From 3998b70fd0ab40a276147a0f55816d383fcbeb54 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 29 Jul 2005 11:41:55 +0200 Subject: [ALSA] WM97xx AC97 codec controls AC97 Codec o Enhanced current WM97xx support to provide additional controls and use the kcontrol suffix naming convention. o Added AC97_HAS_NO_MIC, AC97_HAS_NO_TONE and AC97_HAS_NO_STD_PCM. o Cleaned up WM97xx related comments. o Removed some wm9713 double mono controls and replaced with stereo controls. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 44 +++-- sound/pci/ac97/ac97_patch.c | 429 ++++++++++++++++++++++++++++++++++---------- 2 files changed, 364 insertions(+), 109 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6983eea226da..cbf790270c30 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1307,16 +1307,18 @@ static int snd_ac97_mixer_build(ac97_t * ac97) } /* build master tone controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_TONE)) { - for (idx = 0; idx < 2; idx++) { - if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0) - return err; - if (ac97->id == AC97_ID_YMF753) { - kctl->private_value &= ~(0xff << 16); - kctl->private_value |= 7 << 16; + if (!(ac97->flags & AC97_HAS_NO_TONE)) { + if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_TONE)) { + for (idx = 0; idx < 2; idx++) { + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0) + return err; + if (ac97->id == AC97_ID_YMF753) { + kctl->private_value &= ~(0xff << 16); + kctl->private_value |= 7 << 16; + } } + snd_ac97_write_cache(ac97, AC97_MASTER_TONE, 0x0f0f); } - snd_ac97_write_cache(ac97, AC97_MASTER_TONE, 0x0f0f); } /* build PC Speaker controls */ @@ -1339,11 +1341,13 @@ static int snd_ac97_mixer_build(ac97_t * ac97) } /* build MIC controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_MIC)) { - if ((err = snd_ac97_cmix_new(card, "Mic Playback", AC97_MIC, ac97)) < 0) - return err; - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_boost, ac97))) < 0) - return err; + if (!(ac97->flags & AC97_HAS_NO_MIC)) { + if (snd_ac97_try_volume_mix(ac97, AC97_MIC)) { + if ((err = snd_ac97_cmix_new(card, "Mic Playback", AC97_MIC, ac97)) < 0) + return err; + if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_boost, ac97))) < 0) + return err; + } } /* build Line controls */ @@ -1402,12 +1406,14 @@ static int snd_ac97_mixer_build(ac97_t * ac97) } snd_ac97_write_cache(ac97, AC97_PCM, init_val); } else { - if (ac97->flags & AC97_HAS_NO_PCM_VOL) - err = snd_ac97_cmute_new(card, "PCM Playback Switch", AC97_PCM, ac97); - else - err = snd_ac97_cmix_new(card, "PCM Playback", AC97_PCM, ac97); - if (err < 0) - return err; + if (!(ac97->flags & AC97_HAS_NO_STD_PCM)) { + if (ac97->flags & AC97_HAS_NO_PCM_VOL) + err = snd_ac97_cmute_new(card, "PCM Playback Switch", AC97_PCM, ac97); + else + err = snd_ac97_cmix_new(card, "PCM Playback", AC97_PCM, ac97); + if (err < 0) + return err; + } } /* build Capture controls */ diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 66edc857d3e6..d4bb99fc896c 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -370,141 +370,387 @@ int patch_yamaha_ymf753(ac97_t * ac97) * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ -int patch_wolfson03(ac97_t * ac97) +static const snd_kcontrol_new_t wm97xx_snd_ac97_controls[] = { +AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1), +AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1), +}; + +int patch_wolfson_wm9703_specific(ac97_t * ac97) { /* This is known to work for the ViewSonic ViewPad 1000 - Randolph Bentson */ + * Randolph Bentson + * WM9703/9707/9708/9717 + */ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) + return err; + } + snd_ac97_write_cache(ac97, AC97_WM97XX_FMIXER_VOL, 0x0808); + return 0; +} - // WM9703/9707/9708/9717 - snd_ac97_write_cache(ac97, AC97_WM97XX_FMIXER_VOL, 0x0808); - snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0x8000); +static struct snd_ac97_build_ops patch_wolfson_wm9703_ops = { + .build_specific = patch_wolfson_wm9703_specific, +}; + +int patch_wolfson03(ac97_t * ac97) +{ + ac97->build_ops = &patch_wolfson_wm9703_ops; return 0; } - -int patch_wolfson04(ac97_t * ac97) + +static const snd_kcontrol_new_t wm9704_snd_ac97_controls[] = { +AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1), +AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1), +AC97_DOUBLE("Rear Playback Volume", AC97_WM9704_RMIXER_VOL, 8, 0, 31, 1), +AC97_SINGLE("Rear Playback Switch", AC97_WM9704_RMIXER_VOL, 15, 1, 1), +AC97_DOUBLE("Rear DAC Volume", AC97_WM9704_RPCM_VOL, 8, 0, 31, 1), +AC97_DOUBLE("Surround Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), +}; + +int patch_wolfson_wm9704_specific(ac97_t * ac97) { - // WM9704M/9704Q - // set front and rear mixer volume - snd_ac97_write_cache(ac97, AC97_WM97XX_FMIXER_VOL, 0x0808); - snd_ac97_write_cache(ac97, AC97_WM9704_RMIXER_VOL, 0x0808); - - // patch for DVD noise + int err, i; + for (i = 0; i < ARRAY_SIZE(wm9704_snd_ac97_controls); i++) { + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm9704_snd_ac97_controls[i], ac97))) < 0) + return err; + } + /* patch for DVD noise */ snd_ac97_write_cache(ac97, AC97_WM9704_TEST, 0x0200); - - // init vol - snd_ac97_write_cache(ac97, AC97_WM9704_RPCM_VOL, 0x0808); - - // set rear surround volume - snd_ac97_write_cache(ac97, AC97_SURROUND_MASTER, 0x0000); return 0; } - + +static struct snd_ac97_build_ops patch_wolfson_wm9704_ops = { + .build_specific = patch_wolfson_wm9704_specific, +}; + +int patch_wolfson04(ac97_t * ac97) +{ + /* WM9704M/9704Q */ + ac97->build_ops = &patch_wolfson_wm9704_ops; + return 0; +} + +int patch_wolfson_wm9705_specific(ac97_t * ac97) +{ + int err, i; + for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) + return err; + } + snd_ac97_write_cache(ac97, 0x72, 0x0808); + return 0; +} + +static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = { + .build_specific = patch_wolfson_wm9705_specific, +}; + int patch_wolfson05(ac97_t * ac97) { - // WM9705, WM9710 - // set front mixer volume - snd_ac97_write_cache(ac97, AC97_WM97XX_FMIXER_VOL, 0x0808); + /* WM9705, WM9710 */ + ac97->build_ops = &patch_wolfson_wm9705_ops; + return 0; +} + +static const char* wm9711_alc_select[] = {"None", "Left", "Right", "Stereo"}; +static const char* wm9711_alc_mix[] = {"Stereo", "Right", "Left", "None"}; +static const char* wm9711_out3_src[] = {"Left", "VREF", "Left + Right", "Mono"}; +static const char* wm9711_out3_lrsrc[] = {"Master Mix", "Headphone Mix"}; +static const char* wm9711_rec_adc[] = {"Stereo", "Left", "Right", "Mute"}; +static const char* wm9711_base[] = {"Linear Control", "Adaptive Boost"}; +static const char* wm9711_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; +static const char* wm9711_mic[] = {"Mic 1", "Differential", "Mic 2", "Stereo"}; +static const char* wm9711_rec_sel[] = + {"Mic 1", "NC", "NC", "Master Mix", "Line", "Headphone Mix", "Phone Mix", "Phone"}; +static const char* wm9711_ng_type[] = {"Constant Gain", "Mute"}; + +static const struct ac97_enum wm9711_enum[] = { +AC97_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9711_alc_select), +AC97_ENUM_SINGLE(AC97_VIDEO, 10, 4, wm9711_alc_mix), +AC97_ENUM_SINGLE(AC97_AUX, 9, 4, wm9711_out3_src), +AC97_ENUM_SINGLE(AC97_AUX, 8, 2, wm9711_out3_lrsrc), +AC97_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9711_rec_adc), +AC97_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9711_base), +AC97_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9711_rec_gain), +AC97_ENUM_SINGLE(AC97_MIC, 5, 4, wm9711_mic), +AC97_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, wm9711_rec_sel), +AC97_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9711_ng_type), +}; + +static const snd_kcontrol_new_t wm9711_snd_ac97_controls[] = { +AC97_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), +AC97_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), +AC97_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), +AC97_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), +AC97_ENUM("ALC Function", wm9711_enum[0]), +AC97_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 1), +AC97_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1), +AC97_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), +AC97_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), +AC97_ENUM("ALC NG Type", wm9711_enum[9]), +AC97_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1), + +AC97_SINGLE("Side Tone Switch", AC97_VIDEO, 15, 1, 1), +AC97_SINGLE("Side Tone Volume", AC97_VIDEO, 12, 7, 1), +AC97_ENUM("ALC Headphone Mux", wm9711_enum[1]), +AC97_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1), + +AC97_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1), +AC97_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1), +AC97_ENUM("Out3 Mux", wm9711_enum[2]), +AC97_ENUM("Out3 LR Mux", wm9711_enum[3]), +AC97_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1), + +AC97_SINGLE("Beep to Headphone Switch", AC97_PC_BEEP, 15, 1, 1), +AC97_SINGLE("Beep to Headphone Volume", AC97_PC_BEEP, 12, 7, 1), +AC97_SINGLE("Beep to Side Tone Switch", AC97_PC_BEEP, 11, 1, 1), +AC97_SINGLE("Beep to Side Tone Volume", AC97_PC_BEEP, 8, 7, 1), +AC97_SINGLE("Beep to Phone Switch", AC97_PC_BEEP, 7, 1, 1), +AC97_SINGLE("Beep to Phone Volume", AC97_PC_BEEP, 4, 7, 1), + +AC97_SINGLE("Aux to Headphone Switch", AC97_CD, 15, 1, 1), +AC97_SINGLE("Aux to Headphone Volume", AC97_CD, 12, 7, 1), +AC97_SINGLE("Aux to Side Tone Switch", AC97_CD, 11, 1, 1), +AC97_SINGLE("Aux to Side Tone Volume", AC97_CD, 8, 7, 1), +AC97_SINGLE("Aux to Phone Switch", AC97_CD, 7, 1, 1), +AC97_SINGLE("Aux to Phone Volume", AC97_CD, 4, 7, 1), + +AC97_SINGLE("Phone to Headphone Switch", AC97_PHONE, 15, 1, 1), +AC97_SINGLE("Phone to Master Switch", AC97_PHONE, 14, 1, 1), + +AC97_SINGLE("Line to Headphone Switch", AC97_LINE, 15, 1, 1), +AC97_SINGLE("Line to Master Switch", AC97_LINE, 14, 1, 1), +AC97_SINGLE("Line to Phone Switch", AC97_LINE, 13, 1, 1), + +AC97_SINGLE("PCM Playback to Headphone Switch", AC97_PCM, 15, 1, 1), +AC97_SINGLE("PCM Playback to Master Switch", AC97_PCM, 14, 1, 1), +AC97_SINGLE("PCM Playback to Phone Switch", AC97_PCM, 13, 1, 1), + +AC97_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), +AC97_ENUM("Capture to Phone Mux", wm9711_enum[4]), +AC97_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), +AC97_ENUM("Capture Select", wm9711_enum[8]), + +AC97_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), +AC97_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), + +AC97_ENUM("Bass Control", wm9711_enum[5]), +AC97_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), +AC97_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), +AC97_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), + +AC97_SINGLE("ADC Switch", AC97_REC_GAIN, 15, 1, 1), +AC97_ENUM("Capture Volume Steps", wm9711_enum[6]), +AC97_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 1), +AC97_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), + +AC97_SINGLE("Mic 1 to Phone Switch", AC97_MIC, 14, 1, 1), +AC97_SINGLE("Mic 2 to Phone Switch", AC97_MIC, 13, 1, 1), +AC97_ENUM("Mic Select Source", wm9711_enum[7]), +AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 32, 1), +AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), + +AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), +AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), +AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), +}; + +int patch_wolfson_wm9711_specific(ac97_t * ac97) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm9711_snd_ac97_controls); i++) { + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm9711_snd_ac97_controls[i], ac97))) < 0) + return err; + } + snd_ac97_write_cache(ac97, AC97_CODEC_CLASS_REV, 0x0808); + snd_ac97_write_cache(ac97, AC97_PCI_SVID, 0x0808); + snd_ac97_write_cache(ac97, AC97_VIDEO, 0x0808); + snd_ac97_write_cache(ac97, AC97_AUX, 0x0808); + snd_ac97_write_cache(ac97, AC97_PC_BEEP, 0x0808); + snd_ac97_write_cache(ac97, AC97_CD, 0x0000); return 0; } +static struct snd_ac97_build_ops patch_wolfson_wm9711_ops = { + .build_specific = patch_wolfson_wm9711_specific, +}; + int patch_wolfson11(ac97_t * ac97) { - // WM9711, WM9712 - // set out3 volume - snd_ac97_write_cache(ac97, AC97_WM9711_OUT3VOL, 0x0808); + /* WM9711, WM9712 */ + ac97->build_ops = &patch_wolfson_wm9711_ops; + + ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_MIC | + AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD; + return 0; } -static const char* wm9713_mic_mixer[] = {"Stereo", "Mic1", "Mic2", "Mute"}; +static const char* wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; static const char* wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; -static const char* wm9713_rec_src_l[] = {"Mic1", "Mic2", "Line L", "Mono In", "HP Mix L", "Spk Mix", "Mono Mix", "Zh"}; -static const char* wm9713_rec_src_r[] = {"Mic1", "Mic2", "Line R", "Mono In", "HP Mix R", "Spk Mix", "Mono Mix", "Zh"}; +static const char* wm9713_rec_src[] = + {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone Mix", "Master Mix", + "Mono Mix", "Zh"}; +static const char* wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; +static const char* wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; +static const char* wm9713_mono_pga[] = {"Vmid", "Zh", "Mono Mix", "Inv 1"}; +static const char* wm9713_spk_pga[] = + {"Vmid", "Zh", "Headphone Mix", "Master Mix", "Inv", "NC", "NC", "NC"}; +static const char* wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone Mix", "NC"}; +static const char* wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "NC"}; +static const char* wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "NC"}; +static const char* wm9713_dac_inv[] = + {"Off", "Mono Mix", "Master Mix", "Headphone Mix L", "Headphone Mix R", + "Headphone Mix Mono", "NC", "Vmid"}; +static const char* wm9713_base[] = {"Linear Control", "Adaptive Boost"}; +static const char* wm9713_ng_type[] = {"Constant Gain", "Mute"}; static const struct ac97_enum wm9713_enum[] = { AC97_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), AC97_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), AC97_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), -AC97_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src_l), -AC97_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src_r), +AC97_ENUM_DOUBLE(AC97_VIDEO, 3, 0, 8, wm9713_rec_src), +AC97_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), +AC97_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), +AC97_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), +AC97_ENUM_DOUBLE(AC97_REC_GAIN, 11, 8, 8, wm9713_spk_pga), +AC97_ENUM_DOUBLE(AC97_REC_GAIN, 6, 4, 4, wm9713_hp_pga), +AC97_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), +AC97_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), +AC97_ENUM_DOUBLE(AC97_REC_GAIN_MIC, 13, 10, 8, wm9713_dac_inv), +AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_base), +AC97_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), }; -static const snd_kcontrol_new_t wm13_snd_ac97_controls_line_in[] = { +static const snd_kcontrol_new_t wm13_snd_ac97_controls[] = { AC97_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), -AC97_SINGLE("Line In to Headphone Mute", AC97_PC_BEEP, 15, 1, 1), -AC97_SINGLE("Line In to Speaker Mute", AC97_PC_BEEP, 14, 1, 1), -AC97_SINGLE("Line In to Mono Mute", AC97_PC_BEEP, 13, 1, 1), -}; - -static const snd_kcontrol_new_t wm13_snd_ac97_controls_dac[] = { -AC97_DOUBLE("DAC Volume", AC97_PHONE, 8, 0, 31, 1), -AC97_SINGLE("DAC to Headphone Mute", AC97_PHONE, 15, 1, 1), -AC97_SINGLE("DAC to Speaker Mute", AC97_PHONE, 14, 1, 1), -AC97_SINGLE("DAC to Mono Mute", AC97_PHONE, 13, 1, 1), -}; - -static const snd_kcontrol_new_t wm13_snd_ac97_controls_mic[] = { -AC97_SINGLE("MICA Volume", AC97_MIC, 8, 31, 1), -AC97_SINGLE("MICB Volume", AC97_MIC, 0, 31, 1), -AC97_SINGLE("MICA to Mono Mute", AC97_LINE, 7, 1, 1), -AC97_SINGLE("MICB to Mono Mute", AC97_LINE, 6, 1, 1), -AC97_SINGLE("MIC Boost (+20dB)", AC97_LINE, 5, 1, 1), -AC97_ENUM("MIC Headphone Routing", wm9713_enum[0]), -AC97_SINGLE("MIC Headphone Mixer Volume", AC97_LINE, 0, 7, 1) +AC97_SINGLE("Line In to Headphone Switch", AC97_PC_BEEP, 15, 1, 1), +AC97_SINGLE("Line In to Master Switch", AC97_PC_BEEP, 14, 1, 1), +AC97_SINGLE("Line In to Mono Switch", AC97_PC_BEEP, 13, 1, 1), + +AC97_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), +AC97_SINGLE("PCM Playback to Headphone Switch", AC97_PHONE, 15, 1, 1), +AC97_SINGLE("PCM Playback to Master Switch", AC97_PHONE, 14, 1, 1), +AC97_SINGLE("PCM Playback to Mono Switch", AC97_PHONE, 13, 1, 1), + +AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), +AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), +AC97_SINGLE("Mic 1 to Mono Switch", AC97_LINE, 7, 1, 1), +AC97_SINGLE("Mic 2 to Mono Switch", AC97_LINE, 6, 1, 1), +AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), +AC97_ENUM("Mic to Headphone Mux", wm9713_enum[0]), +AC97_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), + +AC97_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), +AC97_ENUM("Capture Volume Steps", wm9713_enum[4]), +AC97_DOUBLE("Capture Volume", AC97_CD, 8, 0, 15, 0), +AC97_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), + +AC97_ENUM("Capture to Headphone Mux", wm9713_enum[1]), +AC97_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +AC97_ENUM("Capture to Mono Mux", wm9713_enum[2]), +AC97_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), +AC97_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), +AC97_ENUM("Capture Select", wm9713_enum[3]), + +AC97_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), +AC97_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), +AC97_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +AC97_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), +AC97_ENUM("ALC Function", wm9713_enum[5]), +AC97_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), +AC97_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), +AC97_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), +AC97_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), +AC97_ENUM("ALC NG Type", wm9713_enum[13]), +AC97_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), + +AC97_DOUBLE("Master ZC Switch", AC97_MASTER, 14, 6, 1, 0), +AC97_DOUBLE("Headphone ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), +AC97_DOUBLE("Out3/4 ZC Switch", AC97_MASTER_MONO, 14, 6, 1, 0), +AC97_SINGLE("Master Right Switch", AC97_MASTER, 7, 1, 1), +AC97_SINGLE("Headphone Right Switch", AC97_HEADPHONE, 7, 1, 1), +AC97_SINGLE("Out3/4 Right Switch", AC97_MASTER_MONO, 7, 1, 1), + +AC97_SINGLE("Mono In to Headphone Switch", AC97_MASTER_TONE, 15, 1, 1), +AC97_SINGLE("Mono In to Master Switch", AC97_MASTER_TONE, 14, 1, 1), +AC97_SINGLE("Mono In Volume", AC97_MASTER_TONE, 8, 31, 