From 6e4abc40fc125b1dcc2792eacac17606a4d86043 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 26 Mar 2005 19:35:29 +0100 Subject: [ALSA] Adds Capture to P16V chip. EMU10K1/EMU10K2 driver One can select which capture source, but one cannot yet set volumes. Signed-off-by: James Courtier-Dutton --- include/sound/emu10k1.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 43b6786abae5..8221df88053f 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -83,7 +83,8 @@ #define IPR 0x08 /* Global interrupt pending register */ /* Clear pending interrupts by writing a 1 to */ /* the relevant bits and zero to the other bits */ - +#define IPR_P16V 0x80000000 /* Bit set when the CA0151 P16V chip wishes + to interrupt */ #define IPR_GPIOMSG 0x20000000 /* GPIO message interrupt (RE'd, still not sure which INTE bits enable it) */ @@ -1109,7 +1110,9 @@ struct _snd_emu10k1 { emu10k1_voice_t voices[NUM_G]; emu10k1_voice_t p16v_voices[4]; + emu10k1_voice_t p16v_capture_voice; int p16v_device_offset; + u32 p16v_capture_source; emu10k1_pcm_mixer_t pcm_mixer[32]; emu10k1_pcm_mixer_t efx_pcm_mixer[NUM_EFX_PLAYBACK]; snd_kcontrol_t *ctl_send_routing; -- cgit v1.2.2 From 2b637da5a1bb3c128ecdadea6aee693f6ff3b786 Mon Sep 17 00:00:00 2001 From: Lee Revell Date: Wed, 30 Mar 2005 13:51:18 +0200 Subject: [ALSA] clean up card features EMU10K1/EMU10K2 driver This patch converts the emu10k1 driver to use the card capabilities structure for some more things. Not extensively tested but seems to work. Signed-off-by: Lee Revell Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 8221df88053f..b1e8ee8e0fab 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1046,6 +1046,7 @@ typedef struct { unsigned char ca0108_chip; /* Audigy 2 Value */ unsigned char ca0151_chip; /* P16V */ unsigned char spk71; /* Has 7.1 speakers */ + unsigned char sblive51; /* SBLive! 5.1 - extout 0x11 -> center, 0x12 -> lfe */ unsigned char spdif_bug; /* Has Spdif phasing bug */ unsigned char ac97_chip; /* Has an AC97 chip */ unsigned char ecard; /* APS EEPROM */ @@ -1057,11 +1058,8 @@ struct _snd_emu10k1 { int irq; unsigned long port; /* I/O port number */ - unsigned int APS: 1, /* APS flag */ - no_ac97: 1, /* no AC'97 */ - tos_link: 1, /* tos link detected */ - rear_ac97: 1, /* rear channels are on AC'97 */ - spk71:1; /* 7.1 configuration (Audigy 2 ZS) */ + unsigned int tos_link: 1, /* tos link detected */ + rear_ac97: 1; /* rear channels are on AC'97 */ const emu_chip_details_t *card_capabilities; /* Contains profile of card capabilities */ unsigned int audigy; /* is Audigy? */ unsigned int revision; /* chip revision */ @@ -1456,7 +1454,6 @@ int snd_emu10k1_fx8010_unregister_irq_handler(emu10k1_t *emu, #endif typedef struct { - unsigned int card; /* card type */ unsigned int internal_tram_size; /* in samples */ unsigned int external_tram_size; /* in samples */ char fxbus_names[16][32]; /* names of FXBUSes */ -- cgit v1.2.2 From aec72e0a4be407fb69fbee812cf0028d62e75152 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Mar 2005 14:22:25 +0200 Subject: [ALSA] Use old default id strings for compatibility EMU10K1/EMU10K2 driver Use expliciitly the old default id strings for backward compatibility. This will make 'alsactl restore' working again. Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index b1e8ee8e0fab..6647919768bf 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1050,8 +1050,9 @@ typedef struct { unsigned char spdif_bug; /* Has Spdif phasing bug */ unsigned char ac97_chip; /* Has an AC97 chip */ unsigned char ecard; /* APS EEPROM */ - char * driver; - char * name; + const char *driver; + const char *name; + const char *id; /* for backward compatibility - can be NULL if not needed */ } emu_chip_details_t; struct _snd_emu10k1 { -- cgit v1.2.2 From bdaed50292bea3e2b20c68c2ffe9dbde7c0d6910 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2005 15:48:42 +0200 Subject: [ALSA] Check revision for the proper detection of audigy 2 EMU10K1/EMU10K2 driver Check ther revision to detect non-listed audigy 2 boards. Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 6647919768bf..f5babd3f8452 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1040,6 +1040,7 @@ typedef struct { u32 vendor; u32 device; u32 subsystem; + unsigned char revision; unsigned char emu10k1_chip; /* Original SB Live. Not SB Live 24bit. */ unsigned char emu10k2_chip; /* Audigy 1 or Audigy 2. */ unsigned char ca0102_chip; /* Audigy 1 or Audigy 2. Not SB Audigy 2 Value. */ -- cgit v1.2.2 From 001f758990d685e7023008763795f1970ef56614 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 9 Apr 2005 23:38:25 +0200 Subject: [ALSA] Improve SPDIF playback via the P16V/CA0151 chip. EMU10K1/EMU10K2 driver Although we can set 44100 as the output rate, the SPDIF can do it, but the Analog output cannot. The SPDIF has the bug, whereby the Left channel arrives one sample late, so although we don't do any resampling, it is not good for AC3 non-audio output. Signed-off-by: James Courtier-Dutton --- include/sound/emu10k1.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index f5babd3f8452..61a3f418f302 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -747,6 +747,7 @@ /* Assumes sample lock */ /* These three bitfields apply to CDSRCS, GPSRCS, and (except as noted) ZVSRCS. */ +#define SRCS_SPDIFVALID 0x04000000 /* SPDIF stream valid */ #define SRCS_SPDIFLOCKED 0x02000000 /* SPDIF stream locked */ #define SRCS_RATELOCKED 0x01000000 /* Sample rate locked */ #define SRCS_ESTSAMPLERATE 0x0007ffff /* Do not modify this field. */ -- cgit v1.2.2 From 267cdf4036ed9e8565a7d909fdf854b9c7e1c5ff Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 13 Apr 2005 13:25:30 +0200 Subject: [ALSA] replace SNDRV_PCM_HW_PARAMS_RUNTIME -> SNDRV_PCM_HW_PARAMS_NORESAMPLE ALSA Core Signed-off-by: Jaroslav Kysela --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/asound.h b/include/sound/asound.h index a4d149f34541..716227eed3e3 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -344,7 +344,7 @@ enum sndrv_pcm_hw_param { SNDRV_PCM_HW_PARAM_LAST_INTERVAL = SNDRV_PCM_HW_PARAM_TICK_TIME }; -#define SNDRV_PCM_HW_PARAMS_RUNTIME (1<<0) +#define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ struct sndrv_interval { unsigned int min, max; -- cgit v1.2.2 From eb8caf30f4c059ddfdfa32b6034549622953db6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Apr 2005 14:32:57 +0200 Subject: [ALSA] Improve the shared-jack handling on ac97 AC97 Codec The handling of shared surround/clfe output jacks with line/mic-in on some AC97 codecs is improved. Instead of 'Line-In As Surround' or 'Mic As Center/LFE' switch, two new enum controls are introduced: 'Channel Mode' and 'Surround Jack Mode'. The formar changes the current output mode among 2, 4 and 6-channels. The latter controls whether the jacks are shared or independent. Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include/sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 2433e279e071..996eeab683b0 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -437,6 +437,7 @@ struct snd_ac97_build_ops { void (*suspend) (ac97_t *ac97); void (*resume) (ac97_t *ac97); #endif + void (*update_jacks) (ac97_t *ac97); /* for jack-sharing */ }; struct _snd_ac97_bus_ops { @@ -516,6 +517,9 @@ struct _snd_ac97 { } ad18xx; unsigned int dev_flags; /* device specific */ } spec; + /* jack-sharing info */ + unsigned char indep_surround; + unsigned char channel_mode; }; /* conditions */ -- cgit v1.2.2 From 07cf374169699d78721668b4e4bd02097c971f75 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Apr 2005 16:21:03 +0200 Subject: [ALSA] Increase timer protocol number ALSA Core Increase the timer protocl number (to distinguish the fix for TREAD ioctls). Signed-off-by: Takashi Iwai --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/asound.h b/include/sound/asound.h index 716227eed3e3..26db585a1819 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -559,7 +559,7 @@ enum { * Timer section - /dev/snd/timer */ -#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) +#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 3) enum sndrv_timer_class { SNDRV_TIMER_CLASS_NONE = -1, -- cgit v1.2.2 From