From a28287925555c93984115d37a1a25315ea369764 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 19 Jan 2011 12:55:28 +0000 Subject: ASoC: WM8995: Fix incorrect use of snd_soc_update_bits() In the wm8995_set_tristate() function when updating the register bits use the value and not the register index as the value argument. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8995.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 6045cbde492b..608c84c5aa8e 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } /* The size in bits of the FLL divide multiplied by 10 -- cgit v1.2.2 From 78b3fb46753872fc79bffecc8d50355a8622b39b Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 19 Jan 2011 19:10:47 +0800 Subject: ASoC: WM8994: fix wrong value in tristate function fix wrong value in wm8994_set_tristate func. when updating reg bits, it should use "value", not "reg". Signed-off-by: Qiao Zhou Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb8..3351f77607b3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2386,7 +2386,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 -- cgit v1.2.2 From dc5a460a1bfa44273653700e33d4e7051194cbfd Mon Sep 17 00:00:00 2001 From: "Rajashekhara, Sudhakar" Date: Fri, 21 Jan 2011 20:10:01 +0530 Subject: ASoC: da8xx/omap-l1xx: match codec_name with i2c ids The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c is not matching with the i2c ids in the board file. Without this fix the soundcard does not get detected on da850/omap-l138/am18x evm. Signed-off-by: Rajashekhara, Sudhakar Tested-by: Dan Sharon Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org (for 2.6.37) --- sound/soc/davinci/davinci-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 0c2d6bacc681..b36f0b39b090 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, -- cgit v1.2.2 From 20a4e7fc7e213365ea3771d7bf1e10a6bab853be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 21 Jan 2011 12:47:33 +0000 Subject: ASoC: Handle low measured DC offsets for wm_hubs devices The DC servo codes are actually signed numbers so need to be treated as such. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm_hubs.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c466982eed23..613df5db0b32 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + s8 offset; u16 reg, reg_l, reg_r, dcs_cfg; /* If we're using a digital only path and have a previously @@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes); /* HPOUT1L */ - if (reg_l + hubs->dcs_codes > 0 && - reg_l + hubs->dcs_codes < 0xff) - reg_l += hubs->dcs_codes; - dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + offset = reg_l; + offset += hubs->dcs_codes; + dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - if (reg_r + hubs->dcs_codes > 0 && - reg_r + hubs->dcs_codes < 0xff) - reg_r += hubs->dcs_codes; - dcs_cfg |= reg_r; + offset = reg_r; + offset += hubs->dcs_codes; + dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); -- cgit v1.2.2 From 02b6b5b640e773eb4d4d0685fa6c1fbc660b2834 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 21 Jan 2011 13:27:39 +0100 Subject: ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx Four very similar procedures - one for each model - now refactored into one. This isn't all duplicated code, but a step in the right direction. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 81 ++++++++++++------------------------------ 1 file changed, 23 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9bb030a469cd..7cd59b9f0e97 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2387,79 +2387,53 @@ static void cxt5066_hp_automute(struct hda_codec *codec) cxt5066_update_speaker(codec); } -/* unsolicited event for jack sensing */ -static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) +/* Dispatch the right mic autoswitch function */ +static void cxt5066_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - /* ignore mic events in DC mode; we're always using the jack */ - if (!spec->dc_enable) - cxt5066_olpc_automic(codec); - break; - } -} -/* unsolicited event for jack sensing */ -static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + if (spec->dell_vostro) cxt5066_vostro_automic(codec); - break; - } -} - -/* unsolicited event for jack sensing */ -static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + else if (spec->ideapad) cxt5066_ideapad_automic(codec); - break; - } + else if (spec->thinkpad) + cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } /* unsolicited event for jack sensing */ -static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + struct conexant_spec *spec = codec->spec; + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_hp_laptop_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } /* unsolicited event for jack sensing */ -static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_thinkpad_automic(codec); + cxt5066_automic(codec); break; } } + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -3039,20 +3013,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: init\n"); conexant_init(codec); if (codec->patch_ops.unsol_event) { cxt5066_hp_automute(codec); - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); + cxt5066_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3169,7 +3134,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; @@ -3207,7 +3172,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_DELL_VOSTRO: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_vostro_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; @@ -3224,7 +3189,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_IDEAPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_ideapad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_ideapad; @@ -3240,7 +3205,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_THINKPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_thinkpad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; -- cgit v1.2.2 From a1d6906e2d2b4655e248f490ab088c27876a600a Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 21 Jan 2011 13:33:28 +0100 Subject: ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7cd59b9f0e97..19f0daf6497d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -127,6 +127,7 @@ struct conexant_spec { unsigned int ideapad:1; unsigned int thinkpad:1; unsigned int hp_laptop:1; + unsigned int asus:1; unsigned int ext_mic_present; unsigned int recording; @@ -2312,6 +2313,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_asus_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 0); +} + + /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_hp_laptop_automic(struct hda_codec *codec) { @@ -2400,6 +2414,8 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); + else if (spec->asus) + cxt5066_asus_automic(codec); } /* unsolicited event for jack sensing */ @@ -3045,6 +3061,7 @@ enum { CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3056,6 +3073,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", }; @@ -3068,6 +3086,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3077,6 +3096,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3132,13 +3152,15 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_ASUS: case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; - spec->hp_laptop = 1; + spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; + spec->asus = board_config == CXT5066_ASUS; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ -- cgit v1.2.2 From f6a2491ca23d26d829730e33fbdd9e44fc5d1d53 Mon Sep 17 00:00:00 2001 From: Andy Robinson Date: Mon, 24 Jan 2011 10:12:37 -0500 Subject: ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct SPDIF output Changed the Asus A52J quirk to use the asus model instead of the hp_laptop model, which fixes the external mic input. Added an Asus U50F quirk to use the asus model. For the cxt5066 codecs, added checking of the digital output pins to determine which digital output nodes to use instead of always using node 0x21, since some systems have node 0x12 connected to a SPDIF out jack. [A slight modification for better readability by tiwai] Signed-off-by: Andy Robinson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 35 +++++++++++++++++++++++++++++++---- 1 file changed, 31 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 19f0daf6497d..9867afc7895e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -85,6 +85,7 @@ struct conexant_spec { unsigned int auto_mic; int auto_mic_ext; /* autocfg.inputs[] index for ext mic */ unsigned int need_dac_fix; + hda_nid_t slave_dig_outs[2]; /* capture */ unsigned int num_adc_nids; @@ -353,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; } + if (spec->slave_dig_outs[0]) + codec->slave_dig_outs = spec->slave_dig_outs; } return 0; @@ -2101,7 +2104,7 @@ static int patch_cxt5051(struct hda_codec *codec) static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; -#define CXT5066_SPDIF_OUT 0x21 +static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; /* OLPC's microphone port is DC coupled for use with external sensors, * therefore we use a 50% mic bias in order to center the input signal with @@ -2623,6 +2626,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) spec->recording = 0; } +static void conexant_check_dig_outs(struct hda_codec *codec, + hda_nid_t *dig_pins, + int num_pins) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t *nid_loc = &spec->multiout.dig_out_nid; + int i; + + for (i = 0; i < num_pins; i++, dig_pins++) { + unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins); + if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE) + continue; + if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1) + continue; + if (spec->slave_dig_outs[0]) + nid_loc++; + else + nid_loc = spec->slave_dig_outs; + } +} + static struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { @@ -3085,8 +3109,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), + SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3118,7 +3143,8 @@ static int patch_cxt5066(struct hda_codec *codec) spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); spec->multiout.dac_nids = cxt5066_dac_nids; - spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT; + conexant_check_dig_outs(codec, cxt5066_digout_pin_nids, + ARRAY_SIZE(cxt5066_digout_pin_nids)); spec->num_adc_nids = 1; spec->adc_nids = cxt5066_adc_nids; spec->capsrc_nids = cxt5066_capsrc_nids; @@ -3164,7 +3190,8 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; + if (board_config == CXT5066_HP_LAPTOP) + spec->multiout.dig_out_nid = 0; /* input source automatically selected */ spec->input_mux = NULL; spec->port_d_mode = 0; -- cgit v1.2.2 From c9ba374d24882c04e7cc000d8cf3b0fe56511b84 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Tue, 25 Jan 2011 06:46:31 +0100 Subject: ALSA: azt3328 - fix broken AZF_FMT_XLATE macro Cleanly revert to non-macro implementation of snd_azf3328_codec_setfmt(), to fix last-minute functionality breakage induced by following checkpatch.pl recommendations without giving them their due full share of thought ("revolting computer, ensuing PEBKAC"). I would like to thank Jiri Slaby for his very timely (in -rc1 even) and unexpected (uncommon hardware) "recognition of the dangerous situation" due to his very commendable static parser use. :) Reported-by: Jiri Slaby Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 38 ++++++++++++++++---------------------- 1 file changed, 16 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 6117595fc075..573594bf3225 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, snd_azf3328_dbgcallenter(); switch (bitrate) { -#define AZF_FMT_XLATE(in_freq, out_bits) \ - do { \ - case AZF_FREQ_ ## in_freq: \ - freq = SOUNDFORMAT_FREQ_ ## out_bits; \ - break; \ - } while (0); - AZF_FMT_XLATE(4000, SUSPECTED_4000) - AZF_FMT_XLATE(4800, SUSPECTED_4800) - /* the AZF3328 names it "5510" for some strange reason: */ - AZF_FMT_XLATE(5512, 5510) - AZF_FMT_XLATE(6620, 6620) - AZF_FMT_XLATE(8000, 8000) - AZF_FMT_XLATE(9600, 9600) - AZF_FMT_XLATE(11025, 11025) - AZF_FMT_XLATE(13240, SUSPECTED_13240) - AZF_FMT_XLATE(16000, 16000) - AZF_FMT_XLATE(22050, 22050) - AZF_FMT_XLATE(32000, 32000) + case AZF_FREQ_4000: freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break; + case AZF_FREQ_4800: freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break; + case AZF_FREQ_5512: + /* the AZF3328 names it "5510" for some strange reason */ + freq = SOUNDFORMAT_FREQ_5510; break; + case AZF_FREQ_6620: freq = SOUNDFORMAT_FREQ_6620; break; + case AZF_FREQ_8000: freq = SOUNDFORMAT_FREQ_8000; break; + case AZF_FREQ_9600: freq = SOUNDFORMAT_FREQ_9600; break; + case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break; + case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break; + case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break; + case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break; + case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); /* fall-through */ - AZF_FMT_XLATE(44100, 44100) - AZF_FMT_XLATE(48000, 48000) - AZF_FMT_XLATE(66200, SUSPECTED_66200) -#undef AZF_FMT_XLATE + case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; + case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; + case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break; } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ -- cgit v1.2.2 From 81d7da5404aad930a4e4f6111e4f16b752183018 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 24 Jan 2011 22:09:22 +0100 Subject: ASoC: Fix codec device id format used by some dai_links The id part of an I2C device name is created with the "%d-%04x" format string. So for example for an I2C device which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- 6 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e06b0d..5e4d499d8434 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24c1856..ad28663f5bbd 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 3eec610c10f9..9e05e10b86a6 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -401,7 +401,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a24514beb5..cf69e146890b 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index bb4292e3596c..287a97164ab4 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -94,7 +94,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index fbba4e367729..d2b14ba6f9e3 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -85,7 +85,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", -- cgit v1.2.2 From 518aa59f6e45b3c90b849187ae1d56757d074b92 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 24 Jan 2011 22:12:42 +0100 Subject: ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s During the multi-component patch the s3c24xx i2s driver was renamed from "s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not updated to reflect this change as well. As a result there is no match between the dai_link and the i2s driver and no sound card is instantiated. This patch fixes the problem by updating the sound board drivers to use "s3c24xx-iis" for the cpu_dai_name. Signed-off-by: Lars-Peter Clausen Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/samsung/neo1973_gta02_wm8753.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 9e05e10b86a6..0d0ae2b9eef6 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -397,7 +397,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", .stream_name = "WM8753 HiFi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index cf69e146890b..d20815d5ab2e 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -559,7 +559,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .name = "WM8753", .stream_name = "WM8753 HiFi", .platform_name = "samsung-audio", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 287a97164ab4..08fcaaa66907 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -95,7 +95,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", .codec_name = "tlv320aic3x-codec.0-001a", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_hermes_init, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index d2b14ba6f9e3..116e3e670167 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_name = "tlv320aic3x-codec.0-001a", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_tlv320aic23_init, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index cdc8ecbcb8ef..2c09e93dd566 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .stream_name = "UDA134X", .codec_name = "uda134x-hifi", .codec_dai_name = "uda134x-hifi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, .platform_name = "samsung-audio", }; -- cgit v1.2.2 From a3adfa00e8089aa72826c6ba04bcb18cfceaf0a9 Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Fri, 21 Jan 2011 22:14:17 +0300 Subject: ASoC: correct link specifications for corgi, poodle and spitz ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms contained incorrect names for cpu_dai and codec, which effectievly disabled sound on theese platforms. Fix that errors. Signed-off-by: Dmitry Eremin-Solenikov Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index fc592f0d5fc7..784cff5f67e8 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai_name = "pxa-is2-dai", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001a", + .codec_name = "wm8731-codec-0.001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 6298ee115e27..a7d4999f9b24 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .codec_name = "wm8731-codec.0-001b", .init = poodle_wm8731_init, .ops = &poodle_ops, }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2acb69b957a..8e1571350630 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link spitz_dai = { .name = "wm8750", .stream_name = "WM8750", - .cpu_dai_name = "pxa-is2", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750-codec.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; -- cgit v1.2.2 From fd76804f3f5484b35e6a51214c91e916ebba05aa Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Mon, 24 Jan 2011 16:09:56 +0100 Subject: ALSA: fix invalid hardware.h include in ac97c for AVR32 architecture This patch fixes the non-compiling AC97C driver for AVR32 architecture by include mach/hardware.h only for AT91 architecture. The AVR32 architecture does not supply the hardware.h include file. Signed-off-by: Hans-Christian Egtvedt CC: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 10c3a871a12d..b310702c646e 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -33,9 +33,12 @@ #include #include -#include #include +#ifdef CONFIG_ARCH_AT91 +#include +#endif + #include "ac97c.h" enum { -- cgit v1.2.2 From d757534ed15387202e322854cd72dc58bbb975de Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 25 Jan 2011 19:44:26 +0100 Subject: ALSA: HDA: Fix dmesg output of HDMI supported bits This typo caused the dmesg output of the supported bits of HDMI to be cut off early. Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 4a663471dadc..74b0560289c0 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) snd_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) - snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); + snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); else if (a->max_bitrate) snprintf(buf2, sizeof(buf2), ", max bitrate = %d", a->max_bitrate); -- cgit v1.2.2 From ded9f5238bb719737f82b0b5b957937cb0203804 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 26 Jan 2011 11:46:12 +0100 Subject: ALSA: HDA: Fix automute on Thinkpad L412/L512 BugLink: http://bugs.launchpad.net/bugs/707902 More Thinkpad machines with invalid SKU found, that disables automute between speakers and headphones on these machines. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index be4df4c6fd56..2fa9ed99c32f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14954,9 +14954,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), - SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), {} -- cgit v1.2.2 From c73e0c83f512012e7c357e516a0d7c0a832bfa34 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 26 Jan 2011 16:39:37 +0200 Subject: ASoC: Fix module refcount for auxiliary devices Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec driver refcount increments from soc_bind_dai_link into soc_probe_codec. However, the commit didn't remove try_module_get from soc_probe_aux_dev so the auxiliary device reference counts are incremented twice as the soc_probe_codec is called from soc_probe_aux_dev too. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291b6ff6..c4b60610beb0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) goto out; found: - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - ret = soc_probe_codec(card, codec); if (ret < 0) return ret; -- cgit v1.2.2 From 195938753951e70e85303301c37906c7ad72645e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 27 Jan 2011 10:28:46 +0100 Subject: ALSA: HDA: Fix microphone(s) on Lenovo Edge 13 BugLink: http://bugs.launchpad.net/bugs/708521 This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9867afc7895e..7e1ca43bd660 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3120,6 +3120,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ -- cgit v1.2.2 From 0fa63b69284c9bbedf876c677a9e650243cc40be Mon Sep 17 00:00:00 2001 From: "Manjunathappa, Prakash" Date: Thu, 27 Jan 2011 19:17:43 +0530 Subject: ASoC: DaVinci: fix kernel panic due to uninitialized platform_data This patch fixes the Kernel panic issue on accessing davinci_vc in cq93vc_probe function. struct davinci_vc is part of platform device's private driver data(codec->dev->p->driver_data) and this is populated by DaVinci Voice Codec MFD driver. Signed-off-by: Manjunathappa, Prakash Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd067f79..347a567b01e1 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; -- cgit v1.2.2 From e9cf7049330cd44c8af43b0c5c7bef25a086c5b7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 27 Jan 2011 14:54:05 -0700 Subject: ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw() snd_soc_dapm_put_volsw() has variables for both the unshifted and shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in the middle of DAPM sequences) got confused between the two of these. Since there's no need to keep a copy of the unshifted mask fix this and simplify the code by using only one mask variable. [Completely rewrote the changelog to describe the issue -- broonie.] Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 499730ab5638..8194f150bab7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val_mask; + unsigned int val; int connect, change; struct snd_soc_dapm_update update; @@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - val_mask = mask << shift; + mask = mask << shift; val = val << shift; mutex_lock(&widget->codec->mutex); widget->value = val; - change = snd_soc_test_bits(widget->codec, reg, val_mask, val); + change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { if (val) /* new connection */ -- cgit v1.2.2 From fdbc5d1b32e195b7775e103abd6263370f11af11 Mon Sep 17 00:00:00 2001 From: Amerigo Wang Date: Fri, 28 Jan 2011 16:52:00 +0800 Subject: sound: silent echo'ed messages in Makefile Silent these echo's, please. Signed-off-by: WANG Cong Signed-off-by: Takashi Iwai --- sound/oss/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 96f14dcd0cd1..90ffb99c6b17 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y) $(obj)/bin2hex pss_synth < $< > $@ else $(obj)/pss_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * pss_synth = NULL;'; \ echo 'static int pss_synthLen = 0;'; \ ) > $@ @@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y) $(obj)/hex2hex -i trix_boot < $< > $@ else $(obj)/trix_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * trix_boot = NULL;'; \ echo 'static int trix_boot_len = 0;'; \ ) > $@ -- cgit v1.2.2 From efbeb0718126d277c9d7e902eec8da956acf4bd6 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 31 Jan 2011 11:47:52 +0100 Subject: ALSA: oxygen: fix output routing on Xonar DG This card uses separate I2S outputs for the front speakers and headphones, and reverses the order of the three speaker outputs. To work around this, add a model-specific callback to adjust the controller's playback routing. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 2 ++ sound/pci/oxygen/oxygen_mixer.c | 2 ++ sound/pci/oxygen/xonar_dg.c | 36 ++++++++++++++++++++++++++++++++++++ 3 files changed, 40 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index c2ae63d17cd2..f53897a708b4 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -92,6 +92,8 @@ struct oxygen_model { void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed); + unsigned int (*adjust_dac_routing)(struct oxygen *chip, + unsigned int play_routing); void (*gpio_changed)(struct oxygen *chip); void (*uart_input)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 9bff14d5895d..26c7e8bcb229 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip) (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + if (chip->model.adjust_dac_routing) + reg_value = chip->model.adjust_dac_routing(chip, reg_value); oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, OXYGEN_PLAY_DAC0_SOURCE_MASK | OXYGEN_PLAY_DAC1_SOURCE_MASK | diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index e1fa602eba79..bc6eb58be380 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -24,6 +24,11 @@ * * SPI 0 -> CS4245 * + * I²S 1 -> CS4245 + * I²S 2 -> CS4361 (center/LFE) + * I²S 3 -> CS4361 (surround) + * I²S 4 -> CS4361 (front) + * * GPIO 3 <- ? * GPIO 4 <- headphone detect * GPIO 5 -> route input jack to line-in (0) or mic-in (1) @@ -36,6 +41,7 @@ * input 1 <- aux * input 2 <- front mic * input 4 <- line/mic + * DAC out -> headphones * aux out -> front panel headphones */ @@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip, cs4245_write_cached(chip, CS4245_ADC_CTRL, value); } +static inline unsigned int shift_bits(unsigned int value, + unsigned int shift_from, + unsigned int shift_to, + unsigned int mask) +{ + if (shift_from < shift_to) + return (value << (shift_to - shift_from)) & mask; + else + return (value >> (shift_from - shift_to)) & mask; +} + +static unsigned int adjust_dg_dac_routing(struct oxygen *chip, + unsigned int play_routing) +{ + return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC0_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_MASK); +} + static int output_switch_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = { .resume = dg_resume, .set_dac_params = set_cs4245_dac_params, .set_adc_params = set_cs4245_adc_params, + .adjust_dac_routing = adjust_dg_dac_routing, .dump_registers = dump_cs4245_registers, .model_data_size = sizeof(struct dg), .device_config = PLAYBACK_0_TO_I2S | -- cgit v1.2.2 From acd62276773b46810a3292af0c915c9782138ff2 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Tue, 1 Feb 2011 11:11:55 +0100 Subject: ASoC: Amstrad Delta: fix const related build error The Amstrad Delta ASoC driver used to override the digital_mute() callback, expected to be not provided by the on-board CX20442 CODEC driver, with its own implementation. While this is still posssible when substituting the whole empty snd_soc_dai_driver.ops member (the CX20442 case), replacing snd_soc_dai_ops.digital_mute only is no longer correct after the snd_soc_dai_driver.ops member has been constified, and results in build error. Drop this actually not used code path in hope the CX20442 driver never provides its own snd_soc_dai_ops structure. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 2101bdcee21f..3167be689621 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; - } else if (!codec_dai->driver->ops->digital_mute) { - codec_dai->driver->ops->digital_mute = ams_delta_digital_mute; } else { ams_delta_ops.startup = ams_delta_startup; ams_delta_ops.shutdown = ams_delta_shutdown; -- cgit v1.2.2 From f019ee5feb344ff0b22b58df4568676295aae14f Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Tue, 1 Feb 2011 13:01:17 +0100 Subject: ASoC: CX20442: fix NULL pointer dereference The CX20442 codec driver never provided the snd_soc_codec_driver's .reg_cache_default member. With the latest ASoC framework changes, it seems to be referred unconditionally, resulting in a NULL pointer dereference if missing. Provide it. Created and tested on Amstrad Delta against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 03d1e860d229..bb4bf65b9e7e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) return 0; } +static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; + static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache, -- cgit v1.2.2 From 70f7db11c45a313b23922cacf248c613c3b2144c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Feb 2011 17:16:38 +0100 Subject: ALSA: hda - Fix memory leaks in conexant jack arrays The Conexant codec driver adds the jack arrays in init callback which may be called also in each PM resume. This results in the addition of new jack element at each time. The fix is to check whether the requested jack is already present in the array. Reference: Novell bug 668929 https://bugzilla.novell.com/show_bug.cgi?id=668929 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7e1ca43bd660..fbe97d32140d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -407,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec, struct conexant_spec *spec; struct conexant_jack *jack; const char *name; - int err; + int i, err; spec = codec->spec; snd_array_init(&spec->jacks, sizeof(*jack), 32); + + jack = spec->jacks.list; + for (i = 0; i < spec->jacks.used; i++, jack++) + if (jack->nid == nid) + return 0 ; /* already present */ + jack = snd_array_new(&spec->jacks); name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; -- cgit v1.2.2 From ddfb319926462fd9670b7c1678a1f6a14a68e421 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Feb 2011 17:49:53 +0100 Subject: ALSA: use linux/io.h to fix compile warnings For helping to reduce Greert's regression list... src/sound/drivers/mtpav.c: error: implicit declaration of function 'inb' src/sound/drivers/mtpav.c: error: implicit declaration of function 'outb' ... Signed-off-by: Takashi Iwai --- sound/drivers/mtpav.c | 3 +-- sound/pcmcia/pdaudiocf/pdaudiocf.h | 2 +- sound/pcmcia/vx/vxp_ops.c | 2 +- 3 files changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index da03597fc893..5c426df87678 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -55,14 +55,13 @@ #include #include #include +#include #include #include #include #include #include -#include - /* * globals */ diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index bd26e092aead..6ce9ad700290 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -22,7 +22,7 @@ #define __PDAUDIOCF_H #include -#include +#include #include #include #include diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 989e04abb520..fe33e122e372 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -23,8 +23,8 @@ #include #include #include +#include #include -#include #include "vxpocket.h" -- cgit v1.2.2 From 0962bb217ac74c4b8fae34c5367ebc63131c962c Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 2 Feb 2011 21:11:41 +0100 Subject: ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init() The .card member of the snd_soc_pcm_runtime structure pointed to by the snd_soc_dai_link.init() argument used to be initialized before the function being called. This has changed, probably unintentionally, after recent refactorings. Since the function implementations are free to make use of this pointer, move its assignment back before the function is called to avoid NULL pointer dereferences. Created and tested on Amstrad Delta againts linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c4b60610beb0..c3f6f1e72790 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd = &card->rtd_aux[num]; name = aux_dev->name; } + rtd->card = card; /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; @@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->card = card; rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; -- cgit v1.2.2 From f9eb9dd14c2ca2a1f8d979637fb651512d16ad22 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Thu, 3 Feb 2011 16:42:25 +0530 Subject: asoc: davinci: da830/omap-l137: correct cpu_dai_name McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b36f0b39b090..fe7984221eb9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; -- cgit v1.2.2 From 7f94de483f4e37e14d646ad6e85a3c82f66fb487 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 16:27:34 +0000 Subject: ASoC: Create an AIF1ADCDAT signal widget to match AIF2 Due to the different routing for AIF1 and AIF2 we weren't using a single widget to represent the ADCDAT signal. For consistency add one. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3351f77607b3..3e308ad97ddf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ -- cgit v1.2.2 From 6ed8f1485fc82d44ac464bc84a7dcdddd1fa096f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 16:27:35 +0000 Subject: ASoC: Improve WM8994 digital power sequencing On WM8994 revision D and earlier ensure optimal sequencing with simultaneous usage of AIF1 and AIF2 by tying the signals together so if paths through both are connected the streams are started simultaneously. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3e308ad97ddf..37b8aa8a680f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1584,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { + { "AIF1DACDAT", NULL, "AIF2DACDAT" }, + { "AIF2DACDAT", NULL, "AIF1DACDAT" }, + { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, + { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, +}; + static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, @@ -3135,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); + + if (wm8994->revision < 4) + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- cgit v1.2.2 From 460c92fa38ff140f83c269e948e2aaab071d0af0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=C5=81ukasz=20Wojni=C5=82owicz?= Date: Mon, 7 Feb 2011 13:13:27 +0100 Subject: ALSA: hda - switch lfe with side in mixer for 4930g MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Built-in sub-woofer can now be controlled by lfe slider instead of side slider on Acer Aspire 5930g Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2fa9ed99c32f..2571d977df22 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_4930G] = { - .mixers = { alc888_base_mixer, + .mixers = { alc888_acer_aspire_4930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, alc888_acer_aspire_4930g_verbs }, -- cgit v1.2.2 From 1cdfa9f34acb9780e0fe7b8a41fb1a885ab94735 Mon Sep 17 00:00:00 2001 From: Joseph Teichman Date: Tue, 8 Feb 2011 01:22:36 -0500 Subject: ALSA: usbaudio - Enable the E-MU 0204 USB Signed-off-by: Joseph Teichman Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 ++-- sound/usb/quirks-table.h | 7 +++++++ sound/usb/quirks.c | 3 ++- 3 files changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7df89b3d7ded..85af6051b52d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -95,7 +95,7 @@ enum { }; -/*E-mu 0202(0404) eXtension Unit(XU) control*/ +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ enum { USB_XU_CLOCK_RATE = 0xe301, USB_XU_CLOCK_SOURCE = 0xe302, @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 35999874d301..921a86fd9884 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index cf8bf088394b..e314cdb85003 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } -- cgit v1.2.2 From 11839aed21881d7edd65dd79f22a8eb18426f672 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Feb 2011 17:25:49 +0100 Subject: ALSA: hda - Fix missing CA initialization for HDMI/DP The commit 53d7d69d8ffdfa60c5b66cc2e9ee0774aaaef5c0 ALSA: hdmi - support infoframe for DisplayPort dropped the initialization of CA field accidentally. This resulted in only two-channel LPCM mode on Nvidia machines. Reference: kernel bug 28592 https://bugzilla.kernel.org/show_bug.cgi?id=28592 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2d5b83fa8d24..a58767736727 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, hdmi_ai->ver = 0x01; hdmi_ai->len = 0x0a; hdmi_ai->CC02_CT47 = channels - 1; + hdmi_ai->CA = ca; hdmi_checksum_audio_infoframe(hdmi_ai); } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */ struct dp_audio_infoframe *dp_ai; @@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, dp_ai->len = 0x1b; dp_ai->ver = 0x11 << 2; dp_ai->CC02_CT47 = channels - 1; + dp_ai->CA = ca; } else { snd_printd("HDMI: unknown connection type at pin %d\n", pin_nid); -- cgit v1.2.2 From 41a63f18d339ae6aefe73d45a8147f63f3439b30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Feb 2011 17:39:20 +0100 Subject: ALSA: hda - Don't handle empty patch files When an empty string is passed to patch option, the driver should ignore it. Otherwise it gets an error by trying to load it. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2e91a991eb15..0baffcdee8f9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2703,7 +2703,7 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (patch[dev]) { + if (patch[dev] && *patch[dev]) { snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", patch[dev]); err = snd_hda_load_patch(chip->bus, patch[dev]); -- cgit v1.2.2 From a6c47a85b8e7e4a8c47394607c5e5c43224b0892 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 10 Feb 2011 15:39:19 +0100 Subject: ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G According to the reporter, node 0x15 needs to be muted for subwoofer to stop sounding. This pin is marked as unused by BIOS, so fix that. BugLink: http://bugs.launchpad.net/bugs/715877 Cc: stable@kernel.org (2.6.37+) Reported-by: Hans Peter Signed-off-by: David Henningsson 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 2571d977df22..089a7de2439e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19517,6 +19517,7 @@ static const struct alc_fixup alc662_fixups[] = { }; static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), -- cgit v1.2.2 From b1d4f7f4bdcf9915c41ff8cfc4425c84dabb1fde Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 10 Feb 2011 16:15:44 +0100 Subject: ALSA: hrtimer: handle delayed timer interrupts If a timer interrupt was delayed too much, hrtimer_forward_now() will forward the timer expiry more than once. When this happens, the additional number of elapsed ALSA timer ticks must be passed to snd_timer_interrupt() to prevent the ALSA timer from falling behind. This mostly fixes MIDI slowdown problems on highly-loaded systems with badly behaved interrupt handlers. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Arthur Marsh Cc: Signed-off-by: Takashi Iwai --- sound/core/hrtimer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 7730575bfadd..07efa29dfd4a 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; + unsigned long oruns; if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; - hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks); + oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); + snd_timer_interrupt(stime->timer, t->sticks * oruns); if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; -- cgit v1.2.2 From 2243c4d0727ad85aff3f54be9d178632cc9234b2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 10 Feb 2011 16:16:32 +0100 Subject: ALSA: hrtimer: remove superfluous tasklet invocation Commit bb758e9637e5ddc removed snd_hrtimer_callback() from the hardware interrupt handler, thus moving it into a tasklet, but did not tell the ALSA timer framework about this, so the timer handling would now be done in the ALSA timer tasklet scheduled from another tasklet. To fix this, add the flag to tell the ALSA timer framework that the timer handler is already being invoked in a tasklet. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/hrtimer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 07efa29dfd4a..b8b31c433d64 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -105,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t) } static struct snd_timer_hardware hrtimer_hw = { - .flags = SNDRV_TIMER_HW_AUTO, + .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET, .open = snd_hrtimer_open, .close = snd_hrtimer_close, .start = snd_hrtimer_start, -- cgit v1.2.2 From 965b76d23ea354848dea8d34059d04e150dcd464 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Thu, 10 Feb 2011 13:14:44 +0100 Subject: ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662 This netbook has a only one jack output and an internal mic. By default, mic and jack sense aren't working. Using lenovo-101e parameters makes both work. The device seems based on a Sharetronic Q70, so this should fix audio for this model too. Signed-off-by: Anisse Astier 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 089a7de2439e..3328a259a242 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18825,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), {} }; -- cgit v1.2.2 From 5e5677f239ba69fc718ec9a87ac4ba035dafe2c0 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Mon, 14 Feb 2011 07:33:24 +0800 Subject: ALSA: au88x0 - Modify pointer callback to give accurate playback position Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 23f49f356e0f..16c0bdfbb164 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; - int temp; + int temp, page, delta; temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2)); - temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1)); - return temp; + page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT; + if (dma->nr_periods >= 4) + delta = (page - dma->period_real) & 3; + else { + delta = (page - dma->period_real); + if (delta < 0) + delta += dma->nr_periods; + } + return (dma->period_virt + delta) * dma->period_bytes + + (temp & (dma->period_bytes - 1)); } static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma) -- cgit v1.2.2 From eaae55dac6b64c0616046436b294e69fc5311581 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Feb 2011 22:45:59 +0100 Subject: ALSA: caiaq - Fix possible string-buffer overflow Use strlcpy() to assure not to overflow the string array sizes by too long USB device name string. Reported-by: Rafa Cc: stable Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/midi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 68b97477577b..66eabafb1c24 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) } dev->pcm->private_data = dev; - strcpy(dev->pcm->name, dev->product_name); + strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name)); memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 2f218c77fff2..a1a47088fd0c 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (ret < 0) return ret; - strcpy(rmidi->name, device->product_name); + strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name)); rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = device; -- cgit v1.2.2 From b540afc2b3d6e4cd1d1f137ef6d9e9c78d67fecd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 14 Feb 2011 20:27:44 +0100 Subject: ALSA: HDA: Add position_fix quirk for an Asus device The bug reporter claims that position_fix=1 is needed for his microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40). Reported-by: Kjell L. BugLink: http://bugs.launchpad.net/bugs/718402 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0baffcdee8f9..fcedad9a5fef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), -- cgit v1.2.2 From 983345e51e0de144775c7449e5cb01ce6cdd1346 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 15 Feb 2011 19:57:09 +0100 Subject: ALSA: HDA: Conexant auto: Handle multiple connections to ADC node Conexant 20641 has several inputs to its ADC node, with one selector and individual amps for all inputs. This patch adds support in the Conexant auto parser to handle that case. It also means that the pin node's volume is being renamed to "Boost" to avoid name clash with the new volume controls on the ADC node. BugLink: http://bugs.launchpad.net/bugs/719524 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 61 +++++++++++++++++++++++++++++++++--------- 1 file changed, 48 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fbe97d32140d..cd29eafdc0ed 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3729,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } -static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, +static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir) + hda_nid_t nid, int hda_dir, int amp_idx) { static char name[32]; static struct snd_kcontrol_new knew[] = { @@ -3743,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir); + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]); @@ -3759,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, return 0; } +#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -3808,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; static const char *prev_label; - int i, err, cidx; + int i, err, cidx, conn_len; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; + + int