From 6129daaa0d2b84c0e376b6b17b3d3740c4d1d1ca Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 9 Apr 2006 13:01:34 +0100 Subject: [ALSA] ca0106: Add analog capture controls. Signed-off-by: James Courtier-Dutton --- sound/pci/ca0106/ca0106.h | 4 +- sound/pci/ca0106/ca0106_main.c | 44 ++++++++++-- sound/pci/ca0106/ca0106_mixer.c | 152 +++++++++++++++++++++++++++++++++++++--- 3 files changed, 186 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index c8131ea92e..9cb66c59f5 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -537,9 +537,9 @@ #endif #define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux +#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) #define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux #define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux -#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) #define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux #define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ @@ -604,6 +604,8 @@ struct snd_ca0106 { u32 spdif_bits[4]; /* s/pdif out setup */ int spdif_enable; int capture_source; + int i2c_capture_source; + u8 i2c_capture_volume[4][2]; int capture_mic_line_in; struct snd_dma_buffer buffer; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index fd8bfebfbd..3762f58384 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -326,6 +326,7 @@ int snd_ca0106_spi_write(struct snd_ca0106 * emu, return 0; } +/* The ADC does not support i2c read, so only write is implemented */ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value) @@ -340,6 +341,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; + // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -348,8 +350,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, for (retry = 0; retry < 10; retry++) { /* Send the data to i2c */ - tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); - tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); + //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + tmp = 0; tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD); snd_ca0106_ptr_write(emu, I2C_A, 0, tmp); @@ -1200,6 +1203,22 @@ static unsigned int spi_dac_init[] = { 0x1400, }; +static unsigned int i2c_adc_init[][2] = { + { 0x17, 0x00 }, /* Reset */ + { 0x07, 0x00 }, /* Timeout */ + { 0x0b, 0x22 }, /* Interface control */ + { 0x0c, 0x22 }, /* Master mode control */ + { 0x0d, 0x08 }, /* Powerdown control */ + { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */ + { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */ + { 0x10, 0x7b }, /* ALC Control 1 */ + { 0x11, 0x00 }, /* ALC Control 2 */ + { 0x12, 0x32 }, /* ALC Control 3 */ + { 0x13, 0x00 }, /* Noise gate control */ + { 0x14, 0xa6 }, /* Limiter control */ + { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */ +}; + static int __devinit snd_ca0106_create(struct snd_card *card, struct pci_dev *pci, struct snd_ca0106 **rchip) @@ -1361,7 +1380,12 @@ static int __devinit snd_ca0106_create(struct snd_card *card, snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */ chip->capture_source = 3; /* Set CAPTURE_SOURCE */ - if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ + if (chip->details->gpio_type == 2) { /* The SB0410 and SB0413 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + outl(0x0, chip->port+GPIO); + //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + outl(0x005f5301, chip->port+GPIO); /* Analog */ + } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ outl(0x0, chip->port+GPIO); //outl(0x00f0e000, chip->port+GPIO); /* Analog */ @@ -1379,7 +1403,19 @@ static int __devinit snd_ca0106_create(struct snd_card *card, outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */ if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ - snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ + int size, n; + + size = ARRAY_SIZE(i2c_adc_init); + //snd_printk("I2C:array size=0x%x\n", size); + for (n=0; n < size; n++) { + snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); + } + for (n=0; n < 4; n++) { + chip->i2c_capture_volume[n][0]= 0xcf; + chip->i2c_capture_volume[n][1]= 0xcf; + } + chip->i2c_capture_source=2; /* Line in */ + //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ } if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */ int size, n; diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 06fe055674..8a5833317b 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -171,6 +171,62 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, return change; } +static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[6] = { + "Phone", "Mic", "Line in", "Aux" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->i2c_capture_source; + return 0; +} + +static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int source_id; + unsigned int ngain, ogain; + int change = 0; + u32 source; + /* If the capture source has changed, + * update the capture volume from the cached value + * for the particular source. + */ + source_id = ucontrol->value.enumerated.item[0] ; + change = (emu->i2c_capture_source != source_id); + if (change) { + snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ + ngain = emu->i2c_capture_volume[source_id][0]; /* Left */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */ + if (ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff)); + ngain = emu->i2c_capture_volume[source_id][1]; /* Left */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */ + if (ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); + source = 1 << source_id; + snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */ + emu->i2c_capture_source = source_id; + } + return change; +} + static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -207,16 +263,16 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, if (change) { emu->capture_mic_line_in = val; if (val) { - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */ + //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ tmp = inl(emu->port+GPIO) & ~0x400; tmp = tmp | 0x400; outl(tmp, emu->port+GPIO); - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); + //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); } else { - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */ + //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ tmp = inl(emu->port+GPIO) & ~0x400; outl(tmp, emu->port+GPIO); - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); + //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); } } return change; @@ -225,7 +281,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic/Line in Capture", + .name = "Shared Mic/Line in Capture Switch", .info = snd_ca0106_capture_mic_line_in_info, .get = snd_ca0106_capture_mic_line_in_get, .put = snd_ca0106_capture_mic_line_in_put @@ -329,15 +385,81 @@ static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol, return 1; } +static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 255; + return 0; +} + +static int snd_ca0106_i2c_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + int source_id; + + source_id = kcontrol->private_value; + + ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0]; + ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1]; + return 0; +} + +static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int ogain; + unsigned int ngain; + int source_id; + int change = 0; + + source_id = kcontrol->private_value; + ogain = emu->i2c_capture_volume[source_id][0]; /* Left */ + ngain = ucontrol->value.integer.value[0]; + if (ngain > 0xff) + return 0; + if (ogain != ngain) { + if (emu->i2c_capture_source == source_id) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) ); + emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0]; + change = 1; + } + ogain = emu->i2c_capture_volume[source_id][1]; /* Right */ + ngain = ucontrol->value.integer.value[1]; + if (ngain > 0xff) + return 0; + if (ogain != ngain) { + if (emu->i2c_capture_source == source_id) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); + emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1]; + change = 1; + } + + return change; +} + #define CA_VOLUME(xname,chid,reg) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_ca0106_volume_info, \ - .get = snd_ca0106_volume_get, \ - .put = snd_ca0106_volume_put, \ + .info = snd_ca0106_volume_info, \ + .get = snd_ca0106_volume_get, \ + .put = snd_ca0106_volume_put, \ .private_value = ((chid) << 8) | (reg) \ } +#define I2C_VOLUME(xname,chid) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_ca0106_i2c_volume_info, \ + .get = snd_ca0106_i2c_volume_get, \ + .put = snd_ca0106_i2c_volume_put, \ + .private_value = chid \ +} + static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { CA_VOLUME("Analog Front Playback Volume", @@ -361,6 +483,11 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { CA_VOLUME("CAPTURE feedback Playback Volume", 1, CAPTURE_CONTROL), + I2C_VOLUME("Phone Capture Volume", 0), + I2C_VOLUME("Mic Capture Volume", 1), + I2C_VOLUME("Line in Capture Volume", 2), + I2C_VOLUME("Aux Capture Volume", 3), + { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -378,11 +505,18 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", + .name = "Digital Capture Source", .info = snd_ca0106_capture_source_info, .get = snd_ca0106_capture_source_get, .put = snd_ca0106_capture_source_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_ca0106_i2c_capture_source_info, + .get = snd_ca0106_i2c_capture_source_get, + .put = snd_ca0106_i2c_capture_source_put + }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), -- cgit v1.2.2 From 21fdddea8e4cc54341d389916d0c17db8c1ca452 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 9 Apr 2006 17:36:39 +0100 Subject: [ALSA] emu10k1: Add support for Audigy4 (not Pro) Signed-off-by: James Courtier-Dutton --- sound/pci/ac97/ac97_codec.c | 4 +-- sound/pci/ac97/ac97_patch.c | 4 +-- sound/pci/emu10k1/emu10k1_main.c | 56 +++++++++++++++++++++++++++++++++------- sound/pci/emu10k1/emumixer.c | 54 ++++++++++++++++++++++++++++++++++++-- sound/pci/emu10k1/tina2.h | 8 ++---- 5 files changed, 105 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index d05200741a..4544f6aa08 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -563,7 +563,7 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = - AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); + AC97_SINGLE("Mic Boost (+20dB) Capture Switch", AC97_MIC, 6, 1, 0); static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; @@ -605,7 +605,7 @@ AC97_SINGLE("Simulated Stereo Enhancement", AC97_GENERAL_PURPOSE, 14, 1, 0), AC97_SINGLE("3D Control - Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), AC97_SINGLE("Loudness (bass boost)", AC97_GENERAL_PURPOSE, 12, 1, 0), AC97_ENUM("Mono Output Select", std_enum[2]), -AC97_ENUM("Mic Select", std_enum[3]), +AC97_ENUM("Mic Select Capture Switch", std_enum[3]), AC97_SINGLE("ADC/DAC Loopback", AC97_GENERAL_PURPOSE, 7, 1, 0) }; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 4d9cf37300..7ae7bc6524 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -563,7 +563,7 @@ AC97_SINGLE("Mic 1 to Phone Switch", AC97_MIC, 14, 1, 1), AC97_SINGLE("Mic 2 to Phone Switch", AC97_MIC, 13, 1, 1), AC97_ENUM("Mic Select Source", wm9711_enum[7]), AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 32, 1), -AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), +AC97_SINGLE("Mic 20dB Boost Capture Switch", AC97_MIC, 7, 1, 0), AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), @@ -653,7 +653,7 @@ AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic 1 to Mono Switch", AC97_LINE, 7, 1, 1), AC97_SINGLE("Mic 2 to Mono Switch", AC97_LINE, 6, 1, 1), -AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), +AC97_SINGLE("Mic Boost (+20dB) Capture Switch", AC97_LINE, 5, 1, 0), AC97_ENUM("Mic to Headphone Mux", wm9713_enum[0]), AC97_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6bfa08436e..e71485c23c 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -777,14 +777,6 @@ static int snd_emu10k1_dev_free(struct snd_device *device) static struct snd_emu_chip_details emu_chip_details[] = { /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ - /* Audigy4 SB0400 */ - {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, - .driver = "Audigy2", .name = "Audigy 4 [SB0400]", - .id = "Audigy2", - .emu10k2_chip = 1, - .ca0108_chip = 1, - .spk71 = 1, - .ac97_chip = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ /* DSP: CA0108-IAT * DAC: CS4382-KQ @@ -799,13 +791,59 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .ac97_chip = 1} , + /* Audigy4 (Not PRO) SB0610 */ + /* Tested by James@superbug.co.uk 4th April 2006 */ + /* A_IOCFG bits + * Output + * 0: ? + * 1: ? + * 2: ? + * 3: 0 - Digital Out, 1 - Line in + * 4: ? + * 5: ? + * 6: ? + * 7: ? + * Input + * 8: ? + * 9: ? + * A: Green jack sense (Front) + * B: ? + * C: Black jack sense (Rear/Side Right) + * D: Yellow jack sense (Center/LFE/Side Left) + * E: ? + * F: ? + * + * Digital Out/Line in switch using A_IOCFG bit 3 (0x08) + * 0 - Digital Out + * 1 - Line in + */ + /* Mic input not tested. + * Analog CD input not tested + * Digital Out not tested. + * Line in working. + * Audio output 5.1 working. Side outputs not working. + */ + /* DSP: CA10300-IAT LF + * DAC: Cirrus Logic CS4382-KQZ + * ADC: Philips 1361T + * AC97: Sigmatel STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, + .driver = "Audigy2", .name = "Audigy 4 [SB0610]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1} , /* Audigy 2 ZS Notebook Cardbus card.*/ /* Tested by James@superbug.co.uk 22th December 2005 */ /* Audio output 7.1/Headphones working. * Digital output working. (AC3 not checked, only PCM) * Audio inputs not tested. */ - /* DSP: Tiny2 + /* DSP: Tina2 * DAC: Wolfson WM8768/WM8568 * ADC: Wolfson WM8775 * AC97: None diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 2a9d12d106..c31f3d0877 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -777,6 +777,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, }; static char *audigy_remove_ctls[] = { /* Master/PCM controls on ac97 of Audigy has no effect */ + /* On the Audigy2 the AC97 playback is piped into + * the Philips ADC for 24bit capture */ "PCM Playback Switch", "PCM Playback Volume", "Master Mono Playback Switch", @@ -804,6 +806,47 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, "AMic Playback Volume", "Mic Playback Volume", NULL }; + static char *audigy_remove_ctls_1361t_adc[] = { + /* On the Audigy2 the AC97 playback is piped into + * the Philips ADC for 24bit capture */ + "PCM Playback Switch", + "PCM Playback Volume", + "Master Mono Playback Switch", + "Master Mono Playback Volume", + "Capture Source", + "Capture Switch", + "Capture Volume", + "Mic Capture Volume", + "Headphone Playback Switch", + "Headphone Playback Volume", + "3D Control - Center", + "3D Control - Depth", + "3D Control - Switch", + "Line2 Playback Volume", + "Line2 Capture Volume", + NULL + }; + static char *audigy_rename_ctls_1361t_adc[] = { + "Master Playback Switch", "Master Capture Switch", + "Master Playback Volume", "Master Capture Volume", + "Wave Master Playback Volume", "Master Playback Volume", + "PC Speaker Playback Switch", "PC Speaker Capture Switch", + "PC Speaker Playback Volume", "PC Speaker Capture Volume", + "Phone Playback Switch", "Phone Capture Switch", + "Phone Playback Volume", "Phone Capture Volume", + "Mic Playback Switch", "Mic Capture Switch", + "Mic Playback Volume", "Mic Capture Volume", + "Line Playback Switch", "Line Capture Switch", + "Line Playback Volume", "Line Capture Volume", + "CD Playback Switch", "CD Capture Switch", + "CD Playback Volume", "CD Capture Volume", + "Aux Playback Switch", "Aux Capture Switch", + "Aux Playback Volume", "Aux Capture Volume", + "Video Playback Switch", "Video Capture Switch", + "Video Playback Volume", "Video Capture Volume", + + NULL + }; if (emu->card_capabilities->ac97_chip) { struct snd_ac97_bus *pbus; @@ -834,7 +877,10 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ac97_write_cache(emu->ac97, AC97_MASTER, 0x0000); /* set capture source to mic */ snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000); - c = audigy_remove_ctls; + if (emu->card_capabilities->adc_1361t) + c = audigy_remove_ctls_1361t_adc; + else + c = audigy_remove_ctls; } else { /* * Credits for cards based on STAC9758: @@ -863,11 +909,15 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, } if (emu->audigy) - c = audigy_rename_ctls; + if (emu->card_capabilities->adc_1361t) + c = audigy_rename_ctls_1361t_adc; + else + c = audigy_rename_ctls; else c = emu10k1_rename_ctls; for (; *c; c += 2) rename_ctl(card, c[0], c[1]); + if (emu->card_capabilities->subsystem == 0x20071102) { /* Audigy 4 Pro */ rename_ctl(card, "Line2 Capture Volume", "Line1/Mic Capture Volume"); rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume"); diff --git a/sound/pci/emu10k1/tina2.h b/sound/pci/emu10k1/tina2.h index 5c43abf03e..f2d8eb6c89 100644 --- a/sound/pci/emu10k1/tina2.h +++ b/sound/pci/emu10k1/tina2.h @@ -1,11 +1,7 @@ /* * Copyright (c) by James Courtier-Dutton - * Driver p16v chips - * Version: 0.21 - * - * - * This code was initally based on code from ALSA's emu10k1x.c which is: - * Copyright (c) by Francisco Moraes + * Driver tina2 chips + * Version: 0.1 * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by -- cgit v1.2.2 From 3969f6178b86613fd443e70d11b8848451552bdd Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 9 Apr 2006 17:44:13 +0100 Subject: [ALSA] Add p17v.h file. Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/p17v.h | 111 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 111 insertions(+) create mode 100644 sound/pci/emu10k1/p17v.h (limited to 'sound') diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h new file mode 100644 index 0000000000..7ddb5be632 --- /dev/null +++ b/sound/pci/emu10k1/p17v.h @@ -0,0 +1,111 @@ +/* + * Copyright (c) by James Courtier-Dutton + * Driver p17v chips + * Version: 0.01 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/******************************************************************************/ +/* Audigy2Value Tina (P17V) pointer-offset register set, + * accessed through the PTR20 and DATA24 registers */ +/******************************************************************************/ + +/* 00 - 07: Not used */ +#define P17V_PLAYBACK_FIFO_PTR 0x08 /* Current playback fifo pointer + * and number of sound samples in cache. + */ +/* 09 - 12: Not used */ +#define P17V_CAPTURE_FIFO_PTR 0x13 /* Current capture fifo pointer + * and number of sound samples in cache. + */ +/* 14 - 17: Not used */ +#define P17V_PB_CHN_SEL 0x18 /* P17v playback channel select */ +#define P17V_SE_SLOT_SEL_L 0x19 /* Sound Engine slot select low */ +#define P17V_SE_SLOT_SEL_H 0x1a /* Sound Engine slot select high */ +/* 1b - 1f: Not used */ +/* 20 - 2f: Not used */ +/* 30 - 3b: Not used */ +#define P17V_SPI 0x3c /* SPI interface register */ +#define P17V_I2C_ADDR 0x3d /* I2C Address */ +#define P17V_I2C_0 0x3e /* I2C Data */ +#define P17V_I2C_1 0x3f /* I2C Data */ + +#define P17V_START_AUDIO 0x40 /* Start Audio bit */ +/* 41 - 47: Reserved */ +#define P17V_START_CAPTURE 0x48 /* Start Capture bit */ +#define P17V_CAPTURE_FIFO_BASE 0x49 /* Record FIFO base address */ +#define P17V_CAPTURE_FIFO_SIZE 0x4a /* Record FIFO buffer size */ +#define P17V_CAPTURE_FIFO_INDEX 0x4b /* Record FIFO capture index */ +#define P17V_CAPTURE_VOL_H 0x4c /* P17v capture volume control */ +#define P17V_CAPTURE_VOL_L 0x4d /* P17v capture volume control */ +/* 4e - 4f: Not used */ +/* 50 - 5f: Not used */ +#define P17V_SRCSel 0x60 /* SRC48 and SRCMulti sample rate select + * and output select + */ +#define P17V_MIXER_AC97_10K1_VOL_L 0x61 /* 10K to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_10K1_VOL_H 0x62 /* 10K to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_P17V_VOL_L 0x63 /* P17V to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_P17V_VOL_H 0x64 /* P17V to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_SRP_REC_VOL_L 0x65 /* SRP Record to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_SRP_REC_VOL_H 0x66 /* SRP Record to Mixer_AC97 input volume control */ +/* 67 - 68: Reserved */ +#define P17V_MIXER_Spdif_10K1_VOL_L 0x69 /* 10K to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_10K1_VOL_H 0x6A /* 10K to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_P17V_VOL_L 0x6B /* P17V to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_P17V_VOL_H 0x6C /* P17V to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_SRP_REC_VOL_L 0x6D /* SRP Record to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_SRP_REC_VOL_H 0x6E /* SRP Record to Mixer_Spdif input volume control */ +/* 6f - 70: Reserved */ +#define P17V_MIXER_I2S_10K1_VOL_L 0x71 /* 10K to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_10K1_VOL_H 0x72 /* 10K to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_P17V_VOL_L 0x73 /* P17V to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_P17V_VOL_H 0x74 /* P17V to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_SRP_REC_VOL_L 0x75 /* SRP Record to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_SRP_REC_VOL_H 0x76 /* SRP Record to Mixer_I2S input volume control */ +/* 77 - 78: Reserved */ +#define P17V_MIXER_AC97_ENABLE 0x79 /* Mixer AC97 input audio enable */ +#define P17V_MIXER_SPDIF_ENABLE 0x7A /* Mixer SPDIF input audio enable */ +#define P17V_MIXER_I2S_ENABLE 0x7B /* Mixer I2S input audio enable */ +#define P17V_AUDIO_OUT_ENABLE 0x7C /* Audio out enable */ +#define P17V_MIXER_ATT 0x7D /* SRP Mixer Attenuation Select */ +#define P17V_SRP_RECORD_SRR 0x7E /* SRP Record channel source Select */ +#define P17V_SOFT_RESET_SRP_MIXER 0x7F /* SRP and mixer soft reset */ + +#define P17V_AC97_OUT_MASTER_VOL_L 0x80 /* AC97 Output master volume control */ +#define P17V_AC97_OUT_MASTER_VOL_H 0x81 /* AC97 Output master volume control */ +#define P17V_SPDIF_OUT_MASTER_VOL_L 0x82 /* SPDIF Output master volume control */ +#define P17V_SPDIF_OUT_MASTER_VOL_H 0x83 /* SPDIF Output master volume control */ +#define P17V_I2S_OUT_MASTER_VOL_L 0x84 /* I2S Output master volume control */ +#define P17V_I2S_OUT_MASTER_VOL_H 0x85 /* I2S Output master volume control */ +/* 86 - 87: Not used */ +#define P17V_I2S_CHANNEL_SWAP_PHASE_INVERSE 0x88 /* I2S out mono channel swap + * and phase inverse */ +#define P17V_SPDIF_CHANNEL_SWAP_PHASE_INVERSE 0x89 /* SPDIF out mono channel swap + * and phase inverse */ +/* 8A: Not used */ +#define P17V_SRP_P17V_ESR 0x8B /* SRP_P17V estimated sample rate and rate lock */ +#define P17V_SRP_REC_ESR 0x8C /* SRP_REC estimated sample rate and rate lock */ +#define P17V_SRP_BYPASS 0x8D /* srps channel bypass and srps bypass */ +/* 8E - 92: Not used */ +#define P17V_I2S_SRC_SEL 0x93 /* I2SIN mode sel */ + + + + + + -- cgit v1.2.2 From be0b7b0113300c324034e94a12244c4ac3f4b354 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 9 Apr 2006 20:48:44 +0100 Subject: [ALSA] ca0106: Fixes MSI K8N's SB Live 24 bit, no sound from line-in. Fixed bug#1331 Signed-off-by: James Courtier-Dutton --- sound/pci/ca0106/ca0106_main.c | 9 +++++++-- sound/pci/ca0106/ca0106_mixer.c | 29 ++++++++++++++++++++++++++++- 2 files changed, 35 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 3762f58384..b605d7045c 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -195,9 +195,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .i2c_adc = 1, .spi_dac = 1 } , /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ + /* SB0438 + * CTRL:CA0106-DAT + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ { .serial = 0x10091462, .name = "MSI K8N Diamond MB [SB0438]", - .gpio_type = 1, + .gpio_type = 2, .i2c_adc = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX @@ -1380,7 +1385,7 @@ static int __devinit snd_ca0106_create(struct snd_card *card, snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */ chip->capture_source = 3; /* Set CAPTURE_SOURCE */ - if (chip->details->gpio_type == 2) { /* The SB0410 and SB0413 use GPIO differently. */ + if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ outl(0x0, chip->port+GPIO); //outl(0x00f0e000, chip->port+GPIO); /* Analog */ diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 8a5833317b..146eed70dc 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -227,6 +227,20 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, return change; } +static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Side out", "Line in" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item > 1) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -287,6 +301,16 @@ static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata = .put = snd_ca0106_capture_mic_line_in_put }; +static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out __devinitdata = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Shared Line in/Side out Capture Switch", + .info = snd_ca0106_capture_line_in_side_out_info, + .get = snd_ca0106_capture_mic_line_in_get, + .put = snd_ca0106_capture_mic_line_in_put +}; + + static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -611,7 +635,10 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) return err; } if (emu->details->i2c_adc == 1) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); + if (emu->details->gpio_type == 1) + err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); + else /* gpio_type == 2 */ + err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu)); if (err < 0) return err; } -- cgit v1.2.2 From d7f6f1157f73dffe0a6afd12b90557e484b7fb35 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Tue, 11 Apr 2006 21:47:27 +0100 Subject: [ALSA] AC97: Correct Mic Boost label. Signed-off-by: James Courtier-Dutton --- sound/pci/ac97/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 4544f6aa08..6c1937ff0d 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -563,7 +563,7 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = - AC97_SINGLE("Mic Boost (+20dB) Capture Switch", AC97_MIC, 6, 1, 0); + AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_MIC, 6, 1, 0); static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 7ae7bc6524..4d9cf37300 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -563,7 +563,7 @@ AC97_SINGLE("Mic 1 to Phone Switch", AC97_MIC, 14, 1, 1), AC97_SINGLE("Mic 2 to Phone Switch", AC97_MIC, 13, 1, 1), AC97_ENUM("Mic Select Source", wm9711_enum[7]), AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 32, 1), -AC97_SINGLE("Mic 20dB Boost Capture Switch", AC97_MIC, 7, 1, 0), +AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), @@ -653,7 +653,7 @@ AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic 1 to Mono Switch", AC97_LINE, 7, 1, 1), AC97_SINGLE("Mic 2 to Mono Switch", AC97_LINE, 6, 1, 1), -AC97_SINGLE("Mic Boost (+20dB) Capture Switch", AC97_LINE, 5, 1, 0), +AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), AC97_ENUM("Mic to Headphone Mux", wm9713_enum[0]), AC97_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), -- cgit v1.2.2 From 79ca4f3f625e14212310f953b096e2e45110ac6b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Apr 2006 12:54:55 +0200 Subject: [ALSA] vxpocket - Fix a typo Fix a typo of return value from vxpocket_config(). Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vxpocket.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7e0cda2b6e..cafe6640cc 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -261,7 +261,7 @@ static int vxpocket_config(struct pcmcia_device *link) link->dev_node = &vxp->node; kfree(parse); - return 9; + return 0; cs_failed: cs_error(link, last_fn, last_ret); -- cgit v1.2.2 From 7152447df98b3981d621224be947a2c8d77aed06 Mon Sep 17 00:00:00 2001 From: Rene Herman Date: Thu, 13 Apr 2006 12:58:06 +0200 Subject: [ALSA] unregister platform device again if probe was unsuccessful This second one unregisters the platform device again when the probe is unsuccesful for sound/drivers, sound/arm/sa11xx-uda1341.c and sound/ppc/powermac.c. This gets them all. Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- sound/arm/sa11xx-uda1341.c | 14 +++++++++----- sound/drivers/dummy.c | 4 ++++ sound/drivers/mpu401/mpu401.c | 4 ++++ sound/drivers/mtpav.c | 14 +++++++++----- sound/drivers/serial-u16550.c | 4 ++++ sound/drivers/virmidi.c | 4 ++++ sound/ppc/powermac.c | 14 +++++++++----- 7 files changed, 43 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 13057d92f0..9211348824 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -984,11 +984,15 @@ static int __init sa11xx_uda1341_init(void) if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) return err; device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (IS_ERR(device)) { - platform_driver_unregister(&sa11xx_uda1341_driver); - return PTR_ERR(device); - } - return 0; + if (!IS_ERR(device)) { + if (platform_get_drvdata(device)) + return 0; + platform_device_unregister(device); + err = -ENODEV + } else + err = PTR_ERR(device); + platform_driver_unregister(&sa11xx_uda1341_driver); + return err; } static void __exit sa11xx_uda1341_exit(void) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index ae0df549fa..ffeafaf2ec 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -677,6 +677,10 @@ static int __init alsa_card_dummy_init(void) i, NULL, 0); if (IS_ERR(device)) continue; + if (!platform_get_drvdata(device)) { + platform_device_unregister(device); + continue; + } devices[i] = device; cards++; } diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 77b0600973..d3cbbb0475 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -253,6 +253,10 @@ static int __init alsa_card_mpu401_init(void) i, NULL, 0); if (IS_ERR(device)) continue; + if (!platform_get_drvdata(device)) { + platform_device_unregister(device); + continue; + } platform_devices[i] = device; snd_mpu401_devices++; } diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index b7a0b42813..474eed06e7 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -770,11 +770,15 @@ static int __init alsa_card_mtpav_init(void) return err; device = platform_device_register_simple(SND_MTPAV_DRIVER, -1, NULL, 0); - if (IS_ERR(device)) { - platform_driver_unregister(&snd_mtpav_driver); - return PTR_ERR(device); - } - return 0; + if (!IS_ERR(device)) { + if (platform_get_drvdata(device)) + return 0; + platform_device_unregister(device); + err = -ENODEV; + } else + err = PTR_ERR(device); + platform_driver_unregister(&snd_mtpav_driver); + return err; } static void __exit alsa_card_mtpav_exit(void) diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index c01b4c5118..2330fec505 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -998,6 +998,10 @@ static int __init alsa_card_serial_init(void) i, NULL, 0); if (IS_ERR(device)) continue; + if (!platform_get_drvdata(device)) { + platform_device_unregister(device); + continue; + } devices[i] = device; cards++; } diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 26eb2499d4..59171f8200 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -171,6 +171,10 @@ static int __init alsa_card_virmidi_init(void) i, NULL, 0); if (IS_ERR(device)) continue; + if (!platform_get_drvdata(device)) { + platform_device_unregister(device); + continue; + } devices[i] = device; cards++; } diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index f4902a219e..875f1f7bdc 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -188,11 +188,15 @@ static int __init alsa_card_pmac_init(void) if ((err = platform_driver_register(&snd_pmac_driver)) < 0) return err; device = platform_device_register_simple(SND_PMAC_DRIVER, -1, NULL, 0); - if (IS_ERR(device)) { - platform_driver_unregister(&snd_pmac_driver); - return PTR_ERR(device); - } - return 0; + if (!IS_ERR(device)) { + if (platform_get_drvdata(device)) + return 0; + platform_device_unregister(device); + err = -ENODEV; + } else + err = PTR_ERR(device); + platform_driver_unregister(&snd_pmac_driver); + return err; } -- cgit v1.2.2 From 01686c5fce4682350849f9f2c262fcaf67ec73c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Apr 2006 12:54:11 +0200 Subject: [ALSA] hda-codec - Add Thinkpad X60/T60/Z60 support Added the support for Thinkpad X60/T60/Z60 laptops with AD1981HD codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 44 +++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 43 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 40f000ba13..8ddae0a25e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1329,13 +1329,50 @@ static int ad1981_hp_init(struct hda_codec *codec) return 0; } +/* configuration for Lenovo Thinkpad T60 */ +static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1981_thinkpad_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Mix", 0x2 }, + { "CD", 0x4 }, + }, +}; + /* models */ -enum { AD1981_BASIC, AD1981_HP }; +enum { AD1981_BASIC, AD1981_HP, AD1981_THINKPAD }; static struct hda_board_config ad1981_cfg_tbl[] = { { .modelname = "hp", .config = AD1981_HP }, /* All HP models */ { .pci_subvendor = 0x103c, .config = AD1981_HP }, + { .modelname = "thinkpad", .config = AD1981_THINKPAD }, + /* Lenovo Thinkpad T60/X60/Z6xx */ + { .pci_subvendor = 0x17aa, .config = AD1981_THINKPAD }, + { .pci_subvendor = 0x1014, .pci_subsystem = 0x0597, + .config = AD1981_THINKPAD }, /* Z60m/t */ { .modelname = "basic", .config = AD1981_BASIC }, {} }; @@ -1381,6 +1418,11 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops.init = ad1981_hp_init; codec->patch_ops.unsol_event = ad1981_hp_unsol_event; break; + case AD1981_THINKPAD: + spec->mixers[0] = ad1981_thinkpad_mixers; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1981_thinkpad_capture_source; + break; } return 0; -- cgit v1.2.2 From 887709be9063d233eb5abef25aafcd94615b03f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Apr 2006 13:27:31 +0200 Subject: [ALSA] hda-codec - Fix a typo Fixed a typo of 'pci_subsystem' in the last changeset. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8ddae0a25e..3a9b800db8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1371,7 +1371,7 @@ static struct hda_board_config ad1981_cfg_tbl[] = { { .modelname = "thinkpad", .config = AD1981_THINKPAD }, /* Lenovo Thinkpad T60/X60/Z6xx */ { .pci_subvendor = 0x17aa, .config = AD1981_THINKPAD }, - { .pci_subvendor = 0x1014, .pci_subsystem = 0x0597, + { .pci_subvendor = 0x1014, .pci_subdevice = 0x0597, .config = AD1981_THINKPAD }, /* Z60m/t */ { .modelname = "basic", .config = AD1981_BASIC }, {} -- cgit v1.2.2 From 78fc030bdbbeebdea436f2b02a616d67e5f9bd9b Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Fri, 21 Apr 2006 08:39:20 +0200 Subject: [ALSA] Change seq_midi.c so client name is card, rather than port, specific Change snd_seq_midisynth_register_port() in seq_midi.c so that if a new client is created, the client name string is based on card->shortname not (port-specific) info->name. Signed-off-by: Alan Horstmann Signed-off-by: Clemens Ladisch --- sound/core/seq/seq_midi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 9caa1372be..3b316da25e 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -320,8 +320,8 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) } client->seq_client = snd_seq_create_kernel_client( - card, 0, "%s", info->name[0] ? - (const char *)info->name : "External MIDI"); + card, 0, "%s", card->shortname[0] ? + (const char *)card->shortname : "External MIDI"); if (client->seq_client < 0) { kfree(client); mutex_unlock(®ister_mutex); -- cgit v1.2.2 From 3bef229e4f13790402b1387ea8147906f217a766 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Wed, 26 Apr 2006 18:13:59 +0200 Subject: [ALSA] ice1712 - Provides specified midi port names instead of defaults Patch provides for the ice1712 card driver to overwrite the midi port name string given by default in mpu401_uart, with one specified in snd_ice1712_card_info. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ews.c | 2 ++ sound/pci/ice1712/ice1712.c | 17 +++++++++++++++-- sound/pci/ice1712/ice1712.h | 2 ++ 3 files changed, 19 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 2c529e7413..2e1cf11205 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -1031,6 +1031,8 @@ struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = { .model = "dmx6fire", .chip_init = snd_ice1712_ews_init, .build_controls = snd_ice1712_ews_add_controls, + .mpu401_1_name = "MIDI-Front DMX6fire", + .mpu401_2_name = "Wavetable DMX6fire", }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index c56793b381..2821014b26 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2743,8 +2743,14 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, snd_card_free(card); return err; } - - if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) + if (c->mpu401_1_name) + /* Prefered name available in card_info */ + snprintf(ice->rmidi[0]->name, + sizeof(ice->rmidi[0]->name), + "%s %d", c->mpu401_1_name, card->number); + + if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) { + /* 2nd port used */ if ((err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, ICEREG(ice, MPU2_CTRL), 1, ice->irq, 0, @@ -2752,6 +2758,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, snd_card_free(card); return err; } + if (c->mpu401_2_name) + /* Prefered name available in card_info */ + snprintf(ice->rmidi[1]->name, + sizeof(ice->rmidi[1]->name), + "%s %d", c->mpu401_2_name, + card->number); + } } snd_ice1712_set_input_clock_source(ice, 0); diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 053f8e56fd..d4776319a0 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -495,6 +495,8 @@ struct snd_ice1712_card_info { int (*chip_init)(struct snd_ice1712 *); int (*build_controls)(struct snd_ice1712 *); unsigned int no_mpu401: 1; + const char *mpu401_1_name; + const char *mpu401_2_name; unsigned int eeprom_size; unsigned char *eeprom_data; }; -- cgit v1.2.2 From 5e1b1518a53fc62d9f39a13819c849336c6d8dd4 Mon Sep 17 00:00:00 2001 From: Kenneth Crudup Date: Fri, 28 Apr 2006 13:03:48 +0200 Subject: [ALSA] hda-codec - Add support for Sony Vaio VGN-A790 laptop Added the model entry for Sony Vaio VGN-A790 laptop with ALC260 codec. From: Kenneth Crudup Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f0e9a9c907..cf6c100940 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3822,6 +3822,8 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "basic", .config = ALC260_BASIC }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb, .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81cd, + .config = ALC260_BASIC }, /* Sony VAIO */ { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, -- cgit v1.2.2 From 9ac25594e68a4b61516e7c1140d8c0f7ef449e20 Mon Sep 17 00:00:00 2001 From: Jaya Kumar Date: Fri, 28 Apr 2006 14:34:49 +0200 Subject: [ALSA] PM support for cs5535audio Appended is my patch adding PM support to the cs5535audio driver. I also added the ac97 quirk but it's not yet confirmed which boards need to be in the quirk list. The patch also includes some Kconfig and misc cleanup. Signed-off-by: Jaya Kumar Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 9 ++- sound/pci/cs5535audio/Makefile | 4 ++ sound/pci/cs5535audio/cs5535audio.c | 31 ++++++-- sound/pci/cs5535audio/cs5535audio.h | 8 +++ sound/pci/cs5535audio/cs5535audio_pcm.c | 24 ++++++- sound/pci/cs5535audio/cs5535audio_pm.c | 123 ++++++++++++++++++++++++++++++++ 6 files changed, 191 insertions(+), 8 deletions(-) create mode 100644 sound/pci/cs5535audio/cs5535audio_pm.c (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index a2081803a8..d37346b12d 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -216,14 +216,19 @@ config SND_CS46XX_NEW_DSP This works better than the old code, so say Y. config SND_CS5535AUDIO - tristate "CS5535 Audio" + tristate "CS5535/CS5536 Audio" depends on SND && X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help Say Y here to include support for audio on CS5535 chips. It is referred to as NS CS5535 IO or AMD CS5535 IO companion in - various literature. + various literature. This driver also supports the CS5536 audio + device. However, for both chips, on certain boards, you may + need to use ac97_quirk=hp_only if your board has physically + mapped headphone out to master output. If that works for you, + send lspci -vvv output to the mailing list so that your board + can be identified in the quirks list. To compile this driver as a module, choose M here: the module will be called snd-cs5535audio. diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index 08d8ee6547..2911a8adc1 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -4,5 +4,9 @@ snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o +ifdef CONFIG_PM +snd-cs5535audio-objs += cs5535audio_pm.o +endif + # Toplevel Module Dependency obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 2c1213a35d..41f02f05df 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -1,5 +1,5 @@ /* - * Driver for audio on multifunction CS5535 companion device + * Driver for audio on multifunction CS5535/6 companion device * Copyright (C) Jaya Kumar * * Based on Jaroslav Kysela and Takashi Iwai's examples. @@ -40,16 +40,29 @@ #define DRIVER_NAME "cs5535audio" +static char *ac97_quirk; +module_param(ac97_quirk, charp, 0444); +MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds."); + +static struct ac97_quirk ac97_quirks[] __devinitdata = { +#if 0 /* Not yet confirmed if all 5536 boards are HP only */ + { + .subvendor = PCI_VENDOR_ID_AMD, + .subdevice = PCI_DEVICE_ID_AMD_CS5536_AUDIO, + .name = "AMD RDK", + .type = AC97_TUNE_HP_ONLY + }, +#endif + {} +}; static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = { - { PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, + { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} }; @@ -148,6 +161,8 @@ static int snd_cs5535audio_mixer(struct cs5535audio *cs5535au) return err; } + snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk); + return 0; } @@ -347,6 +362,8 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci, if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0) goto probefail_out; + card->private_data = cs5535au; + if ((err = snd_cs5535audio_mixer(cs5535au)) < 0) goto probefail_out; @@ -383,6 +400,10 @@ static struct pci_driver driver = { .id_table = snd_cs5535audio_ids, .probe = snd_cs5535audio_probe, .remove = __devexit_p(snd_cs5535audio_remove), +#ifdef CONFIG_PM + .suspend = snd_cs5535audio_suspend, + .resume = snd_cs5535audio_resume, +#endif }; static int __init alsa_card_cs5535audio_init(void) diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 5e55a1a1ed..4fd1f31a6c 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -74,6 +74,8 @@ #define PRM_RDY_STS 0x00800000 #define ACC_CODEC_CNTL_WR_CMD (~0x80000000) #define ACC_CODEC_CNTL_RD_CMD 0x80000000 +#define ACC_CODEC_CNTL_LNK_SHUTDOWN 0x00040000 +#define ACC_CODEC_CNTL_LNK_WRM_RST 0x00020000 #define PRD_JMP 0x2000 #define PRD_EOP 0x4000 #define PRD_EOT 0x8000 @@ -88,6 +90,7 @@ struct cs5535audio_dma_ops { void (*disable_dma)(struct cs5535audio *cs5535au); void (*pause_dma)(struct cs5535audio *cs5535au); void (*setup_prd)(struct cs5535audio *cs5535au, u32 prd_addr); + u32 (*read_prd)(struct cs5535audio *cs5535au); u32 (*read_dma_pntr)(struct cs5535audio *cs5535au); }; @@ -103,11 +106,14 @@ struct cs5535audio_dma { struct snd_pcm_substream *substream; unsigned int buf_addr, buf_bytes; unsigned int period_bytes, periods; + int suspended; + u32 saved_prd; }; struct cs5535audio { struct snd_card *card; struct snd_ac97 *ac97; + struct snd_pcm *pcm; int irq; struct pci_dev *pci; unsigned long port; @@ -117,6 +123,8 @@ struct cs5535audio { struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS]; }; +int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state); +int snd_cs5535audio_resume(struct pci_dev *pci); int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio); #endif /* __SOUND_CS5535AUDIO_H */ diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 60bb82b2ff..f0a48693d6 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -43,7 +43,8 @@ static struct snd_pcm_hardware snd_cs5535audio_playback = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME ), .formats = ( SNDRV_PCM_FMTBIT_S16_LE @@ -193,6 +194,11 @@ static void cs5535audio_playback_setup_prd(struct cs5535audio *cs5535au, cs_writel(cs5535au, ACC_BM0_PRD, prd_addr); } +static u32 cs5535audio_playback_read_prd(struct cs5535audio *cs5535au) +{ + return cs_readl(cs5535au, ACC_BM0_PRD); +} + static u32 cs5535audio_playback_read_dma_pntr(struct cs5535audio *cs5535au) { return cs_readl(cs5535au, ACC_BM0_PNTR); @@ -219,6 +225,11 @@ static void cs5535audio_capture_setup_prd(struct cs5535audio *cs5535au, cs_writel(cs5535au, ACC_BM1_PRD, prd_addr); } +static u32 cs5535audio_capture_read_prd(struct cs5535audio *cs5535au) +{ + return cs_readl(cs5535au, ACC_BM1_PRD); +} + static u32 cs5535audio_capture_read_dma_pntr(struct cs5535audio *cs5535au) { return cs_readl(cs5535au, ACC_BM1_PNTR); @@ -285,9 +296,17 @@ static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: dma->ops->enable_dma(cs5535au); break; + case SNDRV_PCM_TRIGGER_RESUME: + dma->ops->enable_dma(cs5535au); + dma->suspended = 0; + break; case SNDRV_PCM_TRIGGER_STOP: dma->ops->disable_dma(cs5535au); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + dma->ops->disable_dma(cs5535au); + dma->suspended = 1; + break; default: snd_printk(KERN_ERR "unhandled trigger\n"); err = -EINVAL; @@ -375,6 +394,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = { .enable_dma = cs5535audio_playback_enable_dma, .disable_dma = cs5535audio_playback_disable_dma, .setup_prd = cs5535audio_playback_setup_prd, + .read_prd = cs5535audio_playback_read_prd, .pause_dma = cs5535audio_playback_pause_dma, .read_dma_pntr = cs5535audio_playback_read_dma_pntr, }; @@ -384,6 +404,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_capture_dma_ops = { .enable_dma = cs5535audio_capture_enable_dma, .disable_dma = cs5535audio_capture_disable_dma, .setup_prd = cs5535audio_capture_setup_prd, + .read_prd = cs5535audio_capture_read_prd, .pause_dma = cs5535audio_capture_pause_dma, .read_dma_pntr = cs5535audio_capture_read_dma_pntr, }; @@ -413,6 +434,7 @@ int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535au) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(cs5535au->pci), 64*1024, 128*1024); + cs5535au->pcm = pcm; return 0; } diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c new file mode 100644 index 0000000000..aad0e69db9 --- /dev/null +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -0,0 +1,123 @@ +/* + * Power management for audio on multifunction CS5535 companion device + * Copyright (C) Jaya Kumar + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "cs5535audio.h" + +static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au) +{ + /* + we depend on snd_ac97_suspend to tell the + AC97 codec to shutdown. the amd spec suggests + that the LNK_SHUTDOWN be done at the same time + that the codec power-down is issued. instead, + we do it just after rather than at the same + time. excluding codec specific build_ops->suspend + ac97 powerdown hits: + 0x8000 EAPD + 0x4000 Headphone amplifier + 0x0300 ADC & DAC + 0x0400 Analog Mixer powerdown (Vref on) + I am not sure if this is the best that we can do. + The remainder to be investigated are: + - analog mixer (vref off) 0x0800 + - AC-link powerdown 0x1000 + - codec internal clock 0x2000 + */ + + /* set LNK_SHUTDOWN to shutdown AC link */ + cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_SHUTDOWN); + +} + +int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct cs5535audio *cs5535au = card->private_data; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) { + struct cs5535audio_dma *dma = &cs5535au->dmas[i]; + if (dma && dma->substream && !dma->suspended) + dma->saved_prd = dma->ops->read_prd(cs5535au); + } + snd_pcm_suspend_all(cs5535au->pcm); + snd_ac97_suspend(cs5535au->ac97); + /* save important regs, then disable aclink in hw */ + snd_cs5535audio_stop_hardware(cs5535au); + pci_disable_device(pci); + pci_save_state(pci); + + return 0; +} + +int snd_cs5535audio_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct cs5535audio *cs5535au = card->private_data; + u32 tmp; + int timeout; + int i; + + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + + /* set LNK_WRM_RST to reset AC link */ + cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_WRM_RST); + + timeout = 50; + do { + tmp = cs_readl(cs5535au, ACC_CODEC_STATUS); + if (tmp & PRM_RDY_STS) + break; + udelay(1); + } while (--timeout); + + if (!timeout) + snd_printk(KERN_ERR "Failure getting AC Link ready\n"); + + /* we depend on ac97 to perform the codec power up */ + snd_ac97_resume(cs5535au->ac97); + /* set up rate regs, dma. actual initiation is done in trig */ + for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) { + struct cs5535audio_dma *dma = &cs5535au->dmas[i]; + if (dma && dma->substream && dma->suspended) { + dma->substream->ops->prepare(dma->substream); + dma->ops->setup_prd(cs5535au, dma->saved_prd); + } + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} + -- cgit v1.2.2 From c0d3fb39e9511c6fad17d059a3a50d1be33add24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:39 +0200 Subject: [ALSA] Clean up EXPORT_SYMBOL()s in snd module Move EXPORT_SYMBOL()s to places adjacent to functions/variables. Signed-off-by: Takashi Iwai --- sound/core/control.c | 31 ++++++++++++++++ sound/core/device.c | 6 +++ sound/core/info.c | 20 ++++++++++ sound/core/info_oss.c | 2 + sound/core/init.c | 20 ++++++++++ sound/core/isadma.c | 6 +++ sound/core/memory.c | 5 +++ sound/core/misc.c | 6 +++ sound/core/sound.c | 99 ++++++-------------------------------------------- sound/core/sound_oss.c | 6 +++ 10 files changed, 113 insertions(+), 88 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 22565c9b96..bb397eaa71 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -176,6 +176,8 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, read_unlock(&card->ctl_files_rwlock); } +EXPORT_SYMBOL(snd_ctl_notify); + /** * snd_ctl_new - create a control instance from the template * @control: the control template @@ -204,6 +206,8 @@ struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, unsigned int acce return kctl; } +EXPORT_SYMBOL(snd_ctl_new); + /** * snd_ctl_new1 - create a control instance from the template * @ncontrol: the initialization record @@ -242,6 +246,8 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, return snd_ctl_new(&kctl, access); } +EXPORT_SYMBOL(snd_ctl_new1); + /** * snd_ctl_free_one - release the control instance * @kcontrol: the control instance @@ -259,6 +265,8 @@ void snd_ctl_free_one(struct snd_kcontrol *kcontrol) } } +EXPORT_SYMBOL(snd_ctl_free_one); + static unsigned int snd_ctl_hole_check(struct snd_card *card, unsigned int count) { @@ -347,6 +355,8 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) return err; } +EXPORT_SYMBOL(snd_ctl_add); + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -373,6 +383,8 @@ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) return 0; } +EXPORT_SYMBOL(snd_ctl_remove); + /** * snd_ctl_remove_id - remove the control of the given id and release it * @card: the card instance @@ -399,6 +411,8 @@ int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id) return ret; } +EXPORT_SYMBOL(snd_ctl_remove_id); + /** * snd_ctl_remove_unlocked_id - remove the unlocked control of the given id and release it * @file: active control handle @@ -461,6 +475,8 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, return 0; } +EXPORT_SYMBOL(snd_ctl_rename_id); + /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -487,6 +503,8 @@ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numi return NULL; } +EXPORT_SYMBOL(snd_ctl_find_numid); + /** * snd_ctl_find_id - find the control instance with the given id * @card: the card instance @@ -527,6 +545,8 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; } +EXPORT_SYMBOL(snd_ctl_find_id); + static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, unsigned int cmd, void __user *arg) { @@ -704,6 +724,8 @@ int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control) return result; } +EXPORT_SYMBOL(snd_ctl_elem_read); + static int snd_ctl_elem_read_user(struct snd_card *card, struct snd_ctl_elem_value __user *_control) { @@ -767,6 +789,8 @@ int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, return result; } +EXPORT_SYMBOL(snd_ctl_elem_write); + static int snd_ctl_elem_write_user(struct snd_ctl_file *file, struct snd_ctl_elem_value __user *_control) { @@ -1199,11 +1223,15 @@ int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn) return _snd_ctl_register_ioctl(fcn, &snd_control_ioctls); } +EXPORT_SYMBOL(snd_ctl_register_ioctl); + #ifdef CONFIG_COMPAT int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn) { return _snd_ctl_register_ioctl(fcn, &snd_control_compat_ioctls); } + +EXPORT_SYMBOL(snd_ctl_register_ioctl_compat); #endif /* @@ -1236,12 +1264,15 @@ int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn) return _snd_ctl_unregister_ioctl(fcn, &snd_control_ioctls); } +EXPORT_SYMBOL(snd_ctl_unregister_ioctl); + #ifdef CONFIG_COMPAT int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn) { return _snd_ctl_unregister_ioctl(fcn, &snd_control_compat_ioctls); } +EXPORT_SYMBOL(snd_ctl_unregister_ioctl_compat); #endif static int snd_ctl_fasync(int fd, struct file * file, int on) diff --git a/sound/core/device.c b/sound/core/device.c index b1cf6ec567..6ce4da4a10 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -63,6 +63,8 @@ int snd_device_new(struct snd_card *card, snd_device_type_t type, return 0; } +EXPORT_SYMBOL(snd_device_new); + /** * snd_device_free - release the device from the card * @card: the card instance @@ -107,6 +109,8 @@ int snd_device_free(struct snd_card *card, void *device_data) return -ENXIO; } +EXPORT_SYMBOL(snd_device_free); + /** * snd_device_disconnect - disconnect the device * @card: the card instance @@ -182,6 +186,8 @@ int snd_device_register(struct snd_card *card, void *device_data) return -ENXIO; } +EXPORT_SYMBOL(snd_device_register); + /* * register all the devices on the card. * called from init.c diff --git a/sound/core/info.c b/sound/core/info.c index 2582b74d31..9c288539e9 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -111,12 +111,16 @@ int snd_iprintf(struct snd_info_buffer *buffer, char *fmt,...) return res; } +EXPORT_SYMBOL(snd_iprintf); + /* */ static struct proc_dir_entry *snd_proc_root = NULL; struct snd_info_entry *snd_seq_root = NULL; +EXPORT_SYMBOL(snd_seq_root); + #ifdef CONFIG_SND_OSSEMUL struct snd_info_entry *snd_oss_root = NULL; #endif @@ -687,6 +691,8 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) return 0; } +EXPORT_SYMBOL(snd_info_get_line); + /** * snd_info_get_str - parse a string token * @dest: the buffer to store the string token @@ -723,6 +729,8 @@ char *snd_info_get_str(char *dest, char *src, int len) return src; } +EXPORT_SYMBOL(snd_info_get_str); + /** * snd_info_create_entry - create an info entry * @name: the proc file name @@ -774,6 +782,8 @@ struct snd_info_entry *snd_info_create_module_entry(struct module * module, return entry; } +EXPORT_SYMBOL(snd_info_create_module_entry); + /** * snd_info_create_card_entry - create an info entry for the given card * @card: the card instance @@ -797,6 +807,8 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, return entry; } +EXPORT_SYMBOL(snd_info_create_card_entry); + static int snd_info_dev_free_entry(struct snd_device *device) { struct snd_info_entry *entry = device->device_data; @@ -867,6 +879,8 @@ int snd_card_proc_new(struct snd_card *card, const char *name, return 0; } +EXPORT_SYMBOL(snd_card_proc_new); + /** * snd_info_free_entry - release the info entry * @entry: the info entry @@ -883,6 +897,8 @@ void snd_info_free_entry(struct snd_info_entry * entry) kfree(entry); } +EXPORT_SYMBOL(snd_info_free_entry); + /** * snd_info_register - register the info entry * @entry: the info entry @@ -913,6 +929,8 @@ int snd_info_register(struct snd_info_entry * entry) return 0; } +EXPORT_SYMBOL(snd_info_register); + /** * snd_info_unregister - de-register the info entry * @entry: the info entry @@ -937,6 +955,8 @@ int snd_info_unregister(struct snd_info_entry * entry) return 0; } +EXPORT_SYMBOL(snd_info_unregister); + /* */ diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index f9ce854b3d..f2efca1872 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -64,6 +64,8 @@ int snd_oss_info_register(int dev, int num, char *string) return 0; } +EXPORT_SYMBOL(snd_oss_info_register); + extern void snd_card_info_read_oss(struct snd_info_buffer *buffer); static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int dev) diff --git a/sound/core/init.c b/sound/core/init.c index 39ed2e5bb0..b145d17ba3 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -40,10 +40,13 @@ struct snd_shutdown_f_ops { unsigned int snd_cards_lock = 0; /* locked for registering/using */ struct snd_card *snd_cards[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = NULL}; +EXPORT_SYMBOL(snd_cards); + DEFINE_RWLOCK(snd_card_rwlock); #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); +EXPORT_SYMBOL(snd_mixer_oss_notify_callback); #endif #ifdef CONFIG_PROC_FS @@ -169,6 +172,8 @@ struct snd_card *snd_card_new(int idx, const char *xid, return NULL; } +EXPORT_SYMBOL(snd_card_new); + static loff_t snd_disconnect_llseek(struct file *file, loff_t offset, int orig) { return -ENODEV; @@ -298,6 +303,8 @@ int snd_card_disconnect(struct snd_card *card) return 0; } +EXPORT_SYMBOL(snd_card_disconnect); + /** * snd_card_free - frees given soundcard structure * @card: soundcard structure @@ -360,6 +367,8 @@ int snd_card_free(struct snd_card *card) return 0; } +EXPORT_SYMBOL(snd_card_free); + static void snd_card_free_thread(void * __card) { struct snd_card *card = __card; @@ -405,6 +414,8 @@ int snd_card_free_in_thread(struct snd_card *card) return -EFAULT; } +EXPORT_SYMBOL(snd_card_free_in_thread); + static void choose_default_id(struct snd_card *card) { int i, len, idx_flag = 0, loops = SNDRV_CARDS; @@ -505,6 +516,8 @@ int snd_card_register(struct snd_card *card) return 0; } +EXPORT_SYMBOL(snd_card_register); + #ifdef CONFIG_PROC_FS static struct snd_info_entry *snd_card_info_entry = NULL; @@ -644,6 +657,8 @@ int snd_component_add(struct snd_card *card, const char *component) return 0; } +EXPORT_SYMBOL(snd_component_add); + /** * snd_card_file_add - add the file to the file list of the card * @card: soundcard structure @@ -676,6 +691,8 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return 0; } +EXPORT_SYMBOL(snd_card_file_add); + /** * snd_card_file_remove - remove the file from the file list * @card: soundcard structure @@ -717,6 +734,8 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) return 0; } +EXPORT_SYMBOL(snd_card_file_remove); + #ifdef CONFIG_PM /** * snd_power_wait - wait until the power-state is changed. @@ -753,4 +772,5 @@ int snd_power_wait(struct snd_card *card, unsigned int power_state) return result; } +EXPORT_SYMBOL(snd_power_wait); #endif /* CONFIG_PM */ diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 1a378951da..d52398727f 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -56,6 +56,8 @@ void snd_dma_program(unsigned long dma, release_dma_lock(flags); } +EXPORT_SYMBOL(snd_dma_program); + /** * snd_dma_disable - stop the ISA DMA transfer * @dma: the dma number @@ -72,6 +74,8 @@ void snd_dma_disable(unsigned long dma) release_dma_lock(flags); } +EXPORT_SYMBOL(snd_dma_disable); + /** * snd_dma_pointer - return the current pointer to DMA transfer buffer in bytes * @dma: the dma number @@ -101,3 +105,5 @@ unsigned int snd_dma_pointer(unsigned long dma, unsigned int size) else return size - result; } + +EXPORT_SYMBOL(snd_dma_pointer); diff --git a/sound/core/memory.c b/sound/core/memory.c index 862d62d2e1..fe59850be8 100644 --- a/sound/core/memory.c +++ b/sound/core/memory.c @@ -21,6 +21,7 @@ */ #include +#include #include #include @@ -55,6 +56,8 @@ int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size #endif } +EXPORT_SYMBOL(copy_to_user_fromio); + /** * copy_from_user_toio - copy data from user-space to mmio-space * @dst: the destination pointer on mmio-space @@ -85,3 +88,5 @@ int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size return 0; #endif } + +EXPORT_SYMBOL(copy_from_user_toio); diff --git a/sound/core/misc.c b/sound/core/misc.c index b53e563c09..03fc711f41 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -34,6 +34,8 @@ void release_and_free_resource(struct resource *res) } } +EXPORT_SYMBOL(release_and_free_resource); + #ifdef CONFIG_SND_VERBOSE_PRINTK void snd_verbose_printk(const char *file, int line, const char *format, ...) { @@ -51,6 +53,8 @@ void snd_verbose_printk(const char *file, int line, const char *format, ...) vprintk(format, args); va_end(args); } + +EXPORT_SYMBOL(snd_verbose_printk); #endif #if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK) @@ -71,4 +75,6 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) va_end(args); } + +EXPORT_SYMBOL(snd_verbose_printd); #endif diff --git a/sound/core/sound.c b/sound/core/sound.c index 108e430b50..67cfa06062 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -39,6 +39,8 @@ static int major = CONFIG_SND_MAJOR; int snd_major; +EXPORT_SYMBOL(snd_major); + static int cards_limit = 1; static int device_mode = S_IFCHR | S_IRUGO | S_IWUGO; @@ -60,6 +62,7 @@ MODULE_ALIAS_CHARDEV_MAJOR(CONFIG_SND_MAJOR); * modules are loaded manually, this limit number increases, too. */ int snd_ecards_limit; +EXPORT_SYMBOL(snd_ecards_limit); static struct snd_minor *snd_minors[SNDRV_OS_MINORS]; static DEFINE_MUTEX(sound_mutex); @@ -92,6 +95,8 @@ void snd_request_card(int card) request_module("snd-card-%i", card); } +EXPORT_SYMBOL(snd_request_card); + static void snd_request_other(int minor) { char *str; @@ -133,6 +138,8 @@ void *snd_lookup_minor_data(unsigned int minor, int type) return private_data; } +EXPORT_SYMBOL(snd_lookup_minor_data); + static int snd_open(struct inode *inode, struct file *file) { unsigned int minor = iminor(inode); @@ -281,6 +288,8 @@ int snd_register_device(int type, struct snd_card *card, int dev, return 0; } +EXPORT_SYMBOL(snd_register_device); + /** * snd_unregister_device - unregister the device on the given card * @type: the device type, SNDRV_DEVICE_TYPE_XXX @@ -321,6 +330,8 @@ int snd_unregister_device(int type, struct snd_card *card, int dev) return 0; } +EXPORT_SYMBOL(snd_unregister_device); + #ifdef CONFIG_PROC_FS /* * INFO PART @@ -446,91 +457,3 @@ static void __exit alsa_sound_exit(void) module_init(alsa_sound_init) module_exit(alsa_sound_exit) - - /* sound.c */ -EXPORT_SYMBOL(snd_major); -EXPORT_SYMBOL(snd_ecards_limit); -#if defined(CONFIG_KMOD) -EXPORT_SYMBOL(snd_request_card); -#endif -EXPORT_SYMBOL(snd_register_device); -EXPORT_SYMBOL(snd_unregister_device); -EXPORT_SYMBOL(snd_lookup_minor_data); -#if defined(CONFIG_SND_OSSEMUL) -EXPORT_SYMBOL(snd_register_oss_device); -EXPORT_SYMBOL(snd_unregister_oss_device); -EXPORT_SYMBOL(snd_lookup_oss_minor_data); -#endif - /* memory.c */ -EXPORT_SYMBOL(copy_to_user_fromio); -EXPORT_SYMBOL(copy_from_user_toio); - /* init.c */ -EXPORT_SYMBOL(snd_cards); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) -EXPORT_SYMBOL(snd_mixer_oss_notify_callback); -#endif -EXPORT_SYMBOL(snd_card_new); -EXPORT_SYMBOL(snd_card_disconnect); -EXPORT_SYMBOL(snd_card_free); -EXPORT_SYMBOL(snd_card_free_in_thread); -EXPORT_SYMBOL(snd_card_register); -EXPORT_SYMBOL(snd_component_add); -EXPORT_SYMBOL(snd_card_file_add); -EXPORT_SYMBOL(snd_card_file_remove); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_power_wait); -#endif - /* device.c */ -EXPORT_SYMBOL(snd_device_new); -EXPORT_SYMBOL(snd_device_register); -EXPORT_SYMBOL(snd_device_free); - /* isadma.c */ -#ifdef CONFIG_ISA_DMA_API -EXPORT_SYMBOL(snd_dma_program); -EXPORT_SYMBOL(snd_dma_disable); -EXPORT_SYMBOL(snd_dma_pointer); -#endif - /* info.c */ -#ifdef CONFIG_PROC_FS -EXPORT_SYMBOL(snd_seq_root); -EXPORT_SYMBOL(snd_iprintf); -EXPORT_SYMBOL(snd_info_get_line); -EXPORT_SYMBOL(snd_info_get_str); -EXPORT_SYMBOL(snd_info_create_module_entry); -EXPORT_SYMBOL(snd_info_create_card_entry); -EXPORT_SYMBOL(snd_info_free_entry); -EXPORT_SYMBOL(snd_info_register); -EXPORT_SYMBOL(snd_info_unregister); -EXPORT_SYMBOL(snd_card_proc_new); -#endif - /* info_oss.c */ -#if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS) -EXPORT_SYMBOL(snd_oss_info_register); -#endif - /* control.c */ -EXPORT_SYMBOL(snd_ctl_new); -EXPORT_SYMBOL(snd_ctl_new1); -EXPORT_SYMBOL(snd_ctl_free_one); -EXPORT_SYMBOL(snd_ctl_add); -EXPORT_SYMBOL(snd_ctl_remove); -EXPORT_SYMBOL(snd_ctl_remove_id); -EXPORT_SYMBOL(snd_ctl_rename_id); -EXPORT_SYMBOL(snd_ctl_find_numid); -EXPORT_SYMBOL(snd_ctl_find_id); -EXPORT_SYMBOL(snd_ctl_notify); -EXPORT_SYMBOL(snd_ctl_register_ioctl); -EXPORT_SYMBOL(snd_ctl_unregister_ioctl); -#ifdef CONFIG_COMPAT -EXPORT_SYMBOL(snd_ctl_register_ioctl_compat); -EXPORT_SYMBOL(snd_ctl_unregister_ioctl_compat); -#endif -EXPORT_SYMBOL(snd_ctl_elem_read); -EXPORT_SYMBOL(snd_ctl_elem_write); - /* misc.c */ -EXPORT_SYMBOL(release_and_free_resource); -#ifdef CONFIG_SND_VERBOSE_PRINTK -EXPORT_SYMBOL(snd_verbose_printk); -#endif -#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK) -EXPORT_SYMBOL(snd_verbose_printd); -#endif diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 9055c6de95..c18f6a45e4 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -58,6 +58,8 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) return private_data; } +EXPORT_SYMBOL(snd_lookup_oss_minor_data); + static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev) { int minor; @@ -158,6 +160,8 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, return -EBUSY; } +EXPORT_SYMBOL(snd_register_oss_device); + int snd_unregister_oss_device(int type, struct snd_card *card, int dev) { int minor = snd_oss_kernel_minor(type, card, dev); @@ -197,6 +201,8 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) return 0; } +EXPORT_SYMBOL(snd_unregister_oss_device); + /* * INFO PART */ -- cgit v1.2.2 From 91715ed934fb645948ff17b6c20c6f7fd7688a70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:39 +0200 Subject: [ALSA] Clean up EXPORT_SYMBOL()s in snd-seq module Move EXPORT_SYMBOL()s to places adjacent to functions/variables. Signed-off-by: Takashi Iwai --- sound/core/seq/seq.c | 22 ---------------------- sound/core/seq/seq_clientmgr.c | 12 ++++++++++++ sound/core/seq/seq_lock.c | 2 ++ sound/core/seq/seq_memory.c | 3 +++ sound/core/seq/seq_ports.c | 3 +++ 5 files changed, 20 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 20f954bc7a..2f0d8773ac 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -129,25 +129,3 @@ static void __exit alsa_seq_exit(void) module_init(alsa_seq_init) module_exit(alsa_seq_exit) - - /* seq_clientmgr.c */ -EXPORT_SYMBOL(snd_seq_create_kernel_client); -EXPORT_SYMBOL(snd_seq_delete_kernel_client); -EXPORT_SYMBOL(snd_seq_kernel_client_enqueue); -EXPORT_SYMBOL(snd_seq_kernel_client_enqueue_blocking); -EXPORT_SYMBOL(snd_seq_kernel_client_dispatch); -EXPORT_SYMBOL(snd_seq_kernel_client_ctl); -EXPORT_SYMBOL(snd_seq_kernel_client_write_poll); -EXPORT_SYMBOL(snd_seq_set_queue_tempo); - /* seq_memory.c */ -EXPORT_SYMBOL(snd_seq_expand_var_event); -EXPORT_SYMBOL(snd_seq_dump_var_event); - /* seq_ports.c */ -EXPORT_SYMBOL(snd_seq_event_port_attach); -EXPORT_SYMBOL(snd_seq_event_port_detach); - /* seq_lock.c */ -#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG) -/*EXPORT_SYMBOL(snd_seq_sleep_in_lock);*/ -/*EXPORT_SYMBOL(snd_seq_sleep_timeout_in_lock);*/ -EXPORT_SYMBOL(snd_use_lock_sync_helper); -#endif diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index bb15d9ee88..532a660df5 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1714,6 +1714,8 @@ int snd_seq_set_queue_tempo(int client, struct snd_seq_queue_tempo *tempo) return snd_seq_queue_timer_set_tempo(tempo->queue, client, tempo); } +EXPORT_SYMBOL(snd_seq_set_queue_tempo); + static int snd_seq_ioctl_set_queue_tempo(struct snd_seq_client *client, void __user *arg) { @@ -2264,6 +2266,8 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, return client->number; } +EXPORT_SYMBOL(snd_seq_create_kernel_client); + /* exported to kernel modules */ int snd_seq_delete_kernel_client(int client) { @@ -2280,6 +2284,7 @@ int snd_seq_delete_kernel_client(int client) return 0; } +EXPORT_SYMBOL(snd_seq_delete_kernel_client); /* skeleton to enqueue event, called from snd_seq_kernel_client_enqueue * and snd_seq_kernel_client_enqueue_blocking @@ -2328,6 +2333,8 @@ int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event * ev, return kernel_client_enqueue(client, ev, NULL, 0, atomic, hop); } +EXPORT_SYMBOL(snd_seq_kernel_client_enqueue); + /* * exported, called by kernel clients to enqueue events (with blocking) * @@ -2340,6 +2347,7 @@ int snd_seq_kernel_client_enqueue_blocking(int client, struct snd_seq_event * ev return kernel_client_enqueue(client, ev, file, 1, atomic, hop); } +EXPORT_SYMBOL(snd_seq_kernel_client_enqueue_blocking); /* * exported, called by kernel clients to dispatch events directly to other @@ -2376,6 +2384,7 @@ int snd_seq_kernel_client_dispatch(int client, struct snd_seq_event * ev, return result; } +EXPORT_SYMBOL(snd_seq_kernel_client_dispatch); /* * exported, called by kernel clients to perform same functions as with @@ -2396,6 +2405,7 @@ int snd_seq_kernel_client_ctl(int clientid, unsigned int cmd, void *arg) return result; } +EXPORT_SYMBOL(snd_seq_kernel_client_ctl); /* exported (for OSS emulator) */ int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table *wait) @@ -2413,6 +2423,8 @@ int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table return 0; } +EXPORT_SYMBOL(snd_seq_kernel_client_write_poll); + /*---------------------------------------------------------------------------*/ #ifdef CONFIG_PROC_FS diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c index a837a94b2d..1a34941d42 100644 --- a/sound/core/seq/seq_lock.c +++ b/sound/core/seq/seq_lock.c @@ -44,4 +44,6 @@ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line) } } +EXPORT_SYMBOL(snd_use_lock_sync_helper); + #endif diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 40b4f679c8..4bffe509f7 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -118,6 +118,8 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event, return 0; } +EXPORT_SYMBOL(snd_seq_dump_var_event); + /* * exported: @@ -167,6 +169,7 @@ int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char return err < 0 ? err : newlen; } +EXPORT_SYMBOL(snd_seq_expand_var_event); /* * release this cell, free extended data if available diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 41e078c938..5f46ee9e21 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -677,6 +677,7 @@ int snd_seq_event_port_attach(int client, return ret; } +EXPORT_SYMBOL(snd_seq_event_port_attach); /* * Detach the driver from a port. @@ -696,3 +697,5 @@ int snd_seq_event_port_detach(int client, int port) return err; } + +EXPORT_SYMBOL(snd_seq_event_port_detach); -- cgit v1.2.2 From 7b09679c431ba91551a90203f7e7dadbb4c26d1b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:39 +0200 Subject: [ALSA] ac97 - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 40 ++++++++++++++++++---------------------- sound/pci/ac97/ac97_pcm.c | 10 ++++++++++ 2 files changed, 28 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6c1937ff0d..72e33b9532 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -253,6 +253,8 @@ void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short va ac97->bus->ops->write(ac97, reg, value); } +EXPORT_SYMBOL(snd_ac97_write); + /** * snd_ac97_read - read a value from the given register * @@ -281,6 +283,8 @@ static inline unsigned short snd_ac97_read_cache(struct snd_ac97 *ac97, unsigned return ac97->regs[reg]; } +EXPORT_SYMBOL(snd_ac97_read); + /** * snd_ac97_write_cache - write a value on the given register and update the cache * @ac97: the ac97 instance @@ -302,6 +306,8 @@ void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned sh mutex_unlock(&ac97->reg_mutex); } +EXPORT_SYMBOL(snd_ac97_write_cache); + /** * snd_ac97_update - update the value on the given register * @ac97: the ac97 instance @@ -331,6 +337,8 @@ int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short va return change; } +EXPORT_SYMBOL(snd_ac97_update); + /** * snd_ac97_update_bits - update the bits on the given register * @ac97: the ac97 instance @@ -356,6 +364,8 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho return change; } +EXPORT_SYMBOL(snd_ac97_update_bits); + /* no lock version - see snd_ac97_updat_bits() */ int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value) @@ -1682,6 +1692,7 @@ const char *snd_ac97_get_short_name(struct snd_ac97 *ac97) return "unknown codec"; } +EXPORT_SYMBOL(snd_ac97_get_short_name); /* wait for a while until registers are accessible after RESET * return 0 if ok, negative not ready @@ -1774,6 +1785,8 @@ int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops, return 0; } +EXPORT_SYMBOL(snd_ac97_bus); + /* stop no dev release warning */ static void ac97_device_release(struct device * dev) { @@ -2117,6 +2130,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, return 0; } +EXPORT_SYMBOL(snd_ac97_mixer); /* * Power down the chip. @@ -2166,6 +2180,8 @@ void snd_ac97_suspend(struct snd_ac97 *ac97) snd_ac97_powerdown(ac97); } +EXPORT_SYMBOL(snd_ac97_suspend); + /* * restore ac97 status */ @@ -2267,6 +2283,8 @@ __reset_ready: snd_ac97_restore_iec958(ac97); } } + +EXPORT_SYMBOL(snd_ac97_resume); #endif @@ -2590,29 +2608,7 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, cons return 0; } - -/* - * Exported symbols - */ - -EXPORT_SYMBOL(snd_ac97_write); -EXPORT_SYMBOL(snd_ac97_read); -EXPORT_SYMBOL(snd_ac97_write_cache); -EXPORT_SYMBOL(snd_ac97_update); -EXPORT_SYMBOL(snd_ac97_update_bits); -EXPORT_SYMBOL(snd_ac97_get_short_name); -EXPORT_SYMBOL(snd_ac97_bus); -EXPORT_SYMBOL(snd_ac97_mixer); -EXPORT_SYMBOL(snd_ac97_pcm_assign); -EXPORT_SYMBOL(snd_ac97_pcm_open); -EXPORT_SYMBOL(snd_ac97_pcm_close); -EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules); EXPORT_SYMBOL(snd_ac97_tune_hardware); -EXPORT_SYMBOL(snd_ac97_set_rate); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_ac97_resume); -EXPORT_SYMBOL(snd_ac97_suspend); -#endif /* * INIT part diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 512a3583b0..f684aa2c00 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -317,6 +317,8 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate) return 0; } +EXPORT_SYMBOL(snd_ac97_set_rate); + static unsigned short get_pslots(struct snd_ac97 *ac97, unsigned char *rate_table, unsigned short *spdif_slots) { if (!ac97_is_audio(ac97)) @@ -550,6 +552,8 @@ int snd_ac97_pcm_assign(struct snd_ac97_bus *bus, return 0; } +EXPORT_SYMBOL(snd_ac97_pcm_assign); + /** * snd_ac97_pcm_open - opens the given AC97 pcm * @pcm: the ac97 pcm instance @@ -633,6 +637,8 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, return err; } +EXPORT_SYMBOL(snd_ac97_pcm_open); + /** * snd_ac97_pcm_close - closes the given AC97 pcm * @pcm: the ac97 pcm instance @@ -658,6 +664,8 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm) return 0; } +EXPORT_SYMBOL(snd_ac97_pcm_close); + static int double_rate_hw_constraint_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -709,3 +717,5 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime) SNDRV_PCM_HW_PARAM_RATE, -1); return err; } + +EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules); -- cgit v1.2.2 From ac19e19b3664feda8040fb9fb7885183a9eb7a80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:39 +0200 Subject: [ALSA] opl3 - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_lib.c | 19 ++++++++++--------- sound/drivers/opl3/opl3_synth.c | 4 ++++ 2 files changed, 14 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 4f85569767..87fe376f38 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -316,6 +316,8 @@ void snd_opl3_interrupt(struct snd_hwdep * hw) } } +EXPORT_SYMBOL(snd_opl3_interrupt); + /* */ @@ -369,6 +371,8 @@ int snd_opl3_new(struct snd_card *card, return 0; } +EXPORT_SYMBOL(snd_opl3_new); + int snd_opl3_init(struct snd_opl3 *opl3) { if (! opl3->command) { @@ -393,6 +397,8 @@ int snd_opl3_init(struct snd_opl3 *opl3) return 0; } +EXPORT_SYMBOL(snd_opl3_init); + int snd_opl3_create(struct snd_card *card, unsigned long l_port, unsigned long r_port, @@ -451,6 +457,8 @@ int snd_opl3_create(struct snd_card *card, return 0; } +EXPORT_SYMBOL(snd_opl3_create); + int snd_opl3_timer_new(struct snd_opl3 * opl3, int timer1_dev, int timer2_dev) { int err; @@ -468,6 +476,8 @@ int snd_opl3_timer_new(struct snd_opl3 * opl3, int timer1_dev, int timer2_dev) return 0; } +EXPORT_SYMBOL(snd_opl3_timer_new); + int snd_opl3_hwdep_new(struct snd_opl3 * opl3, int device, int seq_device, struct snd_hwdep ** rhwdep) @@ -526,17 +536,8 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, return 0; } -EXPORT_SYMBOL(snd_opl3_interrupt); -EXPORT_SYMBOL(snd_opl3_new); -EXPORT_SYMBOL(snd_opl3_init); -EXPORT_SYMBOL(snd_opl3_create); -EXPORT_SYMBOL(snd_opl3_timer_new); EXPORT_SYMBOL(snd_opl3_hwdep_new); -/* opl3_synth.c */ -EXPORT_SYMBOL(snd_opl3_regmap); -EXPORT_SYMBOL(snd_opl3_reset); - /* * INIT part */ diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 6db503f025..a4b3543a71 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -58,6 +58,8 @@ char snd_opl3_regmap[MAX_OPL2_VOICES][4] = { 0x12, 0x15, 0x00, 0x00 } /* is selected (only left reg block) */ }; +EXPORT_SYMBOL(snd_opl3_regmap); + /* * prototypes */ @@ -228,6 +230,7 @@ void snd_opl3_reset(struct snd_opl3 * opl3) opl3->rhythm = 0; } +EXPORT_SYMBOL(snd_opl3_reset); static int snd_opl3_play_note(struct snd_opl3 * opl3, struct snd_dm_fm_note * note) { @@ -445,3 +448,4 @@ static int snd_opl3_set_connection(struct snd_opl3 * opl3, int connection) return 0; } + -- cgit v1.2.2 From 4181e5fe4b27b0a049402a359a4a5d8b80308528 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:39 +0200 Subject: [ALSA] opl4 - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/drivers/opl4/opl4_lib.