ade2916109dc53350298f1ccfb8ab03432c590b4 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 27 Apr 2005 16:09:21 +0200 Subject: ALSA CVS update ALSA Version 1.0.9rc3 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/version.h b/include/sound/version.h index 98b4230778ed..f5959de329c8 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by configure. */ -#define CONFIG_SND_VERSION "1.0.9rc2" +#define CONFIG_SND_VERSION "1.0.9rc3" #define CONFIG_SND_DATE " (Thu Mar 24 10:33:39 2005 UTC)" -- cgit v1.2.2 From b259b10c420a59a2fdbcf5a3498253ebcbdffa1e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 29 Apr 2005 16:29:28 +0200 Subject: [ALSA] usb-audio - add Extigy/Audigy 2 NX remote control support ALSA Core,USB generic driver Add an hwdep interface that supports reading remote control data from Sound Blaster Extigy and Audigy 2 NX devices. Signed-off-by: Clemens Ladisch --- include/sound/asound.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/asound.h b/include/sound/asound.h index 26db585a1819..4321e92a7f8b 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -113,9 +113,10 @@ enum sndrv_hwdep_iface { SNDRV_HWDEP_IFACE_BLUETOOTH, /* Bluetooth audio */ SNDRV_HWDEP_IFACE_USX2Y_PCM, /* Tascam US122, US224 & US428 rawusb pcm */ SNDRV_HWDEP_IFACE_PCXHR, /* Digigram PCXHR */ + SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_PCXHR + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC }; struct sndrv_hwdep_info { -- cgit v1.2.2 From 14c7e472aa979eecc15255eec5cec2763649c599 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Wed, 4 May 2005 16:53:53 +0200 Subject: [ALSA] Update A_SAMPLE_RATE register details. EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- include/sound/emu10k1.h | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 61a3f418f302..23dabbceb4b7 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -805,10 +805,26 @@ #define A_FXWC2 0x75 /* Selects 0x9f-0x80 for FX recording */ #define A_SPDIF_SAMPLERATE 0x76 /* Set the sample rate of SPDIF output */ -#define A_SPDIF_RATE_MASK 0x000000c0 +#define A_SAMPLE_RATE 0x76 /* Various sample rate settings. */ +#define A_SAMPLE_RATE_NOT_USED 0x0ffc111e /* Bits that are not used and cannot be set. */ +#define A_SAMPLE_RATE_UNKNOWN 0xf0030001 /* Bits that can be set, but have unknown use. */ +#define A_SPDIF_RATE_MASK 0x000000e0 /* Any other values for rates, just use 48000 */ #define A_SPDIF_48000 0x00000000 -#define A_SPDIF_44100 0x00000080 +#define A_SPDIF_192000 0x00000020 #define A_SPDIF_96000 0x00000040 +#define A_SPDIF_44100 0x00000080 + +#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */ +#define A_I2S_CAPTURE_48000 0x00000000 /* unclear if this sets the ADC rate as well. */ +#define A_I2S_CAPTURE_192000 0x00000200 +#define A_I2S_CAPTURE_96000 0x00000400 +#define A_I2S_CAPTURE_44100 0x00000800 + +#define A_PCM_RATE_MASK 0x0000e000 /* This sets the playback PCM rate on the P16V */ +#define A_PCM_48000 0x00000000 +#define A_PCM_192000 0x00002000 +#define A_PCM_96000 0x00004000 +#define A_PCM_44100 0x00008000 /* 0x77,0x78,0x79 "something i2s-related" - default to 0x01080000 on my audigy 2 ZS --rlrevell */ /* 0x7a, 0x7b - lookup tables */ -- cgit v1.2.2 From f927c8fc648420ad8edd7e4699b4ba510c2e9c6b Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 7 May 2005 15:34:13 +0200 Subject: [ALSA] Implement different capture sources. EMU10K1/EMU10K2 driver e.g. When HD Capture source is set to SPDIF, setting HD Capture channel to 0 captures from CDROM digital input. setting HD Capture channel to 1 captures from SPDIF in. Signed-off-by: James Courtier-Dutton --- include/sound/emu10k1.