1), +AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1), +AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0), +AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1), + +AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1), +AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1), +AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1), +AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1), +AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1), +AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1), + +AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1), +AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1), +AC97_SINGLE("Voice to Master Switch", AC97_PCM, 11, 1, 1), +AC97_SINGLE("Voice to Master Volume", AC97_PCM, 8, 7, 1), +AC97_SINGLE("Voice to Mono Switch", AC97_PCM, 7, 1, 1), +AC97_SINGLE("Voice to Mono Volume", AC97_PCM, 4, 7, 1), + +AC97_SINGLE("Aux to Headphone Switch", AC97_REC_SEL, 15, 1, 1), +AC97_SINGLE("Aux to Headphone Volume", AC97_REC_SEL, 12, 7, 1), +AC97_SINGLE("Aux to Master Switch", AC97_REC_SEL, 11, 1, 1), +AC97_SINGLE("Aux to Master Volume", AC97_REC_SEL, 8, 7, 1), +AC97_SINGLE("Aux to Mono Switch", AC97_REC_SEL, 7, 1, 1), +AC97_SINGLE("Aux to Mono Volume", AC97_REC_SEL, 4, 7, 1), + +AC97_ENUM("Mono Input Mux", wm9713_enum[6]), +AC97_ENUM("Master Input Mux", wm9713_enum[7]), +AC97_ENUM("Headphone Input Mux", wm9713_enum[8]), +AC97_ENUM("Out 3 Input Mux", wm9713_enum[9]), +AC97_ENUM("Out 4 Input Mux", wm9713_enum[10]), + +AC97_ENUM("Bass Control", wm9713_enum[12]), +AC97_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), +AC97_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), +AC97_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), +AC97_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), +AC97_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), }; -static const snd_kcontrol_new_t wm13_snd_ac97_controls_adc[] = { -AC97_SINGLE("ADC Mute", AC97_CD, 15, 1, 1), -AC97_DOUBLE("Gain Step Size (1.5dB/0.75dB)", AC97_CD, 14, 6, 1, 1), -AC97_DOUBLE("ADC Volume",AC97_CD, 8, 0, 15, 0), -AC97_SINGLE("ADC Zero Cross", AC97_CD, 7, 1, 1), +static const snd_kcontrol_new_t wm13_snd_ac97_controls_3d[] = { +AC97_ENUM("Inv Input Mux", wm9713_enum[11]), +AC97_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), +AC97_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), +AC97_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; -static const snd_kcontrol_new_t wm13_snd_ac97_controls_recsel[] = { -AC97_ENUM("Record to Headphone Path", wm9713_enum[1]), -AC97_SINGLE("Record to Headphone Volume", AC97_VIDEO, 11, 7, 0), -AC97_ENUM("Record to Mono Path", wm9713_enum[2]), -AC97_SINGLE("Record to Mono Boost (+20dB)", AC97_VIDEO, 8, 1, 0), -AC97_SINGLE("Record ADC Boost (+20dB)", AC97_VIDEO, 6, 1, 0), -AC97_ENUM("Record Select Left", wm9713_enum[3]), -AC97_ENUM("Record Select Right", wm9713_enum[4]), -}; +static int patch_wolfson_wm9713_3d (ac97_t * ac97) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_3d); i++) { + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_3d[i], ac97))) < 0) + return err; + } + return 0; +} static int patch_wolfson_wm9713_specific(ac97_t * ac97) { int err, i; - for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_line_in); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_line_in[i], ac97))) < 0) + for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls); i++) { + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls[i], ac97))) < 0) return err; } snd_ac97_write_cache(ac97, AC97_PC_BEEP, 0x0808); - - for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_dac); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_dac[i], ac97))) < 0) - return err; - } snd_ac97_write_cache(ac97, AC97_PHONE, 0x0808); - - for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_mic); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_mic[i], ac97))) < 0) - return err; - } snd_ac97_write_cache(ac97, AC97_MIC, 0x0808); snd_ac97_write_cache(ac97, AC97_LINE, 0x00da); - - for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_adc); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_adc[i], ac97))) < 0) - return err; - } snd_ac97_write_cache(ac97, AC97_CD, 0x0808); - - for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_recsel); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_recsel[i], ac97))) < 0) - return err; - } snd_ac97_write_cache(ac97, AC97_VIDEO, 0xd612); snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x1ba0); - return 0; } @@ -525,6 +771,7 @@ static void patch_wolfson_wm9713_resume (ac97_t * ac97) static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { .build_specific = patch_wolfson_wm9713_specific, + .build_3d = patch_wolfson_wm9713_3d, #ifdef CONFIG_PM .suspend = patch_wolfson_wm9713_suspend, .resume = patch_wolfson_wm9713_resume @@ -533,10 +780,12 @@ static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { int patch_wolfson13(ac97_t * ac97) { + /* WM9713, WM9714 */ ac97->build_ops = &patch_wolfson_wm9713_ops; ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_PHONE | - AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD; + AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD | AC97_HAS_NO_TONE | + AC97_HAS_NO_STD_PCM; snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xda00); snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0x3810); -- cgit From cb8e2f83851ff17b1b361644e82420a923fbc318 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Jul 2005 11:54:32 +0200 Subject: [ALSA] hda-codec - Fix reordering of surround channels HDA Codec driver - Fixed the reordering of surround channels. Originally reported by Nicolas GRAZIANO . - Show the selected ssid when debug option is set. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e2cf02387289..d3ac7530ec89 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1541,8 +1541,11 @@ int snd_hda_check_board_config(struct hda_codec *codec, const struct hda_board_c for (c = tbl; c->modelname || c->pci_subvendor; c++) { if (c->pci_subvendor == subsystem_vendor && (! c->pci_subdevice /* all match */|| - (c->pci_subdevice == subsystem_device))) + (c->pci_subdevice == subsystem_device))) { + snd_printdd(KERN_INFO "hda_codec: PCI %x:%x, codec config %d is selected\n", + subsystem_vendor, subsystem_device, c->config); return c->config; + } } } return -1; @@ -1803,11 +1806,25 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->line_out_pins[j] = nid; } - /* Swap surround and CLFE: the association order is front/CLFE/surr/back */ - if (cfg->line_outs >= 3) { + /* Reorder the surround channels + * ALSA sequence is front/surr/clfe/side + * HDA sequence is: + * 4-ch: front/surr => OK as it is + * 6-ch: front/clfe/surr + * 8-ch: front/clfe/side/surr + */ + switch (cfg->line_outs) { + case 3: nid = cfg->line_out_pins[1]; cfg->line_out_pins[1] = cfg->line_out_pins[2]; cfg->line_out_pins[2] = nid; + break; + case 4: + nid = cfg->line_out_pins[1]; + cfg->line_out_pins[1] = cfg->line_out_pins[3]; + cfg->line_out_pins[3] = cfg->line_out_pins[2]; + cfg->line_out_pins[2] = nid; + break; } return 0; -- cgit From d827560c96346e0b8b02c1000cc347d73488c818 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Jul 2005 11:56:41 +0200 Subject: [ALSA] hda-codec - Add default config for ASUS P5AD2 HDA Codec driver Added the default config for ASUS P5AD2. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cmedia.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 86f195f19eef..07fb4f5a54b3 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -647,6 +647,7 @@ static struct hda_board_config cmi9880_cfg_tbl[] = { { .modelname = "min_fp", .config = CMI_MIN_FP }, { .modelname = "full", .config = CMI_FULL }, { .modelname = "full_dig", .config = CMI_FULL_DIG }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x813d, .config = CMI_FULL_DIG }, /* ASUS P5AD2 */ { .modelname = "allout", .config = CMI_ALLOUT }, { .modelname = "auto", .config = CMI_AUTO }, {} /* terminator */ -- cgit From b9f5a89c74e541533766dcda55d34a06253f60f3 Mon Sep 17 00:00:00 2001 From: Nicolas Graziano Date: Fri, 29 Jul 2005 12:17:20 +0200 Subject: [ALSA] hda-codec - Fix LFE volume/switch HDA Codec driver Fixed LFE volume/switch control. Signed-off-by: Nicolas Graziano Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d3ac7530ec89..026ae726d875 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -749,12 +749,14 @@ int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t long *valp = ucontrol->value.integer.value; int change = 0; - if (chs & 1) + if (chs & 1) { change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, 0x7f, *valp); + valp++; + } if (chs & 2) change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x7f, valp[1]); + 0x7f, *valp); return change; } @@ -796,12 +798,15 @@ int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t long *valp = ucontrol->value.integer.value; int change = 0; - if (chs & 1) + if (chs & 1) { change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, 0x80, *valp ? 0 : 0x80); + valp++; + } if (chs & 2) change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x80, valp[1] ? 0 : 0x80); + 0x80, *valp ? 0 : 0x80); + return change; } -- cgit From 42dfd0adab83c95890e562623264999914771763 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Jul 2005 14:18:39 +0200 Subject: [ALSA] via82xx - Add dxs entry for Acer Aspire 1524 WMLi VIA82xx driver Added the dxs entry for Acer Aspire 1524 WMLi. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 4889600387c8..6e1c29f5799a 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2153,6 +2153,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1019, .subdevice = 0x0a81, .action = VIA_DXS_NO_VRA }, /* ECS K7VTA3 v8.0 */ { .subvendor = 0x1019, .subdevice = 0x0a85, .action = VIA_DXS_NO_VRA }, /* ECS L7VMM2 */ { .subvendor = 0x1025, .subdevice = 0x0033, .action = VIA_DXS_NO_VRA }, /* Acer Inspire 1353LM */ + { .subvendor = 0x1025, .subdevice = 0x0046, .action = VIA_DXS_SRC }, /* Acer Aspire 1524 WMLi */ { .subvendor = 0x1043, .subdevice = 0x8095, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8X (FIXME: possibly VIA_DXS_ENABLE?)*/ { .subvendor = 0x1043, .subdevice = 0x80a1, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8-X */ { .subvendor = 0x1043, .subdevice = 0x80b0, .action = VIA_DXS_NO_VRA }, /* ASUS A7V600 & K8V*/ -- cgit From 67ed4161f60d76b3b27efc085b36357a6a90768c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 29 Jul 2005 15:32:58 +0200 Subject: [ALSA] sound - fix .iface field of mixer control elements Documentation,CS46xx driver,EMU10K1/EMU10K2 driver,AD1848 driver SB16/AWE driver,CMIPCI driver,ENS1370/1+ driver,RME32 driver RME96 driver,ICE1712 driver,ICE1724 driver,KORG1212 driver RME HDSP driver,RME9652 driver This patch changes .iface to SNDRV_CTL_ELEM_IFACE_MIXER whre _PCM or _HWDEP was used in controls that are not associated with a specific PCM (sub)stream or hwdep device, and changes some controls that got inconsitent .iface values due to copy+paste errors. Furthermore, it makes sure that all control that do use _PCM or _HWDEP use the correct number in the .device field. Signed-off-by: Clemens Ladisch --- sound/pci/cmipci.c | 2 +- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 4 +++- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/emu10k1/emumixer.c | 15 ++++++++---- sound/pci/emu10k1/emupcm.c | 7 +++++- sound/pci/ens1370.c | 6 ++--- sound/pci/ice1712/delta.c | 10 ++++---- sound/pci/ice1712/ice1712.c | 4 ++-- sound/pci/ice1712/ice1724.c | 4 ++-- sound/pci/korg1212/korg1212.c | 4 ++-- sound/pci/rme32.c | 4 ++-- sound/pci/rme96.c | 4 ++-- sound/pci/rme9652/hdsp.c | 55 +++++++++++++++++++++---------------------- sound/pci/rme9652/hdspm.c | 27 +++++++++++---------- sound/pci/rme9652/rme9652.c | 24 +++++++++---------- 16 files changed, 94 insertions(+), 80 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index f5a4ac1ceef9..b098b51099c2 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1029,7 +1029,7 @@ static int snd_cmipci_spdif_mask_get(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_cmipci_spdif_mask __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), .info = snd_cmipci_spdif_mask_info, .get = snd_cmipci_spdif_mask_get, diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index db212ecd792a..b9fff4ee6f9d 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -113,7 +113,7 @@ static int __devinit snd_card_cs46xx_probe(struct pci_dev *pci, return err; } #endif - if ((err = snd_cs46xx_mixer(chip)) < 0) { + if ((err = snd_cs46xx_mixer(chip, 2)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index ff28af1f658e..1546aec02ea7 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2440,7 +2440,7 @@ static int __devinit cs46xx_detect_codec(cs46xx_t *chip, int codec) return -ENXIO; } -int __devinit snd_cs46xx_mixer(cs46xx_t *chip) +int __devinit snd_cs46xx_mixer(cs46xx_t *chip, int spdif_device) { snd_card_t *card = chip->card; snd_ctl_elem_id_t id; @@ -2476,6 +2476,8 @@ int __devinit snd_cs46xx_mixer(cs46xx_t *chip) for (idx = 0; idx < ARRAY_SIZE(snd_cs46xx_controls); idx++) { snd_kcontrol_t *kctl; kctl = snd_ctl_new1(&snd_cs46xx_controls[idx], chip); + if (kctl && kctl->id.iface == SNDRV_CTL_ELEM_IFACE_PCM) + kctl->id.device = spdif_device; if ((err = snd_ctl_add(card, kctl)) < 0) return err; } diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index b17142cabead..fc377c4b666c 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -149,7 +149,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, } } - if ((err = snd_emu10k1_mixer(emu)) < 0) { + if ((err = snd_emu10k1_mixer(emu, 0, 3)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 6be82c5fe138..279edaeb0663 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -295,7 +295,7 @@ static int snd_emu10k1_send_routing_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_emu10k1_send_routing_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "EMU10K1 PCM Send Routing", .count = 32, .info = snd_emu10k1_send_routing_info, @@ -364,7 +364,7 @@ static int snd_emu10k1_send_volume_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_emu10k1_send_volume_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "EMU10K1 PCM Send Volume", .count = 32, .info = snd_emu10k1_send_volume_info, @@ -427,7 +427,7 @@ static int snd_emu10k1_attn_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_emu10k1_attn_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "EMU10K1 PCM Volume", .count = 32, .info = snd_emu10k1_attn_info, @@ -737,7 +737,8 @@ static int rename_ctl(snd_card_t *card, const char *src, const char *dst) return -ENOENT; } -int __devinit snd_emu10k1_mixer(emu10k1_t *emu) +int __devinit snd_emu10k1_mixer(emu10k1_t *emu, + int pcm_device, int multi_device) { int err, pcm; snd_kcontrol_t *kctl; @@ -852,29 +853,35 @@ int __devinit snd_emu10k1_mixer(emu10k1_t *emu) if ((kctl = emu->ctl_send_routing = snd_ctl_new1(&snd_emu10k1_send_routing_control, emu)) == NULL) return -ENOMEM; + kctl->id.device = pcm_device; if ((err = snd_ctl_add(card, kctl))) return err; if ((kctl = emu->ctl_send_volume = snd_ctl_new1(&snd_emu10k1_send_volume_control, emu)) == NULL) return -ENOMEM; + kctl->id.device = pcm_device; if ((err = snd_ctl_add(card, kctl))) return err; if ((kctl = emu->ctl_attn = snd_ctl_new1(&snd_emu10k1_attn_control, emu)) == NULL) return -ENOMEM; + kctl->id.device = pcm_device; if ((err = snd_ctl_add(card, kctl))) return err; if ((kctl = emu->ctl_efx_send_routing = snd_ctl_new1(&snd_emu10k1_efx_send_routing_control, emu)) == NULL) return -ENOMEM; + kctl->id.device = multi_device; if ((err = snd_ctl_add(card, kctl))) return err; if ((kctl = emu->ctl_efx_send_volume = snd_ctl_new1(&snd_emu10k1_efx_send_volume_control, emu)) == NULL) return -ENOMEM; + kctl->id.device = multi_device; if ((err = snd_ctl_add(card, kctl))) return err; if ((kctl = emu->ctl_efx_attn = snd_ctl_new1(&snd_emu10k1_efx_attn_control, emu)) == NULL) return -ENOMEM; + kctl->id.device = multi_device; if ((err = snd_ctl_add(card, kctl))) return err; diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 520b99af5f55..9c35f6dde1b5 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1682,6 +1682,7 @@ static void snd_emu10k1_pcm_efx_free(snd_pcm_t *pcm) int __devinit snd_emu10k1_pcm_efx(emu10k1_t * emu, int device, snd_pcm_t ** rpcm) { snd_pcm_t *pcm; + snd_kcontrol_t *kctl; int err; if (rpcm) @@ -1714,7 +1715,11 @@ int __devinit snd_emu10k1_pcm_efx(emu10k1_t * emu, int device, snd_pcm_t ** rpcm emu->efx_voices_mask[0] = 0xffff0000; emu->efx_voices_mask[1] = 0; } - snd_ctl_add(emu->card, snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu)); + kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu); + if (!kctl) + return -ENOMEM; + kctl->id.device = device; + snd_ctl_add(emu->card, kctl); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), 64*1024, 64*1024); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 78a81f3912a1..0c64e65bee97 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1446,7 +1446,7 @@ static int snd_es1371_spdif_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t static snd_kcontrol_new_t snd_es1371_mixer_spdif[] __devinitdata = { ES1371_SPDIF("IEC958 Playback Switch"), { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = snd_ens1373_spdif_info, .get = snd_ens1373_spdif_default_get, @@ -1454,13 +1454,13 @@ static snd_kcontrol_new_t snd_es1371_mixer_spdif[] __devinitdata = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), .info = snd_ens1373_spdif_info, .get = snd_ens1373_spdif_mask_get }, { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), .info = snd_ens1373_spdif_info, .get = snd_ens1373_spdif_stream_get, diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index eb20f73be61a..39fbe662965d 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -618,15 +618,15 @@ static int __devinit snd_ice1712_delta_init(ice1712_t *ice) */ static snd_kcontrol_new_t snd_ice1712_delta1010_wordclock_select __devinitdata = -ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_PCM, "Word Clock Sync", 0, ICE1712_DELTA_WORD_CLOCK_SELECT, 1, 0); +ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_WORD_CLOCK_SELECT, 1, 0); static snd_kcontrol_new_t snd_ice1712_delta1010lt_wordclock_select __devinitdata = -ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_PCM, "Word Clock Sync", 0, ICE1712_DELTA_1010LT_WORDCLOCK, 1, 0); +ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_1010LT_WORDCLOCK, 1, 0); static snd_kcontrol_new_t snd_ice1712_delta1010_wordclock_status __devinitdata = -ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_PCM, "Word Clock Status", 0, ICE1712_DELTA_WORD_CLOCK_STATUS, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE); +ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Status", 0, ICE1712_DELTA_WORD_CLOCK_STATUS, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE); static snd_kcontrol_new_t snd_ice1712_deltadio2496_spdif_in_select __devinitdata = -ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_PCM, "IEC958 Input Optical", 0, ICE1712_DELTA_SPDIF_INPUT_SELECT, 0, 0); +ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "IEC958 Input Optical", 0, ICE1712_DELTA_SPDIF_INPUT_SELECT, 0, 0); static snd_kcontrol_new_t snd_ice1712_delta_spdif_in_status __devinitdata = -ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_PCM, "Delta IEC958 Input Status", 0, ICE1712_DELTA_SPDIF_IN_STAT, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE); +ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Delta IEC958 Input Status", 0, ICE1712_DELTA_SPDIF_IN_STAT, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE); static int __devinit snd_ice1712_delta_add_controls(ice1712_t *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index a2545a5b26c4..d51c72b721e0 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1715,7 +1715,7 @@ static int snd_ice1712_spdif_maskp_get(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_ice1712_spdif_maskc __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), .info = snd_ice1712_spdif_info, .get = snd_ice1712_spdif_maskc_get, @@ -1724,7 +1724,7 @@ static snd_kcontrol_new_t snd_ice1712_spdif_maskc __devinitdata = static snd_kcontrol_new_t snd_ice1712_spdif_maskp __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), .info = snd_ice1712_spdif_info, .get = snd_ice1712_spdif_maskp_get, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 79b5f12e06fc..e819b9ffc266 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1414,7 +1414,7 @@ static int snd_vt1724_spdif_maskp_get(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_vt1724_spdif_maskc __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), .info = snd_vt1724_spdif_info, .get = snd_vt1724_spdif_maskc_get, @@ -1423,7 +1423,7 @@ static snd_kcontrol_new_t snd_vt1724_spdif_maskc __devinitdata = static snd_kcontrol_new_t snd_vt1724_spdif_maskp __devinitdata = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), .info = snd_vt1724_spdif_info, .get = snd_vt1724_spdif_maskp_get, diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 79d8eda54f0d..d2aa9c82d41e 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2067,7 +2067,7 @@ static int snd_korg1212_control_sync_put(snd_kcontrol_t * kcontrol, snd_ctl_elem }, \ { \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE, \ - .