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 4bc860ae02..01997f24c8 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -43,6 +43,8 @@ void snd_opl4_write(struct snd_opl4 *opl4, u8 reg, u8 value) outb(value, opl4->pcm_port + 1); } +EXPORT_SYMBOL(snd_opl4_write); + u8 snd_opl4_read(struct snd_opl4 *opl4, u8 reg) { snd_opl4_wait(opl4); @@ -52,6 +54,8 @@ u8 snd_opl4_read(struct snd_opl4 *opl4, u8 reg) return inb(opl4->pcm_port + 1); } +EXPORT_SYMBOL(snd_opl4_read); + void snd_opl4_read_memory(struct snd_opl4 *opl4, char *buf, int offset, int size) { unsigned long flags; @@ -76,6 +80,8 @@ void snd_opl4_read_memory(struct snd_opl4 *opl4, char *buf, int offset, int size spin_unlock_irqrestore(&opl4->reg_lock, flags); } +EXPORT_SYMBOL(snd_opl4_read_memory); + void snd_opl4_write_memory(struct snd_opl4 *opl4, const char *buf, int offset, int size) { unsigned long flags; @@ -100,6 +106,8 @@ void snd_opl4_write_memory(struct snd_opl4 *opl4, const char *buf, int offset, i spin_unlock_irqrestore(&opl4->reg_lock, flags); } +EXPORT_SYMBOL(snd_opl4_write_memory); + static void snd_opl4_enable_opl4(struct snd_opl4 *opl4) { outb(OPL3_REG_MODE, opl4->fm_port + 2); @@ -256,10 +264,6 @@ int snd_opl4_create(struct snd_card *card, return 0; } -EXPORT_SYMBOL(snd_opl4_write); -EXPORT_SYMBOL(snd_opl4_read); -EXPORT_SYMBOL(snd_opl4_write_memory); -EXPORT_SYMBOL(snd_opl4_read_memory); EXPORT_SYMBOL(snd_opl4_create); static int __init alsa_opl4_init(void) -- cgit v1.2.2 From 2dd31deeeb238a4f40c9fc9e219dc210fcbf8765 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:39 +0200 Subject: [ALSA] emu10k1 - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 13 ------------- sound/pci/emu10k1/io.c | 4 ++++ sound/pci/emu10k1/memory.c | 8 ++++++++ sound/pci/emu10k1/voice.c | 4 ++++ 4 files changed, 16 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index e71485c23c..42a358f989 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1459,16 +1459,3 @@ void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu) } } #endif - -/* memory.c */ -EXPORT_SYMBOL(snd_emu10k1_synth_alloc); -EXPORT_SYMBOL(snd_emu10k1_synth_free); -EXPORT_SYMBOL(snd_emu10k1_synth_bzero); -EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user); -EXPORT_SYMBOL(snd_emu10k1_memblk_map); -/* voice.c */ -EXPORT_SYMBOL(snd_emu10k1_voice_alloc); -EXPORT_SYMBOL(snd_emu10k1_voice_free); -/* io.c */ -EXPORT_SYMBOL(snd_emu10k1_ptr_read); -EXPORT_SYMBOL(snd_emu10k1_ptr_write); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index ef5304df8c..029e7856c4 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -62,6 +62,8 @@ unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, un } } +EXPORT_SYMBOL(snd_emu10k1_ptr_read); + void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data) { unsigned int regptr; @@ -92,6 +94,8 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i } } +EXPORT_SYMBOL(snd_emu10k1_ptr_write); + unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index e7ec98649f..4fcaefe5a3 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -287,6 +287,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b return err; } +EXPORT_SYMBOL(snd_emu10k1_memblk_map); + /* * page allocation for DMA */ @@ -387,6 +389,7 @@ snd_emu10k1_synth_alloc(struct snd_emu10k1 *hw, unsigned int size) return (struct snd_util_memblk *)blk; } +EXPORT_SYMBOL(snd_emu10k1_synth_alloc); /* * free a synth sample area @@ -409,6 +412,7 @@ snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk) return 0; } +EXPORT_SYMBOL(snd_emu10k1_synth_free); /* check new allocation range */ static void get_single_page_range(struct snd_util_memhdr *hdr, @@ -540,6 +544,8 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk return 0; } +EXPORT_SYMBOL(snd_emu10k1_synth_bzero); + /* * copy_from_user(blk + offset, data, size) */ @@ -568,3 +574,5 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me } while (offset < end_offset); return 0; } + +EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user); diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index 56ffb7dc3e..94eca82dd4 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -139,6 +139,8 @@ int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int number, return result; } +EXPORT_SYMBOL(snd_emu10k1_voice_alloc); + int snd_emu10k1_voice_free(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *pvoice) { @@ -153,3 +155,5 @@ int snd_emu10k1_voice_free(struct snd_emu10k1 *emu, spin_unlock_irqrestore(&emu->voice_lock, flags); return 0; } + +EXPORT_SYMBOL(snd_emu10k1_voice_free); -- cgit v1.2.2 From cbef55f3d8e4e7efef4703c82302a0021d781483 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] trident - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/pci/trident/trident_main.c | 20 ++++++++++---------- sound/pci/trident/trident_memory.c | 3 +++ 2 files changed, 13 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 52178b8ad4..850579208e 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -306,6 +306,8 @@ void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice) outl(mask, TRID_REG(trident, reg)); } +EXPORT_SYMBOL(snd_trident_start_voice); + /*--------------------------------------------------------------------------- void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice) @@ -328,6 +330,8 @@ void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice) outl(mask, TRID_REG(trident, reg)); } +EXPORT_SYMBOL(snd_trident_stop_voice); + /*--------------------------------------------------------------------------- int snd_trident_allocate_pcm_channel(struct snd_trident *trident) @@ -502,6 +506,8 @@ void snd_trident_write_voice_regs(struct snd_trident * trident, #endif } +EXPORT_SYMBOL(snd_trident_write_voice_regs); + /*--------------------------------------------------------------------------- snd_trident_write_cso_reg @@ -3884,6 +3890,8 @@ struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, return NULL; } +EXPORT_SYMBOL(snd_trident_alloc_voice); + void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice) { unsigned long flags; @@ -3912,6 +3920,8 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi private_free(voice); } +EXPORT_SYMBOL(snd_trident_free_voice); + static void snd_trident_clear_voices(struct snd_trident * trident, unsigned short v_min, unsigned short v_max) { unsigned int i, val, mask[2] = { 0, 0 }; @@ -3993,13 +4003,3 @@ int snd_trident_resume(struct pci_dev *pci) return 0; } #endif /* CONFIG_PM */ - -EXPORT_SYMBOL(snd_trident_alloc_voice); -EXPORT_SYMBOL(snd_trident_free_voice); -EXPORT_SYMBOL(snd_trident_start_voice); -EXPORT_SYMBOL(snd_trident_stop_voice); -EXPORT_SYMBOL(snd_trident_write_voice_regs); -/* trident_memory.c symbols */ -EXPORT_SYMBOL(snd_trident_synth_alloc); -EXPORT_SYMBOL(snd_trident_synth_free); -EXPORT_SYMBOL(snd_trident_synth_copy_from_user); diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index 46c6982c9e..aff3f87413 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -349,6 +349,7 @@ snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size) return blk; } +EXPORT_SYMBOL(snd_trident_synth_alloc); /* * free a synth sample area @@ -365,6 +366,7 @@ snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk) return 0; } +EXPORT_SYMBOL(snd_trident_synth_free); /* * reset TLB entry and free kernel page @@ -486,3 +488,4 @@ int snd_trident_synth_copy_from_user(struct snd_trident *trident, return 0; } +EXPORT_SYMBOL(snd_trident_synth_copy_from_user); -- cgit v1.2.2 From 95ff17564b6db34cad0cd67678fb79174e77531e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] emux - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/synth/emux/emux.c | 12 ++---------- sound/synth/emux/emux_synth.c | 5 +++++ sound/synth/emux/soundfont.c | 4 +++- 3 files changed, 10 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index fc733bbf44..573e3701c1 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -63,6 +63,7 @@ int snd_emux_new(struct snd_emux **remu) return 0; } +EXPORT_SYMBOL(snd_emux_new); /* */ @@ -136,6 +137,7 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch return 0; } +EXPORT_SYMBOL(snd_emux_register); /* */ @@ -171,18 +173,8 @@ int snd_emux_free(struct snd_emux *emu) return 0; } - -EXPORT_SYMBOL(snd_emux_new); -EXPORT_SYMBOL(snd_emux_register); EXPORT_SYMBOL(snd_emux_free); -EXPORT_SYMBOL(snd_emux_terminate_all); -EXPORT_SYMBOL(snd_emux_lock_voice); -EXPORT_SYMBOL(snd_emux_unlock_voice); - -/* soundfont.c */ -EXPORT_SYMBOL(snd_sf_linear_to_log); - /* * INIT part diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 24705d15eb..3733118d39 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -434,6 +434,7 @@ snd_emux_terminate_all(struct snd_emux *emu) spin_unlock_irqrestore(&emu->voice_lock, flags); } +EXPORT_SYMBOL(snd_emux_terminate_all); /* * Terminate all voices associated with the given port @@ -951,6 +952,8 @@ void snd_emux_lock_voice(struct snd_emux *emu, int voice) spin_unlock_irqrestore(&emu->voice_lock, flags); } +EXPORT_SYMBOL(snd_emux_lock_voice); + /* */ void snd_emux_unlock_voice(struct snd_emux *emu, int voice) @@ -965,3 +968,5 @@ void snd_emux_unlock_voice(struct snd_emux *emu, int voice) voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } + +EXPORT_SYMBOL(snd_emux_unlock_voice); diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 32c27162df..7f0bdea0df 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -810,6 +810,9 @@ snd_sf_linear_to_log(unsigned int amount, int offset, int ratio) return v; } +EXPORT_SYMBOL(snd_sf_linear_to_log); + + #define OFFSET_MSEC 653117 /* base = 1000 */ #define OFFSET_ABSCENT 851781 /* base = 8176 */ #define OFFSET_SAMPLERATE 1011119 /* base = 44100 */ @@ -1485,4 +1488,3 @@ snd_soundfont_remove_unlocked(struct snd_sf_list *sflist) unlock_preset(sflist); return 0; } - -- cgit v1.2.2 From fa325eb3afc3cdaf7fba6ee3eaf05b243f5614a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] vx - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_core.c | 30 ++++++++++++++---------------- sound/drivers/vx/vx_hwdep.c | 3 +++ 2 files changed, 17 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index fa4a2b5c2d..e1c3dda157 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -70,6 +70,8 @@ int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int t return -EIO; } +EXPORT_SYMBOL(snd_vx_check_reg_bit); + /* * vx_send_irq_dsp - set command irq bit * @num: the requested IRQ type, IRQ_XXX @@ -465,6 +467,8 @@ int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *boot) return 0; } +EXPORT_SYMBOL(snd_vx_load_boot_image); + /* * vx_test_irq_src - query the source of interrupts * @@ -545,6 +549,7 @@ irqreturn_t snd_vx_irq_handler(int irq, void *dev, struct pt_regs *regs) return IRQ_HANDLED; } +EXPORT_SYMBOL(snd_vx_irq_handler); /* */ @@ -657,6 +662,8 @@ int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *boot) return 0; } +EXPORT_SYMBOL(snd_vx_dsp_boot); + /** * snd_vx_dsp_load - load the DSP image */ @@ -705,6 +712,8 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) return 0; } +EXPORT_SYMBOL(snd_vx_dsp_load); + #ifdef CONFIG_PM /* * suspend @@ -721,6 +730,8 @@ int snd_vx_suspend(struct vx_core *chip, pm_message_t state) return 0; } +EXPORT_SYMBOL(snd_vx_suspend); + /* * resume */ @@ -747,6 +758,7 @@ int snd_vx_resume(struct vx_core *chip) return 0; } +EXPORT_SYMBOL(snd_vx_resume); #endif /** @@ -790,6 +802,8 @@ struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw, return chip; } +EXPORT_SYMBOL(snd_vx_create); + /* * module entries */ @@ -804,19 +818,3 @@ static void __exit alsa_vx_core_exit(void) module_init(alsa_vx_core_init) module_exit(alsa_vx_core_exit) - -/* - * exports - */ -EXPORT_SYMBOL(snd_vx_check_reg_bit); -EXPORT_SYMBOL(snd_vx_create); -EXPORT_SYMBOL(snd_vx_setup_firmware); -EXPORT_SYMBOL(snd_vx_free_firmware); -EXPORT_SYMBOL(snd_vx_irq_handler); -EXPORT_SYMBOL(snd_vx_dsp_boot); -EXPORT_SYMBOL(snd_vx_dsp_load); -EXPORT_SYMBOL(snd_vx_load_boot_image); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_vx_suspend); -EXPORT_SYMBOL(snd_vx_resume); -#endif diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index d837783fb5..e1920af450 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -250,3 +250,6 @@ void snd_vx_free_firmware(struct vx_core *chip) } #endif /* SND_VX_FW_LOADER */ + +EXPORT_SYMBOL(snd_vx_setup_firmware); +EXPORT_SYMBOL(snd_vx_free_firmware); -- cgit v1.2.2 From 57c65c116e1c03c54ac7c4bf38f2b4086d2c1a17 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] i2c - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/i2c/i2c.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index edfe76fb00..b60fb18928 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -106,6 +106,8 @@ int snd_i2c_bus_create(struct snd_card *card, const char *name, return 0; } +EXPORT_SYMBOL(snd_i2c_bus_create); + int snd_i2c_device_create(struct snd_i2c_bus *bus, const char *name, unsigned char addr, struct snd_i2c_device **rdevice) { @@ -124,6 +126,8 @@ int snd_i2c_device_create(struct snd_i2c_bus *bus, const char *name, return 0; } +EXPORT_SYMBOL(snd_i2c_device_create); + int snd_i2c_device_free(struct snd_i2c_device *device) { if (device->bus) @@ -134,22 +138,29 @@ int snd_i2c_device_free(struct snd_i2c_device *device) return 0; } +EXPORT_SYMBOL(snd_i2c_device_free); + int snd_i2c_sendbytes(struct snd_i2c_device *device, unsigned char *bytes, int count) { return device->bus->ops->sendbytes(device, bytes, count); } +EXPORT_SYMBOL(snd_i2c_sendbytes); int snd_i2c_readbytes(struct snd_i2c_device *device, unsigned char *bytes, int count) { return device->bus->ops->readbytes(device, bytes, count); } +EXPORT_SYMBOL(snd_i2c_readbytes); + int snd_i2c_probeaddr(struct snd_i2c_bus *bus, unsigned short addr) { return bus->ops->probeaddr(bus, addr); } +EXPORT_SYMBOL(snd_i2c_probeaddr); + /* * bit-operations */ @@ -320,12 +331,6 @@ static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr) return err; } -EXPORT_SYMBOL(snd_i2c_bus_create); -EXPORT_SYMBOL(snd_i2c_device_create); -EXPORT_SYMBOL(snd_i2c_device_free); -EXPORT_SYMBOL(snd_i2c_sendbytes); -EXPORT_SYMBOL(snd_i2c_readbytes); -EXPORT_SYMBOL(snd_i2c_probeaddr); static int __init alsa_i2c_init(void) { -- cgit v1.2.2 From e5e8a1d4618595ea406336da3cdbd0c6eb6f260d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] hda-codec - Move EXPORT_SYMBOL() to adjacent to each function Move EXPORT_SYMBOL() to adjacent to each exported function/variable. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5bee3b5364..8c2a8174ec 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -86,6 +86,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int dire return res; } +EXPORT_SYMBOL(snd_hda_codec_read); + /** * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec @@ -108,6 +110,8 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, return err; } +EXPORT_SYMBOL(snd_hda_codec_write); + /** * snd_hda_sequence_write - sequence writes * @codec: the HDA codec @@ -122,6 +126,8 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq) snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param); } +EXPORT_SYMBOL(snd_hda_sequence_write); + /** * snd_hda_get_sub_nodes - get the range of sub nodes * @codec: the HDA codec @@ -140,6 +146,8 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *sta return (int)(parm & 0x7fff); } +EXPORT_SYMBOL(snd_hda_get_sub_nodes); + /** * snd_hda_get_connections - get connection list * @codec: the HDA codec @@ -256,6 +264,8 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) return 0; } +EXPORT_SYMBOL(snd_hda_queue_unsol_event); + /* * process queueud unsolicited events */ @@ -384,6 +394,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, return 0; } +EXPORT_SYMBOL(snd_hda_bus_new); /* * find a matching codec preset @@ -587,6 +598,8 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, return 0; } +EXPORT_SYMBOL(snd_hda_codec_new); + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -609,6 +622,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stre snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); } +EXPORT_SYMBOL(snd_hda_codec_setup_stream); /* * amp access functions @@ -1294,6 +1308,7 @@ int snd_hda_build_controls(struct hda_bus *bus) return 0; } +EXPORT_SYMBOL(snd_hda_build_controls); /* * stream formats @@ -1382,6 +1397,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, return val; } +EXPORT_SYMBOL(snd_hda_calc_stream_format); + /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats * @codec: the HDA codec @@ -1663,6 +1680,7 @@ int snd_hda_build_pcms(struct hda_bus *bus) return 0; } +EXPORT_SYMBOL(snd_hda_build_pcms); /** * snd_hda_check_board_config - compare the current codec with the config table @@ -2165,6 +2183,8 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) return 0; } +EXPORT_SYMBOL(snd_hda_suspend); + /** * snd_hda_resume - resume the codecs * @bus: the HDA bus @@ -2187,6 +2207,8 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } +EXPORT_SYMBOL(snd_hda_resume); + /** * snd_hda_resume_ctls - resume controls in the new control list * @codec: the HDA codec @@ -2246,25 +2268,6 @@ int snd_hda_resume_spdif_in(struct hda_codec *codec) } #endif -/* - * symbols exported for controller modules - */ -EXPORT_SYMBOL(snd_hda_codec_read); -EXPORT_SYMBOL(snd_hda_codec_write); -EXPORT_SYMBOL(snd_hda_sequence_write); -EXPORT_SYMBOL(snd_hda_get_sub_nodes); -EXPORT_SYMBOL(snd_hda_queue_unsol_event); -EXPORT_SYMBOL(snd_hda_bus_new); -EXPORT_SYMBOL(snd_hda_codec_new); -EXPORT_SYMBOL(snd_hda_codec_setup_stream); -EXPORT_SYMBOL(snd_hda_calc_stream_format); -EXPORT_SYMBOL(snd_hda_build_pcms); -EXPORT_SYMBOL(snd_hda_build_controls); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_hda_suspend); -EXPORT_SYMBOL(snd_hda_resume); -#endif - /* * INIT part */ -- cgit v1.2.2 From e88e8ae639a4908b903d9406c54e99a729b01a28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] Move OSS-specific hw_params helper to snd-pcm-oss module Move EXPORT_SYMBOL()s to places adjacent to functions/variables. Also move OSS-specific hw_params helper functions to pcm_oss.c. Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 481 +++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 36 +-- sound/core/pcm_lib.c | 649 ++++------------------------------------------- sound/core/pcm_memory.c | 12 + sound/core/pcm_misc.c | 24 ++ sound/core/pcm_native.c | 21 +- 6 files changed, 597 insertions(+), 626 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index ac990bf0b4..0d2e232afe 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -78,6 +78,487 @@ static inline void snd_leave_user(mm_segment_t fs) set_fs(fs); } +/* + * helper functions to process hw_params + */ +static int snd_interval_refine_min(struct snd_interval *i, unsigned int min, int openmin) +{ + int changed = 0; + if (i->min < min) { + i->min = min; + i->openmin = openmin; + changed = 1; + } else if (i->min == min && !i->openmin && openmin) { + i->openmin = 1; + changed = 1; + } + if (i->integer) { + if (i->openmin) { + i->min++; + i->openmin = 0; + } + } + if (snd_interval_checkempty(i)) { + snd_interval_none(i); + return -EINVAL; + } + return changed; +} + +static int snd_interval_refine_max(struct snd_interval *i, unsigned int max, int openmax) +{ + int changed = 0; + if (i->max > max) { + i->max = max; + i->openmax = openmax; + changed = 1; + } else if (i->max == max && !i->openmax && openmax) { + i->openmax = 1; + changed = 1; + } + if (i->integer) { + if (i->openmax) { + i->max--; + i->openmax = 0; + } + } + if (snd_interval_checkempty(i)) { + snd_interval_none(i); + return -EINVAL; + } + return changed; +} + +static int snd_interval_refine_set(struct snd_interval *i, unsigned int val) +{ + struct snd_interval t; + t.empty = 0; + t.min = t.max = val; + t.openmin = t.openmax = 0; + t.integer = 1; + return snd_interval_refine(i, &t); +} + +/** + * snd_pcm_hw_param_value_min + * @params: the hw_params instance + * @var: parameter to retrieve + * @dir: pointer to the direction (-1,0,1) or NULL + * + * Return the minimum value for field PAR. + */ +static unsigned int +snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, int *dir) +{ + if (hw_is_mask(var)) { + if (dir) + *dir = 0; + return snd_mask_min(hw_param_mask_c(params, var)); + } + if (hw_is_interval(var)) { + const struct snd_interval *i = hw_param_interval_c(params, var); + if (dir) + *dir = i->openmin; + return snd_interval_min(i); + } + return -EINVAL; +} + +/** + * snd_pcm_hw_param_value_max + * @params: the hw_params instance + * @var: parameter to retrieve + * @dir: pointer to the direction (-1,0,1) or NULL + * + * Return the maximum value for field PAR. + */ +static unsigned int +snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, int *dir) +{ + if (hw_is_mask(var)) { + if (dir) + *dir = 0; + return snd_mask_max(hw_param_mask_c(params, var)); + } + if (hw_is_interval(var)) { + const struct snd_interval *i = hw_param_interval_c(params, var); + if (dir) + *dir = - (int) i->openmax; + return snd_interval_max(i); + } + return -EINVAL; +} + +static int _snd_pcm_hw_param_mask(struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, + const struct snd_mask *val) +{ + int changed; + changed = snd_mask_refine(hw_param_mask(params, var), val); + if (changed) { + params->cmask |= 1 << var; + params->rmask |= 1 << var; + } + return changed; +} + +static int snd_pcm_hw_param_mask(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, + const struct snd_mask *val) +{ + int changed = _snd_pcm_hw_param_mask(params, var, val); + if (changed < 0) + return changed; + if (params->rmask) { + int err = snd_pcm_hw_refine(pcm, params); + if (err < 0) + return err; + } + return 0; +} + +static int _snd_pcm_hw_param_min(struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int val, + int dir) +{ + int changed; + int open = 0; + if (dir) { + if (dir > 0) { + open = 1; + } else if (dir < 0) { + if (val > 0) { + open = 1; + val--; + } + } + } + if (hw_is_mask(var)) + changed = snd_mask_refine_min(hw_param_mask(params, var), + val + !!open); + else if (hw_is_interval(var)) + changed = snd_interval_refine_min(hw_param_interval(params, var), + val, open); + else + return -EINVAL; + if (changed) { + params->cmask |= 1 << var; + params->rmask |= 1 << var; + } + return changed; +} + +/** + * snd_pcm_hw_param_min + * @pcm: PCM instance + * @params: the hw_params instance + * @var: parameter to retrieve + * @val: minimal value + * @dir: pointer to the direction (-1,0,1) or NULL + * + * Inside configuration space defined by PARAMS remove from PAR all + * values < VAL. Reduce configuration space accordingly. + * Return new minimum or -EINVAL if the configuration space is empty + */ +static int snd_pcm_hw_param_min(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int val, + int *dir) +{ + int changed = _snd_pcm_hw_param_min(params, var, val, dir ? *dir : 0); + if (changed < 0) + return changed; + if (params->rmask) { + int err = snd_pcm_hw_refine(pcm, params); + if (err < 0) + return err; + } + return snd_pcm_hw_param_value_min(params, var, dir); +} + +static int _snd_pcm_hw_param_max(struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int val, + int dir) +{ + int changed; + int open = 0; + if (dir) { + if (dir < 0) { + open = 1; + } else if (dir > 0) { + open = 1; + val++; + } + } + if (hw_is_mask(var)) { + if (val == 0 && open) { + snd_mask_none(hw_param_mask(params, var)); + changed = -EINVAL; + } else + changed = snd_mask_refine_max(hw_param_mask(params, var), + val - !!open); + } else if (hw_is_interval(var)) + changed = snd_interval_refine_max(hw_param_interval(params, var), + val, open); + else + return -EINVAL; + if (changed) { + params->cmask |= 1 << var; + params->rmask |= 1 << var; + } + return changed; +} + +/** + * snd_pcm_hw_param_max + * @pcm: PCM instance + * @params: the hw_params instance + * @var: parameter to retrieve + * @val: maximal value + * @dir: pointer to the direction (-1,0,1) or NULL + * + * Inside configuration space defined by PARAMS remove from PAR all + * values >= VAL + 1. Reduce configuration space accordingly. + * Return new maximum or -EINVAL if the configuration space is empty + */ +static int snd_pcm_hw_param_max(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int val, + int *dir) +{ + int changed = _snd_pcm_hw_param_max(params, var, val, dir ? *dir : 0); + if (changed < 0) + return changed; + if (params->rmask) { + int err = snd_pcm_hw_refine(pcm, params); + if (err < 0) + return err; + } + return snd_pcm_hw_param_value_max(params, var, dir); +} + +static int boundary_sub(int a, int adir, + int b, int bdir, + int *c, int *cdir) +{ + adir = adir < 0 ? -1 : (adir > 0 ? 1 : 0); + bdir = bdir < 0 ? -1 : (bdir > 0 ? 1 : 0); + *c = a - b; + *cdir = adir - bdir; + if (*cdir == -2) { + (*c)--; + } else if (*cdir == 2) { + (*c)++; + } + return 0; +} + +static int boundary_lt(unsigned int a, int adir, + unsigned int b, int bdir) +{ + if (adir < 0) { + a--; + adir = 1; + } else if (adir > 0) + adir = 1; + if (bdir < 0) { + b--; + bdir = 1; + } else if (bdir > 0) + bdir = 1; + return a < b || (a == b && adir < bdir); +} + +/* Return 1 if min is nearer to best than max */ +static int boundary_nearer(int min, int mindir, + int best, int bestdir, + int max, int maxdir) +{ + int dmin, dmindir; + int dmax, dmaxdir; + boundary_sub(best, bestdir, min, mindir, &dmin, &dmindir); + boundary_sub(max, maxdir, best, bestdir, &dmax, &dmaxdir); + return boundary_lt(dmin, dmindir, dmax, dmaxdir); +} + +/** + * snd_pcm_hw_param_near + * @pcm: PCM instance + * @params: the hw_params instance + * @var: parameter to retrieve + * @best: value to set + * @dir: pointer to the direction (-1,0,1) or NULL + * + * Inside configuration space defined by PARAMS set PAR to the available value + * nearest to VAL. Reduce configuration space accordingly. + * This function cannot be called for SNDRV_PCM_HW_PARAM_ACCESS, + * SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SUBFORMAT. + * Return the value found. + */ +static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int best, + int *dir) +{ + struct snd_pcm_hw_params *save = NULL; + int v; + unsigned int saved_min; + int last = 0; + int min, max; + int mindir, maxdir; + int valdir = dir ? *dir : 0; + /* FIXME */ + if (best > INT_MAX) + best = INT_MAX; + min = max = best; + mindir = maxdir = valdir; + if (maxdir > 0) + maxdir = 0; + else if (maxdir == 0) + maxdir = -1; + else { + maxdir = 1; + max--; + } + save = kmalloc(sizeof(*save), GFP_KERNEL); + if (save == NULL) + return -ENOMEM; + *save = *params; + saved_min = min; + min = snd_pcm_hw_param_min(pcm, params, var, min, &mindir); + if (min >= 0) { + struct snd_pcm_hw_params *params1; + if (max < 0) + goto _end; + if ((unsigned int)min == saved_min && mindir == valdir) + goto _end; + params1 = kmalloc(sizeof(*params1), GFP_KERNEL); + if (params1 == NULL) { + kfree(save); + return -ENOMEM; + } + *params1 = *save; + max = snd_pcm_hw_param_max(pcm, params1, var, max, &maxdir); + if (max < 0) { + kfree(params1); + goto _end; + } + if (boundary_nearer(max, maxdir, best, valdir, min, mindir)) { + *params = *params1; + last = 1; + } + kfree(params1); + } else { + *params = *save; + max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir); + snd_assert(max >= 0, return -EINVAL); + last = 1; + } + _end: + kfree(save); + if (last) + v = snd_pcm_hw_param_last(pcm, params, var, dir); + else + v = snd_pcm_hw_param_first(pcm, params, var, dir); + snd_assert(v >= 0, return -EINVAL); + return v; +} + +static int _snd_pcm_hw_param_set(struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int val, + int dir) +{ + int changed; + if (hw_is_mask(var)) { + struct snd_mask *m = hw_param_mask(params, var); + if (val == 0 && dir < 0) { + changed = -EINVAL; + snd_mask_none(m); + } else { + if (dir > 0) + val++; + else if (dir < 0) + val--; + changed = snd_mask_refine_set(hw_param_mask(params, var), val); + } + } else if (hw_is_interval(var)) { + struct snd_interval *i = hw_param_interval(params, var); + if (val == 0 && dir < 0) { + changed = -EINVAL; + snd_interval_none(i); + } else if (dir == 0) + changed = snd_interval_refine_set(i, val); + else { + struct snd_interval t; + t.openmin = 1; + t.openmax = 1; + t.empty = 0; + t.integer = 0; + if (dir < 0) { + t.min = val - 1; + t.max = val; + } else { + t.min = val; + t.max = val+1; + } + changed = snd_interval_refine(i, &t); + } + } else + return -EINVAL; + if (changed) { + params->cmask |= 1 << var; + params->rmask |= 1 << var; + } + return changed; +} + +/** + * snd_pcm_hw_param_set + * @pcm: PCM instance + * @params: the hw_params instance + * @var: parameter to retrieve + * @val: value to set + * @dir: pointer to the direction (-1,0,1) or NULL + * + * Inside configuration space defined by PARAMS remove from PAR all + * values != VAL. Reduce configuration space accordingly. + * Return VAL or -EINVAL if the configuration space is empty + */ +static int snd_pcm_hw_param_set(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, unsigned int val, + int dir) +{ + int changed = _snd_pcm_hw_param_set(params, var, val, dir); + if (changed < 0) + return changed; + if (params->rmask) { + int err = snd_pcm_hw_refine(pcm, params); + if (err < 0) + return err; + } + return snd_pcm_hw_param_value(params, var, NULL); +} + +static int _snd_pcm_hw_param_setinteger(struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var) +{ + int changed; + changed = snd_interval_setinteger(hw_param_interval(params, var)); + if (changed) { + params->cmask |= 1 << var; + params->rmask |= 1 << var; + } + return changed; +} + +/* + * plugin + */ + #ifdef CONFIG_SND_PCM_OSS_PLUGINS static int snd_pcm_oss_plugin_clear(struct snd_pcm_substream *substream) { diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 84b0003823..8c15c01907 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -671,6 +671,8 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) return 0; } +EXPORT_SYMBOL(snd_pcm_new_stream); + /** * snd_pcm_new - create a new PCM instance * @card: the card instance @@ -730,6 +732,8 @@ int snd_pcm_new(struct snd_card *card, char *id, int device, return 0; } +EXPORT_SYMBOL(snd_pcm_new); + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -1022,6 +1026,8 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) return 0; } +EXPORT_SYMBOL(snd_pcm_notify); + #ifdef CONFIG_PROC_FS /* * Info interface @@ -1099,33 +1105,3 @@ static void __exit alsa_pcm_exit(void) module_init(alsa_pcm_init) module_exit(alsa_pcm_exit) - -EXPORT_SYMBOL(snd_pcm_new); -EXPORT_SYMBOL(snd_pcm_new_stream); -EXPORT_SYMBOL(snd_pcm_notify); -EXPORT_SYMBOL(snd_pcm_open_substream); -EXPORT_SYMBOL(snd_pcm_release_substream); - /* pcm_native.c */ -EXPORT_SYMBOL(snd_pcm_link_rwlock); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_pcm_suspend); -EXPORT_SYMBOL(snd_pcm_suspend_all); -#endif -EXPORT_SYMBOL(snd_pcm_kernel_ioctl); -EXPORT_SYMBOL(snd_pcm_mmap_data); -#if SNDRV_PCM_INFO_MMAP_IOMEM -EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); -#endif - /* pcm_misc.c */ -EXPORT_SYMBOL(snd_pcm_format_signed); -EXPORT_SYMBOL(snd_pcm_format_unsigned); -EXPORT_SYMBOL(snd_pcm_format_linear); -EXPORT_SYMBOL(snd_pcm_format_little_endian); -EXPORT_SYMBOL(snd_pcm_format_big_endian); -EXPORT_SYMBOL(snd_pcm_format_width); -EXPORT_SYMBOL(snd_pcm_format_physical_width); -EXPORT_SYMBOL(snd_pcm_format_size); -EXPORT_SYMBOL(snd_pcm_format_silence_64); -EXPORT_SYMBOL(snd_pcm_format_set_silence); -EXPORT_SYMBOL(snd_pcm_build_linear_format); -EXPORT_SYMBOL(snd_pcm_limit_hw_rates); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index eedc6cb038..786f88145e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -289,6 +289,7 @@ void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, struct snd_pcm_ops *ops substream->ops = ops; } +EXPORT_SYMBOL(snd_pcm_set_ops); /** * snd_pcm_sync - set the PCM sync id @@ -306,13 +307,12 @@ void snd_pcm_set_sync(struct snd_pcm_substream *substream) runtime->sync.id32[3] = -1; } +EXPORT_SYMBOL(snd_pcm_set_sync); + /* * Standard ioctl routine */ -/* Code taken from alsa-lib */ -#define assert(a) snd_assert((a), return -EINVAL) - static inline unsigned int div32(unsigned int a, unsigned int b, unsigned int *r) { @@ -369,56 +369,6 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b, return n; } -static int snd_interval_refine_min(struct snd_interval *i, unsigned int min, int openmin) -{ - int changed = 0; - assert(!snd_interval_empty(i)); - if (i->min < min) { - i->min = min; - i->openmin = openmin; - changed = 1; - } else if (i->min == min && !i->openmin && openmin) { - i->openmin = 1; - changed = 1; - } - if (i->integer) { - if (i->openmin) { - i->min++; - i->openmin = 0; - } - } - if (snd_interval_checkempty(i)) { - snd_interval_none(i); - return -EINVAL; - } - return changed; -} - -static int snd_interval_refine_max(struct snd_interval *i, unsigned int max, int openmax) -{ - int changed = 0; - assert(!snd_interval_empty(i)); - if (i->max > max) { - i->max = max; - i->openmax = openmax; - changed = 1; - } else if (i->max == max && !i->openmax && openmax) { - i->openmax = 1; - changed = 1; - } - if (i->integer) { - if (i->openmax) { - i->max--; - i->openmax = 0; - } - } - if (snd_interval_checkempty(i)) { - snd_interval_none(i); - return -EINVAL; - } - return changed; -} - /** * snd_interval_refine - refine the interval value of configurator * @i: the interval value to refine @@ -433,7 +383,7 @@ static int snd_interval_refine_max(struct snd_interval *i, unsigned int max, int int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v) { int changed = 0; - assert(!snd_interval_empty(i)); + snd_assert(!snd_interval_empty(i), return -EINVAL); if (i->min < v->min) { i->min = v->min; i->openmin = v->openmin; @@ -472,9 +422,11 @@ int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v) return changed; } +EXPORT_SYMBOL(snd_interval_refine); + static int snd_interval_refine_first(struct snd_interval *i) { - assert(!snd_interval_empty(i)); + snd_assert(!snd_interval_empty(i), return -EINVAL); if (snd_interval_single(i)) return 0; i->max = i->min; @@ -486,7 +438,7 @@ static int snd_interval_refine_first(struct snd_interval *i) static int snd_interval_refine_last(struct snd_interval *i) { - assert(!snd_interval_empty(i)); + snd_assert(!snd_interval_empty(i), return -EINVAL); if (snd_interval_single(i)) return 0; i->min = i->max; @@ -496,16 +448,6 @@ static int snd_interval_refine_last(struct snd_interval *i) return 1; } -static int snd_interval_refine_set(struct snd_interval *i, unsigned int val) -{ - struct snd_interval t; - t.empty = 0; - t.min = t.max = val; - t.openmin = t.openmax = 0; - t.integer = 1; - return snd_interval_refine(i, &t); -} - void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) { if (a->empty || b->empty) { @@ -621,7 +563,6 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, c->integer = 0; } -#undef assert /* ---- */ @@ -727,6 +668,8 @@ int snd_interval_ratnum(struct snd_interval *i, return err; } +EXPORT_SYMBOL(snd_interval_ratnum); + /** * snd_interval_ratden - refine the interval value * @i: interval to refine @@ -877,6 +820,8 @@ int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int * return changed; } +EXPORT_SYMBOL(snd_interval_list); + static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned int step) { unsigned int n; @@ -953,6 +898,8 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, return 0; } +EXPORT_SYMBOL(snd_pcm_hw_rule_add); + /** * snd_pcm_hw_constraint_mask * @runtime: PCM runtime instance @@ -1007,6 +954,8 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa return snd_interval_setinteger(constrs_interval(constrs, var)); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); + /** * snd_pcm_hw_constraint_minmax * @runtime: PCM runtime instance @@ -1028,6 +977,8 @@ int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return snd_interval_refine(constrs_interval(constrs, var), &t); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax); + static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -1055,6 +1006,8 @@ int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, var, -1); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_list); + static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -1087,6 +1040,8 @@ int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, var, -1); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); + static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -1118,6 +1073,8 @@ int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, var, -1); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_ratdens); + static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -1149,6 +1106,8 @@ int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_msbits); + static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -1173,6 +1132,8 @@ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, var, -1); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_step); + static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { static int pow2_sizes[] = { @@ -1200,6 +1161,8 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, var, -1); } +EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); + /* To use the same code we have in alsa-lib */ #define assert(i) snd_assert((i), return -EINVAL) #ifndef INT_MIN @@ -1224,18 +1187,6 @@ static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params, snd_BUG(); } -#if 0 -/* - * snd_pcm_hw_param_any - */ -int snd_pcm_hw_param_any(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var) -{ - _snd_pcm_hw_param_any(params, var); - return snd_pcm_hw_refine(pcm, params); -} -#endif /* 0 */ - void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params) { unsigned int k; @@ -1247,18 +1198,7 @@ void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params) params->info = ~0U; } -#if 0 -/* - * snd_pcm_hw_params_any - * - * Fill PARAMS with full configuration space boundaries - */ -int snd_pcm_hw_params_any(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params) -{ - _snd_pcm_hw_params_any(params); - return snd_pcm_hw_refine(pcm, params); -} -#endif /* 0 */ +EXPORT_SYMBOL(_snd_pcm_hw_params_any); /** * snd_pcm_hw_param_value @@ -1269,8 +1209,8 @@ int snd_pcm_hw_params_any(struct snd_pcm_substream *pcm, struct snd_pcm_hw_param * Return the value for field PAR if it's fixed in configuration space * defined by PARAMS. Return -EINVAL otherwise */ -static int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, int *dir) +int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, int *dir) { if (hw_is_mask(var)) { const struct snd_mask *mask = hw_param_mask_c(params, var); @@ -1292,57 +1232,7 @@ static int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, return -EINVAL; } -/** - * snd_pcm_hw_param_value_min - * @params: the hw_params instance - * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL - * - * Return the minimum value for field PAR. - */ -unsigned int snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, int *dir) -{ - if (hw_is_mask(var)) { - if (dir) - *dir = 0; - return snd_mask_min(hw_param_mask_c(params, var)); - } - if (hw_is_interval(var)) { - const struct snd_interval *i = hw_param_interval_c(params, var); - if (dir) - *dir = i->openmin; - return snd_interval_min(i); - } - assert(0); - return -EINVAL; -} - -/** - * snd_pcm_hw_param_value_max - * @params: the hw_params instance - * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL - * - * Return the maximum value for field PAR. - */ -unsigned int snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, int *dir) -{ - if (hw_is_mask(var)) { - if (dir) - *dir = 0; - return snd_mask_max(hw_param_mask_c(params, var)); - } - if (hw_is_interval(var)) { - const struct snd_interval *i = hw_param_interval_c(params, var); - if (dir) - *dir = - (int) i->openmax; - return snd_interval_max(i); - } - assert(0); - return -EINVAL; -} +EXPORT_SYMBOL(snd_pcm_hw_param_value); void _snd_pcm_hw_param_setempty(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) @@ -1360,42 +1250,7 @@ void _snd_pcm_hw_param_setempty(struct snd_pcm_hw_params *params, } } -int _snd_pcm_hw_param_setinteger(struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var) -{ - int changed; - assert(hw_is_interval(var)); - changed = snd_interval_setinteger(hw_param_interval(params, var)); - if (changed) { - params->cmask |= 1 << var; - params->rmask |= 1 << var; - } - return changed; -} - -#if 0 -/* - * snd_pcm_hw_param_setinteger - * - * Inside configuration space defined by PARAMS remove from PAR all - * non integer values. Reduce configuration space accordingly. - * Return -EINVAL if the configuration space is empty - */ -int snd_pcm_hw_param_setinteger(struct snd_pcm_substream *pcm, - struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var) -{ - int changed = _snd_pcm_hw_param_setinteger(params, var); - if (changed < 0) - return changed; - if (params->rmask) { - int err = snd_pcm_hw_refine(pcm, params); - if (err < 0) - return err; - } - return 0; -} -#endif /* 0 */ +EXPORT_SYMBOL(_snd_pcm_hw_param_setempty); static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) @@ -1428,9 +1283,9 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, * values > minimum. Reduce configuration space accordingly. * Return the minimum. */ -static int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, - struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, int *dir) +int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, int *dir) { int changed = _snd_pcm_hw_param_first(params, var); if (changed < 0) @@ -1442,6 +1297,8 @@ static int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, return snd_pcm_hw_param_value(params, var, dir); } +EXPORT_SYMBOL(snd_pcm_hw_param_first); + static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { @@ -1473,9 +1330,9 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, * values < maximum. Reduce configuration space accordingly. * Return the maximum. */ -static int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, - struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, int *dir) +int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, + snd_pcm_hw_param_t var, int *dir) { int changed = _snd_pcm_hw_param_last(params, var); if (changed < 0) @@ -1487,367 +1344,7 @@ static int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, return snd_pcm_hw_param_value(params, var, dir); } -int _snd_pcm_hw_param_min(struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int val, int dir) -{ - int changed; - int open = 0; - if (dir) { - if (dir > 0) { - open = 1; - } else if (dir < 0) { - if (val > 0) { - open = 1; - val--; - } - } - } - if (hw_is_mask(var)) - changed = snd_mask_refine_min(hw_param_mask(params, var), val + !!open); - else if (hw_is_interval(var)) - changed = snd_interval_refine_min(hw_param_interval(params, var), val, open); - else { - assert(0); - return -EINVAL; - } - if (changed) { - params->cmask |= 1 << var; - params->rmask |= 1 << var; - } - return changed; -} - -/** - * snd_pcm_hw_param_min - * @pcm: PCM instance - * @params: the hw_params instance - * @var: parameter to retrieve - * @val: minimal value - * @dir: pointer to the direction (-1,0,1) or NULL - * - * Inside configuration space defined by PARAMS remove from PAR all - * values < VAL. Reduce configuration space accordingly. - * Return new minimum or -EINVAL if the configuration space is empty - */ -static int snd_pcm_hw_param_min(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int val, - int *dir) -{ - int changed = _snd_pcm_hw_param_min(params, var, val, dir ? *dir : 0); - if (changed < 0) - return changed; - if (params->rmask) { - int err = snd_pcm_hw_refine(pcm, params); - if (err < 0) - return err; - } - return snd_pcm_hw_param_value_min(params, var, dir); -} - -static int _snd_pcm_hw_param_max(struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int val, - int dir) -{ - int changed; - int open = 0; - if (dir) { - if (dir < 0) { - open = 1; - } else if (dir > 0) { - open = 1; - val++; - } - } - if (hw_is_mask(var)) { - if (val == 0 && open) { - snd_mask_none(hw_param_mask(params, var)); - changed = -EINVAL; - } else - changed = snd_mask_refine_max(hw_param_mask(params, var), val - !!open); - } else if (hw_is_interval(var)) - changed = snd_interval_refine_max(hw_param_interval(params, var), val, open); - else { - assert(0); - return -EINVAL; - } - if (changed) { - params->cmask |= 1 << var; - params->rmask |= 1 << var; - } - return changed; -} - -/** - * snd_pcm_hw_param_max - * @pcm: PCM instance - * @params: the hw_params instance - * @var: parameter to retrieve - * @val: maximal value - * @dir: pointer to the direction (-1,0,1) or NULL - * - * Inside configuration space defined by PARAMS remove from PAR all - * values >= VAL + 1. Reduce configuration space accordingly. - * Return new maximum or -EINVAL if the configuration space is empty - */ -static int snd_pcm_hw_param_max(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int val, - int *dir) -{ - int changed = _snd_pcm_hw_param_max(params, var, val, dir ? *dir : 0); - if (changed < 0) - return changed; - if (params->rmask) { - int err = snd_pcm_hw_refine(pcm, params); - if (err < 0) - return err; - } - return snd_pcm_hw_param_value_max(params, var, dir); -} - -int _snd_pcm_hw_param_set(struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int val, int dir) -{ - int changed; - if (hw_is_mask(var)) { - struct snd_mask *m = hw_param_mask(params, var); - if (val == 0 && dir < 0) { - changed = -EINVAL; - snd_mask_none(m); - } else { - if (dir > 0) - val++; - else if (dir < 0) - val--; - changed = snd_mask_refine_set(hw_param_mask(params, var), val); - } - } else if (hw_is_interval(var)) { - struct snd_interval *i = hw_param_interval(params, var); - if (val == 0 && dir < 0) { - changed = -EINVAL; - snd_interval_none(i); - } else if (dir == 0) - changed = snd_interval_refine_set(i, val); - else { - struct snd_interval t; - t.openmin = 1; - t.openmax = 1; - t.empty = 0; - t.integer = 0; - if (dir < 0) { - t.min = val - 1; - t.max = val; - } else { - t.min = val; - t.max = val+1; - } - changed = snd_interval_refine(i, &t); - } - } else { - assert(0); - return -EINVAL; - } - if (changed) { - params->cmask |= 1 << var; - params->rmask |= 1 << var; - } - return changed; -} - -/** - * snd_pcm_hw_param_set - * @pcm: PCM instance - * @params: the hw_params instance - * @var: parameter to retrieve - * @val: value to set - * @dir: pointer to the direction (-1,0,1) or NULL - * - * Inside configuration space defined by PARAMS remove from PAR all - * values != VAL. Reduce configuration space accordingly. - * Return VAL or -EINVAL if the configuration space is empty - */ -int snd_pcm_hw_param_set(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int val, int dir) -{ - int changed = _snd_pcm_hw_param_set(params, var, val, dir); - if (changed < 0) - return changed; - if (params->rmask) { - int err = snd_pcm_hw_refine(pcm, params); - if (err < 0) - return err; - } - return snd_pcm_hw_param_value(params, var, NULL); -} - -static int _snd_pcm_hw_param_mask(struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, const struct snd_mask *val) -{ - int changed; - assert(hw_is_mask(var)); - changed = snd_mask_refine(hw_param_mask(params, var), val); - if (changed) { - params->cmask |= 1 << var; - params->rmask |= 1 << var; - } - return changed; -} - -/** - * snd_pcm_hw_param_mask - * @pcm: PCM instance - * @params: the hw_params instance - * @var: parameter to retrieve - * @val: mask to apply - * - * Inside configuration space defined by PARAMS remove from PAR all values - * not contained in MASK. Reduce configuration space accordingly. - * This function can be called only for SNDRV_PCM_HW_PARAM_ACCESS, - * SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SUBFORMAT. - * Return 0 on success or -EINVAL - * if the configuration space is empty - */ -int snd_pcm_hw_param_mask(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, const struct snd_mask *val) -{ - int changed = _snd_pcm_hw_param_mask(params, var, val); - if (changed < 0) - return changed; - if (params->rmask) { - int err = snd_pcm_hw_refine(pcm, params); - if (err < 0) - return err; - } - return 0; -} - -static int boundary_sub(int a, int adir, - int b, int bdir, - int *c, int *cdir) -{ - adir = adir < 0 ? -1 : (adir > 0 ? 1 : 0); - bdir = bdir < 0 ? -1 : (bdir > 0 ? 1 : 0); - *c = a - b; - *cdir = adir - bdir; - if (*cdir == -2) { - assert(*c > INT_MIN); - (*c)--; - } else if (*cdir == 2) { - assert(*c < INT_MAX); - (*c)++; - } - return 0; -} - -static int boundary_lt(unsigned int a, int adir, - unsigned int b, int bdir) -{ - assert(a > 0 || adir >= 0); - assert(b > 0 || bdir >= 0); - if (adir < 0) { - a--; - adir = 1; - } else if (adir > 0) - adir = 1; - if (bdir < 0) { - b--; - bdir = 1; - } else if (bdir > 0) - bdir = 1; - return a < b || (a == b && adir < bdir); -} - -/* Return 1 if min is nearer to best than max */ -static int boundary_nearer(int min, int mindir, - int best, int bestdir, - int max, int maxdir) -{ - int dmin, dmindir; - int dmax, dmaxdir; - boundary_sub(best, bestdir, min, mindir, &dmin, &dmindir); - boundary_sub(max, maxdir, best, bestdir, &dmax, &dmaxdir); - return boundary_lt(dmin, dmindir, dmax, dmaxdir); -} - -/** - * snd_pcm_hw_param_near - * @pcm: PCM instance - * @params: the hw_params instance - * @var: parameter to retrieve - * @best: value to set - * @dir: pointer to the direction (-1,0,1) or NULL - * - * Inside configuration space defined by PARAMS set PAR to the available value - * nearest to VAL. Reduce configuration space accordingly. - * This function cannot be called for SNDRV_PCM_HW_PARAM_ACCESS, - * SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SUBFORMAT. - * Return the value found. - */ -int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, - snd_pcm_hw_param_t var, unsigned int best, int *dir) -{ - struct snd_pcm_hw_params *save = NULL; - int v; - unsigned int saved_min; - int last = 0; - int min, max; - int mindir, maxdir; - int valdir = dir ? *dir : 0; - /* FIXME */ - if (best > INT_MAX) - best = INT_MAX; - min = max = best; - mindir = maxdir = valdir; - if (maxdir > 0) - maxdir = 0; - else if (maxdir == 0) - maxdir = -1; - else { - maxdir = 1; - max--; - } - save = kmalloc(sizeof(*save), GFP_KERNEL); - if (save == NULL) - return -ENOMEM; - *save = *params; - saved_min = min; - min = snd_pcm_hw_param_min(pcm, params, var, min, &mindir); - if (min >= 0) { - struct snd_pcm_hw_params *params1; - if (max < 0) - goto _end; - if ((unsigned int)min == saved_min && mindir == valdir) - goto _end; - params1 = kmalloc(sizeof(*params1), GFP_KERNEL); - if (params1 == NULL) { - kfree(save); - return -ENOMEM; - } - *params1 = *save; - max = snd_pcm_hw_param_max(pcm, params1, var, max, &maxdir); - if (max < 0) { - kfree(params1); - goto _end; - } - if (boundary_nearer(max, maxdir, best, valdir, min, mindir)) { - *params = *params1; - last = 1; - } - kfree(params1); - } else { - *params = *save; - max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir); - assert(max >= 0); - last = 1; - } - _end: - kfree(save); - if (last) - v = snd_pcm_hw_param_last(pcm, params, var, dir); - else - v = snd_pcm_hw_param_first(pcm, params, var, dir); - assert(v >= 0); - return v; -} +EXPORT_SYMBOL(snd_pcm_hw_param_last); /** * snd_pcm_hw_param_choose @@ -1967,6 +1464,8 @@ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, return -ENXIO; } +EXPORT_SYMBOL(snd_pcm_lib_ioctl); + /* * Conditions */ @@ -2101,6 +1600,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) kill_fasync(&runtime->fasync, SIGIO, POLL_IN); } +EXPORT_SYMBOL(snd_pcm_period_elapsed); + static int snd_pcm_lib_write_transfer(struct snd_pcm_substream *substream, unsigned int hwoff, unsigned long data, unsigned int off, @@ -2308,6 +1809,8 @@ snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, const v snd_pcm_lib_write_transfer); } +EXPORT_SYMBOL(snd_pcm_lib_write); + static int snd_pcm_lib_writev_transfer(struct snd_pcm_substream *substream, unsigned int hwoff, unsigned long data, unsigned int off, @@ -2370,6 +1873,8 @@ snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream, nonblock, snd_pcm_lib_writev_transfer); } +EXPORT_SYMBOL(snd_pcm_lib_writev); + static int snd_pcm_lib_read_transfer(struct snd_pcm_substream *substream, unsigned int hwoff, unsigned long data, unsigned int off, @@ -2578,6 +2083,8 @@ snd_pcm_sframes_t snd_pcm_lib_read(struct snd_pcm_substream *substream, void __u return snd_pcm_lib_read1(substream, (unsigned long)buf, size, nonblock, snd_pcm_lib_read_transfer); } +EXPORT_SYMBOL(snd_pcm_lib_read); + static int snd_pcm_lib_readv_transfer(struct snd_pcm_substream *substream, unsigned int hwoff, unsigned long data, unsigned int off, @@ -2635,52 +2142,4 @@ snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream, return snd_pcm_lib_read1(substream, (unsigned long)bufs, frames, nonblock, snd_pcm_lib_readv_transfer); } -/* - * Exported symbols - */ - -EXPORT_SYMBOL(snd_interval_refine); -EXPORT_SYMBOL(snd_interval_list); -EXPORT_SYMBOL(snd_interval_ratnum); -EXPORT_SYMBOL(_snd_pcm_hw_params_any); -EXPORT_SYMBOL(_snd_pcm_hw_param_min); -EXPORT_SYMBOL(_snd_pcm_hw_param_set); -EXPORT_SYMBOL(_snd_pcm_hw_param_setempty); -EXPORT_SYMBOL(_snd_pcm_hw_param_setinteger); -EXPORT_SYMBOL(snd_pcm_hw_param_value_min); -EXPORT_SYMBOL(snd_pcm_hw_param_value_max); -EXPORT_SYMBOL(snd_pcm_hw_param_mask); -EXPORT_SYMBOL(snd_pcm_hw_param_first); -EXPORT_SYMBOL(snd_pcm_hw_param_last); -EXPORT_SYMBOL(snd_pcm_hw_param_near); -EXPORT_SYMBOL(snd_pcm_hw_param_set); -EXPORT_SYMBOL(snd_pcm_hw_refine); -EXPORT_SYMBOL(snd_pcm_hw_constraints_init); -EXPORT_SYMBOL(snd_pcm_hw_constraints_complete); -EXPORT_SYMBOL(snd_pcm_hw_constraint_list); -EXPORT_SYMBOL(snd_pcm_hw_constraint_step); -EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); -EXPORT_SYMBOL(snd_pcm_hw_constraint_ratdens); -EXPORT_SYMBOL(snd_pcm_hw_constraint_msbits); -EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax); -EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); -EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); -EXPORT_SYMBOL(snd_pcm_hw_rule_add); -EXPORT_SYMBOL(snd_pcm_set_ops); -EXPORT_SYMBOL(snd_pcm_set_sync); -EXPORT_SYMBOL(snd_pcm_lib_ioctl); -EXPORT_SYMBOL(snd_pcm_stop); -EXPORT_SYMBOL(snd_pcm_period_elapsed); -EXPORT_SYMBOL(snd_pcm_lib_write); -EXPORT_SYMBOL(snd_pcm_lib_read); -EXPORT_SYMBOL(snd_pcm_lib_writev); EXPORT_SYMBOL(snd_pcm_lib_readv); -EXPORT_SYMBOL(snd_pcm_lib_buffer_bytes); -EXPORT_SYMBOL(snd_pcm_lib_period_bytes); -/* pcm_memory.c */ -EXPORT_SYMBOL(snd_pcm_lib_preallocate_free_for_all); -EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages); -EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); -EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page); -EXPORT_SYMBOL(snd_pcm_lib_malloc_pages); -EXPORT_SYMBOL(snd_pcm_lib_free_pages); diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 428f8c169e..eb56167d3b 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -126,6 +126,8 @@ int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm) return 0; } +EXPORT_SYMBOL(snd_pcm_lib_preallocate_free_for_all); + #ifdef CONFIG_SND_VERBOSE_PROCFS /* * read callback for prealloc proc file @@ -253,6 +255,8 @@ int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, return snd_pcm_lib_preallocate_pages1(substream, size, max); } +EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages); + /** * snd_pcm_lib_preallocate_pages_for_all - pre-allocation for continous memory type (all substreams) * @pcm: the pcm instance @@ -280,6 +284,8 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, return 0; } +EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); + /** * snd_pcm_sgbuf_ops_page - get the page struct at the given offset * @substream: the pcm substream instance @@ -298,6 +304,8 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigne return sgbuf->page_table[idx]; } +EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page); + /** * snd_pcm_lib_malloc_pages - allocate the DMA buffer * @substream: the substream to allocate the DMA buffer to @@ -349,6 +357,8 @@ int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size) return 1; /* area was changed */ } +EXPORT_SYMBOL(snd_pcm_lib_malloc_pages); + /** * snd_pcm_lib_free_pages - release the allocated DMA buffer. * @substream: the substream to release the DMA buffer @@ -374,3 +384,5 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) snd_pcm_set_runtime_buffer(substream, NULL); return 0; } + +EXPORT_SYMBOL(snd_pcm_lib_free_pages); diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 593c77f4d1..0019c59a77 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -207,6 +207,8 @@ int snd_pcm_format_signed(snd_pcm_format_t format) return val; } +EXPORT_SYMBOL(snd_pcm_format_signed); + /** * snd_pcm_format_unsigned - Check the PCM format is unsigned linear * @format: the format to check @@ -224,6 +226,8 @@ int snd_pcm_format_unsigned(snd_pcm_format_t format) return !val; } +EXPORT_SYMBOL(snd_pcm_format_unsigned); + /** * snd_pcm_format_linear - Check the PCM format is linear * @format: the format to check @@ -235,6 +239,8 @@ int snd_pcm_format_linear(snd_pcm_format_t format) return snd_pcm_format_signed(format) >= 0; } +EXPORT_SYMBOL(snd_pcm_format_linear); + /** * snd_pcm_format_little_endian - Check the PCM format is little-endian * @format: the format to check @@ -252,6 +258,8 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) return val; } +EXPORT_SYMBOL(snd_pcm_format_little_endian); + /** * snd_pcm_format_big_endian - Check the PCM format is big-endian * @format: the format to check @@ -269,6 +277,8 @@ int snd_pcm_format_big_endian(snd_pcm_format_t format) return !val; } +EXPORT_SYMBOL(snd_pcm_format_big_endian); + /** * snd_pcm_format_width - return the bit-width of the format * @format: the format to check @@ -286,6 +296,8 @@ int snd_pcm_format_width(snd_pcm_format_t format) return val; } +EXPORT_SYMBOL(snd_pcm_format_width); + /** * snd_pcm_format_physical_width - return the physical bit-width of the format * @format: the format to check @@ -303,6 +315,8 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) return val; } +EXPORT_SYMBOL(snd_pcm_format_physical_width); + /** * snd_pcm_format_size - return the byte size of samples on the given format * @format: the format to check @@ -318,6 +332,8 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) return samples * phys_width / 8; } +EXPORT_SYMBOL(snd_pcm_format_size); + /** * snd_pcm_format_silence_64 - return the silent data in 8 bytes array * @format: the format to check @@ -333,6 +349,8 @@ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format) return pcm_formats[format].silence; } +EXPORT_SYMBOL(snd_pcm_format_silence_64); + /** * snd_pcm_format_set_silence - set the silence data on the buffer * @format: the PCM format @@ -402,6 +420,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; } +EXPORT_SYMBOL(snd_pcm_format_set_silence); + /* [width][unsigned][bigendian] */ static int linear_formats[4][2][2] = { {{ SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8}, @@ -432,6 +452,8 @@ snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_end return linear_formats[width][!!unsignd][!!big_endian]; } +EXPORT_SYMBOL(snd_pcm_build_linear_format); + /** * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields * @runtime: the runtime instance @@ -463,3 +485,5 @@ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) } return 0; } + +EXPORT_SYMBOL(snd_pcm_limit_hw_rates); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0860c5a845..7b5729c4b2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -71,8 +71,9 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream); */ DEFINE_RWLOCK(snd_pcm_link_rwlock); -static DECLARE_RWSEM(snd_pcm_link_rwsem); +EXPORT_SYMBOL(snd_pcm_link_rwlock); +static DECLARE_RWSEM(snd_pcm_link_rwsem); static inline mm_segment_t snd_enter_user(void) { @@ -319,6 +320,8 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, return 0; } +EXPORT_SYMBOL(snd_pcm_hw_refine); + static int snd_pcm_hw_refine_user(struct snd_pcm_substream *substream, struct snd_pcm_hw_params __user * _params) { @@ -936,6 +939,8 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) return snd_pcm_action(&snd_pcm_action_stop, substream, state); } +EXPORT_SYMBOL(snd_pcm_stop); + /** * snd_pcm_drain_done * @substream: the PCM substream @@ -1085,6 +1090,8 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream) return err; } +EXPORT_SYMBOL(snd_pcm_suspend); + /** * snd_pcm_suspend_all * @pcm: the PCM instance @@ -1114,6 +1121,8 @@ int snd_pcm_suspend_all(struct snd_pcm *pcm) return 0; } +EXPORT_SYMBOL(snd_pcm_suspend_all); + /* resume */ static int snd_pcm_pre_resume(struct snd_pcm_substream *substream, int state) @@ -2020,6 +2029,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_detach_substream(substream); } +EXPORT_SYMBOL(snd_pcm_release_substream); + int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, struct file *file, struct snd_pcm_substream **rsubstream) @@ -2056,6 +2067,8 @@ int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, return err; } +EXPORT_SYMBOL(snd_pcm_open_substream); + static int snd_pcm_open_file(struct file *file, struct snd_pcm *pcm, int stream, @@ -2768,6 +2781,8 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, return result; } +EXPORT_SYMBOL(snd_pcm_kernel_ioctl); + static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, loff_t * offset) { @@ -3169,6 +3184,8 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, atomic_inc(&substream->runtime->mmap_count); return 0; } + +EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ /* @@ -3212,6 +3229,8 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, return snd_pcm_default_mmap(substream, area); } +EXPORT_SYMBOL(snd_pcm_mmap_data); + static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) { struct snd_pcm_file * pcm_file; -- cgit v1.2.2 From 2f4ca8e5c7cf6a6f7935483d8ee4aa8b039bdd7d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] Clean up ugly hacks in pcm_lib.c Clean up ugly hacks for sync with alsa-lib code in pcm_lib.c. Also, optimize snd_pcm_hw_params_choose() with a loop. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 70 +++++++++++++++++++--------------------------------- 1 file changed, 26 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 786f88145e..a21aa0050e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1163,12 +1163,6 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); -/* To use the same code we have in alsa-lib */ -#define assert(i) snd_assert((i), return -EINVAL) -#ifndef INT_MIN -#define INT_MIN ((int)((unsigned int)INT_MAX+1)) -#endif - static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { @@ -1228,7 +1222,6 @@ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, *dir = i->openmin; return snd_interval_value(i); } - assert(0); return -EINVAL; } @@ -1260,10 +1253,8 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, changed = snd_mask_refine_first(hw_param_mask(params, var)); else if (hw_is_interval(var)) changed = snd_interval_refine_first(hw_param_interval(params, var)); - else { - assert(0); + else return -EINVAL; - } if (changed) { params->cmask |= 1 << var; params->rmask |= 1 << var; @@ -1292,7 +1283,7 @@ int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - assert(err >= 0); + snd_assert(err >= 0, return err); } return snd_pcm_hw_param_value(params, var, dir); } @@ -1307,10 +1298,8 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, changed = snd_mask_refine_last(hw_param_mask(params, var)); else if (hw_is_interval(var)) changed = snd_interval_refine_last(hw_param_interval(params, var)); - else { - assert(0); + else return -EINVAL; - } if (changed) { params->cmask |= 1 << var; params->rmask |= 1 << var; @@ -1339,7 +1328,7 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - assert(err >= 0); + snd_assert(err >= 0, return err); } return snd_pcm_hw_param_value(params, var, dir); } @@ -1356,39 +1345,32 @@ EXPORT_SYMBOL(snd_pcm_hw_param_last); * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time */ -int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params) -{ - int err; - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_ACCESS, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_FORMAT, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_SUBFORMAT, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_CHANNELS, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_RATE, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_PERIOD_TIME, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_last(pcm, params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL); - assert(err >= 0); - - err = snd_pcm_hw_param_first(pcm, params, SNDRV_PCM_HW_PARAM_TICK_TIME, NULL); - assert(err >= 0); +int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params) +{ + static int vars[] = { + SNDRV_PCM_HW_PARAM_ACCESS, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_SUBFORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + SNDRV_PCM_HW_PARAM_TICK_TIME, + -1 + }; + int err, *v; + for (v = vars; *v != -1; v++) { + if (*v != SNDRV_PCM_HW_PARAM_BUFFER_SIZE) + err = snd_pcm_hw_param_first(pcm, params, *v, NULL); + else + err = snd_pcm_hw_param_last(pcm, params, *v, NULL); + snd_assert(err >= 0, return err); + } return 0; } -#undef assert - static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, void *arg) { -- cgit v1.2.2 From 7e4eeec8a30fa9e00cac67a37ca9ddf6cbdb79c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:40 +0200 Subject: [ALSA] Make buffer size of proc text interface variable Make the read/write buffer size of proc text interface variable. Signed-off-by: Takashi Iwai --- sound/core/info.c | 149 +++++++++++++++++++++++++++++------------------------- 1 file changed, 79 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index 9c288539e9..86366839c4 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include @@ -82,6 +81,24 @@ static int snd_info_version_init(void); static int snd_info_version_done(void); +/* resize the proc r/w buffer */ +static int resize_info_buffer(struct snd_info_buffer *buffer, + unsigned int nsize) +{ + char *nbuf; + + nsize = PAGE_ALIGN(nsize); + nbuf = kmalloc(nsize, GFP_KERNEL); + if (! nbuf) + return -ENOMEM; + + memcpy(nbuf, buffer->buffer, buffer->len); + kfree(buffer->buffer); + buffer->buffer = nbuf; + buffer->len = nsize; + return 0; +} + /** * snd_iprintf - printf on the procfs buffer * @buffer: the procfs buffer @@ -95,17 +112,25 @@ int snd_iprintf(struct snd_info_buffer *buffer, char *fmt,...) { va_list args; int len, res; + int err = 0; if (buffer->stop || buffer->error) return 0; len = buffer->len - buffer->size; va_start(args, fmt); - res = vsnprintf(buffer->curr, len, fmt, args); - va_end(args); - if (res >= len) { - buffer->stop = 1; - return 0; + for (;;) { + res = vsnprintf(buffer->buffer + buffer->curr, len, fmt, args); + if (res < len) + break; + err = resize_info_buffer(buffer, buffer->len + PAGE_SIZE); + if (err < 0) + break; + len = buffer->len - buffer->size; } + va_end(args); + + if (err < 0) + return err; buffer->curr += res; buffer->size += res; return res; @@ -225,7 +250,7 @@ static ssize_t snd_info_entry_write(struct file *file, const char __user *buffer struct snd_info_private_data *data; struct snd_info_entry *entry; struct snd_info_buffer *buf; - size_t size = 0; + ssize_t size = 0; loff_t pos; data = file->private_data; @@ -241,14 +266,20 @@ static ssize_t snd_info_entry_write(struct file *file, const char __user *buffer buf = data->wbuffer; if (buf == NULL) return -EIO; - if (pos >= buf->len) - return -ENOMEM; - size = buf->len - pos; - size = min(count, size); - if (copy_from_user(buf->buffer + pos, buffer, size)) + mutex_unlock(&entry->access); + if (pos + count >= buf->len) { + if (resize_info_buffer(buf, pos + count)) { + mutex_unlock(&entry->access); + return -ENOMEM; + } + } + if (copy_from_user(buf->buffer + pos, buffer, count)) { + mutex_unlock(&entry->access); return -EFAULT; - if ((long)buf->size < pos + size) - buf->size = pos + size; + } + buf->size = pos + count; + mutex_unlock(&entry->access); + size = count; break; case SNDRV_INFO_CONTENT_DATA: if (entry->c.ops->write) @@ -283,18 +314,14 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) } mode = file->f_flags & O_ACCMODE; if (mode == O_RDONLY || mode == O_RDWR) { - if ((entry->content == SNDRV_INFO_CONTENT_TEXT && - !entry->c.text.read_size) || - (entry->content == SNDRV_INFO_CONTENT_DATA && + if ((entry->content == SNDRV_INFO_CONTENT_DATA && entry->c.ops->read == NULL)) { err = -ENODEV; goto __error; } } if (mode == O_WRONLY || mode == O_RDWR) { - if ((entry->content == SNDRV_INFO_CONTENT_TEXT && - !entry->c.text.write_size) || - (entry->content == SNDRV_INFO_CONTENT_DATA && + if ((entry->content == SNDRV_INFO_CONTENT_DATA && entry->c.ops->write == NULL)) { err = -ENODEV; goto __error; @@ -310,49 +337,23 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) case SNDRV_INFO_CONTENT_TEXT: if (mode == O_RDONLY || mode == O_RDWR) { buffer = kzalloc(sizeof(*buffer), GFP_KERNEL); - if (buffer == NULL) { - kfree(data); - err = -ENOMEM; - goto __error; - } - buffer->len = (entry->c.text.read_size + - (PAGE_SIZE - 1)) & ~(PAGE_SIZE - 1); - buffer->buffer = vmalloc(buffer->len); - if (buffer->buffer == NULL) { - kfree(buffer); - kfree(data); - err = -ENOMEM; - goto __error; - } - buffer->curr = buffer->buffer; + if (buffer == NULL) + goto __nomem; data->rbuffer = buffer; + buffer->len = PAGE_SIZE; + buffer->buffer = kmalloc(buffer->len, GFP_KERNEL); + if (buffer->buffer == NULL) + goto __nomem; } if (mode == O_WRONLY || mode == O_RDWR) { buffer = kzalloc(sizeof(*buffer), GFP_KERNEL); - if (buffer == NULL) { - if (mode == O_RDWR) { - vfree(data->rbuffer->buffer); - kfree(data->rbuffer); - } - kfree(data); - err = -ENOMEM; - goto __error; - } - buffer->len = (entry->c.text.write_size + - (PAGE_SIZE - 1)) & ~(PAGE_SIZE - 1); - buffer->buffer = vmalloc(buffer->len); - if (buffer->buffer == NULL) { - if (mode == O_RDWR) { - vfree(data->rbuffer->buffer); - kfree(data->rbuffer); - } - kfree(buffer); - kfree(data); - err = -ENOMEM; - goto __error; - } - buffer->curr = buffer->buffer; + if (buffer == NULL) + goto __nomem; data->wbuffer = buffer; + buffer->len = PAGE_SIZE; + buffer->buffer = kmalloc(buffer->len, GFP_KERNEL); + if (buffer->buffer == NULL) + goto __nomem; } break; case SNDRV_INFO_CONTENT_DATA: /* data */ @@ -377,6 +378,17 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) } return 0; + __nomem: + if (data->rbuffer) { + kfree(data->rbuffer->buffer); + kfree(data->rbuffer); + } + if (data->wbuffer) { + kfree(data->wbuffer->buffer); + kfree(data->wbuffer); + } + kfree(data); + err = -ENOMEM; __error: module_put(entry->module); __error1: @@ -395,11 +407,11 @@ static int snd_info_entry_release(struct inode *inode, struct file *file) entry = data->entry; switch (entry->content) { case SNDRV_INFO_CONTENT_TEXT: - if (mode == O_RDONLY || mode == O_RDWR) { - vfree(data->rbuffer->buffer); + if (data->rbuffer) { + kfree(data->rbuffer->buffer); kfree(data->rbuffer); } - if (mode == O_WRONLY || mode == O_RDWR) { + if (data->wbuffer) { if (entry->c.text.write) { entry->c.text.write(entry, data->wbuffer); if (data->wbuffer->error) { @@ -408,7 +420,7 @@ static int snd_info_entry_release(struct inode *inode, struct file *file) data->wbuffer->error); } } - vfree(data->wbuffer->buffer); + kfree(data->wbuffer->buffer); kfree(data->wbuffer); } break; @@ -668,24 +680,22 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) if (len <= 0 || buffer->stop || buffer->error) return 1; while (--len > 0) { - c = *buffer->curr++; + c = buffer->buffer[buffer->curr++]; if (c == '\n') { - if ((buffer->curr - buffer->buffer) >= (long)buffer->size) { + if (buffer->curr >= buffer->size) buffer->stop = 1; - } break; } *line++ = c; - if ((buffer->curr - buffer->buffer) >= (long)buffer->size) { + if (buffer->curr >= buffer->size) { buffer->stop = 1; break; } } while (c != '\n' && !buffer->stop) { - c = *buffer->curr++; - if ((buffer->curr - buffer->buffer) >= (long)buffer->size) { + c = buffer->buffer[buffer->curr++]; + if (buffer->curr >= buffer->size) buffer->stop = 1; - } } *line = '\0'; return 0; @@ -978,7 +988,6 @@ static int __init snd_info_version_init(void) entry = snd_info_create_module_entry(THIS_MODULE, "version", NULL); if (entry == NULL) return -ENOMEM; - entry->c.text.read_size = 256; entry->c.text.read = snd_info_version_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); -- cgit v1.2.2 From bf850204a71a97eb5a6afaf27263bb667f9cab0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:41 +0200 Subject: [ALSA] Remove unneeded read/write_size fields in proc text ops Remove unneeded read/write_size fields in proc text ops. snd_info_set_text_ops() is fixed, too. Signed-off-by: Takashi Iwai --- sound/core/hwdep.c | 1 - sound/core/info_oss.c | 1 - sound/core/init.c | 3 --- sound/core/oss/mixer_oss.c | 2 -- sound/core/oss/pcm_oss.c | 2 -- sound/core/pcm.c | 19 ++++++++++--------- sound/core/pcm_memory.c | 2 -- sound/core/rawmidi.c | 1 - sound/core/seq/oss/seq_oss.c | 1 - sound/core/seq/seq_device.c | 1 - sound/core/seq/seq_info.c | 11 +++++------ sound/core/sound.c | 1 - sound/core/sound_oss.c | 1 - sound/core/timer.c | 1 - sound/drivers/vx/vx_core.c | 2 +- sound/i2c/l3/uda1341.c | 4 ++-- sound/isa/gus/gus_irq.c | 2 +- sound/isa/gus/gus_mem.c | 6 ++---- sound/isa/opti9xx/miro.c | 2 +- sound/isa/sb/sb16_csp.c | 2 +- sound/pci/ac97/ac97_proc.c | 5 ++--- sound/pci/ac97/ak4531_codec.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/ca0106/ca0106_proc.c | 17 +++++++---------- sound/pci/cmipci.c | 2 +- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/dsp_spos.c | 7 ------- sound/pci/cs46xx/dsp_spos_scb_lib.c | 1 - sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 27 ++++++++++----------------- sound/pci/ens1370.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/ice1712/pontis.c | 8 +++----- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/mixart/mixart.c | 1 - sound/pci/pcxhr/pcxhr.c | 4 ++-- sound/pci/riptide/riptide.c | 2 +- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident_main.