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 23dabbceb4b7..c50b91958ff9 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1130,6 +1130,7 @@ struct _snd_emu10k1 { emu10k1_voice_t p16v_capture_voice; int p16v_device_offset; u32 p16v_capture_source; + u32 p16v_capture_channel; emu10k1_pcm_mixer_t pcm_mixer[32]; emu10k1_pcm_mixer_t efx_pcm_mixer[NUM_EFX_PLAYBACK]; snd_kcontrol_t *ctl_send_routing; -- cgit v1.2.2 From 8c50b37c04a026ab6641ecb7eaf0fd479798e8b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 15 May 2005 15:43:54 +0200 Subject: [ALSA] Change some timer ioctls due to confliction Timer Midlevel,ALSA Core Change values of some timer ioctls to avoid confliction with FIO* ioctls. The protocol version is increased to indicate this change. Signed-off-by: Takashi Iwai --- include/sound/asound.h | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'include/sound') diff --git a/include/sound/asound.h b/include/sound/asound.h index 4321e92a7f8b..9974f83cca44 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -560,7 +560,7 @@ enum { * Timer section - /dev/snd/timer */ -#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 3) +#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 4) enum sndrv_timer_class { SNDRV_TIMER_CLASS_NONE = -1, @@ -673,10 +673,11 @@ enum { SNDRV_TIMER_IOCTL_INFO = _IOR('T', 0x11, struct sndrv_timer_info), SNDRV_TIMER_IOCTL_PARAMS = _IOW('T', 0x12, struct sndrv_timer_params), SNDRV_TIMER_IOCTL_STATUS = _IOR('T', 0x14, struct sndrv_timer_status), - SNDRV_TIMER_IOCTL_START = _IO('T', 0x20), - SNDRV_TIMER_IOCTL_STOP = _IO('T', 0x21), - SNDRV_TIMER_IOCTL_CONTINUE = _IO('T', 0x22), - SNDRV_TIMER_IOCTL_PAUSE = _IO('T', 0x23), + /* The following four ioctls are changed since 1.0.9 due to confliction */ + SNDRV_TIMER_IOCTL_START = _IO('T', 0xa0), + SNDRV_TIMER_IOCTL_STOP = _IO('T', 0xa1), + SNDRV_TIMER_IOCTL_CONTINUE = _IO('T', 0xa2), + SNDRV_TIMER_IOCTL_PAUSE = _IO('T', 0xa3), }; struct sndrv_timer_read { -- cgit v1.2.2 From 9502dcad6c1138a3ce2bae23ccd4be44c718d2a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 May 2005 16:25:46 +0200 Subject: [ALSA] Export missing snd_pcm_format_*() PCM Midlevel Export snd_pcm_format_size(). This function is used by some out-of-kernel drivers. Make snd_pcm_format_cpu_endian() macro for optimization. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'include/sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 53fc04d75bad..50a6ee1aeab2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -922,8 +922,22 @@ int snd_pcm_format_unsigned(snd_pcm_format_t format); int snd_pcm_format_linear(snd_pcm_format_t format); int snd_pcm_format_little_endian(snd_pcm_format_t format); int snd_pcm_format_big_endian(snd_pcm_format_t format); +/** + * snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian + * @format: the format to check + * + * Returns 1 if the given PCM format is CPU-endian, 0 if + * opposite, or a negative error code if endian not specified. + */ +/* int snd_pcm_format_cpu_endian(snd_pcm_format_t format); */ +#ifdef SNDRV_LITTLE_ENDIAN +#define snd_pcm_format_cpu_endian snd_pcm_format_little_endian +#else +#define snd_pcm_format_cpu_endian snd_pcm_format_big_endian +#endif int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ +ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames); snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_endian); -- cgit v1.2.2 From 123992f728785e05f385d23893bd5ec69871aeb4 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Wed, 18 May 2005 18:02:04 +0200 Subject: [ALSA] sound/core/: possible cleanups PCM Midlevel,ALSA Core,Timer Midlevel,ALSA sequencer,Virtual Midi This patch contains the following possible cleanups: - make needlessly global code static - #if 0 the following unused global functions - remove the following unneeded EXPORT_SYMBOL's Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 18 ------------------ include/sound/seq_midi_event.h | 2 -- include/sound/seq_virmidi.h | 1 - include/sound/timer.h | 2 -- 4 files changed, 23 deletions(-) (limited to 'include/sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 50a6ee1aeab2..d935417575b5 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -848,23 +848,6 @@ int snd_interval_ratnum(snd_interval_t *i, void _snd_pcm_hw_params_any(snd_pcm_hw_params_t *params); void _snd_pcm_hw_param_setempty(snd_pcm_hw_params_t *params, snd_pcm_hw_param_t var); -int snd_pcm_hw_param_min(snd_pcm_substream_t *substream, - snd_pcm_hw_params_t *params, - snd_pcm_hw_param_t var, - unsigned int val, int *dir); -int