multi_adc_volume = 0; /* If the ADC nid has several input volumes */ + int adc_nid = spec->adc_nids[0]; + + conn_len = snd_hda_get_connections(codec, adc_nid, conn, + HDA_MAX_CONNECTIONS); + if (conn_len < 0) + return conn_len; + + multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1; + if (!multi_adc_volume) { + err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid, + HDA_INPUT); + if (err < 0) + return err; + } - err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0], - HDA_INPUT); - if (err < 0) - return err; prev_label = NULL; cidx = 0; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; const char *label; - if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + int j; + int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP; + if (!pin_amp && !multi_adc_volume) continue; + label = hda_get_autocfg_input_label(codec, cfg, i); if (label == prev_label) cidx++; else cidx = 0; prev_label = label; - err = cx_auto_add_volume(codec, label, " Capture", cidx, - nid, HDA_INPUT); - if (err < 0) - return err; + + if (pin_amp) { + err = cx_auto_add_volume(codec, label, " Boost", cidx, + nid, HDA_INPUT); + if (err < 0) + return err; + } + + if (!multi_adc_volume) + continue; + for (j = 0; j < conn_len; j++) { + if (conn[j] == nid) { + err = cx_auto_add_volume_idx(codec, label, + " Capture", cidx, adc_nid, HDA_INPUT, j); + if (err < 0) + return err; + break; + } + } } return 0; } -- cgit v1.2.2 From 89724958e5d596bb91328644c97dd80399443e87 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 16 Feb 2011 21:34:04 +0100 Subject: ALSA: HDA: Do not announce false surround in Conexant auto Without this patch, one line-out and one speaker and Conexant's auto parser would announce (non-working) surround capabilities. BugLink: http://bugs.launchpad.net/bugs/721126 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cd29eafdc0ed..dd7c5c12225d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3410,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec) } } spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.max_channels = nums * 2; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (cfg->hp_outs > 0) spec->auto_mute = 1; -- cgit v1.2.2 From eeda276bef36026fce3029e6423e1a09a50c359e Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Mon, 21 Feb 2011 13:45:04 +0800 Subject: ALSA: fix one memory leak in sound jack Signed-off-by: Lu Guanqun Reviewed-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/core/jack.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index 4902ae568730..53b53e97c896 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, fail_input: input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return err; } -- cgit v1.2.2 From 306496761745942d8167e9193a738b559a7fb0b3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 21 Feb 2011 10:23:18 +0100 Subject: ALSA: HDA: Fix mic initialization in VIA auto parser This typo caused some microphone inputs not to be correctly initialized on VIA codecs. Reported-By: Mark Goldstein Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a76c3260d941..63b0054200a8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(spec, nid)) ctl = PIN_OUT; - else if (i == AUTO_PIN_MIC) + else if (cfg->inputs[i].type == AUTO_PIN_MIC) ctl = PIN_VREF50; else ctl = PIN_IN; -- cgit v1.2.2 From 382225e62bdb8059b7f915b133426425516dd300 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Feb 2011 10:21:18 +0100 Subject: ALSA: usb-audio: fix oops due to cleanup race when disconnecting When a USB audio device is disconnected, snd_usb_audio_disconnect() kills all audio URBs. At the same time, the application, after being notified of the disconnection, might close the device, in which case ALSA calls the .hw_free callback, which should free the URBs too. Commit de1b8b93a0ba "[ALSA] Fix hang-up at disconnection of usb-audio" prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that resulted from this race, but this introduced another race because the URB callbacks could now be executed after snd_usb_hw_free() has returned, and try to access already freed data. Fix the first race by introducing a mutex to serialize the disconnect callback and all PCM callbacks that manage URBs (hw_free and hw_params). Reported-and-tested-by: Pierre-Louis Bossart Cc: [CL: also serialize hw_params callback] Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.c | 4 ++++ sound/usb/pcm.c | 7 +++++-- sound/usb/usbaudio.h | 1 + 3 files changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 800f7cb4f251..c0f8270bc199 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, return -ENOMEM; } + mutex_init(&chip->shutdown_mutex); chip->index = idx; chip->dev = dev; chip->card = card; @@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) chip = ptr; card = chip->card; mutex_lock(®ister_mutex); + mutex_lock(&chip->shutdown_mutex); chip->shutdown = 1; chip->num_interfaces--; if (chip->num_interfaces <= 0) { @@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) snd_usb_mixer_disconnect(p); } usb_chip[chip->index] = NULL; + mutex_unlock(&chip->shutdown_mutex); mutex_unlock(®ister_mutex); snd_card_free_when_closed(card); } else { + mutex_unlock(&chip->shutdown_mutex); mutex_unlock(®ister_mutex); } } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4132522ac90f..e3f680526cb5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, } if (changed) { + mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ snd_usb_release_substream_urbs(subs, 0); /* influenced: period_bytes, channels, rate, format, */ @@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, params_rate(hw_params), snd_pcm_format_physical_width(params_format(hw_params)) * params_channels(hw_params)); + mutex_unlock(&subs->stream->chip->shutdown_mutex); } return ret; @@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; - if (!subs->stream->chip->shutdown) - snd_usb_release_substream_urbs(subs, 0); + mutex_lock(&subs->stream->chip->shutdown_mutex); + snd_usb_release_substream_urbs(subs, 0); + mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index db3eb21627ee..6e66fffe87f5 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,6 +36,7 @@ struct snd_usb_audio { struct snd_card *card; u32 usb_id; int shutdown; + struct mutex shutdown_mutex; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ int num_interfaces; int num_suspended_intf; -- cgit v1.2.2