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 2 +- sound/sparc/dbri.c | 4 ++-- sound/synth/emux/emux_proc.c | 1 - sound/usb/usbaudio.c | 6 +++--- sound/usb/usbmixer.c | 2 +- 59 files changed, 80 insertions(+), 123 deletions(-) (limited to 'sound') diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 2524e66ecc..8bd0dcc93e 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -486,7 +486,6 @@ static void __init snd_hwdep_proc_init(void) struct snd_info_entry *entry; if ((entry = snd_info_create_module_entry(THIS_MODULE, "hwdep", NULL)) != NULL) { - entry->c.text.read_size = PAGE_SIZE; entry->c.text.read = snd_hwdep_proc_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index f2efca1872..bb2c40d0ab 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -119,7 +119,6 @@ int snd_info_minor_register(void) memset(snd_sndstat_strings, 0, sizeof(snd_sndstat_strings)); if ((entry = snd_info_create_module_entry(THIS_MODULE, "sndstat", snd_oss_root)) != NULL) { - entry->c.text.read_size = 2048; entry->c.text.read = snd_sndstat_proc_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/init.c b/sound/core/init.c index b145d17ba3..2ff0e5e908 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -69,7 +69,6 @@ static inline int init_info_for_card(struct snd_card *card) snd_printd("unable to create card entry\n"); return err; } - entry->c.text.read_size = PAGE_SIZE; entry->c.text.read = snd_card_id_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -592,7 +591,6 @@ int __init snd_card_info_init(void) entry = snd_info_create_module_entry(THIS_MODULE, "cards", NULL); if (! entry) return -ENOMEM; - entry->c.text.read_size = PAGE_SIZE; entry->c.text.read = snd_card_info_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -603,7 +601,6 @@ int __init snd_card_info_init(void) #ifdef MODULE entry = snd_info_create_module_entry(THIS_MODULE, "modules", NULL); if (entry) { - entry->c.text.read_size = PAGE_SIZE; entry->c.text.read = snd_card_module_info_read; if (snd_info_register(entry) < 0) snd_info_free_entry(entry); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 9c68bc3f97..71b5080fa6 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1182,9 +1182,7 @@ static void snd_mixer_oss_proc_init(struct snd_mixer_oss *mixer) return; entry->content = SNDRV_INFO_CONTENT_TEXT; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 8192; entry->c.text.read = snd_mixer_oss_proc_read; - entry->c.text.write_size = 8192; entry->c.text.write = snd_mixer_oss_proc_write; entry->private_data = mixer; if (snd_info_register(entry) < 0) { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 0d2e232afe..d8b7416ee0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2823,9 +2823,7 @@ static void snd_pcm_oss_proc_init(struct snd_pcm *pcm) if ((entry = snd_info_create_card_entry(pcm->card, "oss", pstr->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 8192; entry->c.text.read = snd_pcm_oss_proc_read; - entry->c.text.write_size = 8192; entry->c.text.write = snd_pcm_oss_proc_write; entry->private_data = pstr; if (snd_info_register(entry) < 0) { diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8c15c01907..08223783cf 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -472,7 +472,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) pstr->proc_root = entry; if ((entry = snd_info_create_card_entry(pcm->card, "info", pstr->proc_root)) != NULL) { - snd_info_set_text_ops(entry, pstr, 256, snd_pcm_stream_proc_info_read); + snd_info_set_text_ops(entry, pstr, snd_pcm_stream_proc_info_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -483,9 +483,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) #ifdef CONFIG_SND_PCM_XRUN_DEBUG if ((entry = snd_info_create_card_entry(pcm->card, "xrun_debug", pstr->proc_root)) != NULL) { - entry->c.text.read_size = 64; entry->c.text.read = snd_pcm_xrun_debug_read; - entry->c.text.write_size = 64; entry->c.text.write = snd_pcm_xrun_debug_write; entry->mode |= S_IWUSR; entry->private_data = pstr; @@ -537,7 +535,8 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->proc_root = entry; if ((entry = snd_info_create_card_entry(card, "info", substream->proc_root)) != NULL) { - snd_info_set_text_ops(entry, substream, 256, snd_pcm_substream_proc_info_read); + snd_info_set_text_ops(entry, substream, + snd_pcm_substream_proc_info_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -546,7 +545,8 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->proc_info_entry = entry; if ((entry = snd_info_create_card_entry(card, "hw_params", substream->proc_root)) != NULL) { - snd_info_set_text_ops(entry, substream, 256, snd_pcm_substream_proc_hw_params_read); + snd_info_set_text_ops(entry, substream, + snd_pcm_substream_proc_hw_params_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -555,7 +555,8 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->proc_hw_params_entry = entry; if ((entry = snd_info_create_card_entry(card, "sw_params", substream->proc_root)) != NULL) { - snd_info_set_text_ops(entry, substream, 256, snd_pcm_substream_proc_sw_params_read); + snd_info_set_text_ops(entry, substream, + snd_pcm_substream_proc_sw_params_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -564,7 +565,8 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->proc_sw_params_entry = entry; if ((entry = snd_info_create_card_entry(card, "status", substream->proc_root)) != NULL) { - snd_info_set_text_ops(entry, substream, 256, snd_pcm_substream_proc_status_read); + snd_info_set_text_ops(entry, substream, + snd_pcm_substream_proc_status_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -1062,8 +1064,7 @@ static void snd_pcm_proc_init(void) struct snd_info_entry *entry; if ((entry = snd_info_create_module_entry(THIS_MODULE, "pcm", NULL)) != NULL) { - snd_info_set_text_ops(entry, NULL, SNDRV_CARDS * SNDRV_PCM_DEVICES * 128, - snd_pcm_proc_read); + snd_info_set_text_ops(entry, NULL, snd_pcm_proc_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index eb56167d3b..067d2056db 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -193,9 +193,7 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) struct snd_info_entry *entry; if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc", substream->proc_root)) != NULL) { - entry->c.text.read_size = 64; entry->c.text.read = snd_pcm_lib_preallocate_proc_read; - entry->c.text.write_size = 64; entry->c.text.write = snd_pcm_lib_preallocate_proc_write; entry->mode |= S_IWUSR; entry->private_data = substream; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 87b47c9564..08a41e5023 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1561,7 +1561,6 @@ static int snd_rawmidi_dev_register(struct snd_device *device) entry = snd_info_create_card_entry(rmidi->card, name, rmidi->card->proc_root); if (entry) { entry->private_data = rmidi; - entry->c.text.read_size = 1024; entry->c.text.read = snd_rawmidi_proc_info_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index b991978518..e723413564 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -291,7 +291,6 @@ register_proc(void) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = NULL; - entry->c.text.read_size = 1024; entry->c.text.read = info_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index d9a3e5a18d..1e4bc402f0 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -555,7 +555,6 @@ static int __init alsa_seq_device_init(void) if (info_entry == NULL) return -ENOMEM; info_entry->content = SNDRV_INFO_CONTENT_TEXT; - info_entry->c.text.read_size = 2048; info_entry->c.text.read = snd_seq_device_info; if (snd_info_register(info_entry) < 0) { snd_info_free_entry(info_entry); diff --git a/sound/core/seq/seq_info.c b/sound/core/seq/seq_info.c index acce21afda..142e9e6882 100644 --- a/sound/core/seq/seq_info.c +++ b/sound/core/seq/seq_info.c @@ -34,8 +34,8 @@ static struct snd_info_entry *timer_entry; static struct snd_info_entry * __init -create_info_entry(char *name, int size, void (*read)(struct snd_info_entry *, - struct snd_info_buffer *)) +create_info_entry(char *name, void (*read)(struct snd_info_entry *, + struct snd_info_buffer *)) { struct snd_info_entry *entry; @@ -43,7 +43,6 @@ create_info_entry(char *name, int size, void (*read)(struct snd_info_entry *, if (entry == NULL) return NULL; entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->c.text.read_size = size; entry->c.text.read = read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -55,11 +54,11 @@ create_info_entry(char *name, int size, void (*read)(struct snd_info_entry *, /* create all our /proc entries */ int __init snd_seq_info_init(void) { - queues_entry = create_info_entry("queues", 512 + (256 * SNDRV_SEQ_MAX_QUEUES), + queues_entry = create_info_entry("queues", snd_seq_info_queues_read); - clients_entry = create_info_entry("clients", 512 + (256 * SNDRV_SEQ_MAX_CLIENTS), + clients_entry = create_info_entry("clients", snd_seq_info_clients_read); - timer_entry = create_info_entry("timer", 1024, snd_seq_info_timer_read); + timer_entry = create_info_entry("timer", snd_seq_info_timer_read); return 0; } diff --git a/sound/core/sound.c b/sound/core/sound.c index 67cfa06062..8313f97907 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -392,7 +392,6 @@ int __init snd_minor_info_init(void) entry = snd_info_create_module_entry(THIS_MODULE, "devices", NULL); if (entry) { - entry->c.text.read_size = PAGE_SIZE; entry->c.text.read = snd_minor_info_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index c18f6a45e4..0043c9a97d 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -258,7 +258,6 @@ int __init snd_minor_info_oss_init(void) entry = snd_info_create_module_entry(THIS_MODULE, "devices", snd_oss_root); if (entry) { - entry->c.text.read_size = PAGE_SIZE; entry->c.text.read = snd_minor_info_oss_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/timer.c b/sound/core/timer.c index cdeeb639b6..9a1e51c7c2 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1117,7 +1117,6 @@ static void __init snd_timer_proc_init(void) entry = snd_info_create_module_entry(THIS_MODULE, "timers", NULL); if (entry != NULL) { - entry->c.text.read_size = SNDRV_TIMER_DEVICES * 128; entry->c.text.read = snd_timer_proc_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index e1c3dda157..a60168268d 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -640,7 +640,7 @@ static void vx_proc_init(struct vx_core *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "vx-status", &entry)) - snd_info_set_text_ops(entry, chip, 1024, vx_proc_read); + snd_info_set_text_ops(entry, chip, vx_proc_read); } diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c index 746500e069..b074fdddea 100644 --- a/sound/i2c/l3/uda1341.c +++ b/sound/i2c/l3/uda1341.c @@ -517,9 +517,9 @@ static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_cli struct snd_info_entry *entry; if (! snd_card_proc_new(card, "uda1341", &entry)) - snd_info_set_text_ops(entry, clnt, 1024, snd_uda1341_proc_read); + snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read); if (! snd_card_proc_new(card, "uda1341-regs", &entry)) - snd_info_set_text_ops(entry, clnt, 1024, snd_uda1341_proc_regs_read); + snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read); } /* }}} */ diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index c19ba2910b..42db37552e 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -136,7 +136,7 @@ void snd_gus_irq_profile_init(struct snd_gus_card *gus) struct snd_info_entry *entry; if (! snd_card_proc_new(gus->card, "gusirq", &entry)) - snd_info_set_text_ops(entry, gus, 1024, snd_gus_irq_info_read); + snd_info_set_text_ops(entry, gus, snd_gus_irq_info_read); } #endif diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 3c0d27aa08..f50c276cae 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -264,10 +264,8 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) if (snd_gf1_mem_xalloc(alloc, &block) == NULL) return -ENOMEM; #ifdef CONFIG_SND_DEBUG - if (! snd_card_proc_new(gus->card, "gusmem", &entry)) { - snd_info_set_text_ops(entry, gus, 1024, snd_gf1_mem_info_read); - entry->c.text.read_size = 256 * 1024; - } + if (! snd_card_proc_new(gus->card, "gusmem", &entry)) + snd_info_set_text_ops(entry, gus, snd_gf1_mem_info_read); #endif return 0; } diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index e6bfcf74c1..283817f2de 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -967,7 +967,7 @@ static void __init snd_miro_proc_init(struct snd_miro * miro) struct snd_info_entry *entry; if (! snd_card_proc_new(miro->card, "miro", &entry)) - snd_info_set_text_ops(entry, miro, 1024, snd_miro_proc_read); + snd_info_set_text_ops(entry, miro, snd_miro_proc_read); } /* diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 9703c68e4e..fcd638090a 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -1101,7 +1101,7 @@ static int init_proc_entry(struct snd_sb_csp * p, int device) struct snd_info_entry *entry; sprintf(name, "cspD%d", device); if (! snd_card_proc_new(p->chip->card, name, &entry)) - snd_info_set_text_ops(entry, p, 1024, info_read); + snd_info_set_text_ops(entry, p, info_read); return 0; } diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 4d523df79c..2118df50b9 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -433,7 +433,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) prefix = ac97_is_audio(ac97) ? "ac97" : "mc97"; sprintf(name, "%s#%d-%d", prefix, ac97->addr, ac97->num); if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { - snd_info_set_text_ops(entry, ac97, 1024, snd_ac97_proc_read); + snd_info_set_text_ops(entry, ac97, snd_ac97_proc_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -442,10 +442,9 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) ac97->proc = entry; sprintf(name, "%s#%d-%d+regs", prefix, ac97->addr, ac97->num); if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { - snd_info_set_text_ops(entry, ac97, 1024, snd_ac97_proc_regs_read); + snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read); #ifdef CONFIG_SND_DEBUG entry->mode |= S_IWUSR; - entry->c.text.write_size = 1024; entry->c.text.write = snd_ac97_proc_regs_write; #endif if (snd_info_register(entry) < 0) { diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index 0fb7b34073..94c26ec058 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -453,7 +453,7 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453 struct snd_info_entry *entry; if (! snd_card_proc_new(card, "ak4531", &entry)) - snd_info_set_text_ops(entry, ak4531, 1024, snd_ak4531_proc_read); + snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); } #endif diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index eece1c7e55..d42bf45703 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -753,7 +753,7 @@ snd_ad1889_proc_init(struct snd_ad1889 *chip) struct snd_info_entry *entry; if (!snd_card_proc_new(chip->card, chip->card->driver, &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_ad1889_proc_read); + snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read); } static struct ac97_quirk ac97_quirks[] = { diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index e2dbc21189..4f01ef10fa 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2173,7 +2173,7 @@ static void __devinit snd_ali_proc_init(struct snd_ali *codec) { struct snd_info_entry *entry; if(!snd_card_proc_new(codec->card, "ali5451", &entry)) - snd_info_set_text_ops(entry, codec, 1024, snd_ali_proc_read); + snd_info_set_text_ops(entry, codec, snd_ali_proc_read); } static int __devinit snd_ali_resources(struct snd_ali *codec) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d0f759d86d..f18a8c0e46 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1504,7 +1504,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "atiixp", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_atiixp_proc_read); + snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read); } #else /* !CONFIG_PROC_FS */ #define snd_atiixp_proc_init(chip) diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 12a34c39ca..4073905707 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1177,7 +1177,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp_modem *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "atiixp-modem", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_atiixp_proc_read); + snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read); } #else #define snd_atiixp_proc_init(chip) diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 63757273bf..75ca421eb3 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -431,33 +431,30 @@ int __devinit snd_ca0106_proc_init(struct snd_ca0106 * emu) struct snd_info_entry *entry; if(! snd_card_proc_new(emu->card, "iec958", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_iec958); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958); if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; entry->mode |= S_IWUSR; } if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read8); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8); if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; entry->mode |= S_IWUSR; // entry->private_data = emu; } if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_i2c_write); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_i2c_write); entry->c.text.write = snd_ca0106_proc_i2c_write; entry->mode |= S_IWUSR; // entry->private_data = emu; } if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read2); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); return 0; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index e5ce2dabd0..42ca92be18 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2602,7 +2602,7 @@ static void __devinit snd_cmipci_proc_init(struct cmipci *cm) struct snd_info_entry *entry; if (! snd_card_proc_new(cm->card, "cmipci", &entry)) - snd_info_set_text_ops(entry, cm, 1024, snd_cmipci_proc_read); + snd_info_set_text_ops(entry, cm, snd_cmipci_proc_read); } #else /* !CONFIG_PROC_FS */ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index b3c94d8345..8c150eab45 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1184,7 +1184,7 @@ static void __devinit snd_cs4281_proc_init(struct cs4281 * chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "cs4281", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_cs4281_proc_read); + snd_info_set_text_ops(entry, chip, snd_cs4281_proc_read); if (! snd_card_proc_new(chip->card, "cs4281_BA0", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = chip; diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f407d2a5ce..5c9711c026 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -767,7 +767,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; - entry->c.text.read_size = 512; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -784,7 +783,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_symbol_table_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -797,7 +795,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_modules_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -810,7 +807,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -823,7 +819,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_sample_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -836,7 +831,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_task_tree_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -849,7 +843,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 1024; entry->c.text.read = cs46xx_dsp_proc_scb_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 2c4ee45fe1..3844d18af1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -267,7 +267,6 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, entry->private_data = scb_info; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_scb_info_read; if (snd_info_register(entry) < 0) { diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index d51290c181..0fb27e4be0 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1055,8 +1055,7 @@ static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu) struct snd_info_entry *entry; if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_emu10k1x_proc_reg_read); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; entry->mode |= S_IWUSR; entry->private_data = emu; diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 90f1c52703..b939e03aae 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -532,57 +532,51 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) struct snd_info_entry *entry; #ifdef CONFIG_SND_DEBUG if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_emu_proc_io_reg_read); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00a); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00b); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20a); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20b); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20c); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } #endif if (! snd_card_proc_new(emu->card, "emu10k1", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_read); if (emu->card_capabilities->emu10k2_chip) { if (! snd_card_proc_new(emu->card, "spdif-in", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_spdif_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_spdif_read); } if (emu->card_capabilities->ca0151_chip) { if (! snd_card_proc_new(emu->card, "capture-rates", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_rates_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_rates_read); } if (! snd_card_proc_new(emu->card, "voices", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_voices_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_voices_read); if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; @@ -616,7 +610,6 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; - entry->c.text.read_size = 128*1024; entry->c.text.read = snd_emu10k1_proc_acode_read; } return 0; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index ca9e34e88f..9d46bbee2a 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1915,7 +1915,7 @@ static void __devinit snd_ensoniq_proc_init(struct ensoniq * ensoniq) struct snd_info_entry *entry; if (! snd_card_proc_new(ensoniq->card, "audiopci", &entry)) - snd_info_set_text_ops(entry, ensoniq, 1024, snd_ensoniq_proc_read); + snd_info_set_text_ops(entry, ensoniq, snd_ensoniq_proc_read); } /* diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ca514a6a58..3db009990c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -318,7 +318,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) if (err < 0) return err; - snd_info_set_text_ops(entry, codec, 32 * 1024, print_codec_info); + snd_info_set_text_ops(entry, codec, print_codec_info); return 0; } diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 2821014b26..52de85e21b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1596,7 +1596,7 @@ static void __devinit snd_ice1712_proc_init(struct snd_ice1712 * ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "ice1712", &entry)) - snd_info_set_text_ops(entry, ice, 1024, snd_ice1712_proc_read); + snd_info_set_text_ops(entry, ice, snd_ice1712_proc_read); } /* diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index b1c007e022..1031bcbf70 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1293,7 +1293,7 @@ static void __devinit snd_vt1724_proc_init(struct snd_ice1712 * ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "ice1724", &entry)) - snd_info_set_text_ops(entry, ice, 1024, snd_vt1724_proc_read); + snd_info_set_text_ops(entry, ice, snd_vt1724_proc_read); } /* diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index d23fb3fc21..0efcad9260 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -680,9 +680,8 @@ static void wm_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { - snd_info_set_text_ops(entry, ice, 1024, wm_proc_regs_read); + snd_info_set_text_ops(entry, ice, wm_proc_regs_read); entry->mode |= S_IWUSR; - entry->c.text.write_size = 1024; entry->c.text.write = wm_proc_regs_write; } } @@ -705,9 +704,8 @@ static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff static void cs_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "cs_codec", &entry)) { - snd_info_set_text_ops(entry, ice, 1024, cs_proc_regs_read); - } + if (! snd_card_proc_new(ice->card, "cs_codec", &entry)) + snd_info_set_text_ops(entry, ice, cs_proc_regs_read); } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 0df7602568..a4e5b8115a 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2645,7 +2645,7 @@ static void __devinit snd_intel8x0_proc_init(struct intel8x0 * chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "intel8x0", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_intel8x0_proc_read); + snd_info_set_text_ops(entry, chip, snd_intel8x0_proc_read); } #else #define snd_intel8x0_proc_init(x) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 720635f0cb..20acb1a7e9 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1092,7 +1092,7 @@ static void __devinit snd_intel8x0m_proc_init(struct intel8x0m * chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "intel8x0m", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_intel8x0m_proc_read); + snd_info_set_text_ops(entry, chip, snd_intel8x0m_proc_read); } #else /* !CONFIG_PROC_FS */ #define snd_intel8x0m_proc_init(chip) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index e39fad1a42..6e97932de3 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2085,7 +2085,7 @@ static void __devinit snd_korg1212_proc_init(struct snd_korg1212 *korg1212) struct snd_info_entry *entry; if (! snd_card_proc_new(korg1212->card, "korg1212", &entry)) - snd_info_set_text_ops(entry, korg1212, 1024, snd_korg1212_proc_read); + snd_info_set_text_ops(entry, korg1212, snd_korg1212_proc_read); } static int diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 09cc078649..366c4a7e65 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1244,7 +1244,6 @@ static void __devinit snd_mixart_proc_init(struct snd_mixart *chip) /* text interface to read perf and temp meters */ if (! snd_card_proc_new(chip->card, "board_info", &entry)) { entry->private_data = chip; - entry->c.text.read_size = 1024; entry->c.text.read = snd_mixart_proc_read; } diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index dafa2235ab..8198884b51 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1150,9 +1150,9 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "info", &entry)) - snd_info_set_text_ops(entry, chip, 1024, pcxhr_proc_info); + snd_info_set_text_ops(entry, chip, pcxhr_proc_info); if (! snd_card_proc_new(chip->card, "sync", &entry)) - snd_info_set_text_ops(entry, chip, 1024, pcxhr_proc_sync); + snd_info_set_text_ops(entry, chip, pcxhr_proc_sync); } /* end of proc interface */ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index d8cc985d72..c27cd49997 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1992,7 +1992,7 @@ static void __devinit snd_riptide_proc_init(struct snd_riptide *chip) struct snd_info_entry *entry; if (!snd_card_proc_new(chip->card, "riptide", &entry)) - snd_info_set_text_ops(entry, chip, 4096, snd_riptide_proc_read); + snd_info_set_text_ops(entry, chip, snd_riptide_proc_read); } static int __devinit snd_riptide_mixer(struct snd_riptide *chip) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 55b1d4838d..4dd53bfe03 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1578,7 +1578,7 @@ static void __devinit snd_rme32_proc_init(struct rme32 * rme32) struct snd_info_entry *entry; if (! snd_card_proc_new(rme32->card, "rme32", &entry)) - snd_info_set_text_ops(entry, rme32, 1024, snd_rme32_proc_read); + snd_info_set_text_ops(entry, rme32, snd_rme32_proc_read); } /* diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 3c1bc533d5..75a8b754ef 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1805,7 +1805,7 @@ snd_rme96_proc_init(struct rme96 *rme96) struct snd_info_entry *entry; if (! snd_card_proc_new(rme96->card, "rme96", &entry)) - snd_info_set_text_ops(entry, rme96, 1024, snd_rme96_proc_read); + snd_info_set_text_ops(entry, rme96, snd_rme96_proc_read); } /* diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 61f82f0d5c..da63a1a199 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3470,7 +3470,7 @@ static void __devinit snd_hdsp_proc_init(struct hdsp *hdsp) struct snd_info_entry *entry; if (! snd_card_proc_new(hdsp->card, "hdsp", &entry)) - snd_info_set_text_ops(entry, hdsp, 1024, snd_hdsp_proc_read); + snd_info_set_text_ops(entry, hdsp, snd_hdsp_proc_read); } static void snd_hdsp_free_buffers(struct hdsp *hdsp) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 722b9e6ce5..bba1615504 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2489,7 +2489,7 @@ static void __devinit snd_hdspm_proc_init(struct hdspm * hdspm) struct snd_info_entry *entry; if (!snd_card_proc_new(hdspm->card, "hdspm", &entry)) - snd_info_set_text_ops(entry, hdspm, 1024, + snd_info_set_text_ops(entry, hdspm, snd_hdspm_proc_read); } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 75d6406303..ac14b2733f 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1787,7 +1787,7 @@ static void __devinit snd_rme9652_proc_init(struct snd_rme9652 *rme9652) struct snd_info_entry *entry; if (! snd_card_proc_new(rme9652->card, "rme9652", &entry)) - snd_info_set_text_ops(entry, rme9652, 1024, snd_rme9652_proc_read); + snd_info_set_text_ops(entry, rme9652, snd_rme9652_proc_read); } static void snd_rme9652_free_buffers(struct snd_rme9652 *rme9652) diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 91f8bf3ae9..a783041729 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1144,7 +1144,7 @@ static void __devinit snd_sonicvibes_proc_init(struct sonicvibes * sonic) struct snd_info_entry *entry; if (! snd_card_proc_new(sonic->card, "sonicvibes", &entry)) - snd_info_set_text_ops(entry, sonic, 1024, snd_sonicvibes_proc_read); + snd_info_set_text_ops(entry, sonic, snd_sonicvibes_proc_read); } /* diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 850579208e..d99ed72377 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3338,7 +3338,7 @@ static void __devinit snd_trident_proc_init(struct snd_trident * trident) if (trident->device == TRIDENT_DEVICE_ID_SI7018) s = "sis7018"; if (! snd_card_proc_new(trident->card, s, &entry)) - snd_info_set_text_ops(entry, trident, 1024, snd_trident_proc_read); + snd_info_set_text_ops(entry, trident, snd_trident_proc_read); } static int snd_trident_dev_free(struct snd_device *device) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 39daf62d2b..a1b777e79c 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2015,7 +2015,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "via82xx", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_via82xx_proc_read); + snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read); } /* diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index ef97e50cd6..577a2b0375 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -929,7 +929,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx_modem *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "via82xx", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_via82xx_proc_read); + snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read); } /* diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 8ac5ab50b5..f894752523 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1919,7 +1919,7 @@ static int __devinit snd_ymfpci_proc_init(struct snd_card *card, struct snd_ymfp struct snd_info_entry *entry; if (! snd_card_proc_new(card, "ymfpci", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_ymfpci_proc_read); + snd_info_set_text_ops(entry, chip, snd_ymfpci_proc_read); return 0; } diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index bd0d70ff30..1dfe29b863 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -144,7 +144,7 @@ static void pdacf_proc_init(struct snd_pdacf *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "pdaudiocf", &entry)) - snd_info_set_text_ops(entry, chip, 1024, pdacf_proc_read); + snd_info_set_text_ops(entry, chip, pdacf_proc_read); } struct snd_pdacf *snd_pdacf_create(struct snd_card *card) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index e622d08215..db6539126d 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2521,11 +2521,11 @@ void snd_dbri_proc(struct snd_dbri * dbri) struct snd_info_entry *entry; if (! snd_card_proc_new(dbri->card, "regs", &entry)) - snd_info_set_text_ops(entry, dbri, 1024, dbri_regs_read); + snd_info_set_text_ops(entry, dbri, dbri_regs_read); #ifdef DBRI_DEBUG if (! snd_card_proc_new(dbri->card, "debug", &entry)) { - snd_info_set_text_ops(entry, dbri, 4096, dbri_debug_read); + snd_info_set_text_ops(entry, dbri, dbri_debug_read); entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ } #endif diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index 1ba68ce302..58b9601f3a 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -119,7 +119,6 @@ void snd_emux_proc_init(struct snd_emux *emu, struct snd_card *card, int device) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; - entry->c.text.read_size = 1024; entry->c.text.read = snd_emux_proc_info_read; if (snd_info_register(entry) < 0) snd_info_free_entry(entry); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4e614ac39f..8100516e1f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2138,7 +2138,7 @@ static void proc_pcm_format_add(struct snd_usb_stream *stream) sprintf(name, "stream%d", stream->pcm_index); if (! snd_card_proc_new(card, name, &entry)) - snd_info_set_text_ops(entry, stream, 1024, proc_pcm_format_read); + snd_info_set_text_ops(entry, stream, proc_pcm_format_read); } #else @@ -3197,9 +3197,9 @@ static void snd_usb_audio_create_proc(struct snd_usb_audio *chip) { struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "usbbus", &entry)) - snd_info_set_text_ops(entry, chip, 1024, proc_audio_usbbus_read); + snd_info_set_text_ops(entry, chip, proc_audio_usbbus_read); if (! snd_card_proc_new(chip->card, "usbid", &entry)) - snd_info_set_text_ops(entry, chip, 1024, proc_audio_usbid_read); + snd_info_set_text_ops(entry, chip, proc_audio_usbid_read); } /* diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ce86283ee0..ab921aa9d7 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1998,7 +1998,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif) if ((err = snd_audigy2nx_controls_create(mixer)) < 0) goto _error; if (!snd_card_proc_new(chip->card, "audigy2nx", &entry)) - snd_info_set_text_ops(entry, mixer, 1024, + snd_info_set_text_ops(entry, mixer, snd_audigy2nx_proc_read); } -- cgit v1.2.2 From d689e34b524b69c111db0b7c844d71c8e1a53b15 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:41 +0200 Subject: [ALSA] Remove spinlocks around proc prints Don't lock during showing proc read. snd_iprintf() might sleep. Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 6 ------ sound/core/timer.c | 2 -- 2 files changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 08223783cf..4f5204175d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -351,10 +351,8 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "closed\n"); return; } - snd_pcm_stream_lock_irq(substream); if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - snd_pcm_stream_unlock_irq(substream); return; } snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access)); @@ -375,7 +373,6 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames); } #endif - snd_pcm_stream_unlock_irq(substream); } static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, @@ -387,10 +384,8 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "closed\n"); return; } - snd_pcm_stream_lock_irq(substream); if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - snd_pcm_stream_unlock_irq(substream); return; } snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode)); @@ -403,7 +398,6 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold); snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size); snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary); - snd_pcm_stream_unlock_irq(substream); } static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, diff --git a/sound/core/timer.c b/sound/core/timer.c index 9a1e51c7c2..e37eab7457 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1095,7 +1095,6 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) snd_iprintf(buffer, " SLAVE"); snd_iprintf(buffer, "\n"); - spin_lock_irqsave(&timer->lock, flags); list_for_each(q, &timer->open_list_head) { ti = list_entry(q, struct snd_timer_instance, open_list); snd_iprintf(buffer, " Client %s : %s\n", @@ -1104,7 +1103,6 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, SNDRV_TIMER_IFLG_RUNNING) ? "running" : "stopped"); } - spin_unlock_irqrestore(&timer->lock, flags); } mutex_unlock(®ister_mutex); } -- cgit v1.2.2 From f001c3acf64b8ca18fe40af592629abb261b321e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:41 +0200 Subject: [ALSA] Insert might_sleep() in snd_iprintf() Inserted might_sleep() in snd_iprintf() for sanity check. Signed-off-by: Takashi Iwai --- sound/core/info.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index 86366839c4..4188f76add 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -114,6 +114,7 @@ int snd_iprintf(struct snd_info_buffer *buffer, char *fmt,...) int len, res; int err = 0; + might_sleep(); if (buffer->stop || buffer->error) return 0; len = buffer->len - buffer->size; -- cgit v1.2.2 From 0df63e44c3e315ec0fe427ae62558231864108bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:41 +0200 Subject: [ALSA] Add O_APPEND flag support to PCM Added O_APPEND flag support to PCM to enable shared substreams among multiple processes. This mechanism is used by dmix and dsnoop plugins. Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 17 +++++---- sound/core/pcm.c | 25 ++++++++++++- sound/core/pcm_compat.c | 4 +- sound/core/pcm_lib.c | 8 ++-- sound/core/pcm_native.c | 81 ++++++++++++++++++++++++++++------------- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 6 files changed, 95 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d8b7416ee0..9803a6ce3d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1331,7 +1331,7 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if (runtime->oss.period_ptr == 0 || runtime->oss.period_ptr == runtime->oss.buffer_used) runtime->oss.buffer_used = 0; - else if ((substream->ffile->f_flags & O_NONBLOCK) != 0) + else if ((substream->f_flags & O_NONBLOCK) != 0) return xfer > 0 ? xfer : -EAGAIN; } } else { @@ -1344,7 +1344,7 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha buf += tmp; bytes -= tmp; xfer += tmp; - if ((substream->ffile->f_flags & O_NONBLOCK) != 0 && + if ((substream->f_flags & O_NONBLOCK) != 0 && tmp != runtime->oss.period_bytes) break; } @@ -1582,10 +1582,10 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) * finish sync: drain the buffer */ __direct: - saved_f_flags = substream->ffile->f_flags; - substream->ffile->f_flags &= ~O_NONBLOCK; + saved_f_flags = substream->f_flags; + substream->f_flags &= ~O_NONBLOCK; err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); - substream->ffile->f_flags = saved_f_flags; + substream->f_flags = saved_f_flags; if (err < 0) return err; runtime->oss.prepare = 1; @@ -2164,9 +2164,9 @@ static void snd_pcm_oss_init_substream(struct snd_pcm_substream *substream, substream->oss.oss = 1; substream->oss.setup = *setup; if (setup->nonblock) - substream->ffile->f_flags |= O_NONBLOCK; + substream->f_flags |= O_NONBLOCK; else if (setup->block) - substream->ffile->f_flags &= ~O_NONBLOCK; + substream->f_flags &= ~O_NONBLOCK; runtime = substream->runtime; runtime->oss.params = 1; runtime->oss.trigger = 1; @@ -2223,6 +2223,7 @@ static int snd_pcm_oss_open_file(struct file *file, (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX)) f_mode = FMODE_WRITE; + file->f_flags &= ~O_APPEND; for (idx = 0; idx < 2; idx++) { if (setup[idx].disable) continue; @@ -2540,6 +2541,7 @@ static ssize_t snd_pcm_oss_read(struct file *file, char __user *buf, size_t coun substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; if (substream == NULL) return -ENXIO; + substream->f_flags = file->f_flags & O_NONBLOCK; #ifndef OSS_DEBUG return snd_pcm_oss_read1(substream, buf, count); #else @@ -2561,6 +2563,7 @@ static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream == NULL) return -ENXIO; + substream->f_flags = file->f_flags & O_NONBLOCK; result = snd_pcm_oss_write1(substream, buf, count); #ifdef OSS_DEBUG printk("pcm_oss: write %li bytes (wrote %li bytes)\n", (long)count, (long)result); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 4f5204175d..8136be2e60 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -829,6 +829,26 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return -EINVAL; } + if (file->f_flags & O_APPEND) { + if (prefer_subdevice < 0) { + if (pstr->substream_count > 1) + return -EINVAL; /* must be unique */ + substream = pstr->substream; + } else { + for (substream = pstr->substream; substream; + substream = substream->next) + if (substream->number == prefer_subdevice) + break; + } + if (! substream) + return -ENODEV; + if (! SUBSTREAM_BUSY(substream)) + return -EBADFD; + substream->ref_count++; + *rsubstream = substream; + return 0; + } + if (prefer_subdevice >= 0) { for (substream = pstr->substream; substream; substream = substream->next) if (!SUBSTREAM_BUSY(substream) && substream->number == prefer_subdevice) @@ -873,7 +893,8 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->runtime = runtime; substream->private_data = pcm->private_data; - substream->ffile = file; + substream->ref_count = 1; + substream->f_flags = file->f_flags; pstr->substream_opened++; *rsubstream = substream; return 0; @@ -882,7 +903,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, void snd_pcm_detach_substream(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; - substream->file = NULL; + runtime = substream->runtime; snd_assert(runtime != NULL, return); if (runtime->private_free != NULL) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index e5133033de..2b8aab6fd6 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -497,9 +497,9 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_LINK: case SNDRV_PCM_IOCTL_UNLINK: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return snd_pcm_playback_ioctl1(substream, cmd, argp); + return snd_pcm_playback_ioctl1(file, substream, cmd, argp); else - return snd_pcm_capture_ioctl1(substream, cmd, argp); + return snd_pcm_capture_ioctl1(file, substream, cmd, argp); case SNDRV_PCM_IOCTL_HW_REFINE32: return snd_pcm_ioctl_hw_params_compat(substream, 1, argp); case SNDRV_PCM_IOCTL_HW_PARAMS32: diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a21aa0050e..0bb142a285 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1782,7 +1782,7 @@ snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, const v if (runtime->status->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; - nonblock = !!(substream->ffile->f_flags & O_NONBLOCK); + nonblock = !!(substream->f_flags & O_NONBLOCK); if (runtime->access != SNDRV_PCM_ACCESS_RW_INTERLEAVED && runtime->channels > 1) @@ -1847,7 +1847,7 @@ snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream, if (runtime->status->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; - nonblock = !!(substream->ffile->f_flags & O_NONBLOCK); + nonblock = !!(substream->f_flags & O_NONBLOCK); if (runtime->access != SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) return -EINVAL; @@ -2059,7 +2059,7 @@ snd_pcm_sframes_t snd_pcm_lib_read(struct snd_pcm_substream *substream, void __u if (runtime->status->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; - nonblock = !!(substream->ffile->f_flags & O_NONBLOCK); + nonblock = !!(substream->f_flags & O_NONBLOCK); if (runtime->access != SNDRV_PCM_ACCESS_RW_INTERLEAVED) return -EINVAL; return snd_pcm_lib_read1(substream, (unsigned long)buf, size, nonblock, snd_pcm_lib_read_transfer); @@ -2118,7 +2118,7 @@ snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream, if (runtime->status->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; - nonblock = !!(substream->ffile->f_flags & O_NONBLOCK); + nonblock = !!(substream->f_flags & O_NONBLOCK); if (runtime->access != SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) return -EINVAL; return snd_pcm_lib_read1(substream, (unsigned long)bufs, frames, nonblock, snd_pcm_lib_readv_transfer); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7b5729c4b2..36d6765618 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1284,13 +1284,16 @@ static int snd_pcm_reset(struct snd_pcm_substream *substream) /* * prepare ioctl */ -static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, int state) +/* we use the second argument for updating f_flags */ +static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, + int f_flags) { struct snd_pcm_runtime *runtime = substream->runtime; if (runtime->status->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; if (snd_pcm_running(substream)) return -EBUSY; + substream->f_flags = f_flags; return 0; } @@ -1319,17 +1322,26 @@ static struct action_ops snd_pcm_action_prepare = { /** * snd_pcm_prepare * @substream: the PCM substream instance + * @file: file to refer f_flags * * Prepare the PCM substream to be triggerable. */ -static int snd_pcm_prepare(struct snd_pcm_substream *substream) +static int snd_pcm_prepare(struct snd_pcm_substream *substream, + struct file *file) { int res; struct snd_card *card = substream->pcm->card; + int f_flags; + + if (file) + f_flags = file->f_flags; + else + f_flags = substream->f_flags; snd_power_lock(card); if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0) - res = snd_pcm_action_nonatomic(&snd_pcm_action_prepare, substream, 0); + res = snd_pcm_action_nonatomic(&snd_pcm_action_prepare, + substream, f_flags); snd_power_unlock(card); return res; } @@ -1340,7 +1352,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream) static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { - if (substream->ffile->f_flags & O_NONBLOCK) + if (substream->f_flags & O_NONBLOCK) return -EAGAIN; substream->runtime->trigger_master = substream; return 0; @@ -2015,6 +2027,10 @@ static void pcm_release_private(struct snd_pcm_substream *substream) void snd_pcm_release_substream(struct snd_pcm_substream *substream) { + substream->ref_count--; + if (substream->ref_count > 0) + return; + snd_pcm_drop(substream); if (substream->hw_opened) { if (substream->ops->hw_free != NULL) @@ -2041,6 +2057,11 @@ int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, err = snd_pcm_attach_substream(pcm, stream, file, &substream); if (err < 0) return err; + if (substream->ref_count > 1) { + *rsubstream = substream; + return 0; + } + substream->no_mmap_ctrl = 0; err = snd_pcm_hw_constraints_init(substream); if (err < 0) { @@ -2086,17 +2107,20 @@ static int snd_pcm_open_file(struct file *file, if (err < 0) return err; - pcm_file = kzalloc(sizeof(*pcm_file), GFP_KERNEL); - if (pcm_file == NULL) { - snd_pcm_release_substream(substream); - return -ENOMEM; + if (substream->ref_count > 1) + pcm_file = substream->file; + else { + pcm_file = kzalloc(sizeof(*pcm_file), GFP_KERNEL); + if (pcm_file == NULL) { + snd_pcm_release_substream(substream); + return -ENOMEM; + } + str = substream->pstr; + substream->file = pcm_file; + substream->pcm_release = pcm_release_private; + pcm_file->substream = substream; + snd_pcm_add_file(str, pcm_file); } - str = substream->pstr; - substream->file = pcm_file; - substream->pcm_release = pcm_release_private; - pcm_file->substream = substream; - snd_pcm_add_file(str, pcm_file); - file->private_data = pcm_file; *rpcm_file = pcm_file; return 0; @@ -2506,7 +2530,8 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_common_ioctl1(struct snd_pcm_substream *substream, +static int snd_pcm_common_ioctl1(struct file *file, + struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) { snd_assert(substream != NULL, return -ENXIO); @@ -2531,7 +2556,7 @@ static int snd_pcm_common_ioctl1(struct snd_pcm_substream *substream, case SNDRV_PCM_IOCTL_CHANNEL_INFO: return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: - return snd_pcm_prepare(substream); + return snd_pcm_prepare(substream, file); case SNDRV_PCM_IOCTL_RESET: return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START: @@ -2573,7 +2598,8 @@ static int snd_pcm_common_ioctl1(struct snd_pcm_substream *substream, return -ENOTTY; } -static int snd_pcm_playback_ioctl1(struct snd_pcm_substream *substream, +static int snd_pcm_playback_ioctl1(struct file *file, + struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) { snd_assert(substream != NULL, return -ENXIO); @@ -2649,10 +2675,11 @@ static int snd_pcm_playback_ioctl1(struct snd_pcm_substream *substream, return result < 0 ? result : 0; } } - return snd_pcm_common_ioctl1(substream, cmd, arg); + return snd_pcm_common_ioctl1(file, substream, cmd, arg); } -static int snd_pcm_capture_ioctl1(struct snd_pcm_substream *substream, +static int snd_pcm_capture_ioctl1(struct file *file, + struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) { snd_assert(substream != NULL, return -ENXIO); @@ -2728,7 +2755,7 @@ static int snd_pcm_capture_ioctl1(struct snd_pcm_substream *substream, return result < 0 ? result : 0; } } - return snd_pcm_common_ioctl1(substream, cmd, arg); + return snd_pcm_common_ioctl1(file, substream, cmd, arg); } static long snd_pcm_playback_ioctl(struct file *file, unsigned int cmd, @@ -2741,7 +2768,8 @@ static long snd_pcm_playback_ioctl(struct file *file, unsigned int cmd, if (((cmd >> 8) & 0xff) != 'A') return -ENOTTY; - return snd_pcm_playback_ioctl1(pcm_file->substream, cmd, (void __user *)arg); + return snd_pcm_playback_ioctl1(file, pcm_file->substream, cmd, + (void __user *)arg); } static long snd_pcm_capture_ioctl(struct file *file, unsigned int cmd, @@ -2754,7 +2782,8 @@ static long snd_pcm_capture_ioctl(struct file *file, unsigned int cmd, if (((cmd >> 8) & 0xff) != 'A') return -ENOTTY; - return snd_pcm_capture_ioctl1(pcm_file->substream, cmd, (void __user *)arg); + return snd_pcm_capture_ioctl1(file, pcm_file->substream, cmd, + (void __user *)arg); } int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, @@ -2766,12 +2795,12 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, fs = snd_enter_user(); switch (substream->stream) { case SNDRV_PCM_STREAM_PLAYBACK: - result = snd_pcm_playback_ioctl1(substream, - cmd, (void __user *)arg); + result = snd_pcm_playback_ioctl1(NULL, substream, cmd, + (void __user *)arg); break; case SNDRV_PCM_STREAM_CAPTURE: - result = snd_pcm_capture_ioctl1(substream, - cmd, (void __user *)arg); + result = snd_pcm_capture_ioctl1(NULL, substream, cmd, + (void __user *)arg); break; default: result = -EINVAL; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index fe67a92e2a..88b72b5259 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -632,7 +632,7 @@ static int usX2Y_pcms_lock_check(struct snd_card *card) for (s = 0; s < 2; ++s) { struct snd_pcm_substream *substream; substream = pcm->streams[s].substream; - if (substream && substream->ffile != NULL) + if (SUBSTREAM_BUSY(substream)) err = -EBUSY; } } -- cgit v1.2.2 From 9c323fcbc51493f79f9700cb20830d0857c72d99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:13:41 +0200 Subject: [ALSA] Fix mmap_count with O_APPEND opened streams Move mmap_count to snd_pcm_substream instead of runtime struct so that multiplly opened substreams via O_APPEND can be handled correctly. Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 27 ++++++++++++++------------- sound/core/pcm.c | 2 +- sound/core/pcm_native.c | 10 +++++----- 3 files changed, 20 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 9803a6ce3d..4395285aa6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -684,7 +684,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_buffer_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; oss_buffer_size = 1 << ld2(oss_buffer_size); - if (atomic_read(&runtime->mmap_count)) { + if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) oss_buffer_size = runtime->oss.mmap_bytes; } @@ -819,7 +819,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) goto failure; } - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) direct = 1; else direct = substream->oss.setup.direct; @@ -828,7 +828,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) _snd_pcm_hw_param_setinteger(sparams, SNDRV_PCM_HW_PARAM_PERIODS); _snd_pcm_hw_param_min(sparams, SNDRV_PCM_HW_PARAM_PERIODS, 2, 0); snd_mask_none(&mask); - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) snd_mask_set(&mask, SNDRV_PCM_ACCESS_MMAP_INTERLEAVED); else { snd_mask_set(&mask, SNDRV_PCM_ACCESS_RW_INTERLEAVED); @@ -947,7 +947,8 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) } else { sw_params->start_threshold = runtime->boundary; } - if (atomic_read(&runtime->mmap_count) || substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (atomic_read(&substream->mmap_count) || + substream->stream == SNDRV_PCM_STREAM_CAPTURE) sw_params->stop_threshold = runtime->boundary; else sw_params->stop_threshold = runtime->buffer_size; @@ -957,7 +958,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : runtime->period_size; sw_params->xfer_align = 1; - if (atomic_read(&runtime->mmap_count) || + if (atomic_read(&substream->mmap_count) || substream->oss.setup.nosilence) { sw_params->silence_threshold = 0; sw_params->silence_size = 0; @@ -1301,7 +1302,7 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha ssize_t tmp; struct snd_pcm_runtime *runtime = substream->runtime; - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) return -ENXIO; if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) @@ -1391,7 +1392,7 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use ssize_t tmp; struct snd_pcm_runtime *runtime = substream->runtime; - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) return -ENXIO; if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) @@ -1521,7 +1522,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream != NULL) { runtime = substream->runtime; - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) goto __direct; if ((err = snd_pcm_oss_make_ready(substream)) < 0) return err; @@ -1690,7 +1691,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) if ((err = snd_pcm_oss_get_active_substream(pcm_oss_file, &substream)) < 0) return err; - if (atomic_read(&substream->runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) direct = 1; else direct = substream->oss.setup.direct; @@ -1900,7 +1901,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr if (trigger & PCM_ENABLE_OUTPUT) { if (runtime->oss.trigger) goto _skip1; - if (atomic_read(&psubstream->runtime->mmap_count)) + if (atomic_read(&psubstream->mmap_count)) snd_pcm_oss_simulate_fill(psubstream, runtime->hw_ptr_interrupt); runtime->oss.trigger = 1; runtime->start_threshold = 1; @@ -2018,7 +2019,7 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream if (err < 0) return err; info.ptr = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr % runtime->buffer_size); - if (atomic_read(&runtime->mmap_count)) { + if (atomic_read(&substream->mmap_count)) { snd_pcm_sframes_t n; n = (delay = runtime->hw_ptr_interrupt) - runtime->oss.prev_hw_ptr_interrupt; if (n < 0) @@ -2574,7 +2575,7 @@ static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; else return snd_pcm_playback_avail(runtime) >= runtime->oss.period_frames; @@ -2583,7 +2584,7 @@ static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream) static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; else return snd_pcm_capture_avail(runtime) >= runtime->oss.period_frames; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8136be2e60..bc00f9b00c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -662,6 +662,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) INIT_LIST_HEAD(&substream->self_group.substreams); list_add_tail(&substream->link_list, &substream->self_group.substreams); spin_lock_init(&substream->timer_lock); + atomic_set(&substream->mmap_count, 0); prev = substream; } return 0; @@ -884,7 +885,6 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, memset((void*)runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); - atomic_set(&runtime->mmap_count, 0); init_timer(&runtime->tick_timer); runtime->tick_timer.function = snd_pcm_tick_timer_func; runtime->tick_timer.data = (unsigned long) substream; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 36d6765618..a998f88e3f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -372,7 +372,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) if (!substream->oss.oss) #endif - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) return -EBADFD; params->rmask = ~0U; @@ -485,7 +485,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) return -EBADFD; } snd_pcm_stream_unlock_irq(substream); - if (atomic_read(&runtime->mmap_count)) + if (atomic_read(&substream->mmap_count)) return -EBADFD; if (substream->ops->hw_free) result = substream->ops->hw_free(substream); @@ -2207,7 +2207,7 @@ static int snd_pcm_release(struct inode *inode, struct file *file) pcm_file = file->private_data; substream = pcm_file->substream; snd_assert(substream != NULL, return -ENXIO); - snd_assert(!atomic_read(&substream->runtime->mmap_count), ); + snd_assert(!atomic_read(&substream->mmap_count), ); pcm = substream->pcm; fasync_helper(-1, file, 0, &substream->runtime->fasync); mutex_lock(&pcm->open_mutex); @@ -3178,7 +3178,7 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, area->vm_ops = &snd_pcm_vm_ops_data; area->vm_private_data = substream; area->vm_flags |= VM_RESERVED; - atomic_inc(&substream->runtime->mmap_count); + atomic_inc(&substream->mmap_count); return 0; } @@ -3210,7 +3210,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, size, area->vm_page_prot)) return -EAGAIN; - atomic_inc(&substream->runtime->mmap_count); + atomic_inc(&substream->mmap_count); return 0; } -- cgit v1.2.2 From 170a34605c14a90df5f4a78e0b4ca643be6ef8ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Apr 2006 15:29:15 +0200 Subject: [ALSA] Fix compile warning in timer.c Fix a compile warning in timer.c due to unused variables. Signed-off-by: Takashi Iwai --- sound/core/timer.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index e37eab7457..d92f73c2c6 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1061,7 +1061,6 @@ static int snd_timer_register_system(void) static void snd_timer_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - unsigned long flags; struct snd_timer *timer; struct snd_timer_instance *ti; struct list_head *p, *q; -- cgit v1.2.2 From f4a747f155fe375231196ec3d26fcb6e3675d82f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 May 2006 15:33:25 +0200 Subject: [ALSA] fix a wrong lock fix a typo in the info locking code Signed-off-by: Clemens Ladisch --- sound/core/info.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index 4188f76add..c8eeaea9d6 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -267,7 +267,7 @@ static ssize_t snd_info_entry_write(struct file *file, const char __user *buffer buf = data->wbuffer; if (buf == NULL) return -EIO; - mutex_unlock(&entry->access); + mutex_lock(&entry->access); if (pos + count >= buf->len) { if (resize_info_buffer(buf, pos + count)) { mutex_unlock(&entry->access); -- cgit v1.2.2 From c97f3dd85490e51ba48782dd0c063cdade352c0d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 May 2006 15:50:05 +0200 Subject: [ALSA] fix port type bits Fix the port information about non-MIDI messages that had wrong values for some OPL3 and EmuX ports. Signed-off-by: Clemens Ladisch --- sound/drivers/opl3/opl3_oss.c | 3 +-- sound/drivers/opl3/opl3_seq.c | 2 +- sound/synth/emux/emux_seq.c | 3 +-- 3 files changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index fccf019a6d..d48f8dee2d 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -99,8 +99,7 @@ static int snd_opl3_oss_create_port(struct snd_opl3 * opl3) opl3->oss_chset->port = snd_seq_event_port_attach(opl3->seq_client, &callbacks, SNDRV_SEQ_PORT_CAP_WRITE, SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | - SNDRV_SEQ_PORT_TYPE_MIDI_GM | - SNDRV_SEQ_PORT_TYPE_SYNTH, + SNDRV_SEQ_PORT_TYPE_MIDI_GM, voices, voices, name); if (opl3->oss_chset->port < 0) { diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index 57becf34f4..2aece1b186 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -203,7 +203,7 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) SNDRV_SEQ_PORT_CAP_SUBS_WRITE, SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | SNDRV_SEQ_PORT_TYPE_MIDI_GM | - SNDRV_SEQ_PORT_TYPE_SYNTH, + SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE, 16, voices, name); if (opl3->chset->port < 0) { diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 8f00f07701..58838f7c95 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -54,8 +54,7 @@ static struct snd_midi_op emux_ops = { #define DEFAULT_MIDI_TYPE (SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |\ SNDRV_SEQ_PORT_TYPE_MIDI_GM |\ SNDRV_SEQ_PORT_TYPE_MIDI_GS |\ - SNDRV_SEQ_PORT_TYPE_MIDI_XG |\ - SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE) + SNDRV_SEQ_PORT_TYPE_MIDI_XG) /* * Initialise the EMUX Synth by creating a client and registering -- cgit v1.2.2 From 450047a78f3c35a905576e121abfbee2ccd45993 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 May 2006 16:08:41 +0200 Subject: [ALSA] add more sequencer port type information bits Add four new information flags SNDRV_SEQ_PORT_TYPE_HARDWARE, _SOFTWARE, _SYNTHESIZER, _PORT for sequencer ports. This makes it easier for apps like Rosegarden to make policy decisions based on the port type. Signed-off-by: Clemens Ladisch --- sound/core/seq/seq_dummy.c | 4 +++- sound/core/seq/seq_midi.c | 4 +++- sound/core/seq/seq_virmidi.c | 4 +++- sound/drivers/opl3/opl3_oss.c | 4 +++- sound/drivers/opl3/opl3_seq.c | 4 +++- sound/drivers/opl4/opl4_seq.c | 4 +++- sound/isa/gus/gus_synth.c | 4 +++- sound/pci/trident/trident_synth.c | 4 +++- sound/synth/emux/emux_seq.c | 4 +++- 9 files changed, 27 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index 2a283a59ea..9eb1c744f7 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -171,7 +171,9 @@ create_port(int idx, int type) pinfo.capability |= SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE; if (duplex) pinfo.capability |= SNDRV_SEQ_PORT_CAP_DUPLEX; - pinfo.type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC; + pinfo.type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC + | SNDRV_SEQ_PORT_TYPE_SOFTWARE + | SNDRV_SEQ_PORT_TYPE_PORT; memset(&pcb, 0, sizeof(pcb)); pcb.owner = THIS_MODULE; pcb.unuse = dummy_unuse; diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 3b316da25e..f873742c65 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -376,7 +376,9 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) if ((port->capability & (SNDRV_SEQ_PORT_CAP_WRITE|SNDRV_SEQ_PORT_CAP_READ)) == (SNDRV_SEQ_PORT_CAP_WRITE|SNDRV_SEQ_PORT_CAP_READ) && info->flags & SNDRV_RAWMIDI_INFO_DUPLEX) port->capability |= SNDRV_SEQ_PORT_CAP_DUPLEX; - port->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC; + port->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC + | SNDRV_SEQ_PORT_TYPE_HARDWARE + | SNDRV_SEQ_PORT_TYPE_PORT; port->midi_channels = 16; memset(&pcallbacks, 0, sizeof(pcallbacks)); pcallbacks.owner = THIS_MODULE; diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index f4edec603b..0cfa06c6b8 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -390,7 +390,9 @@ static int snd_virmidi_dev_attach_seq(struct snd_virmidi_dev *rdev) pinfo->capability |= SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SYNC_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE; pinfo->capability |= SNDRV_SEQ_PORT_CAP_READ | SNDRV_SEQ_PORT_CAP_SYNC_READ | SNDRV_SEQ_PORT_CAP_SUBS_READ; pinfo->capability |= SNDRV_SEQ_PORT_CAP_DUPLEX; - pinfo->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC; + pinfo->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC + | SNDRV_SEQ_PORT_TYPE_SOFTWARE + | SNDRV_SEQ_PORT_TYPE_PORT; pinfo->midi_channels = 16; memset(&pcallbacks, 0, sizeof(pcallbacks)); pcallbacks.owner = THIS_MODULE; diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index d48f8dee2d..5fd3a4c956 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -99,7 +99,9 @@ static int snd_opl3_oss_create_port(struct snd_opl3 * opl3) opl3->oss_chset->port = snd_seq_event_port_attach(opl3->seq_client, &callbacks, SNDRV_SEQ_PORT_CAP_WRITE, SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | - SNDRV_SEQ_PORT_TYPE_MIDI_GM, + SNDRV_SEQ_PORT_TYPE_MIDI_GM | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, voices, voices, name); if (opl3->oss_chset->port < 0) { diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index 2aece1b186..96762c9d48 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -203,7 +203,9 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) SNDRV_SEQ_PORT_CAP_SUBS_WRITE, SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | SNDRV_SEQ_PORT_TYPE_MIDI_GM | - SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE, + SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, 16, voices, name); if (opl3->chset->port < 0) { diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index dc0dcdc6c3..43d8a2bdd2 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -164,7 +164,9 @@ static int snd_opl4_seq_new_device(struct snd_seq_device *dev) SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE, SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | - SNDRV_SEQ_PORT_TYPE_MIDI_GM, + SNDRV_SEQ_PORT_TYPE_MIDI_GM | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, 16, 24, "OPL4 Wavetable Port"); if (opl4->chset->port < 0) { diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c index 2767cc187a..3e4d4d6edd 100644 --- a/sound/isa/gus/gus_synth.c +++ b/sound/isa/gus/gus_synth.c @@ -194,7 +194,9 @@ static int snd_gus_synth_create_port(struct snd_gus_card * gus, int idx) &callbacks, SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE, SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE | - SNDRV_SEQ_PORT_TYPE_SYNTH, + SNDRV_SEQ_PORT_TYPE_SYNTH | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, 16, 0, name); if (p->chset->port < 0) { diff --git a/sound/pci/trident/trident_synth.c b/sound/pci/trident/trident_synth.c index cc7af8bc55..9b7dee8474 100644 --- a/sound/pci/trident/trident_synth.c +++ b/sound/pci/trident/trident_synth.c @@ -914,7 +914,9 @@ static int snd_trident_synth_create_port(struct snd_trident * trident, int idx) &callbacks, SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE, SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE | - SNDRV_SEQ_PORT_TYPE_SYNTH, + SNDRV_SEQ_PORT_TYPE_SYNTH | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, 16, 0, name); if (p->chset->port < 0) { diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 58838f7c95..d176cc0174 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -54,7 +54,9 @@ static struct snd_midi_op emux_ops = { #define DEFAULT_MIDI_TYPE (SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |\ SNDRV_SEQ_PORT_TYPE_MIDI_GM |\ SNDRV_SEQ_PORT_TYPE_MIDI_GS |\ - SNDRV_SEQ_PORT_TYPE_MIDI_XG) + SNDRV_SEQ_PORT_TYPE_MIDI_XG |\ + SNDRV_SEQ_PORT_TYPE_HARDWARE |\ + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER) /* * Initialise the EMUX Synth by creating a client and registering -- cgit v1.2.2 From a7b928ac5fcd8e1b5c7c69926d8845b1d0500af3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 May 2006 16:22:12 +0200 Subject: [ALSA] rawmidi: add get_port_info callback for sequencer information flags Add a get_port_info callback to the snd_rawmidi_global_ops structure to allow the USB MIDI driver to supply information flags for the sequencer ports created by seq_midi. Signed-off-by: Clemens Ladisch --- sound/core/seq/seq_midi.c | 3 + sound/usb/usbmidi.c | 200 ++++++++++++++++++++++++++++++---------------- 2 files changed, 133 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index f873742c65..1daa5b069c 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -278,6 +278,7 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) struct seq_midisynth *msynth, *ms; struct snd_seq_port_info *port; struct snd_rawmidi_info *info; + struct snd_rawmidi *rmidi = dev->private_data; int newclient = 0; unsigned int p, ports; struct snd_seq_port_callback pcallbacks; @@ -389,6 +390,8 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) pcallbacks.unuse = midisynth_unuse; pcallbacks.event_input = event_process_midi; port->kernel = &pcallbacks; + if (rmidi->ops && rmidi->ops->get_port_info) + rmidi->ops->get_port_info(rmidi, p, port); if (snd_seq_kernel_client_ctl(client->seq_client, SNDRV_SEQ_IOCTL_CREATE_PORT, port)<0) goto __nomem; ms->seq_client = client->seq_client; diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2b9d940c80..5c53ec8a13 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -48,6 +48,7 @@ #include #include #include +#include #include "usbaudio.h" @@ -1010,97 +1011,157 @@ static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_m * "(product) MIDI (n)" schema because they aren't external MIDI ports, * such as internal control or synthesizer ports. */ -static struct { +static struct port_info { u32 id; - int port; - const char *name_format; -} snd_usbmidi_port_names[] = { + short int port; + short int voices; + const char *name; + unsigned int seq_flags; +} snd_usbmidi_port_info[] = { +#define PORT_INFO(vendor, product, num, name_, voices_, flags) \ + { .id = USB_ID(vendor, product), \ + .port = num, .voices = voices_, \ + .name = name_, .seq_flags = flags } +#define EXTERNAL_PORT(vendor, product, num, name) \ + PORT_INFO(vendor, product, num, name, 0, \ + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \ + SNDRV_SEQ_PORT_TYPE_HARDWARE | \ + SNDRV_SEQ_PORT_TYPE_PORT) +#define CONTROL_PORT(vendor, product, num, name) \ + PORT_INFO(vendor, product, num, name, 0, \ + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \ + SNDRV_SEQ_PORT_TYPE_HARDWARE) +#define ROLAND_SYNTH_PORT(vendor, product, num, name, voices) \ + PORT_INFO(vendor, product, num, name, voices, \ + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \ + SNDRV_SEQ_PORT_TYPE_MIDI_GM | \ + SNDRV_SEQ_PORT_TYPE_MIDI_GM2 | \ + SNDRV_SEQ_PORT_TYPE_MIDI_GS | \ + SNDRV_SEQ_PORT_TYPE_MIDI_XG | \ + SNDRV_SEQ_PORT_TYPE_HARDWARE | \ + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER) +#define SOUNDCANVAS_PORT(vendor, product, num, name, voices) \ + PORT_INFO(vendor, product, num, name, voices, \ + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \ + SNDRV_SEQ_PORT_TYPE_MIDI_GM | \ + SNDRV_SEQ_PORT_TYPE_MIDI_GM2 | \ + SNDRV_SEQ_PORT_TYPE_MIDI_GS | \ + SNDRV_SEQ_PORT_TYPE_MIDI_XG | \ + SNDRV_SEQ_PORT_TYPE_MIDI_MT32 | \ + SNDRV_SEQ_PORT_TYPE_HARDWARE | \ + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER) /* Roland UA-100 */ - { USB_ID(0x0582, 0x0000), 2, "%s Control" }, + CONTROL_PORT(0x0582, 0x0000, 2, "%s Control"), /* Roland SC-8850 */ - { USB_ID(0x0582, 0x0003), 0, "%s Part A" }, - { USB_ID(0x0582, 0x0003), 1, "%s Part B" }, - { USB_ID(0x0582, 0x0003), 2, "%s Part C" }, - { USB_ID(0x0582, 0x0003), 3, "%s Part D" }, - { USB_ID(0x0582, 0x0003), 4, "%s MIDI 1" }, - { USB_ID(0x0582, 0x0003), 5, "%s MIDI 2" }, + SOUNDCANVAS_PORT(0x0582, 0x0003, 0, "%s Part A", 128), + SOUNDCANVAS_PORT(0x0582, 0x0003, 1, "%s Part B", 128), + SOUNDCANVAS_PORT(0x0582, 0x0003, 2, "%s Part C", 128), + SOUNDCANVAS_PORT(0x0582, 0x0003, 3, "%s Part D", 128), + EXTERNAL_PORT(0x0582, 0x0003, 4, "%s MIDI 1"), + EXTERNAL_PORT(0x0582, 0x0003, 5, "%s MIDI 2"), /* Roland U-8 */ - { USB_ID(0x0582, 0x0004), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x0004), 1, "%s Control" }, + EXTERNAL_PORT(0x0582, 0x0004, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0004, 1, "%s Control"), /* Roland SC-8820 */ - { USB_ID(0x0582, 0x0007), 0, "%s Part A" }, - { USB_ID(0x0582, 0x0007), 1, "%s Part B" }, - { USB_ID(0x0582, 0x0007), 2, "%s MIDI" }, + SOUNDCANVAS_PORT(0x0582, 0x0007, 0, "%s Part A", 64), + SOUNDCANVAS_PORT(0x0582, 0x0007, 1, "%s Part B", 64), + EXTERNAL_PORT(0x0582, 0x0007, 2, "%s MIDI"), /* Roland SK-500 */ - { USB_ID(0x0582, 0x000b), 0, "%s Part A" }, - { USB_ID(0x0582, 0x000b), 1, "%s Part B" }, - { USB_ID(0x0582, 0x000b), 2, "%s MIDI" }, + SOUNDCANVAS_PORT(0x0582, 0x000b, 0, "%s Part A", 64), + SOUNDCANVAS_PORT(0x0582, 0x000b, 1, "%s Part B", 64), + EXTERNAL_PORT(0x0582, 0x000b, 2, "%s MIDI"), /* Roland SC-D70 */ - { USB_ID(0x0582, 0x000c), 0, "%s Part A" }, - { USB_ID(0x0582, 0x000c), 1, "%s Part B" }, - { USB_ID(0x0582, 0x000c), 2, "%s MIDI" }, + SOUNDCANVAS_PORT(0x0582, 0x000c, 0, "%s Part A", 64), + SOUNDCANVAS_PORT(0x0582, 0x000c, 1, "%s Part B", 64), + EXTERNAL_PORT(0x0582, 0x000c, 2, "%s MIDI"), /* Edirol UM-880 */ - { USB_ID(0x0582, 0x0014), 8, "%s Control" }, + CONTROL_PORT(0x0582, 0x0014, 8, "%s Control"), /* Edirol SD-90 */ - { USB_ID(0x0582, 0x0016), 0, "%s Part A" }, - { USB_ID(0x0582, 0x0016), 1, "%s Part B" }, - { USB_ID(0x0582, 0x0016), 2, "%s MIDI 1" }, - { USB_ID(0x0582, 0x0016), 3, "%s MIDI 2" }, + ROLAND_SYNTH_PORT(0x0582, 0x0016, 0, "%s Part A", 128), + ROLAND_SYNTH_PORT(0x0582, 0x0016, 1, "%s Part B", 128), + EXTERNAL_PORT(0x0582, 0x0016, 2, "%s MIDI 1"), + EXTERNAL_PORT(0x0582, 0x0016, 3, "%s MIDI 2"), /* Edirol UM-550 */ - { USB_ID(0x0582, 0x0023), 5, "%s Control" }, + CONTROL_PORT(0x0582, 0x0023, 5, "%s Control"), /* Edirol SD-20 */ - { USB_ID(0x0582, 0x0027), 0, "%s Part A" }, - { USB_ID(0x0582, 0x0027), 1, "%s Part B" }, - { USB_ID(0x0582, 0x0027), 2, "%s MIDI" }, + ROLAND_SYNTH_PORT(0x0582, 0x0027, 0, "%s Part A", 64), + ROLAND_SYNTH_PORT(0x0582, 0x0027, 1, "%s Part B", 64), + EXTERNAL_PORT(0x0582, 0x0027, 2, "%s MIDI"), /* Edirol SD-80 */ - { USB_ID(0x0582, 0x0029), 0, "%s Part A" }, - { USB_ID(0x0582, 0x0029), 1, "%s Part B" }, - { USB_ID(0x0582, 0x0029), 2, "%s MIDI 1" }, - { USB_ID(0x0582, 0x0029), 3, "%s MIDI 2" }, + ROLAND_SYNTH_PORT(0x0582, 0x0029, 0, "%s Part A", 128), + ROLAND_SYNTH_PORT(0x0582, 0x0029, 1, "%s Part B", 128), + EXTERNAL_PORT(0x0582, 0x0029, 2, "%s MIDI 1"), + EXTERNAL_PORT(0x0582, 0x0029, 3, "%s MIDI 2"), /* Edirol UA-700 */ - { USB_ID(0x0582, 0x002b), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x002b), 1, "%s Control" }, + EXTERNAL_PORT(0x0582, 0x002b, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x002b, 1, "%s Control"), /* Roland VariOS */ - { USB_ID(0x0582, 0x002f), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x002f), 1, "%s External MIDI" }, - { USB_ID(0x0582, 0x002f), 2, "%s Sync" }, + EXTERNAL_PORT(0x0582, 0x002f, 0, "%s MIDI"), + EXTERNAL_PORT(0x0582, 0x002f, 1, "%s External MIDI"), + EXTERNAL_PORT(0x0582, 0x002f, 2, "%s Sync"), /* Edirol PCR */ - { USB_ID(0x0582, 0x0033), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x0033), 1, "%s 1" }, - { USB_ID(0x0582, 0x0033), 2, "%s 2" }, + EXTERNAL_PORT(0x0582, 0x0033, 0, "%s MIDI"), + EXTERNAL_PORT(0x0582, 0x0033, 1, "%s 1"), + EXTERNAL_PORT(0x0582, 0x0033, 2, "%s 2"), /* BOSS GS-10 */ - { USB_ID(0x0582, 0x003b), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x003b), 1, "%s Control" }, + EXTERNAL_PORT(0x0582, 0x003b, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x003b, 1, "%s Control"), /* Edirol UA-1000 */ - { USB_ID(0x0582, 0x0044), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x0044), 1, "%s Control" }, + EXTERNAL_PORT(0x0582, 0x0044, 0, "%s MIDI"), + CONTROL_PORT(0x0582, 0x0044, 1, "%s Control"), /* Edirol UR-80 */ - { USB_ID(0x0582, 0x0048), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x0048), 1, "%s 1" }, - { USB_ID(0x0582, 0x0048), 2, "%s 2" }, + EXTERNAL_PORT(0x0582, 0x0048, 0, "%s MIDI"), + EXTERNAL_PORT(0x0582, 0x0048, 1, "%s 1"), + EXTERNAL_PORT(0x0582, 0x0048, 2, "%s 2"), /* Edirol PCR-A */ - { USB_ID(0x0582, 0x004d), 0, "%s MIDI" }, - { USB_ID(0x0582, 0x004d), 1, "%s 1" }, - { USB_ID(0x0582, 0x004d), 2, "%s 2" }, + EXTERNAL_PORT(0x0582, 0x004d, 0, "%s MIDI"), + EXTERNAL_PORT(0x0582, 0x004d, 1, "%s 1"), + EXTERNAL_PORT(0x0582, 0x004d, 2, "%s 2"), /* Edirol UM-3EX */ - { USB_ID(0x0582, 0x009a), 3, "%s Control" }, + CONTROL_PORT(0x0582, 0x009a, 3, "%s Control"), /* M-Audio MidiSport 8x8 */ - { USB_ID(0x0763, 0x1031), 8, "%s Control" }, - { USB_ID(0x0763, 0x1033), 8, "%s Control" }, + CONTROL_PORT(0x0763, 0x1031, 8, "%s Control"), + CONTROL_PORT(0x0763, 0x1033, 8, "%s Control"), /* MOTU Fastlane */ - { USB_ID(0x07fd, 0x0001), 0, "%s MIDI A" }, - { USB_ID(0x07fd, 0x0001), 1, "%s MIDI B" }, + EXTERNAL_PORT(0x07fd, 0x0001, 0, "%s MIDI A"), + EXTERNAL_PORT(0x07fd, 0x0001, 1, "%s MIDI B"), /* Emagic Unitor8/AMT8/MT4 */ - { USB_ID(0x086a, 0x0001), 8, "%s Broadcast" }, - { USB_ID(0x086a, 0x0002), 8, "%s Broadcast" }, - { USB_ID(0x086a, 0x0003), 4, "%s Broadcast" }, + EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), + EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), + EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), }; +static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_info); ++i) { + if (snd_usbmidi_port_info[i].id == umidi->chip->usb_id && + snd_usbmidi_port_info[i].port == number) + return &snd_usbmidi_port_info[i]; + } + return NULL; +} + +static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number, + struct snd_seq_port_info *seq_port_info) +{ + struct snd_usb_midi *umidi = rmidi->private_data; + struct port_info *port_info; + + /* TODO: read port flags from descriptors */ + port_info = find_port_info(umidi, number); + if (port_info) { + seq_port_info->type = port_info->seq_flags; + seq_port_info->midi_voices = port_info->voices; + } +} + static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, int stream, int number, struct snd_rawmidi_substream ** rsubstream) { - int i; + struct port_info *port_info; const char *name_format; struct snd_rawmidi_substream *substream = snd_usbmidi_find_substream(umidi, stream, number); @@ -1110,14 +1171,8 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, } /* TODO: read port name from jack descriptor */ - name_format = "%s MIDI %d"; - for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_names); ++i) { - if (snd_usbmidi_port_names[i].id == umidi->chip->usb_id && - snd_usbmidi_port_names[i].port == number) { - name_format = snd_usbmidi_port_names[i].name_format; - break; - } - } + port_info = find_port_info(umidi, number); + name_format = port_info ? port_info->name : "%s MIDI %d"; snprintf(substream->name, sizeof(substream->name), name_format, umidi->chip->card->shortname, number + 1); @@ -1457,6 +1512,10 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, return 0; } +static struct snd_rawmidi_global_ops snd_usbmidi_ops = { + .get_port_info = snd_usbmidi_get_port_info, +}; + static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, int out_ports, int in_ports) { @@ -1472,6 +1531,7 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->ops = &snd_usbmidi_ops; rmidi->private_data = umidi; rmidi->private_free = snd_usbmidi_rawmidi_free; snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usbmidi_output_ops); -- cgit v1.2.2 From 886da8677d2e4e942fc8984b22bfb8da45e810ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 May 2006 18:17:57 +0200 Subject: [ALSA] hda-codec - Add support for LG S1 laptop Added the model entry for LG S1 laptop. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cf6c100940..6876094c91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2174,6 +2174,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "lg", .config = ALC880_LG }, { .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x0068, .config = ALC880_LG }, { .modelname = "lg-lw", .config = ALC880_LG_LW }, { .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW }, -- cgit v1.2.2 From eed656493a459bbc0fdf687fa8f43f87946d8d3a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 May 2006 18:22:06 +0200 Subject: [ALSA] Add a workaround for ASUS A6KM Added a workaround for ASUS A6KM board that requires EAPD rather than SPDIF-in. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 4d9cf37300..720b419e0c 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -2048,7 +2048,10 @@ int patch_alc650(struct snd_ac97 * ac97) /* Enable SPDIF-IN only on Rev.E and above */ val = snd_ac97_read(ac97, AC97_ALC650_CLOCK); /* SPDIF IN with pin 47 */ - if (ac97->spec.dev_flags) + if (ac97->spec.dev_flags && + /* ASUS A6KM requires EAPD */ + ! (ac97->subsystem_vendor == 0x1043 && + ac97->subsystem_device == 0x1103)) val |= 0x03; /* enable */ else val &= ~0x03; /* disable */ -- cgit v1.2.2 From 1dbfd8c56bd7366d86e58b3e510a75de93e1978b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 May 2006 18:31:31 +0200 Subject: [ALSA] cs5535audio - Add missing module_param*() and MODULE_PARM_DESC() Added missing module_param*() and MODULE_PARM_DESC() for cs5535audio driver. Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 41f02f05df..f61c4fa4ed 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -60,6 +60,13 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); + static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, -- cgit v1.2.2 From a9393d70e564e4afe0333b1e26dda48af8b9305e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 May 2006 11:59:03 +0200 Subject: [ALSA] hda-codec - Fix mute switch on VAIO laptops with STAC7661 Fixed the master mute switch on VAIO laptops with STAC7661 codec chip. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c440fb986..d8622951c3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1262,13 +1262,13 @@ static int vaio_master_sw_put(struct snd_kcontrol *kcontrol, int change; change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0, - 0x80, valp[0] & 0x80); + 0x80, (valp[0] ? 0 : 0x80)); change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0, - 0x80, valp[1] & 0x80); + 0x80, (valp[1] ? 0 : 0x80)); snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, - 0x80, valp[0] & 0x80); + 0x80, (valp[0] ? 0 : 0x80)); snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, - 0x80, valp[1] & 0x80); + 0x80, (valp[1] ? 0 : 0x80)); return change; } -- cgit v1.2.2 From a59524faf3a2050e14a1c9038eb006ce96025394 Mon Sep 17 00:00:00 2001 From: Matt Porter Date: Wed, 3 May 2006 14:08:33 +0200 Subject: [ALSA] hda: add sigmatel 9227/9228/9229 ids Adds support for the 9227/9228/9229 sigmatel hda codecs. Signed-off-by: Matt Porter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8622951c3..6d8224dc03 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1370,6 +1370,12 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x }, { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x }, { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x }, + { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac922x }, + { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac922x }, + { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac922x }, + { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac922x }, + { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac922x }, + { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac922x }, { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x }, { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x }, { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x }, -- cgit v1.2.2 From 520290e43f9880da34e542185838816c6d79a340 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Wed, 3 May 2006 17:07:29 +0200 Subject: [ALSA] au88x0 - Init before create components Change the order in vortex_probe to set the card details before creating the components, meaning for example that card->shortname is available when registering the midi port. I have also added extra to card->shortname, and a line to overwrite the midi name following snd_mpu401_uart_new. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.c | 12 +++++++----- sound/pci/au88x0/au88x0_mpu401.c | 3 +++ 2 files changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 126870ec06..8a3b118989 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -261,6 +261,13 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return err; } snd_vortex_workaround(pci, pcifix[dev]); + + // Card details needed in snd_vortex_midi + strcpy(card->driver, CARD_NAME_SHORT); + sprintf(card->shortname, "Aureal Vortex %s", CARD_NAME_SHORT); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->io, chip->irq); + // (4) Alloc components. // ADB pcm. if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) { @@ -323,11 +330,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) #endif // (5) - strcpy(card->driver, CARD_NAME_SHORT); - strcpy(card->shortname, CARD_NAME_SHORT); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->io, chip->irq); - if ((err = pci_read_config_word(pci, PCI_DEVICE_ID, &(chip->device))) < 0) { snd_card_free(card); diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 873f486b07..814bc2db9f 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -107,6 +107,9 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) mpu = rmidi->private_data; mpu->cport = (unsigned long)(vortex->mmio + VORTEX_MIDI_CMD); #endif + /* Overwrite MIDI name */ + snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI %d", CARD_NAME_SHORT , vortex->card->number); + vortex->rmidi = rmidi; return 0; } -- cgit v1.2.2 From cab5c4c97a98e46359faa52e86787c1f0ccd773c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 May 2006 14:36:08 +0200 Subject: [ALSA] cmipci - Disable integrated mpu401 as default Enable the support of mpu401 PCI port only when mpu_port=1 module option is given, i.e. disabled as default. It turned out that the check of integrated midi port isn't perfect and caused hang-ups on some boards. Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 42ca92be18..cb475ada2e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2932,7 +2932,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc } integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff; - if (integrated_midi) + if (integrated_midi && mpu_port[dev] == 1) iomidi = cm->iobase + CM_REG_MPU_PCI; else { iomidi = mpu_port[dev]; -- cgit v1.2.2 From 4d1a70dad0e1c44dc0725de6de25aceead48599e Mon Sep 17 00:00:00 2001 From: Raimonds Cicans Date: Fri, 5 May 2006 09:49:53 +0200 Subject: [ALSA] add support for SB Live! 24-Bit External remote control This patch rewrites the remote control code to use a table for the peculiarities of the various SB models, and adds support for a third model. Signed-off-by: Raimonds Cicans Signed-off-by: Clemens Ladisch --- sound/usb/usbmixer.c | 68 ++++++++++++++++++++++++++++++---------------------- 1 file changed, 40 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ab921aa9d7..491e975a0c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -46,6 +46,27 @@ /* ignore error from controls - for debugging */ /* #define IGNORE_CTL_ERROR */ +/* + * Sound Blaster remote control configuration + * + * format of remote control data: + * Extigy: xx 00 + * Audigy 2 NX: 06 80 xx 00 00 00 + * Live! 24-bit: 06 80 xx yy 22 83 + */ +static const struct rc_config { + u32 usb_id; + u8 offset; + u8 length; + u8 packet_length; + u8 mute_mixer_id; + u32 mute_code; +} rc_configs[] = { + { USB_ID(0x041e, 0x3000), 0, 1, 2, 18, 0x0013 }, /* Extigy */ + { USB_ID(0x041e, 0x3020), 2, 1, 6, 18, 0x0013 }, /* Audigy 2 NX */ + { USB_ID(0x041e, 0x3040), 2, 2, 6, 2, 0x6e91 }, /* Live! 24-bit */ +}; + struct usb_mixer_interface { struct snd_usb_audio *chip; unsigned int ctrlif; @@ -55,11 +76,7 @@ struct usb_mixer_interface { struct usb_mixer_elem_info **id_elems; /* array[256], indexed by unit id */ /* Sound Blaster remote control stuff */ - enum { - RC_NONE, - RC_EXTIGY, - RC_AUDIGY2NX, - } rc_type; + const struct rc_config *rc_cfg; unsigned long rc_hwdep_open; u32 rc_code; wait_queue_head_t rc_waitq; @@ -1647,7 +1664,7 @@ static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, int unitid) { - if (mixer->rc_type == RC_NONE) + if (!mixer->rc_cfg) return; /* unit ids specific to Extigy/Audigy 2 NX: */ switch (unitid) { @@ -1732,20 +1749,19 @@ static void snd_usb_soundblaster_remote_complete(struct urb *urb, struct pt_regs *regs) { struct usb_mixer_interface *mixer = urb->context; - /* - * format of remote control data: - * Extigy: xx 00 - * Audigy 2 NX: 06 80 xx 00 00 00 - */ - int offset = mixer->rc_type == RC_EXTIGY ? 