snd_pcm_hw_param_max(snd_pcm_substream_t *substream, - snd_pcm_hw_params_t *params, - snd_pcm_hw_param_t var, - unsigned int val, int *dir); -int snd_pcm_hw_param_setinteger(snd_pcm_substream_t *substream, - snd_pcm_hw_params_t *params, - snd_pcm_hw_param_t var); -int snd_pcm_hw_param_first(snd_pcm_substream_t *substream, - snd_pcm_hw_params_t *params, - snd_pcm_hw_param_t var, int *dir); -int snd_pcm_hw_param_last(snd_pcm_substream_t *substream, - snd_pcm_hw_params_t *params, - snd_pcm_hw_param_t var, int *dir); int snd_pcm_hw_param_near(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params, snd_pcm_hw_param_t var, @@ -876,7 +859,6 @@ int snd_pcm_hw_param_set(snd_pcm_substream_t *pcm, int snd_pcm_hw_params_choose(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params); int snd_pcm_hw_refine(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params); -int snd_pcm_hw_params(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params); int snd_pcm_hw_constraints_init(snd_pcm_substream_t *substream); int snd_pcm_hw_constraints_complete(snd_pcm_substream_t *substream); diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h index 4357cac07500..8857e2bd31a5 100644 --- a/include/sound/seq_midi_event.h +++ b/include/sound/seq_midi_event.h @@ -41,9 +41,7 @@ struct snd_midi_event_t { }; int snd_midi_event_new(int bufsize, snd_midi_event_t **rdev); -int snd_midi_event_resize_buffer(snd_midi_event_t *dev, int bufsize); void snd_midi_event_free(snd_midi_event_t *dev); -void snd_midi_event_init(snd_midi_event_t *dev); void snd_midi_event_reset_encode(snd_midi_event_t *dev); void snd_midi_event_reset_decode(snd_midi_event_t *dev); void snd_midi_event_no_status(snd_midi_event_t *dev, int on); diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index cf4e2388103f..1ad27e859af3 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -79,6 +79,5 @@ struct _snd_virmidi_dev { #define SNDRV_VIRMIDI_SEQ_DISPATCH 2 int snd_virmidi_new(snd_card_t *card, int device, snd_rawmidi_t **rrmidi); -int snd_virmidi_receive(snd_rawmidi_t *rmidi, snd_seq_event_t *ev); #endif /* __SOUND_SEQ_VIRMIDI */ diff --git a/include/sound/timer.h b/include/sound/timer.h index 57fde990606e..1898511a0f38 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -152,6 +152,4 @@ extern int snd_timer_pause(snd_timer_instance_t * timeri); extern void snd_timer_interrupt(snd_timer_t * timer, unsigned long ticks_left); -extern unsigned int snd_timer_system_resolution(void); - #endif /* __SOUND_TIMER_H */ -- cgit v1.2.2 From 209ac85d76e4edf05779b4bd5c2a92b059e9ab4d Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Mon, 23 May 2005 10:29:53 +0200 Subject: [ALSA] sound/isa/: cleanups GUS Library This patch contains the following possible cleanups: - make needlesly global code static - #if 0 the following unused global functions: - gus/gus_volume.c: snd_gf1_gvol_to_lvol_raw - gus/gus_volume.c: snd_gf1_calc_ramp_rate - gus/gus_volume.c: snd_gf1_compute_vibrato - gus/gus_volume.c: snd_gf1_compute_pitchbend - gus/gus_volume.c: snd_gf1_compute_freq - gus/gus_io.c: snd_gf1_i_adlib_write - gus/gus_io.c: snd_gf1_i_write_addr - gus/gus_io.c: snd_gf1_pokew - gus/gus_io.c: snd_gf1_peekw - gus/gus_io.c: snd_gf1_dram_setmem - gus/gus_io.c: snd_gf1_print_global_registers - gus/gus_io.c: snd_gf1_print_setup_registers - gus/gus_io.c: snd_gf1_peek_print_block - gus/gus_io.c: snd_gf1_print_setup_registers - gus/gus_io.c: snd_gf1_peek_print_block - #if 0 the following unused global variable: - gus/gus_tables.h: snd_gf1_scale_table - remove the following unneeded EXPORT_SYMBOL's: - gus/gus_main.c: snd_gf1_i_write16 - gus/gus_main.c: snd_gf1_start - gus/gus_main.c: snd_gf1_stop Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- include/sound/gus.h | 23 ----------------------- 1 file changed, 23 deletions(-) (limited to 'include/sound') diff --git a/include/sound/gus.h b/include/sound/gus.h index 8b6287a6fff5..b4b461ca173d 100644 --- a/include/sound/gus.h +++ b/include/sound/gus.h @@ -526,9 +526,6 @@ extern void snd_gf1_adlib_write(snd_gus_card_t * gus, unsigned char reg, unsigne extern