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = c_name " Monitor Phase Invert", \ .info = snd_korg1212_control_phase_info, \ .get = snd_korg1212_control_phase_get, \ @@ -2082,7 +2082,7 @@ static snd_kcontrol_new_t snd_korg1212_controls[] = { MON_MIXER(4, "ADAT-5"), MON_MIXER(5, "ADAT-6"), MON_MIXER(6, "ADAT-7"), MON_MIXER(7, "ADAT-8"), { .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Sync Source", .info = snd_korg1212_control_sync_info, .get = snd_korg1212_control_sync_get, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index b7b554df6705..456be39e8e4a 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1900,7 +1900,7 @@ static snd_kcontrol_new_t snd_rme32_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_rme32_control_spdif_mask_info, .get = snd_rme32_control_spdif_mask_get, @@ -1908,7 +1908,7 @@ static snd_kcontrol_new_t snd_rme32_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_rme32_control_spdif_mask_info, .get = snd_rme32_control_spdif_mask_get, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 10c4f45a913c..9645e9004a48 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2266,7 +2266,7 @@ static snd_kcontrol_new_t snd_rme96_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), .info = snd_rme96_control_spdif_mask_info, .get = snd_rme96_control_spdif_mask_get, @@ -2276,7 +2276,7 @@ static snd_kcontrol_new_t snd_rme96_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), .info = snd_rme96_control_spdif_mask_info, .get = snd_rme96_control_spdif_mask_get, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 796621de5009..60eef845a074 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1524,7 +1524,7 @@ static int snd_hdsp_control_spdif_mask_get(snd_kcontrol_t * kcontrol, snd_ctl_el } #define HDSP_SPDIF_IN(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_spdif_in, \ @@ -1584,7 +1584,7 @@ static int snd_hdsp_put_spdif_in(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t } #define HDSP_SPDIF_OUT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_hdsp_info_spdif_bits, \ .get = snd_hdsp_get_spdif_out, .put = snd_hdsp_put_spdif_out } @@ -1638,7 +1638,7 @@ static int snd_hdsp_put_spdif_out(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_ } #define HDSP_SPDIF_PROFESSIONAL(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_hdsp_info_spdif_bits, \ .get = snd_hdsp_get_spdif_professional, .put = snd_hdsp_put_spdif_professional } @@ -1683,7 +1683,7 @@ static int snd_hdsp_put_spdif_professional(snd_kcontrol_t * kcontrol, snd_ctl_el } #define HDSP_SPDIF_EMPHASIS(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_hdsp_info_spdif_bits, \ .get = snd_hdsp_get_spdif_emphasis, .put = snd_hdsp_put_spdif_emphasis } @@ -1728,7 +1728,7 @@ static int snd_hdsp_put_spdif_emphasis(snd_kcontrol_t * kcontrol, snd_ctl_elem_v } #define HDSP_SPDIF_NON_AUDIO(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_hdsp_info_spdif_bits, \ .get = snd_hdsp_get_spdif_nonaudio, .put = snd_hdsp_put_spdif_nonaudio } @@ -1773,7 +1773,7 @@ static int snd_hdsp_put_spdif_nonaudio(snd_kcontrol_t * kcontrol, snd_ctl_elem_v } #define HDSP_SPDIF_SAMPLE_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1834,7 +1834,7 @@ static int snd_hdsp_get_spdif_sample_rate(snd_kcontrol_t * kcontrol, snd_ctl_ele } #define HDSP_SYSTEM_SAMPLE_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1858,7 +1858,7 @@ static int snd_hdsp_get_system_sample_rate(snd_kcontrol_t * kcontrol, snd_ctl_el } #define HDSP_AUTOSYNC_SAMPLE_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1918,7 +1918,7 @@ static int snd_hdsp_get_autosync_sample_rate(snd_kcontrol_t * kcontrol, snd_ctl_ } #define HDSP_SYSTEM_CLOCK_MODE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1958,7 +1958,7 @@ static int snd_hdsp_get_system_clock_mode(snd_kcontrol_t * kcontrol, snd_ctl_ele } #define HDSP_CLOCK_SOURCE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_clock_source, \ @@ -2124,7 +2124,7 @@ static int snd_hdsp_put_clock_source_lock(snd_kcontrol_t * kcontrol, snd_ctl_ele } #define HDSP_DA_GAIN(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_da_gain, \ @@ -2210,7 +2210,7 @@ static int snd_hdsp_put_da_gain(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t } #define HDSP_AD_GAIN(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_ad_gain, \ @@ -2296,7 +2296,7 @@ static int snd_hdsp_put_ad_gain(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t } #define HDSP_PHONE_GAIN(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_phone_gain, \ @@ -2382,7 +2382,7 @@ static int snd_hdsp_put_phone_gain(snd_kcontrol_t * kcontrol, snd_ctl_elem_value } #define HDSP_XLR_BREAKOUT_CABLE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_xlr_breakout_cable, \ @@ -2447,7 +2447,7 @@ static int snd_hdsp_put_xlr_breakout_cable(snd_kcontrol_t * kcontrol, snd_ctl_el Switching this on desactivates external ADAT */ #define HDSP_AEB(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_aeb, \ @@ -2508,7 +2508,7 @@ static int snd_hdsp_put_aeb(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * uc } #define HDSP_PREF_SYNC_REF(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_pref_sync_ref, \ @@ -2641,7 +2641,7 @@ static int snd_hdsp_put_pref_sync_ref(snd_kcontrol_t * kcontrol, snd_ctl_elem_va } #define HDSP_AUTOSYNC_REF(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -2697,7 +2697,7 @@ static int snd_hdsp_get_autosync_ref(snd_kcontrol_t * kcontrol, snd_ctl_elem_val } #define HDSP_LINE_OUT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_line_out, \ @@ -2757,7 +2757,7 @@ static int snd_hdsp_put_line_out(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t } #define HDSP_PRECISE_POINTER(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_precise_pointer, \ @@ -2811,7 +2811,7 @@ static int snd_hdsp_put_precise_pointer(snd_kcontrol_t * kcontrol, snd_ctl_elem_ } #define HDSP_USE_MIDI_TASKLET(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ .name = xname, \ .index = xindex, \ .info = snd_hdsp_info_use_midi_tasklet, \ @@ -2868,6 +2868,7 @@ static int snd_hdsp_put_use_midi_tasklet(snd_kcontrol_t * kcontrol, snd_ctl_elem { .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ .name = xname, \ .index = xindex, \ + .device = 0, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_hdsp_info_mixer, \ @@ -2939,7 +2940,7 @@ static int snd_hdsp_put_mixer(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * } #define HDSP_WC_SYNC_CHECK(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ @@ -2983,7 +2984,7 @@ static int snd_hdsp_get_wc_sync_check(snd_kcontrol_t * kcontrol, snd_ctl_elem_va } #define HDSP_SPDIF_SYNC_CHECK(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ @@ -3015,7 +3016,7 @@ static int snd_hdsp_get_spdif_sync_check(snd_kcontrol_t * kcontrol, snd_ctl_elem } #define HDSP_ADATSYNC_SYNC_CHECK(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ @@ -3046,7 +3047,7 @@ static int snd_hdsp_get_adatsync_sync_check(snd_kcontrol_t * kcontrol, snd_ctl_e } #define HDSP_ADAT_SYNC_CHECK \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_hdsp_info_sync_check, \ .get = snd_hdsp_get_adat_sync_check \ @@ -3103,7 +3104,7 @@ HDSP_XLR_BREAKOUT_CABLE("XLR Breakout Cable", 0) static snd_kcontrol_new_t snd_hdsp_controls[] = { { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = snd_hdsp_control_spdif_info, .get = snd_hdsp_control_spdif_get, @@ -3111,7 +3112,7 @@ static snd_kcontrol_new_t snd_hdsp_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), .info = snd_hdsp_control_spdif_stream_info, .get = snd_hdsp_control_spdif_stream_get, @@ -3146,8 +3147,6 @@ HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), /* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { - /* FIXME: should be PCM or MIXER? */ - /* .iface = SNDRV_CTL_ELEM_IFACE_PCM, */ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Sample Clock Source Locking", .info = snd_hdsp_info_clock_source_lock, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9e86d0eb41ce..f6daec4e4d97 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1168,7 +1168,7 @@ static void hdspm_midi_tasklet(unsigned long arg) /* get the system sample rate which is set */ #define HDSPM_SYSTEM_SAMPLE_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1195,7 +1195,7 @@ static int snd_hdspm_get_system_sample_rate(snd_kcontrol_t * kcontrol, } #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1264,7 +1264,7 @@ static int snd_hdspm_get_autosync_sample_rate(snd_kcontrol_t * kcontrol, } #define HDSPM_SYSTEM_CLOCK_MODE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1310,7 +1310,7 @@ static int snd_hdspm_get_system_clock_mode(snd_kcontrol_t * kcontrol, } #define HDSPM_CLOCK_SOURCE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_clock_source, \ @@ -1457,7 +1457,7 @@ static int snd_hdspm_put_clock_source(snd_kcontrol_t * kcontrol, } #define HDSPM_PREF_SYNC_REF(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_pref_sync_ref, \ @@ -1547,7 +1547,7 @@ static int snd_hdspm_put_pref_sync_ref(snd_kcontrol_t * kcontrol, } #define HDSPM_AUTOSYNC_REF(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ, \ @@ -1604,7 +1604,7 @@ static int snd_hdspm_get_autosync_ref(snd_kcontrol_t * kcontrol, } #define HDSPM_LINE_OUT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_line_out, \ @@ -1668,7 +1668,7 @@ static int snd_hdspm_put_line_out(snd_kcontrol_t * kcontrol, } #define HDSPM_TX_64(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_tx_64, \ @@ -1731,7 +1731,7 @@ static int snd_hdspm_put_tx_64(snd_kcontrol_t * kcontrol, } #define HDSPM_C_TMS(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_c_tms, \ @@ -1794,7 +1794,7 @@ static int snd_hdspm_put_c_tms(snd_kcontrol_t * kcontrol, } #define HDSPM_SAFE_MODE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_safe_mode, \ @@ -1857,7 +1857,7 @@ static int snd_hdspm_put_safe_mode(snd_kcontrol_t * kcontrol, } #define HDSPM_INPUT_SELECT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .info = snd_hdspm_info_input_select, \ @@ -1941,6 +1941,7 @@ static int snd_hdspm_put_input_select(snd_kcontrol_t * kcontrol, { .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ .name = xname, \ .index = xindex, \ + .device = 0, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_hdspm_info_mixer, \ @@ -2124,7 +2125,7 @@ static int snd_hdspm_put_playback_mixer(snd_kcontrol_t * kcontrol, } #define HDSPM_WC_SYNC_CHECK(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ @@ -2170,7 +2171,7 @@ static int snd_hdspm_get_wc_sync_check(snd_kcontrol_t * kcontrol, #define HDSPM_MADI_SYNC_CHECK(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 1bc9d0df8516..6e3a91dba3eb 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -893,7 +893,7 @@ static int snd_rme9652_control_spdif_mask_get(snd_kcontrol_t * kcontrol, snd_ctl } #define RME9652_ADAT1_IN(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_rme9652_info_adat1_in, \ .get = snd_rme9652_get_adat1_in, \ .put = snd_rme9652_put_adat1_in } @@ -971,7 +971,7 @@ static int snd_rme9652_put_adat1_in(snd_kcontrol_t * kcontrol, snd_ctl_elem_valu } #define RME9652_SPDIF_IN(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_rme9652_info_spdif_in, \ .get = snd_rme9652_get_spdif_in, .put = snd_rme9652_put_spdif_in } @@ -1042,7 +1042,7 @@ static int snd_rme9652_put_spdif_in(snd_kcontrol_t * kcontrol, snd_ctl_elem_valu } #define RME9652_SPDIF_OUT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_rme9652_info_spdif_out, \ .get = snd_rme9652_get_spdif_out, .put = snd_rme9652_put_spdif_out } @@ -1110,7 +1110,7 @@ static int snd_rme9652_put_spdif_out(snd_kcontrol_t * kcontrol, snd_ctl_elem_val } #define RME9652_SYNC_MODE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_rme9652_info_sync_mode, \ .get = snd_rme9652_get_sync_mode, .put = snd_rme9652_put_sync_mode } @@ -1195,7 +1195,7 @@ static int snd_rme9652_put_sync_mode(snd_kcontrol_t * kcontrol, snd_ctl_elem_val } #define RME9652_SYNC_PREF(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_rme9652_info_sync_pref, \ .get = snd_rme9652_get_sync_pref, .put = snd_rme9652_put_sync_pref } @@ -1340,7 +1340,7 @@ static int snd_rme9652_put_thru(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t } #define RME9652_PASSTHRU(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_rme9652_info_passthru, \ .put = snd_rme9652_put_passthru, \ .get = snd_rme9652_get_passthru } @@ -1386,7 +1386,7 @@ static int snd_rme9652_put_passthru(snd_kcontrol_t * kcontrol, snd_ctl_elem_valu /* Read-only switches */ #define RME9652_SPDIF_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_rme9652_info_spdif_rate, \ .get = snd_rme9652_get_spdif_rate } @@ -1411,7 +1411,7 @@ static int snd_rme9652_get_spdif_rate(snd_kcontrol_t * kcontrol, snd_ctl_elem_va } #define RME9652_ADAT_SYNC(xname, xindex, xidx) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_rme9652_info_adat_sync, \ .get = snd_rme9652_get_adat_sync, .private_value = xidx } @@ -1447,7 +1447,7 @@ static int snd_rme9652_get_adat_sync(snd_kcontrol_t * kcontrol, snd_ctl_elem_val } #define RME9652_TC_VALID(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = xname, .index = xindex, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_rme9652_info_tc_valid, \ .get = snd_rme9652_get_tc_valid } @@ -1529,7 +1529,7 @@ static int snd_rme9652_get_tc_value(void *private_data, static snd_kcontrol_new_t snd_rme9652_controls[] = { { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = snd_rme9652_control_spdif_info, .get = snd_rme9652_control_spdif_get, @@ -1537,7 +1537,7 @@ static snd_kcontrol_new_t snd_rme9652_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), .info = snd_rme9652_control_spdif_stream_info, .get = snd_rme9652_control_spdif_stream_get, @@ -1568,7 +1568,7 @@ RME9652_SPDIF_OUT("IEC958 Output also on ADAT1", 0), RME9652_SYNC_MODE("Sync Mode", 0), RME9652_SYNC_PREF("Preferred Sync Source", 0), { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channels Thru", .index = 0, .info = snd_rme9652_info_thru, -- cgit From 0ca06a00e206b963587ac471e6d1c52bf33b9a18 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 29 Jul 2005 16:13:36 +0200 Subject: [ALSA] AC97 bus interface for ad-hoc drivers AC97 Codec,PCI drivers I've made the review changes and as requested I've pasted the RFC by Nicolas below:- 'I would like to know what people think of the following patch. It allows for a codec on an AC97 bus to be shared with other drivers which are completely unrelated to audio. It registers a new bus type, and whenever a codec instance is created then a device for it is also registered with the driver model using that bus type. This allows, for example, to use the extra features of the UCB1400 like the touchscreen interface and the additional GPIOs and ADCs available on that chip for battery monitoring. I have a working UCB1400 touchscreen driver here that simply registers with the driver model happily working alongside with audio features using this.' Changes over RFC:- o Now matches codec name within codec group. o Added ac97_dev_release() to stop kernel complaining about no release method for device. o Added 'config SND_AC97_BUS' to sound/pci/Kconfig and moved 'config SND_AC97_CODEC' out with the PCI=n statement. o module is now called snd-ac97-bus Signed-off-by: Liam Girdwood Signed-off-by: Nicolas Pitre Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 10 ++++-- sound/pci/ac97/Makefile | 2 ++ sound/pci/ac97/ac97_bus.c | 79 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/ac97/ac97_codec.c | 35 ++++++++++++++++++++ 4 files changed, 123 insertions(+), 3 deletions(-) create mode 100644 sound/pci/ac97/ac97_bus.c (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 26b42bb20a0a..1e458919cce6 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,11 +1,15 @@ # ALSA PCI drivers -menu "PCI devices" - depends on SND!=n && PCI - config SND_AC97_CODEC tristate select SND_PCM + select SND_AC97_BUS + +config SND_AC97_BUS + tristate + +menu "PCI devices" + depends on SND!