0 : 2; + const struct rc_config *rc = mixer->rc_cfg; u32 code; - if (urb->status < 0 || urb->actual_length <= offset) + if (urb->status < 0 || urb->actual_length < rc->packet_length) return; - code = mixer->rc_buffer[offset]; + + code = mixer->rc_buffer[rc->offset]; + if (rc->length == 2) + code |= mixer->rc_buffer[rc->offset + 1] << 8; + /* the Mute button actually changes the mixer control */ - if (code == 13) - snd_usb_mixer_notify_id(mixer, 18); + if (code == rc->mute_code) + snd_usb_mixer_notify_id(mixer, rc->mute_mixer_id); mixer->rc_code = code; wmb(); wake_up(&mixer->rc_waitq); @@ -1801,21 +1817,17 @@ static unsigned int snd_usb_sbrc_hwdep_poll(struct snd_hwdep *hw, struct file *f static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) { struct snd_hwdep *hwdep; - int err, len; + int err, len, i; - switch (mixer->chip->usb_id) { - case USB_ID(0x041e, 0x3000): - mixer->rc_type = RC_EXTIGY; - len = 2; - break; - case USB_ID(0x041e, 0x3020): - mixer->rc_type = RC_AUDIGY2NX; - len = 6; - break; - default: + for (i = 0; i < ARRAY_SIZE(rc_configs); ++i) + if (rc_configs[i].usb_id == mixer->chip->usb_id) + break; + if (i >= ARRAY_SIZE(rc_configs)) return 0; - } + mixer->rc_cfg = &rc_configs[i]; + len = mixer->rc_cfg->packet_length; + init_waitqueue_head(&mixer->rc_waitq); err = snd_hwdep_new(mixer->chip->card, "SB remote control", 0, &hwdep); if (err < 0) -- cgit v1.2.2 From 62fe78e90dc25b269362034487dc450cd8453e8c Mon Sep 17 00:00:00 2001 From: Sam Revitch Date: Wed, 10 May 2006 15:09:17 +0200 Subject: [ALSA] hda-codec - Add support for Apple Mac Mini (early 2006) Add support for some audio quirks of the Apple Mac Mini (early 2006) Signed-off-by: Sam Revitch Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 68 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 67 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6d8224dc03..36f199442f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -41,6 +41,7 @@ #define STAC_REF 0 #define STAC_D945GTP3 1 #define STAC_D945GTP5 2 +#define STAC_MACMINI 3 struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; @@ -52,6 +53,7 @@ struct sigmatel_spec { unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; + unsigned int gpio_mute: 1; /* playback */ struct hda_multi_out multiout; @@ -293,6 +295,7 @@ static unsigned int *stac922x_brd_tbl[] = { ref922x_pin_configs, d945gtp3_pin_configs, d945gtp5_pin_configs, + NULL, /* STAC_MACMINI */ }; static struct hda_board_config stac922x_cfg_tbl[] = { @@ -324,6 +327,9 @@ static struct hda_board_config stac922x_cfg_tbl[] = { { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0417, .config = STAC_D945GTP5 }, /* Intel D975XBK - 5 Stack */ + { .pci_subvendor = 0x8384, + .pci_subdevice = 0x7680, + .config = STAC_MACMINI }, /* Apple Mac Mini (early 2006) */ {} /* terminator */ }; @@ -841,6 +847,19 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const } } + if (imux->num_items == 1) { + /* + * Set the current input for the muxes. + * The STAC9221 has two input muxes with identical source + * NID lists. Hopefully this won't get confused. + */ + for (i = 0; i < spec->num_muxes; i++) { + snd_hda_codec_write(codec, spec->mux_nids[i], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } + } + return 0; } @@ -946,6 +965,45 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return 1; } +/* + * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a + * funky external mute control using GPIO pins. + */ + +static void stac922x_gpio_mute(struct hda_codec *codec, int pin, int muted) +{ + unsigned int gpiostate, gpiomask, gpiodir; + + gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + + if (!muted) + gpiostate |= (1 << pin); + else + gpiostate &= ~(1 << pin); + + gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_MASK, 0); + gpiomask |= (1 << pin); + + gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + gpiodir |= (1 << pin); + + /* AppleHDA seems to do this -- WTF is this verb?? */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, gpiomask); + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpiodir); + + msleep(1); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpiostate); +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -982,6 +1040,11 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); + if (spec->gpio_mute) { + stac922x_gpio_mute(codec, 0, 0); + stac922x_gpio_mute(codec, 1, 0); + } + return 0; } @@ -1132,7 +1195,7 @@ static int patch_stac922x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl); if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, using BIOS defaults\n"); - else { + else if (stac922x_brd_tbl[spec->board_config] != NULL) { spec->num_pins = 10; spec->pin_nids = stac922x_pin_nids; spec->pin_configs = stac922x_brd_tbl[spec->board_config]; @@ -1154,6 +1217,9 @@ static int patch_stac922x(struct hda_codec *codec) return err; } + if (spec->board_config == STAC_MACMINI) + spec->gpio_mute = 1; + codec->patch_ops = stac92xx_patch_ops; return 0; -- cgit v1.2.2 From 2ce7fb579f842f76a0216618c105bffd334d9233 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 May 2006 16:24:42 +0200 Subject: [ALSA] rme96 - Fix OSS full-duplex Fixed a bug in rme96 driver that the full-duplex on OSS emulation doesn't work due to the invalid period size parameter. Signed-off-by: Takashi Iwai --- sound/pci/rme96.c | 32 +++++++++++++++++++++++--------- 1 file changed, 23 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 75a8b754ef..65611a7d36 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1151,6 +1151,25 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = { .mask = 0 }; +static void +rme96_set_buffer_size_constraint(struct rme96 *rme96, + struct snd_pcm_runtime *runtime) +{ + unsigned int size; + + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); + if ((size = rme96->playback_periodsize) != 0 || + (size = rme96->capture_periodsize) != 0) + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + size, size); + else + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + &hw_constraints_period_bytes); +} + static int snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) { @@ -1180,8 +1199,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); + rme96_set_buffer_size_constraint(rme96, runtime); rme96->wcreg_spdif_stream = rme96->wcreg_spdif; rme96->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1219,9 +1237,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock); - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); - + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1254,8 +1270,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1291,8 +1306,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream) rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock); - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } -- cgit v1.2.2 From 3206b9ca9fba8dc8d6ddd371a3ff455c67ad137f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 May 2006 16:33:11 +0200 Subject: [ALSA] hda-codec - Add support for Sony Vaio VGN-S3HP Added the missing support for Sony Vaio VGN-S3HP with ALC260 codec. The patch taken from ALSA bug#2101. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6876094c91..f6bccd66d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3823,6 +3823,8 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "basic", .config = ALC260_BASIC }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb, .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81cc, + .config = ALC260_BASIC }, /* Sony VAIO VGN-S3HP */ { .pci_subvendor = 0x104d, .pci_subdevice = 0x81cd, .config = ALC260_BASIC }, /* Sony VAIO */ { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, -- cgit v1.2.2 From 0defb2672d7cde8d048eec35c183da7b88adbd9e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 May 2006 18:12:23 +0200 Subject: [ALSA] hda-codec - Fix handling of capture controls on ALC882 3/6-stack models Fixed the handling of capture controls on ALC882 3/6-stack models. Now the driver checks the availability of NID 07h. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 19 ------------------- 1 file changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6bccd66d1..0fc2f77dce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4099,21 +4099,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 3, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, { } /* end */ }; @@ -4347,8 +4332,6 @@ static struct alc_config_preset alc882_presets[] = { .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, .dig_in_nid = ALC882_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, @@ -4360,8 +4343,6 @@ static struct alc_config_preset alc882_presets[] = { .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, .dig_in_nid = ALC882_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), .channel_mode = alc882_sixstack_modes, -- cgit v1.2.2 From 746df94898554b3d8e91d855e934852e626c701c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 May 2006 19:49:05 +0200 Subject: [ALSA] Fix rwlock around snd_iprintf() in sound core Fixed rwlock around snd_iprintf() in sound core part. Replaced with mutex. Also, make mutex and flags static variables with addition of snd_card_locked() function (just for sound.c). Signed-off-by: Takashi Iwai --- sound/core/init.c | 51 +++++++++++++++++++++++++++++++-------------------- sound/core/sound.c | 7 +------ 2 files changed, 32 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 2ff0e5e908..38b2d4a9d6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -38,11 +38,11 @@ struct snd_shutdown_f_ops { struct snd_shutdown_f_ops *next; }; -unsigned int snd_cards_lock = 0; /* locked for registering/using */ +static unsigned int snd_cards_lock = 0; /* locked for registering/using */ struct snd_card *snd_cards[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = NULL}; EXPORT_SYMBOL(snd_cards); -DEFINE_RWLOCK(snd_card_rwlock); +static DEFINE_MUTEX(snd_card_mutex); #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); @@ -112,7 +112,7 @@ struct snd_card *snd_card_new(int idx, const char *xid, strlcpy(card->id, xid, sizeof(card->id)); } err = 0; - write_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); if (idx < 0) { int idx2; for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) @@ -130,12 +130,12 @@ struct snd_card *snd_card_new(int idx, const char *xid, else err = -ENODEV; if (idx < 0 || err < 0) { - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); snd_printk(KERN_ERR "cannot find the slot for index %d (range 0-%i)\n", idx, snd_ecards_limit - 1); goto __error; } snd_cards_lock |= 1 << idx; /* lock it */ - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); card->number = idx; card->module = module; INIT_LIST_HEAD(&card->devices); @@ -173,6 +173,17 @@ struct snd_card *snd_card_new(int idx, const char *xid, EXPORT_SYMBOL(snd_card_new); +/* return non-zero if a card is already locked */ +int snd_card_locked(int card) +{ + int locked; + + mutex_lock(&snd_card_mutex); + locked = snd_cards_lock & (1 << card); + mutex_unlock(&snd_card_mutex); + return locked; +} + static loff_t snd_disconnect_llseek(struct file *file, loff_t offset, int orig) { return -ENODEV; @@ -240,9 +251,9 @@ int snd_card_disconnect(struct snd_card *card) spin_unlock(&card->files_lock); /* phase 1: disable fops (user space) operations for ALSA API */ - write_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); snd_cards[card->number] = NULL; - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); /* phase 2: replace file->f_op with special dummy operations */ @@ -321,9 +332,9 @@ int snd_card_free(struct snd_card *card) if (card == NULL) return -EINVAL; - write_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); snd_cards[card->number] = NULL; - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); #ifdef CONFIG_PM wake_up(&card->power_sleep); @@ -359,9 +370,9 @@ int snd_card_free(struct snd_card *card) card->s_f_ops = s_f_ops->next; kfree(s_f_ops); } - write_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); snd_cards_lock &= ~(1 << card->number); - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); kfree(card); return 0; } @@ -497,16 +508,16 @@ int snd_card_register(struct snd_card *card) snd_assert(card != NULL, return -EINVAL); if ((err = snd_device_register_all(card)) < 0) return err; - write_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); if (snd_cards[card->number]) { /* already registered */ - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); return 0; } if (card->id[0] == '\0') choose_default_id(card); snd_cards[card->number] = card; - write_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); init_info_for_card(card); #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) if (snd_mixer_oss_notify_callback) @@ -527,7 +538,7 @@ static void snd_card_info_read(struct snd_info_entry *entry, struct snd_card *card; for (idx = count = 0; idx < SNDRV_CARDS; idx++) { - read_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); if ((card = snd_cards[idx]) != NULL) { count++; snd_iprintf(buffer, "%2i [%-15s]: %s - %s\n", @@ -538,7 +549,7 @@ static void snd_card_info_read(struct snd_info_entry *entry, snd_iprintf(buffer, " %s\n", card->longname); } - read_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); } if (!count) snd_iprintf(buffer, "--- no soundcards ---\n"); @@ -552,12 +563,12 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer) struct snd_card *card; for (idx = count = 0; idx < SNDRV_CARDS; idx++) { - read_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); if ((card = snd_cards[idx]) != NULL) { count++; snd_iprintf(buffer, "%s\n", card->longname); } - read_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); } if (!count) { snd_iprintf(buffer, "--- no soundcards ---\n"); @@ -575,11 +586,11 @@ static void snd_card_module_info_read(struct snd_info_entry *entry, struct snd_card *card; for (idx = 0; idx < SNDRV_CARDS; idx++) { - read_lock(&snd_card_rwlock); + mutex_lock(&snd_card_mutex); if ((card = snd_cards[idx]) != NULL) snd_iprintf(buffer, "%2i %s\n", idx, card->module->name); - read_unlock(&snd_card_rwlock); + mutex_unlock(&snd_card_mutex); } } #endif diff --git a/sound/core/sound.c b/sound/core/sound.c index 8313f97907..02c8cc4ebf 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -81,14 +81,9 @@ extern struct class *sound_class; */ void snd_request_card(int card) { - int locked; - if (! current->fs->root) return; - read_lock(&snd_card_rwlock); - locked = snd_cards_lock & (1 << card); - read_unlock(&snd_card_rwlock); - if (locked) + if (snd_card_locked(card)) return; if (card < 0 || card >= cards_limit) return; -- cgit v1.2.2 From ca54bde3634360afecd0dada9c59399bbe88bd32 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Wed, 17 May 2006 11:02:24 +0200 Subject: [ALSA] azt3328.c: add suspend/resume support - add suspend/resume handlers - fix problem (private_data members not set) Playing a file while suspending will resume correctly with this patch, so I assume the hardware to get fully correctly reinitialized with this patch. Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 119 ++++++++++++++++++++++++++++++++++++++++++++++++++-- sound/pci/azt3328.h | 20 +++++++-- 2 files changed, 133 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 52a3645242..f197fbac10 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -90,9 +90,11 @@ * * TODO * - test MPU401 MIDI playback etc. - * - power management. See e.g. intel8x0 or cs4281. - * This would be nice since the chip runs a bit hot, and it's *required* - * anyway for proper ACPI power management. + * - add some power micro-management (disable various units of the card + * as long as they're unused). However this requires I/O ports which I + * haven't figured out yet and which thus might not even exist... + * The standard suspend/resume functionality could probably make use of + * some improvement, too... * - figure out what all unknown port bits are responsible for */ @@ -214,6 +216,16 @@ struct snd_azf3328 { struct pci_dev *pci; int irq; + +#ifdef CONFIG_PM + /* register value containers for power management + * Note: not always full I/O range preserved (just like Win driver!) */ + u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2]; + u16 saved_regs_io2 [AZF_IO_SIZE_IO2_PM / 2]; + u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; + u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2]; + u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; +#endif }; static const struct pci_device_id snd_azf3328_ids[] __devinitdata = { @@ -961,6 +973,13 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_playing = 1; snd_azf3328_dbgplay("STARTED PLAYBACK\n"); break; + case SNDRV_PCM_TRIGGER_RESUME: + snd_azf3328_dbgplay("RESUME PLAYBACK\n"); + /* resume playback if we were active */ + if (chip->is_playing) + snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP PLAYBACK\n"); @@ -988,6 +1007,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_playing = 0; snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + snd_azf3328_dbgplay("SUSPEND PLAYBACK\n"); + /* make sure playback is stopped */ + snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); break; @@ -995,6 +1020,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); break; default: + printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } @@ -1068,6 +1094,13 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_recording = 1; snd_azf3328_dbgplay("STARTED CAPTURE\n"); break; + case SNDRV_PCM_TRIGGER_RESUME: + snd_azf3328_dbgplay("RESUME CAPTURE\n"); + /* resume recording if we were active */ + if (chip->is_recording) + snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP CAPTURE\n"); @@ -1088,6 +1121,12 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_recording = 0; snd_azf3328_dbgplay("STOPPED CAPTURE\n"); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + snd_azf3328_dbgplay("SUSPEND CAPTURE\n"); + /* make sure recording is stopped */ + snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); break; @@ -1095,6 +1134,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); break; default: + printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } @@ -1766,6 +1806,8 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) goto out_err; } + card->private_data = chip; + if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, chip->mpu_port, 1, pci->irq, 0, &chip->rmidi)) < 0) { @@ -1791,6 +1833,8 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) } } + opl3->private_data = chip; + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->codec_port, chip->irq); @@ -1834,11 +1878,80 @@ snd_azf3328_remove(struct pci_dev *pci) snd_azf3328_dbgcallleave(); } +#ifdef CONFIG_PM +static int +snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_azf3328 *chip = card->private_data; + int reg; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(chip->pcm); + + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) + chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2); + + /* make sure to disable master volume etc. to prevent looping sound */ + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) + chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) + chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) + chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) + chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2); + + pci_set_power_state(pci, PCI_D3hot); + pci_disable_device(pci); + pci_save_state(pci); + return 0; +} + +static int +snd_azf3328_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_azf3328 *chip = card->private_data; + int reg; + + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_power_state(pci, PCI_D0); + pci_set_master(pci); + + for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) + outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) + outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) + outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) + outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) + outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + + + + static struct pci_driver driver = { .name = "AZF3328", .id_table = snd_azf3328_ids, .probe = snd_azf3328_probe, .remove = __devexit_p(snd_azf3328_remove), +#ifdef CONFIG_PM + .suspend = snd_azf3328_suspend, + .resume = snd_azf3328_resume, +#endif }; static int __init diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index f489bdaf6d..560a4653c0 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -5,6 +5,9 @@ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ +#define AZF_IO_SIZE_CODEC 0x80 +#define AZF_IO_SIZE_CODEC_PM 0x70 + /* the driver initialisation suggests a layout of 4 main areas: * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??). * And another area from 0x60 to 0x6f (DirectX timer, IRQ management, @@ -107,7 +110,8 @@ #define IRQ_UNKNOWN2 0x0080 /* probably unused */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ #define IDX_IO_SOME_VALUE 0x68 /* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */ -#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated; actually inhibits PCM playback!!! maybe power management?? */ +#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */ + #define IO_6A_PAUSE_PLAYBACK 0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */ #define IDX_IO_6CH 0x6C #define IDX_IO_6EH 0x6E /* writing 0xffff returns 0x83fe */ /* further I/O indices not saved/restored, so probably not used */ @@ -115,15 +119,25 @@ /*** I/O 2 area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ +#define AZF_IO_SIZE_IO2 0x08 +#define AZF_IO_SIZE_IO2_PM 0x06 + #define IDX_IO2_LEGACY_ADDR 0x04 #define LEGACY_SOMETHING 0x01 /* OPL3?? */ #define LEGACY_JOY 0x08 +#define AZF_IO_SIZE_MPU 0x04 +#define AZF_IO_SIZE_MPU_PM 0x04 + +#define AZF_IO_SIZE_SYNTH 0x08 +#define AZF_IO_SIZE_SYNTH_PM 0x06 /*** mixer I/O area port indices ***/ /* (only 0x22 of 0x40 bytes saved/restored by Windows driver) - * generally spoken: AC97 register index = AZF3328 mixer reg index + 2 - * (in other words: AZF3328 NOT fully AC97 compliant) */ + * UNFORTUNATELY azf3328 is NOT truly AC97 compliant: see main file intro */ +#define AZF_IO_SIZE_MIXER 0x40 +#define AZF_IO_SIZE_MIXER_PM 0x22 + #define MIXER_VOLUME_RIGHT_MASK 0x001f #define MIXER_VOLUME_LEFT_MASK 0x1f00 #define MIXER_MUTE_MASK 0x8000 -- cgit v1.2.2 From 13769e3f21d6e9c59999c9bf6908278b878d05c5 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Wed, 17 May 2006 11:03:16 +0200 Subject: [ALSA] azt3328.c: add 3D sound mixer switch/rename controls - add 3D sound pre-3D/post-3D switch, as seen in standard AC-97 - rename controls to shorter and more accurate strings Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 41 +++++++++++++++++++++++++++++++---------- sound/pci/azt3328.h | 16 ++++++++-------- 2 files changed, 39 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index f197fbac10..c9af04ed20 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -39,8 +39,15 @@ * for compatibility reasons) has the following features: * * - builtin AC97 conformant codec (SNR over 80dB) - * (really AC97 compliant?? I really doubt it when looking - * at the mixer register layout) + * Note that "conformant" != "compliant"!! this chip's mixer register layout + * *differs* from the standard AC97 layout: + * they chose to not implement the headphone register (which is not a + * problem since it's merely optional), yet when doing this, they committed + * the grave sin of letting other registers follow immediately instead of + * keeping a headphone dummy register, thereby shifting the mixer register + * addresses illegally. So far unfortunately it looks like the very flexible + * ALSA AC97 support is still not enough to easily compensate for such a + * grave layout violation despite all tweaks and quirks mechanisms it offers. * - builtin genuine OPL3 * - full duplex 16bit playback/record at independent sampling rate * - MPU401 (+ legacy address support) FIXME: how to enable legacy addr?? @@ -96,6 +103,9 @@ * The standard suspend/resume functionality could probably make use of * some improvement, too... * - figure out what all unknown port bits are responsible for + * - figure out some cleverly evil scheme to possibly make ALSA AC97 code + * fully accept our quite incompatible ""AC97"" mixer and thus save some + * code (but I'm not too optimistic that doing this is possible at all) */ #include @@ -526,15 +536,18 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts1[] = { - "ModemOut1", "ModemOut2" + "Mic1", "Mic2" }; static const char * const texts2[] = { - "MonoSelectSource1", "MonoSelectSource2" + "Mix", "Mic" }; static const char * const texts3[] = { "Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone" }; + static const char * const texts4[] = { + "pre 3D", "post 3D" + }; struct azf3328_mixer_reg reg; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); @@ -545,10 +558,17 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.item = reg.enum_c - 1U; if (reg.reg == IDX_MIXER_ADVCTL2) { - if (reg.lchan_shift == 8) /* modem out sel */ + switch(reg.lchan_shift) { + case 8: /* modem out sel */ strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); - else /* mono sel source */ + break; + case 9: /* mono sel source */ strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]); + break; + case 15: /* PCM Out Path */ + strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); + break; + } } else strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] @@ -641,13 +661,14 @@ static const struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata AZF3328_MIXER_VOL_MONO("Modem Playback Volume", IDX_MIXER_MODEMOUT, 0x1f, 1), AZF3328_MIXER_SWITCH("Modem Capture Switch", IDX_MIXER_MODEMIN, 15, 1), AZF3328_MIXER_VOL_MONO("Modem Capture Volume", IDX_MIXER_MODEMIN, 0x1f, 1), - AZF3328_MIXER_ENUM("Modem Out Select", IDX_MIXER_ADVCTL2, 2, 8), - AZF3328_MIXER_ENUM("Mono Select Source", IDX_MIXER_ADVCTL2, 2, 9), + AZF3328_MIXER_ENUM("Mic Select", IDX_MIXER_ADVCTL2, 2, 8), + AZF3328_MIXER_ENUM("Mono Output Select", IDX_MIXER_ADVCTL2, 2, 9), + AZF3328_MIXER_ENUM("PCM", IDX_MIXER_ADVCTL2, 2, 15), /* PCM Out Path, place in front since it controls *both* 3D and Bass/Treble! */ AZF3328_MIXER_VOL_SPECIAL("Tone Control - Treble", IDX_MIXER_BASSTREBLE, 0x07, 1, 0), AZF3328_MIXER_VOL_SPECIAL("Tone Control - Bass", IDX_MIXER_BASSTREBLE, 0x07, 9, 0), AZF3328_MIXER_SWITCH("3D Control - Switch", IDX_MIXER_ADVCTL2, 13, 0), - AZF3328_MIXER_VOL_SPECIAL("3D Control - Wide", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */ - AZF3328_MIXER_VOL_SPECIAL("3D Control - Space", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */ + AZF3328_MIXER_VOL_SPECIAL("3D Control - Width", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */ + AZF3328_MIXER_VOL_SPECIAL("3D Control - Depth", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */ #if MIXER_TESTING AZF3328_MIXER_SWITCH("0", IDX_MIXER_ADVCTL2, 0, 0), AZF3328_MIXER_SWITCH("1", IDX_MIXER_ADVCTL2, 1, 0), diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 560a4653c0..b4f3e3cd00 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -90,7 +90,7 @@ #define IDX_IO_REC_DMA_CURROFS 0x34 /* PU:0x00000000 */ #define IDX_IO_REC_SOUNDFORMAT 0x36 /* PU:0x0000 */ -/** hmm, what is this I/O area for? MPU401?? (after playback, recording, ???, timer) **/ +/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/ #define IDX_IO_SOMETHING_FLAGS 0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */ /* general */ #define IDX_IO_42H 0x42 /* PU:0x0001 */ @@ -170,14 +170,14 @@ #define IDX_MIXER_ADVCTL1 0x1e /* unlisted bits are unmodifiable */ #define MIXER_ADVCTL1_3DWIDTH_MASK 0x000e - #define MIXER_ADVCTL1_HIFI3D_MASK 0x0300 -#define IDX_MIXER_ADVCTL2 0x20 /* resembles AC97_GENERAL_PURPOSE reg! */ + #define MIXER_ADVCTL1_HIFI3D_MASK 0x0300 /* yup, this is missing the high bit that official AC97 contains, plus it doesn't have linear bit value range behaviour but instead acts weirdly (possibly we're dealing with two *different* 3D settings here??) */ +#define IDX_MIXER_ADVCTL2 0x20 /* subset of AC97_GENERAL_PURPOSE reg! */ /* unlisted bits are unmodifiable */ - #define MIXER_ADVCTL2_BIT7 0x0080 /* WaveOut 3D Bypass? mutes WaveOut at LineOut */ - #define MIXER_ADVCTL2_BIT8 0x0100 /* is this Modem Out Select? */ - #define MIXER_ADVCTL2_BIT9 0x0200 /* Mono Select Source? */ - #define MIXER_ADVCTL2_BIT13 0x2000 /* 3D enable? */ - #define MIXER_ADVCTL2_BIT15 0x8000 /* unknown */ + #define MIXER_ADVCTL2_LPBK 0x0080 /* Loopback mode -- Win driver: "WaveOut3DBypass"? mutes WaveOut at LineOut */ + #define MIXER_ADVCTL2_MS 0x0100 /* Mic Select 0=Mic1, 1=Mic2 -- Win driver: "ModemOutSelect"?? */ + #define MIXER_ADVCTL2_MIX 0x0200 /* Mono output select 0=Mix, 1=Mic; Win driver: "MonoSelectSource"?? */ + #define MIXER_ADVCTL2_3D 0x2000 /* 3D Enhancement 1=on */ + #define MIXER_ADVCTL2_POP 0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */ #define IDX_MIXER_SOMETHING30H 0x30 /* used, but unknown??? */ -- cgit v1.2.2 From e2f872608af7f3c00beaa61ff6037e3cc5a66cf1 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Wed, 17 May 2006 11:04:19 +0200 Subject: [ALSA] azt3328.c: use kernel coding style Scope braces were not done the One True Kernel Way. Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 70 ++++++++++++++++++----------------------------------- 1 file changed, 23 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index c9af04ed20..e68056c815 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -33,7 +33,7 @@ * in the first place >:-P}), * I was forced to base this driver on reverse engineering * (3 weeks' worth of evenings filled with driver work). - * (and no, I did NOT go the easy way: to pick up a PCI128 for 9 Euros) + * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros) * * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name * for compatibility reasons) has the following features: @@ -339,10 +339,8 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg else dst_vol_left &= ~0x80; - do - { - if (!left_done) - { + do { + if (!left_done) { if (curr_vol_left > dst_vol_left) curr_vol_left--; else @@ -352,8 +350,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg left_done = 1; outb(curr_vol_left, portbase + 1); } - if (!right_done) - { + if (!right_done) { if (curr_vol_right > dst_vol_right) curr_vol_right--; else @@ -368,8 +365,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg } if (delay) mdelay(delay); - } - while ((!left_done) || (!right_done)); + } while ((!left_done) || (!right_done)); snd_azf3328_dbgcallleave(); } @@ -556,8 +552,7 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.items = reg.enum_c; if (uinfo->value.enumerated.item > reg.enum_c - 1U) uinfo->value.enumerated.item = reg.enum_c - 1U; - if (reg.reg == IDX_MIXER_ADVCTL2) - { + if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); @@ -569,8 +564,7 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); break; } - } - else + } else strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] ); return 0; @@ -586,12 +580,10 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol, snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); val = snd_azf3328_mixer_inw(chip, reg.reg); - if (reg.reg == IDX_MIXER_REC_SELECT) - { + if (reg.reg == IDX_MIXER_REC_SELECT) { ucontrol->value.enumerated.item[0] = (val >> 8) & (reg.enum_c - 1); ucontrol->value.enumerated.item[1] = (val >> 0) & (reg.enum_c - 1); - } - else + } else ucontrol->value.enumerated.item[0] = (val >> reg.lchan_shift) & (reg.enum_c - 1); snd_azf3328_dbgmixer("get_enum: %02x is %04x -> %d|%d (shift %02d, enum_c %d)\n", @@ -611,16 +603,13 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); val = oreg; - if (reg.reg == IDX_MIXER_REC_SELECT) - { + if (reg.reg == IDX_MIXER_REC_SELECT) { if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U || ucontrol->value.enumerated.item[1] > reg.enum_c - 1U) return -EINVAL; val = (ucontrol->value.enumerated.item[0] << 8) | (ucontrol->value.enumerated.item[1] << 0); - } - else - { + } else { if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U) return -EINVAL; val &= ~((reg.enum_c - 1) << reg.lchan_shift); @@ -846,22 +835,18 @@ snd_azf3328_setdmaa(struct snd_azf3328 *chip, unsigned int is_running; snd_azf3328_dbgcallenter(); - if (do_recording) - { + if (do_recording) { /* access capture registers, i.e. skip playback reg section */ portbase = chip->codec_port + 0x20; is_running = chip->is_recording; - } - else - { + } else { /* access the playback register section */ portbase = chip->codec_port + 0x00; is_running = chip->is_playing; } /* AZF3328 uses a two buffer pointer DMA playback approach */ - if (!is_running) - { + if (!is_running) { unsigned long addr_area2; unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */ count_areas = size/2; @@ -1224,8 +1209,7 @@ snd_azf3328_interrupt(int irq, void *dev_id, struct pt_regs *regs) snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status); - if (status & IRQ_TIMER) - { + if (status & IRQ_TIMER) { /* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */ if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); @@ -1235,50 +1219,43 @@ snd_azf3328_interrupt(int irq, void *dev_id, struct pt_regs *regs) spin_unlock(&chip->reg_lock); snd_azf3328_dbgplay("azt3328: timer IRQ\n"); } - if (status & IRQ_PLAYBACK) - { + if (status & IRQ_PLAYBACK) { spin_lock(&chip->reg_lock); which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE); /* ack all IRQ types immediately */ snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->playback_substream) - { + if (chip->pcm && chip->playback_substream) { snd_pcm_period_elapsed(chip->playback_substream); snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", which, inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS)); - } - else + } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); } - if (status & IRQ_RECORDING) - { + if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE); /* ack all IRQ types immediately */ snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->capture_substream) - { + if (chip->pcm && chip->capture_substream) { snd_pcm_period_elapsed(chip->capture_substream); snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", which, inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS)); - } - else + } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); } /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ - if (status & IRQ_MPU401) - { + if (status & IRQ_MPU401) { snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data, regs); /* hmm, do we have to ack the IRQ here somehow? @@ -1572,8 +1549,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) snd_azf3328_dbgcallenter(); chip = snd_timer_chip(timer); delay = ((timer->sticks * seqtimer_scaling) - 1) & TIMER_VALUE_MASK; - if (delay < 49) - { + if (delay < 49) { /* uhoh, that's not good, since user-space won't know about * this timing tweak * (we need to do it to avoid a lockup, though) */ -- cgit v1.2.2 From 778b6e1b2da260adf3d3254aaa35bffd1eb05b42 Mon Sep 17 00:00:00 2001 From: Felix Kuehling Date: Wed, 17 May 2006 11:22:21 +0200 Subject: [ALSA] hda - Add support for the ATI RS600 HDMI audio device Add support for the ATI RS600 HDMI audio device. It has a one-stream pure digital stereo codec that isn't handled by the generic codec support. Signed-off-by: Felix Kuehling Signed-off-by: Takashi Iwai --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_intel.c | 16 ++++ sound/pci/hda/hda_patch.h | 3 + sound/pci/hda/patch_atihdmi.c | 165 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 185 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/patch_atihdmi.c (limited to 'sound') diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index ddfb5ff7fb..dbacba6177 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,5 @@ snd-hda-intel-objs := hda_intel.o -snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o +snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o patch_atihdmi.o ifdef CONFIG_PROC_FS snd-hda-codec-objs += hda_proc.o endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e821d65afa..0154389bf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -82,6 +82,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH8}," "{ATI, SB450}," "{ATI, SB600}," + "{ATI, RS600}," "{VIA, VT8251}," "{VIA, VT8237A}," "{SiS, SIS966}," @@ -167,6 +168,12 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ULI_PLAYBACK_INDEX 5 #define ULI_NUM_PLAYBACK 6 +/* ATI HDMI has 1 playback and 0 capture */ +#define ATIHDMI_CAPTURE_INDEX 0 +#define ATIHDMI_NUM_CAPTURE 0 +#define ATIHDMI_PLAYBACK_INDEX 0 +#define ATIHDMI_NUM_PLAYBACK 1 + /* this number is statically defined for simplicity */ #define MAX_AZX_DEV 16 @@ -331,6 +338,7 @@ struct azx { enum { AZX_DRIVER_ICH, AZX_DRIVER_ATI, + AZX_DRIVER_ATIHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -340,6 +348,7 @@ enum { static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_ATI] = "HDA ATI SB", + [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1495,6 +1504,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->playback_index_offset = ULI_PLAYBACK_INDEX; chip->capture_index_offset = ULI_CAPTURE_INDEX; break; + case AZX_DRIVER_ATIHDMI: + chip->playback_streams = ATIHDMI_NUM_PLAYBACK; + chip->capture_streams = ATIHDMI_NUM_CAPTURE; + chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX; + chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX; + break; default: chip->playback_streams = ICH6_NUM_PLAYBACK; chip->capture_streams = ICH6_NUM_CAPTURE; @@ -1621,6 +1636,7 @@ static struct pci_device_id azx_ids[] __devinitdata = { { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ + { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h index acaef3c811..0b668793fa 100644 --- a/sound/pci/hda/hda_patch.h +++ b/sound/pci/hda/hda_patch.h @@ -12,6 +12,8 @@ extern struct hda_codec_preset snd_hda_preset_analog[]; extern struct hda_codec_preset snd_hda_preset_sigmatel[]; /* SiLabs 3054/3055 modem codecs */ extern struct hda_codec_preset snd_hda_preset_si3054[]; +/* ATI HDMI codecs */ +extern struct hda_codec_preset snd_hda_preset_atihdmi[]; static const struct hda_codec_preset *hda_preset_tables[] = { snd_hda_preset_realtek, @@ -19,5 +21,6 @@ static const struct hda_codec_preset *hda_preset_tables[] = { snd_hda_preset_analog, snd_hda_preset_sigmatel, snd_hda_preset_si3054, + snd_hda_preset_atihdmi, NULL }; diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c new file mode 100644 index 0000000000..a27440ffd1 --- /dev/null +++ b/sound/pci/hda/patch_atihdmi.c @@ -0,0 +1,165 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * HD audio interface patch for ATI HDMI codecs + * + * Copyright (c) 2006 ATI Technologies Inc. + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + +struct atihdmi_spec { + struct hda_multi_out multiout; + + struct hda_pcm pcm_rec; +}; + +static struct hda_verb atihdmi_basic_init[] = { + /* enable digital output on pin widget */ + { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {} /* terminator */ +}; + +/* + * Controls + */ +static int atihdmi_build_controls(struct hda_codec *codec) +{ + struct atihdmi_spec *spec = codec->spec; + int err; + + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + + return 0; +} + +static int atihdmi_init(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, atihdmi_basic_init); + return 0; +} + +#ifdef CONFIG_PM +/* + * resume + */ +static int atihdmi_resume(struct hda_codec *codec) +{ + atihdmi_init(codec); + snd_hda_resume_spdif_out(codec); + + return 0; +} +#endif + +/* + * Digital out + */ +static int atihdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct atihdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int atihdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct atihdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static struct hda_pcm_stream atihdmi_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0x2, /* NID to query formats and rates and setup streams */ + .ops = { + .open = atihdmi_dig_playback_pcm_open, + .close = atihdmi_dig_playback_pcm_close + }, +}; + +static int atihdmi_build_pcms(struct hda_codec *codec) +{ + struct atihdmi_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "ATI HDMI"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback; + + return 0; +} + +static void atihdmi_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +static struct hda_codec_ops atihdmi_patch_ops = { + .build_controls = atihdmi_build_controls, + .build_pcms = atihdmi_build_pcms, + .init = atihdmi_init, + .free = atihdmi_free, +#ifdef CONFIG_PM + .resume = atihdmi_resume, +#endif +}; + +static int patch_atihdmi(struct hda_codec *codec) +{ + struct atihdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = 0x2; /* NID for copying analog to digital, + * seems to be unused in pure-digital + * case. */ + + codec->patch_ops = atihdmi_patch_ops; + + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_atihdmi[] = { + { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, + {} /* terminator */ +}; -- cgit v1.2.2 From 0fbf405c583e6ee6d7227eb938a096d0998f7e78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 May 2006 17:10:35 +0200 Subject: [ALSA] Fix pcm-draining of capture stream in PCM middle layer Fix the draining of PCM capture stream in the PCM middle layer. snd_pcm_drain() ignored capture streams, but it should change the state to SNDRV_PCM_DRAINING. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a998f88e3f..9e495244ee 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1469,8 +1469,6 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } up_read(&snd_pcm_link_rwsem); - if (! num_drecs) - goto _error; snd_pcm_stream_lock_irq(substream); /* resume pause */ -- cgit v1.2.2 From 6581f4e74d8541dd7d579f64e94822622cbb1654 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 May 2006 17:14:51 +0200 Subject: [ALSA] Remove zero-initialization of static variables Removed zero-initializations of static variables. A tiny optimization. Signed-off-by: Takashi Iwai --- sound/arm/sa11xx-uda1341.c | 2 +- sound/core/info.c | 8 ++++---- sound/core/init.c | 6 +++--- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 2 +- sound/core/rawmidi.c | 2 +- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_dummy.c | 2 +- sound/core/sound.c | 2 +- sound/core/sound_oss.c | 2 +- sound/core/timer.c | 2 +- sound/drivers/virmidi.c | 2 +- sound/isa/gus/interwave.c | 4 ++-- sound/isa/opl3sa2.c | 2 +- sound/isa/sb/emu8000_patch.c | 2 +- sound/isa/sb/sb16.c | 2 +- sound/isa/wavefront/wavefront.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/au88x0/au88x0_xtalk.c | 29 ++++++----------------------- sound/pci/bt87x.c | 2 +- sound/pci/cs46xx/cs46xx.c | 4 ++-- sound/pci/emu10k1/emu10k1.c | 8 ++++---- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sonicvibes.c | 4 ++-- sound/sparc/dbri.c | 4 ++-- 29 files changed, 46 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 9211348824..b88fb0c5a6 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -112,7 +112,7 @@ MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); -static char *id = NULL; /* ID for this card */ +static char *id; /* ID for this card */ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); diff --git a/sound/core/info.c b/sound/core/info.c index c8eeaea9d6..10c1772bf3 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -143,12 +143,12 @@ EXPORT_SYMBOL(snd_iprintf); */ -static struct proc_dir_entry *snd_proc_root = NULL; -struct snd_info_entry *snd_seq_root = NULL; +static struct proc_dir_entry *snd_proc_root; +struct snd_info_entry *snd_seq_root; EXPORT_SYMBOL(snd_seq_root); #ifdef CONFIG_SND_OSSEMUL -struct snd_info_entry *snd_oss_root = NULL; +struct snd_info_entry *snd_oss_root; #endif static inline void snd_info_entry_prepare(struct proc_dir_entry *de) @@ -972,7 +972,7 @@ EXPORT_SYMBOL(snd_info_unregister); */ -static struct snd_info_entry *snd_info_version_entry = NULL; +static struct snd_info_entry *snd_info_version_entry; static void snd_info_version_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { diff --git a/sound/core/init.c b/sound/core/init.c index 38b2d4a9d6..4d9258884e 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -38,8 +38,8 @@ struct snd_shutdown_f_ops { struct snd_shutdown_f_ops *next; }; -static unsigned int snd_cards_lock = 0; /* locked for registering/using */ -struct snd_card *snd_cards[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = NULL}; +static unsigned int snd_cards_lock; /* locked for registering/using */ +struct snd_card *snd_cards[SNDRV_CARDS]; EXPORT_SYMBOL(snd_cards); static DEFINE_MUTEX(snd_card_mutex); @@ -529,7 +529,7 @@ int snd_card_register(struct snd_card *card) EXPORT_SYMBOL(snd_card_register); #ifdef CONFIG_PROC_FS -static struct snd_info_entry *snd_card_info_entry = NULL; +static struct snd_info_entry *snd_card_info_entry; static void snd_card_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 4395285aa6..f5ff4f4a16 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -45,7 +45,7 @@ #define OSS_ALSAEMULVER _SIOR ('M', 249, int) -static int dsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 0}; +static int dsp_map[SNDRV_CARDS]; static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static int nonblock_open = 1; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index bc00f9b00c..7581edd7b9 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1072,7 +1072,7 @@ static void snd_pcm_proc_read(struct snd_info_entry *entry, mutex_unlock(®ister_mutex); } -static struct snd_info_entry *snd_pcm_proc_entry = NULL; +static struct snd_info_entry *snd_pcm_proc_entry; static void snd_pcm_proc_init(void) { diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 08a41e5023..8c15c66eb4 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -43,7 +43,7 @@ MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA."); MODULE_LICENSE("GPL"); #ifdef CONFIG_SND_OSSEMUL -static int midi_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 0}; +static int midi_map[SNDRV_CARDS]; static int amidi_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; module_param_array(midi_map, int, NULL, 0444); MODULE_PARM_DESC(midi_map, "Raw MIDI device number assigned to 1st OSS device."); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 1e4bc402f0..d812dc8863 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -80,7 +80,7 @@ static LIST_HEAD(opslist); static int num_ops; static DEFINE_MUTEX(ops_mutex); #ifdef CONFIG_PROC_FS -static struct snd_info_entry *info_entry = NULL; +static struct snd_info_entry *info_entry; #endif /* diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index 9eb1c744f7..e55488d123 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -66,7 +66,7 @@ MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-seq-client-" __stringify(SNDRV_SEQ_CLIENT_DUMMY)); static int ports = 1; -static int duplex = 0; +static int duplex; module_param(ports, int, 0444); MODULE_PARM_DESC(ports, "number of ports to be created"); diff --git a/sound/core/sound.c b/sound/core/sound.c index 02c8cc4ebf..cd86272834 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -332,7 +332,7 @@ EXPORT_SYMBOL(snd_unregister_device); * INFO PART */ -static struct snd_info_entry *snd_minor_info_entry = NULL; +static struct snd_info_entry *snd_minor_info_entry; static const char *snd_device_type_name(int type) { diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 0043c9a97d..74f0fe5a1b 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -209,7 +209,7 @@ EXPORT_SYMBOL(snd_unregister_oss_device); #ifdef CONFIG_PROC_FS -static struct snd_info_entry *snd_minor_info_oss_entry = NULL; +static struct snd_info_entry *snd_minor_info_oss_entry; static const char *snd_oss_device_type_name(int type) { diff --git a/sound/core/timer.c b/sound/core/timer.c index d92f73c2c6..78199f58b9 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1106,7 +1106,7 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, mutex_unlock(®ister_mutex); } -static struct snd_info_entry *snd_timer_proc_entry = NULL; +static struct snd_info_entry *snd_timer_proc_entry; static void __init snd_timer_proc_init(void) { diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 59171f8200..72d09b304d 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -65,7 +65,7 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual rawmidi device}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; +static int enable[SNDRV_CARDS]; static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; module_param_array(index, int, NULL, 0444); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 4298d339e7..866300f2ac 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -70,9 +70,9 @@ static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ static int joystick_dac[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 29}; /* 0 to 31, (0.59V-4.52V or 0.389V-2.98V) */ -static int midi[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int midi[SNDRV_CARDS]; static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; -static int effect[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int effect[SNDRV_CARDS]; #ifdef SNDRV_STB #define PFX "interwave-stb: " diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 6d889052c3..931ff75e54 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -59,7 +59,7 @@ static long midi_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x330,0x300 */ static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 0,1,3,5,9,11,12,15 */ static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */ static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */ -static int opl3sa3_ymode[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* 0,1,2,3 */ /*SL Added*/ +static int opl3sa3_ymode[SNDRV_CARDS]; /* 0,1,2,3 */ /*SL Added*/ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for OPL3-SA soundcard."); diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index 80b1cf84a1..1be16c9700 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -23,7 +23,7 @@ #include #include -static int emu8000_reset_addr = 0; +static int emu8000_reset_addr; module_param(emu8000_reset_addr, int, 0444); MODULE_PARM_DESC(emu8000_reset_addr, "reset write address at each time (makes slowdown)"); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 6333f900ea..7f7f05fa51 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -85,7 +85,7 @@ static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 5,6,7 */ static int mic_agc[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; #ifdef CONFIG_SND_SB16_CSP -static int csp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int csp[SNDRV_CARDS]; #endif #ifdef SNDRV_SBAWE_EMU8000 static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 7ae86f82c3..9eb27082c6 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -50,7 +50,7 @@ static int ics2115_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 2,9,11,12,15 */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ -static int use_cs4232_midi[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int use_cs4232_midi[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for WaveFront soundcard."); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 4f01ef10fa..5dfdbf6657 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -49,7 +49,7 @@ MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}"); static int index = SNDRV_DEFAULT_IDX1; /* Index */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int pcm_channels = 32; -static int spdif = 0; +static int spdif; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio."); diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index 4534e1882a..b4151e208b 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -66,31 +66,20 @@ static xtalk_gains_t const asXtalkGainsAllChan = { 0 //0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff }; -static xtalk_gains_t const asXtalkGainsZeros = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -}; +static xtalk_gains_t const asXtalkGainsZeros; -static xtalk_dline_t const alXtalkDlineZeros = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0 -}; +static xtalk_dline_t const alXtalkDlineZeros; static xtalk_dline_t const alXtalkDlineTest = { 0xFC18, 0x03E8FFFF, 0x186A0, 0x7960FFFE, 1, 0xFFFFFFFF, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }; -static xtalk_instate_t const asXtalkInStateZeros = { 0, 0, 0, 0 }; +static xtalk_instate_t const asXtalkInStateZeros; static xtalk_instate_t const asXtalkInStateTest = { 0xFF80, 0x0080, 0xFFFF, 0x0001 }; -static xtalk_state_t const asXtalkOutStateZeros = { - {0, 0, 0, 0}, - {0, 0, 0, 0}, - {0, 0, 0, 0}, - {0, 0, 0, 0}, - {0, 0, 0, 0} -}; +static xtalk_state_t const asXtalkOutStateZeros; + static short const sDiamondKLeftEq = 0x401d; static short const sDiamondKRightEq = 0x401d; static short const sDiamondKLeftXt = 0xF90E; @@ -162,13 +151,7 @@ static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkCoefsZeros = { - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; +static xtalk_coefs_t const asXtalkCoefsZeros; static xtalk_coefs_t const asXtalkCoefsPipe = { {0, 0, 0x0FA0, 0, 0}, {0, 0, 0x0FA0, 0, 0}, diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 9ee07d4aac..aa21cc74a8 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -44,7 +44,7 @@ MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878}," static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int digital_rate[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* digital input rate */ +static int digital_rate[SNDRV_CARDS]; /* digital input rate */ static int load_all; /* allow to load the non-whitelisted cards */ module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 848d772ae3..772dc52bfe 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -48,8 +48,8 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static int thinkpad[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int external_amp[SNDRV_CARDS]; +static int thinkpad[SNDRV_CARDS]; static int mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 42b11ba1d2..549673ea14 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -46,13 +46,13 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int extin[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static int extout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int extin[SNDRV_CARDS]; +static int extout[SNDRV_CARDS]; static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; -static int enable_ir[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static uint subsystem[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; /* Force card subsystem model */ +static int enable_ir[SNDRV_CARDS]; +static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5ff4175c7b..f43bd380ac 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -132,7 +132,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * static int total_bufsize[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1024 }; static int pcm_substreams_p[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4 }; static int pcm_substreams_c[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1 }; -static int clock[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int clock[SNDRV_CARDS]; static int use_pm[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; #ifdef SUPPORT_JOYSTICK diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index d72fc28c58..0ec90f3773 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -56,7 +56,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 3 = MediaForte 64-PCR * High 16-bits are video (radio) device number + 1 */ -static int tea575x_tuner[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; +static int tea575x_tuner[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the FM801 soundcard."); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index a4e5b8115a..e09fb7f9e7 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -66,7 +66,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ -static int ac97_clock = 0; +static int ac97_clock; static char *ac97_quirk; static int buggy_semaphore; static int buggy_irq = -1; /* auto-check */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 20acb1a7e9..24703d75b6 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -59,7 +59,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," static int index = -2; /* Exclude the first card */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ -static int ac97_clock = 0; +static int ac97_clock; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel i8x0 modemcard."); diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index ac14b2733f..3b945e8c1b 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -41,7 +41,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int precise_ptr[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* Enable precise pointer */ +static int precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard."); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index a783041729..51775706c8 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -54,8 +54,8 @@ MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int reverb[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static int mge[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int reverb[SNDRV_CARDS]; +static int mge[SNDRV_CARDS]; static unsigned int dmaio = 0x7a00; /* DDMA i/o address */ module_param_array(index, int, NULL, 0444); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index db6539126d..5eecdd09a7 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -92,7 +92,7 @@ MODULE_PARM_DESC(enable, "Enable Sun DBRI soundcard."); #define D_USR (1<<4) #define D_DESC (1<<5) -static int dbri_debug = 0; +static int dbri_debug; module_param(dbri_debug, int, 0644); MODULE_PARM_DESC(dbri_debug, "Debug value for Sun DBRI soundcard."); @@ -593,7 +593,7 @@ struct snd_dbri { /* Return a pointer to dbri_streaminfo */ #define DBRI_STREAM(dbri, substream) &dbri->stream_info[DBRI_STREAMNO(substream)] -static struct snd_dbri *dbri_list = NULL; /* All DBRI devices */ +static struct snd_dbri *dbri_list; /* All DBRI devices */ /* * Short data pipes transmit LSB first. The CS4215 receives MSB first. Grrr. -- cgit v1.2.2 From 474167d646cb2147b9fcd7bacf5cdf8177ed43c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 May 2006 17:17:43 +0200 Subject: [ALSA] hda-codec - Fix init verbs for ALC260 hp model Use the basic init verbs for ALC260 instead of hp init verbs since hp init verbs seem incomplete and not working on some machines. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fc2f77dce..ceb103b93b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3106,6 +3106,7 @@ static struct hda_verb alc260_init_verbs[] = { { } }; +#if 0 /* should be identical with alc260_init_verbs? */ static struct hda_verb alc260_hp_init_verbs[] = { /* Headphone and output */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, @@ -3152,6 +3153,7 @@ static struct hda_verb alc260_hp_init_verbs[] = { {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, { } }; +#endif static struct hda_verb alc260_hp_3013_init_verbs[] = { /* Line out and output */ @@ -3867,7 +3869,7 @@ static struct alc_config_preset alc260_presets[] = { .mixers = { alc260_base_output_mixer, alc260_input_mixer, alc260_capture_alt_mixer }, - .init_verbs = { alc260_hp_init_verbs }, + .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), -- cgit v1.2.2 From faf8d11743961c720c85be191f8a08c00e5c5d60 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 18 May 2006 09:35:15 +0200 Subject: [ALSA] usb-audio: add workaround for CSR Bluetooth Headphones (Saitek A-250) Some devices don't have the required class-specific endpoint descriptors. Instead of making this case an error, this patch makes the driver guess the endpoint attributes. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8100516e1f..770642a595 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2627,9 +2627,10 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) { - snd_printk(KERN_ERR "%d:%u:%d : no or invalid class specific endpoint descriptor\n", + snd_printk(KERN_WARN "%d:%u:%d : no or invalid" + " class specific endpoint descriptor\n", dev->devnum, iface_no, altno); - continue; + csep = NULL; } fp = kmalloc(sizeof(*fp), GFP_KERNEL); @@ -2648,7 +2649,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); - fp->attributes = csep[3]; + fp->attributes = csep ? csep[3] : 0; /* some quirks for attributes here */ -- cgit v1.2.2 From c77a03551b3fd8ef6434153dfadff83ae404e526 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Thu, 18 May 2006 14:24:30 +0200 Subject: [ALSA] Remove ENTER_UART from au88x0 init Remove an unnecessary ENTER_UART instruction during au88x0 init as it makes the first/subsequent midi open to fail. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_mpu401.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 814bc2db9f..1e128a3c8d 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -70,9 +70,6 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) temp |= (MIDI_CLOCK_DIV << 8) | ((mode >> 24) & 0xff) << 4; hwwrite(vortex->mmio, VORTEX_CTRL2, temp); hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_RESET); - /* Set some kind of mode */ - if (mode) - hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_ENTER_UART); /* Check if anything is OK. */ temp = hwread(vortex->mmio, VORTEX_MIDI_DATA); -- cgit v1.2.2 From f8c7579051763d6be275bf88a430ffb1c5234bad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 May 2006 14:47:03 +0200 Subject: [ALSA] usbaudio - Fix a typo Fix a typo introduced by the last fix. Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 770642a595..30cadec9a3 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2627,7 +2627,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) { - snd_printk(KERN_WARN "%d:%u:%d : no or invalid" + snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" " class specific endpoint descriptor\n", dev->devnum, iface_no, altno); csep = NULL; -- cgit v1.2.2 From 2851d963e0038c53d2175970daac4217abed7af2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 May 2006 14:48:26 +0200 Subject: [ALSA] mpu401_uart - Fix coding style and code clean up - fold lines and fix spaces to follow the standard style - added some comments - moved EXPORT_SYMBOL() near the definition - some code clean up Signed-off-by: Takashi Iwai --- sound/drivers/mpu401/mpu401_uart.c | 108 ++++++++++++++++++++++--------------- 1 file changed, 66 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index b49a45cbf6..cd64d3eb9e 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -58,22 +58,26 @@ static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu); #define MPU401_ACK 0xfe /* Build in lowlevel io */ -static void mpu401_write_port(struct snd_mpu401 *mpu, unsigned char data, unsigned long addr) +static void mpu401_write_port(struct snd_mpu401 *mpu, unsigned char data, + unsigned long addr) { outb(data, addr); } -static unsigned char mpu401_read_port(struct snd_mpu401 *mpu, unsigned long addr) +static unsigned char mpu401_read_port(struct snd_mpu401 *mpu, + unsigned long addr) { return inb(addr); } -static void mpu401_write_mmio(struct snd_mpu401 *mpu, unsigned char data, unsigned long addr) +static void mpu401_write_mmio(struct snd_mpu401 *mpu, unsigned char data, + unsigned long addr) { writeb(data, (void __iomem *)addr); } -static unsigned char mpu401_read_mmio(struct snd_mpu401 *mpu, unsigned long addr) +static unsigned char mpu401_read_mmio(struct snd_mpu401 *mpu, + unsigned long addr) { return readb((void __iomem *)addr); } @@ -86,20 +90,22 @@ static void snd_mpu401_uart_clear_rx(struct snd_mpu401 *mpu) mpu->read(mpu, MPU401D(mpu)); #ifdef CONFIG_SND_DEBUG if (timeout <= 0) - snd_printk("cmd: clear rx timeout (status = 0x%x)\n", mpu->read(mpu, MPU401C(mpu))); + snd_printk(KERN_ERR "cmd: clear rx timeout (status = 0x%x)\n", + mpu->read(mpu, MPU401C(mpu))); #endif } static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu) { spin_lock(&mpu->input_lock); - if (test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode)) { + if (test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode)) snd_mpu401_uart_input_read(mpu); - } else { + else snd_mpu401_uart_clear_rx(mpu); - } spin_unlock(&mpu->input_lock); - /* ok. for better Tx performance try do some output when input is done */ + /* ok. for better Tx performance try do some output when + * input is done + */ if (test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode) && test_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode)) { spin_lock(&mpu->output_lock); @@ -116,7 +122,8 @@ static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu) * * Processes the interrupt for MPU401-UART i/o. */ -irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id, struct pt_regs *regs) +irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id, + struct pt_regs *regs) { struct snd_mpu401 *mpu = dev_id; @@ -126,6 +133,8 @@ irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id, struct pt_regs *reg return IRQ_HANDLED; } +EXPORT_SYMBOL(snd_mpu401_uart_interrupt); + /* * timer callback * reprogram the timer and call the interrupt job @@ -159,7 +168,8 @@ static void snd_mpu401_uart_add_timer (struct snd_mpu401 *mpu, int input) mpu->timer.expires = 1 + jiffies; add_timer(&mpu->timer); } - mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER : MPU401_MODE_OUTPUT_TIMER; + mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER : + MPU401_MODE_OUTPUT_TIMER; spin_unlock_irqrestore (&mpu->timer_lock, flags); } @@ -172,7 +182,8 @@ static void snd_mpu401_uart_remove_timer (struct snd_mpu401 *mpu, int input) spin_lock_irqsave (&mpu->timer_lock, flags); if (mpu->timer_invoked) { - mpu->timer_invoked &= input ? ~MPU401_MODE_INPUT_TIMER : ~MPU401_MODE_OUTPUT_TIMER; + mpu->timer_invoked &= input ? ~MPU401_MODE_INPUT_TIMER : + ~MPU401_MODE_OUTPUT_TIMER; if (! mpu->timer_invoked) del_timer(&mpu->timer); } @@ -180,11 +191,12 @@ static void snd_mpu401_uart_remove_timer (struct snd_mpu401 *mpu, int input) } /* - + * send a UART command + * return zero if successful, non-zero for some errors */ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd, - int ack) + int ack) { unsigned long flags; int timeout, ok; @@ -196,11 +208,13 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd, } /* ok. standard MPU-401 initialization */ if (mpu->hardware != MPU401_HW_SB) { - for (timeout = 1000; timeout > 0 && !snd_mpu401_output_ready(mpu); timeout--) + for (timeout = 1000; timeout > 0 && + !snd_mpu401_output_ready(mpu); timeout--) udelay(10); #ifdef CONFIG_SND_DEBUG if (!timeout) - snd_printk("cmd: tx timeout (status = 0x%x)\n", mpu->read(mpu, MPU401C(mpu))); + snd_printk(KERN_ERR "cmd: tx timeout (status = 0x%x)\n", + mpu->read(mpu, MPU401C(mpu))); #endif } mpu->write(mpu, cmd, MPU401C(mpu)); @@ -215,12 +229,14 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd, } if (!ok && mpu->read(mpu, MPU401D(mpu)) == MPU401_ACK) ok = 1; - } else { + } else ok = 1; - } spin_unlock_irqrestore(&mpu->input_lock, flags); if (!ok) { - snd_printk("cmd: 0x%x failed at 0x%lx (status = 0x%x, data = 0x%x)\n", cmd, mpu->port, mpu->read(mpu, MPU401C(mpu)), mpu->read(mpu, MPU401D(mpu))); + snd_printk(KERN_ERR "cmd: 0x%x failed at 0x%lx " + "(status = 0x%x, data = 0x%x)\n", cmd, mpu->port, + mpu->read(mpu, MPU401C(mpu)), + mpu->read(mpu, MPU401D(mpu))); return 1; } return 0; @@ -314,7 +330,8 @@ static int snd_mpu401_uart_output_close(struct snd_rawmidi_substream *substream) /* * trigger input callback */ -static void snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) +static void +snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) { unsigned long flags; struct snd_mpu401 *mpu; @@ -322,7 +339,8 @@ static void snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substrea mpu = substream->rmidi->private_data; if (up) { - if (! test_and_set_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) { + if (! test_and_set_bit(MPU401_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) { /* first time - flush FIFO */ while (max-- > 0) mpu->read(mpu, MPU401D(mpu)); @@ -352,13 +370,11 @@ static void snd_mpu401_uart_input_read(struct snd_mpu401 * mpu) unsigned char byte; while (max-- > 0) { - if (snd_mpu401_input_avail(mpu)) { - byte = mpu->read(mpu, MPU401D(mpu)); - if (test_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) - snd_rawmidi_receive(mpu->substream_input, &byte, 1); - } else { + if (! snd_mpu401_input_avail(mpu)) break; /* input not available */ - } + byte = mpu->read(mpu, MPU401D(mpu)); + if (test_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, &byte, 1); } } @@ -380,16 +396,16 @@ static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu) int max = 256, timeout; do { - if (snd_rawmidi_transmit_peek(mpu->substream_output, &byte, 1) == 1) { + if (snd_rawmidi_transmit_peek(mpu->substream_output, + &byte, 1) == 1) { for (timeout = 100; timeout > 0; timeout--) { - if (snd_mpu401_output_ready(mpu)) { - mpu->write(mpu, byte, MPU401D(mpu)); - snd_rawmidi_transmit_ack(mpu->substream_output, 1); + if (snd_mpu401_output_ready(mpu)) break; - } } if (timeout == 0) break; /* Tx FIFO full - try again later */ + mpu->write(mpu, byte, MPU401D(mpu)); + snd_rawmidi_transmit_ack(mpu->substream_output, 1); } else { snd_mpu401_uart_remove_timer (mpu, 0); break; /* no other data - leave the tx loop */ @@ -400,7 +416,8 @@ static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu) /* * output trigger callback */ -static void snd_mpu401_uart_output_trigger(struct snd_rawmidi_substream *substream, int up) +static void +snd_mpu401_uart_output_trigger(struct snd_rawmidi_substream *substream, int up) { unsigned long flags; struct snd_mpu401 *mpu; @@ -499,8 +516,11 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, mpu->hardware = hardware; if (!integrated) { int res_size = hardware == MPU401_HW_PC98II ? 4 : 2; - if ((mpu->res = request_region(port, res_size, "MPU401 UART")) == NULL) { - snd_printk(KERN_ERR "mpu401_uart: unable to grab port 0x%lx size %d\n", port, res_size); + mpu->res = request_region(port, res_size, "MPU401 UART"); + if (mpu->res == NULL) { + snd_printk(KERN_ERR "mpu401_uart: " + "unable to grab port 0x%lx size %d\n", + port, res_size); snd_device_free(card, rmidi); return -EBUSY; } @@ -521,8 +541,10 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, else mpu->cport = port + 1; if (irq >= 0 && irq_flags) { - if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags, "MPU401 UART", (void *) mpu)) { - snd_printk(KERN_ERR "mpu401_uart: unable to grab IRQ %d\n", irq); + if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags, + "MPU401 UART", (void *) mpu)) { + snd_printk(KERN_ERR "mpu401_uart: " + "unable to grab IRQ %d\n", irq); snd_device_free(card, rmidi); return -EBUSY; } @@ -530,11 +552,14 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, mpu->irq = irq; mpu->irq_flags = irq_flags; if (card->shortname[0]) - snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI", card->shortname); + snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI", + card->shortname); else - sprintf(rmidi->name, "MPU-401 MIDI %d-%d", card->number, device); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mpu401_uart_output); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mpu401_uart_input); + sprintf(rmidi->name, "MPU-401 MIDI %d-%d",card->number, device); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &snd_mpu401_uart_output); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_mpu401_uart_input); rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; @@ -544,7 +569,6 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, return 0; } -EXPORT_SYMBOL(snd_mpu401_uart_interrupt); EXPORT_SYMBOL(snd_mpu401_uart_new); /* -- cgit v1.2.2 From 77389b432344c811832962ca7f8181b8b3da3449 Mon Sep 17 00:00:00 2001 From: Jaya Kumar Date: Fri, 19 May 2006 12:04:22 +0200 Subject: [ALSA] Single variables for cs5535audio As per Takashi's feedback, this is a cleanup to make cs5535audio be single device per system. The diff is against 2.6.17-rc4 with Takashi's patch adding the module_params for index, id and enable. Signed-off-by: Jaya Kumar Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index f61c4fa4ed..8f46190f24 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -56,16 +56,17 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { {} }; -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static int index = SNDRV_DEFAULT_IDX1; +static char *id = SNDRV_DEFAULT_STR1; +/* for backward compatibility */ +static int enable; -module_param_array(index, int, NULL, 0444); +module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME); -module_param_array(id, charp, NULL, 0444); +module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable for " DRIVER_NAME); static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, @@ -357,12 +358,8 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci, if (dev >= SNDRV_CARDS) return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + card = snd_card_new(index, id, THIS_MODULE, 0); if (card == NULL) return -ENOMEM; -- cgit v1.2.2 From 140432fd2fbe68d59fe6fcddbcd4bcd0f84e951a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 May 2006 14:31:57 +0200 Subject: [ALSA] hdsp - Fix compilation with hdsp driver built in kernel Fixed the compilation with hdsp driver built in kernel. The traditional hwdep loader is used in this case. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index da63a1a199..bf89ab9e31 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -389,7 +389,7 @@ MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," /* use hotplug firmeare loader? */ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#ifndef HDSP_USE_HWDEP_LOADER +#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP) #define HDSP_FW_LOADER #endif #endif -- cgit v1.2.2 From 302e4c2f9e2b9f07c69649782330a61c60001ac4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 May 2006 13:24:30 +0200 Subject: [ALSA] Change an arugment of snd_mpu401_uart_new() to bit flags Change the 5th argument of snd_mpu401_uart_new() to bit flags instead of a boolean. The argument takes bits that consist of MPU401_INFO_XXX flags. The callers that used the value 1 there are replaced with MPU401_INFO_INTEGRATED. Signed-off-by: Takashi Iwai --- sound/drivers/mpu401/mpu401_uart.c | 96 ++++++++++++++++++++++++++----------- sound/isa/sscape.c | 3 +- sound/pci/als4000.c | 4 +- sound/pci/au88x0/au88x0_mpu401.c | 3 +- sound/pci/azt3328.c | 4 +- sound/pci/cmipci.c | 4 +- sound/pci/cs5535audio/cs5535audio.c | 21 ++++---- sound/pci/es1938.c | 3 +- sound/pci/es1968.c | 3 +- sound/pci/fm801.c | 3 +- sound/pci/ice1712/ice1712.c | 6 ++- sound/pci/ice1712/ice1724.c | 3 +- sound/pci/maestro3.c | 3 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 3 +- sound/pci/via82xx.c | 2 +- sound/pci/ymfpci/ymfpci.c | 3 +- 17 files changed, 110 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index cd64d3eb9e..4bf07ca9b1 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -95,17 +95,8 @@ static void snd_mpu401_uart_clear_rx(struct snd_mpu401 *mpu) #endif } -static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu) +static void uart_interrupt_tx(struct snd_mpu401 *mpu) { - spin_lock(&mpu->input_lock); - if (test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode)) - snd_mpu401_uart_input_read(mpu); - else - snd_mpu401_uart_clear_rx(mpu); - spin_unlock(&mpu->input_lock); - /* ok. for better Tx performance try do some output when - * input is done - */ if (test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode) && test_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode)) { spin_lock(&mpu->output_lock); @@ -114,6 +105,22 @@ static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu) } } +static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu) +{ + if (mpu->info_flags & MPU401_INFO_INPUT) { + spin_lock(&mpu->input_lock); + if (test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode)) + snd_mpu401_uart_input_read(mpu); + else + snd_mpu401_uart_clear_rx(mpu); + spin_unlock(&mpu->input_lock); + } + if (! (mpu->info_flags & MPU401_INFO_TX_IRQ)) + /* ok. for better Tx performance try do some output + when input is done */ + uart_interrupt_tx(mpu); +} + /** * snd_mpu401_uart_interrupt - generic MPU401-UART interrupt handler * @irq: the irq number @@ -135,6 +142,27 @@ irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id, EXPORT_SYMBOL(snd_mpu401_uart_interrupt); +/** + * snd_mpu401_uart_interrupt_tx - generic MPU401-UART transmit irq handler + * @irq: the irq number + * @dev_id: mpu401 instance + * @regs: the reigster + * + * Processes the interrupt for MPU401-UART output. + */ +irqreturn_t snd_mpu401_uart_interrupt_tx(int irq, void *dev_id, + struct pt_regs *regs) +{ + struct snd_mpu401 *mpu = dev_id; + + if (mpu == NULL) + return IRQ_NONE; + uart_interrupt_tx(mpu); + return IRQ_HANDLED; +} + +EXPORT_SYMBOL(snd_mpu401_uart_interrupt_tx); + /* * timer callback * reprogram the timer and call the interrupt job @@ -430,14 +458,16 @@ snd_mpu401_uart_output_trigger(struct snd_rawmidi_substream *substream, int up) * since the output timer might have been removed in * snd_mpu401_uart_output_write(). */ - snd_mpu401_uart_add_timer(mpu, 0); + if (! (mpu->info_flags & MPU401_INFO_TX_IRQ)) + snd_mpu401_uart_add_timer(mpu, 0); /* output pending data */ spin_lock_irqsave(&mpu->output_lock, flags); snd_mpu401_uart_output_write(mpu); spin_unlock_irqrestore(&mpu->output_lock, flags); } else { - snd_mpu401_uart_remove_timer(mpu, 0); + if (! (mpu->info_flags & MPU401_INFO_TX_IRQ)) + snd_mpu401_uart_remove_timer(mpu, 0); clear_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode); } } @@ -475,7 +505,7 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) * @device: the device index, zero-based * @hardware: the hardware type, MPU401_HW_XXXX * @port: the base address of MPU401 port - * @integrated: non-zero if the port was already reserved by the chip + * @info_flags: bitflags MPU401_INFO_XXX * @irq: the irq number, -1 if no interrupt for mpu * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved. * @rrawmidi: the pointer to store the new rawmidi instance @@ -490,17 +520,24 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) */ int snd_mpu401_uart_new(struct snd_card *card, int device, unsigned short hardware, - unsigned long port, int integrated, + unsigned long port, + unsigned int info_flags, int irq, int irq_flags, struct snd_rawmidi ** rrawmidi) { struct snd_mpu401 *mpu; struct snd_rawmidi *rmidi; + int in_enable, out_enable; int err; if (rrawmidi) *rrawmidi = NULL; - if ((err = snd_rawmidi_new(card, "MPU-401U", device, 1, 1, &rmidi)) < 0) + if (! (info_flags & (MPU401_INFO_INPUT | MPU401_INFO_OUTPUT))) + info_flags |= MPU401_INFO_INPUT | MPU401_INFO_OUTPUT; + in_enable = (info_flags & MPU401_INFO_INPUT) ? 1 : 0; + out_enable = (info_flags & MPU401_INFO_OUTPUT) ? 1 : 0; + if ((err = snd_rawmidi_new(card, "MPU-401U", device, + out_enable, in_enable, &rmidi)) < 0) return err; mpu = kzalloc(sizeof(*mpu), GFP_KERNEL); if (mpu == NULL) { @@ -514,7 +551,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, spin_lock_init(&mpu->output_lock); spin_lock_init(&mpu->timer_lock); mpu->hardware = hardware; - if (!integrated) { + if (! (info_flags & MPU401_INFO_INTEGRATED)) { int res_size = hardware == MPU401_HW_PC98II ? 4 : 2; mpu->res = request_region(port, res_size, "MPU401 UART"); if (mpu->res == NULL) { @@ -525,15 +562,12 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, return -EBUSY; } } - switch (hardware) { - case MPU401_HW_AUREAL: + if (info_flags & MPU401_INFO_MMIO) { mpu->write = mpu401_write_mmio; mpu->read = mpu401_read_mmio; - break; - default: + } else { mpu->write = mpu401_write_port; mpu->read = mpu401_read_port; - break; } mpu->port = port; if (hardware == MPU401_HW_PC98II) @@ -549,6 +583,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, return -EBUSY; } } + mpu->info_flags = info_flags; mpu->irq = irq; mpu->irq_flags = irq_flags; if (card->shortname[0]) @@ -556,13 +591,18 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, card->shortname); else sprintf(rmidi->name, "MPU-401 MIDI %d-%d",card->number, device); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, - &snd_mpu401_uart_output); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, - &snd_mpu401_uart_input); - rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; + if (out_enable) { + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &snd_mpu401_uart_output); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + } + if (in_enable) { + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_mpu401_uart_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + if (out_enable) + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; + } mpu->rmidi = rmidi; if (rrawmidi) *rrawmidi = rmidi; diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index d2a856f0fd..27271c9446 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -897,10 +897,9 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_rawmidi *rawmidi; int err; -#define MPU401_SHARE_HARDWARE 1 if ((err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, - port, MPU401_SHARE_HARDWARE, + port, MPU401_INFO_INTEGRATED, irq, SA_INTERRUPT, &rawmidi)) == 0) { struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 60423b1c67..a9f0806645 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -746,8 +746,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, card->shortname, chip->alt_port, chip->irq); if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, - gcr+0x30, 1, pci->irq, 0, - &chip->rmidi)) < 0) { + gcr+0x30, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi)) < 0) { printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", gcr+0x30); goto out_err; } diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 1e128a3c8d..5a0c53530f 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -95,7 +95,8 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA); if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port, - 1, 0, 0, &rmidi)) != 0) { + MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO, + 0, 0, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index e68056c815..6e62dafb66 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1806,8 +1806,8 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) card->private_data = chip; if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, - chip->mpu_port, 1, pci->irq, 0, - &chip->rmidi)) < 0) { + chip->mpu_port, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi)) < 0) { snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port); goto out_err; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index cb475ada2e..79bc60a520 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2981,7 +2981,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc if (iomidi > 0) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, - iomidi, integrated_midi, + iomidi, + (integrated_midi ? + MPU401_INFO_INTEGRATED : 0), cm->irq, 0, &cm->rmidi)) < 0) { printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi); } diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 8f46190f24..f61c4fa4ed 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -56,17 +56,16 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { {} }; -static int index = SNDRV_DEFAULT_IDX1; -static char *id = SNDRV_DEFAULT_STR1; -/* for backward compatibility */ -static int enable; +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -module_param(index, int, 0444); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME); -module_param(id, charp, 0444); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); -module_param(enable, bool, 0444); -MODULE_PARM_DESC(enable, "Enable for " DRIVER_NAME); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, @@ -358,8 +357,12 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci, if (dev >= SNDRV_CARDS) return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } - card = snd_card_new(index, id, THIS_MODULE, 0); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 6f9094ca4f..ca6603fe0b 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1756,7 +1756,8 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, } } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 1, chip->irq, 0, &chip->rmidi) < 0) { + chip->mpu_port, MPU401_INFO_INTEGRATED, + chip->irq, 0, &chip->rmidi) < 0) { printk(KERN_ERR "es1938: unable to initialize MPU-401\n"); } else { // this line is vital for MIDI interrupt handling on ess-solo1 diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index f43bd380ac..bfa0876e71 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2727,7 +2727,8 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, } if (enable_mpu[dev]) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->io_port + ESM_MPU401_PORT, 1, + chip->io_port + ESM_MPU401_PORT, + MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi)) < 0) { printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n"); } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 0ec90f3773..0afa573dd2 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1448,7 +1448,8 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, return err; } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801, - FM801_REG(chip, MPU401_DATA), 1, + FM801_REG(chip, MPU401_DATA), + MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 52de85e21b..00e565e1db 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2737,7 +2737,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (! c->no_mpu401) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG(ice, MPU1_CTRL), 1, + ICEREG(ice, MPU1_CTRL), + MPU401_INFO_INTEGRATED, ice->irq, 0, &ice->rmidi[0])) < 0) { snd_card_free(card); @@ -2752,7 +2753,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) { /* 2nd port used */ if ((err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, - ICEREG(ice, MPU2_CTRL), 1, + ICEREG(ice, MPU2_CTRL), + MPU401_INFO_INTEGRATED, ice->irq, 0, &ice->rmidi[1])) < 0) { snd_card_free(card); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1031bcbf70..34a58c629f 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2388,7 +2388,8 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (! c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG1724(ice, MPU_CTRL), 1, + ICEREG1724(ice, MPU_CTRL), + MPU401_INFO_INTEGRATED, ice->irq, 0, &ice->rmidi[0])) < 0) { snd_card_free(card); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 1928e06b6d..1c344fbd96 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2861,7 +2861,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) #if 0 /* TODO: not supported yet */ /* TODO enable MIDI IRQ and I/O */ err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401, - chip->iobase + MPU401_DATA_PORT, 1, + chip->iobase + MPU401_DATA_PORT, + MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi); if (err < 0) printk(KERN_WARNING "maestro3: no MIDI support.