void snd_gf1_dram_addr(snd_gus_card_t * gus, unsigned int addr); extern void snd_gf1_poke(snd_gus_card_t * gus, unsigned int addr, unsigned char data); extern unsigned char snd_gf1_peek(snd_gus_card_t * gus, unsigned int addr); -extern void snd_gf1_pokew(snd_gus_card_t * gus, unsigned int addr, unsigned short data); -extern unsigned short snd_gf1_peekw(snd_gus_card_t * gus, unsigned int addr); -extern void snd_gf1_dram_setmem(snd_gus_card_t * gus, unsigned int addr, unsigned short value, unsigned int count); extern void snd_gf1_write_addr(snd_gus_card_t * gus, unsigned char reg, unsigned int addr, short w_16bit); extern unsigned int snd_gf1_read_addr(snd_gus_card_t * gus, unsigned char reg, short w_16bit); extern void snd_gf1_i_ctrl_stop(snd_gus_card_t * gus, unsigned char reg); @@ -544,9 +541,6 @@ extern inline unsigned short snd_gf1_i_read16(snd_gus_card_t * gus, unsigned cha { return snd_gf1_i_look16(gus, reg | 0x80); } -extern void snd_gf1_i_adlib_write(snd_gus_card_t * gus, unsigned char reg, unsigned char data); -extern void snd_gf1_i_write_addr(snd_gus_card_t * gus, unsigned char reg, unsigned int addr, short w_16bit); -extern unsigned int snd_gf1_i_read_addr(snd_gus_card_t * gus, unsigned char reg, short w_16bit); extern void snd_gf1_select_active_voices(snd_gus_card_t * gus); @@ -580,10 +574,6 @@ extern void snd_gf1_lfo_command(snd_gus_card_t * gus, int voice, unsigned char * void snd_gf1_mem_lock(snd_gf1_mem_t * alloc, int xup); int snd_gf1_mem_xfree(snd_gf1_mem_t * alloc, snd_gf1_mem_block_t * block); -snd_gf1_mem_block_t *snd_gf1_mem_look(snd_gf1_mem_t * alloc, - unsigned int address); -snd_gf1_mem_block_t *snd_gf1_mem_share(snd_gf1_mem_t * alloc, - unsigned int *share_id); snd_gf1_mem_block_t *snd_gf1_mem_alloc(snd_gf1_mem_t * alloc, int owner, char *name, int size, int w_16, int align, unsigned int *share_id); @@ -608,23 +598,13 @@ int snd_gf1_dma_transfer_block(snd_gus_card_t * gus, /* gus_volume.c */ unsigned short snd_gf1_lvol_to_gvol_raw(unsigned int vol); -unsigned int snd_gf1_gvol_to_lvol_raw(unsigned short gf1_vol); -unsigned int snd_gf1_calc_ramp_rate(snd_gus_card_t * gus, - unsigned short start, - unsigned short end, - unsigned int us); unsigned short snd_gf1_translate_freq(snd_gus_card_t * gus, unsigned int freq2); -unsigned short snd_gf1_compute_pitchbend(unsigned short pitchbend, unsigned short sens); -unsigned short snd_gf1_compute_freq(unsigned int freq, - unsigned int rate, - unsigned short mix_rate); /* gus_reset.c */ void snd_gf1_set_default_handlers(snd_gus_card_t * gus, unsigned int what); void snd_gf1_smart_stop_voice(snd_gus_card_t * gus, unsigned short voice); void snd_gf1_stop_voice(snd_gus_card_t * gus, unsigned short voice); -void snd_gf1_clear_voices(snd_gus_card_t * gus, unsigned short v_min, unsigned short v_max); void snd_gf1_stop_voices(snd_gus_card_t * gus, unsigned short v_min, unsigned short v_max); snd_gus_voice_t *snd_gf1_alloc_voice(snd_gus_card_t * gus, int type, int client, int port); void snd_gf1_free_voice(snd_gus_card_t * gus, snd_gus_voice_t *voice); @@ -641,9 +621,6 @@ int snd_gf1_pcm_new(snd_gus_card_t * gus, int pcm_dev, int control_index, snd_pc #ifdef CONFIG_SND_DEBUG extern void snd_gf1_print_voice_registers(snd_gus_card_t * gus); -extern void snd_gf1_print_global_registers(snd_gus_card_t * gus); -extern void snd_gf1_print_setup_registers(snd_gus_card_t * gus); -extern void snd_gf1_peek_print_block(snd_gus_card_t * gus, unsigned int addr, int count, int w_16bit); #endif /* gus.c */ -- cgit v1.2.2 From bbc0274e9bb2e3f1d724d445a2bd32566b9b66f7 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 29 May 2005 10:32:48 +0200 Subject: [ALSA] version 1.0.9 --- include/sound/version.