=n && PCI config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index 3c3222122d8b..77b3482cb133 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -10,9 +10,11 @@ snd-ac97-codec-objs += ac97_proc.o endif snd-ak4531-codec-objs := ak4531_codec.o +snd-ac97-bus-objs := ac97_bus.o # Toplevel Module Dependency obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o +obj-$(CONFIG_SND_AC97_BUS) += snd-ac97-bus.o obj-m := $(sort $(obj-m)) diff --git a/sound/pci/ac97/ac97_bus.c b/sound/pci/ac97/ac97_bus.c new file mode 100644 index 000000000000..227f8b9f67ce --- /dev/null +++ b/sound/pci/ac97/ac97_bus.c @@ -0,0 +1,79 @@ +/* + * Linux driver model AC97 bus interface + * + * Author: Nicolas Pitre + * Created: Jan 14, 2005 + * Copyright: (C) MontaVista Software Inc. + * + * 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. + */ + +#include +#include +#include +#include + +/* + * Codec families have names seperated by commas, so we search for an + * individual codec name within the family string. + */ +static int ac97_bus_match(struct device *dev, struct device_driver *drv) +{ + return (strstr(dev->bus_id, drv->name) != NULL); +} + +static int ac97_bus_suspend(struct device *dev, pm_message_t state) +{ + int ret = 0; + + if (dev->driver && dev->driver->suspend) { + ret = dev->driver->suspend(dev, state, SUSPEND_DISABLE); + if (ret == 0) + ret = dev->driver->suspend(dev, state, SUSPEND_SAVE_STATE); + if (ret == 0) + ret = dev->driver->suspend(dev, state, SUSPEND_POWER_DOWN); + } + return ret; +} + +static int ac97_bus_resume(struct device *dev) +{ + int ret = 0; + + if (dev->driver && dev->driver->resume) { + ret = dev->driver->resume(dev, RESUME_POWER_ON); + if (ret == 0) + ret = dev->driver->resume(dev, RESUME_RESTORE_STATE); + if (ret == 0) + ret = dev->driver->resume(dev, RESUME_ENABLE); + } + return ret; +} + +struct bus_type ac97_bus_type = { + .name = "ac97", + .match = ac97_bus_match, + .suspend = ac97_bus_suspend, + .resume = ac97_bus_resume, +}; + +static int __init ac97_bus_init(void) +{ + return bus_register(&ac97_bus_type); +} + +subsys_initcall(ac97_bus_init); + +static void __exit ac97_bus_exit(void) +{ + bus_unregister(&ac97_bus_type); +} + +module_exit(ac97_bus_exit); + +EXPORT_SYMBOL(ac97_bus_type); + +MODULE_LICENSE("GPL"); diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index cbf790270c30..33dba10f03e8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1813,6 +1813,39 @@ int snd_ac97_bus(snd_card_t *card, int num, ac97_bus_ops_t *ops, return 0; } +/* stop no dev release warning */ +static void ac97_device_release(struct device * dev) +{ +} + +/* register ac97 codec to bus */ +static int snd_ac97_dev_register(snd_device_t *device) +{ + ac97_t *ac97 = device->device_data; + int err; + + ac97->dev.bus = &ac97_bus_type; + ac97->dev.parent = ac97->bus->card->dev; + ac97->dev.platform_data = ac97; + ac97->dev.release = ac97_device_release; + strncpy(ac97->dev.bus_id, snd_ac97_get_short_name(ac97), BUS_ID_SIZE); + if ((err = device_register(&ac97->dev)) < 0) { + snd_printk(KERN_ERR "Can't register ac97 bus\n"); + ac97->dev.bus = NULL; + return err; + } + return 0; +} + +/* unregister ac97 codec */ +static int snd_ac97_dev_unregister(snd_device_t *device) +{ + ac97_t *ac97 = device->device_data; + if (ac97->dev.bus) + device_unregister(&ac97->dev); + return snd_ac97_free(ac97); +} + /* build_ops to do nothing */ static struct snd_ac97_build_ops null_build_ops; @@ -1846,6 +1879,8 @@ int snd_ac97_mixer(ac97_bus_t *bus, ac97_template_t *template, ac97_t **rac97) const ac97_codec_id_t *pid; static snd_device_ops_t ops = { .dev_free = snd_ac97_dev_free, + .dev_register = snd_ac97_dev_register, + .dev_unregister = snd_ac97_dev_unregister, }; snd_assert(rac97 != NULL, return -EINVAL); -- cgit From 10e8d78a94fc57f1bf11d50b97ff85b005e46d0b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 3 Aug 2005 13:40:08 +0200 Subject: [ALSA] use SNDRV_CTL_NAME_IEC958 macro CMI8330 driver,ENS1370/1+ driver,FM801 driver,VIA82xx driver,AC97 Codec ALI5451 driver,CS46xx driver,EMU10K1/EMU10K2 driver,HDA Codec driver ICE1712 driver,ICE1724 driver Where appropriate, use the SNDRV_CTL_NAME_IEC958 instead of a literal string. Signed-off-by: Clemens Ladisch --- sound/pci/ac97/ac97_patch.c | 8 ++++---- sound/pci/ali5451/ali5451.c | 4 ++-- sound/pci/cs46xx/cs46xx_lib.c | 6 +++--- sound/pci/emu10k1/emufx.c | 26 +++++++++++++------------- sound/pci/ens1370.c | 2 +- sound/pci/fm801.c | 8 ++++---- sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/ice1712/ice1712.c | 6 +++--- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/via82xx.c | 2 +- 10 files changed, 34 insertions(+), 34 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index d4bb99fc896c..00fb51992460 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1955,7 +1955,7 @@ static const snd_kcontrol_new_t snd_ac97_controls_alc650[] = { }; static const snd_kcontrol_new_t snd_ac97_spdif_controls_alc650[] = { - AC97_SINGLE("IEC958 Capture Switch", AC97_ALC650_MULTICH, 11, 1, 0), + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_ALC650_MULTICH, 11, 1, 0), AC97_SINGLE("Analog to IEC958 Output", AC97_ALC650_MULTICH, 12, 1, 0), /* disable this controls since it doesn't work as expected */ /* AC97_SINGLE("IEC958 Input Monitor", AC97_ALC650_MULTICH, 13, 1, 0), */ @@ -2098,12 +2098,12 @@ static int alc655_iec958_route_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_ } static const snd_kcontrol_new_t snd_ac97_spdif_controls_alc655[] = { - AC97_PAGE_SINGLE("IEC958 Capture Switch", AC97_ALC650_MULTICH, 11, 1, 0, 0), + AC97_PAGE_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_ALC650_MULTICH, 11, 1, 0, 0), /* disable this controls since it doesn't work as expected */ /* AC97_PAGE_SINGLE("IEC958 Input Monitor", AC97_ALC650_MULTICH, 14, 1, 0, 0), */ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", .info = alc655_iec958_route_info, .get = alc655_iec958_route_get, .put = alc655_iec958_route_put, @@ -2682,7 +2682,7 @@ static const snd_kcontrol_new_t snd_ac97_controls_it2646[] = { }; static const snd_kcontrol_new_t snd_ac97_spdif_controls_it2646[] = { - AC97_SINGLE("IEC958 Capture Switch", 0x76, 11, 1, 0), + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0x76, 11, 1, 0), AC97_SINGLE("Analog to IEC958 Output", 0x76, 12, 1, 0), AC97_SINGLE("IEC958 Input Monitor", 0x76, 13, 1, 0), }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index f08ae71f902d..cb12d780a6c6 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1959,9 +1959,9 @@ static int snd_ali5451_spdif_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t static snd_kcontrol_new_t snd_ali5451_mixer_spdif[] __devinitdata = { /* spdif aplayback switch */ /* FIXME: "IEC958 Playback Switch" may conflict with one on ac97_codec */ - ALI5451_SPDIF("IEC958 Output switch", 0, 0), + ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), 0, 0), /* spdif out to spdif channel */ - ALI5451_SPDIF("IEC958 Channel Output Switch", 0, 1), + ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("Channel Output ",NONE,SWITCH), 0, 1), /* spdif in from spdif channel */ ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, 2) }; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1546aec02ea7..5eb9e0bb9100 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2231,7 +2231,7 @@ static snd_kcontrol_new_t snd_cs46xx_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Output Switch", + .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), .info = snd_mixer_boolean_info, .get = snd_cs46xx_iec958_get, .put = snd_cs46xx_iec958_put, @@ -2239,7 +2239,7 @@ static snd_kcontrol_new_t snd_cs46xx_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Input Switch", + .name = SNDRV_CTL_NAME_IEC958("Input ",NONE,SWITCH), .info = snd_mixer_boolean_info, .get = snd_cs46xx_iec958_get, .put = snd_cs46xx_iec958_put, @@ -2249,7 +2249,7 @@ static snd_kcontrol_new_t snd_cs46xx_controls[] __devinitdata = { /* Input IEC958 volume does not work for the moment. (Benny) */ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Input Volume", + .name = SNDRV_CTL_NAME_IEC958("Input ",NONE,VOLUME), .info = snd_cs46xx_vol_info, .get = snd_cs46xx_vol_iec958_get, .put = snd_cs46xx_vol_iec958_put, diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 0529fb281125..637c555cfdb1 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1159,12 +1159,12 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* Optical SPDIF Playback Volume */ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_OPT_SPDIF_L); A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_OPT_SPDIF_R); - snd_emu10k1_init_stereo_control(&controls[nctl++], "IEC958 Optical Playback Volume", gpr, 0); + snd_emu10k1_init_stereo_control(&controls[nctl++], SNDRV_CTL_NAME_IEC958("Optical ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; /* Optical SPDIF Capture Volume */ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_OPT_SPDIF_L); A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_OPT_SPDIF_R); - snd_emu10k1_init_stereo_control(&controls[nctl++], "IEC958 Optical Capture Volume", gpr, 0); + snd_emu10k1_init_stereo_control(&controls[nctl++], SNDRV_CTL_NAME_IEC958("Optical ",CAPTURE,VOLUME), gpr, 0); gpr += 2; /* Line2 Playback Volume */ @@ -1389,7 +1389,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_OP(icode, &ptr, iACC3, A_EXTOUT(A_EXTOUT_FRONT_L + z), A_GPR(tmp + 0), A_GPR(tmp + 1), A_C_00000000); } } - snd_emu10k1_init_stereo_onoff_control(controls + nctl++, "IEC958 Optical Raw Playback Switch", gpr, 0); + snd_emu10k1_init_stereo_onoff_control(controls + nctl++, SNDRV_CTL_NAME_IEC958("Optical Raw ",PLAYBACK,SWITCH), gpr, 0); gpr += 2; A_PUT_STEREO_OUTPUT(A_EXTOUT_REAR_L, A_EXTOUT_REAR_R, playback+2 + SND_EMU10K1_PLAYBACK_CHANNELS); @@ -1716,7 +1716,7 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) /* IEC958 TTL Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_SPDIF_CD_L + z, gpr + z); - snd_emu10k1_init_stereo_control(controls + i++, "IEC958 TTL Playback Volume", gpr, 0); + snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; /* IEC958 TTL Capture Volume + Switch */ @@ -1724,8 +1724,8 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_SPDIF_CD_L + z, gpr + 2 + z); VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z); } - snd_emu10k1_init_stereo_control(controls + i++, "IEC958 TTL Capture Volume", gpr, 0); - snd_emu10k1_init_stereo_onoff_control(controls + i++, "IEC958 TTL Capture Switch", gpr + 2, 0); + snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",CAPTURE,VOLUME), gpr, 0); + snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",CAPTURE,SWITCH), gpr + 2, 0); gpr += 4; } @@ -1750,7 +1750,7 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) /* IEC958 Optical Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_TOSLINK_L + z, gpr + z); - snd_emu10k1_init_stereo_control(controls + i++, "IEC958 LiveDrive Playback Volume", gpr, 0); + snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; /* IEC958 Optical Capture Volume */ @@ -1758,8 +1758,8 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_TOSLINK_L + z, gpr + 2 + z); VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z); } - snd_emu10k1_init_stereo_control(controls + i++, "IEC958 LiveDrive Capture Volume", gpr, 0); - snd_emu10k1_init_stereo_onoff_control(controls + i++, "IEC958 LiveDrive Capture Switch", gpr + 2, 0); + snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",CAPTURE,VOLUME), gpr, 0); + snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",CAPTURE,SWITCH), gpr + 2, 0); gpr += 4; } @@ -1784,7 +1784,7 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) /* IEC958 Coax Playback Volume */ for (z = 0; z < 2; z++) VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_COAX_SPDIF_L + z, gpr + z); - snd_emu10k1_init_stereo_control(controls + i++, "IEC958 Coaxial Playback Volume", gpr, 0); + snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",PLAYBACK,VOLUME), gpr, 0); gpr += 2; /* IEC958 Coax Capture Volume + Switch */ @@ -1792,8 +1792,8 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_COAX_SPDIF_L + z, gpr + 2 + z); VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z); } - snd_emu10k1_init_stereo_control(controls + i++, "IEC958 Coaxial Capture Volume", gpr, 0); - snd_emu10k1_init_stereo_onoff_control(controls + i++, "IEC958 Coaxial Capture Switch", gpr + 2, 0); + snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",CAPTURE,VOLUME), gpr, 0); + snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",CAPTURE,SWITCH), gpr + 2, 0); gpr += 4; } @@ -1920,7 +1920,7 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu) #endif } - snd_emu10k1_init_stereo_onoff_control(controls + i++, "IEC958 Optical Raw Playback Switch", gpr, 0); + snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("Optical Raw ",PLAYBACK,SWITCH), gpr, 0); gpr += 2; } diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 0c64e65bee97..bc8272be0031 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1444,7 +1444,7 @@ static int snd_es1371_spdif_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t /* spdif controls */ static snd_kcontrol_new_t snd_es1371_mixer_spdif[] __devinitdata = { - ES1371_SPDIF("IEC958 Playback Switch"), + ES1371_SPDIF(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH)), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index ff10e637a95e..36b2f62e8573 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1155,10 +1155,10 @@ FM801_SINGLE("FM Playback Switch", FM801_FM_VOL, 15, 1, 1), static snd_kcontrol_new_t snd_fm801_controls_multi[] __devinitdata = { FM801_SINGLE("AC97 2ch->4ch Copy Switch", FM801_CODEC_CTRL, 7, 1, 0), FM801_SINGLE("AC97 18-bit Switch", FM801_CODEC_CTRL, 10, 1, 0), -FM801_SINGLE("IEC958 Capture Switch", FM801_I2S_MODE, 8, 1, 0), -FM801_SINGLE("IEC958 Raw Data Playback Switch", FM801_I2S_MODE, 9, 1, 0), -FM801_SINGLE("IEC958 Raw Data Capture Switch", FM801_I2S_MODE, 10, 1, 0), -FM801_SINGLE("IEC958 Playback Switch", FM801_GEN_CTRL, 2, 1, 0), +FM801_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), FM801_I2S_MODE, 8, 1, 0), +FM801_SINGLE(SNDRV_CTL_NAME_IEC958("Raw Data ",PLAYBACK,SWITCH), FM801_I2S_MODE, 9, 1, 0), +FM801_SINGLE(SNDRV_CTL_NAME_IEC958("Raw Data ",CAPTURE,SWITCH), FM801_I2S_MODE, 10, 1, 0), +FM801_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), FM801_GEN_CTRL, 2, 1, 0), }; static void snd_fm801_mixer_free_ac97_bus(ac97_bus_t *bus) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2fd05bb84136..bceb83a42a38 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -572,7 +572,7 @@ static snd_kcontrol_new_t ad1983_mixers[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", .info = ad1983_spdif_route_info, .get = ad1983_spdif_route_get, .put = ad1983_spdif_route_put, @@ -705,7 +705,7 @@ static snd_kcontrol_new_t ad1981_mixers[] = { /* identical with AD1983 */ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", .info = ad1983_spdif_route_info, .get = ad1983_spdif_route_get, .put = ad1983_spdif_route_put, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index d51c72b721e0..b97f50d10ba3 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1422,7 +1422,7 @@ static snd_kcontrol_new_t snd_ice1712_multi_capture_analog_switch __devinitdata static snd_kcontrol_new_t snd_ice1712_multi_capture_spdif_switch __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Multi Capture Switch", + .name = SNDRV_CTL_NAME_IEC958("Multi ",CAPTURE,SWITCH), .info = snd_ice1712_pro_mixer_switch_info, .get = snd_ice1712_pro_mixer_switch_get, .put = snd_ice1712_pro_mixer_switch_put, @@ -1441,7 +1441,7 @@ static snd_kcontrol_new_t snd_ice1712_multi_capture_analog_volume __devinitdata static snd_kcontrol_new_t snd_ice1712_multi_capture_spdif_volume __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Multi Capture Volume", + .name = SNDRV_CTL_NAME_IEC958("Multi ",CAPTURE,VOLUME), .info = snd_ice1712_pro_mixer_volume_info, .get = snd_ice1712_pro_mixer_volume_get, .put = snd_ice1712_pro_mixer_volume_put, @@ -2203,7 +2203,7 @@ static snd_kcontrol_new_t snd_ice1712_mixer_pro_analog_route __devinitdata = { static snd_kcontrol_new_t snd_ice1712_mixer_pro_spdif_route __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", .info = snd_ice1712_pro_route_info, .get = snd_ice1712_pro_route_spdif_get, .put = snd_ice1712_pro_route_spdif_put, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index e819b9ffc266..c7af5e5fee13 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1466,7 +1466,7 @@ static snd_kcontrol_new_t snd_vt1724_spdif_switch __devinitdata = .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* FIXME: the following conflict with IEC958 Playback Route */ // .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), - .name = "IEC958 Output Switch", + .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), .info = snd_vt1724_spdif_sw_info, .get = snd_vt1724_spdif_sw_get, .put = snd_vt1724_spdif_sw_put diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 6e1c29f5799a..8ddc023a5b7f 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1484,7 +1484,7 @@ static int snd_via8233_dxs3_spdif_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_val } static snd_kcontrol_new_t snd_via8233_dxs3_spdif_control __devinitdata = { - .name = "IEC958 Output Switch", + .