\n"); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 51775706c8..dcf4029483 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1456,7 +1456,7 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, return err; } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES, - sonic->midi_port, 1, + sonic->midi_port, MPU401_INFO_INTEGRATED, sonic->irq, 0, &midi_uart)) < 0) { snd_card_free(card); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 9624a5f2b8..5629b7eba9 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -148,7 +148,8 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, } if (trident->device != TRIDENT_DEVICE_ID_SI7018 && (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE, - trident->midi_port, 1, + trident->midi_port, + MPU401_INFO_INTEGRATED, trident->irq, 0, &trident->rmidi)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index a1b777e79c..12ce22ef16 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1973,7 +1973,7 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg); if (chip->mpu_res) { if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A, - mpu_port, 1, + mpu_port, MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi) < 0) { printk(KERN_WARNING "unable to initialize MPU-401" " at 0x%lx, skipping\n", mpu_port); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 65ebf5f193..26aa775b7b 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -308,7 +308,8 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, } if (chip->mpu_res) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI, - mpu_port[dev], 1, + mpu_port[dev], + MPU401_INFO_INTEGRATED, pci->irq, 0, &chip->rawmidi)) < 0) { printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]); legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */ -- cgit v1.2.2 From 721b8a297279276699900a662fa8299232ebc0e8 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Tue, 23 May 2006 13:29:51 +0200 Subject: [ALSA] ice1712 - Disable AC97 for DMX6fire Consumer AC97 is not used by the Terratec DMX6fire, but eeprom bit indicates it is; change the stored value to disable failing consumer mode. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 00e565e1db..aa5a41fecb 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2398,13 +2398,14 @@ static int __devinit snd_ice1712_chip_init(struct snd_ice1712 *ice) udelay(200); outb(ICE1712_NATIVE, ICEREG(ice, CONTROL)); udelay(200); - if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE && !ice->dxr_enable) { - /* Limit active ADCs and DACs to 6; */ - /* Note: DXR extension not supported */ - pci_write_config_byte(ice->pci, 0x60, 0x2a); - } else { - pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]); - } + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE && + !ice->dxr_enable) + /* Set eeprom value to limit active ADCs and DACs to 6; + * Also disable AC97 as no hardware in standard 6fire card/box + * Note: DXR extensions are not currently supported + */ + ice->eeprom.data[ICE_EEP1_CODEC] = 0x3a; + pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]); pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]); pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]); pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]); -- cgit v1.2.2 From d5a31b8b6e79145c832d530743ca80bf5f58a965 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 May 2006 13:30:59 +0200 Subject: [ALSA] au88x0 - Fix 64bit address of MPU401 MMIO port Fix 64bit address of MPU401 MMIO port on au88x0 chip. Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_mpu401.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 5a0c53530f..c75d368ea0 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -47,7 +47,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) struct snd_rawmidi *rmidi; int temp, mode; struct snd_mpu401 *mpu; - int port; + unsigned long port; #ifdef VORTEX_MPU401_LEGACY /* EnableHardCodedMPU401Port() */ -- cgit v1.2.2 From c51302710546f075e65b1e70487707e8324abf2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 May 2006 15:46:10 +0200 Subject: [ALSA] ice1724 - Add functionality for Audiotrak Prodigy 7.1 LT This patch adds support for useable front audio channels, user controllable headphone channel and optical output. From: Anho Ki Signed-off-by: Matt Taylor Signed-off-by: Takashi Iwai --- sound/pci/ice1712/aureon.c | 26 ++++++++++++++++---------- sound/pci/ice1712/aureon.h | 1 + 2 files changed, 17 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 336dc489ae..ca74f5b85f 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1281,9 +1281,15 @@ static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) tmp2 = tmp = snd_ice1712_gpio_read(ice); if (enable) - tmp |= AUREON_HP_SEL; + if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) + tmp |= AUREON_HP_SEL; + else + tmp |= PRODIGY_HP_SEL; else - tmp &= ~ AUREON_HP_SEL; + if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) + tmp &= ~ AUREON_HP_SEL; + else + tmp &= ~ PRODIGY_HP_SEL; if (tmp != tmp2) { snd_ice1712_gpio_write(ice, tmp); return 1; @@ -2079,16 +2085,16 @@ static unsigned char prodigy71_eeprom[] __devinitdata = { }; static unsigned char prodigy71lt_eeprom[] __devinitdata = { - 0x0b, /* SYSCINF: clock 512, spdif-in/ADC, 4DACs */ + 0x4b, /* SYSCINF: clock 512, spdif-in/ADC, 4DACs */ 0x80, /* ACLINK: I2S */ 0xfc, /* I2S: vol, 96k, 24bit, 192k */ - 0xc3, /* SPDUF: out-en, out-int */ - 0x00, /* GPIO_DIR */ - 0x07, /* GPIO_DIR1 */ - 0x00, /* GPIO_DIR2 */ - 0xff, /* GPIO_MASK */ - 0xf8, /* GPIO_MASK1 */ - 0xff, /* GPIO_MASK2 */ + 0xc3, /* SPDIF: out-en, out-int, spdif-in */ + 0xff, /* GPIO_DIR */ + 0xff, /* GPIO_DIR1 */ + 0x5f, /* GPIO_DIR2 */ + 0x00, /* GPIO_MASK */ + 0x00, /* GPIO_MASK1 */ + 0x00, /* GPIO_MASK2 */ 0x00, /* GPIO_STATE */ 0x00, /* GPIO_STATE1 */ 0x00, /* GPIO_STATE2 */ diff --git a/sound/pci/ice1712/aureon.h b/sound/pci/ice1712/aureon.h index 98a6752280..3b7bea656c 100644 --- a/sound/pci/ice1712/aureon.h +++ b/sound/pci/ice1712/aureon.h @@ -58,5 +58,6 @@ extern struct snd_ice1712_card_info snd_vt1724_aureon_cards[]; #define PRODIGY_WM_CS (1 << 8) #define PRODIGY_SPI_MOSI (1 << 10) #define PRODIGY_SPI_CLK (1 << 9) +#define PRODIGY_HP_SEL (1 << 5) #endif /* __SOUND_AUREON_H */ -- cgit v1.2.2 From 766a6c36f3a0b12e1c55dddc1df6673db6b22bfb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 26 May 2006 14:58:29 +0200 Subject: [ALSA] hda-codec - Fix model for HP dc7600 Changed the assigned model for HP dc7600 with ALC260 codec to match better with the actual I/O assignment. Patch taken from ALSA bug#2157. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ceb103b93b..98b9f16c26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3834,7 +3834,7 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "hp", .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, - { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3015, .config = ALC260_HP }, -- cgit v1.2.2 From cf78ee2ccc96d59e602188e0e6e3fe3522b6d3f6 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Fri, 26 May 2006 17:19:34 +0200 Subject: [ALSA] ice1712 - Set mpu401 info flags from _card_info MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To permit use, in ice1712, of the mpu401 info flags recently added to mpu401_uart, adds info_flags in snd_ice1712_card_info so that additional flags can be set, if desired.  'MPU401_INFO_INTEGRATED' is always set with the ice1712.  The flags are passed on to snd_mpu401_uart_new(). _INFO_OUTPUT is set for DMX6fire mpu2. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ews.c | 1 + sound/pci/ice1712/ice1712.c | 7 ++++--- sound/pci/ice1712/ice1712.h | 3 +++ 3 files changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 2e1cf11205..b135389fec 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -1033,6 +1033,7 @@ struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = { .build_controls = snd_ice1712_ews_add_controls, .mpu401_1_name = "MIDI-Front DMX6fire", .mpu401_2_name = "Wavetable DMX6fire", + .mpu401_2_info_flags = MPU401_INFO_OUTPUT, }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index aa5a41fecb..845907159b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -61,7 +61,6 @@ #include #include #include -#include #include #include @@ -2739,7 +2738,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (! c->no_mpu401) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, ICEREG(ice, MPU1_CTRL), - MPU401_INFO_INTEGRATED, + (c->mpu401_1_info_flags | + MPU401_INFO_INTEGRATED), ice->irq, 0, &ice->rmidi[0])) < 0) { snd_card_free(card); @@ -2755,7 +2755,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, /* 2nd port used */ if ((err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, ICEREG(ice, MPU2_CTRL), - MPU401_INFO_INTEGRATED, + (c->mpu401_2_info_flags | + MPU401_INFO_INTEGRATED), ice->irq, 0, &ice->rmidi[1])) < 0) { snd_card_free(card); diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index d4776319a0..ce27eac40d 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -29,6 +29,7 @@ #include #include #include +#include /* @@ -495,6 +496,8 @@ struct snd_ice1712_card_info { int (*chip_init)(struct snd_ice1712 *); int (*build_controls)(struct snd_ice1712 *); unsigned int no_mpu401: 1; + unsigned int mpu401_1_info_flags; + unsigned int mpu401_2_info_flags; const char *mpu401_1_name; const char *mpu401_2_name; unsigned int eeprom_size; -- cgit v1.2.2 From f26eb78fcfb5b76fbe6d3e740b6fedda611f8395 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 May 2006 19:05:28 +0200 Subject: [ALSA] cmipci - Fix a typo in 'PC Speaker Playback Switch' control Fixed a typo in 'PC Speaker Playback Switch' control name. Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 79bc60a520..0938c158b5 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2121,7 +2121,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7), CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7), CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0), - CMIPCI_DOUBLE("PC Speaker Playnack Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), + CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0), }; -- cgit v1.2.2 From d20cad602fc3d92902dc3b4ed252359ab05eae0f Mon Sep 17 00:00:00 2001 From: Eric Sesterhenn Date: Wed, 31 May 2006 11:55:17 +0200 Subject: [ALSA] NULL pointer dereference in sound/synth/emux/soundfont.c this is about coverity id #100. It seems the if statement is negated, since the else branch calls remove_info() with sflist->currsf as a parameter where it gets dereferenced. Signed-off-by: Eric Sesterhenn Signed-off-by: Takashi Iwai --- sound/synth/emux/soundfont.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 7f0bdea0df..455e535933 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -195,7 +195,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, break; case SNDRV_SFNT_REMOVE_INFO: /* patch must be opened */ - if (sflist->currsf) { + if (!sflist->currsf) { snd_printk("soundfont: remove_info: patch not opened\n"); rc = -EINVAL; } else { -- cgit v1.2.2 From 29463dfea754b8b360f638244f002c751aaad1b0 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 1 Jun 2006 08:33:48 +0200 Subject: [ALSA] bt87x: add Voodoo TV 200 whitelist entry This adds a whitelist entry for the digital audio input of the Voodoo TV 200. Signed-off-by: Clemens Ladisch --- sound/pci/bt87x.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index aa21cc74a8..c33642d8d9 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -781,10 +781,12 @@ static struct pci_device_id snd_bt87x_ids[] __devinitdata = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000), /* Viewcast Osprey 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100), - /* AVerMedia Studio No. 103, 203, ...? */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), /* Leadtek Winfast tv 2000xp delux */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000), + /* Voodoo TV 200 */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000), + /* AVerMedia Studio No. 103, 203, ...? */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), { } }; MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); -- cgit v1.2.2 From f079c25ab8a7d223875c5bac9b23b484e4a18f88 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Jun 2006 11:42:14 +0200 Subject: [ALSA] hda-intel - Fix race in remove Call iounmap after free_irq to avoid invalid accesses in the shared irq. The patch is taken from https://bugzilla.novell.com/show_bug.cgi?id=167869 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0154389bf9..4070b5cd9b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1402,10 +1402,10 @@ static int azx_free(struct azx *chip) msleep(1); } - if (chip->remap_addr) - iounmap(chip->remap_addr); if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); + if (chip->remap_addr) + iounmap(chip->remap_addr); if (chip->bdl.area) snd_dma_free_pages(&chip->bdl); -- cgit v1.2.2 From 0a50d2b2951cb7ae12726814f9a198e1c699aa0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Jun 2006 14:47:29 +0200 Subject: [ALSA] Fix possible races in PCI driver removal Call free_irq() before releasing others to avoid races when shared irq is issued. Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 5 +++-- sound/pci/riptide/riptide.c | 4 ++-- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 69dbf542a6..5c21144392 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2877,14 +2877,15 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) if (chip->region.idx[0].resource) snd_cs46xx_hw_stop(chip); + if (chip->irq >= 0) + free_irq(chip->irq, chip); + for (idx = 0; idx < 5; idx++) { struct snd_cs46xx_region *region = &chip->region.idx[idx]; if (region->remap_addr) iounmap(region->remap_addr); release_and_free_resource(region->resource); } - if (chip->irq >= 0) - free_irq(chip->irq, chip); if (chip->active_ctrl) chip->active_ctrl(chip, -chip->amplifier); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index c27cd49997..5618ec9740 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1836,11 +1836,11 @@ static int snd_riptide_free(struct snd_riptide *chip) UNSET_GRESET(cif->hwport); kfree(chip->cif); } + if (chip->irq >= 0) + free_irq(chip->irq, chip); if (chip->fw_entry) release_firmware(chip->fw_entry); release_and_free_resource(chip->res_port); - if (chip->irq >= 0) - free_irq(chip->irq, chip); kfree(chip); return 0; } -- cgit v1.2.2 From a43c4d4d7326c2894be9fd04519b109c438ee78b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Jun 2006 17:16:41 +0200 Subject: [ALSA] ac97 - Add Thinkpad T41p to AD1981 jack-sense blacklist Added Thinkpad T41p to the blacklist to disable HP/line jack-sensing with AD1981B. The jack-sensing is just harmful on this laptop. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 720b419e0c..1c47ee3d7e 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1627,6 +1627,7 @@ static const struct snd_kcontrol_new snd_ac97_ad1981x_jack_sense[] = { * (SS vendor << 16 | device) */ static unsigned int ad1981_jacks_blacklist[] = { + 0x10140537, /* Thinkpad T41p */ 0x10140554, /* Thinkpad T42p/R50p */ 0 /* end */ }; -- cgit v1.2.2 From f8e9f340da753c021c071f318f97ac9046c1316a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Jun 2006 21:08:53 +0200 Subject: [ALSA] hda-codec - Add model entry for HP nx6320 Added a model entry for HP nx6320 with AD1981HD codec. It wasn't covered by the generic HP entry because of a hardware bug (the SSID is reversed). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3a9b800db8..e83c7b01cb 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1368,6 +1368,8 @@ static struct hda_board_config ad1981_cfg_tbl[] = { { .modelname = "hp", .config = AD1981_HP }, /* All HP models */ { .pci_subvendor = 0x103c, .config = AD1981_HP }, + { .pci_subvendor = 0x30b0, .pci_subdevice = 0x103c, + .config = AD1981_HP }, /* HP nx6320 (reversed SSID, H/W bug) */ { .modelname = "thinkpad", .config = AD1981_THINKPAD }, /* Lenovo Thinkpad T60/X60/Z6xx */ { .pci_subvendor = 0x17aa, .config = AD1981_THINKPAD }, -- cgit v1.2.2 From c5533bf36b4a6629dab0e08c4951247050928853 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 2 Jun 2006 09:15:44 +0200 Subject: [ALSA] virmidi: revert erroneous removal of zero initialization The last patch that tried to remove zero initializations of static variables accidentally removed a not-quite-zero initialization too. Signed-off-by: Clemens Ladisch --- sound/drivers/virmidi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 72d09b304d..59171f8200 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -65,7 +65,7 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual rawmidi device}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS]; +static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; module_param_array(index, int, NULL, 0444); -- cgit v1.2.2 From 688956f23bdbfb1c3551bfafc819f989b36bb8ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Jun 2006 15:44:34 +0200 Subject: [ALSA] Fix races in irq handler and ioremap Call ioremap before request_irq for avoiding possible races in the irq handler. Signed-off-by: Takashi Iwai Signed-off-by: Takashi Iwai --- sound/pci/cs4281.c | 14 +++++++------- sound/pci/rme32.c | 12 ++++++------ sound/pci/rme96.c | 10 +++++----- 3 files changed, 18 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 8c150eab45..e77a4ce314 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1379,6 +1379,13 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); + chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); + chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + if (!chip->ba0 || !chip->ba1) { + snd_cs4281_free(chip); + return -ENOMEM; + } + if (request_irq(pci->irq, snd_cs4281_interrupt, SA_INTERRUPT|SA_SHIRQ, "CS4281", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); @@ -1387,13 +1394,6 @@ static int __devinit snd_cs4281_create(struct snd_card *card, } chip->irq = pci->irq; - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); - if (!chip->ba0 || !chip->ba1) { - snd_cs4281_free(chip); - return -ENOMEM; - } - tmp = snd_cs4281_chip_init(chip); if (tmp) { snd_cs4281_free(chip); diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 4dd53bfe03..2cb9fe98db 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1368,18 +1368,18 @@ static int __devinit snd_rme32_create(struct rme32 * rme32) return err; rme32->port = pci_resource_start(rme32->pci, 0); - if (request_irq(pci->irq, snd_rme32_interrupt, SA_INTERRUPT | SA_SHIRQ, "RME32", (void *) rme32)) { - snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); - return -EBUSY; - } - rme32->irq = pci->irq; - if ((rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE)) == 0) { snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme32->port, rme32->port + RME32_IO_SIZE - 1); return -ENOMEM; } + if (request_irq(pci->irq, snd_rme32_interrupt, SA_INTERRUPT | SA_SHIRQ, "RME32", (void *) rme32)) { + snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); + return -EBUSY; + } + rme32->irq = pci->irq; + /* read the card's revision number */ pci_read_config_byte(pci, 8, &rme32->rev); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 65611a7d36..991cb18c14 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1583,17 +1583,17 @@ snd_rme96_create(struct rme96 *rme96) return err; rme96->port = pci_resource_start(rme96->pci, 0); + if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) { + snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1); + return -ENOMEM; + } + if (request_irq(pci->irq, snd_rme96_interrupt, SA_INTERRUPT|SA_SHIRQ, "RME96", (void *)rme96)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); return -EBUSY; } rme96->irq = pci->irq; - if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) { - snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1); - return -ENOMEM; - } - /* read the card's revision number */ pci_read_config_byte(pci, 8, &rme96->rev); -- cgit v1.2.2 From 3758d4e601552a3d9066913a31ccb8dc6a25ee69 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Jun 2006 18:12:31 +0200 Subject: [ALSA] Remove bogus check of mmap_count in snd_pcm_release() Removed a bogus check of mmap_count in snd_pcm_release(). This is no longer true for the shared streams. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9e495244ee..439f047929 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2205,7 +2205,6 @@ static int snd_pcm_release(struct inode *inode, struct file *file) pcm_file = file->private_data; substream = pcm_file->substream; snd_assert(substream != NULL, return -ENXIO); - snd_assert(!atomic_read(&substream->mmap_count), ); pcm = substream->pcm; fasync_helper(-1, file, 0, &substream->runtime->fasync); mutex_lock(&pcm->open_mutex); -- cgit v1.2.2 From 3f3488b84c261ab3cb64b9f0b9f40b61e33edb98 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Thu, 8 Jun 2006 12:01:44 +0200 Subject: [ALSA] sound/vxpocket: fix printk warning Fix printk format warning: sound/pcmcia/vx/vxp_ops.c:205: warning: format '%x' expects type 'unsigned int', but argument 5 has type 'size_t' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vxp_ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 7f82f619f9..1ee0918c3b 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -202,7 +202,7 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware * c |= (int)vx_inb(chip, RXM) << 8; c |= vx_inb(chip, RXL); - snd_printdd(KERN_DEBUG "xilinx: dsp size received 0x%x, orig 0x%x\n", c, fw->size); + snd_printdd(KERN_DEBUG "xilinx: dsp size received 0x%x, orig 0x%Zx\n", c, fw->size); vx_outb(chip, ICR, ICR_HF0); -- cgit v1.2.2 From 2e74eba3e2f000184ade92833c80e799af41c180 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Jun 2006 15:28:07 +0200 Subject: [ALSA] Fix invalid __init in ALSA ISA drivers Replaced invalid __init with __devinit in snd-sbawe and snd-opl3sa2 drivers. Signed-off-by: Takashi Iwai --- sound/isa/opl3sa2.c | 10 +++++----- sound/isa/sb/emu8000.c | 22 +++++++++++----------- 2 files changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 931ff75e54..647a996791 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -221,7 +221,7 @@ static void snd_opl3sa2_write(struct snd_opl3sa2 *chip, unsigned char reg, unsig spin_unlock_irqrestore(&chip->reg_lock, flags); } -static int __init snd_opl3sa2_detect(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) { struct snd_card *card; unsigned long port; @@ -489,7 +489,7 @@ static void snd_opl3sa2_master_free(struct snd_kcontrol *kcontrol) chip->master_volume = NULL; } -static int __init snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -583,8 +583,8 @@ static int snd_opl3sa2_resume(struct snd_card *card) #endif /* CONFIG_PM */ #ifdef CONFIG_PNP -static int __init snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, - struct pnp_dev *pdev) +static int __devinit snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, + struct pnp_dev *pdev) { struct pnp_resource_table * cfg; int err; @@ -862,7 +862,7 @@ static struct pnp_card_driver opl3sa2_pnpc_driver = { }; #endif /* CONFIG_PNP */ -static int __init snd_opl3sa2_nonpnp_probe(struct platform_device *pdev) +static int __devinit snd_opl3sa2_nonpnp_probe(struct platform_device *pdev) { struct snd_card *card; int err; diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index c0b8d61b75..658179e861 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -131,7 +131,7 @@ snd_emu8000_dma_chan(struct snd_emu8000 *emu, int ch, int mode) /* */ -static void __init +static void __devinit snd_emu8000_read_wait(struct snd_emu8000 *emu) { while ((EMU8000_SMALR_READ(emu) & 0x80000000) != 0) { @@ -143,7 +143,7 @@ snd_emu8000_read_wait(struct snd_emu8000 *emu) /* */ -static void __init +static void __devinit snd_emu8000_write_wait(struct snd_emu8000 *emu) { while ((EMU8000_SMALW_READ(emu) & 0x80000000) != 0) { @@ -156,7 +156,7 @@ snd_emu8000_write_wait(struct snd_emu8000 *emu) /* * detect a card at the given port */ -static int __init +static int __devinit snd_emu8000_detect(struct snd_emu8000 *emu) { /* Initialise */ @@ -182,7 +182,7 @@ snd_emu8000_detect(struct snd_emu8000 *emu) /* * intiailize audio channels */ -static void __init +static void __devinit init_audio(struct snd_emu8000 *emu) { int ch; @@ -223,7 +223,7 @@ init_audio(struct snd_emu8000 *emu) /* * initialize DMA address */ -static void __init +static void __devinit init_dma(struct snd_emu8000 *emu) { EMU8000_SMALR_WRITE(emu, 0); @@ -327,7 +327,7 @@ static unsigned short init4[128] /*__devinitdata*/ = { * Taken from the oss driver, not obvious from the doc how this * is meant to work */ -static void __init +static void __devinit send_array(struct snd_emu8000 *emu, unsigned short *data, int size) { int i; @@ -349,7 +349,7 @@ send_array(struct snd_emu8000 *emu, unsigned short *data, int size) * Send initialization arrays to start up, this just follows the * initialisation sequence in the adip. */ -static void __init +static void __devinit init_arrays(struct snd_emu8000 *emu) { send_array(emu, init1, ARRAY_SIZE(init1)/4); @@ -375,7 +375,7 @@ init_arrays(struct snd_emu8000 *emu) * seems that the only way to do this is to use the one channel and keep * reallocating between read and write. */ -static void __init +static void __devinit size_dram(struct snd_emu8000 *emu) { int i, size; @@ -500,7 +500,7 @@ snd_emu8000_init_fm(struct snd_emu8000 *emu) /* * The main initialization routine. */ -static void __init +static void __devinit snd_emu8000_init_hw(struct snd_emu8000 *emu) { int i; @@ -1019,7 +1019,7 @@ static struct snd_kcontrol_new *mixer_defs[EMU8000_NUM_CONTROLS] = { /* * create and attach mixer elements for WaveTable treble/bass controls */ -static int __init +static int __devinit snd_emu8000_create_mixer(struct snd_card *card, struct snd_emu8000 *emu) { int i, err = 0; @@ -1069,7 +1069,7 @@ static int snd_emu8000_dev_free(struct snd_device *device) /* * initialize and register emu8000 synth device. */ -int __init +int __devinit snd_emu8000_new(struct snd_card *card, int index, long port, int seq_ports, struct snd_seq_device **awe_ret) { -- cgit v1.2.2 From 58398895663f855aa32b440b164c426cfae4450c Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 10 Jun 2006 09:16:49 +0100 Subject: [ALSA] snd-ca0106: Update playback to 24bit. Fix typo is comment. Signed-off-by: James Courtier-Dutton --- sound/pci/ca0106/ca0106_main.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index b605d7045c..59bf9bd025 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -186,8 +186,8 @@ static struct snd_ca0106_details ca0106_chip_details[] = { /* New Audigy SE. Has a different DAC. */ /* SB0570: * CTRL:CA0106-DAT - * ADC: WM8768GEDS - * DAC: WM8775EDS + * ADC: WM8775EDS + * DAC: WM8768GEDS */ { .serial = 0x100a1102, .name = "Audigy SE [SB0570]", @@ -1189,7 +1189,7 @@ static unsigned int spi_dac_init[] = { 0x02ff, 0x0400, 0x0520, - 0x0600, + 0x0620, /* Set 24 bit. Was 0x0600 */ 0x08ff, 0x0aff, 0x0cff, -- cgit v1.2.2 From ecb594e66e740dc390a301768d89662777f1fde2 Mon Sep 17 00:00:00 2001 From: Remy Bruno Date: Mon, 12 Jun 2006 09:25:22 +0200 Subject: [ALSA] RME HDSP - fixed proc interface (missing {}) From: Remy Bruno Signed-off-by: Jaroslav Kysela --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index bf89ab9e31..eaf3c22449 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3169,9 +3169,10 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) char *clock_source; int x; - if (hdsp_check_for_iobox (hdsp)) + if (hdsp_check_for_iobox (hdsp)) { snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n"); return; + } if (hdsp_check_for_firmware(hdsp, 0)) { if (hdsp->state & HDSP_FirmwareCached) { -- cgit v1.2.2 From 70c5acbdcc7bb1651bb166f9e4b2345759a9fb18 Mon Sep 17 00:00:00 2001 From: Jaya Kumar Date: Mon, 12 Jun 2006 10:08:02 +0200 Subject: [ALSA] ac97_codec - fix duplicate control creation in AC97 This patch conditions AC97 control creation by whether or not the codec is an AD18xx codec. This fixes the case where the default control would get created and then snd_ac97_mixer_build fails out when creation of ad18xx specific control would get attempted. This problem was found and debuged by Marcelo Tosatti. Signed-off-by: Jaya Kumar Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 72e33b9532..f8389c2d3e 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1236,7 +1236,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080; /* build center controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) { + if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) + && !(ac97->flags & AC97_AD_MULTI)) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_center[0], ac97))) < 0) return err; if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_center[1], ac97))) < 0) @@ -1248,7 +1249,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build LFE controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1)) { + if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1)) + && !(ac97->flags & AC97_AD_MULTI)) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_lfe[0], ac97))) < 0) return err; if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_lfe[1], ac97))) < 0) @@ -1260,7 +1262,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build surround controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) { + if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) + && !(ac97->flags & AC97_AD_MULTI)) { /* Surround Master (0x38) is with stereo mutes */ if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", AC97_SURROUND_MASTER, 1, ac97)) < 0) return err; -- cgit v1.2.2 From 6540dffa6ecfe0d99fb263548dcc4b35ccefe784 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Jun 2006 11:57:22 +0200 Subject: [ALSA] hda-codec - Add SPDIF support to Thinkpad T/X/Z60 Added IEC958 (SPDIF) output support to Thinkpad T/X/Z60 with AD1981HD codec. The spdif jack is on docking station. Also, renamed 'IEC958 Playback Route' to 'IEC958 Playback Source' to avoid the mixer name confliction with IEC958 switch. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e83c7b01cb..e13e36aefb 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -963,7 +963,7 @@ static struct snd_kcontrol_new ad1983_mixers[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", .info = ad1983_spdif_route_info, .get = ad1983_spdif_route_get, .put = ad1983_spdif_route_put, @@ -1103,7 +1103,7 @@ static struct snd_kcontrol_new ad1981_mixers[] = { /* identical with AD1983 */ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", .info = ad1983_spdif_route_info, .get = ad1983_spdif_route_get, .put = ad1983_spdif_route_put, @@ -1349,6 +1349,14 @@ static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { .get = ad198x_mux_enum_get, .put = ad198x_mux_enum_put, }, + /* identical with AD1983 */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, { } /* end */ }; @@ -1422,7 +1430,6 @@ static int patch_ad1981(struct hda_codec *codec) break; case AD1981_THINKPAD: spec->mixers[0] = ad1981_thinkpad_mixers; - spec->multiout.dig_out_nid = 0; spec->input_mux = &ad1981_thinkpad_capture_source; break; } -- cgit v1.2.2 From 63eb1e4bd2975f1d1102c1f44e4fd6fcd76f7792 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Tue, 13 Jun 2006 11:58:12 +0200 Subject: [ALSA] fix potential NULL pointer deref in snd_sb8dsp_midi_interrupt() First testing if a pointer is NULL and if it is (or might be), proceeding with code that dereferences that same pointer is clearly a mistake. This happens in sound/isa/sb/sb8_midi.c::snd_sb8dsp_midi_interrupt() The patch below reworks the code so this unfortunate case doesn't happen. Also remove some blank comments. Found by the Coverity checker as bug #367 Patch is compile testted only due to lack of hardware. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/isa/sb/sb8_midi.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index c549aceea2..0b67edd7ac 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -32,20 +32,22 @@ #include #include -/* - - */ -irqreturn_t snd_sb8dsp_midi_interrupt(struct snd_sb * chip) +irqreturn_t snd_sb8dsp_midi_interrupt(struct snd_sb *chip) { struct snd_rawmidi *rmidi; int max = 64; char byte; - if (chip == NULL || (rmidi = chip->rmidi) == NULL) { + if (!chip) + return IRQ_NONE; + + rmidi = chip->rmidi; + if (!rmidi) { inb(SBP(chip, DATA_AVAIL)); /* ack interrupt */ return IRQ_NONE; } + spin_lock(&chip->midi_input_lock); while (max-- > 0) { if (inb(SBP(chip, DATA_AVAIL)) & 0x80) { @@ -59,10 +61,6 @@ irqreturn_t snd_sb8dsp_midi_interrupt(struct snd_sb * chip) return IRQ_HANDLED; } -/* - - */ - static int snd_sb8dsp_midi_input_open(struct snd_rawmidi_substream *substream) { unsigned long flags; @@ -252,10 +250,6 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre snd_sb8dsp_midi_output_write(substream); } -/* - - */ - static struct snd_rawmidi_ops snd_sb8dsp_midi_output = { .open = snd_sb8dsp_midi_output_open, -- cgit v1.2.2 From 40a4f7a014339712a9f81b5fad99558611e99ca1 Mon Sep 17 00:00:00 2001 From: Jaya Kumar Date: Tue, 13 Jun 2006 12:01:14 +0200 Subject: [ALSA] cs5535audio - trivial debug printk Following is a trivial patch to get more info for boards where the AC97_VENDOR_ID2 register (or others) time out. Signed-off-by: Jaya Kumar Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index f61c4fa4ed..91c18a11fe 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -110,7 +110,8 @@ static unsigned short snd_cs5535audio_codec_read(struct cs5535audio *cs5535au, udelay(1); } while (--timeout); if (!timeout) - snd_printk(KERN_ERR "Failure reading cs5535 codec\n"); + snd_printk(KERN_ERR "Failure reading codec reg 0x%x," + "Last value=0x%x\n", reg, val); return (unsigned short) val; } -- cgit v1.2.2 From 1781a9af1d95256ed45abac4b0b87f48f64b9b87 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 16 Jun 2006 12:13:00 +0200 Subject: [ALSA] sound/pci/: Add hp_only quirk for Dell D800 laptops http://www.kernel.org/git/?p=linux/kernel/git/bcollins/ubuntu-dapper.git;a=commitdiff;h=9ad787cd9670c3f3b8f3db235e84baf00a2ea526 Anders Ostling comments in Malone #41015 that his Dell D800 laptop's volume control works correctly when the hp_only quirk is passed to modprobe. This commit adds his hardware's sub{vendor,device} ids to the quirk list for the intel8x0 driver. Signed-off-by: Daniel T Chen Signed-off-by: Ben Collins Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index e09fb7f9e7..edc14475ef 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1805,6 +1805,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "Dell Optiplex GX270", /* AD1981B */ .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x1028, + .subdevice = 0x014e, + .name = "Dell D800", /* STAC9750/51 */ + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x1028, .subdevice = 0x0163, -- cgit v1.2.2 From d4199f01750f3fa6a5b8bacdac9bd0051fee95ef Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 16 Jun 2006 16:21:54 +0200 Subject: [ALSA] Add hp_only quirk for pci id [161f:2032] to via82xx http://www.kernel.org/git/?p=linux/kernel/git/bcollins/ubuntu-dapper.git;a=commitdiff;h=eae2cc78de39502595f67b7fc1f821f5963bb8ae UpstreamStatus: Not merged Christian Bjalevik reports in LP#38546 that his sound chipset requires the 'hp_only' quirk to allow him to control sound volume correctly when headphones are inserted. This patch adds the appropriate pci id to the via82xx ALSA driver so that the quirk is applied automatically, thereby removing the need for users to modify /etc/modprobe.d/alsa-base (or to unload and reload snd-via82xx with ac97_quirk=hp_only). This patch closes LP#38546. Signed-off-by: Daniel T Chen Signed-off-by: Ben Collins Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 12ce22ef16..cbad9b22f0 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1775,6 +1775,12 @@ static struct ac97_quirk ac97_quirks[] = { .name = "Targa Traveller 811", .type = AC97_TUNE_HP_ONLY, }, + { + .subvendor = 0x161f, + .subdevice = 0x2032, + .name = "m680x", + .type = AC97_TUNE_HP_ONLY, /* http://launchpad.net/bugs/38546 */ + }, { } /* terminator */ }; -- cgit v1.2.2 From 396f739e21f3b7ea9ece08bf0abf0a45693c3047 Mon Sep 17 00:00:00 2001 From: Karsten Wiese Date: Mon, 19 Jun 2006 13:22:52 +0200 Subject: [ALSA] via82xx - Default to variable samplerate enabled for MSI K8T Neo2-FI Default to variable samplerate enabled for MSI K8T Neo2-FI No crackles here with 44100. Signed-off-by: Karsten Wiese Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index cbad9b22f0..2527bbd958 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2371,7 +2371,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci, int revision) { .subvendor = 0x1462, .subdevice = 0x0470, .action = VIA_DXS_SRC }, /* MSI KT880 Delta-FSR */ { .subvendor = 0x1462, .subdevice = 0x3800, .action = VIA_DXS_ENABLE }, /* MSI KT266 */ { .subvendor = 0x1462, .subdevice = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */ - { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */ + { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_SRC }, /* MSI K8T Neo2-FI */ { .subvendor = 0x1462, .subdevice = 0x7120, .action = VIA_DXS_ENABLE }, /* MSI KT4V */ { .subvendor = 0x1462, .subdevice = 0x7142, .action = VIA_DXS_ENABLE }, /* MSI K8MM-V */ { .subvendor = 0x1462, .subdevice = 0xb012, .action = VIA_DXS_SRC }, /* P4M800/VIA8237R */ -- cgit v1.2.2 From 1459c7849ea24fd71e4d2e678caa1cc3fef754e2 Mon Sep 17 00:00:00 2001 From: Rodolfo Giometti Date: Mon, 19 Jun 2006 15:04:54 +0200 Subject: [ALSA] Disable AC97 AUX and VIDEO controls for WM9705 touchscreen This patch by Rodolfo Giometti disables the AC97 AUX and VIDEO controls on the WM9705 when the touchscreen is selected as the AUX and VIDEO lines are shared with the touch controller. Changes:- o Added AC97_HAS_NO_AUX flag o Test for AC97_HAS_NO_AUX flag in snd_ac97_mixer_build() o Sets AC97_HAS_NO_VIDEO and AC97_HAS_NO_AUX in patch_wolfson05() when WM9705 touch driver is selected. Signed-off-by: Rodolfo Giometti Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 8 +++++--- sound/pci/ac97/ac97_patch.c | 4 ++++ 2 files changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index f8389c2d3e..0abf2808d5 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1348,9 +1348,11 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build Aux controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) { - if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0) - return err; + if (!(ac97->flags & AC97_HAS_NO_AUX)) { + if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) { + if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0) + return err; + } } /* build PCM controls */ diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1c47ee3d7e..cc93ee6190 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -464,6 +464,10 @@ int patch_wolfson05(struct snd_ac97 * ac97) { /* WM9705, WM9710 */ ac97->build_ops = &patch_wolfson_wm9705_ops; +#ifdef CONFIG_TOUCHSCREEN_WM9705 + /* WM9705 touchscreen uses AUX and VIDEO for touch */ + ac97->flags |=3D AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; +#endif return 0; } -- cgit v1.2.2 From 1561f09a2f91bc258a72225f919807c9e51c8290 Mon Sep 17 00:00:00 2001 From: Jaya Kumar Date: Mon, 19 Jun 2006 15:06:14 +0200 Subject: [ALSA] AD1888 suspend/resume fix This patch adds a write to an undocumented register, 0x60 Extended Codec Register Page in the AD1888 codec. It is neccessary in order to make suspend/resume work with the AD1888. Signed-off-by: Jaya Kumar Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index cc93ee6190..7f197c7808 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1371,6 +1371,13 @@ static void ad18xx_resume(struct snd_ac97 *ac97) snd_ac97_restore_iec958(ac97); } + +static void ad1888_resume(struct snd_ac97 *ac97) +{ + ad18xx_resume(ac97); + snd_ac97_write_cache(ac97, AC97_CODEC_CLASS_REV, 0x8080); +} + #endif int patch_ad1819(struct snd_ac97 * ac97) @@ -1844,7 +1851,7 @@ static struct snd_ac97_build_ops patch_ad1888_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1888_specific, #ifdef CONFIG_PM - .resume = ad18xx_resume, + .resume = ad1888_resume, #endif .update_jacks = ad1888_update_jacks, }; -- cgit v1.2.2 From c4a87ef4de9860d00460dce30776f7cc17e77459 Mon Sep 17 00:00:00 2001 From: Ben Williamson Date: Mon, 19 Jun 2006 17:20:09 +0200 Subject: [ALSA] USB midi: Remove duplicate CS_AUDIO_* #defines Removed the CS_AUDIO_* #defines, which were duplicates of the class-specific USB_DT_CS_* #defines in . Signed-off-by: Ben Williamson Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 2 +- sound/usb/usbaudio.h | 7 ------- sound/usb/usbmidi.c | 2 +- 3 files changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 30cadec9a3..627de9525a 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2981,7 +2981,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); - if (alts->extralen != 11 || alts->extra[1] != CS_AUDIO_INTERFACE || + if (alts->extralen != 11 || alts->extra[1] != USB_DT_CS_INTERFACE || altsd->bNumEndpoints != 1) return -ENXIO; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 88733524d0..0f4b2b8541 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,13 +30,6 @@ #define USB_SUBCLASS_MIDI_STREAMING 0x03 #define USB_SUBCLASS_VENDOR_SPEC 0xff -#define CS_AUDIO_UNDEFINED 0x20 -#define CS_AUDIO_DEVICE 0x21 -#define CS_AUDIO_CONFIGURATION 0x22 -#define CS_AUDIO_STRING 0x23 -#define CS_AUDIO_INTERFACE 0x24 -#define CS_AUDIO_ENDPOINT 0x25 - #define HEADER 0x01 #define INPUT_TERMINAL 0x02 #define OUTPUT_TERMINAL 0x03 diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 5c53ec8a13..5105b6b057 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1413,7 +1413,7 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, for (cs_desc = hostif->extra; cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; cs_desc += cs_desc[0]) { - if (cs_desc[1] == CS_AUDIO_INTERFACE) { + if (cs_desc[1] == USB_DT_CS_INTERFACE) { if (cs_desc[2] == MIDI_IN_JACK) endpoint->in_cables = (endpoint->in_cables << 1) | 1; else if (cs_desc[2] == MIDI_OUT_JACK) -- cgit v1.2.2 From 6dac9a65f05600bc29316e3cf3365236efe69041 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 21 Jun 2006 08:51:07 +0200 Subject: [ALSA] HDA - Lenovo 3000 N100-07684JU - enable laptop-eapd by default Justin Sunseri reports that sound is audible on his Lenovo 3000 N100-07684JU by passing 'model=laptop-eapd' to modprobe, so this patch adds the pci ids for his sound device to patch_analog.c . This commit closes LP#39517. Alexey Parshin also confirmed the fix at http://bugs.gentoo.org/137245 TODO: Mute onboard speakers when device is plugged into the headphone jack. Muting the 'External Amplifier' mixer element while a device is plugged into the headphone jack allows sound to be played only from the headphone jack. From: Daniel T Chen Signed-off-by: Daniel T Chen Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e13e36aefb..9a6015ad61 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -809,6 +809,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_LAPTOP_EAPD }, /* ASUS Z62F */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x30af, .config = AD1986A_LAPTOP_EAPD }, /* HP Compaq Presario B2800 */ + { .pci_subvendor = 0x17aa, .pci_subdevice = 0x2066, + .config = AD1986A_LAPTOP_EAPD }, /* Lenovo 3000 N100-07684JU */ {} }; -- cgit v1.2.2 From 41f0cd3a0c4c6547860cf3b1c2d7968008e6c071 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jun 2006 12:14:40 +0200 Subject: [ALSA] hda-codec - Use 3stack model for ASUS P5RD2-VM / P5GPL-X SE Use 3stack model as default for ASUS P5RD2-VM and P5GPL-X SE boards with AD1986A codec (ALSA bug#2103). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9a6015ad61..dd4e00a82b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -789,6 +789,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { { .modelname = "3stack", .config = AD1986A_3STACK }, { .pci_subvendor = 0x10de, .pci_subdevice = 0xcb84, .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3, + .config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */ { .modelname = "laptop", .config = AD1986A_LAPTOP }, { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, .config = AD1986A_LAPTOP }, /* FSC V2060 */ -- cgit v1.2.2 From f3d9478b2ce468c3115b02ecae7e975990697f15 Mon Sep 17 00:00:00 2001 From: Johannes Berg Date: Wed, 21 Jun 2006 15:42:43 +0200 Subject: [ALSA] snd-aoa: add snd-aoa This large patch adds all of snd-aoa. Consisting of many modules, it currently replaces snd-powermac for all layout-id based machines and handles many more (for example new powerbooks and powermacs with digital output that previously couldn't be used at all). It also has support for all layout-IDs that Apple has (judging from their Info.plist file) but not all are tested. The driver currently has 2 known regressions over snd-powermac: * it doesn't handle powermac 7,2 and 7,3 * it doesn't have a DRC control on snapper-based machines I will fix those during the 2.6.18 development cycle. Signed-off-by: Johannes Berg Signed-off-by: Takashi Iwai --- sound/Kconfig | 2 + sound/Makefile | 2 +- sound/aoa/Kconfig | 17 + sound/aoa/Makefile | 4 + sound/aoa/aoa-gpio.h | 81 ++ sound/aoa/aoa.h | 131 +++ sound/aoa/codecs/Kconfig | 32 + sound/aoa/codecs/Makefile | 3 + sound/aoa/codecs/snd-aoa-codec-onyx.c | 1113 +++++++++++++++++++++++ sound/aoa/codecs/snd-aoa-codec-onyx.h | 76 ++ sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h | 209 +++++ sound/aoa/codecs/snd-aoa-codec-tas.c | 654 +++++++++++++ sound/aoa/codecs/snd-aoa-codec-tas.h | 47 + sound/aoa/codecs/snd-aoa-codec-toonie.c | 141 +++ sound/aoa/core/Makefile | 5 + sound/aoa/core/snd-aoa-alsa.c | 98 ++ sound/aoa/core/snd-aoa-alsa.h | 16 + sound/aoa/core/snd-aoa-core.c | 162 ++++ sound/aoa/core/snd-aoa-gpio-feature.c | 399 ++++++++ sound/aoa/core/snd-aoa-gpio-pmf.c | 246 +++++ sound/aoa/fabrics/Kconfig | 12 + sound/aoa/fabrics/Makefile | 1 + sound/aoa/fabrics/snd-aoa-fabric-layout.c | 1109 ++++++++++++++++++++++ sound/aoa/soundbus/Kconfig | 14 + sound/aoa/soundbus/Makefile | 3 + sound/aoa/soundbus/core.c | 250 +++++ sound/aoa/soundbus/i2sbus/Makefile | 2 + sound/aoa/soundbus/i2sbus/i2sbus-control.c | 192 ++++ sound/aoa/soundbus/i2sbus/i2sbus-control.h | 37 + sound/aoa/soundbus/i2sbus/i2sbus-core.c | 387 ++++++++ sound/aoa/soundbus/i2sbus/i2sbus-interface.h | 187 ++++ sound/aoa/soundbus/i2sbus/i2sbus-pcm.c | 1021 +++++++++++++++++++++ sound/aoa/soundbus/i2sbus/i2sbus.h | 112 +++ sound/aoa/soundbus/soundbus.h | 202 ++++ sound/aoa/soundbus/sysfs.c | 43 + 35 files changed, 7009 insertions(+), 1 deletion(-) create mode 100644 sound/aoa/Kconfig create mode 100644 sound/aoa/Makefile create mode 100644 sound/aoa/aoa-gpio.h create mode 100644 sound/aoa/aoa.h create mode 100644 sound/aoa/codecs/Kconfig create mode 100644 sound/aoa/codecs/Makefile create mode 100644 sound/aoa/codecs/snd-aoa-codec-onyx.c create mode 100644 sound/aoa/codecs/snd-aoa-codec-onyx.h create mode 100644 sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h create mode 100644 sound/aoa/codecs/snd-aoa-codec-tas.c create mode 100644 sound/aoa/codecs/snd-aoa-codec-tas.h create mode 100644 sound/aoa/codecs/snd-aoa-codec-toonie.c create mode 100644 sound/aoa/core/Makefile create mode 100644 sound/aoa/core/snd-aoa-alsa.c create mode 100644 sound/aoa/core/snd-aoa-alsa.h create mode 100644 sound/aoa/core/snd-aoa-core.c create mode 100644 sound/aoa/core/snd-aoa-gpio-feature.c create mode 100644 sound/aoa/core/snd-aoa-gpio-pmf.c create mode 100644 sound/aoa/fabrics/Kconfig create mode 100644 sound/aoa/fabrics/Makefile create mode 100644 sound/aoa/fabrics/snd-aoa-fabric-layout.c create mode 100644 sound/aoa/soundbus/Kconfig create mode 100644 sound/aoa/soundbus/Makefile create mode 100644 sound/aoa/soundbus/core.c create mode 100644 sound/aoa/soundbus/i2sbus/Makefile create mode 100644 sound/aoa/soundbus/i2sbus/i2sbus-control.c create mode 100644 sound/aoa/soundbus/i2sbus/i2sbus-control.h create mode 100644 sound/aoa/soundbus/i2sbus/i2sbus-core.c create mode 100644 sound/aoa/soundbus/i2sbus/i2sbus-interface.h create mode 100644 sound/aoa/soundbus/i2sbus/i2sbus-pcm.c create mode 100644 sound/aoa/soundbus/i2sbus/i2sbus.h create mode 100644 sound/aoa/soundbus/soundbus.h create mode 100644 sound/aoa/soundbus/sysfs.c (limited to 'sound') diff --git a/sound/Kconfig b/sound/Kconfig index b65ee4701f..e0d791a984 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -58,6 +58,8 @@ source "sound/pci/Kconfig" source "sound/ppc/Kconfig" +source "sound/aoa/Kconfig" + source "sound/arm/Kconfig" source "sound/mips/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index f352bb2359..a682ea30f0 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -4,7 +4,7 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ aoa/ ifeq ($(CONFIG_SND),y) obj-y += last.o diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig new file mode 100644 index 0000000000..b11ccf6dba --- /dev/null +++ b/sound/aoa/Kconfig @@ -0,0 +1,17 @@ +menu "Apple Onboard Audio driver" + depends on SND != n && PPC + +config SND_AOA + tristate "Apple Onboard Audio driver" + depends on SOUND && SND_PCM + ---help--- + This option enables the new driver for the various + Apple Onboard Audio components. + +source "sound/aoa/fabrics/Kconfig" + +source "sound/aoa/codecs/Kconfig" + +source "sound/aoa/soundbus/Kconfig" + +endmenu diff --git a/sound/aoa/Makefile b/sound/aoa/Makefile new file mode 100644 index 0000000000..d8de3e7df4 --- /dev/null +++ b/sound/aoa/Makefile @@ -0,0 +1,4 @@ +obj-$(CONFIG_SND_AOA) += core/ +obj-$(CONFIG_SND_AOA) += codecs/ +obj-$(CONFIG_SND_AOA) += fabrics/ +obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/ diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h new file mode 100644 index 0000000000..3a61f31155 --- /dev/null +++ b/sound/aoa/aoa-gpio.h @@ -0,0 +1,81 @@ +/* + * Apple Onboard Audio GPIO definitions + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#ifndef __AOA_GPIO_H +#define __AOA_GPIO_H +#include +#include +#include + +typedef void (*notify_func_t)(void *data); + +enum notify_type { + AOA_NOTIFY_HEADPHONE, + AOA_NOTIFY_LINE_IN, + AOA_NOTIFY_LINE_OUT, +}; + +struct gpio_runtime; +struct gpio_methods { + /* for initialisation/de-initialisation of the GPIO layer */ + void (*init)(struct gpio_runtime *rt); + void (*exit)(struct gpio_runtime *rt); + + /* turn off headphone, speakers, lineout */ + void (*all_amps_off)(struct gpio_runtime *rt); + /* turn headphone, speakers, lineout back to previous setting */ + void (*all_amps_restore)(struct gpio_runtime *rt); + + void (*set_headphone)(struct gpio_runtime *rt, int on); + void (*set_speakers)(struct gpio_runtime *rt, int on); + void (*set_lineout)(struct gpio_runtime *rt, int on); + + int (*get_headphone)(struct gpio_runtime *rt); + int (*get_speakers)(struct gpio_runtime *rt); + int (*get_lineout)(struct gpio_runtime *rt); + + void (*set_hw_reset)(struct gpio_runtime *rt, int on); + + /* use this to be notified of any events. The notification + * function is passed the data, and is called in process + * context by the use of schedule_work. + * The interface for it is that setting a function to NULL + * removes it, and they return 0 if the operation succeeded, + * and -EBUSY if the notification is already assigned by + * someone else. */ + int (*set_notify)(struct gpio_runtime *rt, + enum notify_type type, + notify_func_t notify, + void *data); + /* returns 0 if not plugged in, 1 if plugged in + * or a negative error code */ + int (*get_detect)(struct gpio_runtime *rt, + enum notify_type type); +}; + +struct gpio_notification { + notify_func_t notify; + void *data; + void *gpio_private; + struct work_struct work; + struct mutex mutex; +}; + +struct gpio_runtime { + /* to be assigned by fabric */ + struct device_node *node; + /* since everyone needs this pointer anyway... */ + struct gpio_methods *methods; + /* to be used by the gpio implementation */ + int implementation_private; + struct gpio_notification headphone_notify; + struct gpio_notification line_in_notify; + struct gpio_notification line_out_notify; +}; + +#endif /* __AOA_GPIO_H */ diff --git a/sound/aoa/aoa.h b/sound/aoa/aoa.h new file mode 100644 index 0000000000..378ef1e987 --- /dev/null +++ b/sound/aoa/aoa.h @@ -0,0 +1,131 @@ +/* + * Apple Onboard Audio definitions + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#ifndef __AOA_H +#define __AOA_H +#include +#include +/* So apparently there's a reason for requiring driver.h to be included first! */ +#include +#include +#include +#include +#include "aoa-gpio.h" +#include "soundbus/soundbus.h" + +#define MAX_CODEC_NAME_LEN 32 + +struct aoa_codec { + char name[MAX_CODEC_NAME_LEN]; + + struct module *owner; + + /* called when the fabric wants to init this codec. + * Do alsa card manipulations from here. */ + int (*init)(struct aoa_codec *codec); + + /* called when the fabric is done with the codec. + * The alsa card will be cleaned up so don't bother. */ + void (*exit)(struct aoa_codec *codec); + + /* May be NULL, but can be used by the fabric. + * Refcounting is the codec driver's responsibility */ + struct device_node *node; + + /* assigned by fabric before init() is called, points + * to the soundbus device. Cannot be NULL. */ + struct soundbus_dev *soundbus_dev; + + /* assigned by the fabric before init() is called, points + * to the fabric's gpio runtime record for the relevant + * device. */ + struct gpio_runtime *gpio; + + /* assigned by the fabric before init() is called, contains + * a codec specific bitmask of what outputs and inputs are + * actually connected */ + u32 connected; + + /* data the fabric can associate with this structure */ + void *fabric_data; + + /* private! */ + struct list_head list; + struct aoa_fabric *fabric; +}; + +/* return 0 on success */ +extern int +aoa_codec_register(struct aoa_codec *codec); +extern void +aoa_codec_unregister(struct aoa_codec *codec); + +#define MAX_LAYOUT_NAME_LEN 32 + +struct aoa_fabric { + char name[MAX_LAYOUT_NAME_LEN]; + + struct module *owner; + + /* once codecs register, they are passed here after. + * They are of course not initialised, since the + * fabric is responsible for initialising some fields + * in the codec structure! */ + int (*found_codec)(struct aoa_codec *codec); + /* called for each codec when it is removed, + * also in the case that aoa_fabric_unregister + * is called and all codecs are removed + * from this fabric. + * Also called if found_codec returned 0 but + * the codec couldn't initialise. */ + void (*remove_codec)(struct aoa_codec *codec); + /* If found_codec returned 0, and the codec + * could be initialised, this is called. */ + void (*attached_codec)(struct aoa_codec *codec); +}; + +/* return 0 on success, -EEXIST if another fabric is + * registered, -EALREADY if the same fabric is registered. + * Passing NULL can be used to test for the presence + * of another fabric, if -EALREADY is returned there is + * no other fabric present. + * In the case that the function returns -EALREADY + * and the fabric passed is not NULL, all codecs + * that are not assigned yet are passed to the fabric + * again for reconsideration. */ +extern int +aoa_fabric_register(struct aoa_fabric *fabric); + +/* it is vital to call this when the fabric exits! + * When calling, the remove_codec will be called + * for all codecs, unless it is NULL. */ +extern void +aoa_fabric_unregister(struct aoa_fabric *fabric); + +/* if for some reason you want to get rid of a codec + * before the fabric is removed, use this. + * Note that remove_codec is called for it! */ +extern void +aoa_fabric_unlink_codec(struct aoa_codec *codec); + +/* alsa help methods */ +struct aoa_card { + struct snd_card *alsa_card; +}; + +extern int aoa_snd_device_new(snd_device_type_t type, + void * device_data, struct snd_device_ops * ops); +extern struct snd_card *aoa_get_card(void); +extern int aoa_snd_ctl_add(struct snd_kcontrol* control); + +/* GPIO stuff */ +extern struct gpio_methods *pmf_gpio_methods; +extern struct gpio_methods *ftr_gpio_methods; +/* extern struct gpio_methods *map_gpio_methods; */ + +#endif /* __AOA_H */ diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig new file mode 100644 index 0000000000..90cf58f686 --- /dev/null +++ b/sound/aoa/codecs/Kconfig @@ -0,0 +1,32 @@ +config SND_AOA_ONYX + tristate "support Onyx chip" + depends on SND_AOA + ---help--- + This option enables support for the Onyx (pcm3052) + codec chip found in the latest Apple machines + (most of those with digital audio output). + +#config SND_AOA_TOPAZ +# tristate "support Topaz chips" +# depends on SND_AOA +# ---help--- +# This option enables support for the Topaz (CS84xx) +# codec chips found in the latest Apple machines, +# these chips do the digital input and output on +# some PowerMacs. + +config SND_AOA_TAS + tristate "support TAS chips" + depends on SND_AOA + ---help--- + This option enables support for the tas chips + found in a lot of Apple Machines, especially + iBooks and PowerBooks without digital. + +config SND_AOA_TOONIE + tristate "support Toonie chip" + depends on SND_AOA + ---help--- + This option enables support for the toonie codec + found in the Mac Mini. If you have a Mac Mini and + want to hear sound, select this option. diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile new file mode 100644 index 0000000000..31cbe68fd4 --- /dev/null +++ b/sound/aoa/codecs/Makefile @@ -0,0 +1,3 @@ +obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o +obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o +obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c new file mode 100644 index 0000000000..0b7650788f --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -0,0 +1,1113 @@ +/* + * Apple Onboard Audio driver for Onyx codec + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + * + * + * This is a driver for the pcm3052 codec chip (codenamed Onyx) + * that is present in newer Apple hardware (with digital output). + * + * The Onyx codec has the following connections (listed by the bit + * to be used in aoa_codec.connected): + * 0: analog output + * 1: digital output + * 2: line input + * 3: microphone input + * Note that even though I know of no machine that has for example + * the digital output connected but not the analog, I have handled + * all the different cases in the code so that this driver may serve + * as a good example of what to do. + * + * NOTE: This driver assumes that there's at most one chip to be + * used with one alsa card, in form of creating all kinds + * of mixer elements without regard for their existence. + * But snd-aoa assumes that there's at most one card, so + * this means you can only have one onyx on a system. This + * should probably be fixed by changing the assumption of + * having just a single card on a system, and making the + * 'card' pointer accessible to anyone who needs it instead + * of hiding it in the aoa_snd_* functions... + * + */ +#include +#include +MODULE_AUTHOR("Johannes Berg "); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); + +#include "snd-aoa-codec-onyx.h" +#include "../aoa.h" +#include "../soundbus/soundbus.h" + + +#define PFX "snd-aoa-codec-onyx: " + +struct onyx { + /* cache registers 65 to 80, they are write-only! */ + u8 cache[16]; + struct i2c_client i2c; + struct aoa_codec codec; + u32 initialised:1, + spdif_locked:1, + analog_locked:1, + original_mute:2; + int open_count; + struct codec_info *codec_info; + + /* mutex serializes concurrent access to the device + * and this structure. + */ + struct mutex mutex; +}; +#define codec_to_onyx(c) container_of(c, struct onyx, codec) + +/* both return 0 if all ok, else on error */ +static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) +{ + s32 v; + + if (reg != ONYX_REG_CONTROL) { + *value = onyx->cache[reg-FIRSTREGISTER]; + return 0; + } + v = i2c_smbus_read_byte_data(&onyx->i2c, reg); + if (v < 0) + return -1; + *value = (u8)v; + onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; + return 0; +} + +static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value) +{ + int result; + + result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value); + if (!result) + onyx->cache[reg-FIRSTREGISTER] = value; + return result; +} + +/* alsa stuff */ + +static int onyx_dev_register(struct snd_device *dev) +{ + return 0; +} + +static struct snd_device_ops ops = { + .dev_register = onyx_dev_register, +}; + +/* this is necessary because most alsa mixer programs + * can't properly handle the negative range */ +#define VOLUME_RANGE_SHIFT 128 + +static int onyx_snd_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = -128 + VOLUME_RANGE_SHIFT; + uinfo->value.integer.max = -1 + VOLUME_RANGE_SHIFT; + return 0; +} + +static int onyx_snd_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + s8 l, r; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = l + VOLUME_RANGE_SHIFT; + ucontrol->value.integer.value[1] = r + VOLUME_RANGE_SHIFT; + + return 0; +} + +static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + s8 l, r; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r); + + if (l + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[0] && + r + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[1]) { + mutex_unlock(&onyx->mutex); + return 0; + } + + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, + ucontrol->value.integer.value[0] + - VOLUME_RANGE_SHIFT); + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, + ucontrol->value.integer.value[1] + - VOLUME_RANGE_SHIFT); + mutex_unlock(&onyx->mutex); + + return 1; +} + +static struct snd_kcontrol_new volume_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = onyx_snd_vol_info, + .get = onyx_snd_vol_get, + .put = onyx_snd_vol_put, +}; + +/* like above, this is necessary because a lot + * of alsa mixer programs don't handle ranges + * that don't start at 0 properly. + * even alsamixer is one of them... */ +#define INPUTGAIN_RANGE_SHIFT (-3) + +static int onyx_snd_inputgain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 3 + INPUTGAIN_RANGE_SHIFT; + uinfo->value.integer.max = 28 + INPUTGAIN_RANGE_SHIFT; + return 0; +} + +static int onyx_snd_inputgain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 ig; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &ig); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = + (ig & ONYX_ADC_PGA_GAIN_MASK) + INPUTGAIN_RANGE_SHIFT; + + return 0; +} + +static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v, n; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v); + n = v; + n &= ~ONYX_ADC_PGA_GAIN_MASK; + n |= (ucontrol->value.integer.value[0] - INPUTGAIN_RANGE_SHIFT) + & ONYX_ADC_PGA_GAIN_MASK; + onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, n); + mutex_unlock(&onyx->mutex); + + return n != v; +} + +static struct snd_kcontrol_new inputgain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = onyx_snd_inputgain_info, + .get = onyx_snd_inputgain_get, + .put = onyx_snd_inputgain_put, +}; + +static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { "Line-In", "Microphone" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item > 1) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int onyx_snd_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + s8 v; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v); + mutex_unlock(&onyx->mutex); + + ucontrol->value.enumerated.item[0] = !!(v&ONYX_ADC_INPUT_MIC); + + return 0; +} + +static void onyx_set_capture_source(struct onyx *onyx, int mic) +{ + s8 v; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v); + v &= ~ONYX_ADC_INPUT_MIC; + if (mic) + v |= ONYX_ADC_INPUT_MIC; + onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, v); + mutex_unlock(&onyx->mutex); +} + +static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + onyx_set_capture_source(snd_kcontrol_chip(kcontrol), + ucontrol->value.enumerated.item[0]); + return 1; +} + +static struct snd_kcontrol_new capture_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* If we name this 'Input Source', it properly shows up in + * alsamixer as a selection, * but it's shown under the + * 'Playback' category. + * If I name it 'Capture Source', it shows up in strange + * ways (two bools of which one can be selected at a + * time) but at least it's shown in the 'Capture' + * category. + * I was told that this was due to backward compatibility, + * but I don't understand then why the mangling is *not* + * done when I name it "Input Source"..... + */ + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = onyx_snd_capture_source_info, + .get = onyx_snd_capture_source_get, + .put = onyx_snd_capture_source_put, +}; + +static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 c; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = !(c & ONYX_MUTE_LEFT); + ucontrol->value.integer.value[1] = !(c & ONYX_MUTE_RIGHT); + + return 0; +} + +static int onyx_snd_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v = 0, c = 0; + int err = -EBUSY; + + mutex_lock(&onyx->mutex); + if (onyx->analog_locked) + goto out_unlock; + + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); + c = v; + c &= ~(ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT); + if (!ucontrol->value.integer.value[0]) + c |= ONYX_MUTE_LEFT; + if (!ucontrol->value.integer.value[1]) + c |= ONYX_MUTE_RIGHT; + err = onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, c); + + out_unlock: + mutex_unlock(&onyx->mutex); + + return !err ? (v != c) : err; +} + +static struct snd_kcontrol_new mute_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = onyx_snd_mute_info, + .get = onyx_snd_mute_get, + .put = onyx_snd_mute_put, +}; + + +static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +#define FLAG_POLARITY_INVERT 1 +#define FLAG_SPDIFLOCK 2 + +static int onyx_snd_single_bit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 c; + long int pv = kcontrol->private_value; + u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT; + u8 address = (pv >> 8) & 0xff; + u8 mask = pv & 0xff; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, address, &c); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = !!(c & mask) ^ polarity; + + return 0; +} + +static int onyx_snd_single_bit_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v = 0, c = 0; + int err; + long int pv = kcontrol->private_value; + u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT; + u8 spdiflock = (pv >> 16) & FLAG_SPDIFLOCK; + u8 address = (pv >> 8) & 0xff; + u8 mask = pv & 0xff; + + mutex_lock(&onyx->mutex); + if (spdiflock && onyx->spdif_locked) { + /* even if alsamixer doesn't care.. */ + err = -EBUSY; + goto out_unlock; + } + onyx_read_register(onyx, address, &v); + c = v; + c &= ~(mask); + if (!!ucontrol->value.integer.value[0] ^ polarity) + c |= mask; + err = onyx_write_register(onyx, address, c); + + out_unlock: + mutex_unlock(&onyx->mutex); + + return !err ? (v != c) : err; +} + +#define SINGLE_BIT(n, type, description, address, mask, flags) \ +static struct snd_kcontrol_new n##_control = { \ + .iface = SNDRV_CTL_ELEM_IFACE_##type, \ + .name = description, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = onyx_snd_single_bit_info, \ + .get = onyx_snd_single_bit_get, \ + .put = onyx_snd_single_bit_put, \ + .private_value = (flags << 16) | (address << 8) | mask \ +} + +SINGLE_BIT(spdif, + MIXER, + SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + ONYX_REG_DIG_INFO4, + ONYX_SPDIF_ENABLE, + FLAG_SPDIFLOCK); +SINGLE_BIT(ovr1, + MIXER, + "Oversampling Rate", + ONYX_REG_DAC_CONTROL, + ONYX_OVR1, + 0); +SINGLE_BIT(flt0, + MIXER, + "Fast Digital Filter Rolloff", + ONYX_REG_DAC_FILTER, + ONYX_ROLLOFF_FAST, + FLAG_POLARITY_INVERT); +SINGLE_BIT(hpf, + MIXER, + "Highpass Filter", + ONYX_REG_ADC_HPF_BYPASS, + ONYX_HPF_DISABLE, + FLAG_POLARITY_INVERT); +SINGLE_BIT(dm12, + MIXER, + "Digital De-Emphasis", + ONYX_REG_DAC_DEEMPH, + ONYX_DIGDEEMPH_CTRL, + 0); + +static int onyx_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* datasheet page 30, all others are 0 */ + ucontrol->value.iec958.status[0] = 0x3e; + ucontrol->value.iec958.status[1] = 0xff; + + ucontrol->value.iec958.status[3] = 0x3f; + ucontrol->value.iec958.status[4] = 0x0f; + + return 0; +} + +static struct snd_kcontrol_new onyx_spdif_mask = { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .info = onyx_spdif_info, + .get = onyx_spdif_mask_get, +}; + +static int onyx_spdif_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v); + ucontrol->value.iec958.status[0] = v & 0x3e; + + onyx_read_register(onyx, ONYX_REG_DIG_INFO2, &v); + ucontrol->value.iec958.status[1] = v; + + onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v); + ucontrol->value.iec958.status[3] = v & 0x3f; + + onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); + ucontrol->value.iec958.status[4] = v & 0x0f; + mutex_unlock(&onyx->mutex); + + return 0; +} + +static int onyx_spdif_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v); + v = (v & ~0x3e) | (ucontrol->value.iec958.status[0] & 0x3e); + onyx_write_register(onyx, ONYX_REG_DIG_INFO1, v); + + v = ucontrol->value.iec958.status[1]; + onyx_write_register(onyx, ONYX_REG_DIG_INFO2, v); + + onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v); + v = (v & ~0x3f) | (ucontrol->value.iec958.status[3] & 0x3f); + onyx_write_register(onyx, ONYX_REG_DIG_INFO3, v); + + onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); + v = (v & ~0x0f) | (ucontrol->value.iec958.status[4] & 0x0f); + onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v); + mutex_unlock(&onyx->mutex); + + return 1; +} + +static struct snd_kcontrol_new onyx_spdif_ctrl = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .info = onyx_spdif_info, + .get = onyx_spdif_get, + .put = onyx_spdif_put, +}; + +/* our registers */ + +static u8 register_map[] = { + ONYX_REG_DAC_ATTEN_LEFT, + ONYX_REG_DAC_ATTEN_RIGHT, + ONYX_REG_CONTROL, + ONYX_REG_DAC_CONTROL, + ONYX_REG_DAC_DEEMPH, + ONYX_REG_DAC_FILTER, + ONYX_REG_DAC_OUTPHASE, + ONYX_REG_ADC_CONTROL, + ONYX_REG_ADC_HPF_BYPASS, + ONYX_REG_DIG_INFO1, + ONYX_REG_DIG_INFO2, + ONYX_REG_DIG_INFO3, + ONYX_REG_DIG_INFO4 +}; + +static u8 initial_values[ARRAY_SIZE(register_map)] = { + 0x80, 0x80, /* muted */ + ONYX_MRST | ONYX_SRST, /* but handled specially! */ + ONYX_MUTE_LEFT | ONYX_MUTE_RIGHT, + 0, /* no deemphasis */ + ONYX_DAC_FILTER_ALWAYS, + ONYX_OUTPHASE_INVERTED, + (-1 /*dB*/ + 8) & 0xF, /* line in selected, -1 dB gain*/ + ONYX_ADC_HPF_ALWAYS, + (1<<2), /* pcm audio */ + 2, /* category: pcm coder */ + 0, /* sampling frequency 44.1 kHz, clock accuracy level II */ + 1 /* 24 bit depth */ +}; + +/* reset registers of chip, either to initial or to previous values */ +static int onyx_register_init(struct onyx *onyx) +{ + int i; + u8 val; + u8 regs[sizeof(initial_values)]; + + if (!onyx->initialised) { + memcpy(regs, initial_values, sizeof(initial_values)); + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &val)) + return -1; + val &= ~ONYX_SILICONVERSION; + val |= initial_values[3]; + regs[3] = val; + } else { + for (i=0; icache[register_map[i]-FIRSTREGISTER]; + } + + for (i=0; iinitialised = 1; + return 0; +} + +static struct transfer_info onyx_transfers[] = { + /* this is first so we can skip it if no input is present... + * No hardware exists with that, but it's here as an example + * of what to do :) */ + { + /* analog input */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_8000_96000, + .transfer_in = 1, + .must_be_clock_source = 0, + .tag = 0, + }, + { + /* if analog and digital are currently off, anything should go, + * so this entry describes everything we can do... */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE +#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE + | SNDRV_PCM_FMTBIT_COMPRESSED_16BE +#endif + , + .rates = SNDRV_PCM_RATE_8000_96000, + .tag = 0, + }, + { + /* analog output */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_8000_96000, + .transfer_in = 0, + .must_be_clock_source = 0, + .tag = 1, + }, + { + /* digital pcm output, also possible for analog out */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .transfer_in = 0, + .must_be_clock_source = 0, + .tag = 2, + }, +#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE +Once alsa gets supports for this kind of thing we can add it... + { + /* digital compressed output */ + .formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .tag = 2, + }, +#endif + {} +}; + +static int onyx_usable(struct codec_info_item *cii, + struct transfer_info *ti, + struct transfer_info *out) +{ + u8 v; + struct onyx *onyx = cii->codec_data; + int spdif_enabled, analog_enabled; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); + spdif_enabled = !!(v & ONYX_SPDIF_ENABLE); + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); + analog_enabled = + (v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT)) + != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT); + mutex_unlock(&onyx->mutex); + + switch (ti->tag) { + case 0: return 1; + case 1: return analog_enabled; + case 2: return spdif_enabled; + } + return 1; +} + +static int onyx_prepare(struct codec_info_item *cii, + struct bus_info *bi, + struct snd_pcm_substream *substream) +{ + u8 v; + struct onyx *onyx = cii->codec_data; + int err = -EBUSY; + + mutex_lock(&onyx->mutex); + +#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE + if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) { + /* mute and lock analog output */ + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); + if (onyx_write_register(onyx + ONYX_REG_DAC_CONTROL, + v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT)) + goto out_unlock; + onyx->analog_locked = 1; + err = 0; + goto out_unlock; + } +#endif + switch (substream->runtime->rate) { + case 32000: + case 44100: + case 48000: + /* these rates are ok for all outputs */ + /* FIXME: program spdif channel control bits here so that + * userspace doesn't have to if it only plays pcm! */ + err = 0; + goto out_unlock; + default: + /* got some rate that the digital output can't do, + * so disable and lock it */ + onyx_read_register(cii->codec_data, ONYX_REG_DIG_INFO4, &v); + if (onyx_write_register(onyx, + ONYX_REG_DIG_INFO4, + v & ~ONYX_SPDIF_ENABLE)) + goto out_unlock; + onyx->spdif_locked = 1; + err = 0; + goto out_unlock; + } + + out_unlock: + mutex_unlock(&onyx->mutex); + + return err; +} + +static int onyx_open(struct codec_info_item *cii, + struct snd_pcm_substream *substream) +{ + struct onyx *onyx = cii->codec_data; + + mutex_lock(&onyx->mutex); + onyx->open_count++; + mutex_unlock(&onyx->mutex); + + return 0; +} + +static int onyx_close(struct codec_info_item *cii, + struct snd_pcm_substream *substream) +{ + struct onyx *onyx = cii->codec_data; + + mutex_lock(&onyx->mutex); + onyx->open_count--; + if (!onyx->open_count) + onyx->spdif_locked = onyx->analog_locked = 0; + mutex_unlock(&onyx->mutex); + + return 0; +} + +static int onyx_switch_clock(struct codec_info_item *cii, + enum clock_switch what) +{ + struct onyx *onyx = cii->codec_data; + + mutex_lock(&onyx->mutex); + /* this *MUST* be more elaborate later... */ + switch (what) { + case CLOCK_SWITCH_PREPARE_SLAVE: + onyx->codec.gpio->methods->all_amps_off(onyx->codec.gpio); + break; + case CLOCK_SWITCH_SLAVE: + onyx->codec.gpio->methods->all_amps_restore(onyx->codec.gpio); + break; + default: /* silence warning */ + break; + } + mutex_unlock(&onyx->mutex); + + return 0; +} + +#ifdef CONFIG_PM + +static int onyx_suspend(struct codec_info_item *cii, pm_message_t state) +{ + struct onyx *onyx = cii->codec_data; + u8 v; + int err = -ENXIO; + + mutex_lock(&onyx->mutex); + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v)) + goto out_unlock; + onyx_write_register(onyx, ONYX_REG_CONTROL, v | ONYX_ADPSV | ONYX_DAPSV); + /* Apple does a sleep here but the datasheet says to do it on resume */ + err = 0; + out_unlock: + mutex_unlock(&onyx->mutex); + + return err; +} + +static int onyx_resume(struct codec_info_item *cii) +{ + struct onyx *onyx = cii->codec_data; + u8 v; + int err = -ENXIO; + + mutex_lock(&onyx->mutex); + /* take codec out of suspend */ + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v)) + goto out_unlock; + onyx_write_register(onyx, ONYX_REG_CONTROL, v & ~(ONYX_ADPSV | ONYX_DAPSV)); + /* FIXME: should divide by sample rate, but 8k is the lowest we go */ + msleep(2205000/8000); + /* reset all values */ + onyx_register_init(onyx); + err = 0; + out_unlock: + mutex_unlock(&onyx->mutex); + + return err; +} + +#endif /* CONFIG_PM */ + +static struct codec_info onyx_codec_info = { + .transfers = onyx_transfers, + .sysclock_factor = 256, + .bus_factor = 64, + .owner = THIS_MODULE, + .usable = onyx_usable, + .prepare = onyx_prepare, + .open = onyx_open, + .close = onyx_close, + .switch_clock = onyx_switch_clock, +#ifdef CONFIG_PM + .suspend = onyx_suspend, + .resume = onyx_resume, +#endif +}; + +static int onyx_init_codec(struct aoa_codec *codec) +{ + struct onyx *onyx = codec_to_onyx(codec); + struct snd_kcontrol *ctl; + struct codec_info *ci = &onyx_codec_info; + u8 v; + int err; + + if (!onyx->codec.gpio || !onyx->codec.gpio->methods) { + printk(KERN_ERR PFX "gpios not assigned!!\n"); + return -EINVAL; + } + + onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0); + msleep(1); + onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1); + msleep(1); + onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0); + msleep(1); + + if (onyx_register_init(onyx)) { + printk(KERN_ERR PFX "failed to initialise onyx registers\n"); + return -ENODEV; + } + + if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, onyx, &ops)) { + printk(KERN_ERR PFX "failed to create onyx snd device!