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/version.h b/include/sound/version.h index f5959de329c8..46acfa8c9988 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by configure. */ -#define CONFIG_SND_VERSION "1.0.9rc3" -#define CONFIG_SND_DATE " (Thu Mar 24 10:33:39 2005 UTC)" +#define CONFIG_SND_VERSION "1.0.9" +#define CONFIG_SND_DATE " (Sun May 29 07:31:02 2005 UTC)" -- cgit v1.2.2 From 69ad07cf98d0ef65cac67bac2ea4381bb499bea8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 30 May 2005 14:48:16 +0200 Subject: [ALSA] AC97 - renamed vendor/device to subvendor/subdevice where appropriate AC97 Codec,ATIIXP driver,VIA82xx driver To avoid confusion, the structure members vendor/device were renamed to subvendor/subdevice, because we compare them with PCI subsystem vendor and subsystem device. Signed-off-by: Jaroslav Kysela --- include/sound/ac97_codec.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 996eeab683b0..1309c12b8f71 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -573,8 +573,8 @@ enum { }; struct ac97_quirk { - unsigned short vendor; /* PCI vendor id */ - unsigned short device; /* PCI device id */ + unsigned short subvendor; /* PCI subsystem vendor id */ + unsigned short subdevice; /* PCI sybsystem device id */ unsigned short mask; /* device id bit mask, 0 = accept all */ unsigned int codec_id; /* codec id (if any), 0 = accept all */ const char *name; /* name shown as info */ -- cgit v1.2.2 From 763f356cd8de9e158836d236b3fd9dd149d696f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jun 2005 11:25:34 +0200 Subject: [ALSA] Add HDSP MADI driver HDSPM driver,PCI drivers,RME9652 driver Added RME Hammerfall DSP MADI driver by Winfried Ritsch. (Moved from alsa-driver tree to mainline.) Signed-off-by: Takashi Iwai --- include/sound/hdspm.h | 131 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 131 insertions(+) create mode 100644 include/sound/hdspm.h (limited to 'include/sound') diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h new file mode 100644 index 000000000000..c34427ccd0b3 --- /dev/null +++ b/include/sound/hdspm.h @@ -0,0 +1,131 @@ +#ifndef __SOUND_HDSPM_H /* -*- linux-c -*- */ +#define __SOUND_HDSPM_H +/* + * Copyright (C) 2003 Winfried Ritsch (IEM) + * based on hdsp.h from Thomas Charbonnel (thomas@undata.org) + * + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +/* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ +#define HDSPM_MAX_CHANNELS 64 + +/* -------------------- IOCTL Peak/RMS Meters -------------------- */ + +typedef struct _snd_hdspm_peak_rms hdspm_peak_rms_t; + +/* peam rms level structure like we get from hardware + + maybe in future we can memory map it so I just copy it + to user on ioctl call now an dont change anything + rms are made out of low and high values + where (long) ????_rms = (????_rms_l >> 8) + ((????_rms_h & 0xFFFFFF00)<<24) + (i asume so from the code) +*/ + +struct _snd_hdspm_peak_rms { + + unsigned int level_offset[1024]; + + unsigned int input_peak[64]; + unsigned int playback_peak[64]; + unsigned int output_peak[64]; + unsigned int xxx_peak[64]; /* not used */ + + unsigned int reserved[256]; /* not used */ + + unsigned int input_rms_l[64]; + unsigned int playback_rms_l[64]; + unsigned int output_rms_l[64]; + unsigned int xxx_rms_l[64]; /* not used */ + + unsigned int input_rms_h[64]; + unsigned int playback_rms_h[64]; + unsigned int output_rms_h[64]; + unsigned int xxx_rms_h[64]; /* not used */ +}; + +struct sndrv_hdspm_peak_rms_ioctl { + hdspm_peak_rms_t *peak; +}; + +/* use indirect access due to the limit of ioctl bit size */ +#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct sndrv_hdspm_peak_rms_ioctl) + +/* ------------ CONFIG block IOCTL ---------------------- */ + +typedef struct _snd_hdspm_config_info hdspm_config_info_t; + +struct _snd_hdspm_config_info { + unsigned char pref_sync_ref; + unsigned char wordclock_sync_check; + unsigned char madi_sync_check; + unsigned int system_sample_rate; + unsigned int autosync_sample_rate; + unsigned char system_clock_mode; + unsigned char clock_source; + unsigned char autosync_ref; + unsigned char line_out; + unsigned int passthru; + unsigned int analog_out; +}; + +#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, hdspm_config_info_t) + + +/* get Soundcard Version */ + +typedef struct _snd_hdspm_version hdspm_version_t; + +struct _snd_hdspm_version { + unsigned short firmware_rev; +}; + +#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, hdspm_version_t) + + +/* ------------- get