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_via8233_dxs3_spdif_info, .get = snd_via8233_dxs3_spdif_get, -- cgit From 5549d54992391d81a8cbfbaac45a958876fbc9cb Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 3 Aug 2005 13:50:30 +0200 Subject: [ALSA] use PCM interface for IEC958 controls Digigram VX core,ENS1370/1+ driver,CA0106 driver,EMU10K1/EMU10K2 driver RME HDSP driver,RME9652 driver For consistency, use the PCM interface instead of MIXER for IEC958 default/mask/stream mixer controls. Signed-off-by: Clemens Ladisch --- sound/pci/ca0106/ca0106_mixer.c | 4 ++-- sound/pci/emu10k1/emu10k1x.c | 4 ++-- sound/pci/emu10k1/emumixer.c | 8 ++++++-- sound/pci/ens1370.c | 6 +++--- sound/pci/rme9652/hdsp.c | 8 ++++---- sound/pci/rme9652/rme9652.c | 8 ++++---- 6 files changed, 21 insertions(+), 17 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 0e5e9ce0ff28..b6b8882ce704 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -297,7 +297,7 @@ static int snd_ca0106_spdif_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_ca0106_spdif_mask_control = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), .count = 4, .info = snd_ca0106_spdif_info, @@ -306,7 +306,7 @@ static snd_kcontrol_new_t snd_ca0106_spdif_mask_control = static snd_kcontrol_new_t snd_ca0106_spdif_control = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .count = 4, .info = snd_ca0106_spdif_info, diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index e90c5ddd1d17..52c7826df440 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1183,7 +1183,7 @@ static int snd_emu10k1x_spdif_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_emu10k1x_spdif_mask_control = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), .count = 3, .info = snd_emu10k1x_spdif_info, @@ -1192,7 +1192,7 @@ static snd_kcontrol_new_t snd_emu10k1x_spdif_mask_control = static snd_kcontrol_new_t snd_emu10k1x_spdif_control = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .count = 3, .info = snd_emu10k1x_spdif_info, diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 279edaeb0663..d71a72e84bcc 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -181,7 +181,7 @@ static int snd_emu10k1_spdif_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_emu10k1_spdif_mask_control = { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), .count = 4, .info = snd_emu10k1_spdif_info, @@ -190,7 +190,7 @@ static snd_kcontrol_new_t snd_emu10k1_spdif_mask_control = static snd_kcontrol_new_t snd_emu10k1_spdif_control = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .count = 4, .info = snd_emu10k1_spdif_info, @@ -931,10 +931,14 @@ int __devinit snd_emu10k1_mixer(emu10k1_t *emu, /* sb live! and audigy */ if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu)) == NULL) return -ENOMEM; + if (!emu->audigy) + kctl->id.device = emu->pcm_efx->device; if ((err = snd_ctl_add(card, kctl))) return err; if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_control, emu)) == NULL) return -ENOMEM; + if (!emu->audigy) + kctl->id.device = emu->pcm_efx->device; if ((err = snd_ctl_add(card, kctl))) return err; } diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index bc8272be0031..f06b95f41a1d 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1446,7 +1446,7 @@ static int snd_es1371_spdif_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t static snd_kcontrol_new_t snd_es1371_mixer_spdif[] __devinitdata = { ES1371_SPDIF(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH)), { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = snd_ens1373_spdif_info, .get = snd_ens1373_spdif_default_get, @@ -1454,13 +1454,13 @@ static snd_kcontrol_new_t snd_es1371_mixer_spdif[] __devinitdata = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), .info = snd_ens1373_spdif_info, .get = snd_ens1373_spdif_mask_get }, { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), .info = snd_ens1373_spdif_info, .get = snd_ens1373_spdif_stream_get, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 60eef845a074..6694866089b5 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3104,7 +3104,7 @@ HDSP_XLR_BREAKOUT_CABLE("XLR Breakout Cable", 0) static snd_kcontrol_new_t snd_hdsp_controls[] = { { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = snd_hdsp_control_spdif_info, .get = snd_hdsp_control_spdif_get, @@ -3112,7 +3112,7 @@ static snd_kcontrol_new_t snd_hdsp_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), .info = snd_hdsp_control_spdif_stream_info, .get = snd_hdsp_control_spdif_stream_get, @@ -3120,7 +3120,7 @@ static snd_kcontrol_new_t snd_hdsp_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), .info = snd_hdsp_control_spdif_mask_info, .get = snd_hdsp_control_spdif_mask_get, @@ -3130,7 +3130,7 @@ static snd_kcontrol_new_t snd_hdsp_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), .info = snd_hdsp_control_spdif_mask_info, .get = snd_hdsp_control_spdif_mask_get, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 6e3a91dba3eb..8ee4d6fd6ea7 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1529,7 +1529,7 @@ static int snd_rme9652_get_tc_value(void *private_data, static snd_kcontrol_new_t snd_rme9652_controls[] = { { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = snd_rme9652_control_spdif_info, .get = snd_rme9652_control_spdif_get, @@ -1537,7 +1537,7 @@ static snd_kcontrol_new_t snd_rme9652_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), .info = snd_rme9652_control_spdif_stream_info, .get = snd_rme9652_control_spdif_stream_get, @@ -1545,7 +1545,7 @@ static snd_kcontrol_new_t snd_rme9652_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), .info = snd_rme9652_control_spdif_mask_info, .get = snd_rme9652_control_spdif_mask_get, @@ -1555,7 +1555,7 @@ static snd_kcontrol_new_t snd_rme9652_controls[] = { }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), .info = snd_rme9652_control_spdif_mask_info, .get = snd_rme9652_control_spdif_mask_get, -- cgit From 064d2112ff24937f9aabb6baae8de88b6e5ef453 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 5 Aug 2005 10:25:08 +0200 Subject: [ALSA] WM9713 modem detection AC97 Codec This patch fixes a problem whereby the WM9713 has modem functionality incorrectly detected after an AC97 cold reset. Changes:- o Cleared AC97_SCAP_MODEM in wm9713 scaps Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 00fb51992460..a51b61d5066b 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -786,6 +786,7 @@ int patch_wolfson13(ac97_t * ac97) ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_PHONE | AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD | AC97_HAS_NO_TONE | AC97_HAS_NO_STD_PCM; + ac97->scaps &= ~AC97_SCAP_MODEM; snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xda00); snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0x3810); -- cgit From 9bcf655109ae06a8e652671a0de6fe2da5c213c2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 10 Aug 2005 10:21:43 +0200 Subject: [ALSA] ymfpci: add per-voice volume controls YMFPCI driver Implements mixer controls for the volume of each playback substream of the main PCM device. Signed-off-by: Clemens Ladisch --- sound/pci/ymfpci/ymfpci_main.c | 232 ++++++++++++++++++++++++++++------------- 1 file changed, 160 insertions(+), 72 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index d54f88a1b525..054836412dc4 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -321,6 +321,26 @@ static void snd_ymfpci_pcm_interrupt(ymfpci_t *chip, ymfpci_voice_t *voice) snd_pcm_period_elapsed(ypcm->substream); spin_lock(&chip->reg_lock); } + + if (unlikely(ypcm->update_pcm_vol)) { + unsigned int subs = ypcm->substream->number; + unsigned int next_bank = 1 - chip->active_bank; + snd_ymfpci_playback_bank_t *bank; + u32 volume; + + bank = &voice->bank[next_bank]; + volume = cpu_to_le32(chip->pcm_mixer[subs].left << 15); + bank->left_gain_end = volume; + if (ypcm->output_rear) + bank->eff2_gain_end = volume; + if (ypcm->voices[1]) + bank = &ypcm->voices[1]->bank[next_bank]; + volume = cpu_to_le32(chip->pcm_mixer[subs].right << 15); + bank->right_gain_end = volume; + if (ypcm->output_rear) + bank->eff3_gain_end = volume; + ypcm->update_pcm_vol--; + } } spin_unlock(&chip->reg_lock); } @@ -451,87 +471,74 @@ static int snd_ymfpci_pcm_voice_alloc(ymfpci_pcm_t *ypcm, int voices) return 0; } -static void snd_ymfpci_pcm_init_voice(ymfpci_voice_t *voice, int stereo, - int rate, int w_16, unsigned long addr, - unsigned int end, - int output_front, int output_rear) +static void snd_ymfpci_pcm_init_voice(ymfpci_pcm_t *ypcm, unsigned int voiceidx, + snd_pcm_runtime_t *runtime, + int has_pcm_volume) { + ymfpci_voice_t *voice = ypcm->voices[voiceidx]; u32 format; - u32 delta = snd_ymfpci_calc_delta(rate); - u32 lpfQ = snd_ymfpci_calc_lpfQ(rate); - u32 lpfK = snd_ymfpci_calc_lpfK(rate); + u32 delta = snd_ymfpci_calc_delta(runtime->rate); + u32 lpfQ = snd_ymfpci_calc_lpfQ(runtime->rate); + u32 lpfK = snd_ymfpci_calc_lpfK(runtime->rate); snd_ymfpci_playback_bank_t *bank; unsigned int nbank; + u32 vol_left, vol_right; + u8 use_left, use_right; snd_assert(voice != NULL, return); - format = (stereo ? 0x00010000 : 0) | (w_16 ? 0 : 0x80000000); + if (runtime->channels == 1) { + use_left = 1; + use_right = 1; + } else { + use_left = (voiceidx & 1) == 0; + use_right = !use_left; + } + if (has_pcm_volume) { + vol_left = cpu_to_le32(ypcm->chip->pcm_mixer + [ypcm->substream->number].left << 15); + vol_right = cpu_to_le32(ypcm->chip->pcm_mixer + [ypcm->substream->number].right << 15); + } else { + vol_left = cpu_to_le32(0x40000000); + vol_right = cpu_to_le32(0x40000000); + } + format = runtime->channels == 2 ? 0x00010000 : 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= 0x80000000; + if (runtime->channels == 2 && (voiceidx & 1) != 0) + format |= 1; for (nbank = 0; nbank < 2; nbank++) { bank = &voice->bank[nbank]; + memset(bank, 0, sizeof(*bank)); bank->format = cpu_to_le32(format); - bank->loop_default = 0; - bank->base = cpu_to_le32(addr); - bank->loop_start = 0; - bank->loop_end = cpu_to_le32(end); - bank->loop_frac = 0; - bank->eg_gain_end = cpu_to_le32(0x40000000); + bank->base = cpu_to_le32(runtime->dma_addr); + bank->loop_end = cpu_to_le32(ypcm->buffer_size); bank->lpfQ = cpu_to_le32(lpfQ); - bank->status = 0; - bank->num_of_frames = 0; - bank->loop_count = 0; - bank->start = 0; - bank->start_frac = 0; bank->delta = bank->delta_end = cpu_to_le32(delta); bank->lpfK = bank->lpfK_end = cpu_to_le32(lpfK); - bank->eg_gain = cpu_to_le32(0x40000000); - bank->lpfD1 = - bank->lpfD2 = 0; - - bank->left_gain = - bank->right_gain = - bank->left_gain_end = - bank->right_gain_end = - bank->eff1_gain = - bank->eff2_gain = - bank->eff3_gain = - bank->eff1_gain_end = - bank->eff2_gain_end = - bank->eff3_gain_end = 0; - - if (!stereo) { - if (output_front) { - bank->left_gain = + bank->eg_gain = + bank->eg_gain_end = cpu_to_le32(0x40000000); + + if (ypcm->output_front) { + if (use_left) { + bank->left_gain = + bank->left_gain_end = vol_left; + } + if (use_right) { bank->right_gain = - bank->left_gain_end = - bank->right_gain_end = cpu_to_le32(0x40000000); + bank->right_gain_end = vol_right; } - if (output_rear) { + } + if (ypcm->output_rear) { + if (use_left) { bank->eff2_gain = - bank->eff2_gain_end = - bank->eff3_gain = - bank->eff3_gain_end = cpu_to_le32(0x40000000); - } - } else { - if (output_front) { - if ((voice->number & 1) == 0) { - bank->left_gain = - bank->left_gain_end = cpu_to_le32(0x40000000); - } else { - bank->format |= cpu_to_le32(1); - bank->right_gain = - bank->right_gain_end = cpu_to_le32(0x40000000); - } + bank->eff2_gain_end = vol_left; } - if (output_rear) { - if ((voice->number & 1) == 0) { - bank->eff3_gain = - bank->eff3_gain_end = cpu_to_le32(0x40000000); - } else { - bank->format |= cpu_to_le32(1); - bank->eff2_gain = - bank->eff2_gain_end = cpu_to_le32(0x40000000); - } + if (use_right) { + bank->eff3_gain = + bank->eff3_gain_end = vol_right; } } } @@ -613,7 +620,7 @@ static int snd_ymfpci_playback_hw_free(snd_pcm_substream_t * substream) static int snd_ymfpci_playback_prepare(snd_pcm_substream_t * substream) { - // ymfpci_t *chip = snd_pcm_substream_chip(substream); + ymfpci_t *chip = snd_pcm_substream_chip(substream); snd_pcm_runtime_t *runtime = substream->runtime; ymfpci_pcm_t *ypcm = runtime->private_data; unsigned int nvoice; @@ -623,14 +630,8 @@ static int snd_ymfpci_playback_prepare(snd_pcm_substream_t * substream) ypcm->period_pos = 0; ypcm->last_pos = 0; for (nvoice = 0; nvoice < runtime->channels; nvoice++) - snd_ymfpci_pcm_init_voice(ypcm->voices[nvoice], - runtime->channels == 2, - runtime->rate, - snd_pcm_format_width(runtime->format) == 16, - runtime->dma_addr, - ypcm->buffer_size, - ypcm->output_front, - ypcm->output_rear); + snd_ymfpci_pcm_init_voice(ypcm, nvoice, runtime, + substream->pcm == chip->pcm); return 0; } @@ -882,6 +883,7 @@ static int snd_ymfpci_playback_open(snd_pcm_substream_t * substream) ymfpci_t *chip = snd_pcm_substream_chip(substream); snd_pcm_runtime_t *runtime = substream->runtime; ymfpci_pcm_t *ypcm; + snd_kcontrol_t *kctl; int err; if ((err = snd_ymfpci_playback_open_1(substream)) < 0) @@ -895,6 +897,10 @@ static int snd_ymfpci_playback_open(snd_pcm_substream_t * substream) chip->rear_opened++; } spin_unlock_irq(&chip->reg_lock); + + kctl = chip->pcm_mixer[substream->number].ctl; + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id); return 0; } @@ -987,6 +993,7 @@ static int snd_ymfpci_playback_close(snd_pcm_substream_t * substream) { ymfpci_t *chip = snd_pcm_substream_chip(substream); ymfpci_pcm_t *ypcm = substream->runtime->private_data; + snd_kcontrol_t *kctl; spin_lock_irq(&chip->reg_lock); if (ypcm->output_rear && chip->rear_opened > 0) { @@ -994,6 +1001,9 @@ static int snd_ymfpci_playback_close(snd_pcm_substream_t * substream) ymfpci_close_extension(chip); } spin_unlock_irq(&chip->reg_lock); + kctl = chip->pcm_mixer[substream->number].ctl; + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id); return snd_ymfpci_playback_close_1(substream); } @@ -1665,6 +1675,66 @@ static snd_kcontrol_new_t snd_ymfpci_rear_shared __devinitdata = { .private_value = 2, }; +/* + * PCM voice volume + */ + +static int snd_ymfpci_pcm_vol_info(snd_kcontrol_t *kcontrol, + snd_ctl_elem_info_t *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x8000; + return 0; +} + +static int snd_ymfpci_pcm_vol_get(snd_kcontrol_t *kcontrol, + snd_ctl_elem_value_t *ucontrol) +{ + ymfpci_t *chip = snd_kcontrol_chip(kcontrol); + unsigned int subs = kcontrol->id.subdevice; + + ucontrol->value.integer.value[0] = chip->pcm_mixer[subs].left; + ucontrol->value.integer.value[1] = chip->pcm_mixer[subs].right; + return 0; +} + +static int snd_ymfpci_pcm_vol_put(snd_kcontrol_t *kcontrol, + snd_ctl_elem_value_t *ucontrol) +{ + ymfpci_t *chip = snd_kcontrol_chip(kcontrol); + unsigned int subs = kcontrol->id.subdevice; + snd_pcm_substream_t *substream; + unsigned long flags; + + if (ucontrol->value.integer.value[0] != chip->pcm_mixer[subs].left || + ucontrol->value.integer.value[1] != chip->pcm_mixer[subs].right) { + chip->pcm_mixer[subs].left = ucontrol->value.integer.value[0]; + chip->pcm_mixer[subs].right = ucontrol->value.integer.value[1]; + + substream = (snd_pcm_substream_t *)kcontrol->private_value; + spin_lock_irqsave(&chip->voice_lock, flags); + if (substream->runtime && substream->runtime->private_data) { + ymfpci_pcm_t *ypcm = substream->runtime->private_data; + ypcm->update_pcm_vol = 2; + } + spin_unlock_irqrestore(&chip->voice_lock, flags); + return 1; + } + return 0; +} + +static snd_kcontrol_new_t snd_ymfpci_pcm_volume __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, + .info = snd_ymfpci_pcm_vol_info, + .get = snd_ymfpci_pcm_vol_get, + .put = snd_ymfpci_pcm_vol_put, +}; + /* * Mixer routines @@ -1686,6 +1756,7 @@ int __devinit snd_ymfpci_mixer(ymfpci_t *chip, int rear_switch) { ac97_template_t ac97; snd_kcontrol_t *kctl; + snd_pcm_substream_t *substream; unsigned int idx; int err; static ac97_bus_ops_t ops = { @@ -1739,6 +1810,23 @@ int __devinit snd_ymfpci_mixer(ymfpci_t *chip, int rear_switch) return err; } + /* per-voice volume */ + substream = chip->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + for (idx = 0; idx < 32; ++idx) { + kctl = snd_ctl_new1(&snd_ymfpci_pcm_volume, chip); + if (!kctl) + return -ENOMEM; + kctl->id.device = chip->pcm->device; + kctl->id.subdevice = idx; + kctl->private_value = (unsigned long)substream; + if ((err = snd_ctl_add(chip->card, kctl)) < 0) + return err; + chip->pcm_mixer[idx].left = 0x8000; + chip->pcm_mixer[idx].right = 0x8000; + chip->pcm_mixer[idx].ctl = kctl; + substream = substream->next; + } + return 0; } -- cgit From 673b683a07272bdc1f757aa32784b9fcc4b3a014 Mon Sep 17 00:00:00 2001 From: Sasha Khapyorsky Date: Thu, 11 Aug 2005 11:00:16 +0200 Subject: ALSA CVS update HDA Codec driver,HDA generic driver Summar: hda-codec - MFG support This adds Modem Functional Group (MFG) support and option for 9600 sample rate. Signed-off-by: Sasha Khapyorsky Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 28 ++++++++++++++++------------ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_generic.c | 5 +++++ 3 files changed, 22 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 026ae726d875..6bfb081d12dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -432,22 +432,26 @@ void snd_hda_get_codec_name(struct hda_codec *codec, } /* - * look for an AFG node - * - * return 0 if not found + * look for an AFG and MFG nodes */ -static int look_for_afg_node(struct hda_codec *codec) +static void setup_fg_nodes(struct hda_codec *codec) { int i, total_nodes; hda_nid_t nid; total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - if ((snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff) == - AC_GRP_AUDIO_FUNCTION) - return nid; + switch((snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff)) { + case AC_GRP_AUDIO_FUNCTION: + codec->afg = nid; + break; + case AC_GRP_MODEM_FUNCTION: + codec->mfg = nid; + break; + default: + break; + } } - return 0; } /* @@ -507,10 +511,9 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, codec->subsystem_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_SUBSYSTEM_ID); codec->revision_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_REV_ID); - /* FIXME: support for multiple AFGs? */ - codec->afg = look_for_afg_node(codec); - if (! codec->afg) { - snd_printdd("hda_codec: no AFG node found\n"); + setup_fg_nodes(codec); + if (! codec->afg && ! codec->mfg) { + snd_printdd("hda_codec: no AFG or MFG node found\n"); snd_hda_codec_free(codec); return -ENODEV; } @@ -1163,6 +1166,7 @@ int snd_hda_build_controls(struct hda_bus *bus) static unsigned int rate_bits[][3] = { /* rate in Hz, ALSA rate bitmask, HDA format value */ { 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */ + { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */ { 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */ { 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */ { 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index dd0d99d2ad27..63a29a8a2860 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -514,6 +514,7 @@ struct hda_codec { struct list_head list; /* list point */ hda_nid_t afg; /* AFG node id */ + hda_nid_t mfg; /* MFG node id */ /* ids */ u32 vendor_id; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2d046abb5911..1229227af5b5 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -881,6 +881,11 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec) struct hda_gspec *spec; int err; + if(!codec->afg) { + snd_printdd("hda_generic: no generic modem yet\n"); + return -ENODEV; + } + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) { printk(KERN_ERR "hda_generic: can't allocate spec\n"); -- cgit From b65f824c1ea954ea2b974e42c064f72bfbfe3dd2 Mon Sep 17 00:00:00 2001 From: Sasha Khapyorsky Date: Thu, 11 Aug 2005 11:18:38 +0200 Subject: [ALSA] hda-codec - support for Si3054/5 HDA modems HDA generic driver,HDA Codec driver Support for Si3054/5 HDA modem codecs. Signed-off-by: Sasha Khapyorsky Signed-off-by: Takashi Iwai --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_patch.h | 3 + sound/pci/hda/patch_si3054.c | 300 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 304 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/patch_si3054.c (limited to 'sound/pci') diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index bd8cb33c4fb4..ddfb5ff7fb8f 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,5 @@ snd-hda-intel-objs := hda_intel.o -snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o +snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o ifdef CONFIG_PROC_FS snd-hda-codec-objs += hda_proc.o endif diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h index a5de684b6944..acaef3c811b8 100644 --- a/sound/pci/hda/hda_patch.h +++ b/sound/pci/hda/hda_patch.h @@ -10,11 +10,14 @@ extern struct hda_codec_preset snd_hda_preset_cmedia[]; extern struct hda_codec_preset snd_hda_preset_analog[]; /* SigmaTel codecs */ extern struct