\n"); + return -ENODEV; + } + + /* nothing connected? what a joke! */ + if ((onyx->codec.connected & 0xF) == 0) + return -ENOTCONN; + + /* if no inputs are present... */ + if ((onyx->codec.connected & 0xC) == 0) { + if (!onyx->codec_info) + onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL); + if (!onyx->codec_info) + return -ENOMEM; + ci = onyx->codec_info; + *ci = onyx_codec_info; + ci->transfers++; + } + + /* if no outputs are present... */ + if ((onyx->codec.connected & 3) == 0) { + if (!onyx->codec_info) + onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL); + if (!onyx->codec_info) + return -ENOMEM; + ci = onyx->codec_info; + /* this is fine as there have to be inputs + * if we end up in this part of the code */ + *ci = onyx_codec_info; + ci->transfers[1].formats = 0; + } + + if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev, + aoa_get_card(), + ci, onyx)) { + printk(KERN_ERR PFX "error creating onyx pcm\n"); + return -ENODEV; + } +#define ADDCTL(n) \ + do { \ + ctl = snd_ctl_new1(&n, onyx); \ + if (ctl) { \ + ctl->id.device = \ + onyx->codec.soundbus_dev->pcm->device; \ + err = aoa_snd_ctl_add(ctl); \ + if (err) \ + goto error; \ + } \ + } while (0) + + if (onyx->codec.soundbus_dev->pcm) { + /* give the user appropriate controls + * depending on what inputs are connected */ + if ((onyx->codec.connected & 0xC) == 0xC) + ADDCTL(capture_source_control); + else if (onyx->codec.connected & 4) + onyx_set_capture_source(onyx, 0); + else + onyx_set_capture_source(onyx, 1); + if (onyx->codec.connected & 0xC) + ADDCTL(inputgain_control); + + /* depending on what output is connected, + * give the user appropriate controls */ + if (onyx->codec.connected & 1) { + ADDCTL(volume_control); + ADDCTL(mute_control); + ADDCTL(ovr1_control); + ADDCTL(flt0_control); + ADDCTL(hpf_control); + ADDCTL(dm12_control); + /* spdif control defaults to off */ + } + if (onyx->codec.connected & 2) { + ADDCTL(onyx_spdif_mask); + ADDCTL(onyx_spdif_ctrl); + } + if ((onyx->codec.connected & 3) == 3) + ADDCTL(spdif_control); + /* if only S/PDIF is connected, enable it unconditionally */ + if ((onyx->codec.connected & 3) == 2) { + onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); + v |= ONYX_SPDIF_ENABLE; + onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v); + } + } +#undef ADDCTL + printk(KERN_INFO PFX "attached to onyx codec via i2c\n"); + + return 0; + error: + onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); + snd_device_free(aoa_get_card(), onyx); + return err; +} + +static void onyx_exit_codec(struct aoa_codec *codec) +{ + struct onyx *onyx = codec_to_onyx(codec); + + if (!onyx->codec.soundbus_dev) { + printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n"); + return; + } + onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); +} + +static struct i2c_driver onyx_driver; + +static int onyx_create(struct i2c_adapter *adapter, + struct device_node *node, + int addr) +{ + struct onyx *onyx; + u8 dummy; + + onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL); + + if (!onyx) + return -ENOMEM; + + mutex_init(&onyx->mutex); + onyx->i2c.driver = &onyx_driver; + onyx->i2c.adapter = adapter; + onyx->i2c.addr = addr & 0x7f; + strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE-1); + + if (i2c_attach_client(&onyx->i2c)) { + printk(KERN_ERR PFX "failed to attach to i2c\n"); + goto fail; + } + + /* we try to read from register ONYX_REG_CONTROL + * to check if the codec is present */ + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { + i2c_detach_client(&onyx->i2c); + printk(KERN_ERR PFX "failed to read control register\n"); + goto fail; + } + + strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN-1); + onyx->codec.owner = THIS_MODULE; + onyx->codec.init = onyx_init_codec; + onyx->codec.exit = onyx_exit_codec; + onyx->codec.node = of_node_get(node); + + if (aoa_codec_register(&onyx->codec)) { + i2c_detach_client(&onyx->i2c); + goto fail; + } + printk(KERN_DEBUG PFX "created and attached onyx instance\n"); + return 0; + fail: + kfree(onyx); + return -EINVAL; +} + +static int onyx_i2c_attach(struct i2c_adapter *adapter) +{ + struct device_node *busnode, *dev = NULL; + struct pmac_i2c_bus *bus; + + bus = pmac_i2c_adapter_to_bus(adapter); + if (bus == NULL) + return -ENODEV; + busnode = pmac_i2c_get_bus_node(bus); + + while ((dev = of_get_next_child(busnode, dev)) != NULL) { + if (device_is_compatible(dev, "pcm3052")) { + u32 *addr; + printk(KERN_DEBUG PFX "found pcm3052\n"); + addr = (u32 *) get_property(dev, "reg", NULL); + if (!addr) + return -ENODEV; + return onyx_create(adapter, dev, (*addr)>>1); + } + } + + /* if that didn't work, try desperate mode for older + * machines that have stuff missing from the device tree */ + + if (!device_is_compatible(busnode, "k2-i2c")) + return -ENODEV; + + printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n"); + /* probe both possible addresses for the onyx chip */ + if (onyx_create(adapter, NULL, 0x46) == 0) + return 0; + return onyx_create(adapter, NULL, 0x47); +} + +static int onyx_i2c_detach(struct i2c_client *client) +{ + struct onyx *onyx = container_of(client, struct onyx, i2c); + int err; + + if ((err = i2c_detach_client(client))) + return err; + aoa_codec_unregister(&onyx->codec); + of_node_put(onyx->codec.node); + if (onyx->codec_info) + kfree(onyx->codec_info); + kfree(onyx); + return 0; +} + +static struct i2c_driver onyx_driver = { + .driver = { + .name = "aoa_codec_onyx", + .owner = THIS_MODULE, + }, + .attach_adapter = onyx_i2c_attach, + .detach_client = onyx_i2c_detach, +}; + +static int __init onyx_init(void) +{ + return i2c_add_driver(&onyx_driver); +} + +static void __exit onyx_exit(void) +{ + i2c_del_driver(&onyx_driver); +} + +module_init(onyx_init); +module_exit(onyx_exit); diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.h b/sound/aoa/codecs/snd-aoa-codec-onyx.h new file mode 100644 index 0000000000..aeedda7736 --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.h @@ -0,0 +1,76 @@ +/* + * Apple Onboard Audio driver for Onyx codec (header) + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#ifndef __SND_AOA_CODEC_ONYX_H +#define __SND_AOA_CODEC_ONYX_H +#include +#include +#include +#include +#include + +/* PCM3052 register definitions */ + +/* the attenuation registers take values from + * -1 (0dB) to -127 (-63.0 dB) or others (muted) */ +#define ONYX_REG_DAC_ATTEN_LEFT 65 +#define FIRSTREGISTER ONYX_REG_DAC_ATTEN_LEFT +#define ONYX_REG_DAC_ATTEN_RIGHT 66 + +#define ONYX_REG_CONTROL 67 +# define ONYX_MRST (1<<7) +# define ONYX_SRST (1<<6) +# define ONYX_ADPSV (1<<5) +# define ONYX_DAPSV (1<<4) +# define ONYX_SILICONVERSION (1<<0) +/* all others reserved */ + +#define ONYX_REG_DAC_CONTROL 68 +# define ONYX_OVR1 (1<<6) +# define ONYX_MUTE_RIGHT (1<<1) +# define ONYX_MUTE_LEFT (1<<0) + +#define ONYX_REG_DAC_DEEMPH 69 +# define ONYX_DIGDEEMPH_SHIFT 5 +# define ONYX_DIGDEEMPH_MASK (3< +#include +int main() { + int dB2; + printf("/" "* This file is only included exactly once!\n"); + printf(" *\n"); + printf(" * If they'd only tell us that generating this table was\n"); + printf(" * as easy as calculating\n"); + printf(" * hwvalue = 1048576.0*exp(0.057564628*dB*2)\n"); + printf(" * :) *" "/\n"); + printf("static int tas_gaintable[] = {\n"); + printf(" 0x000000, /" "* -infinity dB *" "/\n"); + for (dB2=-140;dB2<=36;dB2++) + printf(" 0x%.6x, /" "* %-02.1f dB *" "/\n", (int)(1048576.0*exp(0.057564628*dB2)), dB2/2.0); + printf("};\n\n"); +} + +*/ + +/* This file is only included exactly once! + * + * If they'd only tell us that generating this table was + * as easy as calculating + * hwvalue = 1048576.0*exp(0.057564628*dB*2) + * :) */ +static int tas_gaintable[] = { + 0x000000, /* -infinity dB */ + 0x00014b, /* -70.0 dB */ + 0x00015f, /* -69.5 dB */ + 0x000174, /* -69.0 dB */ + 0x00018a, /* -68.5 dB */ + 0x0001a1, /* -68.0 dB */ + 0x0001ba, /* -67.5 dB */ + 0x0001d4, /* -67.0 dB */ + 0x0001f0, /* -66.5 dB */ + 0x00020d, /* -66.0 dB */ + 0x00022c, /* -65.5 dB */ + 0x00024d, /* -65.0 dB */ + 0x000270, /* -64.5 dB */ + 0x000295, /* -64.0 dB */ + 0x0002bc, /* -63.5 dB */ + 0x0002e6, /* -63.0 dB */ + 0x000312, /* -62.5 dB */ + 0x000340, /* -62.0 dB */ + 0x000372, /* -61.5 dB */ + 0x0003a6, /* -61.0 dB */ + 0x0003dd, /* -60.5 dB */ + 0x000418, /* -60.0 dB */ + 0x000456, /* -59.5 dB */ + 0x000498, /* -59.0 dB */ + 0x0004de, /* -58.5 dB */ + 0x000528, /* -58.0 dB */ + 0x000576, /* -57.5 dB */ + 0x0005c9, /* -57.0 dB */ + 0x000620, /* -56.5 dB */ + 0x00067d, /* -56.0 dB */ + 0x0006e0, /* -55.5 dB */ + 0x000748, /* -55.0 dB */ + 0x0007b7, /* -54.5 dB */ + 0x00082c, /* -54.0 dB */ + 0x0008a8, /* -53.5 dB */ + 0x00092b, /* -53.0 dB */ + 0x0009b6, /* -52.5 dB */ + 0x000a49, /* -52.0 dB */ + 0x000ae5, /* -51.5 dB */ + 0x000b8b, /* -51.0 dB */ + 0x000c3a, /* -50.5 dB */ + 0x000cf3, /* -50.0 dB */ + 0x000db8, /* -49.5 dB */ + 0x000e88, /* -49.0 dB */ + 0x000f64, /* -48.5 dB */ + 0x00104e, /* -48.0 dB */ + 0x001145, /* -47.5 dB */ + 0x00124b, /* -47.0 dB */ + 0x001361, /* -46.5 dB */ + 0x001487, /* -46.0 dB */ + 0x0015be, /* -45.5 dB */ + 0x001708, /* -45.0 dB */ + 0x001865, /* -44.5 dB */ + 0x0019d8, /* -44.0 dB */ + 0x001b60, /* -43.5 dB */ + 0x001cff, /* -43.0 dB */ + 0x001eb7, /* -42.5 dB */ + 0x002089, /* -42.0 dB */ + 0x002276, /* -41.5 dB */ + 0x002481, /* -41.0 dB */ + 0x0026ab, /* -40.5 dB */ + 0x0028f5, /* -40.0 dB */ + 0x002b63, /* -39.5 dB */ + 0x002df5, /* -39.0 dB */ + 0x0030ae, /* -38.5 dB */ + 0x003390, /* -38.0 dB */ + 0x00369e, /* -37.5 dB */ + 0x0039db, /* -37.0 dB */ + 0x003d49, /* -36.5 dB */ + 0x0040ea, /* -36.0 dB */ + 0x0044c3, /* -35.5 dB */ + 0x0048d6, /* -35.0 dB */ + 0x004d27, /* -34.5 dB */ + 0x0051b9, /* -34.0 dB */ + 0x005691, /* -33.5 dB */ + 0x005bb2, /* -33.0 dB */ + 0x006121, /* -32.5 dB */ + 0x0066e3, /* -32.0 dB */ + 0x006cfb, /* -31.5 dB */ + 0x007370, /* -31.0 dB */ + 0x007a48, /* -30.5 dB */ + 0x008186, /* -30.0 dB */ + 0x008933, /* -29.5 dB */ + 0x009154, /* -29.0 dB */ + 0x0099f1, /* -28.5 dB */ + 0x00a310, /* -28.0 dB */ + 0x00acba, /* -27.5 dB */ + 0x00b6f6, /* -27.0 dB */ + 0x00c1cd, /* -26.5 dB */ + 0x00cd49, /* -26.0 dB */ + 0x00d973, /* -25.5 dB */ + 0x00e655, /* -25.0 dB */ + 0x00f3fb, /* -24.5 dB */ + 0x010270, /* -24.0 dB */ + 0x0111c0, /* -23.5 dB */ + 0x0121f9, /* -23.0 dB */ + 0x013328, /* -22.5 dB */ + 0x01455b, /* -22.0 dB */ + 0x0158a2, /* -21.5 dB */ + 0x016d0e, /* -21.0 dB */ + 0x0182af, /* -20.5 dB */ + 0x019999, /* -20.0 dB */ + 0x01b1de, /* -19.5 dB */ + 0x01cb94, /* -19.0 dB */ + 0x01e6cf, /* -18.5 dB */ + 0x0203a7, /* -18.0 dB */ + 0x022235, /* -17.5 dB */ + 0x024293, /* -17.0 dB */ + 0x0264db, /* -16.5 dB */ + 0x02892c, /* -16.0 dB */ + 0x02afa3, /* -15.5 dB */ + 0x02d862, /* -15.0 dB */ + 0x03038a, /* -14.5 dB */ + 0x033142, /* -14.0 dB */ + 0x0361af, /* -13.5 dB */ + 0x0394fa, /* -13.0 dB */ + 0x03cb50, /* -12.5 dB */ + 0x0404de, /* -12.0 dB */ + 0x0441d5, /* -11.5 dB */ + 0x048268, /* -11.0 dB */ + 0x04c6d0, /* -10.5 dB */ + 0x050f44, /* -10.0 dB */ + 0x055c04, /* -9.5 dB */ + 0x05ad50, /* -9.0 dB */ + 0x06036e, /* -8.5 dB */ + 0x065ea5, /* -8.0 dB */ + 0x06bf44, /* -7.5 dB */ + 0x07259d, /* -7.0 dB */ + 0x079207, /* -6.5 dB */ + 0x0804dc, /* -6.0 dB */ + 0x087e80, /* -5.5 dB */ + 0x08ff59, /* -5.0 dB */ + 0x0987d5, /* -4.5 dB */ + 0x0a1866, /* -4.0 dB */ + 0x0ab189, /* -3.5 dB */ + 0x0b53be, /* -3.0 dB */ + 0x0bff91, /* -2.5 dB */ + 0x0cb591, /* -2.0 dB */ + 0x0d765a, /* -1.5 dB */ + 0x0e4290, /* -1.0 dB */ + 0x0f1adf, /* -0.5 dB */ + 0x100000, /* 0.0 dB */ + 0x10f2b4, /* 0.5 dB */ + 0x11f3c9, /* 1.0 dB */ + 0x13041a, /* 1.5 dB */ + 0x14248e, /* 2.0 dB */ + 0x15561a, /* 2.5 dB */ + 0x1699c0, /* 3.0 dB */ + 0x17f094, /* 3.5 dB */ + 0x195bb8, /* 4.0 dB */ + 0x1adc61, /* 4.5 dB */ + 0x1c73d5, /* 5.0 dB */ + 0x1e236d, /* 5.5 dB */ + 0x1fec98, /* 6.0 dB */ + 0x21d0d9, /* 6.5 dB */ + 0x23d1cd, /* 7.0 dB */ + 0x25f125, /* 7.5 dB */ + 0x2830af, /* 8.0 dB */ + 0x2a9254, /* 8.5 dB */ + 0x2d1818, /* 9.0 dB */ + 0x2fc420, /* 9.5 dB */ + 0x3298b0, /* 10.0 dB */ + 0x35982f, /* 10.5 dB */ + 0x38c528, /* 11.0 dB */ + 0x3c224c, /* 11.5 dB */ + 0x3fb278, /* 12.0 dB */ + 0x4378b0, /* 12.5 dB */ + 0x477829, /* 13.0 dB */ + 0x4bb446, /* 13.5 dB */ + 0x5030a1, /* 14.0 dB */ + 0x54f106, /* 14.5 dB */ + 0x59f980, /* 15.0 dB */ + 0x5f4e52, /* 15.5 dB */ + 0x64f403, /* 16.0 dB */ + 0x6aef5e, /* 16.5 dB */ + 0x714575, /* 17.0 dB */ + 0x77fbaa, /* 17.5 dB */ + 0x7f17af, /* 18.0 dB */ +}; + diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c new file mode 100644 index 0000000000..2e39ff6ee3 --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -0,0 +1,654 @@ +/* + * Apple Onboard Audio driver for tas codec + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + * + * Open questions: + * - How to distinguish between 3004 and versions? + * + * FIXMEs: + * - This codec driver doesn't honour the 'connected' + * property of the aoa_codec struct, hence if + * it is used in machines where not everything is + * connected it will display wrong mixer elements. + * - Driver assumes that the microphone is always + * monaureal and connected to the right channel of + * the input. This should also be a codec-dependent + * flag, maybe the codec should have 3 different + * bits for the three different possibilities how + * it can be hooked up... + * But as long as I don't see any hardware hooked + * up that way... + * - As Apple notes in their code, the tas3004 seems + * to delay the right channel by one sample. You can + * see this when for example recording stereo in + * audacity, or recording the tas output via cable + * on another machine (use a sinus generator or so). + * I tried programming the BiQuads but couldn't + * make the delay work, maybe someone can read the + * datasheet and fix it. The relevant Apple comment + * is in AppleTAS3004Audio.cpp lines 1637 ff. Note + * that their comment describing how they program + * the filters sucks... + * + * Other things: + * - this should actually register *two* aoa_codec + * structs since it has two inputs. Then it must + * use the prepare callback to forbid running the + * secondary output on a different clock. + * Also, whatever bus knows how to do this must + * provide two soundbus_dev devices and the fabric + * must be able to link them correctly. + * + * I don't even know if Apple ever uses the second + * port on the tas3004 though, I don't think their + * i2s controllers can even do it. OTOH, they all + * derive the clocks from common clocks, so it + * might just be possible. The framework allows the + * codec to refine the transfer_info items in the + * usable callback, so we can simply remove the + * rates the second instance is not using when it + * actually is in use. + * Maybe we'll need to make the sound busses have + * a 'clock group id' value so the codec can + * determine if the two outputs can be driven at + * the same time. But that is likely overkill, up + * to the fabric to not link them up incorrectly, + * and up to the hardware designer to not wire + * them up in some weird unusable way. + */ +#include +#include +#include +#include +#include +#include +#include +MODULE_AUTHOR("Johannes Berg "); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("tas codec driver for snd-aoa"); + +#include "snd-aoa-codec-tas.h" +#include "snd-aoa-codec-tas-gain-table.h" +#include "../aoa.h" +#include "../soundbus/soundbus.h" + + +#define PFX "snd-aoa-codec-tas: " + +struct tas { + struct aoa_codec codec; + struct i2c_client i2c; + u32 muted_l:1, muted_r:1, + controls_created:1; + u8 cached_volume_l, cached_volume_r; + u8 mixer_l[3], mixer_r[3]; + u8 acr; +}; + +static struct tas *codec_to_tas(struct aoa_codec *codec) +{ + return container_of(codec, struct tas, codec); +} + +static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) +{ + if (len == 1) + return i2c_smbus_write_byte_data(&tas->i2c, reg, *data); + else + return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data); +} + +static void tas_set_volume(struct tas *tas) +{ + u8 block[6]; + int tmp; + u8 left, right; + + left = tas->cached_volume_l; + right = tas->cached_volume_r; + + if (left > 177) left = 177; + if (right > 177) right = 177; + + if (tas->muted_l) left = 0; + if (tas->muted_r) right = 0; + + /* analysing the volume and mixer tables shows + * that they are similar enough when we shift + * the mixer table down by 4 bits. The error + * is miniscule, in just one item the error + * is 1, at a value of 0x07f17b (mixer table + * value is 0x07f17a) */ + tmp = tas_gaintable[left]; + block[0] = tmp>>20; + block[1] = tmp>>12; + block[2] = tmp>>4; + tmp = tas_gaintable[right]; + block[3] = tmp>>20; + block[4] = tmp>>12; + block[5] = tmp>>4; + tas_write_reg(tas, TAS_REG_VOL, 6, block); +} + +static void tas_set_mixer(struct tas *tas) +{ + u8 block[9]; + int tmp, i; + u8 val; + + for (i=0;i<3;i++) { + val = tas->mixer_l[i]; + if (val > 177) val = 177; + tmp = tas_gaintable[val]; + block[3*i+0] = tmp>>16; + block[3*i+1] = tmp>>8; + block[3*i+2] = tmp; + } + tas_write_reg(tas, TAS_REG_LMIX, 9, block); + + for (i=0;i<3;i++) { + val = tas->mixer_r[i]; + if (val > 177) val = 177; + tmp = tas_gaintable[val]; + block[3*i+0] = tmp>>16; + block[3*i+1] = tmp>>8; + block[3*i+2] = tmp; + } + tas_write_reg(tas, TAS_REG_RMIX, 9, block); +} + +/* alsa stuff */ + +static int tas_dev_register(struct snd_device *dev) +{ + return 0; +} + +static struct snd_device_ops ops = { + .dev_register = tas_dev_register, +}; + +static int tas_snd_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 177; + return 0; +} + +static int tas_snd_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas->cached_volume_l; + ucontrol->value.integer.value[1] = tas->cached_volume_r; + return 0; +} + +static int tas_snd_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + if (tas->cached_volume_l == ucontrol->value.integer.value[0] + && tas->cached_volume_r == ucontrol->value.integer.value[1]) + return 0; + + tas->cached_volume_l = ucontrol->value.integer.value[0]; + tas->cached_volume_r = ucontrol->value.integer.value[1]; + tas_set_volume(tas); + return 1; +} + +static struct snd_kcontrol_new volume_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_vol_info, + .get = tas_snd_vol_get, + .put = tas_snd_vol_put, +}; + +static int tas_snd_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int tas_snd_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = !tas->muted_l; + ucontrol->value.integer.value[1] = !tas->muted_r; + return 0; +} + +static int tas_snd_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + if (tas->muted_l == !ucontrol->value.integer.value[0] + && tas->muted_r == !ucontrol->value.integer.value[1]) + return 0; + + tas->muted_l = !ucontrol->value.integer.value[0]; + tas->muted_r = !ucontrol->value.integer.value[1]; + tas_set_volume(tas); + return 1; +} + +static struct snd_kcontrol_new mute_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_mute_info, + .get = tas_snd_mute_get, + .put = tas_snd_mute_put, +}; + +static int tas_snd_mixer_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 177; + return 0; +} + +static int tas_snd_mixer_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + int idx = kcontrol->private_value; + + ucontrol->value.integer.value[0] = tas->mixer_l[idx]; + ucontrol->value.integer.value[1] = tas->mixer_r[idx]; + + return 0; +} + +static int tas_snd_mixer_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + int idx = kcontrol->private_value; + + if (tas->mixer_l[idx] == ucontrol->value.integer.value[0] + && tas->mixer_r[idx] == ucontrol->value.integer.value[1]) + return 0; + + tas->mixer_l[idx] = ucontrol->value.integer.value[0]; + tas->mixer_r[idx] = ucontrol->value.integer.value[1]; + + tas_set_mixer(tas); + return 1; +} + +#define MIXER_CONTROL(n,descr,idx) \ +static struct snd_kcontrol_new n##_control = { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = descr " Playback Volume", \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = tas_snd_mixer_info, \ + .get = tas_snd_mixer_get, \ + .put = tas_snd_mixer_put, \ + .private_value = idx, \ +} + +MIXER_CONTROL(pcm1, "PCM1", 0); +MIXER_CONTROL(monitor, "Monitor", 2); + +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { "Line-In", "Microphone" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item > 1) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B); + return 0; +} + +static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + int oldacr = tas->acr; + + tas->acr &= ~TAS_ACR_INPUT_B; + if (ucontrol->value.enumerated.item[0]) + tas->acr |= TAS_ACR_INPUT_B; + if (oldacr == tas->acr) + return 0; + tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr); + return 1; +} + +static struct snd_kcontrol_new capture_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* If we name this 'Input Source', it properly shows up in + * alsamixer as a selection, * but it's shown under the + * 'Playback' category. + * If I name it 'Capture Source', it shows up in strange + * ways (two bools of which one can be selected at a + * time) but at least it's shown in the 'Capture' + * category. + * I was told that this was due to backward compatibility, + * but I don't understand then why the mangling is *not* + * done when I name it "Input Source"..... + */ + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_capture_source_info, + .get = tas_snd_capture_source_get, + .put = tas_snd_capture_source_put, +}; + + +static struct transfer_info tas_transfers[] = { + { + /* input */ + .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .transfer_in = 1, + }, + { + /* output */ + .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .transfer_in = 0, + }, + {} +}; + +static int tas_usable(struct codec_info_item *cii, + struct transfer_info *ti, + struct transfer_info *out) +{ + return 1; +} + +static int tas_reset_init(struct tas *tas) +{ + u8 tmp; + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0); + msleep(1); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1); + msleep(1); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0); + msleep(1); + + tas->acr &= ~TAS_ACR_ANALOG_PDOWN; + tas->acr |= TAS_ACR_B_MONAUREAL | TAS_ACR_B_MON_SEL_RIGHT; + if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr)) + return -ENODEV; + + tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT; + if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp)) + return -ENODEV; + + tmp = 0; + if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp)) + return -ENODEV; + + return 0; +} + +/* we are controlled via i2c and assume that is always up + * If that wasn't the case, we'd have to suspend once + * our i2c device is suspended, and then take note of that! */ +static int tas_suspend(struct tas *tas) +{ + tas->acr |= TAS_ACR_ANALOG_PDOWN; + tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr); + return 0; +} + +static int tas_resume(struct tas *tas) +{ + /* reset codec */ + tas_reset_init(tas); + tas_set_volume(tas); + tas_set_mixer(tas); + return 0; +} + +#ifdef CONFIG_PM +static int _tas_suspend(struct codec_info_item *cii, pm_message_t state) +{ + return tas_suspend(cii->codec_data); +} + +static int _tas_resume(struct codec_info_item *cii) +{ + return tas_resume(cii->codec_data); +} +#endif + +static struct codec_info tas_codec_info = { + .transfers = tas_transfers, + /* in theory, we can drive it at 512 too... + * but so far the framework doesn't allow + * for that and I don't see much point in it. */ + .sysclock_factor = 256, + /* same here, could be 32 for just one 16 bit format */ + .bus_factor = 64, + .owner = THIS_MODULE, + .usable = tas_usable, +#ifdef CONFIG_PM + .suspend = _tas_suspend, + .resume = _tas_resume, +#endif +}; + +static int tas_init_codec(struct aoa_codec *codec) +{ + struct tas *tas = codec_to_tas(codec); + int err; + + if (!tas->codec.gpio || !tas->codec.gpio->methods) { + printk(KERN_ERR PFX "gpios not assigned!!\n"); + return -EINVAL; + } + + if (tas_reset_init(tas)) { + printk(KERN_ERR PFX "tas failed to initialise\n"); + return -ENXIO; + } + + if (tas->codec.soundbus_dev->attach_codec(tas->codec.soundbus_dev, + aoa_get_card(), + &tas_codec_info, tas)) { + printk(KERN_ERR PFX "error attaching tas to soundbus\n"); + return -ENODEV; + } + + if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, tas, &ops)) { + printk(KERN_ERR PFX "failed to create tas snd device!\n"); + return -ENODEV; + } + err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas)); + if (err) + goto error; + + return 0; + error: + tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas); + snd_device_free(aoa_get_card(), tas); + return err; +} + +static void tas_exit_codec(struct aoa_codec *codec) +{ + struct tas *tas = codec_to_tas(codec); + + if (!tas->codec.soundbus_dev) + return; + tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas); +} + + +static struct i2c_driver tas_driver; + +static int tas_create(struct i2c_adapter *adapter, + struct device_node *node, + int addr) +{ + struct tas *tas; + + tas = kzalloc(sizeof(struct tas), GFP_KERNEL); + + if (!tas) + return -ENOMEM; + + tas->i2c.driver = &tas_driver; + tas->i2c.adapter = adapter; + tas->i2c.addr = addr; + strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE-1); + + if (i2c_attach_client(&tas->i2c)) { + printk(KERN_ERR PFX "failed to attach to i2c\n"); + goto fail; + } + + strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN-1); + tas->codec.owner = THIS_MODULE; + tas->codec.init = tas_init_codec; + tas->codec.exit = tas_exit_codec; + tas->codec.node = of_node_get(node); + + if (aoa_codec_register(&tas->codec)) { + goto detach; + } + printk(KERN_DEBUG "snd-aoa-codec-tas: created and attached tas instance\n"); + return 0; + detach: + i2c_detach_client(&tas->i2c); + fail: + kfree(tas); + return -EINVAL; +} + +static int tas_i2c_attach(struct i2c_adapter *adapter) +{ + struct device_node *busnode, *dev = NULL; + struct pmac_i2c_bus *bus; + + bus = pmac_i2c_adapter_to_bus(adapter); + if (bus == NULL) + return -ENODEV; + busnode = pmac_i2c_get_bus_node(bus); + + while ((dev = of_get_next_child(busnode, dev)) != NULL) { + if (device_is_compatible(dev, "tas3004")) { + u32 *addr; + printk(KERN_DEBUG PFX "found tas3004\n"); + addr = (u32 *) get_property(dev, "reg", NULL); + if (!addr) + continue; + return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f); + } + /* older machines have no 'codec' node with a 'compatible' + * property that says 'tas3004', they just have a 'deq' + * node without any such property... */ + if (strcmp(dev->name, "deq") == 0) { + u32 *_addr, addr; + printk(KERN_DEBUG PFX "found 'deq' node\n"); + _addr = (u32 *) get_property(dev, "i2c-address", NULL); + if (!_addr) + continue; + addr = ((*_addr) >> 1) & 0x7f; + /* now, if the address doesn't match any of the two + * that a tas3004 can have, we cannot handle this. + * I doubt it ever happens but hey. */ + if (addr != 0x34 && addr != 0x35) + continue; + return tas_create(adapter, dev, addr); + } + } + return -ENODEV; +} + +static int tas_i2c_detach(struct i2c_client *client) +{ + struct tas *tas = container_of(client, struct tas, i2c); + int err; + u8 tmp = TAS_ACR_ANALOG_PDOWN; + + if ((err = i2c_detach_client(client))) + return err; + aoa_codec_unregister(&tas->codec); + of_node_put(tas->codec.node); + + /* power down codec chip */ + tas_write_reg(tas, TAS_REG_ACR, 1, &tmp); + + kfree(tas); + return 0; +} + +static struct i2c_driver tas_driver = { + .driver = { + .name = "aoa_codec_tas", + .owner = THIS_MODULE, + }, + .attach_adapter = tas_i2c_attach, + .detach_client = tas_i2c_detach, +}; + +static int __init tas_init(void) +{ + return i2c_add_driver(&tas_driver); +} + +static void __exit tas_exit(void) +{ + i2c_del_driver(&tas_driver); +} + +module_init(tas_init); +module_exit(tas_exit); diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.h b/sound/aoa/codecs/snd-aoa-codec-tas.h new file mode 100644 index 0000000000..daf81f45d8 --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-tas.h @@ -0,0 +1,47 @@ +/* + * Apple Onboard Audio driver for tas codec (header) + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#ifndef __SND_AOA_CODECTASH +#define __SND_AOA_CODECTASH + +#define TAS_REG_MCS 0x01 /* main control */ +# define TAS_MCS_FASTLOAD (1<<7) +# define TAS_MCS_SCLK64 (1<<6) +# define TAS_MCS_SPORT_MODE_MASK (3<<4) +# define TAS_MCS_SPORT_MODE_I2S (2<<4) +# define TAS_MCS_SPORT_MODE_RJ (1<<4) +# define TAS_MCS_SPORT_MODE_LJ (0<<4) +# define TAS_MCS_SPORT_WL_MASK (3<<0) +# define TAS_MCS_SPORT_WL_16BIT (0<<0) +# define TAS_MCS_SPORT_WL_18BIT (1<<0) +# define TAS_MCS_SPORT_WL_20BIT (2<<0) +# define TAS_MCS_SPORT_WL_24BIT (3<<0) + +#define TAS_REG_DRC 0x02 +#define TAS_REG_VOL 0x04 +#define TAS_REG_TREBLE 0x05 +#define TAS_REG_BASS 0x06 +#define TAS_REG_LMIX 0x07 +#define TAS_REG_RMIX 0x08 + +#define TAS_REG_ACR 0x40 /* analog control */ +# define TAS_ACR_B_MONAUREAL (1<<7) +# define TAS_ACR_B_MON_SEL_RIGHT (1<<6) +# define TAS_ACR_DEEMPH_MASK (3<<2) +# define TAS_ACR_DEEMPH_OFF (0<<2) +# define TAS_ACR_DEEMPH_48KHz (1<<2) +# define TAS_ACR_DEEMPH_44KHz (2<<2) +# define TAS_ACR_INPUT_B (1<<1) +# define TAS_ACR_ANALOG_PDOWN (1<<0) + +#define TAS_REG_MCS2 0x43 /* main control 2 */ +# define TAS_MCS2_ALLPASS (1<<1) + +#define TAS_REG_LEFT_BIQUAD6 0x10 +#define TAS_REG_RIGHT_BIQUAD6 0x19 + +#endif /* __SND_AOA_CODECTASH */ diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/snd-aoa-codec-toonie.c new file mode 100644 index 0000000000..bcc555647e --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-toonie.c @@ -0,0 +1,141 @@ +/* + * Apple Onboard Audio driver for Toonie codec + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + * + * + * This is a driver for the toonie codec chip. This chip is present + * on the Mac Mini and is nothing but a DAC. + */ +#include +#include +MODULE_AUTHOR("Johannes Berg "); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("toonie codec driver for snd-aoa"); + +#include "../aoa.h" +#include "../soundbus/soundbus.h" + + +#define PFX "snd-aoa-codec-toonie: " + +struct toonie { + struct aoa_codec codec; +}; +#define codec_to_toonie(c) container_of(c, struct toonie, codec) + +static int toonie_dev_register(struct snd_device *dev) +{ + return 0; +} + +static struct snd_device_ops ops = { + .dev_register = toonie_dev_register, +}; + +static struct transfer_info toonie_transfers[] = { + /* This thing *only* has analog output, + * the rates are taken from Info.plist + * from Darwin. */ + { + .formats = SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + }, + {} +}; + +#ifdef CONFIG_PM +static int toonie_suspend(struct codec_info_item *cii, pm_message_t state) +{ + /* can we turn it off somehow? */ + return 0; +} + +static int toonie_resume(struct codec_info_item *cii) +{ + return 0; +} +#endif /* CONFIG_PM */ + +static struct codec_info toonie_codec_info = { + .transfers = toonie_transfers, + .sysclock_factor = 256, + .bus_factor = 64, + .owner = THIS_MODULE, +#ifdef CONFIG_PM + .suspend = toonie_suspend, + .resume = toonie_resume, +#endif +}; + +static int toonie_init_codec(struct aoa_codec *codec) +{ + struct toonie *toonie = codec_to_toonie(codec); + + if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, toonie, &ops)) { + printk(KERN_ERR PFX "failed to create toonie snd device!\n"); + return -ENODEV; + } + + /* nothing connected? what a joke! */ + if (toonie->codec.connected != 1) + return -ENOTCONN; + + if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev, + aoa_get_card(), + &toonie_codec_info, toonie)) { + printk(KERN_ERR PFX "error creating toonie pcm\n"); + return -ENODEV; + } + + return 0; +} + +static void toonie_exit_codec(struct aoa_codec *codec) +{ + struct toonie *toonie = codec_to_toonie(codec); + + if (!toonie->codec.soundbus_dev) { + printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n"); + return; + } + toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie); +} + +static struct toonie *toonie; + +static int __init toonie_init(void) +{ + toonie = kzalloc(sizeof(struct toonie), GFP_KERNEL); + + if (!toonie) + return -ENOMEM; + + strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name)); + toonie->codec.owner = THIS_MODULE; + toonie->codec.init = toonie_init_codec; + toonie->codec.exit = toonie_exit_codec; + + if (aoa_codec_register(&toonie->codec)) { + kfree(toonie); + return -EINVAL; + } + + return 0; +} + +static void __exit toonie_exit(void) +{ + aoa_codec_unregister(&toonie->codec); + kfree(toonie); +} + +module_init(toonie_init); +module_exit(toonie_exit); diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile new file mode 100644 index 0000000000..62dc7287f6 --- /dev/null +++ b/sound/aoa/core/Makefile @@ -0,0 +1,5 @@ +obj-$(CONFIG_SND_AOA) += snd-aoa.o +snd-aoa-objs := snd-aoa-core.o \ + snd-aoa-alsa.o \ + snd-aoa-gpio-pmf.o \ + snd-aoa-gpio-feature.o diff --git a/sound/aoa/core/snd-aoa-alsa.c b/sound/aoa/core/snd-aoa-alsa.c new file mode 100644 index 0000000000..b42fdea77e --- /dev/null +++ b/sound/aoa/core/snd-aoa-alsa.c @@ -0,0 +1,98 @@ +/* + * Apple Onboard Audio Alsa helpers + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#include +#include "snd-aoa-alsa.h" + +static int index = -1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "index for AOA sound card."); + +static struct aoa_card *aoa_card; + +int aoa_alsa_init(char *name, struct module *mod) +{ + struct snd_card *alsa_card; + int err; + + if (aoa_card) + /* cannot be EEXIST due to usage in aoa_fabric_register */ + return -EBUSY; + + alsa_card = snd_card_new(index, name, mod, sizeof(struct aoa_card)); + if (!alsa_card) + return -ENOMEM; + aoa_card = alsa_card->private_data; + aoa_card->alsa_card = alsa_card; + strlcpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver)); + strlcpy(alsa_card->shortname, name, sizeof(alsa_card->shortname)); + strlcpy(alsa_card->longname, name, sizeof(alsa_card->longname)); + strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername)); + err = snd_card_register(aoa_card->alsa_card); + if (err < 0) { + printk(KERN_ERR "snd-aoa: couldn't register alsa card\n"); + snd_card_free(aoa_card->alsa_card); + aoa_card = NULL; + return err; + } + return 0; +} + +struct snd_card *aoa_get_card(void) +{ + if (aoa_card) + return aoa_card->alsa_card; + return NULL; +} +EXPORT_SYMBOL_GPL(aoa_get_card); + +void aoa_alsa_cleanup(void) +{ + if (aoa_card) { + snd_card_free(aoa_card->alsa_card); + aoa_card = NULL; + } +} + +int aoa_snd_device_new(snd_device_type_t type, + void * device_data, struct snd_device_ops * ops) +{ + struct snd_card *card = aoa_get_card(); + int err; + + if (!card) return -ENOMEM; + + err = snd_device_new(card, type, device_data, ops); + if (err) { + printk(KERN_ERR "snd-aoa: failed to create snd device (%d)\n", err); + return err; + } + err = snd_device_register(card, device_data); + if (err) { + printk(KERN_ERR "snd-aoa: failed to register " + "snd device (%d)\n", err); + printk(KERN_ERR "snd-aoa: have you forgotten the " + "dev_register callback?\n"); + snd_device_free(card, device_data); + } + return err; +} +EXPORT_SYMBOL_GPL(aoa_snd_device_new); + +int aoa_snd_ctl_add(struct snd_kcontrol* control) +{ + int err; + + if (!aoa_card) return -ENODEV; + + err = snd_ctl_add(aoa_card->alsa_card, control); + if (err) + printk(KERN_ERR "snd-aoa: failed to add alsa control (%d)\n", + err); + return err; +} +EXPORT_SYMBOL_GPL(aoa_snd_ctl_add); diff --git a/sound/aoa/core/snd-aoa-alsa.h b/sound/aoa/core/snd-aoa-alsa.h new file mode 100644 index 0000000000..660d2f1793 --- /dev/null +++ b/sound/aoa/core/snd-aoa-alsa.h @@ -0,0 +1,16 @@ +/* + * Apple Onboard Audio Alsa private helpers + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#ifndef __SND_AOA_ALSA_H +#define __SND_AOA_ALSA_H +#include "../aoa.h" + +extern int aoa_alsa_init(char *name, struct module *mod); +extern void aoa_alsa_cleanup(void); + +#endif /* __SND_AOA_ALSA_H */ diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/snd-aoa-core.c new file mode 100644 index 0000000000..ecd2d8263f --- /dev/null +++ b/sound/aoa/core/snd-aoa-core.c @@ -0,0 +1,162 @@ +/* + * Apple Onboard Audio driver core + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#include +#include +#include +#include "../aoa.h" +#include "snd-aoa-alsa.h" + +MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver"); +MODULE_AUTHOR("Johannes Berg "); +MODULE_LICENSE("GPL"); + +/* We allow only one fabric. This simplifies things, + * and more don't really make that much sense */ +static struct aoa_fabric *fabric; +static LIST_HEAD(codec_list); + +static int attach_codec_to_fabric(struct aoa_codec *c) +{ + int err; + + if (!try_module_get(c->owner)) + return -EBUSY; + /* found_codec has to be assigned */ + err = -ENOENT; + if (fabric->found_codec) + err = fabric->found_codec(c); + if (err) { + module_put(c->owner); + printk(KERN_ERR "snd-aoa: fabric didn't like codec %s\n", + c->name); + return err; + } + c->fabric = fabric; + + err = 0; + if (c->init) + err = c->init(c); + if (err) { + printk(KERN_ERR "snd-aoa: codec %s didn't init\n", c->name); + c->fabric = NULL; + if (fabric->remove_codec) + fabric->remove_codec(c); + module_put(c->owner); + return err; + } + if (fabric->attached_codec) + fabric->attached_codec(c); + return 0; +} + +int aoa_codec_register(struct aoa_codec *codec) +{ + int err = 0; + + /* if there's a fabric already, we can tell if we + * will want to have this codec, so propagate error + * through. Otherwise, this will happen later... */ + if (fabric) + err = attach_codec_to_fabric(codec); + if (!err) + list_add(&codec->list, &codec_list); + return err; +} +EXPORT_SYMBOL_GPL(aoa_codec_register); + +void aoa_codec_unregister(struct aoa_codec *codec) +{ + list_del(&codec->list); + if (codec->fabric && codec->exit) + codec->exit(codec); + if (fabric && fabric->remove_codec) + fabric->remove_codec(codec); + codec->fabric = NULL; + module_put(codec->owner); +} +EXPORT_SYMBOL_GPL(aoa_codec_unregister); + +int aoa_fabric_register(struct aoa_fabric *new_fabric) +{ + struct aoa_codec *c; + int err; + + /* allow querying for presence of fabric + * (i.e. do this test first!) */ + if (new_fabric == fabric) { + err = -EALREADY; + goto attach; + } + if (fabric) + return -EEXIST; + if (!new_fabric) + return -EINVAL; + + err = aoa_alsa_init(new_fabric->name, new_fabric->owner); + if (err) + return err; + + fabric = new_fabric; + + attach: + list_for_each_entry(c, &codec_list, list) { + if (c->fabric != fabric) + attach_codec_to_fabric(c); + } + return err; +} +EXPORT_SYMBOL_GPL(aoa_fabric_register); + +void aoa_fabric_unregister(struct aoa_fabric *old_fabric) +{ + struct aoa_codec *c; + + if (fabric != old_fabric) + return; + + list_for_each_entry(c, &codec_list, list) { + if (c->fabric) + aoa_fabric_unlink_codec(c); + } + + aoa_alsa_cleanup(); + + fabric = NULL; +} +EXPORT_SYMBOL_GPL(aoa_fabric_unregister); + +void aoa_fabric_unlink_codec(struct aoa_codec *codec) +{ + if (!codec->fabric) { + printk(KERN_ERR "snd-aoa: fabric unassigned " + "in aoa_fabric_unlink_codec\n"); + dump_stack(); + return; + } + if (codec->exit) + codec->exit(codec); + if (codec->fabric->remove_codec) + codec->fabric->remove_codec(codec); + codec->fabric = NULL; + module_put(codec->owner); +} +EXPORT_SYMBOL_GPL(aoa_fabric_unlink_codec); + +static int __init aoa_init(void) +{ + return 0; +} + +static void __exit aoa_exit(void) +{ + aoa_alsa_cleanup(); +} + +module_init(aoa_init); +module_exit(aoa_exit); diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/snd-aoa-gpio-feature.c new file mode 100644 index 0000000000..2c6eb7784c --- /dev/null +++ b/sound/aoa/core/snd-aoa-gpio-feature.c @@ -0,0 +1,399 @@ +/* + * Apple Onboard Audio feature call GPIO control + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + * + * This file contains the GPIO control routines for + * direct (through feature calls) access to the GPIO + * registers. + */ + +#include +#include +#include "../aoa.h" + +/* TODO: these are 20 global variables + * that aren't used on most machines... + * Move them into a dynamically allocated + * structure and use that. + */ + +/* these are the GPIO numbers (register addresses as offsets into + * the GPIO space) */ +static int headphone_mute_gpio; +static int amp_mute_gpio; +static int lineout_mute_gpio; +static int hw_reset_gpio; +static int lineout_detect_gpio; +static int headphone_detect_gpio; +static int linein_detect_gpio; + +/* see the SWITCH_GPIO macro */ +static int headphone_mute_gpio_activestate; +static int amp_mute_gpio_activestate; +static int lineout_mute_gpio_activestate; +static int hw_reset_gpio_activestate; +static int lineout_detect_gpio_activestate; +static int headphone_detect_gpio_activestate; +static int linein_detect_gpio_activestate; + +/* node pointers that we save when getting the GPIO number + * to get the interrupt later */ +static struct device_node *lineout_detect_node; +static struct device_node *linein_detect_node; +static struct device_node *headphone_detect_node; + +static int lineout_detect_irq; +static int linein_detect_irq; +static int headphone_detect_irq; + +static struct device_node *get_gpio(char *name, + char *altname, + int *gpioptr, + int *gpioactiveptr) +{ + struct device_node *np, *gpio; + u32 *reg; + char *audio_gpio; + + *gpioptr = -1; + + /* check if we can get it the easy way ... */ + np = of_find_node_by_name(NULL, name); + if (!np) { + /* some machines have only gpioX/extint-gpioX nodes, + * and an audio-gpio property saying what it is ... + * So what we have to do is enumerate all children + * of the gpio node and check them all. */ + gpio = of_find_node_by_name(NULL, "gpio"); + if (!gpio) + return NULL; + while ((np = of_get_next_child(gpio, np))) { + audio_gpio = get_property(np, "audio-gpio", NULL); + if (!audio_gpio) + continue; + if (strcmp(audio_gpio, name) == 0) + break; + if (altname && (strcmp(audio_gpio, altname) == 0)) + break; + } + /* still not found, assume not there */ + if (!np) + return NULL; + } + + reg = (u32 *)get_property(np, "reg", NULL); + if (!reg) + return NULL; + + *gpioptr = *reg; + + /* this is a hack, usually the GPIOs 'reg' property + * should have the offset based from the GPIO space + * which is at 0x50, but apparently not always... */ + if (*gpioptr < 0x50) + *gpioptr += 0x50; + + reg = (u32 *)get_property(np, "audio-gpio-active-state", NULL); + if (!reg) + /* Apple seems to default to 1, but + * that doesn't seem right at least on most + * machines. So until proven that the opposite + * is necessary, we default to 0 + * (which, incidentally, snd-powermac also does...) */ + *gpioactiveptr = 0; + else + *gpioactiveptr = *reg; + + return np; +} + +static void get_irq(struct device_node * np, int *irqptr) +{ + *irqptr = -1; + if (!np) + return; + if (np->n_intrs != 1) + return; + *irqptr = np->intrs[0].line; +} + +/* 0x4 is outenable, 0x1 is out, thus 4 or 5 */ +#define SWITCH_GPIO(name, v, on) \ + (((v)&~1) | ((on)? \ + (name##_gpio_activestate==0?4:5): \ + (name##_gpio_activestate==0?5:4))) + +#define FTR_GPIO(name, bit) \ +static void ftr_gpio_set_##name(struct gpio_runtime *rt, int on)\ +{ \ + int v; \ + \ + if (unlikely(!rt)) return; \ + \ + if (name##_mute_gpio < 0) \ + return; \ + \ + v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, \ + name##_mute_gpio, \ + 0); \ + \ + /* muted = !on... */ \ + v = SWITCH_GPIO(name##_mute, v, !on); \ + \ + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, \ + name##_mute_gpio, v); \ + \ + rt->implementation_private &= ~(1<implementation_private |= (!!on << bit); \ +} \ +static int ftr_gpio_get_##name(struct gpio_runtime *rt) \ +{ \ + if (unlikely(!rt)) return 0; \ + return (rt->implementation_private>>bit)&1; \ +} + +FTR_GPIO(headphone, 0); +FTR_GPIO(amp, 1); +FTR_GPIO(lineout, 2); + +static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) +{ + int v; + + if (unlikely(!rt)) return; + if (hw_reset_gpio < 0) + return; + + v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, + hw_reset_gpio, 0); + v = SWITCH_GPIO(hw_reset, v, on); + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, + hw_reset_gpio, v); +} + +static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) +{ + int saved; + + if (unlikely(!rt)) return; + saved = rt->implementation_private; + ftr_gpio_set_headphone(rt, 0); + ftr_gpio_set_amp(rt, 0); + ftr_gpio_set_lineout(rt, 0); + rt->implementation_private = saved; +} + +static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt) +{ + int s; + + if (unlikely(!rt)) return; + s = rt->implementation_private; + ftr_gpio_set_headphone(rt, (s>>0)&1); + ftr_gpio_set_amp(rt, (s>>1)&1); + ftr_gpio_set_lineout(rt, (s>>2)&1); +} + +static void ftr_handle_notify(void *data) +{ + struct gpio_notification *notif = data; + + mutex_lock(¬if->mutex); + if (notif->notify) + notif->notify(notif->data); + mutex_unlock(¬if->mutex); +} + +static void ftr_gpio_init(struct gpio_runtime *rt) +{ + get_gpio("headphone-mute", NULL, + &headphone_mute_gpio, + &headphone_mute_gpio_activestate); + get_gpio("amp-mute", NULL, + &_mute_gpio, + &_mute_gpio_activestate); + get_gpio("lineout-mute", NULL, + &lineout_mute_gpio, + &lineout_mute_gpio_activestate); + get_gpio("hw-reset", "audio-hw-reset", + &hw_reset_gpio, + &hw_reset_gpio_activestate); + + headphone_detect_node = get_gpio("headphone-detect", NULL, + &headphone_detect_gpio, + &headphone_detect_gpio_activestate); + /* go Apple, and thanks for giving these different names + * across the board... */ + lineout_detect_node = get_gpio("lineout-detect", "line-output-detect", + &lineout_detect_gpio, + &lineout_detect_gpio_activestate); + linein_detect_node = get_gpio("linein-detect", "line-input-detect", + &linein_detect_gpio, + &linein_detect_gpio_activestate); + + get_irq(headphone_detect_node, &headphone_detect_irq); + get_irq(lineout_detect_node, &lineout_detect_irq); + get_irq(linein_detect_node, &linein_detect_irq); + + ftr_gpio_all_amps_off(rt); + rt->implementation_private = 0; + INIT_WORK(&rt->headphone_notify.work, ftr_handle_notify, + &rt->headphone_notify); + INIT_WORK(&rt->line_in_notify.work, ftr_handle_notify, + &rt->line_in_notify); + INIT_WORK(&rt->line_out_notify.work, ftr_handle_notify, + &rt->line_out_notify); + mutex_init(&rt->headphone_notify.mutex); + mutex_init(&rt->line_in_notify.mutex); + mutex_init(&rt->line_out_notify.mutex); +} + +static void ftr_gpio_exit(struct gpio_runtime *rt) +{ + ftr_gpio_all_amps_off(rt); + rt->implementation_private = 0; + if (rt->headphone_notify.notify) + free_irq(headphone_detect_irq, &rt->headphone_notify); + if (rt->line_in_notify.gpio_private) + free_irq(linein_detect_irq, &rt->line_in_notify); + if (rt->line_out_notify.gpio_private) + free_irq(lineout_detect_irq, &rt->line_out_notify); + cancel_delayed_work(&rt->headphone_notify.work); + cancel_delayed_work(&rt->line_in_notify.work); + cancel_delayed_work(&rt->line_out_notify.work); + flush_scheduled_work(); + mutex_destroy(&rt->headphone_notify.mutex); + mutex_destroy(&rt->line_in_notify.mutex); + mutex_destroy(&rt->line_out_notify.mutex); +} + +static irqreturn_t ftr_handle_notify_irq(int xx, + void *data, + struct pt_regs *regs) +{ + struct gpio_notification *notif = data; + + schedule_work(¬if->work); + + return IRQ_HANDLED; +} + +static int ftr_set_notify(struct gpio_runtime *rt, + enum notify_type type, + notify_func_t notify, + void *data) +{ + struct gpio_notification *notif; + notify_func_t old; + int irq; + char *name; + int err = -EBUSY; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + notif = &rt->headphone_notify; + name = "headphone-detect"; + irq = headphone_detect_irq; + break; + case AOA_NOTIFY_LINE_IN: + notif = &rt->line_in_notify; + name = "linein-detect"; + irq = linein_detect_irq; + break; + case AOA_NOTIFY_LINE_OUT: + notif = &rt->line_out_notify; + name = "lineout-detect"; + irq = lineout_detect_irq; + break; + default: + return -EINVAL; + } + + if (irq == -1) + return -ENODEV; + + mutex_lock(¬if->mutex); + + old = notif->notify; + + if (!old && !notify) { + err = 0; + goto out_unlock; + } + + if (old && notify) { + if (old == notify && notif->data == data) + err = 0; + goto out_unlock; + } + + if (old && !notify) + free_irq(irq, notif); + + if (!old && notify) { + err = request_irq(irq, ftr_handle_notify_irq, 0, name, notif); + if (err) + goto out_unlock; + } + + notif->notify = notify; + notif->data = data; + + err = 0; + out_unlock: + mutex_unlock(¬if->mutex); + return err; +} + +static int ftr_get_detect(struct gpio_runtime *rt, + enum notify_type type) +{ + int gpio, ret, active; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + gpio = headphone_detect_gpio; + active = headphone_detect_gpio_activestate; + break; + case AOA_NOTIFY_LINE_IN: + gpio = linein_detect_gpio; + active = linein_detect_gpio_activestate; + break; + case AOA_NOTIFY_LINE_OUT: + gpio = lineout_detect_gpio; + active = lineout_detect_gpio_activestate; + break; + default: + return -EINVAL; + } + + if (gpio == -1) + return -ENODEV; + + ret = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0); + if (ret < 0) + return ret; + return ((ret >> 1) & 1) == active; +} + +static struct gpio_methods methods = { + .init = ftr_gpio_init, + .exit = ftr_gpio_exit, + .all_amps_off = ftr_gpio_all_amps_off, + .all_amps_restore = ftr_gpio_all_amps_restore, + .set_headphone = ftr_gpio_set_headphone, + .set_speakers = ftr_gpio_set_amp, + .set_lineout = ftr_gpio_set_lineout, + .set_hw_reset = ftr_gpio_set_hw_reset, + .get_headphone = ftr_gpio_get_headphone, + .get_speakers = ftr_gpio_get_amp, + .get_lineout = ftr_gpio_get_lineout, + .set_notify = ftr_set_notify, + .get_detect = ftr_get_detect, +}; + +struct gpio_methods *ftr_gpio_methods = &methods; +EXPORT_SYMBOL_GPL(ftr_gpio_methods); diff --git a/sound/aoa/core/snd-aoa-gpio-pmf.c b/sound/aoa/core/snd-aoa-gpio-pmf.c new file mode 100644 index 0000000000..0e9b9bb2a6 --- /dev/null +++ b/sound/aoa/core/snd-aoa-gpio-pmf.c @@ -0,0 +1,246 @@ +/* + * Apple Onboard Audio pmf GPIOs + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#include +#include +#include "../aoa.h" + +#define PMF_GPIO(name, bit) \ +static void pmf_gpio_set_##name(struct gpio_runtime *rt, int on)\ +{ \ + struct pmf_args args = { .count = 1, .u[0].v = !on }; \ + \ + if (unlikely(!rt)) return; \ + pmf_call_function(rt->node, #name "-mute", &args); \ + rt->implementation_private &= ~(1<implementation_private |= (!!on << bit); \ +} \ +static int pmf_gpio_get_##name(struct gpio_runtime *rt) \ +{ \ + if (unlikely(!rt)) return 0; \ + return (rt->implementation_private>>bit)&1; \ +} + +PMF_GPIO(headphone, 0); +PMF_GPIO(amp, 1); +PMF_GPIO(lineout, 2); + +static void pmf_gpio_set_hw_reset(struct gpio_runtime *rt, int on) +{ + struct pmf_args args = { .count = 1, .u[0].v = !!on }; + + if (unlikely(!rt)) return; + pmf_call_function(rt->node, "hw-reset", &args); +} + +static void pmf_gpio_all_amps_off(struct gpio_runtime *rt) +{ + int saved; + + if (unlikely(!rt)) return; + saved = rt->implementation_private; + pmf_gpio_set_headphone(rt, 0); + pmf_gpio_set_amp(rt, 0); + pmf_gpio_set_lineout(rt, 0); + rt->implementation_private = saved; +} + +static void pmf_gpio_all_amps_restore(struct gpio_runtime *rt) +{ + int s; + + if (unlikely(!rt)) return; + s = rt->implementation_private; + pmf_gpio_set_headphone(rt, (s>>0)&1); + pmf_gpio_set_amp(rt, (s>>1)&1); + pmf_gpio_set_lineout(rt, (s>>2)&1); +} + +static void pmf_handle_notify(void *data) +{ + struct gpio_notification *notif = data; + + mutex_lock(¬if->mutex); + if (notif->notify) + notif->notify(notif->data); + mutex_unlock(¬if->mutex); +} + +static void pmf_gpio_init(struct gpio_runtime *rt) +{ + pmf_gpio_all_amps_off(rt); + rt->implementation_private = 0; + INIT_WORK(&rt->headphone_notify.work, pmf_handle_notify, + &rt->headphone_notify); + INIT_WORK(&rt->line_in_notify.work, pmf_handle_notify, + &rt->line_in_notify); + INIT_WORK(&rt->line_out_notify.work, pmf_handle_notify, + &rt->line_out_notify); + mutex_init(&rt->headphone_notify.mutex); + mutex_init(&rt->line_in_notify.mutex); + mutex_init(&rt->line_out_notify.mutex); +} + +static void pmf_gpio_exit(struct gpio_runtime *rt) +{ + pmf_gpio_all_amps_off(rt); + rt->implementation_private = 0; + + if (rt->headphone_notify.gpio_private) + pmf_unregister_irq_client(rt->headphone_notify.gpio_private); + if (rt->line_in_notify.gpio_private) + pmf_unregister_irq_client(rt->line_in_notify.gpio_private); + if (rt->line_out_notify.gpio_private) + pmf_unregister_irq_client(rt->line_out_notify.gpio_private); + + /* make sure no work is pending before freeing + * all things */ + cancel_delayed_work(&rt->headphone_notify.work); + cancel_delayed_work(&rt->line_in_notify.work); + cancel_delayed_work(&rt->line_out_notify.work); + flush_scheduled_work(); + + mutex_destroy(&rt->headphone_notify.mutex); + mutex_destroy(&rt->line_in_notify.mutex); + mutex_destroy(&rt->line_out_notify.mutex); + + if (rt->headphone_notify.gpio_private) + kfree(rt->headphone_notify.gpio_private); + if (rt->line_in_notify.gpio_private) + kfree(rt->line_in_notify.gpio_private); + if (rt->line_out_notify.gpio_private) + kfree(rt->line_out_notify.gpio_private); +} + +static void pmf_handle_notify_irq(void *data) +{ + struct gpio_notification *notif = data; + + schedule_work(¬if->work); +} + +static int pmf_set_notify(struct gpio_runtime *rt, + enum notify_type type, + notify_func_t notify, + void *data) +{ + struct gpio_notification *notif; + notify_func_t old; + struct pmf_irq_client *irq_client; + char *name; + int err = -EBUSY; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + notif = &rt->headphone_notify; + name = "headphone-detect"; + break; + case AOA_NOTIFY_LINE_IN: + notif = &rt->line_in_notify; + name = "linein-detect"; + break; + case AOA_NOTIFY_LINE_OUT: + notif = &rt->line_out_notify; + name = "lineout-detect"; + break; + default: + return -EINVAL; + } + + mutex_lock(¬if->mutex); + + old = notif->notify; + + if (!old && !notify) { + err = 0; + goto out_unlock; + } + + if (old && notify) { + if (old == notify && notif->data == data) + err = 0; + goto out_unlock; + } + + if (old && !notify) { + irq_client = notif->gpio_private; + pmf_unregister_irq_client(irq_client); + kfree(irq_client); + notif->gpio_private = NULL; + } + if (!old && notify) { + irq_client = kzalloc(sizeof(struct pmf_irq_client), + GFP_KERNEL); + irq_client->data = notif; + irq_client->handler = pmf_handle_notify_irq; + irq_client->owner = THIS_MODULE; + err = pmf_register_irq_client(rt->node, + name, + irq_client); + if (err) { + printk(KERN_ERR "snd-aoa: gpio layer failed to" + " register %s irq (%d)\n", name, err); + kfree(irq_client); + goto out_unlock; + } + notif->gpio_private = irq_client; + } + notif->notify = notify; + notif->data = data; + + err = 0; + out_unlock: + mutex_unlock(¬if->mutex); + return err; +} + +static int pmf_get_detect(struct gpio_runtime *rt, + enum notify_type type) +{ + char *name; + int err = -EBUSY, ret; + struct pmf_args args = { .count = 1, .u[0].p = &ret }; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + name = "headphone-detect"; + break; + case AOA_NOTIFY_LINE_IN: + name = "linein-detect"; + break; + case AOA_NOTIFY_LINE_OUT: + name = "lineout-detect"; + break; + default: + return -EINVAL; + } + + err = pmf_call_function(rt->node, name, &args); + if (err) + return err; + return ret; +} + +static struct gpio_methods methods = { + .init = pmf_gpio_init, + .exit = pmf_gpio_exit, + .all_amps_off = pmf_gpio_all_amps_off, + .all_amps_restore = pmf_gpio_all_amps_restore, + .set_headphone = pmf_gpio_set_headphone, + .set_speakers = pmf_gpio_set_amp, + .set_lineout = pmf_gpio_set_lineout, + .set_hw_reset = pmf_gpio_set_hw_reset, + .get_headphone = pmf_gpio_get_headphone, + .get_speakers = pmf_gpio_get_amp, + .get_lineout = pmf_gpio_get_lineout, + .set_notify = pmf_set_notify, + .get_detect = pmf_get_detect, +}; + +struct gpio_methods *pmf_gpio_methods = &methods; +EXPORT_SYMBOL_GPL(pmf_gpio_methods); diff --git a/sound/aoa/fabrics/Kconfig b/sound/aoa/fabrics/Kconfig new file mode 100644 index 0000000000..c3bc7705c8 --- /dev/null +++ b/sound/aoa/fabrics/Kconfig @@ -0,0 +1,12 @@ +config SND_AOA_FABRIC_LAYOUT + tristate "layout-id fabric" + depends SND_AOA + select SND_AOA_SOUNDBUS + select SND_AOA_SOUNDBUS_I2S + ---help--- + This enables the layout-id fabric for the Apple Onboard + Audio driver, the module holding it all together + based on the device-tree's layout-id property. + + If you are unsure and have a later Apple machine, + compile it as a module. diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile new file mode 100644 index 0000000000..55fc5e7e52 --- /dev/null +++ b/sound/aoa/fabrics/Makefile @@ -0,0 +1 @@ +obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c new file mode 100644 index 0000000000..04a7238e94 --- /dev/null +++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c @@ -0,0 +1,1109 @@ +/* + * Apple Onboard Audio driver -- layout fabric + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + * + * + * This fabric module looks for sound codecs + * based on the layout-id property in the device tree. + * + */ + +#include +#include +#include +#include "../aoa.h" +#include "../soundbus/soundbus.h" + +MODULE_AUTHOR("Johannes Berg "); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Layout-ID fabric for snd-aoa"); + +#define MAX_CODECS_PER_BUS 2 + +/* These are the connections the layout fabric + * knows about. It doesn't really care about the + * input ones, but I thought I'd separate them + * to give them proper names. The thing is that + * Apple usually will distinguish the active output + * by GPIOs, while the active input is set directly + * on the codec. Hence we here tell the codec what + * we think is connected. This information is hard- + * coded below ... */ +#define CC_SPEAKERS (1<<0) +#define CC_HEADPHONE (1<<1) +#define CC_LINEOUT (1<<2) +#define CC_DIGITALOUT (1<<3) +#define CC_LINEIN (1<<4) +#define CC_MICROPHONE (1<<5) +#define CC_DIGITALIN (1<<6) +/* pretty bogus but users complain... + * This is a flag saying that the LINEOUT + * should be renamed to HEADPHONE. + * be careful with input detection! */ +#define CC_LINEOUT_LABELLED_HEADPHONE (1<<7) + +struct codec_connection { + /* CC_ flags from above */ + int connected; + /* codec dependent bit to be set in the aoa_codec.connected field. + * This intentionally doesn't have any generic flags because the + * fabric has to know the codec anyway and all codecs might have + * different connectors */ + int codec_bit; +}; + +struct codec_connect_info { + char *name; + struct codec_connection *connections; +}; + +#define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) + +struct layout { + unsigned int layout_id; + struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; + int flags; + + /* if busname is not assigned, we use 'Master' below, + * so that our layout table doesn't need to be filled + * too much. + * We only assign these two if we expect to find more + * than one soundbus, i.e. on those machines with + * multiple layout-ids */ + char *busname; + int pcmid; +}; + +MODULE_ALIAS("sound-layout-41"); +MODULE_ALIAS("sound-layout-45"); +MODULE_ALIAS("sound-layout-51"); +MODULE_ALIAS("sound-layout-58"); +MODULE_ALIAS("sound-layout-60"); +MODULE_ALIAS("sound-layout-61"); +MODULE_ALIAS("sound-layout-64"); +MODULE_ALIAS("sound-layout-65"); +MODULE_ALIAS("sound-layout-68"); +MODULE_ALIAS("sound-layout-69"); +MODULE_ALIAS("sound-layout-70"); +MODULE_ALIAS("sound-layout-72"); +MODULE_ALIAS("sound-layout-80"); +MODULE_ALIAS("sound-layout-82"); +MODULE_ALIAS("sound-layout-84"); +MODULE_ALIAS("sound-layout-86"); +MODULE_ALIAS("sound-layout-92"); + +/* onyx with all but microphone connected */ +static struct codec_connection onyx_connections_nomic[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* onyx on machines without headphone */ +static struct codec_connection onyx_connections_noheadphones[] = { + { + .connected = CC_SPEAKERS | CC_LINEOUT | + CC_LINEOUT_LABELLED_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + /* FIXME: are these correct? probably not for all the machines + * below ... If not this will need separating. */ + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* onyx on machines with real line-out */ +static struct codec_connection onyx_connections_reallineout[] = { + { + .connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines without line out */ +static struct codec_connection tas_connections_nolineout[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines with neither line out nor line in */ +static struct codec_connection tas_connections_noline[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines without microphone */ +static struct codec_connection tas_connections_nomic[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, + .codec_bit = 0, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines with everything connected */ +static struct codec_connection tas_connections_all[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, + .codec_bit = 0, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection toonie_connections[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE, + .codec_bit = 0, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection topaz_input[] = { + { + .connected = CC_DIGITALIN, + .codec_bit = 0, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection topaz_output[] = { + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection topaz_inout[] = { + { + .connected = CC_DIGITALIN, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct layout layouts[] = { + /* last PowerBooks (15" Oct 2005) */ + { .layout_id = 82, + .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerMac9,1 */ + { .layout_id = 60, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_reallineout, + }, + }, + /* PowerMac9,1 */ + { .layout_id = 61, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerBook5,7 */ + { .layout_id = 64, + .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + /* PowerBook5,7 */ + { .layout_id = 65, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerBook5,9 [17" Oct 2005] */ + { .layout_id = 84, + .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerMac8,1 */ + { .layout_id = 45, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* Quad PowerMac (analog in, analog/digital out) */ + { .layout_id = 68, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_nomic, + }, + }, + /* Quad PowerMac (digital in) */ + { .layout_id = 69, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + .busname = "digital in", .pcmid = 1 }, + /* Early 2005 PowerBook (PowerBook 5,6) */ + { .layout_id = 70, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerBook 5,4 */ + { .layout_id = 51, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerBook6,7 */ + { .layout_id = 80, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerBook6,8 */ + { .layout_id = 72, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerMac8,2 */ + { .layout_id = 86, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_nomic, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerBook6,7 */ + { .layout_id = 92, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerMac10,1 (Mac Mini) */ + { .layout_id = 58, + .codecs[0] = { + .name = "toonie", + .connections = toonie_connections, + }, + }, + /* unknown, untested, but this comes from Apple */ + { .layout_id = 41, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + { .layout_id = 36, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nomic, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_inout, + }, + }, + { .layout_id = 47, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 48, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 49, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_nomic, + }, + }, + { .layout_id = 50, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 56, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 57, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 62, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_output, + }, + }, + { .layout_id = 66, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 67, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 76, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nomic, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_inout, + }, + }, + { .layout_id = 90, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + { .layout_id = 94, + .codecs[0] = { + .name = "onyx", + /* but it has an external mic?? how to select? */ + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 96, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 98, + .codecs[0] = { + .name = "toonie", + .connections = toonie_connections, + }, + }, + { .layout_id = 100, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + .codecs[1] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + {} +}; + +static struct layout *find_layout_by_id(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->layout_id) { + if (l->layout_id == id) + return l; + l++; + } + return NULL; +} + +static void use_layout(struct layout *l) +{ + int i; + + for (i=0; icodecs[i].name) { + request_module("snd-aoa-codec-%s", l->codecs[i].name); + } + } + /* now we wait for the codecs to call us back */ +} + +struct layout_dev; + +struct layout_dev_ptr { + struct layout_dev *ptr; +}; + +struct layout_dev { + struct list_head list; + struct soundbus_dev *sdev; + struct device_node *sound; + struct aoa_codec *codecs[MAX_CODECS_PER_BUS]; + struct layout *layout; + struct gpio_runtime gpio; + + /* we need these for headphone/lineout detection */ + struct snd_kcontrol *headphone_ctrl; + struct snd_kcontrol *lineout_ctrl; + struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *headphone_detected_ctrl; + struct snd_kcontrol *lineout_detected_ctrl; + + struct layout_dev_ptr selfptr_headphone; + struct layout_dev_ptr selfptr_lineout; + + u32 have_lineout_detect:1, + have_headphone_detect:1, + switch_on_headphone:1, + switch_on_lineout:1; +}; + +static LIST_HEAD(layouts_list); +static int layouts_list_items; +/* this can go away but only if we allow multiple cards, + * make the fabric handle all the card stuff, etc... */ +static struct layout_dev *layout_device; + +static int control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +#define AMP_CONTROL(n, description) \ +static int n##_control_get(struct snd_kcontrol *kcontrol, \ + struct snd_ctl_elem_value *ucontrol) \ +{ \ + struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \ + if (gpio->methods && gpio->methods->get_##n) \ + ucontrol->value.integer.value[0] = \ + gpio->methods->get_##n(gpio); \ + return 0; \ +} \ +static int n##_control_put(struct snd_kcontrol *kcontrol, \ + struct snd_ctl_elem_value *ucontrol) \ +{ \ + struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \ + if (gpio->methods && gpio->methods->get_##n) \ + gpio->methods->set_##n(gpio, \ + ucontrol->value.integer.value[0]); \ + return 1; \ +} \ +static struct snd_kcontrol_new n##_ctl = { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = description, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = control_info, \ + .get = n##_control_get, \ + .put = n##_control_put, \ +} + +AMP_CONTROL(headphone, "Headphone Switch"); +AMP_CONTROL(speakers, "Speakers Switch"); +AMP_CONTROL(lineout, "Line-Out Switch"); + +static int detect_choice_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct layout_dev *ldev = snd_kcontrol_chip(kcontrol); + + switch (kcontrol->private_value) { + case 0: + ucontrol->value.integer.value[0] = ldev->switch_on_headphone; + break; + case 1: + ucontrol->value.integer.value[0] = ldev->switch_on_lineout; + break; + default: + return -ENODEV; + } + return 0; +} + +static int detect_choice_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct layout_dev *ldev = snd_kcontrol_chip(kcontrol); + + switch (kcontrol->private_value) { + case 0: + ldev->switch_on_headphone = !!ucontrol->value.integer.value[0]; + break; + case 1: + ldev->switch_on_lineout = !!ucontrol->value.integer.value[0]; + break; + default: + return -ENODEV; + } + return 1; +} + +static struct snd_kcontrol_new headphone_detect_choice = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Detect Autoswitch", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .get = detect_choice_get, + .put = detect_choice_put, + .private_value = 0, +}; + +static struct snd_kcontrol_new lineout_detect_choice = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line-Out Detect Autoswitch", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .get = detect_choice_get, + .put = detect_choice_put, + .private_value = 1, +}; + +static int detected_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct layout_dev *ldev = snd_kcontrol_chip(kcontrol); + int v; + + switch (kcontrol->private_value) { + case 0: + v = ldev->gpio.methods->get_detect(&ldev->gpio, + AOA_NOTIFY_HEADPHONE); + break; + case 1: + v = ldev->gpio.methods->get_detect(&ldev->gpio, + AOA_NOTIFY_LINE_OUT); + break; + default: + return -ENODEV; + } + ucontrol->value.integer.value[0] = v; + return 0; +} + +static struct snd_kcontrol_new headphone_detected = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Detected", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .get = detected_get, + .private_value = 0, +}; + +static struct snd_kcontrol_new lineout_detected = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line-Out Detected", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .get = detected_get, + .private_value = 1, +}; + +static int check_codec(struct aoa_codec *codec, + struct layout_dev *ldev, + struct codec_connect_info *cci) +{ + u32 *ref; + char propname[32]; + struct codec_connection *cc; + + /* if the codec has a 'codec' node, we require a reference */ + if (codec->node && (strcmp(codec->node->name, "codec") == 0)) { + snprintf(propname, sizeof(propname), + "platform-%s-codec-ref", codec->name); + ref = (u32*)get_property(ldev->sound, propname, NULL); + if (!ref) { + printk(KERN_INFO "snd-aoa-fabric-layout: " + "required property %s not present\n", propname); + return -ENODEV; + } + if (*ref != codec->node->linux_phandle) { + printk(KERN_INFO "snd-aoa-fabric-layout: " + "%s doesn't match!\n", propname); + return -ENODEV; + } + } else { + if (layouts_list_items != 1) { + printk(KERN_INFO "snd-aoa-fabric-layout: " + "more than one soundbus, but no references.\n"); + return -ENODEV; + } + } + codec->soundbus_dev = ldev->sdev; + codec->gpio = &ldev->gpio; + + cc = cci->connections; + if (!cc) + return -EINVAL; + + printk(KERN_INFO "snd-aoa-fabric-layout: can use this codec\n"); + + codec->connected = 0; + codec->fabric_data = cc; + + while (cc->connected) { + codec->connected |= 1<codec_bit; + cc++; + } + + return 0; +} + +static int layout_found_codec(struct aoa_codec *codec) +{ + struct layout_dev *ldev; + int i; + + list_for_each_entry(ldev, &layouts_list, list) { + for (i=0; ilayout->codecs[i].name) + continue; + if (strcmp(ldev->layout->codecs[i].name, codec->name) == 0) { + if (check_codec(codec, + ldev, + &ldev->layout->codecs[i]) == 0) + return 0; + } + } + } + return -ENODEV; +} + +static void layout_remove_codec(struct aoa_codec *codec) +{ + int i; + /* here remove the codec from the layout dev's + * codec reference */ + + codec->soundbus_dev = NULL; + codec->gpio = NULL; + for (i=0; iptr; + if (data == &ldev->selfptr_headphone) { + v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_HEADPHONE); + detected = ldev->headphone_detected_ctrl; + update = ldev->switch_on_headphone; + if (update) { + ldev->gpio.methods->set_speakers(&ldev->gpio, !v); + ldev->gpio.methods->set_headphone(&ldev->gpio, v); + ldev->gpio.methods->set_lineout(&ldev->gpio, 0); + } + } else if (data == &ldev->selfptr_lineout) { + v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_LINE_OUT); + detected = ldev->lineout_detected_ctrl; + update = ldev->switch_on_lineout; + if (update) { + ldev->gpio.methods->set_speakers(&ldev->gpio, !v); + ldev->gpio.methods->set_headphone(&ldev->gpio, 0); + ldev->gpio.methods->set_lineout(&ldev->gpio, v); + } + } else + return; + + if (detected) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &detected->id); + if (update) { + c = ldev->headphone_ctrl; + if (c) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id); + c = ldev->speaker_ctrl; + if (c) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id); + c = ldev->lineout_ctrl; + if (c) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id); + } +} + +static void layout_attached_codec(struct aoa_codec *codec) +{ + struct codec_connection *cc; + struct snd_kcontrol *ctl; + int headphones, lineout; + struct layout_dev *ldev = layout_device; + + /* need to add this codec to our codec array! */ + + cc = codec->fabric_data; + + headphones = codec->gpio->methods->get_detect(codec->gpio, + AOA_NOTIFY_HEADPHONE); + lineout = codec->gpio->methods->get_detect(codec->gpio, + AOA_NOTIFY_LINE_OUT); + + while (cc->connected) { + if (cc->connected & CC_SPEAKERS) { + if (headphones <= 0 && lineout <= 0) + ldev->gpio.methods->set_speakers(codec->gpio, 1); + ctl = snd_ctl_new1(&speakers_ctl, codec->gpio); + ldev->speaker_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + if (cc->connected & CC_HEADPHONE) { + if (headphones == 1) + ldev->gpio.methods->set_headphone(codec->gpio, 1); + ctl = snd_ctl_new1(&headphone_ctl, codec->gpio); + ldev->headphone_ctrl = ctl; + aoa_snd_ctl_add(ctl); + ldev->have_headphone_detect = + !ldev->gpio.methods + ->set_notify(&ldev->gpio, + AOA_NOTIFY_HEADPHONE, + layout_notify, + &ldev->selfptr_headphone); + if (ldev->have_headphone_detect) { + ctl = snd_ctl_new1(&headphone_detect_choice, + ldev); + aoa_snd_ctl_add(ctl); + ctl = snd_ctl_new1(&headphone_detected, + ldev); + ldev->headphone_detected_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + } + if (cc->connected & CC_LINEOUT) { + if (lineout == 1) + ldev->gpio.methods->set_lineout(codec->gpio, 1); + ctl = snd_ctl_new1(&lineout_ctl, codec->gpio); + if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) + strlcpy(ctl->id.name, + "Headphone Switch", sizeof(ctl->id.name)); + ldev->lineout_ctrl = ctl; + aoa_snd_ctl_add(ctl); + ldev->have_lineout_detect = + !ldev->gpio.methods + ->set_notify(&ldev->gpio, + AOA_NOTIFY_LINE_OUT, + layout_notify, + &ldev->selfptr_lineout); + if (ldev->have_lineout_detect) { + ctl = snd_ctl_new1(&lineout_detect_choice, + ldev); + if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) + strlcpy(ctl->id.name, + "Headphone Detect Autoswitch", + sizeof(ctl->id.name)); + aoa_snd_ctl_add(ctl); + ctl = snd_ctl_new1(&lineout_detected, + ldev); + if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) + strlcpy(ctl->id.name, + "Headphone Detected", + sizeof(ctl->id.name)); + ldev->lineout_detected_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + } + cc++; + } + /* now update initial state */ + if (ldev->have_headphone_detect) + layout_notify(&ldev->selfptr_headphone); + if (ldev->have_lineout_detect) + layout_notify(&ldev->selfptr_lineout); +} + +static struct aoa_fabric layout_fabric = { + .name = "SoundByLayout", + .owner = THIS_MODULE, + .found_codec = layout_found_codec, + .remove_codec = layout_remove_codec, + .attached_codec = layout_attached_codec, +}; + +static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) +{ + struct device_node *sound = NULL; + unsigned int *layout_id; + struct layout *layout; + struct layout_dev *ldev = NULL; + int err; + + /* hm, currently we can only have one ... */ + if (layout_device) + return -ENODEV; + + /* by breaking out we keep a reference */ + while ((sound = of_get_next_child(sdev->ofdev.node, sound))) { + if (sound->type && strcasecmp(sound->type, "soundchip") == 0) + break; + } + if (!sound) return -ENODEV; + + layout_id = (unsigned int *) get_property(sound, "layout-id", NULL); + if (!layout_id) + goto outnodev; + printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d ", *layout_id); + + layout = find_layout_by_id(*layout_id); + if (!layout) { + printk("(no idea how to handle)\n"); + goto outnodev; + } + + ldev = kzalloc(sizeof(struct layout_dev), GFP_KERNEL); + if (!ldev) + goto outnodev; + + layout_device = ldev; + ldev->sdev = sdev; + ldev->sound = sound; + ldev->layout = layout; + ldev->gpio.node = sound->parent; + switch (layout->layout_id) { + case 41: /* that unknown machine no one seems to have */ + case 51: /* PowerBook5,4 */ + case 58: /* Mac Mini */ + ldev->gpio.methods = ftr_gpio_methods; + break; + default: + ldev->gpio.methods = pmf_gpio_methods; + } + ldev->selfptr_headphone.ptr = ldev; + ldev->selfptr_lineout.ptr = ldev; + sdev->ofdev.dev.driver_data = ldev; + + printk("(using)\n"); + list_add(&ldev->list, &layouts_list); + layouts_list_items++; + + /* assign these before registering ourselves, so + * callbacks that are done during registration + * already have the values */ + sdev->pcmid = ldev->layout->pcmid; + if (ldev->layout->busname) { + sdev->pcmname = ldev->layout->busname; + } else { + sdev->pcmname = "Master"; + } + + ldev->gpio.methods->init(&ldev->gpio); + + err = aoa_fabric_register(&layout_fabric); + if (err && err != -EALREADY) { + printk(KERN_INFO "snd-aoa-fabric-layout: can't use," + " another fabric is active!\n"); + goto outlistdel; + } + + use_layout(layout); + ldev->switch_on_headphone = 1; + ldev->switch_on_lineout = 1; + return 0; + outlistdel: + /* we won't be using these then... */ + ldev->gpio.methods->exit(&ldev->gpio); + /* reset if we didn't use it */ + sdev->pcmname = NULL; + sdev->pcmid = -1; + list_del(&ldev->list); + layouts_list_items--; + outnodev: + if (sound) of_node_put(sound); + layout_device = NULL; + if (ldev) kfree(ldev); + return -ENODEV; +} + +static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) +{ + struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + int i; + + for (i=0; icodecs[i]) { + aoa_fabric_unlink_codec(ldev->codecs[i]); + } + ldev->codecs[i] = NULL; + } + list_del(&ldev->list); + layouts_list_items--; + of_node_put(ldev->sound); + + ldev->gpio.methods->set_notify(&ldev->gpio, + AOA_NOTIFY_HEADPHONE, + NULL, + NULL); + ldev->gpio.methods->set_notify(&ldev->gpio, + AOA_NOTIFY_LINE_OUT, + NULL, + NULL); + + ldev->gpio.methods->exit(&ldev->gpio); + layout_device = NULL; + kfree(ldev); + sdev->pcmid = -1; + sdev->pcmname = NULL; + return 0; +} + +#ifdef CONFIG_PM +static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state) +{ + struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + + printk("aoa_fabric_layout_suspend()\n"); + + if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) + ldev->gpio.methods->all_amps_off(&ldev->gpio); + + return 0; +} + +static int aoa_fabric_layout_resume(struct soundbus_dev *sdev) +{ + struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + + printk("aoa_fabric_layout_resume()\n"); + + if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) + ldev->gpio.methods->all_amps_restore(&ldev->gpio); + + return 0; +} +#endif + +static struct soundbus_driver aoa_soundbus_driver = { + .name = "snd_aoa_soundbus_drv", + .owner = THIS_MODULE, + .probe = aoa_fabric_layout_probe, + .remove = aoa_fabric_layout_remove, +#ifdef CONFIG_PM + .suspend = aoa_fabric_layout_suspend, + .resume = aoa_fabric_layout_resume, +#endif +}; + +static int __init aoa_fabric_layout_init(void) +{ + int err; + + err = soundbus_register_driver(&aoa_soundbus_driver); + if (err) + return err; + return 0; +} + +static void __exit aoa_fabric_layout_exit(void) +{ + soundbus_unregister_driver(&aoa_soundbus_driver); + aoa_fabric_unregister(&layout_fabric); +} + +module_init(aoa_fabric_layout_init); +module_exit(aoa_fabric_layout_exit); diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig new file mode 100644 index 0000000000..d532d27a9f --- /dev/null +++ b/sound/aoa/soundbus/Kconfig @@ -0,0 +1,14 @@ +config SND_AOA_SOUNDBUS + tristate "Apple Soundbus support" + depends on SOUND && SND_PCM && EXPERIMENTAL + ---help--- + This option enables the generic driver for the soundbus + support on Apple machines. + + It is required for the sound bus implementations. + +config SND_AOA_SOUNDBUS_I2S + tristate "I2S bus support" + depends on SND_AOA_SOUNDBUS && PCI + ---help--- + This option enables support for Apple I2S busses. diff --git a/sound/aoa/soundbus/Makefile b/sound/aoa/soundbus/Makefile new file mode 100644 index 0000000000..0e61f5aa06 --- /dev/null +++ b/sound/aoa/soundbus/Makefile @@ -0,0 +1,3 @@ +obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o +snd-aoa-soundbus-objs := core.o sysfs.o +obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/ diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c new file mode 100644 index 0000000000..abe84a76c8 --- /dev/null +++ b/sound/aoa/soundbus/core.c @@ -0,0 +1,250 @@ +/* + * soundbus + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#include +#include "soundbus.h" + +MODULE_AUTHOR("Johannes Berg "); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Apple Soundbus"); + +struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev) +{ + struct device *tmp; + + if (!dev) + return NULL; + tmp = get_device(&dev->ofdev.dev); + if (tmp) + return to_soundbus_device(tmp); + else + return NULL; +} +EXPORT_SYMBOL_GPL(soundbus_dev_get); + +void soundbus_dev_put(struct soundbus_dev *dev) +{ + if (dev) + put_device(&dev->ofdev.dev); +} +EXPORT_SYMBOL_GPL(soundbus_dev_put); + +static int soundbus_probe(struct device *dev) +{ + int error = -ENODEV; + struct soundbus_driver *drv; + struct soundbus_dev *soundbus_dev; + + drv = to_soundbus_driver(dev->driver); + soundbus_dev = to_soundbus_device(dev); + + if (!drv->probe) + return error; + + soundbus_dev_get(soundbus_dev); + + error = drv->probe(soundbus_dev); + if (error) + soundbus_dev_put(soundbus_dev); + + return error; +} + + +static int soundbus_uevent(struct device *dev, char **envp, int num_envp, + char *buffer, int buffer_size) +{ + struct soundbus_dev * soundbus_dev; + struct of_device * of; + char *scratch, *compat, *compat2; + int i = 0; + int length, cplen, cplen2, seen = 0; + + if (!dev) + return -ENODEV; + + soundbus_dev = to_soundbus_device(dev); + if (!soundbus_dev) + return -ENODEV; + + of = &soundbus_dev->ofdev; + + /* stuff we want to pass to /sbin/hotplug */ + envp[i++] = scratch = buffer; + length = scnprintf (scratch, buffer_size, "OF_NAME=%s", of->node->name); + ++length; + buffer_size -= length; + if ((buffer_size <= 0) || (i >= num_envp)) + return -ENOMEM; + scratch += length; + + envp[i++] = scratch; + length = scnprintf (scratch, buffer_size, "OF_TYPE=%s", of->node->type); + ++length; + buffer_size -= length; + if ((buffer_size <= 0) || (i >= num_envp)) + return -ENOMEM; + scratch += length; + + /* Since the compatible field can contain pretty much anything + * it's not really legal to split it out with commas. We split it + * up using a number of environment variables instead. */ + + compat = (char *) get_property(of->node, "compatible", &cplen); + compat2 = compat; + cplen2= cplen; + while (compat && cplen > 0) { + envp[i++] = scratch; + length = scnprintf (scratch, buffer_size, + "OF_COMPATIBLE_%d=%s", seen, compat); + ++length; + buffer_size -= length; + if ((buffer_size <= 0) || (i >= num_envp)) + return -ENOMEM; + scratch += length; + length = strlen (compat) + 1; + compat += length; + cplen -= length; + seen++; + } + + envp[i++] = scratch; + length = scnprintf (scratch, buffer_size, "OF_COMPATIBLE_N=%d", seen); + ++length; + buffer_size -= length; + if ((buffer_size <= 0) || (i >= num_envp)) + return -ENOMEM; + scratch += length; + + envp[i++] = scratch; + length = scnprintf (scratch, buffer_size, "MODALIAS=%s", + soundbus_dev->modalias); + + buffer_size -= length; + if ((buffer_size <= 0) || (i >= num_envp)) + return -ENOMEM; + + envp[i] = NULL; + + return 0; +} + +static int soundbus_device_remove(struct device *dev) +{ + struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); + struct soundbus_driver * drv = to_soundbus_driver(dev->driver); + + if (dev->driver && drv->remove) + drv->remove(soundbus_dev); + soundbus_dev_put(soundbus_dev); + + return 0; +} + +static void soundbus_device_shutdown(struct device *dev) +{ + struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); + struct soundbus_driver * drv = to_soundbus_driver(dev->driver); + + if (dev->driver && drv->shutdown) + drv->shutdown(soundbus_dev); +} + +#ifdef CONFIG_PM + +static int soundbus_device_suspend(struct device *dev, pm_message_t state) +{ + struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); + struct soundbus_driver * drv = to_soundbus_driver(dev->driver); + + if (dev->driver && drv->suspend) + return drv->suspend(soundbus_dev, state); + return 0; +} + +static int soundbus_device_resume(struct device * dev) +{ + struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); + struct soundbus_driver * drv = to_soundbus_driver(dev->driver); + + if (dev->driver && drv->resume) + return drv->resume(soundbus_dev); + return 0; +} + +#endif /* CONFIG_PM */ + +extern struct device_attribute soundbus_dev_attrs[]; + +static struct bus_type soundbus_bus_type = { + .name = "aoa-soundbus", + .probe = soundbus_probe, + .uevent = soundbus_uevent, + .remove = soundbus_device_remove, + .shutdown = soundbus_device_shutdown, +#ifdef CONFIG_PM + .suspend = soundbus_device_suspend, + .resume = soundbus_device_resume, +#endif + .dev_attrs = soundbus_dev_attrs, +}; + +static int __init soundbus_init(void) +{ + return bus_register(&soundbus_bus_type); +} + +static void __exit soundbus_exit(void) +{ + bus_unregister(&soundbus_bus_type); +} + +int soundbus_add_one(struct soundbus_dev *dev) +{ + static int devcount; + + /* sanity checks */ + if (!dev->attach_codec || + !dev->ofdev.node || + dev->pcmname || + dev->pcmid != -1) { + printk(KERN_ERR "soundbus: adding device failed sanity check!