Matrix Mixer IOCTL --------------- */ + +/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = 32768 Bytes */ + +/* organisation is 64 channelfader in a continous memory block */ +/* equivalent to hardware definition, maybe for future feature of mmap of them */ +/* each of 64 outputs has 64 infader and 64 outfader: + Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */ + +#define HDSPM_MIXER_CHANNELS HDSPM_MAX_CHANNELS + +typedef struct _snd_hdspm_channelfader snd_hdspm_channelfader_t; + +struct _snd_hdspm_channelfader { + unsigned int in[HDSPM_MIXER_CHANNELS]; + unsigned int pb[HDSPM_MIXER_CHANNELS]; +}; + +typedef struct _snd_hdspm_mixer hdspm_mixer_t; + +struct _snd_hdspm_mixer { + snd_hdspm_channelfader_t ch[HDSPM_MIXER_CHANNELS]; +}; + +struct sndrv_hdspm_mixer_ioctl { + hdspm_mixer_t *mixer; +}; + +/* use indirect access due to the limit of ioctl bit size */ +#define SNDRV_HDSPM_IOCTL_GET_MIXER _IOR('H', 0x44, struct sndrv_hdspm_mixer_ioctl) + +#endif /* __SOUND_HDSPM_H */ -- cgit v1.2.2 From b636a71d9b9525ee51ca872d461817a5bd5c39fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Jun 2005 14:13:09 +0200 Subject: [ALSA] Add const prefix Control Midlevel Add const prefix to snd_kcontrol_new_t pointer for better protection. Signed-off-by: Takashi Iwai --- include/sound/control.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/control.h b/include/sound/control.h index 7b9444cd02f4..ef7903c7a327 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -106,7 +106,7 @@ typedef int (*snd_kctl_ioctl_func_t) (snd_card_t * card, void snd_ctl_notify(snd_card_t * card, unsigned int mask, snd_ctl_elem_id_t * id); snd_kcontrol_t *snd_ctl_new(snd_kcontrol_t * kcontrol, unsigned int access); -snd_kcontrol_t *snd_ctl_new1(snd_kcontrol_new_t * kcontrolnew, void * private_data); +snd_kcontrol_t *snd_ctl_new1(const snd_kcontrol_new_t * kcontrolnew, void * private_data); void snd_ctl_free_one(snd_kcontrol_t * kcontrol); int snd_ctl_add(snd_card_t * card, snd_kcontrol_t * kcontrol); int snd_ctl_remove(snd_card_t * card, snd_kcontrol_t * kcontrol); -- cgit v1.2.2 From 543537bd922692bc978e2e356fcd8bfc9c2ee7d5 Mon Sep 17 00:00:00 2001 From: Paulo Marques Date: Thu, 23 Jun 2005 00:09:02 -0700 Subject: [PATCH] create a kstrdup library function This patch creates a new kstrdup library function and changes the "local" implementations in several places to use this function. Most of the changes come from the sound and net subsystems. The sound part had already been acknowledged by Takashi Iwai and the net part by David S. Miller. I left UML alone for now because I would need more time to read the code carefully before making changes there. Signed-off-by: Paulo Marques Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- include/sound/core.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/core.h b/include/sound/core.h index 9117c23e3a01..f8c4ef0aa352 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -292,6 +292,7 @@ void *snd_hidden_kcalloc(size_t n, size_t size, int flags); void snd_hidden_kfree(const void *obj); void *snd_hidden_vmalloc(unsigned long size); void snd_hidden_vfree(void *obj); +char *snd_hidden_kstrdup(const char *s, int flags); #define kmalloc(size, flags) snd_hidden_kmalloc(size, flags) #define kcalloc(n, size, flags) snd_hidden_kcalloc(n, size, flags) #define kfree(obj) snd_hidden_kfree(obj) @@ -301,6 +302,7 @@ void snd_hidden_vfree(void *obj); #define vmalloc_nocheck(size) snd_wrapper_vmalloc(size) #define kfree_nocheck(obj) snd_wrapper_kfree(obj) #define vfree_nocheck(obj) snd_wrapper_vfree(obj) +#define kstrdup(s, flags) snd_hidden_kstrdup(s, flags) #else #define snd_memory_init() /*NOP*/ #define snd_memory_done() /*NOP*/ @@ -311,7 +313,6 @@ void snd_hidden_vfree(void *obj); #define kfree_nocheck(obj) kfree(obj) #define vfree_nocheck(obj) vfree(obj) #endif -char *snd_kmalloc_strdup(const char *string, int flags); int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count); int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count); -- cgit v1.2.2