hda_codec_preset snd_hda_preset_sigmatel[]; +/* SiLabs 3054/3055 modem codecs */ +extern struct hda_codec_preset snd_hda_preset_si3054[]; static const struct hda_codec_preset *hda_preset_tables[] = { snd_hda_preset_realtek, snd_hda_preset_cmedia, snd_hda_preset_analog, snd_hda_preset_sigmatel, + snd_hda_preset_si3054, NULL }; diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c new file mode 100644 index 000000000000..b0270d1b64ce --- /dev/null +++ b/sound/pci/hda/patch_si3054.c @@ -0,0 +1,300 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * HD audio interface patch for Silicon Labs 3054/5 modem codec + * + * Copyright (c) 2005 Sasha Khapyorsky + * Takashi Iwai + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + + +/* si3054 verbs */ +#define SI3054_VERB_READ_NODE 0x900 +#define SI3054_VERB_WRITE_NODE 0x100 + +/* si3054 nodes (registers) */ +#define SI3054_EXTENDED_MID 2 +#define SI3054_LINE_RATE 3 +#define SI3054_LINE_LEVEL 4 +#define SI3054_GPIO_CFG 5 +#define SI3054_GPIO_POLARITY 6 +#define SI3054_GPIO_STICKY 7 +#define SI3054_GPIO_WAKEUP 8 +#define SI3054_GPIO_STATUS 9 +#define SI3054_GPIO_CONTROL 10 +#define SI3054_MISC_AFE 11 +#define SI3054_CHIPID 12 +#define SI3054_LINE_CFG1 13 +#define SI3054_LINE_STATUS 14 +#define SI3054_DC_TERMINATION 15 +#define SI3054_LINE_CONFIG 16 +#define SI3054_CALLPROG_ATT 17 +#define SI3054_SQ_CONTROL 18 +#define SI3054_MISC_CONTROL 19 +#define SI3054_RING_CTRL1 20 +#define SI3054_RING_CTRL2 21 + +/* extended MID */ +#define SI3054_MEI_READY 0xf + +/* line level */ +#define SI3054_ATAG_MASK 0x00f0 +#define SI3054_DTAG_MASK 0xf000 + +/* GPIO bits */ +#define SI3054_GPIO_OH 0x0001 +#define SI3054_GPIO_CID 0x0002 + +/* chipid and revisions */ +#define SI3054_CHIPID_CODEC_REV_MASK 0x000f +#define SI3054_CHIPID_DAA_REV_MASK 0x00f0 +#define SI3054_CHIPID_INTERNATIONAL 0x0100 +#define SI3054_CHIPID_DAA_ID 0x0f00 +#define SI3054_CHIPID_CODEC_ID (1<<12) + +/* si3054 codec registers (nodes) access macros */ +#define GET_REG(codec,reg) (snd_hda_codec_read(codec,reg,0,SI3054_VERB_READ_NODE,0)) +#define SET_REG(codec,reg,val) (snd_hda_codec_write(codec,reg,0,SI3054_VERB_WRITE_NODE,val)) + + +struct si3054_spec { + unsigned international; + struct hda_pcm pcm; +}; + + +/* + * Modem mixer + */ + +#define PRIVATE_VALUE(reg,mask) ((reg<<16)|(mask&0xffff)) +#define PRIVATE_REG(val) ((val>>16)&0xffff) +#define PRIVATE_MASK(val) (val&0xffff) + +static int si3054_switch_info(snd_kcontrol_t *kcontrol, + snd_ctl_elem_info_t *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int si3054_switch_get(snd_kcontrol_t *kcontrol, + snd_ctl_elem_value_t *uvalue) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + u16 reg = PRIVATE_REG(kcontrol->private_value); + u16 mask = PRIVATE_MASK(kcontrol->private_value); + uvalue->value.integer.value[0] = (GET_REG(codec, reg)) & mask ? 1 : 0 ; + return 0; +} + +static int si3054_switch_put(snd_kcontrol_t *kcontrol, + snd_ctl_elem_value_t *uvalue) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + u16 reg = PRIVATE_REG(kcontrol->private_value); + u16 mask = PRIVATE_MASK(kcontrol->private_value); + if (uvalue->value.integer.value[0]) + SET_REG(codec, reg, (GET_REG(codec, reg)) | mask); + else + SET_REG(codec, reg, (GET_REG(codec, reg)) & ~mask); + return 0; +} + +#define SI3054_KCONTROL(kname,reg,mask) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = kname, \ + .info = si3054_switch_info, \ + .get = si3054_switch_get, \ + .put = si3054_switch_put, \ + .private_value = PRIVATE_VALUE(reg,mask), \ +} + + +static snd_kcontrol_new_t si3054_modem_mixer[] = { + SI3054_KCONTROL("Off-hook Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_OH), + SI3054_KCONTROL("Caller ID Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_CID), + {} +}; + +static int si3054_build_controls(struct hda_codec *codec) +{ + return snd_hda_add_new_ctls(codec, si3054_modem_mixer); +} + + +/* + * PCM callbacks + */ + +static int si3054_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) +{ + u16 val; + + SET_REG(codec, SI3054_LINE_RATE, substream->runtime->rate); + val = GET_REG(codec, SI3054_LINE_LEVEL); + val &= 0xff << (8 * (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)); + val |= ((stream_tag & 0xf) << 4) << (8 * (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)); + SET_REG(codec, SI3054_LINE_LEVEL, val); + + snd_hda_codec_setup_stream(codec, hinfo->nid, + stream_tag, 0, format); + return 0; +} + +static int si3054_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) +{ + static unsigned int rates[] = { 8000, 9600, 16000 }; + static snd_pcm_hw_constraint_list_t hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + substream->runtime->hw.period_bytes_min = 80; + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); +} + + +static struct hda_pcm_stream si3054_pcm = { + .substreams = 1, + .channels_min = 1, + .channels_max = 1, + .nid = 0x1, + .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .maxbps = 16, + .ops = { + .open = si3054_pcm_open, + .prepare = si3054_pcm_prepare, + }, +}; + + +static int si3054_build_pcms(struct hda_codec *codec) +{ + struct si3054_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm; + si3054_pcm.nid = codec->mfg; + codec->num_pcms = 1; + codec->pcm_info = info; + info->name = "Si3054 Modem"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; + return 0; +} + + +/* + * Init part + */ + +static int si3054_init(struct hda_codec *codec) +{ + struct si3054_spec *spec = codec->spec; + unsigned wait_count; + u16 val; + + snd_hda_codec_write(codec, AC_NODE_ROOT, 0, AC_VERB_SET_CODEC_RESET, 0); + snd_hda_codec_write(codec, codec->mfg, 0, AC_VERB_SET_STREAM_FORMAT, 0); + SET_REG(codec, SI3054_LINE_RATE, 9600); + SET_REG(codec, SI3054_LINE_LEVEL, SI3054_DTAG_MASK|SI3054_ATAG_MASK); + SET_REG(codec, SI3054_EXTENDED_MID, 0); + + wait_count = 10; + do { + msleep(2); + val = GET_REG(codec, SI3054_EXTENDED_MID); + } while ((val & SI3054_MEI_READY) != SI3054_MEI_READY && wait_count--); + + if((val&SI3054_MEI_READY) != SI3054_MEI_READY) { + snd_printk(KERN_ERR "si3054: cannot initialize. EXT MID = %04x\n", val); + return -EACCES; + } + + SET_REG(codec, SI3054_GPIO_POLARITY, 0xffff); + SET_REG(codec, SI3054_GPIO_CFG, 0x0); + SET_REG(codec, SI3054_MISC_AFE, 0); + SET_REG(codec, SI3054_LINE_CFG1,0x200); + + if((GET_REG(codec,SI3054_LINE_STATUS) & (1<<6)) == 0) { + snd_printd("Link Frame Detect(FDT) is not ready (line status: %04x)\n", + GET_REG(codec,SI3054_LINE_STATUS)); + } + + spec->international = GET_REG(codec, SI3054_CHIPID) & SI3054_CHIPID_INTERNATIONAL; + + return 0; +} + +static void si3054_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + + +/* + */ + +static struct hda_codec_ops si3054_patch_ops = { + .build_controls = si3054_build_controls, + .build_pcms = si3054_build_pcms, + .init = si3054_init, + .free = si3054_free, +#ifdef CONFIG_PM + //.suspend = si3054_suspend, + .resume = si3054_init, +#endif +}; + +static int patch_si3054(struct hda_codec *codec) +{ + struct si3054_spec *spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + codec->spec = spec; + codec->patch_ops = si3054_patch_ops; + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_si3054[] = { + { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, + {} +}; + -- cgit From a53fc188ec6fc406276799da465fe789c40d96b2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 11 Aug 2005 15:59:17 +0200 Subject: [ALSA] make local objects static Memalloc module,PCM Midlevel,Timer Midlevel,GUS Library,AC97 Codec ALI5451 driver,RME9652 driver Make some functions/variables that are used in only one file static. Signed-off-by: Clemens Ladisch --- sound/pci/ac97/ac97_patch.c | 8 ++++---- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/rme9652/hdspm.c | 4 ++-- 3 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a51b61d5066b..b0b31f4015aa 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -375,7 +375,7 @@ AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1), AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1), }; -int patch_wolfson_wm9703_specific(ac97_t * ac97) +static int patch_wolfson_wm9703_specific(ac97_t * ac97) { /* This is known to work for the ViewSonic ViewPad 1000 * Randolph Bentson @@ -410,7 +410,7 @@ AC97_DOUBLE("Rear DAC Volume", AC97_WM9704_RPCM_VOL, 8, 0, 31, 1), AC97_DOUBLE("Surround Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), }; -int patch_wolfson_wm9704_specific(ac97_t * ac97) +static int patch_wolfson_wm9704_specific(ac97_t * ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm9704_snd_ac97_controls); i++) { @@ -433,7 +433,7 @@ int patch_wolfson04(ac97_t * ac97) return 0; } -int patch_wolfson_wm9705_specific(ac97_t * ac97) +static int patch_wolfson_wm9705_specific(ac97_t * ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { @@ -558,7 +558,7 @@ AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), }; -int patch_wolfson_wm9711_specific(ac97_t * ac97) +static int patch_wolfson_wm9711_specific(ac97_t * ac97) { int err, i; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index cb12d780a6c6..ce6c9fadb594 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1842,7 +1842,7 @@ static int __devinit snd_ali_pcm(ali_t * codec, int device, struct ali_pcm_descr return 0; } -struct ali_pcm_description ali_pcms[] = { +static struct ali_pcm_description ali_pcms[] = { { "ALI 5451", ALI_CHANNELS, 1, &snd_ali_playback_ops, &snd_ali_capture_ops }, { "ALI 5451 modem", 1, 1, &snd_ali_modem_playback_ops, &snd_ali_modem_capture_ops } }; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f6daec4e4d97..d4a0c2c56cdb 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1104,14 +1104,14 @@ static int snd_hdspm_midi_output_close(snd_rawmidi_substream_t * substream) return 0; } -snd_rawmidi_ops_t snd_hdspm_midi_output = +static snd_rawmidi_ops_t snd_hdspm_midi_output = { .open = snd_hdspm_midi_output_open, .close = snd_hdspm_midi_output_close, .trigger = snd_hdspm_midi_output_trigger, }; -snd_rawmidi_ops_t snd_hdspm_midi_input = +static snd_rawmidi_ops_t snd_hdspm_midi_input = { .open = snd_hdspm_midi_input_open, .close = snd_hdspm_midi_input_close, -- cgit From 47672310789431c16de22f16934febf4d55b4bb0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Aug 2005 16:44:04 +0200 Subject: [ALSA] hda-intel - Add SiS966 support HDA Intel driver Added SiS966 pci id to snd-hda-intel driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 288ab0764830..2b6bd3139b86 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -71,7 +71,8 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ESB2}," "{ATI, SB450}," "{VIA, VT8251}," - "{VIA, VT8237A}}"); + "{VIA, VT8237A}," + "{SiS, SIS966}}"); MODULE_DESCRIPTION("Intel HDA driver"); #define SFX "hda-intel: " @@ -1464,6 +1465,7 @@ static struct pci_device_id azx_ids[] = { { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ESB2 */ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ATI SB450 */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* VIA VT8251/VT8237A */ + { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ALI 5461? */ { 0, } }; -- cgit From 4b4994869d68328276f6a3cc9df90cce8c82b43d Mon Sep 17 00:00:00 2001 From: Philip Prindeville Date: Fri, 12 Aug 2005 16:46:17 +0200 Subject: [ALSA] Add VT1617A codec support AC97 Codec Added (minimal) support of VT1617A codec with SPDIF. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 6 ++++-- sound/pci/ac97/ac97_patch.c | 18 ++++++++++++++---- sound/pci/ac97/ac97_patch.h | 1 + 3 files changed, 19 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 33dba10f03e8..600e053dfd35 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -157,6 +157,7 @@ static const ac97_codec_id_t snd_ac97_codec_ids[] = { { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF +{ 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -2580,8 +2581,6 @@ int snd_ac97_tune_hardware(ac97_t *ac97, struct ac97_quirk *quirk, const char *o { int result; - snd_assert(quirk, return -EINVAL); - /* quirk overriden? */ if (override && strcmp(override, "-1") && strcmp(override, "default")) { result = apply_quirk_str(ac97, override); @@ -2590,6 +2589,9 @@ int snd_ac97_tune_hardware(ac97_t *ac97, struct ac97_quirk *quirk, const char *o return result; } + if (! quirk) + return -EINVAL; + for (; quirk->subvendor; quirk++) { if (quirk->subvendor != ac97->subsystem_vendor) continue; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index b0b31f4015aa..a0d515536177 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -375,7 +375,7 @@ AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1), AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1), }; -static int patch_wolfson_wm9703_specific(ac97_t * ac97) +int patch_wolfson_wm9703_specific(ac97_t * ac97) { /* This is known to work for the ViewSonic ViewPad 1000 * Randolph Bentson @@ -410,7 +410,7 @@ AC97_DOUBLE("Rear DAC Volume", AC97_WM9704_RPCM_VOL, 8, 0, 31, 1), AC97_DOUBLE("Surround Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), }; -static int patch_wolfson_wm9704_specific(ac97_t * ac97) +int patch_wolfson_wm9704_specific(ac97_t * ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm9704_snd_ac97_controls); i++) { @@ -433,7 +433,7 @@ int patch_wolfson04(ac97_t * ac97) return 0; } -static int patch_wolfson_wm9705_specific(ac97_t * ac97) +int patch_wolfson_wm9705_specific(ac97_t * ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { @@ -558,7 +558,7 @@ AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), }; -static int patch_wolfson_wm9711_specific(ac97_t * ac97) +int patch_wolfson_wm9711_specific(ac97_t * ac97) { int err, i; @@ -2665,6 +2665,16 @@ int patch_vt1616(ac97_t * ac97) return 0; } +/* + * VT1617A codec + */ +int patch_vt1617a(ac97_t * ac97) +{ + ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ + ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + return 0; +} + /* */ static void it2646_update_jacks(ac97_t *ac97) diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h index 7b7377d0f2ae..ec1811320106 100644 --- a/sound/pci/ac97/ac97_patch.h +++ b/sound/pci/ac97/ac97_patch.h @@ -56,5 +56,6 @@ int patch_cm9739(ac97_t * ac97); int patch_cm9761(ac97_t * ac97); int patch_cm9780(ac97_t * ac97); int patch_vt1616(ac97_t * ac97); +int patch_vt1617a(ac97_t * ac97); int patch_it2646(ac97_t * ac97); int mpatch_si3036(ac97_t * ac97); -- cgit From f347c774c37c83e7b2d2b75d962ca40b9a283b2b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Aug 2005 16:47:49 +0200 Subject: [ALSA] via82xx - Add DXS entry for MSI K8MM-V VIA82xx driver Added the DXS entry for MSI K8MM-V. Also fixed a typo in last patch for Acer Aspire 1524 WLMi. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8ddc023a5b7f..38b96eabea60 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2153,7 +2153,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1019, .subdevice = 0x0a81, .action = VIA_DXS_NO_VRA }, /* ECS K7VTA3 v8.0 */ { .subvendor = 0x1019, .subdevice = 0x0a85, .action = VIA_DXS_NO_VRA }, /* ECS L7VMM2 */ { .subvendor = 0x1025, .subdevice = 0x0033, .action = VIA_DXS_NO_VRA }, /* Acer Inspire 1353LM */ - { .subvendor = 0x1025, .subdevice = 0x0046, .action = VIA_DXS_SRC }, /* Acer Aspire 1524 WMLi */ + { .subvendor = 0x1025, .subdevice = 0x0046, .action = VIA_DXS_SRC }, /* Acer Aspire 1524 WLMi */ { .subvendor = 0x1043, .subdevice = 0x8095, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8X (FIXME: possibly VIA_DXS_ENABLE?)*/ { .subvendor = 0x1043, .subdevice = 0x80a1, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8-X */ { .subvendor = 0x1043, .subdevice = 0x80b0, .action = VIA_DXS_NO_VRA }, /* ASUS A7V600 & K8V*/ @@ -2173,6 +2173,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1462, .subdevice = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */ { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */ { .subvendor = 0x1462, .subdevice = 0x7120, .action = VIA_DXS_ENABLE }, /* MSI KT4V */ + { .subvendor = 0x1462, .subdevice = 0x7142, .action = VIA_DXS_ENABLE }, /* MSI K8MM-V */ { .subvendor = 0x147b, .subdevice = 0x1401, .action = VIA_DXS_ENABLE }, /* ABIT KD7(-RAID) */ { .subvendor = 0x147b, .subdevice = 0x1411, .action = VIA_DXS_ENABLE }, /* ABIT VA-20 */ { .subvendor = 0x147b, .subdevice = 0x1413, .action = VIA_DXS_ENABLE }, /* ABIT KV8 Pro */ -- cgit From 1b05962e8b2d8a1b1f5934087a4a00f7532fa2d1 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Fri, 12 Aug 2005 23:50:13 +0200 Subject: [ALSA] Add new ID. Fixes ALSA bug #1298 CA0106 driver Signed-off-by: James Courtier-Dutton --- sound/pci/ca0106/ca0106_main.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 95c289284267..7e27bfc37439 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -188,6 +188,14 @@ static ca0106_details_t ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 1, .i2c_adc = 1 } , + /* Shuttle XPC SD31P which has an onboard Creative Labs Sound Blaster Live! 24-bit EAX + * high-definition 7.1 audio processor". + * Added using info from andrewvegan in alsa bug #1298 + */ + { .serial = 0x30381297, + .name = "Shuttle XPC SD31P [SD31P]", + .gpio_type = 1, + .i2c_adc = 1 } , { .serial = 0, .name = "AudigyLS [Unknown]" } }; -- cgit From 99250872fc619bb5b5ddddcf1c58714a774526fc Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Fri, 12 Aug 2005 23:54:58 +0200 Subject: [ALSA] Add new card ID. Fixes ALSA bug #1297 EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1_main.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 746b51ef3966..c0b67b70e345 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -747,11 +747,11 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , + /* Tested by alsa bugtrack user "hus" 12th Sept 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]", .id = "Live", .emu10k1_chip = 1, - .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", -- cgit From 3e6c6139f0cf9c196c3fd700055f2e22ca22e726 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 15 Aug 2005 09:13:32 +0200 Subject: [ALSA] ac97: make patch_wolfson_wm97??_specific() functions static AC97 Codec Make the patch_wolfson_wm97??_specific() functions static again. This patch was accidentally reverted in rev. 1.92. Signed-off-by: Clemens Ladisch --- sound/pci/ac97/ac97_patch.