\n"); + return -EINVAL; + } + + snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount); + dev->ofdev.dev.bus = &soundbus_bus_type; + return of_device_register(&dev->ofdev); +} +EXPORT_SYMBOL_GPL(soundbus_add_one); + +void soundbus_remove_one(struct soundbus_dev *dev) +{ + of_device_unregister(&dev->ofdev); +} +EXPORT_SYMBOL_GPL(soundbus_remove_one); + +int soundbus_register_driver(struct soundbus_driver *drv) +{ + /* initialize common driver fields */ + drv->driver.name = drv->name; + drv->driver.bus = &soundbus_bus_type; + + /* register with core */ + return driver_register(&drv->driver); +} +EXPORT_SYMBOL_GPL(soundbus_register_driver); + +void soundbus_unregister_driver(struct soundbus_driver *drv) +{ + driver_unregister(&drv->driver); +} +EXPORT_SYMBOL_GPL(soundbus_unregister_driver); + +module_init(soundbus_init); +module_exit(soundbus_exit); diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile new file mode 100644 index 0000000000..e57a5cf656 --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/Makefile @@ -0,0 +1,2 @@ +obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o +snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-control.c b/sound/aoa/soundbus/i2sbus/i2sbus-control.c new file mode 100644 index 0000000000..f50407952d --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/i2sbus-control.c @@ -0,0 +1,192 @@ +/* + * i2sbus driver -- bus control routines + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#include +#include +#include +#include +#include +#include +#include "i2sbus.h" + +int i2sbus_control_init(struct macio_dev* dev, struct i2sbus_control **c) +{ + *c = kzalloc(sizeof(struct i2sbus_control), GFP_KERNEL); + if (!*c) + return -ENOMEM; + + INIT_LIST_HEAD(&(*c)->list); + + if (of_address_to_resource(dev->ofdev.node, 0, &(*c)->rsrc)) + goto err; + /* we really should be using feature calls instead of mapping + * these registers. It's safe for now since no one else is + * touching them... */ + (*c)->controlregs = ioremap((*c)->rsrc.start, + sizeof(struct i2s_control_regs)); + if (!(*c)->controlregs) + goto err; + + return 0; + err: + kfree(*c); + *c = NULL; + return -ENODEV; +} + +void i2sbus_control_destroy(struct i2sbus_control *c) +{ + iounmap(c->controlregs); + kfree(c); +} + +/* this is serialised externally */ +int i2sbus_control_add_dev(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev) +{ + struct device_node *np; + + np = i2sdev->sound.ofdev.node; + i2sdev->enable = pmf_find_function(np, "enable"); + i2sdev->cell_enable = pmf_find_function(np, "cell-enable"); + i2sdev->clock_enable = pmf_find_function(np, "clock-enable"); + i2sdev->cell_disable = pmf_find_function(np, "cell-disable"); + i2sdev->clock_disable = pmf_find_function(np, "clock-disable"); + + /* if the bus number is not 0 or 1 we absolutely need to use + * the platform functions -- there's nothing in Darwin that + * would allow seeing a system behind what the FCRs are then, + * and I don't want to go parsing a bunch of platform functions + * by hand to try finding a system... */ + if (i2sdev->bus_number != 0 && i2sdev->bus_number != 1 && + (!i2sdev->enable || + !i2sdev->cell_enable || !i2sdev->clock_enable || + !i2sdev->cell_disable || !i2sdev->clock_disable)) { + pmf_put_function(i2sdev->enable); + pmf_put_function(i2sdev->cell_enable); + pmf_put_function(i2sdev->clock_enable); + pmf_put_function(i2sdev->cell_disable); + pmf_put_function(i2sdev->clock_disable); + return -ENODEV; + } + + list_add(&i2sdev->item, &c->list); + + return 0; +} + +void i2sbus_control_remove_dev(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev) +{ + /* this is serialised externally */ + list_del(&i2sdev->item); + if (list_empty(&c->list)) + i2sbus_control_destroy(c); +} + +int i2sbus_control_enable(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev) +{ + struct pmf_args args = { .count = 0 }; + int cc; + + if (i2sdev->enable) + return pmf_call_one(i2sdev->enable, &args); + + switch (i2sdev->bus_number) { + case 0: + cc = in_le32(&c->controlregs->cell_control); + out_le32(&c->controlregs->cell_control, cc | CTRL_CLOCK_INTF_0_ENABLE); + break; + case 1: + cc = in_le32(&c->controlregs->cell_control); + out_le32(&c->controlregs->cell_control, cc | CTRL_CLOCK_INTF_1_ENABLE); + break; + default: + return -ENODEV; + } + return 0; +} + +int i2sbus_control_cell(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev, + int enable) +{ + struct pmf_args args = { .count = 0 }; + int cc; + + switch (enable) { + case 0: + if (i2sdev->cell_disable) + return pmf_call_one(i2sdev->cell_disable, &args); + break; + case 1: + if (i2sdev->cell_enable) + return pmf_call_one(i2sdev->cell_enable, &args); + break; + default: + printk(KERN_ERR "i2sbus: INVALID CELL ENABLE VALUE\n"); + return -ENODEV; + } + switch (i2sdev->bus_number) { + case 0: + cc = in_le32(&c->controlregs->cell_control); + cc &= ~CTRL_CLOCK_CELL_0_ENABLE; + cc |= enable * CTRL_CLOCK_CELL_0_ENABLE; + out_le32(&c->controlregs->cell_control, cc); + break; + case 1: + cc = in_le32(&c->controlregs->cell_control); + cc &= ~CTRL_CLOCK_CELL_1_ENABLE; + cc |= enable * CTRL_CLOCK_CELL_1_ENABLE; + out_le32(&c->controlregs->cell_control, cc); + break; + default: + return -ENODEV; + } + return 0; +} + +int i2sbus_control_clock(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev, + int enable) +{ + struct pmf_args args = { .count = 0 }; + int cc; + + switch (enable) { + case 0: + if (i2sdev->clock_disable) + return pmf_call_one(i2sdev->clock_disable, &args); + break; + case 1: + if (i2sdev->clock_enable) + return pmf_call_one(i2sdev->clock_enable, &args); + break; + default: + printk(KERN_ERR "i2sbus: INVALID CLOCK ENABLE VALUE\n"); + return -ENODEV; + } + switch (i2sdev->bus_number) { + case 0: + cc = in_le32(&c->controlregs->cell_control); + cc &= ~CTRL_CLOCK_CLOCK_0_ENABLE; + cc |= enable * CTRL_CLOCK_CLOCK_0_ENABLE; + out_le32(&c->controlregs->cell_control, cc); + break; + case 1: + cc = in_le32(&c->controlregs->cell_control); + cc &= ~CTRL_CLOCK_CLOCK_1_ENABLE; + cc |= enable * CTRL_CLOCK_CLOCK_1_ENABLE; + out_le32(&c->controlregs->cell_control, cc); + break; + default: + return -ENODEV; + } + return 0; +} diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-control.h b/sound/aoa/soundbus/i2sbus/i2sbus-control.h new file mode 100644 index 0000000000..bb05550f73 --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/i2sbus-control.h @@ -0,0 +1,37 @@ +/* + * i2sbus driver -- bus register definitions + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#ifndef __I2SBUS_CONTROLREGS_H +#define __I2SBUS_CONTROLREGS_H + +/* i2s control registers, at least what we know about them */ + +#define __PAD(m,n) u8 __pad##m[n] +#define _PAD(line, n) __PAD(line, n) +#define PAD(n) _PAD(__LINE__, (n)) +struct i2s_control_regs { + PAD(0x38); + __le32 fcr0; /* 0x38 (unknown) */ + __le32 cell_control; /* 0x3c (fcr1) */ + __le32 fcr2; /* 0x40 (unknown) */ + __le32 fcr3; /* 0x44 (fcr3) */ + __le32 clock_control; /* 0x48 (unknown) */ + PAD(4); + /* total size: 0x50 bytes */ +} __attribute__((__packed__)); + +#define CTRL_CLOCK_CELL_0_ENABLE (1<<10) +#define CTRL_CLOCK_CLOCK_0_ENABLE (1<<12) +#define CTRL_CLOCK_SWRESET_0 (1<<11) +#define CTRL_CLOCK_INTF_0_ENABLE (1<<13) + +#define CTRL_CLOCK_CELL_1_ENABLE (1<<17) +#define CTRL_CLOCK_CLOCK_1_ENABLE (1<<18) +#define CTRL_CLOCK_SWRESET_1 (1<<19) +#define CTRL_CLOCK_INTF_1_ENABLE (1<<20) + +#endif /* __I2SBUS_CONTROLREGS_H */ diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c new file mode 100644 index 0000000000..f268dacdaa --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -0,0 +1,387 @@ +/* + * i2sbus driver + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../soundbus.h" +#include "i2sbus.h" + +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Johannes Berg "); +MODULE_DESCRIPTION("Apple Soundbus: I2S support"); +/* for auto-loading, declare that we handle this weird + * string that macio puts into the relevant device */ +MODULE_ALIAS("of:Ni2sTi2sC"); + +static struct of_device_id i2sbus_match[] = { + { .name = "i2s" }, + { } +}; + +static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, + struct dbdma_command_mem *r, + int numcmds) +{ + /* one more for rounding */ + r->size = (numcmds+1) * sizeof(struct dbdma_cmd); + /* We use the PCI APIs for now until the generic one gets fixed + * enough or until we get some macio-specific versions + */ + r->space = dma_alloc_coherent( + &macio_get_pci_dev(i2sdev->macio)->dev, + r->size, + &r->bus_addr, + GFP_KERNEL); + + if (!r->space) return -ENOMEM; + + memset(r->space, 0, r->size); + r->cmds = (void*)DBDMA_ALIGN(r->space); + r->bus_cmd_start = r->bus_addr + + (dma_addr_t)((char*)r->cmds - (char*)r->space); + + return 0; +} + +static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, + struct dbdma_command_mem *r) +{ + if (!r->space) return; + + dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, + r->size, r->space, r->bus_addr); +} + +static void i2sbus_release_dev(struct device *dev) +{ + struct i2sbus_dev *i2sdev; + int i; + + i2sdev = container_of(dev, struct i2sbus_dev, sound.ofdev.dev); + + if (i2sdev->intfregs) iounmap(i2sdev->intfregs); + if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma); + if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma); + for (i=0;i<3;i++) + if (i2sdev->allocated_resource[i]) + release_and_free_resource(i2sdev->allocated_resource[i]); + free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring); + free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring); + for (i=0;i<3;i++) + free_irq(i2sdev->interrupts[i], i2sdev); + i2sbus_control_remove_dev(i2sdev->control, i2sdev); + mutex_destroy(&i2sdev->lock); + kfree(i2sdev); +} + +static irqreturn_t i2sbus_bus_intr(int irq, void *devid, struct pt_regs *regs) +{ + struct i2sbus_dev *dev = devid; + u32 intreg; + + spin_lock(&dev->low_lock); + intreg = in_le32(&dev->intfregs->intr_ctl); + + /* acknowledge interrupt reasons */ + out_le32(&dev->intfregs->intr_ctl, intreg); + + spin_unlock(&dev->low_lock); + + return IRQ_HANDLED; +} + +static int force; +module_param(force, int, 0444); +MODULE_PARM_DESC(force, "Force loading i2sbus even when" + " no layout-id property is present"); + +/* FIXME: look at device node refcounting */ +static int i2sbus_add_dev(struct macio_dev *macio, + struct i2sbus_control *control, + struct device_node *np) +{ + struct i2sbus_dev *dev; + struct device_node *child = NULL, *sound = NULL; + int i; + static const char *rnames[] = { "i2sbus: %s (control)", + "i2sbus: %s (tx)", + "i2sbus: %s (rx)" }; + static irqreturn_t (*ints[])(int irq, void *devid, + struct pt_regs *regs) = { + i2sbus_bus_intr, + i2sbus_tx_intr, + i2sbus_rx_intr + }; + + if (strlen(np->name) != 5) + return 0; + if (strncmp(np->name, "i2s-", 4)) + return 0; + + if (np->n_intrs != 3) + return 0; + + dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); + if (!dev) + return 0; + + i = 0; + while ((child = of_get_next_child(np, child))) { + if (strcmp(child->name, "sound") == 0) { + i++; + sound = child; + } + } + if (i == 1) { + u32 *layout_id; + layout_id = (u32*) get_property(sound, "layout-id", NULL); + if (layout_id) { + snprintf(dev->sound.modalias, 32, + "sound-layout-%d", *layout_id); + force = 1; + } + } + /* for the time being, until we can handle non-layout-id + * things in some fabric, refuse to attach if there is no + * layout-id property or we haven't been forced to attach. + * When there are two i2s busses and only one has a layout-id, + * then this depends on the order, but that isn't important + * either as the second one in that case is just a modem. */ + if (!force) { + kfree(dev); + return -ENODEV; + } + + mutex_init(&dev->lock); + spin_lock_init(&dev->low_lock); + dev->sound.ofdev.node = np; + dev->sound.ofdev.dma_mask = macio->ofdev.dma_mask; + dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.dma_mask; + dev->sound.ofdev.dev.parent = &macio->ofdev.dev; + dev->sound.ofdev.dev.release = i2sbus_release_dev; + dev->sound.attach_codec = i2sbus_attach_codec; + dev->sound.detach_codec = i2sbus_detach_codec; + dev->sound.pcmid = -1; + dev->macio = macio; + dev->control = control; + dev->bus_number = np->name[4] - 'a'; + INIT_LIST_HEAD(&dev->sound.codec_list); + + for (i=0;i<3;i++) { + dev->interrupts[i] = -1; + snprintf(dev->rnames[i], sizeof(dev->rnames[i]), rnames[i], np->name); + } + for (i=0;i<3;i++) { + if (request_irq(np->intrs[i].line, ints[i], 0, dev->rnames[i], dev)) + goto err; + dev->interrupts[i] = np->intrs[i].line; + } + + for (i=0;i<3;i++) { + if (of_address_to_resource(np, i, &dev->resources[i])) + goto err; + /* if only we could use our resource dev->resources[i]... + * but request_resource doesn't know about parents and + * contained resources... */ + dev->allocated_resource[i] = + request_mem_region(dev->resources[i].start, + dev->resources[i].end - + dev->resources[i].start + 1, + dev->rnames[i]); + if (!dev->allocated_resource[i]) { + printk(KERN_ERR "i2sbus: failed to claim resource %d!\n", i); + goto err; + } + } + /* should do sanity checking here about length of them */ + dev->intfregs = ioremap(dev->resources[0].start, + dev->resources[0].end-dev->resources[0].start+1); + dev->out.dbdma = ioremap(dev->resources[1].start, + dev->resources[1].end-dev->resources[1].start+1); + dev->in.dbdma = ioremap(dev->resources[2].start, + dev->resources[2].end-dev->resources[2].start+1); + if (!dev->intfregs || !dev->out.dbdma || !dev->in.dbdma) + goto err; + + if (alloc_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring, + MAX_DBDMA_COMMANDS)) + goto err; + if (alloc_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring, + MAX_DBDMA_COMMANDS)) + goto err; + + if (i2sbus_control_add_dev(dev->control, dev)) { + printk(KERN_ERR "i2sbus: control layer didn't like bus\n"); + goto err; + } + + if (soundbus_add_one(&dev->sound)) { + printk(KERN_DEBUG "i2sbus: device registration error!\n"); + goto err; + } + + /* enable this cell */ + i2sbus_control_cell(dev->control, dev, 1); + i2sbus_control_enable(dev->control, dev); + i2sbus_control_clock(dev->control, dev, 1); + + return 1; + err: + for (i=0;i<3;i++) + if (dev->interrupts[i] != -1) + free_irq(dev->interrupts[i], dev); + free_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring); + free_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring); + if (dev->intfregs) iounmap(dev->intfregs); + if (dev->out.dbdma) iounmap(dev->out.dbdma); + if (dev->in.dbdma) iounmap(dev->in.dbdma); + for (i=0;i<3;i++) + if (dev->allocated_resource[i]) + release_and_free_resource(dev->allocated_resource[i]); + mutex_destroy(&dev->lock); + kfree(dev); + return 0; +} + +static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) +{ + struct device_node *np = NULL; + int got = 0, err; + struct i2sbus_control *control = NULL; + + err = i2sbus_control_init(dev, &control); + if (err) + return err; + if (!control) { + printk(KERN_ERR "i2sbus_control_init API breakage\n"); + return -ENODEV; + } + + while ((np = of_get_next_child(dev->ofdev.node, np))) { + if (device_is_compatible(np, "i2sbus") || + device_is_compatible(np, "i2s-modem")) { + got += i2sbus_add_dev(dev, control, np); + } + } + + if (!got) { + /* found none, clean up */ + i2sbus_control_destroy(control); + return -ENODEV; + } + + dev->ofdev.dev.driver_data = control; + + return 0; +} + +static int i2sbus_remove(struct macio_dev* dev) +{ + struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_dev *i2sdev, *tmp; + + list_for_each_entry_safe(i2sdev, tmp, &control->list, item) + soundbus_remove_one(&i2sdev->sound); + + return 0; +} + +#ifdef CONFIG_PM +static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state) +{ + struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct codec_info_item *cii; + struct i2sbus_dev* i2sdev; + int err, ret = 0; + + list_for_each_entry(i2sdev, &control->list, item) { + /* Notify Alsa */ + if (i2sdev->sound.pcm) { + /* Suspend PCM streams */ + snd_pcm_suspend_all(i2sdev->sound.pcm); + /* Probably useless as we handle + * power transitions ourselves */ + snd_power_change_state(i2sdev->sound.pcm->card, + SNDRV_CTL_POWER_D3hot); + } + /* Notify codecs */ + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + err = 0; + if (cii->codec->suspend) + err = cii->codec->suspend(cii, state); + if (err) + ret = err; + } + } + return ret; +} + +static int i2sbus_resume(struct macio_dev* dev) +{ + struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct codec_info_item *cii; + struct i2sbus_dev* i2sdev; + int err, ret = 0; + + list_for_each_entry(i2sdev, &control->list, item) { + /* Notify codecs so they can re-initialize */ + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + err = 0; + if (cii->codec->resume) + err = cii->codec->resume(cii); + if (err) + ret = err; + } + /* Notify Alsa */ + if (i2sdev->sound.pcm) { + /* Same comment as above, probably useless */ + snd_power_change_state(i2sdev->sound.pcm->card, + SNDRV_CTL_POWER_D0); + } + } + + return ret; +} +#endif /* CONFIG_PM */ + +static int i2sbus_shutdown(struct macio_dev* dev) +{ + return 0; +} + +static struct macio_driver i2sbus_drv = { + .name = "soundbus-i2s", + .owner = THIS_MODULE, + .match_table = i2sbus_match, + .probe = i2sbus_probe, + .remove = i2sbus_remove, +#ifdef CONFIG_PM + .suspend = i2sbus_suspend, + .resume = i2sbus_resume, +#endif + .shutdown = i2sbus_shutdown, +}; + +static int __init soundbus_i2sbus_init(void) +{ + return macio_register_driver(&i2sbus_drv); +} + +static void __exit soundbus_i2sbus_exit(void) +{ + macio_unregister_driver(&i2sbus_drv); +} + +module_init(soundbus_i2sbus_init); +module_exit(soundbus_i2sbus_exit); diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h b/sound/aoa/soundbus/i2sbus/i2sbus-interface.h new file mode 100644 index 0000000000..c6b5f5452d --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/i2sbus-interface.h @@ -0,0 +1,187 @@ +/* + * i2sbus driver -- interface register definitions + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#ifndef __I2SBUS_INTERFACE_H +#define __I2SBUS_INTERFACE_H + +/* i2s bus control registers, at least what we know about them */ + +#define __PAD(m,n) u8 __pad##m[n] +#define _PAD(line, n) __PAD(line, n) +#define PAD(n) _PAD(__LINE__, (n)) +struct i2s_interface_regs { + __le32 intr_ctl; /* 0x00 */ + PAD(12); + __le32 serial_format; /* 0x10 */ + PAD(12); + __le32 codec_msg_out; /* 0x20 */ + PAD(12); + __le32 codec_msg_in; /* 0x30 */ + PAD(12); + __le32 frame_count; /* 0x40 */ + PAD(12); + __le32 frame_match; /* 0x50 */ + PAD(12); + __le32 data_word_sizes; /* 0x60 */ + PAD(12); + __le32 peak_level_sel; /* 0x70 */ + PAD(12); + __le32 peak_level_in0; /* 0x80 */ + PAD(12); + __le32 peak_level_in1; /* 0x90 */ + PAD(12); + /* total size: 0x100 bytes */ +} __attribute__((__packed__)); + +/* interrupt register is just a bitfield with + * interrupt enable and pending bits */ +#define I2S_REG_INTR_CTL 0x00 +# define I2S_INT_FRAME_COUNT (1<<31) +# define I2S_PENDING_FRAME_COUNT (1<<30) +# define I2S_INT_MESSAGE_FLAG (1<<29) +# define I2S_PENDING_MESSAGE_FLAG (1<<28) +# define I2S_INT_NEW_PEAK (1<<27) +# define I2S_PENDING_NEW_PEAK (1<<26) +# define I2S_INT_CLOCKS_STOPPED (1<<25) +# define I2S_PENDING_CLOCKS_STOPPED (1<<24) +# define I2S_INT_EXTERNAL_SYNC_ERROR (1<<23) +# define I2S_PENDING_EXTERNAL_SYNC_ERROR (1<<22) +# define I2S_INT_EXTERNAL_SYNC_OK (1<<21) +# define I2S_PENDING_EXTERNAL_SYNC_OK (1<<20) +# define I2S_INT_NEW_SAMPLE_RATE (1<<19) +# define I2S_PENDING_NEW_SAMPLE_RATE (1<<18) +# define I2S_INT_STATUS_FLAG (1<<17) +# define I2S_PENDING_STATUS_FLAG (1<<16) + +/* serial format register is more interesting :) + * It contains: + * - clock source + * - MClk divisor + * - SClk divisor + * - SClk master flag + * - serial format (sony, i2s 64x, i2s 32x, dav, silabs) + * - external sample frequency interrupt (don't understand) + * - external sample frequency + */ +#define I2S_REG_SERIAL_FORMAT 0x10 +/* clock source. You get either 18.432, 45.1584 or 49.1520 MHz */ +# define I2S_SF_CLOCK_SOURCE_SHIFT 30 +# define I2S_SF_CLOCK_SOURCE_MASK (3< + * + * GPL v2, can be found in COPYING. + */ + +#include +#include +/* So apparently there's a reason for requiring driver.h + * to be included first, even if I don't know it... */ +#include +#include +#include +#include +#include "../soundbus.h" +#include "i2sbus.h" + +static inline void get_pcm_info(struct i2sbus_dev *i2sdev, int in, + struct pcm_info **pi, struct pcm_info **other) +{ + if (in) { + if (pi) + *pi = &i2sdev->in; + if (other) + *other = &i2sdev->out; + } else { + if (pi) + *pi = &i2sdev->out; + if (other) + *other = &i2sdev->in; + } +} + +static int clock_and_divisors(int mclk, int sclk, int rate, int *out) +{ + /* sclk must be derived from mclk! */ + if (mclk % sclk) + return -1; + /* derive sclk register value */ + if (i2s_sf_sclkdiv(mclk / sclk, out)) + return -1; + + if (I2S_CLOCK_SPEED_18MHz % (rate * mclk) == 0) { + if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_18MHz / (rate * mclk), out)) { + *out |= I2S_SF_CLOCK_SOURCE_18MHz; + return 0; + } + } + if (I2S_CLOCK_SPEED_45MHz % (rate * mclk) == 0) { + if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_45MHz / (rate * mclk), out)) { + *out |= I2S_SF_CLOCK_SOURCE_45MHz; + return 0; + } + } + if (I2S_CLOCK_SPEED_49MHz % (rate * mclk) == 0) { + if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_49MHz / (rate * mclk), out)) { + *out |= I2S_SF_CLOCK_SOURCE_49MHz; + return 0; + } + } + return -1; +} + +#define CHECK_RATE(rate) \ + do { if (rates & SNDRV_PCM_RATE_ ##rate) { \ + int dummy; \ + if (clock_and_divisors(sysclock_factor, \ + bus_factor, rate, &dummy)) \ + rates &= ~SNDRV_PCM_RATE_ ##rate; \ + } } while (0) + +static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in) +{ + struct pcm_info *pi, *other; + struct soundbus_dev *sdev; + int masks_inited = 0, err; + struct codec_info_item *cii, *rev; + struct snd_pcm_hardware *hw; + u64 formats = 0; + unsigned int rates = 0; + struct transfer_info v; + int result = 0; + int bus_factor = 0, sysclock_factor = 0; + int found_this; + + mutex_lock(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, &other); + + hw = &pi->substream->runtime->hw; + sdev = &i2sdev->sound; + + if (pi->active) { + /* alsa messed up */ + result = -EBUSY; + goto out_unlock; + } + + /* we now need to assign the hw */ + list_for_each_entry(cii, &sdev->codec_list, list) { + struct transfer_info *ti = cii->codec->transfers; + bus_factor = cii->codec->bus_factor; + sysclock_factor = cii->codec->sysclock_factor; + while (ti->formats && ti->rates) { + v = *ti; + if (ti->transfer_in == in + && cii->codec->usable(cii, ti, &v)) { + if (masks_inited) { + formats &= v.formats; + rates &= v.rates; + } else { + formats = v.formats; + rates = v.rates; + masks_inited = 1; + } + } + ti++; + } + } + if (!masks_inited || !bus_factor || !sysclock_factor) { + result = -ENODEV; + goto out_unlock; + } + /* bus dependent stuff */ + hw->info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME; + + CHECK_RATE(5512); + CHECK_RATE(8000); + CHECK_RATE(11025); + CHECK_RATE(16000); + CHECK_RATE(22050); + CHECK_RATE(32000); + CHECK_RATE(44100); + CHECK_RATE(48000); + CHECK_RATE(64000); + CHECK_RATE(88200); + CHECK_RATE(96000); + CHECK_RATE(176400); + CHECK_RATE(192000); + hw->rates = rates; + + /* well. the codec might want 24 bits only, and we'll + * ever only transfer 24 bits, but they are top-aligned! + * So for alsa, we claim that we're doing full 32 bit + * while in reality we'll ignore the lower 8 bits of + * that when doing playback (they're transferred as 0 + * as far as I know, no codecs we have are 32-bit capable + * so I can't really test) and when doing recording we'll + * always have those lower 8 bits recorded as 0 */ + if (formats & SNDRV_PCM_FMTBIT_S24_BE) + formats |= SNDRV_PCM_FMTBIT_S32_BE; + if (formats & SNDRV_PCM_FMTBIT_U24_BE) + formats |= SNDRV_PCM_FMTBIT_U32_BE; + /* now mask off what we can support. I suppose we could + * also support S24_3LE and some similar formats, but I + * doubt there's a codec that would be able to use that, + * so we don't support it here. */ + hw->formats = formats & (SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE | + SNDRV_PCM_FMTBIT_S32_BE | + SNDRV_PCM_FMTBIT_U32_BE); + + /* we need to set the highest and lowest rate possible. + * These are the highest and lowest rates alsa can + * support properly in its bitfield. + * Below, we'll use that to restrict to the rate + * currently in use (if any). */ + hw->rate_min = 5512; + hw->rate_max = 192000; + /* if the other stream is active, then we can only + * support what it is currently using. + * FIXME: I lied. This comment is wrong. We can support + * anything that works with the same serial format, ie. + * when recording 24 bit sound we can well play 16 bit + * sound at the same time iff using the same transfer mode. + */ + if (other->active) { + /* FIXME: is this guaranteed by the alsa api? */ + hw->formats &= (1ULL << i2sdev->format); + /* see above, restrict rates to the one we already have */ + hw->rate_min = i2sdev->rate; + hw->rate_max = i2sdev->rate; + } + + hw->channels_min = 2; + hw->channels_max = 2; + /* these are somewhat arbitrary */ + hw->buffer_bytes_max = 131072; + hw->period_bytes_min = 256; + hw->period_bytes_max = 16384; + hw->periods_min = 3; + hw->periods_max = MAX_DBDMA_COMMANDS; + list_for_each_entry(cii, &sdev->codec_list, list) { + if (cii->codec->open) { + err = cii->codec->open(cii, pi->substream); + if (err) { + result = err; + /* unwind */ + found_this = 0; + list_for_each_entry_reverse(rev, + &sdev->codec_list, list) { + if (found_this && rev->codec->close) { + rev->codec->close(rev, + pi->substream); + } + if (rev == cii) + found_this = 1; + } + goto out_unlock; + } + } + } + + out_unlock: + mutex_unlock(&i2sdev->lock); + return result; +} + +#undef CHECK_RATE + +static int i2sbus_pcm_close(struct i2sbus_dev *i2sdev, int in) +{ + struct codec_info_item *cii; + struct pcm_info *pi; + int err = 0, tmp; + + mutex_lock(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, NULL); + + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + if (cii->codec->close) { + tmp = cii->codec->close(cii, pi->substream); + if (tmp) + err = tmp; + } + } + + pi->substream = NULL; + pi->active = 0; + mutex_unlock(&i2sdev->lock); + return err; +} + +static int i2sbus_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); +} + +static int i2sbus_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) +{ + /* whee. Hard work now. The user has selected a bitrate + * and bit format, so now we have to program our + * I2S controller appropriately. */ + struct snd_pcm_runtime *runtime; + struct dbdma_cmd *command; + int i, periodsize; + dma_addr_t offset; + struct bus_info bi; + struct codec_info_item *cii; + int sfr = 0; /* serial format register */ + int dws = 0; /* data word sizes reg */ + int input_16bit; + struct pcm_info *pi, *other; + int cnt; + int result = 0; + + mutex_lock(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, &other); + + if (pi->dbdma_ring.running) { + result = -EBUSY; + goto out_unlock; + } + + runtime = pi->substream->runtime; + pi->active = 1; + if (other->active && + ((i2sdev->format != runtime->format) + || (i2sdev->rate != runtime->rate))) { + result = -EINVAL; + goto out_unlock; + } + + i2sdev->format = runtime->format; + i2sdev->rate = runtime->rate; + + periodsize = snd_pcm_lib_period_bytes(pi->substream); + pi->current_period = 0; + + /* generate dbdma command ring first */ + command = pi->dbdma_ring.cmds; + offset = runtime->dma_addr; + for (i = 0; i < pi->substream->runtime->periods; + i++, command++, offset += periodsize) { + memset(command, 0, sizeof(struct dbdma_cmd)); + command->command = + cpu_to_le16((in ? INPUT_MORE : OUTPUT_MORE) | INTR_ALWAYS); + command->phy_addr = cpu_to_le32(offset); + command->req_count = cpu_to_le16(periodsize); + command->xfer_status = cpu_to_le16(0); + } + /* last one branches back to first */ + command--; + command->command |= cpu_to_le16(BR_ALWAYS); + command->cmd_dep = cpu_to_le32(pi->dbdma_ring.bus_cmd_start); + + /* ok, let's set the serial format and stuff */ + switch (runtime->format) { + /* 16 bit formats */ + case SNDRV_PCM_FORMAT_S16_BE: + case SNDRV_PCM_FORMAT_U16_BE: + /* FIXME: if we add different bus factors we need to + * do more here!! */ + bi.bus_factor = 0; + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + bi.bus_factor = cii->codec->bus_factor; + break; + } + if (!bi.bus_factor) { + result = -ENODEV; + goto out_unlock; + } + input_16bit = 1; + break; + case SNDRV_PCM_FORMAT_S32_BE: + case SNDRV_PCM_FORMAT_U32_BE: + /* force 64x bus speed, otherwise the data cannot be + * transferred quickly enough! */ + bi.bus_factor = 64; + input_16bit = 0; + break; + default: + result = -EINVAL; + goto out_unlock; + } + /* we assume all sysclocks are the same! */ + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + bi.sysclock_factor = cii->codec->sysclock_factor; + break; + } + + if (clock_and_divisors(bi.sysclock_factor, + bi.bus_factor, + runtime->rate, + &sfr) < 0) { + result = -EINVAL; + goto out_unlock; + } + switch (bi.bus_factor) { + case 32: + sfr |= I2S_SF_SERIAL_FORMAT_I2S_32X; + break; + case 64: + sfr |= I2S_SF_SERIAL_FORMAT_I2S_64X; + break; + } + /* FIXME: THIS ASSUMES MASTER ALL THE TIME */ + sfr |= I2S_SF_SCLK_MASTER; + + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + int err = 0; + if (cii->codec->prepare) + err = cii->codec->prepare(cii, &bi, pi->substream); + if (err) { + result = err; + goto out_unlock; + } + } + /* codecs are fine with it, so set our clocks */ + if (input_16bit) + dws = (2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) | + (2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) | + I2S_DWS_DATA_IN_16BIT | I2S_DWS_DATA_OUT_16BIT; + else + dws = (2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) | + (2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) | + I2S_DWS_DATA_IN_24BIT | I2S_DWS_DATA_OUT_24BIT; + + /* early exit if already programmed correctly */ + /* not locking these is fine since we touch them only in this function */ + if (in_le32(&i2sdev->intfregs->serial_format) == sfr + && in_le32(&i2sdev->intfregs->data_word_sizes) == dws) + goto out_unlock; + + /* let's notify the codecs about clocks going away. + * For now we only do mastering on the i2s cell... */ + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) + if (cii->codec->switch_clock) + cii->codec->switch_clock(cii, CLOCK_SWITCH_PREPARE_SLAVE); + + i2sbus_control_enable(i2sdev->control, i2sdev); + i2sbus_control_cell(i2sdev->control, i2sdev, 1); + + out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED); + + i2sbus_control_clock(i2sdev->control, i2sdev, 0); + + msleep(1); + + /* wait for clock stopped. This can apparently take a while... */ + cnt = 100; + while (cnt-- && + !(in_le32(&i2sdev->intfregs->intr_ctl) & I2S_PENDING_CLOCKS_STOPPED)) { + msleep(5); + } + out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED); + + /* not locking these is fine since we touch them only in this function */ + out_le32(&i2sdev->intfregs->serial_format, sfr); + out_le32(&i2sdev->intfregs->data_word_sizes, dws); + + i2sbus_control_enable(i2sdev->control, i2sdev); + i2sbus_control_cell(i2sdev->control, i2sdev, 1); + i2sbus_control_clock(i2sdev->control, i2sdev, 1); + msleep(1); + + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) + if (cii->codec->switch_clock) + cii->codec->switch_clock(cii, CLOCK_SWITCH_SLAVE); + + out_unlock: + mutex_unlock(&i2sdev->lock); + return result; +} + +static struct dbdma_cmd STOP_CMD = { + .command = __constant_cpu_to_le16(DBDMA_STOP), +}; + +static int i2sbus_pcm_trigger(struct i2sbus_dev *i2sdev, int in, int cmd) +{ + struct codec_info_item *cii; + struct pcm_info *pi; + int timeout; + struct dbdma_cmd tmp; + int result = 0; + unsigned long flags; + + spin_lock_irqsave(&i2sdev->low_lock, flags); + + get_pcm_info(i2sdev, in, &pi, NULL); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + if (pi->dbdma_ring.running) { + result = -EALREADY; + goto out_unlock; + } + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) + if (cii->codec->start) + cii->codec->start(cii, pi->substream); + pi->dbdma_ring.running = 1; + + /* reset dma engine */ + out_le32(&pi->dbdma->control, + 0 | (RUN | PAUSE | FLUSH | WAKE) << 16); + timeout = 100; + while (in_le32(&pi->dbdma->status) & RUN && timeout--) + udelay(1); + if (timeout <= 0) { + printk(KERN_ERR + "i2sbus: error waiting for dma reset\n"); + result = -ENXIO; + goto out_unlock; + } + + /* write dma command buffer address to the dbdma chip */ + out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start); + /* post PCI write */ + mb(); + (void)in_le32(&pi->dbdma->status); + + /* change first command to STOP */ + tmp = *pi->dbdma_ring.cmds; + *pi->dbdma_ring.cmds = STOP_CMD; + + /* set running state, remember that the first command is STOP */ + out_le32(&pi->dbdma->control, RUN | (RUN << 16)); + timeout = 100; + /* wait for STOP to be executed */ + while (in_le32(&pi->dbdma->status) & ACTIVE && timeout--) + udelay(1); + if (timeout <= 0) { + printk(KERN_ERR "i2sbus: error waiting for dma stop\n"); + result = -ENXIO; + goto out_unlock; + } + /* again, write dma command buffer address to the dbdma chip, + * this time of the first real command */ + *pi->dbdma_ring.cmds = tmp; + out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start); + /* post write */ + mb(); + (void)in_le32(&pi->dbdma->status); + + /* reset dma engine again */ + out_le32(&pi->dbdma->control, + 0 | (RUN | PAUSE | FLUSH | WAKE) << 16); + timeout = 100; + while (in_le32(&pi->dbdma->status) & RUN && timeout--) + udelay(1); + if (timeout <= 0) { + printk(KERN_ERR + "i2sbus: error waiting for dma reset\n"); + result = -ENXIO; + goto out_unlock; + } + + /* wake up the chip with the next descriptor */ + out_le32(&pi->dbdma->control, + (RUN | WAKE) | ((RUN | WAKE) << 16)); + /* get the frame count */ + pi->frame_count = in_le32(&i2sdev->intfregs->frame_count); + + /* off you go! */ + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (!pi->dbdma_ring.running) { + result = -EALREADY; + goto out_unlock; + } + + /* turn off all relevant bits */ + out_le32(&pi->dbdma->control, + (RUN | WAKE | FLUSH | PAUSE) << 16); + { + /* FIXME: move to own function */ + int timeout = 5000; + while ((in_le32(&pi->dbdma->status) & RUN) + && --timeout > 0) + udelay(1); + if (!timeout) + printk(KERN_ERR + "i2sbus: timed out turning " + "off dbdma engine!\n"); + } + + pi->dbdma_ring.running = 0; + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) + if (cii->codec->stop) + cii->codec->stop(cii, pi->substream); + break; + default: + result = -EINVAL; + goto out_unlock; + } + + out_unlock: + spin_unlock_irqrestore(&i2sdev->low_lock, flags); + return result; +} + +static snd_pcm_uframes_t i2sbus_pcm_pointer(struct i2sbus_dev *i2sdev, int in) +{ + struct pcm_info *pi; + u32 fc; + + get_pcm_info(i2sdev, in, &pi, NULL); + + fc = in_le32(&i2sdev->intfregs->frame_count); + fc = fc - pi->frame_count; + + return (bytes_to_frames(pi->substream->runtime, + pi->current_period * + snd_pcm_lib_period_bytes(pi->substream)) + + fc) % pi->substream->runtime->buffer_size; +} + +static inline void handle_interrupt(struct i2sbus_dev *i2sdev, int in) +{ + struct pcm_info *pi; + u32 fc; + u32 delta; + + spin_lock(&i2sdev->low_lock); + get_pcm_info(i2sdev, in, &pi, NULL); + + if (!pi->dbdma_ring.running) { + /* there was still an interrupt pending + * while we stopped. or maybe another + * processor (not the one that was stopping + * the DMA engine) was spinning above + * waiting for the lock. */ + goto out_unlock; + } + + fc = in_le32(&i2sdev->intfregs->frame_count); + /* a counter overflow does not change the calculation. */ + delta = fc - pi->frame_count; + + /* update current_period */ + while (delta >= pi->substream->runtime->period_size) { + pi->current_period++; + delta = delta - pi->substream->runtime->period_size; + } + + if (unlikely(delta)) { + /* Some interrupt came late, so check the dbdma. + * This special case exists to syncronize the frame_count with + * the dbdma transfer, but is hit every once in a while. */ + int period; + + period = (in_le32(&pi->dbdma->cmdptr) + - pi->dbdma_ring.bus_cmd_start) + / sizeof(struct dbdma_cmd); + pi->current_period = pi->current_period + % pi->substream->runtime->periods; + + while (pi->current_period != period) { + pi->current_period++; + pi->current_period %= pi->substream->runtime->periods; + /* Set delta to zero, as the frame_count value is too + * high (otherwise the code path will not be executed). + * This corrects the fact that the frame_count is too + * low at the beginning due to buffering. */ + delta = 0; + } + } + + pi->frame_count = fc - delta; + pi->current_period %= pi->substream->runtime->periods; + + spin_unlock(&i2sdev->low_lock); + /* may call _trigger again, hence needs to be unlocked */ + snd_pcm_period_elapsed(pi->substream); + return; + out_unlock: + spin_unlock(&i2sdev->low_lock); +} + +irqreturn_t i2sbus_tx_intr(int irq, void *devid, struct pt_regs *regs) +{ + handle_interrupt((struct i2sbus_dev *)devid, 0); + return IRQ_HANDLED; +} + +irqreturn_t i2sbus_rx_intr(int irq, void *devid, struct pt_regs * regs) +{ + handle_interrupt((struct i2sbus_dev *)devid, 1); + return IRQ_HANDLED; +} + +static int i2sbus_playback_open(struct snd_pcm_substream *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + i2sdev->out.substream = substream; + return i2sbus_pcm_open(i2sdev, 0); +} + +static int i2sbus_playback_close(struct snd_pcm_substream *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + int err; + + if (!i2sdev) + return -EINVAL; + if (i2sdev->out.substream != substream) + return -EINVAL; + err = i2sbus_pcm_close(i2sdev, 0); + if (!err) + i2sdev->out.substream = NULL; + return err; +} + +static int i2sbus_playback_prepare(struct snd_pcm_substream *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + if (i2sdev->out.substream != substream) + return -EINVAL; + return i2sbus_pcm_prepare(i2sdev, 0); +} + +static int i2sbus_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + if (i2sdev->out.substream != substream) + return -EINVAL; + return i2sbus_pcm_trigger(i2sdev, 0, cmd); +} + +static snd_pcm_uframes_t i2sbus_playback_pointer(struct snd_pcm_substream + *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + if (i2sdev->out.substream != substream) + return 0; + return i2sbus_pcm_pointer(i2sdev, 0); +} + +static struct snd_pcm_ops i2sbus_playback_ops = { + .open = i2sbus_playback_open, + .close = i2sbus_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = i2sbus_hw_params, + .hw_free = i2sbus_hw_free, + .prepare = i2sbus_playback_prepare, + .trigger = i2sbus_playback_trigger, + .pointer = i2sbus_playback_pointer, +}; + +static int i2sbus_record_open(struct snd_pcm_substream *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + i2sdev->in.substream = substream; + return i2sbus_pcm_open(i2sdev, 1); +} + +static int i2sbus_record_close(struct snd_pcm_substream *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + int err; + + if (!i2sdev) + return -EINVAL; + if (i2sdev->in.substream != substream) + return -EINVAL; + err = i2sbus_pcm_close(i2sdev, 1); + if (!err) + i2sdev->in.substream = NULL; + return err; +} + +static int i2sbus_record_prepare(struct snd_pcm_substream *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + if (i2sdev->in.substream != substream) + return -EINVAL; + return i2sbus_pcm_prepare(i2sdev, 1); +} + +static int i2sbus_record_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + if (i2sdev->in.substream != substream) + return -EINVAL; + return i2sbus_pcm_trigger(i2sdev, 1, cmd); +} + +static snd_pcm_uframes_t i2sbus_record_pointer(struct snd_pcm_substream + *substream) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + + if (!i2sdev) + return -EINVAL; + if (i2sdev->in.substream != substream) + return 0; + return i2sbus_pcm_pointer(i2sdev, 1); +} + +static struct snd_pcm_ops i2sbus_record_ops = { + .open = i2sbus_record_open, + .close = i2sbus_record_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = i2sbus_hw_params, + .hw_free = i2sbus_hw_free, + .prepare = i2sbus_record_prepare, + .trigger = i2sbus_record_trigger, + .pointer = i2sbus_record_pointer, +}; + +static void i2sbus_private_free(struct snd_pcm *pcm) +{ + struct i2sbus_dev *i2sdev = snd_pcm_chip(pcm); + struct codec_info_item *p, *tmp; + + i2sdev->sound.pcm = NULL; + i2sdev->out.created = 0; + i2sdev->in.created = 0; + list_for_each_entry_safe(p, tmp, &i2sdev->sound.codec_list, list) { + printk(KERN_ERR "i2sbus: a codec didn't unregister!\n"); + list_del(&p->list); + module_put(p->codec->owner); + kfree(p); + } + soundbus_dev_put(&i2sdev->sound); + module_put(THIS_MODULE); +} + +/* FIXME: this function needs an error handling strategy with labels */ +int +i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, + struct codec_info *ci, void *data) +{ + int err, in = 0, out = 0; + struct transfer_info *tmp; + struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev); + struct codec_info_item *cii; + + if (!dev->pcmname || dev->pcmid == -1) { + printk(KERN_ERR "i2sbus: pcm name and id must be set!\n"); + return -EINVAL; + } + + list_for_each_entry(cii, &dev->codec_list, list) { + if (cii->codec_data == data) + return -EALREADY; + } + + if (!ci->transfers || !ci->transfers->formats + || !ci->transfers->rates || !ci->usable) + return -EINVAL; + + /* we currently code the i2s transfer on the clock, and support only + * 32 and 64 */ + if (ci->bus_factor != 32 && ci->bus_factor != 64) + return -EINVAL; + + /* If you want to fix this, you need to keep track of what transport infos + * are to be used, which codecs they belong to, and then fix all the + * sysclock/busclock stuff above to depend on which is usable */ + list_for_each_entry(cii, &dev->codec_list, list) { + if (cii->codec->sysclock_factor != ci->sysclock_factor) { + printk(KERN_DEBUG + "cannot yet handle multiple different sysclocks!\n"); + return -EINVAL; + } + if (cii->codec->bus_factor != ci->bus_factor) { + printk(KERN_DEBUG + "cannot yet handle multiple different bus clocks!\n"); + return -EINVAL; + } + } + + tmp = ci->transfers; + while (tmp->formats && tmp->rates) { + if (tmp->transfer_in) + in = 1; + else + out = 1; + tmp++; + } + + cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); + if (!cii) { + printk(KERN_DEBUG "i2sbus: failed to allocate cii\n"); + return -ENOMEM; + } + + /* use the private data to point to the codec info */ + cii->sdev = soundbus_dev_get(dev); + cii->codec = ci; + cii->codec_data = data; + + if (!cii->sdev) { + printk(KERN_DEBUG + "i2sbus: failed to get soundbus dev reference\n"); + kfree(cii); + return -ENODEV; + } + + if (!try_module_get(THIS_MODULE)) { + printk(KERN_DEBUG "i2sbus: failed to get module reference!\n"); + soundbus_dev_put(dev); + kfree(cii); + return -EBUSY; + } + + if (!try_module_get(ci->owner)) { + printk(KERN_DEBUG + "i2sbus: failed to get module reference to codec owner!\n"); + module_put(THIS_MODULE); + soundbus_dev_put(dev); + kfree(cii); + return -EBUSY; + } + + if (!dev->pcm) { + err = snd_pcm_new(card, + dev->pcmname, + dev->pcmid, + 0, + 0, + &dev->pcm); + if (err) { + printk(KERN_DEBUG "i2sbus: failed to create pcm\n"); + kfree(cii); + module_put(ci->owner); + soundbus_dev_put(dev); + module_put(THIS_MODULE); + return err; + } + } + + /* ALSA yet again sucks. + * If it is ever fixed, remove this line. See below. */ + out = in = 1; + + if (!i2sdev->out.created && out) { + if (dev->pcm->card != card) { + /* eh? */ + printk(KERN_ERR + "Can't attach same bus to different cards!\n"); + module_put(ci->owner); + kfree(cii); + soundbus_dev_put(dev); + module_put(THIS_MODULE); + return -EINVAL; + } + if ((err = + snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1))) { + module_put(ci->owner); + kfree(cii); + soundbus_dev_put(dev); + module_put(THIS_MODULE); + return err; + } + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &i2sbus_playback_ops); + i2sdev->out.created = 1; + } + + if (!i2sdev->in.created && in) { + if (dev->pcm->card != card) { + printk(KERN_ERR + "Can't attach same bus to different cards!\n"); + module_put(ci->owner); + kfree(cii); + soundbus_dev_put(dev); + module_put(THIS_MODULE); + return -EINVAL; + } + if ((err = + snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1))) { + module_put(ci->owner); + kfree(cii); + soundbus_dev_put(dev); + module_put(THIS_MODULE); + return err; + } + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + &i2sbus_record_ops); + i2sdev->in.created = 1; + } + + /* so we have to register the pcm after adding any substream + * to it because alsa doesn't create the devices for the + * substreams when we add them later. + * Therefore, force in and out on both busses (above) and + * register the pcm now instead of just after creating it. + */ + err = snd_device_register(card, dev->pcm); + if (err) { + printk(KERN_ERR "i2sbus: error registering new pcm\n"); + module_put(ci->owner); + kfree(cii); + soundbus_dev_put(dev); + module_put(THIS_MODULE); + return err; + } + /* no errors any more, so let's add this to our list */ + list_add(&cii->list, &dev->codec_list); + + dev->pcm->private_data = i2sdev; + dev->pcm->private_free = i2sbus_private_free; + + /* well, we really should support scatter/gather DMA */ + snd_pcm_lib_preallocate_pages_for_all( + dev->pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(macio_get_pci_dev(i2sdev->macio)), + 64 * 1024, 64 * 1024); + + return 0; +} + +void i2sbus_detach_codec(struct soundbus_dev *dev, void *data) +{ + struct codec_info_item *cii = NULL, *i; + + list_for_each_entry(i, &dev->codec_list, list) { + if (i->codec_data == data) { + cii = i; + break; + } + } + if (cii) { + list_del(&cii->list); + module_put(cii->codec->owner); + kfree(cii); + } + /* no more codecs, but still a pcm? */ + if (list_empty(&dev->codec_list) && dev->pcm) { + /* the actual cleanup is done by the callback above! */ + snd_device_free(dev->pcm->card, dev->pcm); + } +} diff --git a/sound/aoa/soundbus/i2sbus/i2sbus.h b/sound/aoa/soundbus/i2sbus/i2sbus.h new file mode 100644 index 0000000000..cfa5162e3b --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/i2sbus.h @@ -0,0 +1,112 @@ +/* + * i2sbus driver -- private definitions + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#ifndef __I2SBUS_H +#define __I2SBUS_H +#include +#include +#include +#include +#include +#include +#include "i2sbus-interface.h" +#include "i2sbus-control.h" +#include "../soundbus.h" + +struct i2sbus_control { + volatile struct i2s_control_regs __iomem *controlregs; + struct resource rsrc; + struct list_head list; +}; + +#define MAX_DBDMA_COMMANDS 32 + +struct dbdma_command_mem { + dma_addr_t bus_addr; + dma_addr_t bus_cmd_start; + struct dbdma_cmd *cmds; + void *space; + int size; + u32 running:1; +}; + +struct pcm_info { + u32 created:1, /* has this direction been created with alsa? */ + active:1; /* is this stream active? */ + /* runtime information */ + struct snd_pcm_substream *substream; + int current_period; + u32 frame_count; + struct dbdma_command_mem dbdma_ring; + volatile struct dbdma_regs __iomem *dbdma; +}; + +struct i2sbus_dev { + struct soundbus_dev sound; + struct macio_dev *macio; + struct i2sbus_control *control; + volatile struct i2s_interface_regs __iomem *intfregs; + + struct resource resources[3]; + struct resource *allocated_resource[3]; + int interrupts[3]; + char rnames[3][32]; + + /* info about currently active substreams */ + struct pcm_info out, in; + snd_pcm_format_t format; + unsigned int rate; + + /* list for a single controller */ + struct list_head item; + /* number of bus on controller */ + int bus_number; + /* for use by control layer */ + struct pmf_function *enable, + *cell_enable, + *cell_disable, + *clock_enable, + *clock_disable; + + /* locks */ + /* spinlock for low-level interrupt locking */ + spinlock_t low_lock; + /* mutex for high-level consistency */ + struct mutex lock; +}; + +#define soundbus_dev_to_i2sbus_dev(sdev) \ + container_of(sdev, struct i2sbus_dev, sound) + +/* pcm specific functions */ +extern int +i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, + struct codec_info *ci, void *data); +extern void +i2sbus_detach_codec(struct soundbus_dev *dev, void *data); +extern irqreturn_t +i2sbus_tx_intr(int irq, void *devid, struct pt_regs *regs); +extern irqreturn_t +i2sbus_rx_intr(int irq, void *devid, struct pt_regs *regs); + +/* control specific functions */ +extern int i2sbus_control_init(struct macio_dev* dev, + struct i2sbus_control **c); +extern void i2sbus_control_destroy(struct i2sbus_control *c); +extern int i2sbus_control_add_dev(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev); +extern void i2sbus_control_remove_dev(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev); +extern int i2sbus_control_enable(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev); +extern int i2sbus_control_cell(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev, + int enable); +extern int i2sbus_control_clock(struct i2sbus_control *c, + struct i2sbus_dev *i2sdev, + int enable); +#endif /* __I2SBUS_H */ diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h new file mode 100644 index 0000000000..5c27297835 --- /dev/null +++ b/sound/aoa/soundbus/soundbus.h @@ -0,0 +1,202 @@ +/* + * soundbus generic definitions + * + * Copyright 2006 Johannes Berg + * + * GPL v2, can be found in COPYING. + */ +#ifndef __SOUNDBUS_H +#define __SOUNDBUS_H + +#include +#include +#include + + +/* When switching from master to slave or the other way around, + * you don't want to have the codec chip acting as clock source + * while the bus still is. + * More importantly, while switch from slave to master, you need + * to turn off the chip's master function first, but then there's + * no clock for a while and other chips might reset, so we notify + * their drivers after having switched. + * The constants here are codec-point of view, so when we switch + * the soundbus to master we tell the codec we're going to switch + * and give it CLOCK_SWITCH_PREPARE_SLAVE! + */ +enum clock_switch { + CLOCK_SWITCH_PREPARE_SLAVE, + CLOCK_SWITCH_PREPARE_MASTER, + CLOCK_SWITCH_SLAVE, + CLOCK_SWITCH_MASTER, + CLOCK_SWITCH_NOTIFY, +}; + +/* information on a transfer the codec can take */ +struct transfer_info { + u64 formats; /* SNDRV_PCM_FMTBIT_* */ + unsigned int rates; /* SNDRV_PCM_RATE_* */ + /* flags */ + u32 transfer_in:1, /* input = 1, output = 0 */ + must_be_clock_source:1; + /* for codecs to distinguish among their TIs */ + int tag; +}; + +struct codec_info_item { + struct codec_info *codec; + void *codec_data; + struct soundbus_dev *sdev; + /* internal, to be used by the soundbus provider */ + struct list_head list; +}; + +/* for prepare, where the codecs need to know + * what we're going to drive the bus with */ +struct bus_info { + /* see below */ + int sysclock_factor; + int bus_factor; +}; + +/* information on the codec itself, plus function pointers */ +struct codec_info { + /* the module this lives in */ + struct module *owner; + + /* supported transfer possibilities, array terminated by + * formats or rates being 0. */ + struct transfer_info *transfers; + + /* Master clock speed factor + * to be used (master clock speed = sysclock_factor * sampling freq) + * Unused if the soundbus provider has no such notion. + */ + int sysclock_factor; + + /* Bus factor, bus clock speed = bus_factor * sampling freq) + * Unused if the soundbus provider has no such notion. + */ + int bus_factor; + + /* operations */ + /* clock switching, see above */ + int (*switch_clock)(struct codec_info_item *cii, + enum clock_switch clock); + + /* called for each transfer_info when the user + * opens the pcm device to determine what the + * hardware can support at this point in time. + * That can depend on other user-switchable controls. + * Return 1 if usable, 0 if not. + * out points to another instance of a transfer_info + * which is initialised to the values in *ti, and + * it's format and rate values can be modified by + * the callback if it is necessary to further restrict + * the formats that can be used at the moment, for + * example when one codec has multiple logical codec + * info structs for multiple inputs. + */ + int (*usable)(struct codec_info_item *cii, + struct transfer_info *ti, + struct transfer_info *out); + + /* called when pcm stream is opened, probably not implemented + * most of the time since it isn't too useful */ + int (*open)(struct codec_info_item *cii, + struct snd_pcm_substream *substream); + + /* called when the pcm stream is closed, at this point + * the user choices can all be unlocked (see below) */ + int (*close)(struct codec_info_item *cii, + struct snd_pcm_substream *substream); + + /* if the codec must forbid some user choices because + * they are not valid with the substream/transfer info, + * it must do so here. Example: no digital output for + * incompatible framerate, say 8KHz, on Onyx. + * If the selected stuff in the substream is NOT + * compatible, you have to reject this call! */ + int (*prepare)(struct codec_info_item *cii, + struct bus_info *bi, + struct snd_pcm_substream *substream); + + /* start() is called before data is pushed to the codec. + * Note that start() must be atomic! */ + int (*start)(struct codec_info_item *cii, + struct snd_pcm_substream *substream); + + /* stop() is called after data is no longer pushed to the codec. + * Note that stop() must be atomic! */ + int (*stop)(struct codec_info_item *cii, + struct snd_pcm_substream *substream); + + int (*suspend)(struct codec_info_item *cii, pm_message_t state); + int (*resume)(struct codec_info_item *cii); +}; + +/* information on a soundbus device */ +struct soundbus_dev { + /* the bus it belongs to */ + struct list_head onbuslist; + + /* the of device it represents */ + struct of_device ofdev; + + /* what modules go by */ + char modalias[32]; + + /* These fields must be before attach_codec can be called. + * They should be set by the owner of the alsa card object + * that is needed, and whoever sets them must make sure + * that they are unique within that alsa card object. */ + char *pcmname; + int pcmid; + + /* this is assigned by the soundbus provider in attach_codec */ + struct snd_pcm *pcm; + + /* operations */ + /* attach a codec to this soundbus, give the alsa + * card object the PCMs for this soundbus should be in. + * The 'data' pointer must be unique, it is used as the + * key for detach_codec(). */ + int (*attach_codec)(struct soundbus_dev *dev, struct snd_card *card, + struct codec_info *ci, void *data); + void (*detach_codec)(struct soundbus_dev *dev, void *data); + /* TODO: suspend/resume */ + + /* private for the soundbus provider */ + struct list_head codec_list; + u32 have_out:1, have_in:1; +}; +#define to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev.dev) +#define of_to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev) + +extern int soundbus_add_one(struct soundbus_dev *dev); +extern void soundbus_remove_one(struct soundbus_dev *dev); + +extern struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev); +extern void soundbus_dev_put(struct soundbus_dev *dev); + +struct soundbus_driver { + char *name; + struct module *owner; + + /* we don't implement any matching at all */ + + int (*probe)(struct soundbus_dev* dev); + int (*remove)(struct soundbus_dev* dev); + + int (*suspend)(struct soundbus_dev* dev, pm_message_t state); + int (*resume)(struct soundbus_dev* dev); + int (*shutdown)(struct soundbus_dev* dev); + + struct device_driver driver; +}; +#define to_soundbus_driver(drv) container_of(drv,struct soundbus_driver, driver) + +extern int soundbus_register_driver(struct soundbus_driver *drv); +extern void soundbus_unregister_driver(struct soundbus_driver *drv); + +#endif /* __SOUNDBUS_H */ diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c new file mode 100644 index 0000000000..d31f814695 --- /dev/null +++ b/sound/aoa/soundbus/sysfs.c @@ -0,0 +1,43 @@ +#include +#include +#include +/* FIX UP */ +#include "soundbus.h" + +#define soundbus_config_of_attr(field, format_string) \ +static ssize_t \ +field##_show (struct device *dev, struct device_attribute *attr, \ + char *buf) \ +{ \ + struct soundbus_dev *mdev = to_soundbus_device (dev); \ + return sprintf (buf, format_string, mdev->ofdev.node->field); \ +} + +static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, + char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct of_device *of = &sdev->ofdev; + int length; + + if (*sdev->modalias) { + strlcpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1); + strcat(buf, "\n"); + length = strlen(buf); + } else { + length = sprintf(buf, "of:N%sT%s\n", + of->node->name, of->node->type); + } + + return length; +} + +soundbus_config_of_attr (name, "%s\n"); +soundbus_config_of_attr (type, "%s\n"); + +struct device_attribute soundbus_dev_attrs[] = { + __ATTR_RO(name), + __ATTR_RO(type), + __ATTR_RO(modalias), + __ATTR_NULL +}; -- cgit v1.2.2 From 55c385ad5e1f3cda887cd6a8ad69a6d74b4b9125 Mon Sep 17 00:00:00 2001 From: Johannes Berg Date: Wed, 21 Jun 2006 15:43:44 +0200 Subject: [ALSA] snd-powermac: no longer handle anything with a layout-id property This patch removes from snd-powermac the code that check for the layout-id and instead adds code that makes it refuse loading when a layout-id property is present, nothing that snd-aoa should be used. It also removes the 'toonie' codec from snd-powermac which was only ever used on the mac mini which has a layout-id property. Signed-off-by: Johannes Berg Signed-off-by: Takashi Iwai --- sound/ppc/Makefile | 2 +- sound/ppc/pmac.c | 44 ++++++++------------------------------------ sound/ppc/pmac.h | 3 +-- sound/ppc/powermac.c | 7 ------- 4 files changed, 10 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile index d6ba995909..4d95c652c8 100644 --- a/sound/ppc/Makefile +++ b/sound/ppc/Makefile @@ -3,7 +3,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o toonie.o keywest.o beep.o +snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o # Toplevel Module Dependency obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index f0794ef9d1..b678814975 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -867,8 +867,6 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) unsigned int *prop, l; struct macio_chip* macio; - u32 layout_id = 0; - if (!machine_is(powermac)) return -ENODEV; @@ -929,8 +927,14 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) if (prop && *prop < 16) chip->subframe = *prop; prop = (unsigned int *) get_property(sound, "layout-id", NULL); - if (prop) - layout_id = *prop; + if (prop) { + /* partly deprecate snd-powermac, for those machines + * that have a layout-id property for now */ + printk(KERN_INFO "snd-powermac no longer handles any " + "machines with a layout-id property " + "in the device-tree, use snd-aoa.\n"); + return -ENODEV; + } /* This should be verified on older screamers */ if (device_is_compatible(sound, "screamer")) { chip->model = PMAC_SCREAMER; @@ -963,38 +967,6 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) chip->freq_table = tumbler_freqs; chip->control_mask = MASK_IEPC | 0x11; /* disable IEE */ } - if (device_is_compatible(sound, "AOAKeylargo") || - device_is_compatible(sound, "AOAbase") || - device_is_compatible(sound, "AOAK2")) { - /* For now, only support very basic TAS3004 based machines with - * single frequency until proper i2s control is implemented - */ - switch(layout_id) { - case 0x24: - case 0x29: - case 0x33: - case 0x46: - case 0x48: - case 0x50: - case 0x5c: - chip->num_freqs = ARRAY_SIZE(tumbler_freqs); - chip->model = PMAC_SNAPPER; - chip->can_byte_swap = 0; /* FIXME: check this */ - chip->control_mask = MASK_IEPC | 0x11;/* disable IEE */ - break; - case 0x3a: - chip->num_freqs = ARRAY_SIZE(tumbler_freqs); - chip->model = PMAC_TOONIE; - chip->can_byte_swap = 0; /* FIXME: check this */ - chip->control_mask = MASK_IEPC | 0x11;/* disable IEE */ - break; - default: - printk(KERN_ERR "snd: Unknown layout ID 0x%x\n", - layout_id); - return -ENODEV; - - } - } prop = (unsigned int *)get_property(sound, "device-id", NULL); if (prop) chip->device_id = *prop; diff --git a/sound/ppc/pmac.h b/sound/ppc/pmac.h index 3a9bd4dbb9..8394e66ceb 100644 --- a/sound/ppc/pmac.h +++ b/sound/ppc/pmac.h @@ -85,7 +85,7 @@ struct pmac_stream { enum snd_pmac_model { PMAC_AWACS, PMAC_SCREAMER, PMAC_BURGUNDY, PMAC_DACA, PMAC_TUMBLER, - PMAC_SNAPPER, PMAC_TOONIE + PMAC_SNAPPER }; struct snd_pmac { @@ -188,7 +188,6 @@ int snd_pmac_burgundy_init(struct snd_pmac *chip); int snd_pmac_daca_init(struct snd_pmac *chip); int snd_pmac_tumbler_init(struct snd_pmac *chip); int snd_pmac_tumbler_post_init(void); -int snd_pmac_toonie_init(struct snd_pmac *chip); /* i2c functions */ struct pmac_keywest { diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 875f1f7bdc..fa9a44ab48 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -94,13 +94,6 @@ static int __init snd_pmac_probe(struct platform_device *devptr) if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0) goto __error; break; - case PMAC_TOONIE: - strcpy(card->driver, "PMac Toonie"); - strcpy(card->shortname, "PowerMac Toonie"); - strcpy(card->longname, card->shortname); - if ((err = snd_pmac_toonie_init(chip)) < 0) - goto __error; - break; case PMAC_AWACS: case PMAC_SCREAMER: name_ext = chip->model == PMAC_SCREAMER ? "Screamer" : "AWACS"; -- cgit v1.2.2 From 555fdc2e9fb2071fdd10ff1d86b8d63265d80241 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jun 2006 16:23:31 +0200 Subject: [ALSA] Remove ppc/toonie.c Remove obsoleted ppc/toonie.c. The function is replaced with new snd-aoa driver. Signed-off-by: Takashi Iwai --- sound/ppc/toonie.c | 378 ----------------------------------------------------- 1 file changed, 378 deletions(-) (limited to 'sound') diff --git a/sound/ppc/toonie.c b/sound/ppc/toonie.c index 1ac7c8552f..e69de29bb2 100644 --- a/sound/ppc/toonie.c +++ b/sound/ppc/toonie.c @@ -1,378 +0,0 @@ -/* - * Mac Mini "toonie" mixer control - * - * Copyright (c) 2005 by Benjamin Herrenschmidt - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "pmac.h" - -#undef DEBUG - -#ifdef DEBUG -#define DBG(fmt...) printk(fmt) -#else -#define DBG(fmt...) -#endif - -struct pmac_gpio { - unsigned int addr; - u8 active_val; - u8 inactive_val; - u8 active_state; -}; - -struct pmac_toonie -{ - struct pmac_gpio hp_detect_gpio; - struct pmac_gpio hp_mute_gpio; - struct pmac_gpio amp_mute_gpio; - int hp_detect_irq; - int auto_mute_notify; - struct work_struct detect_work; -}; - - -/* - * gpio access - */ -#define do_gpio_write(gp, val) \ - pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, (gp)->addr, val) -#define do_gpio_read(gp) \ - pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, (gp)->addr, 0) -#define tumbler_gpio_free(gp) /* NOP */ - -static void write_audio_gpio(struct pmac_gpio *gp, int active) -{ - if (! gp->addr) - return; - active = active ? gp->active_val : gp->inactive_val; - do_gpio_write(gp, active); - DBG("(I) gpio %x write %d\n", gp->addr, active); -} - -static int check_audio_gpio(struct pmac_gpio *gp) -{ - int ret; - - if (! gp->addr) - return 0; - - ret = do_gpio_read(gp); - - return (ret & 0xd) == (gp->active_val & 0xd); -} - -static int read_audio_gpio(struct pmac_gpio *gp) -{ - int ret; - if (! gp->addr) - return 0; - ret = ((do_gpio_read(gp) & 0x02) !=0); - return ret == gp->active_state; -} - - -enum { TOONIE_MUTE_HP, TOONIE_MUTE_AMP }; - -static int toonie_get_mute_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - struct pmac_toonie *mix = chip->mixer_data; - struct pmac_gpio *gp; - - if (mix == NULL) - return -ENODEV; - switch(kcontrol->private_value) { - case TOONIE_MUTE_HP: - gp = &mix->hp_mute_gpio; - break; - case TOONIE_MUTE_AMP: - gp = &mix->amp_mute_gpio; - break; - default: - return -EINVAL; - } - ucontrol->value.integer.value[0] = !check_audio_gpio(gp); - return 0; -} - -static int toonie_put_mute_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_pmac *chip = snd_kcontrol_chip(kcontrol); - struct pmac_toonie *mix = chip->mixer_data; - struct pmac_gpio *gp; - int val; - - if (chip->update_automute && chip->auto_mute) - return 0; /* don't touch in the auto-mute mode */ - - if (mix == NULL) - return -ENODEV; - - switch(kcontrol->private_value) { - case TOONIE_MUTE_HP: - gp = &mix->hp_mute_gpio; - break; - case TOONIE_MUTE_AMP: - gp = &mix->amp_mute_gpio; - break; - default: - return -EINVAL; - } - val = ! check_audio_gpio(gp); - if (val != ucontrol->value.integer.value[0]) { - write_audio_gpio(gp, ! ucontrol->value.integer.value[0]); - return 1; - } - return 0; -} - -static struct snd_kcontrol_new toonie_hp_sw __initdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Headphone Playback Switch", - .info = snd_pmac_boolean_mono_info, - .get = toonie_get_mute_switch, - .put = toonie_put_mute_switch, - .private_value = TOONIE_MUTE_HP, -}; -static struct snd_kcontrol_new toonie_speaker_sw __initdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", - .info = snd_pmac_boolean_mono_info, - .get = toonie_get_mute_switch, - .put = toonie_put_mute_switch, - .private_value = TOONIE_MUTE_AMP, -}; - -/* - * auto-mute stuffs - */ -static int toonie_detect_headphone(struct snd_pmac *chip) -{ - struct pmac_toonie *mix = chip->mixer_data; - int detect = 0; - - if (mix->hp_detect_gpio.addr) - detect |= read_audio_gpio(&mix->hp_detect_gpio); - return detect; -} - -static void toonie_check_mute(struct snd_pmac *chip, struct pmac_gpio *gp, int val, - int do_notify, struct snd_kcontrol *sw) -{ - if (check_audio_gpio(gp) != val) { - write_audio_gpio(gp, val); - if (do_notify) - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &sw->id); - } -} - -static void toonie_detect_handler(void *self) -{ - struct snd_pmac *chip = (struct snd_pmac *) self; - struct pmac_toonie *mix; - int headphone; - - if (!chip) - return; - - mix = chip->mixer_data; - snd_assert(mix, return); - - headphone = toonie_detect_headphone(chip); - - DBG("headphone: %d, lineout: %d\n", headphone, lineout); - - if (headphone) { - /* unmute headphone/lineout & mute speaker */ - toonie_check_mute(chip, &mix->hp_mute_gpio, 0, - mix->auto_mute_notify, chip->master_sw_ctl); - toonie_check_mute(chip, &mix->amp_mute_gpio, 1, - mix->auto_mute_notify, chip->speaker_sw_ctl); - } else { - /* unmute speaker, mute others */ - toonie_check_mute(chip, &mix->amp_mute_gpio, 0, - mix->auto_mute_notify, chip->speaker_sw_ctl); - toonie_check_mute(chip, &mix->hp_mute_gpio, 1, - mix->auto_mute_notify, chip->master_sw_ctl); - } - if (mix->auto_mute_notify) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->hp_detect_ctl->id); - } -} - -static void toonie_update_automute(struct snd_pmac *chip, int do_notify) -{ - if (chip->auto_mute) { - struct pmac_toonie *mix; - mix = chip->mixer_data; - snd_assert(mix, return); - mix->auto_mute_notify = do_notify; - schedule_work(&mix->detect_work); - } -} - -/* interrupt - headphone plug changed */ -static irqreturn_t toonie_hp_intr(int irq, void *devid, struct pt_regs *regs) -{ - struct snd_pmac *chip = devid; - - if (chip->update_automute && chip->initialized) { - chip->update_automute(chip, 1); - return IRQ_HANDLED; - } - return IRQ_NONE; -} - -/* look for audio gpio device */ -static int find_audio_gpio(const char *name, const char *platform, - struct pmac_gpio *gp) -{ - struct device_node *np; - u32 *base, addr; - - if (! (np = find_devices("gpio"))) - return -ENODEV; - - for (np = np->child; np; np = np->sibling) { - char *property = get_property(np, "audio-gpio", NULL); - if (property && strcmp(property, name) == 0) - break; - if (device_is_compatible(np, name)) - break; - } - if (np == NULL) - return -ENODEV; - - base = (u32 *)get_property(np, "AAPL,address", NULL); - if (! base) { - base = (u32 *)get_property(np, "reg", NULL); - if (!base) { - DBG("(E) cannot find address for device %s !\n", name); - return -ENODEV; - } - addr = *base; - if (addr < 0x50) - addr += 0x50; - } else - addr = *base; - - gp->addr = addr & 0x0000ffff; - - /* Try to find the active state, default to 0 ! */ - base = (u32 *)get_property(np, "audio-gpio-active-state", NULL); - if (base) { - gp->active_state = *base; - gp->active_val = (*base) ? 0x5 : 0x4; - gp->inactive_val = (*base) ? 0x4 : 0x5; - } else { - u32 *prop = NULL; - gp->active_state = 0; - gp->active_val = 0x4; - gp->inactive_val = 0x5; - /* Here are some crude hacks to extract the GPIO polarity and - * open collector informations out of the do-platform script - * as we don't yet have an interpreter for these things - */ - if (platform) - prop = (u32 *)get_property(np, platform, NULL); - if (prop) { - if (prop[3] == 0x9 && prop[4] == 0x9) { - gp->active_val = 0xd; - gp->inactive_val = 0xc; - } - if (prop[3] == 0x1 && prop[4] == 0x1) { - gp->active_val = 0x5; - gp->inactive_val = 0x4; - } - } - } - - DBG("(I) GPIO device %s found, offset: %x, active state: %d !\n", - name, gp->addr, gp->active_state); - - return (np->n_intrs > 0) ? np->intrs[0].line : 0; -} - -static void toonie_cleanup(struct snd_pmac *chip) -{ - struct pmac_toonie *mix = chip->mixer_data; - if (! mix) - return; - if (mix->hp_detect_irq >= 0) - free_irq(mix->hp_detect_irq, chip); - kfree(mix); - chip->mixer_data = NULL; -} - -int __init snd_pmac_toonie_init(struct snd_pmac *chip) -{ - struct pmac_toonie *mix; - - mix = kmalloc(sizeof(*mix), GFP_KERNEL); - if (! mix) - return -ENOMEM; - - chip->mixer_data = mix; - chip->mixer_free = toonie_cleanup; - - find_audio_gpio("headphone-mute", NULL, &mix->hp_mute_gpio); - find_audio_gpio("amp-mute", NULL, &mix->amp_mute_gpio); - mix->hp_detect_irq = find_audio_gpio("headphone-detect", - NULL, &mix->hp_detect_gpio); - - strcpy(chip->card->mixername, "PowerMac Toonie"); - - chip->master_sw_ctl = snd_ctl_new1(&toonie_hp_sw, chip); - snd_ctl_add(chip->card, chip->master_sw_ctl); - - chip->speaker_sw_ctl = snd_ctl_new1(&toonie_speaker_sw, chip); - snd_ctl_add(chip->card, chip->speaker_sw_ctl); - - INIT_WORK(&mix->detect_work, toonie_detect_handler, (void *)chip); - - if (mix->hp_detect_irq >= 0) { - snd_pmac_add_automute(chip); - - chip->detect_headphone = toonie_detect_headphone; - chip->update_automute = toonie_update_automute; - toonie_update_automute(chip, 0); - - if (request_irq(mix->hp_detect_irq, toonie_hp_intr, 0, - "Sound Headphone Detection", chip) < 0) - mix->hp_detect_irq = -1; - } - - return 0; -} - -- cgit v1.2.2 From 45df379798b5c3b2ea937735ef04c58ce0f532a7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jun 2006 16:37:54 +0200 Subject: [ALSA] Remove nested mutexes in seq_ports.c Removed nested mutexes in the removal routine of port connections. The port is guaranteed to be offline before calling it, so no mutex is needed. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ports.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 5f46ee9e21..334579a9f2 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -221,7 +221,6 @@ static void clear_subscriber_list(struct snd_seq_client *client, { struct list_head *p, *n; - down_write(&grp->list_mutex); list_for_each_safe(p, n, &grp->list_head) { struct snd_seq_subscribers *subs; struct snd_seq_client *c; @@ -259,7 +258,6 @@ static void clear_subscriber_list(struct snd_seq_client *client, snd_seq_client_unlock(c); } } - up_write(&grp->list_mutex); } /* delete port data */ -- cgit v1.2.2 From 5885492ab4fb18c155000d12f920754f7f35fbab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jun 2006 19:19:25 +0200 Subject: [ALSA] hda-codec - Show EAPD and pin-detection capabilities in proc Show EAPD and pin-detection capabilities in proc files. They are often required to support the proper audio functionality. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 3db009990c..c2f0fe85bf 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -182,6 +182,10 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " OUT"); if (caps & AC_PINCAP_HP_DRV) snd_iprintf(buffer, " HP"); + if (caps & AC_PINCAP_EAPD) + snd_iprintf(buffer, " EAPD"); + if (caps & AC_PINCAP_PRES_DETECT) + snd_iprintf(buffer, " Detect"); snd_iprintf(buffer, "\n"); caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, -- cgit v1.2.2 From 607c0fbee7272be4d5455d6b29f6ffb092573eff Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 22 Jun 2006 17:49:58 +0200 Subject: [ALSA] aoa driver - Kconfig - remove spaces for SND!=n Signed-off-by: Jaroslav Kysela --- sound/aoa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig index b11ccf6dba..a85194fe0b 100644 --- a/sound/aoa/Kconfig +++ b/sound/aoa/Kconfig @@ -1,5 +1,5 @@ menu "Apple Onboard Audio driver" - depends on SND != n && PPC + depends on SND!=n && PPC config SND_AOA tristate "Apple Onboard Audio driver" -- cgit v1.2.2 From f2c780c1fdbe5008c902c2d7e37242ac5e60f0b9 Mon Sep 17 00:00:00 2001 From: Sergei Shtylyov Date: Fri, 23 Jun 2006 02:04:13 -0700 Subject: [PATCH] Au1550/1200: add missing PSC #define's, make OSS driver use the proper ones Add missing PSC #define's required for the drivers using PSC on DBAu1550 board (also fixing Au1550 PSC3 address) and all Au1200-based boards as well. Make the OSS driver use the correct PSC definitions fo each board. Signed-off-by: Sergei Shtylyov Cc: Ralf Baechle Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/oss/au1550_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index c1168fae6b..9011abe241 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -57,9 +57,9 @@ #include #include #include -#include #include #include +#include #undef OSS_DOCUMENTED_MIXER_SEMANTICS -- cgit v1.2.2 From 2eebb1925d25cfd7e7cd2eb18ac4d4e6d189dba0 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Fri, 23 Jun 2006 02:05:26 -0700 Subject: [PATCH] OSS: cs46xx cleanup and tiny bugfix Here's a patch for cs46xx that - (mostly) cleans up the cs46xx driver according to CodingStyle - removes a bunch of pointless casts - fixes a small, potential use of uninitialized variable, bug - reduces the size of the compiled code by 36 bytes - reduces the size of the source file by 1831 bytes I know I should probably have split this into bits, but since I only thought of that *after* doing all the edits, splitting it up would have been a royal pain. And since these are all pretty trivial changes I thought I'd just submit the one huge patch and hope people could live with it (if not, then just tell me and I'll split it). The bug fix that's in there may be hard to spot, so I'll point it out. It's the - int val, valsave, mapped, ret; + int val, valsave, ret; + int mapped = 0; bit. Without that change we may use `mapped' uninitialized if, in cs_ioctl, the first test of "if(state)" is false and the second "if(state)" test is true. Signed-off-by: Jesper Juhl Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/oss/cs46xx.c | 1272 ++++++++++++++++++++-------------------------------- 1 file changed, 489 insertions(+), 783 deletions(-) (limited to 'sound') diff --git a/sound/oss/cs46xx.c b/sound/oss/cs46xx.c index 53881bc91b..994c71e986 100644 --- a/sound/oss/cs46xx.c +++ b/sound/oss/cs46xx.c @@ -147,7 +147,7 @@ * that should be printed on any released driver. */ #if CSDEBUG -#define CS_DBGOUT(mask,level,x) if((cs_debuglevel >= (level)) && ((mask) & cs_debugmask)) {x;} +#define CS_DBGOUT(mask,level,x) if ((cs_debuglevel >= (level)) && ((mask) & cs_debugmask)) {x;} #else #define CS_DBGOUT(mask,level,x) #endif @@ -175,19 +175,19 @@ #define CS_IOCTL_CMD_RESUME 0x2 // resume #if CSDEBUG -static unsigned long cs_debuglevel=1; /* levels range from 1-9 */ +static unsigned long cs_debuglevel = 1; /* levels range from 1-9 */ module_param(cs_debuglevel, ulong, 0644); -static unsigned long cs_debugmask=CS_INIT | CS_ERROR; /* use CS_DBGOUT with various mask values */ +static unsigned long cs_debugmask = CS_INIT | CS_ERROR; /* use CS_DBGOUT with various mask values */ module_param(cs_debugmask, ulong, 0644); #endif static unsigned long hercules_egpio_disable; /* if non-zero set all EGPIO to 0 */ module_param(hercules_egpio_disable, ulong, 0); -static unsigned long initdelay=700; /* PM delay in millisecs */ +static unsigned long initdelay = 700; /* PM delay in millisecs */ module_param(initdelay, ulong, 0); -static unsigned long powerdown=-1; /* turn on/off powerdown processing in driver */ +static unsigned long powerdown = -1; /* turn on/off powerdown processing in driver */ module_param(powerdown, ulong, 0); #define DMABUF_DEFAULTORDER 3 -static unsigned long defaultorder=DMABUF_DEFAULTORDER; +static unsigned long defaultorder = DMABUF_DEFAULTORDER; module_param(defaultorder, ulong, 0); static int external_amp; @@ -200,8 +200,8 @@ module_param(thinkpad, bool, 0); * powerdown. also set thinkpad to 1 to disable powerdown, * but also to enable the clkrun functionality. */ -static unsigned cs_powerdown=1; -static unsigned cs_laptop_wait=1; +static unsigned cs_powerdown = 1; +static unsigned cs_laptop_wait = 1; /* An instance of the 4610 channel */ struct cs_channel @@ -319,7 +319,7 @@ struct cs_card { atomic_t mixer_use_cnt; /* PCI device stuff */ - struct pci_dev * pci_dev; + struct pci_dev *pci_dev; struct list_head list; unsigned int pctl, cctl; /* Hardware DMA flag sets */ @@ -384,7 +384,7 @@ struct cs_card { static int cs_open_mixdev(struct inode *inode, struct file *file); static int cs_release_mixdev(struct inode *inode, struct file *file); static int cs_ioctl_mixdev(struct inode *inode, struct file *file, unsigned int cmd, - unsigned long arg); + unsigned long arg); static int cs_hardware_init(struct cs_card *card); static int cs46xx_powerup(struct cs_card *card, unsigned int type); static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspendflag); @@ -423,8 +423,7 @@ static void printioctl(unsigned int x) [SOUND_MIXER_VOLUME] = 9 /* Master Volume */ }; - switch(x) - { + switch (x) { case SOUND_MIXER_CS_GETDBGMASK: CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CS_GETDBGMASK: ") ); break; @@ -521,7 +520,6 @@ static void printioctl(unsigned int x) case SOUND_PCM_READ_FILTER: CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_FILTER: ") ); break; - case SOUND_MIXER_PRIVATE1: CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE1: ") ); break; @@ -543,10 +541,8 @@ static void printioctl(unsigned int x) case SOUND_OLD_MIXER_INFO: CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_OLD_MIXER_INFO: ") ); break; - default: - switch (_IOC_NR(x)) - { + switch (_IOC_NR(x)) { case SOUND_MIXER_VOLUME: CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_VOLUME: ") ); break; @@ -579,14 +575,11 @@ static void printioctl(unsigned int x) break; default: i = _IOC_NR(x); - if (i >= SOUND_MIXER_NRDEVICES || !