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a0d515536177..616504ef3702 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -375,7 +375,7 @@ AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1), AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1), }; -int patch_wolfson_wm9703_specific(ac97_t * ac97) +static int patch_wolfson_wm9703_specific(ac97_t * ac97) { /* This is known to work for the ViewSonic ViewPad 1000 * Randolph Bentson @@ -410,7 +410,7 @@ AC97_DOUBLE("Rear DAC Volume", AC97_WM9704_RPCM_VOL, 8, 0, 31, 1), AC97_DOUBLE("Surround Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), }; -int patch_wolfson_wm9704_specific(ac97_t * ac97) +static int patch_wolfson_wm9704_specific(ac97_t * ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm9704_snd_ac97_controls); i++) { @@ -433,7 +433,7 @@ int patch_wolfson04(ac97_t * ac97) return 0; } -int patch_wolfson_wm9705_specific(ac97_t * ac97) +static int patch_wolfson_wm9705_specific(ac97_t * ac97) { int err, i; for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { @@ -558,7 +558,7 @@ AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), }; -int patch_wolfson_wm9711_specific(ac97_t * ac97) +static int patch_wolfson_wm9711_specific(ac97_t * ac97) { int err, i; -- cgit From 16d3f140fc265c9b9c0f8975e0b36fe15912508f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Aug 2005 15:02:28 +0200 Subject: [ALSA] via82xx - Fix SPDIF sample rates VIA82xx driver Fixed the sample rates set in the fourth DXS channel with Non-VRA mode. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 38b96eabea60..819f27b2762b 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -929,12 +929,12 @@ static int snd_via8233_playback_prepare(snd_pcm_substream_t *substream) if ((rate_changed = via_lock_rate(&chip->rates[0], ac97_rate)) < 0) return rate_changed; - if (rate_changed) { + if (rate_changed) snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, chip->no_vra ? 48000 : runtime->rate); - snd_ac97_set_rate(chip->ac97, AC97_SPDIF, - chip->no_vra ? 48000 : runtime->rate); - } + if (chip->spdif_on && viadev->reg_offset == 0x30) + snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate); + if (runtime->rate == 48000) rbits = 0xfffff; else -- cgit From 5a47fe3c5e8f394fb1578bbe8117f3dcc250b6fd Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 15 Aug 2005 20:01:40 +0200 Subject: [ALSA] hda-patch-realtek - added pci subdevice ID for Acer TravelMate 8100 (3 stack model + digital out) HDA Codec driver Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b8569900787..d19da2bae663 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1524,6 +1524,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { /* Back 3 jack plus 1 SPDIF out jack, front 2 jack */ { .modelname = "3stack-digout", .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x0070, .config = ALC880_3ST_DIG }, /* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG }, -- cgit From 47123197c5522f4ae3dc5914e7832dd047f9ddc8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 15 Aug 2005 20:53:07 +0200 Subject: [ALSA] hda-intel: Suspend/resume fixes for PCM devices HDA Intel driver - removed SNDRV_PCM_INFO_RESUME (the driver cannot do PCM resume at the time) - fixed chip->pcm_devs initialization Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6bd3139b86..58e15b8896ac 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -900,8 +900,8 @@ static snd_pcm_hardware_t azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_PAUSE /*|*/ + /*SNDRV_PCM_INFO_RESUME*/), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -1050,6 +1050,7 @@ static int azx_pcm_trigger(snd_pcm_substream_t *substream, int cmd) azx_dev->running = 1; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: azx_stream_stop(chip, azx_dev); azx_dev->running = 0; @@ -1059,6 +1060,7 @@ static int azx_pcm_trigger(snd_pcm_substream_t *substream, int cmd) } spin_unlock(&chip->reg_lock); if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH || + cmd == SNDRV_PCM_TRIGGER_SUSPEND || cmd == SNDRV_PCM_TRIGGER_STOP) { int timeout = 5000; while (azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START && --timeout) @@ -1137,6 +1139,7 @@ static int __devinit create_codec_pcm(azx_t *chip, struct hda_codec *codec, snd_dma_pci_data(chip->pci), 1024 * 64, 1024 * 128); chip->pcm[pcm_dev] = pcm; + chip->pcm_devs = pcm_dev + 1; return 0; } -- cgit From 1cfe43d21bc5ff751e95b6a62410e05956c09e38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Aug 2005 16:52:24 +0200 Subject: [ALSA] intel8x0 - Fix PM Intel8x0 driver Fixed the PCM resume of intel8x0. Restores the requested register setting. Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 39 +++++++++++++++++++++++++++++++-------- 1 file changed, 31 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index d7af3e474432..390b6c8f9a10 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -389,6 +389,7 @@ typedef struct { struct ac97_pcm *pcm; int pcm_open_flag; unsigned int page_attr_changed: 1; + unsigned int suspended: 1; } ichdev_t; typedef struct _snd_intel8x0 intel8x0_t; @@ -862,12 +863,16 @@ static int snd_intel8x0_pcm_trigger(snd_pcm_substream_t *substream, int cmd) unsigned long port = ichdev->reg_offset; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + ichdev->suspended = 0; + /* fallthru */ + case SNDRV_PCM_TRIGGER_START: val = ICH_IOCE | ICH_STARTBM; break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + ichdev->suspended = 1; + /* fallthru */ + case SNDRV_PCM_TRIGGER_STOP: val = 0; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -899,9 +904,11 @@ static int snd_intel8x0_ali_trigger(snd_pcm_substream_t *substream, int cmd) val = igetdword(chip, ICHREG(ALI_DMACR)); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + ichdev->suspended = 0; + /* fallthru */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* clear FIFO for synchronization of channels */ fifo = igetdword(chip, fiforeg[ichdev->ali_slot / 4]); @@ -913,9 +920,11 @@ static int snd_intel8x0_ali_trigger(snd_pcm_substream_t *substream, int cmd) val &= ~(1 << (ichdev->ali_slot + 16)); /* clear PAUSE flag */ iputdword(chip, ICHREG(ALI_DMACR), val | (1 << ichdev->ali_slot)); /* start DMA */ break; + case SNDRV_PCM_TRIGGER_SUSPEND: + ichdev->suspended = 1; + /* fallthru */ case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: iputdword(chip, ICHREG(ALI_DMACR), val | (1 << (ichdev->ali_slot + 16))); /* pause */ iputbyte(chip, port + ICH_REG_OFF_CR, 0); while (igetbyte(chip, port + ICH_REG_OFF_CR)) @@ -994,6 +1003,8 @@ static void snd_intel8x0_setup_pcm_out(intel8x0_t *chip, { unsigned int cnt; int dbl = runtime->rate > 48000; + + spin_lock_irq(&chip->reg_lock); switch (chip->device_type) { case DEVICE_ALI: cnt = igetdword(chip, ICHREG(ALI_SCR)); @@ -1037,6 +1048,7 @@ static void snd_intel8x0_setup_pcm_out(intel8x0_t *chip, iputdword(chip, ICHREG(GLOB_CNT), cnt); break; } + spin_unlock_irq(&chip->reg_lock); } static int snd_intel8x0_pcm_prepare(snd_pcm_substream_t * substream) @@ -1048,15 +1060,12 @@ static int snd_intel8x0_pcm_prepare(snd_pcm_substream_t * substream) ichdev->physbuf = runtime->dma_addr; ichdev->size = snd_pcm_lib_buffer_bytes(substream); ichdev->fragsize = snd_pcm_lib_period_bytes(substream); - spin_lock_irq(&chip->reg_lock); if (ichdev->ichd == ICHD_PCMOUT) { snd_intel8x0_setup_pcm_out(chip, runtime); - if (chip->device_type == DEVICE_INTEL_ICH4) { + if (chip->device_type == DEVICE_INTEL_ICH4) ichdev->pos_shift = (runtime->sample_bits > 16) ? 2 : 1; - } } snd_intel8x0_setup_periods(chip, ichdev); - spin_unlock_irq(&chip->reg_lock); return 0; } @@ -2424,6 +2433,20 @@ static int intel8x0_resume(snd_card_t *card) } } + /* resume status */ + for (i = 0; i < chip->bdbars_count; i++) { + ichdev_t *ichdev = &chip->ichd[i]; + unsigned long port = ichdev->reg_offset; + if (! ichdev->substream || ! ichdev->suspended) + continue; + if (ichdev->ichd == ICHD_PCMOUT) + snd_intel8x0_setup_pcm_out(chip, ichdev->substream->runtime); + iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr); + iputbyte(chip, port + ICH_REG_OFF_LVI, ichdev->lvi); + iputbyte(chip, port + ICH_REG_OFF_CIV, ichdev->civ); + iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); + } + return 0; } #endif /* CONFIG_PM */ -- cgit From 1204de32d0df87892e56062042e25c775ca0e08c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Aug 2005 16:54:12 +0200 Subject: [ALSA] nm256 - Fix PM and irq handling NM256 driver - Fixed the PCM resume - restoring the rate setting - Fixed the handling of buggy irqs - Dynamically acquire/release irq handler to make the driver more robust to unknown irq storms (as OSS driver does). Signed-off-by: Takashi Iwai --- sound/pci/nm256/nm256.c | 93 +++++++++++++++++++++++++++++++++++++------------ 1 file changed, 70 insertions(+), 23 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 7eb20b8f89f6..2bbeb10ff7c4 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -189,6 +189,7 @@ struct snd_nm256_stream { nm256_t *chip; snd_pcm_substream_t *substream; int running; + int suspended; u32 buf; /* offset from chip->buffer */ int bufsize; /* buffer size in bytes */ @@ -231,8 +232,10 @@ struct snd_nm256 { int mixer_status_mask; /* bit mask to test the mixer status */ int irq; + int irq_acks; irqreturn_t (*interrupt)(int, void *, struct pt_regs *); int badintrcount; /* counter to check bogus interrupts */ + struct semaphore irq_mutex; nm256_stream_t streams[2]; @@ -464,6 +467,37 @@ snd_nm256_set_format(nm256_t *chip, nm256_stream_t *s, snd_pcm_substream_t *subs } } +/* acquire interrupt */ +static int snd_nm256_acquire_irq(nm256_t *chip) +{ + down(&chip->irq_mutex); + if (chip->irq < 0) { + if (request_irq(chip->pci->irq, chip->interrupt, SA_INTERRUPT|SA_SHIRQ, + chip->card->driver, (void*)chip)) { + snd_printk("unable to grab IRQ %d\n", chip->pci->irq); + up(&chip->irq_mutex); + return -EBUSY; + } + chip->irq = chip->pci->irq; + } + chip->irq_acks++; + up(&chip->irq_mutex); + return 0; +} + +/* release interrupt */ +static void snd_nm256_release_irq(nm256_t *chip) +{ + down(&chip->irq_mutex); + if (chip->irq_acks > 0) + chip->irq_acks--; + if (chip->irq_acks == 0 && chip->irq >= 0) { + free_irq(chip->irq, (void*)chip); + chip->irq = -1; + } + up(&chip->irq_mutex); +} + /* * start / stop */ @@ -538,15 +572,19 @@ snd_nm256_playback_trigger(snd_pcm_substream_t *substream, int cmd) spin_lock(&chip->reg_lock); switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + s->suspended = 0; + /* fallthru */ + case SNDRV_PCM_TRIGGER_START: if (! s->running) { snd_nm256_playback_start(chip, s, substream); s->running = 1; } break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + s->suspended = 1; + /* fallthru */ + case SNDRV_PCM_TRIGGER_STOP: if (s->running) { snd_nm256_playback_stop(chip); s->running = 0; @@ -818,6 +856,8 @@ snd_nm256_playback_open(snd_pcm_substream_t *substream) { nm256_t *chip = snd_pcm_substream_chip(substream); + if (snd_nm256_acquire_irq(chip) < 0) + return -EBUSY; snd_nm256_setup_stream(chip, &chip->streams[SNDRV_PCM_STREAM_PLAYBACK], substream, &snd_nm256_playback); return 0; @@ -828,6 +868,8 @@ snd_nm256_capture_open(snd_pcm_substream_t *substream) { nm256_t *chip = snd_pcm_substream_chip(substream); + if (snd_nm256_acquire_irq(chip) < 0) + return -EBUSY; snd_nm256_setup_stream(chip, &chip->streams[SNDRV_PCM_STREAM_CAPTURE], substream, &snd_nm256_capture); return 0; @@ -839,6 +881,9 @@ snd_nm256_capture_open(snd_pcm_substream_t *substream) static int snd_nm256_playback_close(snd_pcm_substream_t *substream) { + nm256_t *chip = snd_pcm_substream_chip(substream); + + snd_nm256_release_irq(chip); return 0; } @@ -846,6 +891,9 @@ snd_nm256_playback_close(snd_pcm_substream_t *substream) static int snd_nm256_capture_close(snd_pcm_substream_t *substream) { + nm256_t *chip = snd_pcm_substream_chip(substream); + + snd_nm256_release_irq(chip); return 0; } @@ -915,18 +963,16 @@ snd_nm256_pcm(nm256_t *chip, int device) static void snd_nm256_init_chip(nm256_t *chip) { - spin_lock_irq(&chip->reg_lock); /* Reset everything. */ snd_nm256_writeb(chip, 0x0, 0x11); snd_nm256_writew(chip, 0x214, 0); /* stop sounds.. */ //snd_nm256_playback_stop(chip); //snd_nm256_capture_stop(chip); - spin_unlock_irq(&chip->reg_lock); } -static inline void +static irqreturn_t snd_nm256_intr_check(nm256_t *chip) { if (chip->badintrcount++ > 1000) { @@ -947,7 +993,9 @@ snd_nm256_intr_check(nm256_t *chip) if (chip->streams[SNDRV_PCM_STREAM_CAPTURE].running) snd_nm256_capture_stop(chip); chip->badintrcount = 0; + return IRQ_HANDLED; } + return IRQ_NONE; } /* @@ -969,10 +1017,8 @@ snd_nm256_interrupt(int irq, void *dev_id, struct pt_regs *dummy) status = snd_nm256_readw(chip, NM_INT_REG); /* Not ours. */ - if (status == 0) { - snd_nm256_intr_check(chip); - return IRQ_NONE; - } + if (status == 0) + return snd_nm256_intr_check(chip); chip->badintrcount = 0; @@ -1036,10 +1082,8 @@ snd_nm256_interrupt_zx(int irq, void *dev_id, struct pt_regs *dummy) status = snd_nm256_readl(chip, NM_INT_REG); /* Not ours. */ - if (status == 0) { - snd_nm256_intr_check(chip); - return IRQ_NONE; - } + if (status == 0) + return snd_nm256_intr_check(chip); chip->badintrcount = 0; @@ -1192,7 +1236,7 @@ snd_nm256_mixer(nm256_t *chip) AC97_PC_BEEP, AC97_PHONE, AC97_MIC, AC97_LINE, AC97_CD, AC97_VIDEO, AC97_AUX, AC97_PCM, AC97_REC_SEL, AC97_REC_GAIN, AC97_GENERAL_PURPOSE, AC97_3D_CONTROL, - AC97_EXTENDED_ID, + /*AC97_EXTENDED_ID,*/ AC97_VENDOR_ID1, AC97_VENDOR_ID2, -1 }; @@ -1206,6 +1250,7 @@ snd_nm256_mixer(nm256_t *chip) for (i = 0; mixer_regs[i] >= 0; i++) set_bit(mixer_regs[i], ac97.reg_accessed); ac97.private_data = chip; + pbus->no_vra = 1; err = snd_ac97_mixer(pbus, &ac97, &chip->ac97); if (err < 0) return err; @@ -1281,6 +1326,7 @@ static int nm256_suspend(snd_card_t *card, pm_message_t state) static int nm256_resume(snd_card_t *card) { nm256_t *chip = card->pm_private_data; + int i; /* Perform a full reset on the hardware */ pci_enable_device(chip->pci); @@ -1289,6 +1335,15 @@ static int nm256_resume(snd_card_t *card) /* restore ac97 */ snd_ac97_resume(chip->ac97); + for (i = 0; i < 2; i++) { + nm256_stream_t *s = &chip->streams[i]; + if (s->substream && s->suspended) { + spin_lock_irq(&chip->reg_lock); + snd_nm256_set_format(chip, s, s->substream); + spin_unlock_irq(&chip->reg_lock); + } + } + return 0; } #endif /* CONFIG_PM */ @@ -1360,6 +1415,7 @@ snd_nm256_create(snd_card_t *card, struct pci_dev *pci, chip->use_cache = usecache; spin_lock_init(&chip->reg_lock); chip->irq = -1; + init_MUTEX(&chip->irq_mutex); chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize = play_bufsize; chip->streams[SNDRV_PCM_STREAM_CAPTURE].bufsize = capt_bufsize; @@ -1470,15 +1526,6 @@ snd_nm256_create(snd_card_t *card, struct pci_dev *pci, chip->coeff_buf[SNDRV_PCM_STREAM_CAPTURE] = addr; } - /* acquire interrupt */ - if (request_irq(pci->irq, chip->interrupt, SA_INTERRUPT|SA_SHIRQ, - card->driver, (void*)chip)) { - err = -EBUSY; - snd_printk("unable to grab IRQ %d\n", pci->irq); - goto __error; - } - chip->irq = pci->irq; - /* Fixed setting. */ chip->mixer_base = NM_MIXER_OFFSET; -- cgit From e8da2fbc2dd1fb4d603442f7220d23a2192955fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Aug 2005 16:55:13 +0200 Subject: [ALSA] hdspm - Fix module parameter description RME9652 driver Fix the module parameter description after proofreading. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d4a0c2c56cdb..5d786d113b25 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -65,7 +65,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable/disable specific HDSPM soundcards."); module_param_array(precise_ptr, bool, NULL, 0444); -MODULE_PARM_DESC(precise_ptr, "Enable precise pointer, or disable."); +MODULE_PARM_DESC(precise_ptr, "Enable or disable precise pointer."); module_param_array(line_outs_monitor, bool, NULL, 0444); MODULE_PARM_DESC(line_outs_monitor, -- cgit From 61be3ce0f2d9d80bc271e58c42cb9b021b3d48d8 Mon Sep 17 00:00:00 2001 From: Bjorge Dijkstra Date: Tue, 16 Aug 2005 17:05:05 +0200 Subject: [ALSA] ac97 - make ac97 codec device name unique AC97 Codec The patch fixes the bus_id conflict error when registering two codecs of the same type (ALSA bug#1334). Signed-off-by: Bjorge Dijkstra Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 600e053dfd35..5501f4440c92 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1829,7 +1829,7 @@ static int snd_ac97_dev_register(snd_device_t *device) ac97->dev.parent = ac97->bus->card->dev; ac97->dev.platform_data = ac97; ac97->dev.release = ac97_device_release; - strncpy(ac97->dev.bus_id, snd_ac97_get_short_name(ac97), BUS_ID_SIZE); + snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "card%d-%d", ac97->bus->card->number, ac97->num); if ((err = device_register(&ac97->dev)) < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); ac97->dev.bus = NULL; -- cgit From 41e4845c426024ec41724326397a36b3aa2cb6a3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 18 Aug 2005 13:43:12 +0200 Subject: [ALSA] PCM resume cleanups Digigram VX core,CS4231 driver,ATIIXP driver,VIA82xx driver VIA82xx-modem driver,au88x0 driver,CS46xx driver,Trident driver This patch disables SNDRV_PCM_INFO_RESUME flag for drivers which does not support the full resume. Signed-off-by: Jaroslav Kysela --- sound/pci/atiixp.c | 10 +++++++++- sound/pci/au88x0/au88x0_pcm.c | 6 +++--- sound/pci/cs46xx/cs46xx_lib.c | 8 ++++---- sound/pci/trident/trident_main.c | 10 +++++----- sound/pci/via82xx.c | 4 +++- sound/pci/via82xx_modem.c | 3 ++- 6 files changed, 26 insertions(+), 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index cafab4af5c57..904d17394e1c 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -248,6 +248,7 @@ struct snd_atiixp_dma { unsigned int period_bytes, periods; int opened; int running; + int suspended; int pcm_open_flag; int ac97_pcm_type; /* index # of ac97_pcm to access, -1 = not used */ unsigned int saved_curptr; @@ -699,12 +700,18 @@ static int snd_atiixp_pcm_trigger(snd_pcm_substream_t *substream, int cmd) spin_lock(&chip->reg_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: dma->ops->enable_transfer(chip, 1); dma->running = 1; + dma->suspended = 0; break; case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: dma->ops->enable_transfer(chip, 0); dma->running = 0; + dma->suspended = cmd == SNDRV_PCM_TRIGGER_SUSPEND; break; default: err = -EINVAL; @@ -975,6 +982,7 @@ static snd_pcm_hardware_t snd_atiixp_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, @@ -1443,7 +1451,7 @@ static int snd_atiixp_resume(snd_card_t *card) for (i = 0; i < NUM_ATI_PCMDEVS; i++) if (chip->pcmdevs[i]) { atiixp_dma_t *dma = &chip->dmas[i]; - if (dma->substream && dma->running) { + if (dma->substream && dma->suspended) { dma->ops->enable_dma(chip, 1); writel((u32)dma->desc_buf.addr | ATI_REG_LINKPTR_EN, chip->remap_addr + dma->ops->llp_offset); diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 04dcefd8b8ff..de95bd6d1ee1 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -33,7 +33,7 @@ /* hardware definition */ static snd_pcm_hardware_t snd_vortex_playback_hw_adb = { .info = - (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_RESUME | + (SNDRV_PCM_INFO_MMAP | /* SNDRV_PCM_INFO_RESUME | */ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID), .formats = @@ -58,7 +58,7 @@ static snd_pcm_hardware_t snd_vortex_playback_hw_adb = { #ifndef CHIP_AU8820 static snd_pcm_hardware_t snd_vortex_playback_hw_a3d = { .info = - (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_RESUME | + (SNDRV_PCM_INFO_MMAP | /* SNDRV_PCM_INFO_RESUME | */ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID), .formats = @@ -78,7 +78,7 @@ static snd_pcm_hardware_t snd_vortex_playback_hw_a3d = { #endif static snd_pcm_hardware_t snd_vortex_playback_hw_spdif = { .info = - (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_RESUME | + (SNDRV_PCM_INFO_MMAP | /* SNDRV_PCM_INFO_RESUME | */ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID), .formats = diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 5eb9e0bb9100..4b052158ee33 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1243,8 +1243,8 @@ static snd_pcm_hardware_t snd_cs46xx_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/ + /*SNDRV_PCM_INFO_RESUME*/), .