(vidx = mixtable1[i])) - { + if (i >= SOUND_MIXER_NRDEVICES || !(vidx = mixtable1[i])) { CS_DBGOUT(CS_IOCTL, 4, printk("UNKNOWN IOCTL: 0x%.8x NR=%d ",x,i) ); - } - else - { + } else { CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_IOCTL AC9x: 0x%.8x NR=%d ", - x,i) ); + x,i)); } break; } @@ -601,22 +594,22 @@ static void printioctl(unsigned int x) static void cs461x_poke(struct cs_card *codec, unsigned long reg, unsigned int val) { - writel(val, codec->ba1.idx[(reg >> 16) & 3]+(reg&0xffff)); + writel(val, codec->ba1.idx[(reg >> 16) & 3] + (reg & 0xffff)); } static unsigned int cs461x_peek(struct cs_card *codec, unsigned long reg) { - return readl(codec->ba1.idx[(reg >> 16) & 3]+(reg&0xffff)); + return readl(codec->ba1.idx[(reg >> 16) & 3] + (reg & 0xffff)); } static void cs461x_pokeBA0(struct cs_card *codec, unsigned long reg, unsigned int val) { - writel(val, codec->ba0+reg); + writel(val, codec->ba0 + reg); } static unsigned int cs461x_peekBA0(struct cs_card *codec, unsigned long reg) { - return readl(codec->ba0+reg); + return readl(codec->ba0 + reg); } @@ -625,26 +618,26 @@ static void cs_ac97_set(struct ac97_codec *dev, u8 reg, u16 data); static struct cs_channel *cs_alloc_pcm_channel(struct cs_card *card) { - if(card->channel[1].used==1) + if (card->channel[1].used == 1) return NULL; - card->channel[1].used=1; - card->channel[1].num=1; + card->channel[1].used = 1; + card->channel[1].num = 1; return &card->channel[1]; } static struct cs_channel *cs_alloc_rec_pcm_channel(struct cs_card *card) { - if(card->channel[0].used==1) + if (card->channel[0].used == 1) return NULL; - card->channel[0].used=1; - card->channel[0].num=0; + card->channel[0].used = 1; + card->channel[0].num = 0; return &card->channel[0]; } static void cs_free_pcm_channel(struct cs_card *card, int channel) { card->channel[channel].state = NULL; - card->channel[channel].used=0; + card->channel[channel].used = 0; } /* @@ -655,15 +648,15 @@ static void cs_free_pcm_channel(struct cs_card *card, int channel) */ static void cs_set_divisor(struct dmabuf *dmabuf) { - if(dmabuf->type == CS_TYPE_DAC) + if (dmabuf->type == CS_TYPE_DAC) dmabuf->divisor = 1; - else if( !(dmabuf->fmt & CS_FMT_STEREO) && + else if (!(dmabuf->fmt & CS_FMT_STEREO) && (dmabuf->fmt & CS_FMT_16BIT)) dmabuf->divisor = 2; - else if( (dmabuf->fmt & CS_FMT_STEREO) && + else if ((dmabuf->fmt & CS_FMT_STEREO) && !(dmabuf->fmt & CS_FMT_16BIT)) dmabuf->divisor = 2; - else if( !(dmabuf->fmt & CS_FMT_STEREO) && + else if (!(dmabuf->fmt & CS_FMT_STEREO) && !(dmabuf->fmt & CS_FMT_16BIT)) dmabuf->divisor = 4; else @@ -680,13 +673,12 @@ static void cs_set_divisor(struct dmabuf *dmabuf) */ static void cs_mute(struct cs_card *card, int state) { - struct ac97_codec *dev=card->ac97_codec[0]; + struct ac97_codec *dev = card->ac97_codec[0]; CS_DBGOUT(CS_FUNCTION, 2, printk(KERN_INFO "cs46xx: cs_mute()+ %s\n", - (state == CS_TRUE) ? "Muting" : "UnMuting") ); + (state == CS_TRUE) ? "Muting" : "UnMuting")); - if(state == CS_TRUE) - { + if (state == CS_TRUE) { /* * fix pops when powering up on thinkpads */ @@ -703,9 +695,7 @@ static void cs_mute(struct cs_card *card, int state) cs_ac97_set(dev, (u8)BA0_AC97_HEADPHONE_VOLUME, 0x8000); cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME_MONO, 0x8000); cs_ac97_set(dev, (u8)BA0_AC97_PCM_OUT_VOLUME, 0x8000); - } - else - { + } else { cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME, card->pm.u32AC97_master_volume); cs_ac97_set(dev, (u8)BA0_AC97_HEADPHONE_VOLUME, card->pm.u32AC97_headphone_volume); cs_ac97_set(dev, (u8)BA0_AC97_MASTER_VOLUME_MONO, card->pm.u32AC97_master_volume_mono); @@ -757,7 +747,6 @@ static unsigned int cs_set_dac_rate(struct cs_state * state, unsigned int rate) /* * Fill in the SampleRateConverter control block. */ - spin_lock_irqsave(&state->card->lock, flags); cs461x_poke(state->card, BA1_PSRC, ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF)); @@ -770,7 +759,7 @@ static unsigned int cs_set_dac_rate(struct cs_state * state, unsigned int rate) } /* set recording sample rate */ -static unsigned int cs_set_adc_rate(struct cs_state * state, unsigned int rate) +static unsigned int cs_set_adc_rate(struct cs_state *state, unsigned int rate) { struct dmabuf *dmabuf = &state->dmabuf; struct cs_card *card = state->card; @@ -815,7 +804,6 @@ static unsigned int cs_set_adc_rate(struct cs_state * state, unsigned int rate) * dividend:remainder(ulOther / GOF_PER_SEC) * initialDelay = dividend(((24 * Fs,in) + Fs,out - 1) / Fs,out) */ - tmp1 = rate << 16; coeffIncr = tmp1 / 48000; tmp1 -= coeffIncr * 48000; @@ -891,7 +879,7 @@ static void cs_play_setup(struct cs_state *state) CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_play_setup()+\n") ); cs461x_poke(card, BA1_PVOL, 0x80008000); - if(!dmabuf->SGok) + if (!dmabuf->SGok) cs461x_poke(card, BA1_PBA, virt_to_bus(dmabuf->pbuf)); Count = 4; @@ -899,16 +887,14 @@ static void cs_play_setup(struct cs_state *state) if ((dmabuf->fmt & CS_FMT_STEREO)) { playFormat &= ~DMA_RQ_C2_AC_MONO_TO_STEREO; Count *= 2; - } - else + } else playFormat |= DMA_RQ_C2_AC_MONO_TO_STEREO; if ((dmabuf->fmt & CS_FMT_16BIT)) { playFormat &= ~(DMA_RQ_C2_AC_8_TO_16_BIT | DMA_RQ_C2_AC_SIGNED_CONVERT); Count *= 2; - } - else + } else playFormat |= (DMA_RQ_C2_AC_8_TO_16_BIT | DMA_RQ_C2_AC_SIGNED_CONVERT); @@ -919,7 +905,6 @@ static void cs_play_setup(struct cs_state *state) cs461x_poke(card, BA1_PDTC, tmp | --Count); CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_play_setup()-\n") ); - } static struct InitStruct @@ -944,8 +929,7 @@ static void SetCaptureSPValues(struct cs_card *card) { unsigned i, offset; CS_DBGOUT(CS_FUNCTION, 8, printk("cs46xx: SetCaptureSPValues()+\n") ); - for(i=0; icard; struct dmabuf *dmabuf = &state->dmabuf; - CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_rec_setup()+\n") ); + CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_rec_setup()+\n")); SetCaptureSPValues(card); /* @@ -994,14 +978,11 @@ static inline unsigned cs_get_dma_addr(struct cs_state *state) /* * granularity is byte boundary, good part. */ - if(dmabuf->enable & DAC_RUNNING) - { + if (dmabuf->enable & DAC_RUNNING) offset = cs461x_peek(state->card, BA1_PBA); - } else /* ADC_RUNNING must be set */ - { offset = cs461x_peek(state->card, BA1_CBA); - } + CS_DBGOUT(CS_PARMS | CS_FUNCTION, 9, printk("cs46xx: cs_get_dma_addr() %d\n",offset) ); offset = (u32)bus_to_virt((unsigned long)offset) - (u32)dmabuf->rawbuf; @@ -1015,8 +996,7 @@ static void resync_dma_ptrs(struct cs_state *state) struct dmabuf *dmabuf; CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: resync_dma_ptrs()+ \n") ); - if(state) - { + if (state) { dmabuf = &state->dmabuf; dmabuf->hwptr=dmabuf->swptr = 0; dmabuf->pringbuf = 0; @@ -1149,13 +1129,13 @@ static int alloc_dmabuf(struct cs_state *state) /* * check for order within limits, but do not overwrite value. */ - if((defaultorder > 1) && (defaultorder < 12)) + if ((defaultorder > 1) && (defaultorder < 12)) df = defaultorder; else df = 2; for (order = df; order >= DMABUF_MINORDER; order--) - if ( (rawbuf = (void *) pci_alloc_consistent( + if ((rawbuf = (void *)pci_alloc_consistent( card->pci_dev, PAGE_SIZE << order, &dmabuf->dmaaddr))) break; if (!rawbuf) { @@ -1181,8 +1161,7 @@ static int alloc_dmabuf(struct cs_state *state) /* * only allocate the conversion buffer for the ADC */ - if(dmabuf->type == CS_TYPE_DAC) - { + if (dmabuf->type == CS_TYPE_DAC) { dmabuf->tmpbuff = NULL; dmabuf->buforder_tmpbuff = 0; return 0; @@ -1258,8 +1237,7 @@ static int __prog_dmabuf(struct cs_state *state) /* * check for CAPTURE and use only non-sg for initial release */ - if(dmabuf->type == CS_TYPE_ADC) - { + if (dmabuf->type == CS_TYPE_ADC) { CS_DBGOUT(CS_FUNCTION, 4, printk("cs46xx: prog_dmabuf() ADC\n")); /* * add in non-sg support for capture. @@ -1313,9 +1291,7 @@ static int __prog_dmabuf(struct cs_state *state) CS_DBGOUT(CS_FUNCTION, 4, printk("cs46xx: prog_dmabuf()- 0 \n")); return 0; - } - else if (dmabuf->type == CS_TYPE_DAC) - { + } else if (dmabuf->type == CS_TYPE_DAC) { /* * Must be DAC */ @@ -1337,8 +1313,7 @@ static int __prog_dmabuf(struct cs_state *state) allocated_pages = 1 << dmabuf->buforder; allocated_bytes = allocated_pages*PAGE_SIZE; - if(allocated_pages < 2) - { + if (allocated_pages < 2) { CS_DBGOUT(CS_FUNCTION, 4, printk( "cs46xx: prog_dmabuf() Error: allocated_pages too small (%d)\n", (unsigned)allocated_pages)); @@ -1353,14 +1328,14 @@ static int __prog_dmabuf(struct cs_state *state) /* Set up S/G variables. */ *ptmp = virt_to_bus(dmabuf->rawbuf); - *(ptmp+1) = 0x00000008; - for(tmp1= 1; tmp1 < nSGpages; tmp1++) { - *(ptmp+2*tmp1) = virt_to_bus( (dmabuf->rawbuf)+4096*tmp1); - if( tmp1 == nSGpages-1) + *(ptmp + 1) = 0x00000008; + for (tmp1 = 1; tmp1 < nSGpages; tmp1++) { + *(ptmp + 2 * tmp1) = virt_to_bus((dmabuf->rawbuf) + 4096 * tmp1); + if (tmp1 == nSGpages - 1) tmp2 = 0xbfff0000; else - tmp2 = 0x80000000+8*(tmp1+1); - *(ptmp+2*tmp1+1) = tmp2; + tmp2 = 0x80000000 + 8 * (tmp1 + 1); + *(ptmp + 2 * tmp1 + 1) = tmp2; } SGarray[0] = 0x82c0200d; SGarray[1] = 0xffff0000; @@ -1368,18 +1343,17 @@ static int __prog_dmabuf(struct cs_state *state) SGarray[3] = 0x00010600; SGarray[4] = *(ptmp+2); SGarray[5] = 0x80000010; - SGarray[6] = *ptmp; - SGarray[7] = *(ptmp+2); - SGarray[8] = (virt_to_bus(dmabuf->pbuf) & 0xffff000) | 0x10; - - if (dmabuf->SGok) { - dmabuf->numfrag = nSGpages; - dmabuf->fragsize = 4096; - dmabuf->fragsamples = 4096 >> sample_shift[dmabuf->fmt]; - dmabuf->fragshift = 12; - dmabuf->dmasize = dmabuf->numfrag*4096; - } - else { + SGarray[6] = *ptmp; + SGarray[7] = *(ptmp+2); + SGarray[8] = (virt_to_bus(dmabuf->pbuf) & 0xffff000) | 0x10; + + if (dmabuf->SGok) { + dmabuf->numfrag = nSGpages; + dmabuf->fragsize = 4096; + dmabuf->fragsamples = 4096 >> sample_shift[dmabuf->fmt]; + dmabuf->fragshift = 12; + dmabuf->dmasize = dmabuf->numfrag * 4096; + } else { SGarray[0] = 0xf2c0000f; SGarray[1] = 0x00000200; SGarray[2] = 0; @@ -1391,8 +1365,8 @@ static int __prog_dmabuf(struct cs_state *state) dmabuf->dmasize = 4096; dmabuf->fragshift = 11; } - for(tmp1 = 0; tmp1 < sizeof(SGarray)/4; tmp1++) - cs461x_poke( state->card, BA1_PDTC+tmp1*4, SGarray[tmp1]); + for (tmp1 = 0; tmp1 < sizeof(SGarray) / 4; tmp1++) + cs461x_poke(state->card, BA1_PDTC+tmp1 * 4, SGarray[tmp1]); memset(dmabuf->rawbuf, (dmabuf->fmt & CS_FMT_16BIT) ? 0 : 0x80, dmabuf->dmasize); @@ -1416,9 +1390,7 @@ static int __prog_dmabuf(struct cs_state *state) CS_DBGOUT(CS_FUNCTION, 4, printk("cs46xx: prog_dmabuf()- \n")); return 0; - } - else - { + } else { CS_DBGOUT(CS_FUNCTION, 4, printk("cs46xx: prog_dmabuf()- Invalid Type %d\n", dmabuf->type)); } @@ -1489,8 +1461,7 @@ static int drain_dac(struct cs_state *state, int nonblock) } remove_wait_queue(&dmabuf->wait, &wait); current->state = TASK_RUNNING; - if (signal_pending(current)) - { + if (signal_pending(current)) { CS_DBGOUT(CS_FUNCTION, 4, printk("cs46xx: drain_dac()- -ERESTARTSYS\n")); /* * set to silence and let that clear the fifos. @@ -1514,8 +1485,7 @@ static void cs_update_ptr(struct cs_card *card, int wake) /* error handling and process wake up for ADC */ state = card->states[0]; - if(state) - { + if (state) { dmabuf = &state->dmabuf; if (dmabuf->enable & ADC_RUNNING) { /* update hardware pointer */ @@ -1531,12 +1501,10 @@ static void cs_update_ptr(struct cs_card *card, int wake) if (dmabuf->count > dmabuf->dmasize) dmabuf->count = dmabuf->dmasize; - if(dmabuf->mapped) - { + if (dmabuf->mapped) { if (wake && dmabuf->count >= (signed)dmabuf->fragsize) wake_up(&dmabuf->wait); - } else - { + } else { if (wake && dmabuf->count > 0) wake_up(&dmabuf->wait); } @@ -1547,8 +1515,7 @@ static void cs_update_ptr(struct cs_card *card, int wake) * Now the DAC */ state = card->states[1]; - if(state) - { + if (state) { dmabuf = &state->dmabuf; /* error handling and process wake up for DAC */ if (dmabuf->enable & DAC_RUNNING) { @@ -1570,7 +1537,7 @@ static void cs_update_ptr(struct cs_card *card, int wake) * in that, since dmasize is the buffer asked for * via mmap. */ - if( dmabuf->count > dmabuf->dmasize) + if (dmabuf->count > dmabuf->dmasize) dmabuf->count &= dmabuf->dmasize-1; } else { dmabuf->count -= diff; @@ -1578,13 +1545,10 @@ static void cs_update_ptr(struct cs_card *card, int wake) * backfill with silence and clear out the last * "diff" number of bytes. */ - if(hwptr >= diff) - { + if (hwptr >= diff) { memset(dmabuf->rawbuf + hwptr - diff, (dmabuf->fmt & CS_FMT_16BIT) ? 0 : 0x80, diff); - } - else - { + } else { memset(dmabuf->rawbuf, (dmabuf->fmt & CS_FMT_16BIT) ? 0 : 0x80, (unsigned)hwptr); @@ -1602,12 +1566,12 @@ static void cs_update_ptr(struct cs_card *card, int wake) * buffer underrun or buffer overrun, reset the * count of bytes written back to 0. */ - if(dmabuf->count < 0) - dmabuf->underrun=1; + if (dmabuf->count < 0) + dmabuf->underrun = 1; dmabuf->count = 0; dmabuf->error++; } - if (wake && dmabuf->count < (signed)dmabuf->dmasize/2) + if (wake && dmabuf->count < (signed)dmabuf->dmasize / 2) wake_up(&dmabuf->wait); } } @@ -1661,8 +1625,7 @@ static irqreturn_t cs_interrupt(int irq, void *dev_id, struct pt_regs *regs) status = cs461x_peekBA0(card, BA0_HISR); - if ((status & 0x7fffffff) == 0) - { + if ((status & 0x7fffffff) == 0) { cs461x_pokeBA0(card, BA0_HICR, HICR_CHGM|HICR_IEV); spin_unlock(&card->lock); return IRQ_HANDLED; /* Might be IRQ_NONE.. */ @@ -1671,15 +1634,14 @@ static irqreturn_t cs_interrupt(int irq, void *dev_id, struct pt_regs *regs) /* * check for playback or capture interrupt only */ - if( ((status & HISR_VC0) && playstate && playstate->dmabuf.ready) || - (((status & HISR_VC1) && recstate && recstate->dmabuf.ready)) ) - { + if (((status & HISR_VC0) && playstate && playstate->dmabuf.ready) || + (((status & HISR_VC1) && recstate && recstate->dmabuf.ready))) { CS_DBGOUT(CS_INTERRUPT, 8, printk( "cs46xx: cs_interrupt() interrupt bit(s) set (0x%x)\n",status)); cs_update_ptr(card, CS_TRUE); } - if( status & HISR_MIDI ) + if (status & HISR_MIDI) cs_handle_midi(card); /* clear 'em */ @@ -1694,7 +1656,7 @@ static irqreturn_t cs_interrupt(int irq, void *dev_id, struct pt_regs *regs) static ssize_t cs_midi_read(struct file *file, char __user *buffer, size_t count, loff_t *ppos) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; ssize_t ret; unsigned long flags; unsigned ptr; @@ -1737,7 +1699,7 @@ static ssize_t cs_midi_read(struct file *file, char __user *buffer, size_t count static ssize_t cs_midi_write(struct file *file, const char __user *buffer, size_t count, loff_t *ppos) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; ssize_t ret; unsigned long flags; unsigned ptr; @@ -1785,7 +1747,7 @@ static ssize_t cs_midi_write(struct file *file, const char __user *buffer, size_ static unsigned int cs_midi_poll(struct file *file, struct poll_table_struct *wait) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; unsigned long flags; unsigned int mask = 0; @@ -1810,12 +1772,11 @@ static unsigned int cs_midi_poll(struct file *file, struct poll_table_struct *wa static int cs_midi_open(struct inode *inode, struct file *file) { unsigned int minor = iminor(inode); - struct cs_card *card=NULL; + struct cs_card *card = NULL; unsigned long flags; struct list_head *entry; - list_for_each(entry, &cs46xx_devs) - { + list_for_each(entry, &cs46xx_devs) { card = list_entry(entry, struct cs_card, list); if (card->dev_midi == minor) break; @@ -1823,8 +1784,7 @@ static int cs_midi_open(struct inode *inode, struct file *file) if (entry == &cs46xx_devs) return -ENODEV; - if (!card) - { + if (!card) { CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO "cs46xx: cs46xx_midi_open(): Error - unable to find card struct\n")); return -ENODEV; @@ -1852,12 +1812,10 @@ static int cs_midi_open(struct inode *inode, struct file *file) cs461x_pokeBA0(card, BA0_MIDCR, 0x0000000f); /* Enable xmit, rcv. */ cs461x_pokeBA0(card, BA0_HICR, HICR_IEV | HICR_CHGM); /* Enable interrupts */ } - if (file->f_mode & FMODE_READ) { + if (file->f_mode & FMODE_READ) card->midi.ird = card->midi.iwr = card->midi.icnt = 0; - } - if (file->f_mode & FMODE_WRITE) { + if (file->f_mode & FMODE_WRITE) card->midi.ord = card->midi.owr = card->midi.ocnt = 0; - } spin_unlock_irqrestore(&card->midi.lock, flags); card->midi.open_mode |= (file->f_mode & (FMODE_READ | FMODE_WRITE)); mutex_unlock(&card->midi.open_mutex); @@ -1867,7 +1825,7 @@ static int cs_midi_open(struct inode *inode, struct file *file) static int cs_midi_release(struct inode *inode, struct file *file) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; DECLARE_WAITQUEUE(wait, current); unsigned long flags; unsigned count, tmo; @@ -1933,11 +1891,10 @@ static /*const*/ struct file_operations cs_midi_fops = { static void CopySamples(char *dst, char *src, int count, unsigned fmt, struct dmabuf *dmabuf) { - s32 s32AudioSample; - s16 *psSrc=(s16 *)src; - s16 *psDst=(s16 *)dst; - u8 *pucDst=(u8 *)dst; + s16 *psSrc = (s16 *)src; + s16 *psDst = (s16 *)dst; + u8 *pucDst = (u8 *)dst; CS_DBGOUT(CS_FUNCTION, 2, printk(KERN_INFO "cs46xx: CopySamples()+ ") ); CS_DBGOUT(CS_WAVE_READ, 8, printk(KERN_INFO @@ -1947,34 +1904,29 @@ static void CopySamples(char *dst, char *src, int count, unsigned fmt, /* * See if the data should be output as 8-bit unsigned stereo. */ - if((fmt & CS_FMT_STEREO) && !(fmt & CS_FMT_16BIT)) - { + if ((fmt & CS_FMT_STEREO) && !(fmt & CS_FMT_16BIT)) { /* * Convert each 16-bit signed stereo sample to 8-bit unsigned * stereo using rounding. */ psSrc = (s16 *)src; - count = count/2; - while(count--) - { + count = count / 2; + while (count--) *(pucDst++) = (u8)(((s16)(*psSrc++) + (s16)0x8000) >> 8); - } } /* * See if the data should be output at 8-bit unsigned mono. */ - else if(!(fmt & CS_FMT_STEREO) && !(fmt & CS_FMT_16BIT)) - { + else if (!(fmt & CS_FMT_STEREO) && !(fmt & CS_FMT_16BIT)) { /* * Convert each 16-bit signed stereo sample to 8-bit unsigned * mono using averaging and rounding. */ psSrc = (s16 *)src; - count = count/2; - while(count--) - { - s32AudioSample = ((*psSrc)+(*(psSrc + 1)))/2 + (s32)0x80; - if(s32AudioSample > 0x7fff) + count = count / 2; + while (count--) { + s32AudioSample = ((*psSrc) + (*(psSrc + 1))) / 2 + (s32)0x80; + if (s32AudioSample > 0x7fff) s32AudioSample = 0x7fff; *(pucDst++) = (u8)(((s16)s32AudioSample + (s16)0x8000) >> 8); psSrc += 2; @@ -1983,17 +1935,15 @@ static void CopySamples(char *dst, char *src, int count, unsigned fmt, /* * See if the data should be output at 16-bit signed mono. */ - else if(!(fmt & CS_FMT_STEREO) && (fmt & CS_FMT_16BIT)) - { + else if (!(fmt & CS_FMT_STEREO) && (fmt & CS_FMT_16BIT)) { /* * Convert each 16-bit signed stereo sample to 16-bit signed * mono using averaging. */ psSrc = (s16 *)src; - count = count/2; - while(count--) - { - *(psDst++) = (s16)((*psSrc)+(*(psSrc + 1)))/2; + count = count / 2; + while (count--) { + *(psDst++) = (s16)((*psSrc) + (*(psSrc + 1))) / 2; psSrc += 2; } } @@ -2020,20 +1970,15 @@ static unsigned cs_copy_to_user( "cs_copy_to_user()+ fmt=0x%x cnt=%d dest=%p\n", dmabuf->fmt,(unsigned)cnt,dest) ); - if(cnt > dmabuf->dmasize) - { + if (cnt > dmabuf->dmasize) cnt = dmabuf->dmasize; - } - if(!cnt) - { + if (!cnt) { *copied = 0; return 0; } - if(dmabuf->divisor != 1) - { - if(!dmabuf->tmpbuff) - { - *copied = cnt/dmabuf->divisor; + if (dmabuf->divisor != 1) { + if (!dmabuf->tmpbuff) { + *copied = cnt / dmabuf->divisor; return 0; } @@ -2042,17 +1987,16 @@ static unsigned cs_copy_to_user( src = dmabuf->tmpbuff; cnt = cnt/dmabuf->divisor; } - if (copy_to_user(dest, src, cnt)) - { + if (copy_to_user(dest, src, cnt)) { CS_DBGOUT(CS_FUNCTION, 2, printk(KERN_ERR "cs46xx: cs_copy_to_user()- fault dest=%p src=%p cnt=%d\n", - dest,src,cnt) ); + dest,src,cnt)); *copied = 0; return -EFAULT; } *copied = cnt; CS_DBGOUT(CS_FUNCTION, 2, printk(KERN_INFO - "cs46xx: cs_copy_to_user()- copied bytes is %d \n",cnt) ); + "cs46xx: cs_copy_to_user()- copied bytes is %d \n",cnt)); return 0; } @@ -2060,7 +2004,7 @@ static unsigned cs_copy_to_user( the user's buffer. it is filled by the dma machine and drained by this loop. */ static ssize_t cs_read(struct file *file, char __user *buffer, size_t count, loff_t *ppos) { - struct cs_card *card = (struct cs_card *) file->private_data; + struct cs_card *card = file->private_data; struct cs_state *state; DECLARE_WAITQUEUE(wait, current); struct dmabuf *dmabuf; @@ -2068,12 +2012,12 @@ static ssize_t cs_read(struct file *file, char __user *buffer, size_t count, lof unsigned long flags; unsigned swptr; int cnt; - unsigned copied=0; + unsigned copied = 0; CS_DBGOUT(CS_WAVE_READ | CS_FUNCTION, 4, printk("cs46xx: cs_read()+ %zd\n",count) ); - state = (struct cs_state *)card->states[0]; - if(!state) + state = card->states[0]; + if (!state) return -ENODEV; dmabuf = &state->dmabuf; @@ -2088,11 +2032,11 @@ static ssize_t cs_read(struct file *file, char __user *buffer, size_t count, lof add_wait_queue(&state->dmabuf.wait, &wait); while (count > 0) { - while(!(card->pm.flags & CS46XX_PM_IDLE)) - { + while (!(card->pm.flags & CS46XX_PM_IDLE)) { schedule(); if (signal_pending(current)) { - if(!ret) ret = -ERESTARTSYS; + if (!ret) + ret = -ERESTARTSYS; goto out; } } @@ -2112,19 +2056,20 @@ static ssize_t cs_read(struct file *file, char __user *buffer, size_t count, lof recorded */ start_adc(state); if (file->f_flags & O_NONBLOCK) { - if (!ret) ret = -EAGAIN; + if (!ret) + ret = -EAGAIN; goto out; } mutex_unlock(&state->sem); schedule(); if (signal_pending(current)) { - if(!ret) ret = -ERESTARTSYS; + if (!ret) + ret = -ERESTARTSYS; goto out; } mutex_lock(&state->sem); - if (dmabuf->mapped) - { - if(!ret) + if (dmabuf->mapped) { + if (!ret) ret = -ENXIO; goto out; } @@ -2135,12 +2080,12 @@ static ssize_t cs_read(struct file *file, char __user *buffer, size_t count, lof "_read() copy_to cnt=%d count=%zd ", cnt,count) ); CS_DBGOUT(CS_WAVE_READ, 8, printk(KERN_INFO " .dmasize=%d .count=%d buffer=%p ret=%zd\n", - dmabuf->dmasize,dmabuf->count,buffer,ret) ); + dmabuf->dmasize,dmabuf->count,buffer,ret)); if (cs_copy_to_user(state, buffer, - (char *)dmabuf->rawbuf + swptr, cnt, &copied)) - { - if (!ret) ret = -EFAULT; + (char *)dmabuf->rawbuf + swptr, cnt, &copied)) { + if (!ret) + ret = -EFAULT; goto out; } swptr = (swptr + cnt) % dmabuf->dmasize; @@ -2167,7 +2112,7 @@ out2: the soundcard. it is drained by the dma machine and filled by this loop. */ static ssize_t cs_write(struct file *file, const char __user *buffer, size_t count, loff_t *ppos) { - struct cs_card *card = (struct cs_card *) file->private_data; + struct cs_card *card = file->private_data; struct cs_state *state; DECLARE_WAITQUEUE(wait, current); struct dmabuf *dmabuf; @@ -2178,16 +2123,15 @@ static ssize_t cs_write(struct file *file, const char __user *buffer, size_t cou CS_DBGOUT(CS_WAVE_WRITE | CS_FUNCTION, 4, printk("cs46xx: cs_write called, count = %zd\n", count) ); - state = (struct cs_state *)card->states[1]; - if(!state) + state = card->states[1]; + if (!state) return -ENODEV; if (!access_ok(VERIFY_READ, buffer, count)) return -EFAULT; dmabuf = &state->dmabuf; mutex_lock(&state->sem); - if (dmabuf->mapped) - { + if (dmabuf->mapped) { ret = -ENXIO; goto out; } @@ -2201,11 +2145,11 @@ static ssize_t cs_write(struct file *file, const char __user *buffer, size_t cou * check for PM events and underrun/overrun in the loop. */ while (count > 0) { - while(!(card->pm.flags & CS46XX_PM_IDLE)) - { + while (!(card->pm.flags & CS46XX_PM_IDLE)) { schedule(); if (signal_pending(current)) { - if(!ret) ret = -ERESTARTSYS; + if (!ret) + ret = -ERESTARTSYS; goto out; } } @@ -2216,8 +2160,7 @@ static ssize_t cs_write(struct file *file, const char __user *buffer, size_t cou dmabuf->count = 0; dmabuf->swptr = dmabuf->hwptr; } - if (dmabuf->underrun) - { + if (dmabuf->underrun) { dmabuf->underrun = 0; dmabuf->hwptr = cs_get_dma_addr(state); dmabuf->swptr = dmabuf->hwptr; @@ -2238,34 +2181,35 @@ static ssize_t cs_write(struct file *file, const char __user *buffer, size_t cou played */ start_dac(state); if (file->f_flags & O_NONBLOCK) { - if (!ret) ret = -EAGAIN; + if (!ret) + ret = -EAGAIN; goto out; } mutex_unlock(&state->sem); schedule(); if (signal_pending(current)) { - if(!ret) ret = -ERESTARTSYS; + if (!ret) + ret = -ERESTARTSYS; goto out; } mutex_lock(&state->sem); - if (dmabuf->mapped) - { - if(!ret) + if (dmabuf->mapped) { + if (!ret) ret = -ENXIO; goto out; } continue; } if (copy_from_user(dmabuf->rawbuf + swptr, buffer, cnt)) { - if (!ret) ret = -EFAULT; + if (!ret) + ret = -EFAULT; goto out; } spin_lock_irqsave(&state->card->lock, flags); swptr = (swptr + cnt) % dmabuf->dmasize; dmabuf->swptr = swptr; dmabuf->count += cnt; - if(dmabuf->count > dmabuf->dmasize) - { + if (dmabuf->count > dmabuf->dmasize) { CS_DBGOUT(CS_WAVE_WRITE | CS_ERROR, 2, printk( "cs46xx: cs_write() d->count > dmasize - resetting\n")); dmabuf->count = dmabuf->dmasize; @@ -2284,38 +2228,32 @@ out: set_current_state(TASK_RUNNING); CS_DBGOUT(CS_WAVE_WRITE | CS_FUNCTION, 2, - printk("cs46xx: cs_write()- ret=%zd\n", ret) ); + printk("cs46xx: cs_write()- ret=%zd\n", ret)); return ret; } static unsigned int cs_poll(struct file *file, struct poll_table_struct *wait) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; struct dmabuf *dmabuf; struct cs_state *state; - unsigned long flags; unsigned int mask = 0; CS_DBGOUT(CS_FUNCTION, 2, printk("cs46xx: cs_poll()+ \n")); - if (!(file->f_mode & (FMODE_WRITE | FMODE_READ))) - { + if (!(file->f_mode & (FMODE_WRITE | FMODE_READ))) { return -EINVAL; } - if (file->f_mode & FMODE_WRITE) - { + if (file->f_mode & FMODE_WRITE) { state = card->states[1]; - if(state) - { + if (state) { dmabuf = &state->dmabuf; poll_wait(file, &dmabuf->wait, wait); } } - if (file->f_mode & FMODE_READ) - { + if (file->f_mode & FMODE_READ) { state = card->states[0]; - if(state) - { + if (state) { dmabuf = &state->dmabuf; poll_wait(file, &dmabuf->wait, wait); } @@ -2325,8 +2263,7 @@ static unsigned int cs_poll(struct file *file, struct poll_table_struct *wait) cs_update_ptr(card, CS_FALSE); if (file->f_mode & FMODE_READ) { state = card->states[0]; - if(state) - { + if (state) { dmabuf = &state->dmabuf; if (dmabuf->count >= (signed)dmabuf->fragsize) mask |= POLLIN | POLLRDNORM; @@ -2334,8 +2271,7 @@ static unsigned int cs_poll(struct file *file, struct poll_table_struct *wait) } if (file->f_mode & FMODE_WRITE) { state = card->states[1]; - if(state) - { + if (state) { dmabuf = &state->dmabuf; if (dmabuf->mapped) { if (dmabuf->count >= (signed)dmabuf->fragsize) @@ -2364,7 +2300,7 @@ static unsigned int cs_poll(struct file *file, struct poll_table_struct *wait) static int cs_mmap(struct file *file, struct vm_area_struct *vma) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; struct cs_state *state; struct dmabuf *dmabuf; int ret = 0; @@ -2376,8 +2312,7 @@ static int cs_mmap(struct file *file, struct vm_area_struct *vma) if (vma->vm_flags & VM_WRITE) { state = card->states[1]; - if(state) - { + if (state) { CS_DBGOUT(CS_OPEN, 2, printk( "cs46xx: cs_mmap() VM_WRITE - state TRUE prog_dmabuf DAC\n") ); if ((ret = prog_dmabuf(state)) != 0) @@ -2385,8 +2320,7 @@ static int cs_mmap(struct file *file, struct vm_area_struct *vma) } } else if (vma->vm_flags & VM_READ) { state = card->states[0]; - if(state) - { + if (state) { CS_DBGOUT(CS_OPEN, 2, printk( "cs46xx: cs_mmap() VM_READ - state TRUE prog_dmabuf ADC\n") ); if ((ret = prog_dmabuf(state)) != 0) @@ -2414,8 +2348,7 @@ static int cs_mmap(struct file *file, struct vm_area_struct *vma) mutex_lock(&state->sem); dmabuf = &state->dmabuf; - if (cs4x_pgoff(vma) != 0) - { + if (cs4x_pgoff(vma) != 0) { ret = -EINVAL; goto out; } @@ -2423,15 +2356,13 @@ static int cs_mmap(struct file *file, struct vm_area_struct *vma) CS_DBGOUT(CS_PARMS, 2, printk("cs46xx: cs_mmap(): size=%d\n",(unsigned)size) ); - if (size > (PAGE_SIZE << dmabuf->buforder)) - { + if (size > (PAGE_SIZE << dmabuf->buforder)) { ret = -EINVAL; goto out; } if (remap_pfn_range(vma, vma->vm_start, virt_to_phys(dmabuf->rawbuf) >> PAGE_SHIFT, - size, vma->vm_page_prot)) - { + size, vma->vm_page_prot)) { ret = -EAGAIN; goto out; } @@ -2445,25 +2376,24 @@ out: static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long arg) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; struct cs_state *state; - struct dmabuf *dmabuf=NULL; + struct dmabuf *dmabuf = NULL; unsigned long flags; audio_buf_info abinfo; count_info cinfo; - int val, valsave, mapped, ret; + int val, valsave, ret; + int mapped = 0; void __user *argp = (void __user *)arg; int __user *p = argp; - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; mapped = (file->f_mode & FMODE_READ) && dmabuf->mapped; } - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; mapped |= (file->f_mode & FMODE_WRITE) && dmabuf->mapped; } @@ -2472,17 +2402,14 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un printioctl(cmd); #endif - switch (cmd) - { + switch (cmd) { case OSS_GETVERSION: return put_user(SOUND_VERSION, p); - case SNDCTL_DSP_RESET: /* FIXME: spin_lock ? */ if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; stop_dac(state); synchronize_irq(card->irq); @@ -2495,9 +2422,8 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; stop_adc(state); synchronize_irq(card->irq); @@ -2511,20 +2437,17 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } CS_DBGOUT(CS_IOCTL, 2, printk("cs46xx: DSP_RESET()-\n") ); return 0; - case SNDCTL_DSP_SYNC: if (file->f_mode & FMODE_WRITE) return drain_dac(state, file->f_flags & O_NONBLOCK); return 0; - case SNDCTL_DSP_SPEED: /* set sample rate */ if (get_user(val, p)) return -EFAULT; if (val >= 0) { if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; stop_adc(state); dmabuf->ready = 0; @@ -2534,9 +2457,8 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; stop_dac(state); dmabuf->ready = 0; @@ -2553,19 +2475,17 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un return put_user(dmabuf->rate, p); } return put_user(0, p); - case SNDCTL_DSP_STEREO: /* set stereo or mono channel */ if (get_user(val, p)) return -EFAULT; if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; stop_dac(state); dmabuf->ready = 0; dmabuf->SGok = 0; - if(val) + if (val) dmabuf->fmt |= CS_FMT_STEREO; else dmabuf->fmt &= ~CS_FMT_STEREO; @@ -2577,14 +2497,13 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; stop_adc(state); dmabuf->ready = 0; dmabuf->SGok = 0; - if(val) + if (val) dmabuf->fmt |= CS_FMT_STEREO; else dmabuf->fmt &= ~CS_FMT_STEREO; @@ -2596,12 +2515,10 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } return 0; - case SNDCTL_DSP_GETBLKSIZE: if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; if ((val = prog_dmabuf(state))) return val; @@ -2609,9 +2526,8 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; if ((val = prog_dmabuf(state))) return val; @@ -2620,10 +2536,8 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } return put_user(0, p); - case SNDCTL_DSP_GETFMTS: /* Returns a mask of supported sample format*/ return put_user(AFMT_S16_LE | AFMT_U8, p); - case SNDCTL_DSP_SETFMT: /* Select sample format */ if (get_user(val, p)) return -EFAULT; @@ -2635,88 +2549,75 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un val == AFMT_U8 ? "8Bit Unsigned" : "") ); valsave = val; if (val != AFMT_QUERY) { - if(val==AFMT_S16_LE || val==AFMT_U8) - { + if (val==AFMT_S16_LE || val==AFMT_U8) { if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; stop_dac(state); dmabuf->ready = 0; dmabuf->SGok = 0; - if(val==AFMT_S16_LE) + if (val == AFMT_S16_LE) dmabuf->fmt |= CS_FMT_16BIT; else dmabuf->fmt &= ~CS_FMT_16BIT; cs_set_divisor(dmabuf); - if((ret = prog_dmabuf(state))) + if ((ret = prog_dmabuf(state))) return ret; } } if (file->f_mode & FMODE_READ) { val = valsave; - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; stop_adc(state); dmabuf->ready = 0; dmabuf->SGok = 0; - if(val==AFMT_S16_LE) + if (val == AFMT_S16_LE) dmabuf->fmt |= CS_FMT_16BIT; else dmabuf->fmt &= ~CS_FMT_16BIT; cs_set_divisor(dmabuf); - if((ret = prog_dmabuf(state))) + if ((ret = prog_dmabuf(state))) return ret; } } - } - else - { + } else { CS_DBGOUT(CS_IOCTL | CS_ERROR, 2, printk( "cs46xx: DSP_SETFMT() Unsupported format (0x%x)\n", valsave) ); } - } - else - { - if(file->f_mode & FMODE_WRITE) - { - state = (struct cs_state *)card->states[1]; - if(state) + } else { + if (file->f_mode & FMODE_WRITE) { + state = card->states[1]; + if (state) dmabuf = &state->dmabuf; - } - else if(file->f_mode & FMODE_READ) - { - state = (struct cs_state *)card->states[0]; - if(state) + } else if (file->f_mode & FMODE_READ) { + state = card->states[0]; + if (state) dmabuf = &state->dmabuf; } } - if(dmabuf) - { - if(dmabuf->fmt & CS_FMT_16BIT) + if (dmabuf) { + if (dmabuf->fmt & CS_FMT_16BIT) return put_user(AFMT_S16_LE, p); else return put_user(AFMT_U8, p); } return put_user(0, p); - case SNDCTL_DSP_CHANNELS: if (get_user(val, p)) return -EFAULT; if (val != 0) { if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; stop_dac(state); dmabuf->ready = 0; dmabuf->SGok = 0; - if(val>1) + if (val > 1) dmabuf->fmt |= CS_FMT_STEREO; else dmabuf->fmt &= ~CS_FMT_STEREO; @@ -2726,14 +2627,13 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; stop_adc(state); dmabuf->ready = 0; dmabuf->SGok = 0; - if(val>1) + if (val > 1) dmabuf->fmt |= CS_FMT_STEREO; else dmabuf->fmt &= ~CS_FMT_STEREO; @@ -2745,19 +2645,16 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } return put_user((dmabuf->fmt & CS_FMT_STEREO) ? 2 : 1, p); - case SNDCTL_DSP_POST: /* * There will be a longer than normal pause in the data. * so... do nothing, because there is nothing that we can do. */ return 0; - case SNDCTL_DSP_SUBDIVIDE: if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; if (dmabuf->subdivision) return -EINVAL; @@ -2769,9 +2666,8 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; if (dmabuf->subdivision) return -EINVAL; @@ -2783,37 +2679,31 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } return 0; - case SNDCTL_DSP_SETFRAGMENT: if (get_user(val, p)) return -EFAULT; - if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; dmabuf->ossfragshift = val & 0xffff; dmabuf->ossmaxfrags = (val >> 16) & 0xffff; } } if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; dmabuf->ossfragshift = val & 0xffff; dmabuf->ossmaxfrags = (val >> 16) & 0xffff; } } return 0; - case SNDCTL_DSP_GETOSPACE: if (!(file->f_mode & FMODE_WRITE)) return -EINVAL; - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; spin_lock_irqsave(&state->card->lock, flags); cs_update_ptr(card, CS_TRUE); @@ -2832,13 +2722,11 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un return copy_to_user(argp, &abinfo, sizeof(abinfo)) ? -EFAULT : 0; } return -ENODEV; - case SNDCTL_DSP_GETISPACE: if (!(file->f_mode & FMODE_READ)) return -EINVAL; - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; spin_lock_irqsave(&state->card->lock, flags); cs_update_ptr(card, CS_TRUE); @@ -2850,48 +2738,39 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un return copy_to_user(argp, &abinfo, sizeof(abinfo)) ? -EFAULT : 0; } return -ENODEV; - case SNDCTL_DSP_NONBLOCK: file->f_flags |= O_NONBLOCK; return 0; - case SNDCTL_DSP_GETCAPS: return put_user(DSP_CAP_REALTIME|DSP_CAP_TRIGGER|DSP_CAP_MMAP, p); - case SNDCTL_DSP_GETTRIGGER: val = 0; CS_DBGOUT(CS_IOCTL, 2, printk("cs46xx: DSP_GETTRIGGER()+\n") ); - if (file->f_mode & FMODE_WRITE) - { - state = (struct cs_state *)card->states[1]; - if(state) - { + if (file->f_mode & FMODE_WRITE) { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; - if(dmabuf->enable & DAC_RUNNING) + if (dmabuf->enable & DAC_RUNNING) val |= PCM_ENABLE_INPUT; } } - if (file->f_mode & FMODE_READ) - { - if(state) - { - state = (struct cs_state *)card->states[0]; + if (file->f_mode & FMODE_READ) { + if (state) { + state = card->states[0]; dmabuf = &state->dmabuf; - if(dmabuf->enable & ADC_RUNNING) + if (dmabuf->enable & ADC_RUNNING) val |= PCM_ENABLE_OUTPUT; } } CS_DBGOUT(CS_IOCTL, 2, printk("cs46xx: DSP_GETTRIGGER()- val=0x%x\n",val) ); return put_user(val, p); - case SNDCTL_DSP_SETTRIGGER: if (get_user(val, p)) return -EFAULT; if (file->f_mode & FMODE_READ) { - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; if (val & PCM_ENABLE_INPUT) { if (!dmabuf->ready && (ret = prog_dmabuf(state))) @@ -2902,9 +2781,8 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } if (file->f_mode & FMODE_WRITE) { - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; if (val & PCM_ENABLE_OUTPUT) { if (!dmabuf->ready && (ret = prog_dmabuf(state))) @@ -2915,13 +2793,11 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un } } return 0; - case SNDCTL_DSP_GETIPTR: if (!(file->f_mode & FMODE_READ)) return -EINVAL; - state = (struct cs_state *)card->states[0]; - if(state) - { + state = card->states[0]; + if (state) { dmabuf = &state->dmabuf; spin_lock_irqsave(&state->card->lock, flags); cs_update_ptr(card, CS_TRUE); @@ -2934,28 +2810,23 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un return 0; } return -ENODEV; - case SNDCTL_DSP_GETOPTR: if (!(file->f_mode & FMODE_WRITE)) return -EINVAL; - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; spin_lock_irqsave(&state->card->lock, flags); cs_update_ptr(card, CS_TRUE); cinfo.bytes = dmabuf->total_bytes; - if (dmabuf->mapped) - { + if (dmabuf->mapped) { cinfo.blocks = (cinfo.bytes >> dmabuf->fragshift) - dmabuf->blocks; CS_DBGOUT(CS_PARMS, 8, printk("total_bytes=%d blocks=%d dmabuf->blocks=%d\n", cinfo.bytes,cinfo.blocks,dmabuf->blocks) ); dmabuf->blocks = cinfo.bytes >> dmabuf->fragshift; - } - else - { + } else { cinfo.blocks = dmabuf->count >> dmabuf->fragshift; } cinfo.ptr = dmabuf->hwptr; @@ -2969,66 +2840,54 @@ static int cs_ioctl(struct inode *inode, struct file *file, unsigned int cmd, un return 0; } return -ENODEV; - case SNDCTL_DSP_SETDUPLEX: return 0; - case SNDCTL_DSP_GETODELAY: if (!(file->f_mode & FMODE_WRITE)) return -EINVAL; - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; spin_lock_irqsave(&state->card->lock, flags); cs_update_ptr(card, CS_TRUE); val = dmabuf->count; spin_unlock_irqrestore(&state->card->lock, flags); - } - else + } else val = 0; return put_user(val, p); - case SOUND_PCM_READ_RATE: - if(file->f_mode & FMODE_READ) - state = (struct cs_state *)card->states[0]; + if (file->f_mode & FMODE_READ) + state = card->states[0]; else - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; return put_user(dmabuf->rate, p); } return put_user(0, p); - - case SOUND_PCM_READ_CHANNELS: - if(file->f_mode & FMODE_READ) - state = (struct cs_state *)card->states[0]; + if (file->f_mode & FMODE_READ) + state = card->states[0]; else - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; return put_user((dmabuf->fmt & CS_FMT_STEREO) ? 2 : 1, p); } return put_user(0, p); - case SOUND_PCM_READ_BITS: - if(file->f_mode & FMODE_READ) - state = (struct cs_state *)card->states[0]; + if (file->f_mode & FMODE_READ) + state = card->states[0]; else - state = (struct cs_state *)card->states[1]; - if(state) - { + state = card->states[1]; + if (state) { dmabuf = &state->dmabuf; return put_user((dmabuf->fmt & CS_FMT_16BIT) ? AFMT_S16_LE : AFMT_U8, p); } return put_user(0, p); - case SNDCTL_DSP_MAPINBUF: case SNDCTL_DSP_MAPOUTBUF: case SNDCTL_DSP_SETSYNCRO: @@ -3057,18 +2916,15 @@ static void amp_voyetra(struct cs_card *card, int change) /* Manage the EAPD bit on the Crystal 4297 and the Analog AD1885 */ - int old=card->amplifier; + int old = card->amplifier; card->amplifier+=change; - if(card->amplifier && !old) - { + if (card->amplifier && !old) { /* Turn the EAPD amp on */ cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) | 0x8000); - } - else if(old && !card->amplifier) - { + } else if(old && !card->amplifier) { /* Turn the EAPD amp off */ cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & @@ -3083,25 +2939,21 @@ static void amp_voyetra(struct cs_card *card, int change) static void amp_hercules(struct cs_card *card, int change) { - int old=card->amplifier; - if(!card) - { + int old = card->amplifier; + if (!card) { CS_DBGOUT(CS_ERROR, 2, printk(KERN_INFO "cs46xx: amp_hercules() called before initialized.\n")); return; } card->amplifier+=change; - if( (card->amplifier && !old) && !(hercules_egpio_disable)) - { + if ((card->amplifier && !old) && !(hercules_egpio_disable)) { CS_DBGOUT(CS_PARMS, 4, printk(KERN_INFO "cs46xx: amp_hercules() external amp enabled\n")); cs461x_pokeBA0(card, BA0_EGPIODR, EGPIODR_GPOE2); /* enable EGPIO2 output */ cs461x_pokeBA0(card, BA0_EGPIOPTR, EGPIOPTR_GPPT2); /* open-drain on output */ - } - else if(old && !card->amplifier) - { + } else if (old && !card->amplifier) { CS_DBGOUT(CS_PARMS, 4, printk(KERN_INFO "cs46xx: amp_hercules() external amp disabled\n")); cs461x_pokeBA0(card, BA0_EGPIODR, 0); /* disable */ @@ -3124,31 +2976,28 @@ static void clkrun_hack(struct cs_card *card, int change) u16 control; u8 pp; unsigned long port; - int old=card->active; + int old = card->active; card->active+=change; acpi_dev = pci_find_device(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82371AB_3, NULL); - if(acpi_dev == NULL) + if (acpi_dev == NULL) return; /* Not a thinkpad thats for sure */ /* Find the control port */ pci_read_config_byte(acpi_dev, 0x41, &pp); - port=pp<<8; + port = pp << 8; /* Read ACPI port */ - control=inw(port+0x10); + control = inw(port + 0x10); /* Flip CLKRUN off while running */ - if(!card->active && old) - { + if (!card->active && old) { CS_DBGOUT(CS_PARMS , 9, printk( KERN_INFO "cs46xx: clkrun() enable clkrun - change=%d active=%d\n", change,card->active)); outw(control|0x2000, port+0x10); - } - else - { + } else { /* * sometimes on a resume the bit is set, so always reset the bit. */ @@ -3162,20 +3011,19 @@ static void clkrun_hack(struct cs_card *card, int change) static int cs_open(struct inode *inode, struct file *file) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; struct cs_state *state = NULL; struct dmabuf *dmabuf = NULL; struct list_head *entry; unsigned int minor = iminor(inode); - int ret=0; + int ret = 0; unsigned int tmp; CS_DBGOUT(CS_OPEN | CS_FUNCTION, 2, printk("cs46xx: cs_open()+ file=%p %s %s\n", file, file->f_mode & FMODE_WRITE ? "FMODE_WRITE" : "", file->f_mode & FMODE_READ ? "FMODE_READ" : "") ); - list_for_each(entry, &cs46xx_devs) - { + list_for_each(entry, &cs46xx_devs) { card = list_entry(entry, struct cs_card, list); if (!((card->dev_audio ^ minor) & ~0xf)) @@ -3192,11 +3040,10 @@ static int cs_open(struct inode *inode, struct file *file) /* * hardcode state[0] for capture, [1] for playback */ - if(file->f_mode & FMODE_READ) - { + if (file->f_mode & FMODE_READ) { CS_DBGOUT(CS_WAVE_READ, 2, printk("cs46xx: cs_open() FMODE_READ\n") ); if (card->states[0] == NULL) { - state = card->states[0] = (struct cs_state *) + state = card->states[0] = kmalloc(sizeof(struct cs_state), GFP_KERNEL); if (state == NULL) return -ENOMEM; @@ -3204,36 +3051,32 @@ static int cs_open(struct inode *inode, struct file *file) mutex_init(&state->sem); dmabuf = &state->dmabuf; dmabuf->pbuf = (void *)get_zeroed_page(GFP_KERNEL | GFP_DMA); - if(dmabuf->pbuf==NULL) - { + if (dmabuf->pbuf == NULL) { kfree(state); - card->states[0]=NULL; + card->states[0] = NULL; return -ENOMEM; } - } - else - { + } else { state = card->states[0]; - if(state->open_mode & FMODE_READ) + if (state->open_mode & FMODE_READ) return -EBUSY; } dmabuf->channel = card->alloc_rec_pcm_channel(card); if (dmabuf->channel == NULL) { - kfree (card->states[0]); + kfree(card->states[0]); card->states[0] = NULL; return -ENODEV; } /* Now turn on external AMP if needed */ state->card = card; - state->card->active_ctrl(state->card,1); - state->card->amplifier_ctrl(state->card,1); + state->card->active_ctrl(state->card, 1); + state->card->amplifier_ctrl(state->card, 1); - if( (tmp = cs46xx_powerup(card, CS_POWER_ADC)) ) - { + if ((tmp = cs46xx_powerup(card, CS_POWER_ADC))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO - "cs46xx: cs46xx_powerup of ADC failed (0x%x)\n",tmp) ); + "cs46xx: cs46xx_powerup of ADC failed (0x%x)\n", tmp)); return -EIO; } @@ -3263,11 +3106,10 @@ static int cs_open(struct inode *inode, struct file *file) state->open_mode |= FMODE_READ; mutex_unlock(&state->open_mutex); } - if(file->f_mode & FMODE_WRITE) - { + if (file->f_mode & FMODE_WRITE) { CS_DBGOUT(CS_OPEN, 2, printk("cs46xx: cs_open() FMODE_WRITE\n") ); if (card->states[1] == NULL) { - state = card->states[1] = (struct cs_state *) + state = card->states[1] = kmalloc(sizeof(struct cs_state), GFP_KERNEL); if (state == NULL) return -ENOMEM; @@ -3275,36 +3117,32 @@ static int cs_open(struct inode *inode, struct file *file) mutex_init(&state->sem); dmabuf = &state->dmabuf; dmabuf->pbuf = (void *)get_zeroed_page(GFP_KERNEL | GFP_DMA); - if(dmabuf->pbuf==NULL) - { + if (dmabuf->pbuf == NULL) { kfree(state); - card->states[1]=NULL; + card->states[1] = NULL; return -ENOMEM; } - } - else - { + } else { state = card->states[1]; - if(state->open_mode & FMODE_WRITE) + if (state->open_mode & FMODE_WRITE) return -EBUSY; } dmabuf->channel = card->alloc_pcm_channel(card); if (dmabuf->channel == NULL) { - kfree (card->states[1]); + kfree(card->states[1]); card->states[1] = NULL; return -ENODEV; } /* Now turn on external AMP if needed */ state->card = card; - state->card->active_ctrl(state->card,1); - state->card->amplifier_ctrl(state->card,1); + state->card->active_ctrl(state->card, 1); + state->card->amplifier_ctrl(state->card, 1); - if( (tmp = cs46xx_powerup(card, CS_POWER_DAC)) ) - { + if ((tmp = cs46xx_powerup(card, CS_POWER_DAC))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO - "cs46xx: cs46xx_powerup of DAC failed (0x%x)\n",tmp) ); + "cs46xx: cs46xx_powerup of DAC failed (0x%x)\n", tmp)); return -EIO; } @@ -3333,33 +3171,29 @@ static int cs_open(struct inode *inode, struct file *file) state->open_mode |= FMODE_WRITE; mutex_unlock(&state->open_mutex); - if((ret = prog_dmabuf(state))) + if ((ret = prog_dmabuf(state))) return ret; } - CS_DBGOUT(CS_OPEN | CS_FUNCTION, 2, printk("cs46xx: cs_open()- 0\n") ); + CS_DBGOUT(CS_OPEN | CS_FUNCTION, 2, printk("cs46xx: cs_open()- 0\n")); return nonseekable_open(inode, file); } static int cs_release(struct inode *inode, struct file *file) { - struct cs_card *card = (struct cs_card *)file->private_data; + struct cs_card *card = file->private_data; struct dmabuf *dmabuf; struct cs_state *state; unsigned int tmp; CS_DBGOUT(CS_RELEASE | CS_FUNCTION, 2, printk("cs46xx: cs_release()+ file=%p %s %s\n", file, file->f_mode & FMODE_WRITE ? "FMODE_WRITE" : "", - file->f_mode & FMODE_READ ? "FMODE_READ" : "") ); + file->f_mode & FMODE_READ ? "FMODE_READ" : "")); if (!(file->f_mode & (FMODE_WRITE | FMODE_READ))) - { return -EINVAL; - } state = card->states[1]; - if(state) - { - if ( (state->open_mode & FMODE_WRITE) & (file->f_mode & FMODE_WRITE) ) - { - CS_DBGOUT(CS_RELEASE, 2, printk("cs46xx: cs_release() FMODE_WRITE\n") ); + if (state) { + if ((state->open_mode & FMODE_WRITE) & (file->f_mode & FMODE_WRITE)) { + CS_DBGOUT(CS_RELEASE, 2, printk("cs46xx: cs_release() FMODE_WRITE\n")); dmabuf = &state->dmabuf; cs_clear_tail(state); drain_dac(state, file->f_flags & O_NONBLOCK); @@ -3375,8 +3209,7 @@ static int cs_release(struct inode *inode, struct file *file) state->card->states[state->virt] = NULL; state->open_mode &= (~file->f_mode) & (FMODE_READ|FMODE_WRITE); - if( (tmp = cs461x_powerdown(card, CS_POWER_DAC, CS_FALSE )) ) - { + if ((tmp = cs461x_powerdown(card, CS_POWER_DAC, CS_FALSE))) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_INFO "cs46xx: cs_release_mixdev() powerdown DAC failure (0x%x)\n",tmp) ); } @@ -3384,17 +3217,14 @@ static int cs_release(struct inode *inode, struct file *file) /* Now turn off external AMP if needed */ state->card->amplifier_ctrl(state->card, -1); state->card->active_ctrl(state->card, -1); - kfree(state); } } state = card->states[0]; - if(state) - { - if ( (state->open_mode & FMODE_READ) & (file->f_mode & FMODE_READ) ) - { - CS_DBGOUT(CS_RELEASE, 2, printk("cs46xx: cs_release() FMODE_READ\n") ); + if (state) { + if ((state->open_mode & FMODE_READ) & (file->f_mode & FMODE_READ)) { + CS_DBGOUT(CS_RELEASE, 2, printk("cs46xx: cs_release() FMODE_READ\n")); dmabuf = &state->dmabuf; mutex_lock(&state->open_mutex); stop_adc(state); @@ -3407,8 +3237,7 @@ static int cs_release(struct inode *inode, struct file *file) state->card->states[state->virt] = NULL; state->open_mode &= (~file->f_mode) & (FMODE_READ|FMODE_WRITE); - if( (tmp = cs461x_powerdown(card, CS_POWER_ADC, CS_FALSE )) ) - { + if ((tmp = cs461x_powerdown(card, CS_POWER_ADC, CS_FALSE))) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_INFO "cs46xx: cs_release_mixdev() powerdown ADC failure (0x%x)\n",tmp) ); } @@ -3416,12 +3245,11 @@ static int cs_release(struct inode *inode, struct file *file) /* Now turn off external AMP if needed */ state->card->amplifier_ctrl(state->card, -1); state->card->active_ctrl(state->card, -1); - kfree(state); } } - CS_DBGOUT(CS_FUNCTION | CS_RELEASE, 2, printk("cs46xx: cs_release()- 0\n") ); + CS_DBGOUT(CS_FUNCTION | CS_RELEASE, 2, printk("cs46xx: cs_release()- 0\n")); return 0; } @@ -3474,21 +3302,18 @@ static void cs46xx_ac97_suspend(struct cs_card *card) CS_DBGOUT(CS_PM, 9, printk("cs46xx: cs46xx_ac97_suspend()+\n")); - if(card->states[1]) - { + if (card->states[1]) { stop_dac(card->states[1]); resync_dma_ptrs(card->states[1]); } - if(card->states[0]) - { + if (card->states[0]) { stop_adc(card->states[0]); resync_dma_ptrs(card->states[0]); } - for(Count = 0x2, i=0; (Count <= CS46XX_AC97_HIGHESTREGTORESTORE) - && (i < CS46XX_AC97_NUMBER_RESTORE_REGS); - Count += 2, i++) - { + for (Count = 0x2, i = 0; (Count <= CS46XX_AC97_HIGHESTREGTORESTORE) + && (i < CS46XX_AC97_NUMBER_RESTORE_REGS); + Count += 2, i++) { card->pm.ac97[i] = cs_ac97_get(dev, BA0_AC97_RESET + Count); } /* @@ -3522,11 +3347,10 @@ static void cs46xx_ac97_suspend(struct cs_card *card) * well, for now, only power down the DAC/ADC and MIXER VREFON components. * trouble with removing VREF. */ - if( (tmp = cs461x_powerdown(card, CS_POWER_DAC | CS_POWER_ADC | - CS_POWER_MIXVON, CS_TRUE )) ) - { + if ((tmp = cs461x_powerdown(card, CS_POWER_DAC | CS_POWER_ADC | + CS_POWER_MIXVON, CS_TRUE))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO - "cs46xx: cs46xx_ac97_suspend() failure (0x%x)\n",tmp) ); + "cs46xx: cs46xx_ac97_suspend() failure (0x%x)\n",tmp)); } CS_DBGOUT(CS_PM, 9, printk("cs46xx: cs46xx_ac97_suspend()-\n")); @@ -3566,16 +3390,13 @@ static void cs46xx_ac97_resume(struct cs_card *card) * Restore just the first set of registers, from register number * 0x02 to the register number that ulHighestRegToRestore specifies. */ - for( Count = 0x2, i=0; - (Count <= CS46XX_AC97_HIGHESTREGTORESTORE) - && (i < CS46XX_AC97_NUMBER_RESTORE_REGS); - Count += 2, i++) - { + for (Count = 0x2, i=0; (Count <= CS46XX_AC97_HIGHESTREGTORESTORE) && + (i < CS46XX_AC97_NUMBER_RESTORE_REGS); Count += 2, i++) { cs_ac97_set(dev, (u8)(BA0_AC97_RESET + Count), (u16)card->pm.ac97[i]); } /* Check if we have to init the amplifier */ - if(card->amp_init) + if (card->amp_init) card->amp_init(card); CS_DBGOUT(CS_PM, 9, printk("cs46xx: cs46xx_ac97_resume()-\n")); @@ -3585,30 +3406,27 @@ static void cs46xx_ac97_resume(struct cs_card *card) static int cs46xx_restart_part(struct cs_card *card) { struct dmabuf *dmabuf; + CS_DBGOUT(CS_PM | CS_FUNCTION, 4, printk( "cs46xx: cs46xx_restart_part()+\n")); - if(card->states[1]) - { + if (card->states[1]) { dmabuf = &card->states[1]->dmabuf; dmabuf->ready = 0; resync_dma_ptrs(card->states[1]); cs_set_divisor(dmabuf); - if(__prog_dmabuf(card->states[1])) - { + if (__prog_dmabuf(card->states[1])) { CS_DBGOUT(CS_PM | CS_ERROR, 1, printk("cs46xx: cs46xx_restart_part()- (-1) prog_dmabuf() dac error\n")); return -1; } cs_set_dac_rate(card->states[1], dmabuf->rate); } - if(card->states[0]) - { + if (card->states[0]) { dmabuf = &card->states[0]->dmabuf; dmabuf->ready = 0; resync_dma_ptrs(card->states[0]); cs_set_divisor(dmabuf); - if(__prog_dmabuf(card->states[0])) - { + if (__prog_dmabuf(card->states[0])) { CS_DBGOUT(CS_PM | CS_ERROR, 1, printk("cs46xx: cs46xx_restart_part()- (-1) prog_dmabuf() adc error\n")); return -1; @@ -3616,17 +3434,17 @@ static int cs46xx_restart_part(struct cs_card *card) cs_set_adc_rate(card->states[0], dmabuf->rate); } card->pm.flags |= CS46XX_PM_RESUMED; - if(card->states[0]) + if (card->states[0]) start_adc(card->states[0]); - if(card->states[1]) + if (card->states[1]) start_dac(card->states[1]); card->pm.flags |= CS46XX_PM_IDLE; card->pm.flags &= ~(CS46XX_PM_SUSPENDING | CS46XX_PM_SUSPENDED | CS46XX_PM_RESUMING | CS46XX_PM_RESUMED); - if(card->states[0]) + if (card->states[0]) wake_up(&card->states[0]->dmabuf.wait); - if(card->states[1]) + if (card->states[1]) wake_up(&card->states[1]->dmabuf.wait); CS_DBGOUT(CS_PM | CS_FUNCTION, 4, @@ -3634,20 +3452,19 @@ static int cs46xx_restart_part(struct cs_card *card) return 0; } - static void cs461x_reset(struct cs_card *card); static void cs461x_proc_stop(struct cs_card *card); static int cs46xx_suspend(struct cs_card *card, pm_message_t state) { unsigned int tmp; + CS_DBGOUT(CS_PM | CS_FUNCTION, 4, printk("cs46xx: cs46xx_suspend()+ flags=0x%x s=%p\n", (unsigned)card->pm.flags,card)); /* * check the current state, only suspend if IDLE */ - if(!(card->pm.flags & CS46XX_PM_IDLE)) - { + if (!(card->pm.flags & CS46XX_PM_IDLE)) { CS_DBGOUT(CS_PM | CS_ERROR, 2, printk("cs46xx: cs46xx_suspend() unable to suspend, not IDLE\n")); return 1; @@ -3679,13 +3496,11 @@ static int cs46xx_suspend(struct cs_card *card, pm_message_t state) tmp = cs461x_peek(card, BA1_CCTL); cs461x_poke(card, BA1_CCTL, tmp & 0xffff0000); - if(card->states[1]) - { + if (card->states[1]) { card->pm.dmabuf_swptr_play = card->states[1]->dmabuf.swptr; card->pm.dmabuf_count_play = card->states[1]->dmabuf.count; } - if(card->states[0]) - { + if (card->states[0]) { card->pm.dmabuf_swptr_capture = card->states[0]->dmabuf.swptr; card->pm.dmabuf_count_capture = card->states[0]->dmabuf.count; } @@ -3736,8 +3551,7 @@ static int cs46xx_resume(struct cs_card *card) CS_DBGOUT(CS_PM | CS_FUNCTION, 4, printk( "cs46xx: cs46xx_resume()+ flags=0x%x\n", (unsigned)card->pm.flags)); - if(!(card->pm.flags & CS46XX_PM_SUSPENDED)) - { + if (!(card->pm.flags & CS46XX_PM_SUSPENDED)) { CS_DBGOUT(CS_PM | CS_ERROR, 2, printk("cs46xx: cs46xx_resume() unable to resume, not SUSPENDED\n")); return 1; @@ -3747,10 +3561,8 @@ static int cs46xx_resume(struct cs_card *card) printpm(card); card->active_ctrl(card, 1); - for(i=0;i<5;i++) - { - if (cs_hardware_init(card) != 0) - { + for (i = 0; i < 5; i++) { + if (cs_hardware_init(card) != 0) { CS_DBGOUT(CS_PM | CS_ERROR, 4, printk( "cs46xx: cs46xx_resume()- ERROR in cs_hardware_init()\n")); mdelay(10 * cs_laptop_wait); @@ -3759,15 +3571,13 @@ static int cs46xx_resume(struct cs_card *card) } break; } - if(i>=4) - { + if (i >= 4) { CS_DBGOUT(CS_PM | CS_ERROR, 1, printk( "cs46xx: cs46xx_resume()- cs_hardware_init() failed, retried %d times.\n",i)); return 0; } - if(cs46xx_restart_part(card)) - { + if (cs46xx_restart_part(card)) { CS_DBGOUT(CS_PM | CS_ERROR, 4, printk( "cs46xx: cs46xx_resume(): cs46xx_restart_part() returned error\n")); } @@ -3835,7 +3645,7 @@ static u16 _cs_ac97_get(struct ac97_codec *dev, u8 reg) /* * Wait for the read to occur. */ - if(!(card->pm.flags & CS46XX_PM_IDLE)) + if (!(card->pm.flags & CS46XX_PM_IDLE)) loopcnt = 2000; else loopcnt = 500 * cs_laptop_wait; @@ -3866,7 +3676,7 @@ static u16 _cs_ac97_get(struct ac97_codec *dev, u8 reg) * Wait for the valid status bit to go active. */ - if(!(card->pm.flags & CS46XX_PM_IDLE)) + if (!(card->pm.flags & CS46XX_PM_IDLE)) loopcnt = 2000; else loopcnt = 1000; @@ -3885,7 +3695,7 @@ static u16 _cs_ac97_get(struct ac97_codec *dev, u8 reg) /* * Make sure we got valid status. */ - if (!( (tmp=cs461x_peekBA0(card, BA0_ACSTS)) & ACSTS_VSTS)) { + if (!((tmp = cs461x_peekBA0(card, BA0_ACSTS)) & ACSTS_VSTS)) { CS_DBGOUT(CS_ERROR, 2, printk(KERN_WARNING "cs46xx: AC'97 read problem (ACSTS_VSTS), reg = 0x%x val=0x%x 0xffff \n", reg, tmp)); @@ -3923,12 +3733,9 @@ static void cs_ac97_set(struct ac97_codec *dev, u8 reg, u16 val) spin_lock(&card->ac97_lock); - if(reg == AC97_CD_VOL) - { + if (reg == AC97_CD_VOL) val2 = _cs_ac97_get(dev, AC97_CD_VOL); - } - - + /* * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97 @@ -3970,8 +3777,7 @@ static void cs_ac97_set(struct ac97_codec *dev, u8 reg, u16 val) /* * Make sure the write completed. */ - if (cs461x_peekBA0(card, BA0_ACCTL) & ACCTL_DCV) - { + if (cs461x_peekBA0(card, BA0_ACCTL) & ACCTL_DCV) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: AC'97 write problem, reg = 0x%x, val = 0x%x\n", reg, val)); } @@ -3998,25 +3804,23 @@ static void cs_ac97_set(struct ac97_codec *dev, u8 reg, u16 val) /* CD mute change ? */ - if(reg==AC97_CD_VOL) - { + if (reg == AC97_CD_VOL) { /* Mute bit change ? */ - if((val2^val)&0x8000 || ((val2 == 0x1f1f || val == 0x1f1f) && val2 != val)) - { + if ((val2^val) & 0x8000 || + ((val2 == 0x1f1f || val == 0x1f1f) && val2 != val)) { /* This is a hack but its cleaner than the alternatives. Right now card->ac97_codec[0] might be NULL as we are still doing codec setup. This does an early assignment to avoid the problem if it occurs */ - if(card->ac97_codec[0]==NULL) - card->ac97_codec[0]=dev; + if (card->ac97_codec[0] == NULL) + card->ac97_codec[0] = dev; /* Mute on */ - if(val&0x8000 || val == 0x1f1f) + if (val & 0x8000 || val == 0x1f1f) card->amplifier_ctrl(card, -1); - else /* Mute off power on */ - { - if(card->amp_init) + else { /* Mute off power on */ + if (card->amp_init) card->amp_init(card); card->amplifier_ctrl(card, 1); } @@ -4024,46 +3828,41 @@ static void cs_ac97_set(struct ac97_codec *dev, u8 reg, u16 val) } } - /* OSS /dev/mixer file operation methods */ static int cs_open_mixdev(struct inode *inode, struct file *file) { - int i=0; + int i = 0; unsigned int minor = iminor(inode); - struct cs_card *card=NULL; + struct cs_card *card = NULL; struct list_head *entry; unsigned int tmp; CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, printk(KERN_INFO "cs46xx: cs_open_mixdev()+\n")); - list_for_each(entry, &cs46xx_devs) - { + list_for_each(entry, &cs46xx_devs) { card = list_entry(entry, struct cs_card, list); for (i = 0; i < NR_AC97; i++) if (card->ac97_codec[i] != NULL && card->ac97_codec[i]->dev_mixer == minor) goto match; } - if (!card) - { + if (!card) { CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, printk(KERN_INFO "cs46xx: cs46xx_open_mixdev()- -ENODEV\n")); return -ENODEV; } match: - if(!card->ac97_codec[i]) + if (!card->ac97_codec[i]) return -ENODEV; file->private_data = card->ac97_codec[i]; card->active_ctrl(card,1); - if(!CS_IN_USE(&card->mixer_use_cnt)) - { - if( (tmp = cs46xx_powerup(card, CS_POWER_MIXVON )) ) - { + if (!CS_IN_USE(&card->mixer_use_cnt)) { + if ((tmp = cs46xx_powerup(card, CS_POWER_MIXVON))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO - "cs46xx: cs_open_mixdev() powerup failure (0x%x)\n",tmp) ); + "cs46xx: cs_open_mixdev() powerup failure (0x%x)\n", tmp)); return -EIO; } } @@ -4077,7 +3876,7 @@ static int cs_open_mixdev(struct inode *inode, struct file *file) static int cs_release_mixdev(struct inode *inode, struct file *file) { unsigned int minor = iminor(inode); - struct cs_card *card=NULL; + struct cs_card *card = NULL; struct list_head *entry; int i; unsigned int tmp; @@ -4092,15 +3891,13 @@ static int cs_release_mixdev(struct inode *inode, struct file *file) card->ac97_codec[i]->dev_mixer == minor) goto match; } - if (!card) - { + if (!card) { CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, printk(KERN_INFO "cs46xx: cs46xx_open_mixdev()- -ENODEV\n")); return -ENODEV; } match: - if(!CS_DEC_AND_TEST(&card->mixer_use_cnt)) - { + if (!CS_DEC_AND_TEST(&card->mixer_use_cnt)) { CS_DBGOUT(CS_FUNCTION | CS_RELEASE, 4, printk(KERN_INFO "cs46xx: cs_release_mixdev()- no powerdown, usecnt>0\n")); card->active_ctrl(card, -1); @@ -4110,10 +3907,9 @@ match: /* * ok, no outstanding mixer opens, so powerdown. */ - if( (tmp = cs461x_powerdown(card, CS_POWER_MIXVON, CS_FALSE )) ) - { + if ((tmp = cs461x_powerdown(card, CS_POWER_MIXVON, CS_FALSE))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO - "cs46xx: cs_release_mixdev() powerdown MIXVON failure (0x%x)\n",tmp) ); + "cs46xx: cs_release_mixdev() powerdown MIXVON failure (0x%x)\n", tmp)); card->active_ctrl(card, -1); card->amplifier_ctrl(card, -1); return -EIO; @@ -4126,76 +3922,60 @@ match: } static int cs_ioctl_mixdev(struct inode *inode, struct file *file, unsigned int cmd, - unsigned long arg) + unsigned long arg) { - struct ac97_codec *codec = (struct ac97_codec *)file->private_data; - struct cs_card *card=NULL; + struct ac97_codec *codec = file->private_data; + struct cs_card *card = NULL; struct list_head *entry; unsigned long __user *p = (long __user *)arg; - #if CSDEBUG_INTERFACE int val; - if( (cmd == SOUND_MIXER_CS_GETDBGMASK) || + if ( (cmd == SOUND_MIXER_CS_GETDBGMASK) || (cmd == SOUND_MIXER_CS_SETDBGMASK) || (cmd == SOUND_MIXER_CS_GETDBGLEVEL) || (cmd == SOUND_MIXER_CS_SETDBGLEVEL) || - (cmd == SOUND_MIXER_CS_APM)) - { - switch(cmd) - { - + (cmd == SOUND_MIXER_CS_APM)) { + switch (cmd) { case SOUND_MIXER_CS_GETDBGMASK: return put_user(cs_debugmask, p); - case SOUND_MIXER_CS_GETDBGLEVEL: return put_user(cs_debuglevel, p); - case SOUND_MIXER_CS_SETDBGMASK: if (get_user(val, p)) return -EFAULT; cs_debugmask = val; return 0; - case SOUND_MIXER_CS_SETDBGLEVEL: if (get_user(val, p)) return -EFAULT; cs_debuglevel = val; return 0; - case SOUND_MIXER_CS_APM: if (get_user(val, p)) return -EFAULT; - if(val == CS_IOCTL_CMD_SUSPEND) - { - list_for_each(entry, &cs46xx_devs) - { + if (val == CS_IOCTL_CMD_SUSPEND) { + list_for_each(entry, &cs46xx_devs) { card = list_entry(entry, struct cs_card, list); cs46xx_suspend(card, PMSG_ON); } - } - else if(val == CS_IOCTL_CMD_RESUME) - { - list_for_each(entry, &cs46xx_devs) - { + } else if (val == CS_IOCTL_CMD_RESUME) { + list_for_each(entry, &cs46xx_devs) { card = list_entry(entry, struct cs_card, list); cs46xx_resume(card); } - } - else - { + } else { CS_DBGOUT(CS_ERROR, 1, printk(KERN_INFO "cs46xx: mixer_ioctl(): invalid APM cmd (%d)\n", val)); } return 0; - default: CS_DBGOUT(CS_ERROR, 1, printk(KERN_INFO - "cs46xx: mixer_ioctl(): ERROR unknown debug cmd\n") ); + "cs46xx: mixer_ioctl(): ERROR unknown debug cmd\n")); return 0; - } + } } #endif return codec->mixer_ioctl(codec, cmd, arg); @@ -4232,8 +4012,7 @@ static int __init cs_ac97_init(struct cs_card *card) codec->codec_read = cs_ac97_get; codec->codec_write = cs_ac97_set; - if (ac97_probe_codec(codec) == 0) - { + if (ac97_probe_codec(codec) == 0) { CS_DBGOUT(CS_FUNCTION | CS_INIT, 2, printk(KERN_INFO "cs46xx: cs_ac97_init()- codec number %d not found\n", num_ac97) ); @@ -4241,12 +4020,11 @@ static int __init cs_ac97_init(struct cs_card *card) break; } CS_DBGOUT(CS_FUNCTION | CS_INIT, 2, printk(KERN_INFO - "cs46xx: cs_ac97_init() found codec %d\n",num_ac97) ); + "cs46xx: cs_ac97_init() found codec %d\n",num_ac97)); eid = cs_ac97_get(codec, AC97_EXTENDED_ID); - if(eid==0xFFFF) - { + if (eid == 0xFFFF) { printk(KERN_WARNING "cs46xx: codec %d not present\n",num_ac97); ac97_release_codec(codec); break; @@ -4285,27 +4063,23 @@ static void cs461x_download_image(struct cs_card *card) { unsigned i, j, temp1, temp2, offset, count; unsigned char __iomem *pBA1 = ioremap(card->ba1_addr, 0x40000); - for( i=0; i < CLEAR__COUNT; i++) - { + for (i = 0; i < CLEAR__COUNT; i++) { offset = ClrStat[i].BA1__DestByteOffset; count = ClrStat[i].BA1__SourceSize; - for( temp1 = offset; temp1<(offset+count); temp1+=4 ) + for (temp1 = offset; temp1 < (offset + count); temp1 += 4) writel(0, pBA1+temp1); } - for(i=0; iac97_codec[0], AC97_POWER_CONTROL); - if (tmp & CS_AC97_POWER_CONTROL_MIXVOFF_ON) - { - if(!muted) - { + if (tmp & CS_AC97_POWER_CONTROL_MIXVOFF_ON) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp |= CS_AC97_POWER_CONTROL_MIXVOFF; cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); @@ -4492,16 +4259,14 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend * Check the status.. */ if (cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_MIXVOFF_ON) - { + CS_AC97_POWER_CONTROL_MIXVOFF_ON) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerdown MIXVOFF failed\n")); return 1; } } } - if(type & CS_POWER_MIXVON) - { + if (type & CS_POWER_MIXVON) { CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs461x_powerdown()+ MIXVON\n")); @@ -4509,15 +4274,13 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend * Power down the MIXER (VREF ON) on the AC97 card. */ tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (tmp & CS_AC97_POWER_CONTROL_MIXVON_ON) - { - if(!muted) - { + if (tmp & CS_AC97_POWER_CONTROL_MIXVON_ON) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp |= CS_AC97_POWER_CONTROL_MIXVON; - cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); + cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp); /* * Now, we wait until we sample a ready state. */ @@ -4540,30 +4303,26 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend * Check the status.. */ if (cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_MIXVON_ON) - { + CS_AC97_POWER_CONTROL_MIXVON_ON) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerdown MIXVON failed\n")); return 1; } } } - if(type & CS_POWER_ADC) - { + if (type & CS_POWER_ADC) { /* * Power down the ADC on the AC97 card. */ CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs461x_powerdown()+ ADC\n")); tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (tmp & CS_AC97_POWER_CONTROL_ADC_ON) - { - if(!muted) - { + if (tmp & CS_AC97_POWER_CONTROL_ADC_ON) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp |= CS_AC97_POWER_CONTROL_ADC; - cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); + cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp); /* * Now, we wait until we sample a ready state. @@ -4587,16 +4346,14 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend * Check the status.. */ if (cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_ADC_ON) - { + CS_AC97_POWER_CONTROL_ADC_ON) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerdown ADC failed\n")); return 1; } } } - if(type & CS_POWER_DAC) - { + if (type & CS_POWER_DAC) { /* * Power down the DAC on the AC97 card. */ @@ -4604,15 +4361,13 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs461x_powerdown()+ DAC\n")); tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (tmp & CS_AC97_POWER_CONTROL_DAC_ON) - { - if(!muted) - { + if (tmp & CS_AC97_POWER_CONTROL_DAC_ON) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp |= CS_AC97_POWER_CONTROL_DAC; - cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); + cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp); /* * Now, we wait until we sample a ready state. */ @@ -4635,8 +4390,7 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend * Check the status.. */ if (cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_DAC_ON) - { + CS_AC97_POWER_CONTROL_DAC_ON) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerdown DAC failed\n")); return 1; @@ -4644,7 +4398,7 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend } } tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if(muted) + if (muted) cs_mute(card, CS_FALSE); CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs461x_powerdown()- 0 tmp=0x%x\n",tmp)); @@ -4654,23 +4408,22 @@ static int cs461x_powerdown(struct cs_card *card, unsigned int type, int suspend static int cs46xx_powerup(struct cs_card *card, unsigned int type) { int count; - unsigned int tmp=0,muted=0; + unsigned int tmp = 0, muted = 0; CS_DBGOUT(CS_FUNCTION, 8, printk(KERN_INFO "cs46xx: cs46xx_powerup()+ type=0x%x\n",type)); /* * check for VREF and powerup if need to. */ - if(type & CS_POWER_MIXVON) + if (type & CS_POWER_MIXVON) type |= CS_POWER_MIXVOFF; - if(type & (CS_POWER_DAC | CS_POWER_ADC)) + if (type & (CS_POWER_DAC | CS_POWER_ADC)) type |= CS_POWER_MIXVON | CS_POWER_MIXVOFF; /* * Power up indicated areas. */ - if(type & CS_POWER_MIXVOFF) - { + if (type & CS_POWER_MIXVOFF) { CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs46xx_powerup()+ MIXVOFF\n")); @@ -4678,12 +4431,10 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) * Power up the MIXER (VREF ON) on the AC97 card. */ tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (!(tmp & CS_AC97_POWER_CONTROL_MIXVOFF_ON)) - { - if(!muted) - { + if (!(tmp & CS_AC97_POWER_CONTROL_MIXVOFF_ON)) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp &= ~CS_AC97_POWER_CONTROL_MIXVOFF; cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); @@ -4709,16 +4460,14 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) * Check the status.. */ if (!(cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_MIXVOFF_ON)) - { + CS_AC97_POWER_CONTROL_MIXVOFF_ON)) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerup MIXVOFF failed\n")); return 1; } } } - if(type & CS_POWER_MIXVON) - { + if(type & CS_POWER_MIXVON) { CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs46xx_powerup()+ MIXVON\n")); @@ -4726,12 +4475,10 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) * Power up the MIXER (VREF ON) on the AC97 card. */ tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (!(tmp & CS_AC97_POWER_CONTROL_MIXVON_ON)) - { - if(!muted) - { + if (!(tmp & CS_AC97_POWER_CONTROL_MIXVON_ON)) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp &= ~CS_AC97_POWER_CONTROL_MIXVON; cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); @@ -4757,27 +4504,23 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) * Check the status.. */ if (!(cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_MIXVON_ON)) - { + CS_AC97_POWER_CONTROL_MIXVON_ON)) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerup MIXVON failed\n")); return 1; } } } - if(type & CS_POWER_ADC) - { + if (type & CS_POWER_ADC) { /* * Power up the ADC on the AC97 card. */ CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs46xx_powerup()+ ADC\n")); tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (!(tmp & CS_AC97_POWER_CONTROL_ADC_ON)) - { - if(!muted) - { + if (!(tmp & CS_AC97_POWER_CONTROL_ADC_ON)) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp &= ~CS_AC97_POWER_CONTROL_ADC; cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); @@ -4804,16 +4547,14 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) * Check the status.. */ if (!(cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_ADC_ON)) - { + CS_AC97_POWER_CONTROL_ADC_ON)) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerup ADC failed\n")); return 1; } } } - if(type & CS_POWER_DAC) - { + if (type & CS_POWER_DAC) { /* * Power up the DAC on the AC97 card. */ @@ -4821,12 +4562,10 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs46xx_powerup()+ DAC\n")); tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if (!(tmp & CS_AC97_POWER_CONTROL_DAC_ON)) - { - if(!muted) - { + if (!(tmp & CS_AC97_POWER_CONTROL_DAC_ON)) { + if (!muted) { cs_mute(card, CS_TRUE); - muted=1; + muted = 1; } tmp &= ~CS_AC97_POWER_CONTROL_DAC; cs_ac97_set(card->ac97_codec[0], AC97_POWER_CONTROL, tmp ); @@ -4852,8 +4591,7 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) * Check the status.. */ if (!(cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL) & - CS_AC97_POWER_CONTROL_DAC_ON)) - { + CS_AC97_POWER_CONTROL_DAC_ON)) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_WARNING "cs46xx: powerup DAC failed\n")); return 1; @@ -4861,14 +4599,13 @@ static int cs46xx_powerup(struct cs_card *card, unsigned int type) } } tmp = cs_ac97_get(card->ac97_codec[0], AC97_POWER_CONTROL); - if(muted) + if (muted) cs_mute(card, CS_FALSE); CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO "cs46xx: cs46xx_powerup()- 0 tmp=0x%x\n",tmp)); return 0; } - static void cs461x_proc_start(struct cs_card *card) { int cnt; @@ -4965,7 +4702,7 @@ static int cs_hardware_init(struct cs_card *card) * is not enough for some platforms! tested on an IBM Thinkpads and * reference cards. */ - if(!(card->pm.flags & CS46XX_PM_IDLE)) + if (!(card->pm.flags & CS46XX_PM_IDLE)) mdelay(initdelay); /* * Write the selected clock control setup to the hardware. Do not turn on @@ -5017,8 +4754,7 @@ static int cs_hardware_init(struct cs_card *card) * If we are resuming under 2.2.x then we can not schedule a timeout. * so, just spin the CPU. */ - if(card->pm.flags & CS46XX_PM_IDLE) - { + if (card->pm.flags & CS46XX_PM_IDLE) { /* * Wait for the card ready signal from the AC97 card. */ @@ -5033,9 +4769,7 @@ static int cs_hardware_init(struct cs_card *card) current->state = TASK_UNINTERRUPTIBLE; schedule_timeout(1); } while (time_before(jiffies, end_time)); - } - else - { + } else { for (count = 0; count < 100; count++) { // First, we want to wait for a short time. udelay(25 * cs_laptop_wait); @@ -5064,8 +4798,7 @@ static int cs_hardware_init(struct cs_card *card) */ cs461x_pokeBA0(card, BA0_ACCTL, ACCTL_VFRM | ACCTL_ESYN | ACCTL_RSTN); - if(card->pm.flags & CS46XX_PM_IDLE) - { + if (card->pm.flags & CS46XX_PM_IDLE) { /* * Wait until we've sampled input slots 3 and 4 as valid, meaning that * the card is pumping ADC data across the AC-link. @@ -5081,9 +4814,7 @@ static int cs_hardware_init(struct cs_card *card) current->state = TASK_UNINTERRUPTIBLE; schedule_timeout(1); } while (time_before(jiffies, end_time)); - } - else - { + } else { for (count = 0; count < 100; count++) { // First, we want to wait for a short time. udelay(25 * cs_laptop_wait); @@ -5140,17 +4871,13 @@ static int cs_hardware_init(struct cs_card *card) cs461x_poke(card, BA1_CCTL, tmp & 0xffff0000); /* initialize AC97 codec and register /dev/mixer */ - if(card->pm.flags & CS46XX_PM_IDLE) - { - if (cs_ac97_init(card) <= 0) - { + if (card->pm.flags & CS46XX_PM_IDLE) { + if (cs_ac97_init(card) <= 0) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO - "cs46xx: cs_ac97_init() failure\n") ); + "cs46xx: cs_ac97_init() failure\n")); return -EIO; } - } - else - { + } else { cs46xx_ac97_resume(card); } @@ -5174,23 +4901,17 @@ static int cs_hardware_init(struct cs_card *card) * If IDLE then Power down the part. We will power components up * when we need them. */ - if(card->pm.flags & CS46XX_PM_IDLE) - { - if(!cs_powerdown) - { - if( (tmp = cs46xx_powerup(card, CS_POWER_DAC | CS_POWER_ADC | - CS_POWER_MIXVON )) ) - { + if (card->pm.flags & CS46XX_PM_IDLE) { + if (!cs_powerdown) { + if ((tmp = cs46xx_powerup(card, CS_POWER_DAC | CS_POWER_ADC | + CS_POWER_MIXVON))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO "cs46xx: cs461x_powerup() failure (0x%x)\n",tmp) ); return -EIO; } - } - else - { - if( (tmp = cs461x_powerdown(card, CS_POWER_DAC | CS_POWER_ADC | - CS_POWER_MIXVON, CS_FALSE )) ) - { + } else { + if ((tmp = cs461x_powerdown(card, CS_POWER_DAC | CS_POWER_ADC | + CS_POWER_MIXVON, CS_FALSE))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO "cs46xx: cs461x_powerdown() failure (0x%x)\n",tmp) ); return -EIO; @@ -5310,14 +5031,13 @@ MODULE_AUTHOR("Alan Cox , Jaroslav Kysela, name) { - if(cp->vendor == ss_vendor && cp->id == ss_card) - { + if (cp->vendor == ss_vendor && cp->id == ss_card) { card->amplifier_ctrl = cp->amp; - if(cp->active) + if (cp->active) card->active_ctrl = cp->active; - if(cp->amp_init) + if (cp->amp_init) card->amp_init = cp->amp_init; break; } cp++; } - if (cp->name==NULL) - { + if (cp->name == NULL) { printk(KERN_INFO "cs46xx: Unknown card (%04X:%04X) at 0x%08lx/0x%08lx, IRQ %d\n", ss_vendor, ss_card, card->ba0_addr, card->ba1_addr, card->irq); - } - else - { + } else { printk(KERN_INFO "cs46xx: %s (%04X:%04X) at 0x%08lx/0x%08lx, IRQ %d\n", cp->name, ss_vendor, ss_card, card->ba0_addr, card->ba1_addr, card->irq); } - if (card->amplifier_ctrl==NULL) - { + if (card->amplifier_ctrl == NULL) { card->amplifier_ctrl = amp_none; card->active_ctrl = clkrun_hack; } - if (external_amp == 1) - { + if (external_amp == 1) { printk(KERN_INFO "cs46xx: Crystal EAPD support forced on.\n"); card->amplifier_ctrl = amp_voyetra; } - if (thinkpad == 1) - { + if (thinkpad == 1) { printk(KERN_INFO "cs46xx: Activating CLKRUN hack for Thinkpad.\n"); card->active_ctrl = clkrun_hack; } @@ -5425,13 +5138,11 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, * and mdelay kernel code is replaced by a pm timer, or the delays * work well for battery and/or AC power both. */ - if(card->active_ctrl == clkrun_hack) - { + if (card->active_ctrl == clkrun_hack) { initdelay = 2100; cs_laptop_wait = 5; } - if((card->active_ctrl == clkrun_hack) && !(powerdown == 1)) - { + if ((card->active_ctrl == clkrun_hack) && !(powerdown == 1)) { /* * for some currently unknown reason, powering down the DAC and ADC component * blocks on thinkpads causes some funky behavior... distoorrrtion and ac97 @@ -5440,7 +5151,7 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, */ cs_powerdown = 0; } - if(powerdown == 0) + if (powerdown == 0) cs_powerdown = 0; card->active_ctrl(card, 1); @@ -5461,7 +5172,7 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, card->ba1.name.pmem, card->ba1.name.reg) ); - if(card->ba0 == 0 || card->ba1.name.data0 == 0 || + if (card->ba0 == 0 || card->ba1.name.data0 == 0 || card->ba1.name.data1 == 0 || card->ba1.name.pmem == 0 || card->ba1.name.reg == 0) goto fail2; @@ -5477,14 +5188,12 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, } /* register /dev/midi */ - if((card->dev_midi = register_sound_midi(&cs_midi_fops, -1)) < 0) + if ((card->dev_midi = register_sound_midi(&cs_midi_fops, -1)) < 0) printk(KERN_ERR "cs46xx: unable to register midi\n"); card->pm.flags |= CS46XX_PM_IDLE; - for(i=0;i<5;i++) - { - if (cs_hardware_init(card) != 0) - { + for (i = 0; i < 5; i++) { + if (cs_hardware_init(card) != 0) { CS_DBGOUT(CS_ERROR, 4, printk( "cs46xx: ERROR in cs_hardware_init()... retrying\n")); for (j = 0; j < NR_AC97; j++) @@ -5497,12 +5206,11 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, } break; } - if(i>=4) - { + if(i >= 4) { CS_DBGOUT(CS_PM | CS_ERROR, 1, printk( "cs46xx: cs46xx_probe()- cs_hardware_init() failed, retried %d times.\n",i)); unregister_sound_dsp(card->dev_audio); - if(card->dev_midi) + if (card->dev_midi) unregister_sound_midi(card->dev_midi); goto fail; } @@ -5518,7 +5226,7 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, * Check if we have to init the amplifier, but probably already done * since the CD logic in the ac97 init code will turn on the ext amp. */ - if(cp->amp_init) + if (cp->amp_init) cp->amp_init(card); card->active_ctrl(card, -1); @@ -5536,15 +5244,15 @@ static int __devinit cs46xx_probe(struct pci_dev *pci_dev, fail: free_irq(card->irq, card); fail2: - if(card->ba0) + if (card->ba0) iounmap(card->ba0); - if(card->ba1.name.data0) + if (card->ba1.name.data0) iounmap(card->ba1.name.data0); - if(card->ba1.name.data1) + if (card->ba1.name.data1) iounmap(card->ba1.name.data1); - if(card->ba1.name.pmem) + if (card->ba1.name.pmem) iounmap(card->ba1.name.pmem); - if(card->ba1.name.reg) + if (card->ba1.name.reg) iounmap(card->ba1.name.reg); kfree(card); CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_INFO @@ -5598,9 +5306,8 @@ static void __devexit cs46xx_remove(struct pci_dev *pci_dev) * Power down the DAC and ADC. We will power them up (if) when we need * them. */ - if( (tmp = cs461x_powerdown(card, CS_POWER_DAC | CS_POWER_ADC | - CS_POWER_MIXVON, CS_TRUE )) ) - { + if ((tmp = cs461x_powerdown(card, CS_POWER_DAC | CS_POWER_ADC | + CS_POWER_MIXVON, CS_TRUE))) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk(KERN_INFO "cs46xx: cs461x_powerdown() failure (0x%x)\n",tmp) ); } @@ -5634,7 +5341,7 @@ static void __devexit cs46xx_remove(struct pci_dev *pci_dev) ac97_release_codec(card->ac97_codec[i]); } unregister_sound_dsp(card->dev_audio); - if(card->dev_midi) + if (card->dev_midi) unregister_sound_midi(card->dev_midi); list_del(&card->list); kfree(card); @@ -5693,8 +5400,7 @@ static int __init cs46xx_init_module(void) "cs46xx: cs46xx_init_module()+ \n")); rtn = pci_register_driver(&cs46xx_pci_driver); - if(rtn == -ENODEV) - { + if (rtn == -ENODEV) { CS_DBGOUT(CS_ERROR | CS_INIT, 1, printk( "cs46xx: Unable to detect valid cs46xx device\n")); } -- cgit v1.2.2