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE), @@ -1265,8 +1265,8 @@ static snd_pcm_hardware_t snd_cs46xx_capture = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/ + /*SNDRV_PCM_INFO_RESUME*/), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, .rate_min = 5500, diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 29d89bfba0a4..f30d9d947862 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -1689,7 +1689,7 @@ static snd_pcm_hardware_t snd_trident_playback = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */), .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, @@ -1714,7 +1714,7 @@ static snd_pcm_hardware_t snd_trident_capture = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */), .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, @@ -1739,7 +1739,7 @@ static snd_pcm_hardware_t snd_trident_foldback = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -1763,7 +1763,7 @@ static snd_pcm_hardware_t snd_trident_spdif = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), @@ -1784,7 +1784,7 @@ static snd_pcm_hardware_t snd_trident_spdif_7018 = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 819f27b2762b..52ec785c550c 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -663,10 +663,12 @@ static int snd_via82xx_pcm_trigger(snd_pcm_substream_t * substream, int cmd) val = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: val |= VIA_REG_CTRL_START; viadev->running = 1; break; case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: val = VIA_REG_CTRL_TERMINATE; viadev->running = 0; break; @@ -1035,7 +1037,7 @@ static snd_pcm_hardware_t snd_via82xx_hw = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | + /* SNDRV_PCM_INFO_RESUME | */ SNDRV_PCM_INFO_PAUSE), .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 4a9779cc9733..5872d438a04a 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -521,6 +521,7 @@ static int snd_via82xx_pcm_trigger(snd_pcm_substream_t * substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_SUSPEND: val |= VIA_REG_CTRL_START; viadev->running = 1; break; @@ -697,7 +698,7 @@ static snd_pcm_hardware_t snd_via82xx_hw = .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | + /* SNDRV_PCM_INFO_RESUME | */ SNDRV_PCM_INFO_PAUSE), .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_KNOT, -- cgit From 352dbfd0e6eb7733bcca79c14f8711bb86b55e4e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Aug 2005 17:49:10 +0200 Subject: [ALSA] via82xx - Add DXS entry for MSI 7142 VIA82xx driver Added the DXS entry for MSI 7142 (K8MM-V). Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 52ec785c550c..56c6e52d7264 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2171,6 +2171,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1297, .subdevice = 0xc160, .action = VIA_DXS_ENABLE }, /* Shuttle SK41G */ { .subvendor = 0x1458, .subdevice = 0xa002, .action = VIA_DXS_ENABLE }, /* Gigabyte GA-7VAXP */ { .subvendor = 0x1462, .subdevice = 0x0080, .action = VIA_DXS_SRC }, /* MSI K8T Neo-FIS2R */ + { .subvendor = 0x1462, .subdevice = 0x0430, .action = VIA_DXS_SRC }, /* MSI 7142 (K8MM-V) */ { .subvendor = 0x1462, .subdevice = 0x3800, .action = VIA_DXS_ENABLE }, /* MSI KT266 */ { .subvendor = 0x1462, .subdevice = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */ { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */ -- cgit From 34ad73b9c5b5b25b795136c23f65b9a3ac21130b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 22 Aug 2005 11:30:37 +0200 Subject: [ALSA] ac97_bus - replace with for consistency AC97 Codec Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_bus.c b/sound/pci/ac97/ac97_bus.c index 227f8b9f67ce..013a919c2544 100644 --- a/sound/pci/ac97/ac97_bus.c +++ b/sound/pci/ac97/ac97_bus.c @@ -11,7 +11,7 @@ * (at your option) any later version. */ -#include +#include #include #include #include -- cgit From 5049c35b1389097a1c0c7686519928eaaf31e40b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 22 Aug 2005 12:19:14 +0200 Subject: [ALSA] ac97_bus - revert last change and do it only in the ALSA tree AC97 Codec Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_bus.c b/sound/pci/ac97/ac97_bus.c index 013a919c2544..227f8b9f67ce 100644 --- a/sound/pci/ac97/ac97_bus.c +++ b/sound/pci/ac97/ac97_bus.c @@ -11,7 +11,7 @@ * (at your option) any later version. */ -#include +#include #include #include #include -- cgit From 66d10647feb32178e1204d5b5c7126b1731efe9c Mon Sep 17 00:00:00 2001 From: Sergey Vlasov Date: Mon, 22 Aug 2005 13:43:39 +0200 Subject: [ALSA] HP nx6110 quirks Intel8x0 driver,AC97 Codec The HP nx6110 laptop needs to have Headphone Jack Sense enabled so that the internal speakers will be turned off when headphones are plugged in. Also ac97_quirk=hp_only is needed to make a single Master volume in the mixer instead of separate volumes for internal speakers and headphones, which just confuses the user. Signed-off-by: Sergey Vlasov Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + sound/pci/intel8x0.c | 6 ++++++ 2 files changed, 7 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 616504ef3702..b584172c1104 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1629,6 +1629,7 @@ static void check_ad1981_hp_jack_sense(ac97_t *ac97) u32 subid = ((u32)ac97->subsystem_vendor << 16) | ac97->subsystem_device; switch (subid) { case 0x103c0890: /* HP nc6000 */ + case 0x103c099c: /* HP nx6110 */ case 0x103c006d: /* HP nx9105 */ case 0x17340088: /* FSC Scenic-W */ /* enable headphone jack sense */ diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 390b6c8f9a10..e38984dc4d95 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1824,6 +1824,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP nc6000", .type = AC97_TUNE_MUTE_LED }, + { + .subvendor = 0x103c, + .subdevice = 0x099c, + .name = "HP nx6110", /* AD1981B */ + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x103c, .subdevice = 0x129d, -- cgit From 9d8f53f2bba3c2c06e1e78126222aecf91f8ecdd Mon Sep 17 00:00:00 2001 From: Nicolas Graziano Date: Mon, 22 Aug 2005 13:47:16 +0200 Subject: [ALSA] hda-intel - correct a bug in detection of rate supported HDA Codec driver The insertion of the rate 9600 make a shift in detection of supported rate, put this rate at the end of the list. Signed-off-by: Nicolas Graziano Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6bfb081d12dd..e067a14a2d9e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1165,8 +1165,9 @@ int snd_hda_build_controls(struct hda_bus *bus) */ static unsigned int rate_bits[][3] = { /* rate in Hz, ALSA rate bitmask, HDA format value */ + + /* autodetected value used in snd_hda_query_supported_pcm */ { 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */ - { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */ { 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */ { 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */ { 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */ @@ -1177,6 +1178,9 @@ static unsigned int rate_bits[][3] = { { 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */ { 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */ { 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */ + + /* not autodetected value */ + { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */ { 0 } }; -- cgit From befdf316eaba02ed52284fb78a8027ff35c6a736 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Aug 2005 13:57:55 +0200 Subject: [ALSA] hda-codec - Code clean up HDA Codec driver Use struct instead of array to improve the readability of hda_codec.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 29 ++++++++++++++++++----------- 1 file changed, 18 insertions(+), 11 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e067a14a2d9e..20f7762f7144 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1163,7 +1163,13 @@ int snd_hda_build_controls(struct hda_bus *bus) /* * stream formats */ -static unsigned int rate_bits[][3] = { +struct hda_rate_tbl { + unsigned int hz; + unsigned int alsa_bits; + unsigned int hda_fmt; +}; + +static struct hda_rate_tbl rate_bits[] = { /* rate in Hz, ALSA rate bitmask, HDA format value */ /* autodetected value used in snd_hda_query_supported_pcm */ @@ -1181,7 +1187,8 @@ static unsigned int rate_bits[][3] = { /* not autodetected value */ { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */ - { 0 } + + { 0 } /* terminator */ }; /** @@ -1203,12 +1210,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, int i; unsigned int val = 0; - for (i = 0; rate_bits[i][0]; i++) - if (rate_bits[i][0] == rate) { - val = rate_bits[i][2]; + for (i = 0; rate_bits[i].hz; i++) + if (rate_bits[i].hz == rate) { + val = rate_bits[i].hda_fmt; break; } - if (! rate_bits[i][0]) { + if (! rate_bits[i].hz) { snd_printdd("invalid rate %d\n", rate); return 0; } @@ -1271,9 +1278,9 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (ratesp) { u32 rates = 0; - for (i = 0; rate_bits[i][0]; i++) { + for (i = 0; rate_bits[i].hz; i++) { if (val & (1 << i)) - rates |= rate_bits[i][1]; + rates |= rate_bits[i].alsa_bits; } *ratesp = rates; } @@ -1365,13 +1372,13 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, } rate = format & 0xff00; - for (i = 0; rate_bits[i][0]; i++) - if (rate_bits[i][2] == rate) { + for (i = 0; rate_bits[i].hz; i++) + if (rate_bits[i].hda_fmt == rate) { if (val & (1 << i)) break; return 0; } - if (! rate_bits[i][0]) + if (! rate_bits[i].hz) return 0; stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); -- cgit From 3c10a9d9f8c76fd87c92a14c201ae211d0b81288 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2005 20:02:27 +0200 Subject: [ALSA] hda-codec - Add beep support for Uniwill HDA Codec driver Added the missing beep support for Uniwill laptop (ALSA bug#1358). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d19da2bae663..eeb900ab79af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -687,6 +687,12 @@ static snd_kcontrol_new_t alc880_asus_w1v_mixer[] = { { } /* end */ }; +/* additional mixers to alc880_asus_mixer */ +static snd_kcontrol_new_t alc880_pcbeep_mixer[] = { + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + { } /* end */ +}; /* * build control elements @@ -1735,7 +1741,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { - .mixers = { alc880_asus_mixer }, + .mixers = { alc880_asus_mixer, alc880_pcbeep_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), .dac_nids = alc880_asus_dac_nids, -- cgit From 07e4ca50a5f82aa6eab52e348059579b250c63db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Aug 2005 14:14:57 +0200 Subject: [ALSA] hda-intel - Fix ULI M5461 support HDA Intel driver Fix and clean up for the support of ULI M5461 - set CORB/RIRB sizes explicitly - add workarounds for ULI on ia32 - max number of streams depends on the chip type now - increase the size of BDL Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 150 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 117 insertions(+), 33 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 58e15b8896ac..15107df1f490 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -72,7 +72,8 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{ATI, SB450}," "{VIA, VT8251}," "{VIA, VT8237A}," - "{SiS, SIS966}}"); + "{SiS, SIS966}," + "{ULI, M5461}}"); MODULE_DESCRIPTION("Intel HDA driver"); #define SFX "hda-intel: " @@ -142,9 +143,24 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; */ /* max number of SDs */ -#define MAX_ICH6_DEV 8 +/* ICH, ATI and VIA have 4 playback and 4 capture */ +#define ICH6_CAPTURE_INDEX 0 +#define ICH6_NUM_CAPTURE 4 +#define ICH6_PLAYBACK_INDEX 4 +#define ICH6_NUM_PLAYBACK 4 + +/* ULI has 6 playback and 5 capture */ +#define ULI_CAPTURE_INDEX 0 +#define ULI_NUM_CAPTURE 5 +#define ULI_PLAYBACK_INDEX 5 +#define ULI_NUM_PLAYBACK 6 + +/* this number is statically defined for simplicity */ +#define MAX_AZX_DEV 16 + /* max number of fragments - we may use more if allocating more pages for BDL */ -#define AZX_MAX_FRAG (PAGE_SIZE / (MAX_ICH6_DEV * 16)) +#define BDL_SIZE PAGE_ALIGN(8192) +#define AZX_MAX_FRAG (BDL_SIZE / (MAX_AZX_DEV * 16)) /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) /* max number of PCM devics per card */ @@ -201,7 +217,6 @@ enum { }; /* Defines for ATI HD Audio support in SB450 south bridge */ -#define ATI_SB450_HDAUDIO_PCI_DEVICE_ID 0x437b #define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42 #define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02 @@ -259,6 +274,14 @@ struct snd_azx { snd_card_t *card; struct pci_dev *pci; + /* chip type specific */ + int driver_type; + int playback_streams; + int playback_index_offset; + int capture_streams; + int capture_index_offset; + int num_streams; + /* pci resources */ unsigned long addr; void __iomem *remap_addr; @@ -268,8 +291,8 @@ struct snd_azx { spinlock_t reg_lock; struct semaphore open_mutex; - /* streams */ - azx_dev_t azx_dev[MAX_ICH6_DEV]; + /* streams (x num_streams) */ + azx_dev_t *azx_dev; /* PCM */ unsigned int pcm_devs; @@ -293,6 +316,23 @@ struct snd_azx { unsigned int initialized: 1; }; +/* driver types */ +enum { + AZX_DRIVER_ICH, + AZX_DRIVER_ATI, + AZX_DRIVER_VIA, + AZX_DRIVER_SIS, + AZX_DRIVER_ULI, +}; + +static char *driver_short_names[] __devinitdata = { + [AZX_DRIVER_ICH] = "HDA Intel", + [AZX_DRIVER_ATI] = "HDA ATI SB", + [AZX_DRIVER_VIA] = "HDA VIA VT82xx", + [AZX_DRIVER_SIS] = "HDA SIS966", + [AZX_DRIVER_ULI] = "HDA ULI M5461" +}; + /* * macros for easy use */ @@ -361,6 +401,8 @@ static void azx_init_cmd_io(azx_t *chip) azx_writel(chip, CORBLBASE, (u32)chip->corb.addr); azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr)); + /* set the corb size to 256 entries (ULI requires explicitly) */ + azx_writeb(chip, CORBSIZE, 0x02); /* set the corb write pointer to 0 */ azx_writew(chip, CORBWP, 0); /* reset the corb hw read pointer */ @@ -374,6 +416,8 @@ static void azx_init_cmd_io(azx_t *chip) azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr)); + /* set the rirb size to 256 entries (ULI requires explicitly) */ + azx_writeb(chip, RIRBSIZE, 0x02); /* reset the rirb hw write pointer */ azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR); /* set N=1, get RIRB response interrupt for new entry */ @@ -597,7 +641,7 @@ static void azx_int_disable(azx_t *chip) int i; /* disable interrupts in stream descriptor */ - for (i = 0; i < MAX_ICH6_DEV; i++) { + for (i = 0; i < chip->num_streams; i++) { azx_dev_t *azx_dev = &chip->azx_dev[i]; azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~SD_INT_MASK); @@ -617,7 +661,7 @@ static void azx_int_clear(azx_t *chip) int i; /* clear stream status */ - for (i = 0; i < MAX_ICH6_DEV; i++) { + for (i = 0; i < chip->num_streams; i++) { azx_dev_t *azx_dev = &chip->azx_dev[i]; azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); } @@ -687,8 +731,7 @@ static void azx_init_chip(azx_t *chip) } /* For ATI SB450 azalia HD audio, we need to enable snoop */ - if (chip->pci->vendor == PCI_VENDOR_ID_ATI && - chip->pci->device == ATI_SB450_HDAUDIO_PCI_DEVICE_ID) { + if (chip->driver_type == AZX_DRIVER_ATI) { pci_read_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, &ati_misc_cntl2); pci_write_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, @@ -715,7 +758,7 @@ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) return IRQ_NONE; } - for (i = 0; i < MAX_ICH6_DEV; i++) { + for (i = 0; i < chip->num_streams; i++) { azx_dev = &chip->azx_dev[i]; if (status & azx_dev->sd_int_sta_mask) { azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); @@ -880,9 +923,15 @@ static int __devinit azx_codec_create(azx_t *chip, const char *model) /* assign a stream for the PCM */ static inline azx_dev_t *azx_assign_device(azx_t *chip, int stream) { - int dev, i; - dev = stream == SNDRV_PCM_STREAM_PLAYBACK ? 4 : 0; - for (i = 0; i < 4; i++, dev++) + int dev, i, nums; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev = chip->playback_index_offset; + nums = chip->playback_streams; + } else { + dev = chip->capture_index_offset; + nums = chip->capture_streams; + } + for (i = 0; i < nums; i++, dev++) if (! chip->azx_dev[dev].opened) { chip->azx_dev[dev].opened = 1; return &chip->azx_dev[dev]; @@ -1190,7 +1239,7 @@ static int __devinit azx_init_stream(azx_t *chip) /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) and initialize */ - for (i = 0; i < MAX_ICH6_DEV; i++) { + for (i = 0; i < chip->num_streams; i++) { unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4); azx_dev_t *azx_dev = &chip->azx_dev[i]; azx_dev->bdl = (u32 *)(chip->bdl.area + off); @@ -1249,7 +1298,7 @@ static int azx_free(azx_t *chip) if (chip->initialized) { int i; - for (i = 0; i < MAX_ICH6_DEV; i++) + for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); /* disable interrupts */ @@ -1265,10 +1314,10 @@ static int azx_free(azx_t *chip) /* wait a little for interrupts to finish */ msleep(1); - - iounmap(chip->remap_addr); } + if (chip->remap_addr) + iounmap(chip->remap_addr); if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); @@ -1280,6 +1329,7 @@ static int azx_free(azx_t *chip) snd_dma_free_pages(&chip->posbuf); pci_release_regions(chip->pci); pci_disable_device(chip->pci); + kfree(chip->azx_dev); kfree(chip); return 0; @@ -1294,7 +1344,8 @@ static int azx_dev_free(snd_device_t *device) * constructor */ static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, - int posfix, azx_t **rchip) + int posfix, int driver_type, + azx_t **rchip) { azx_t *chip; int err = 0; @@ -1320,9 +1371,20 @@ static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, chip->card = card; chip->pci = pci; chip->irq = -1; + chip->driver_type = driver_type; chip->position_fix = posfix; +#if BITS_PER_LONG != 64 + /* Fix up base address on ULI M5461 */ + if (chip->driver_type == AZX_DRIVER_ULI) { + u16 tmp3; + pci_read_config_word(pci, 0x40, &tmp3); + pci_write_config_word(pci, 0x40, tmp3 | 0x10); + pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); + } +#endif + if ((err = pci_request_regions(pci, "ICH HD audio")) < 0) { kfree(chip); pci_disable_device(pci); @@ -1348,16 +1410,37 @@ static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, pci_set_master(pci); synchronize_irq(chip->irq); + switch (chip->driver_type) { + case AZX_DRIVER_ULI: + chip->playback_streams = ULI_NUM_PLAYBACK; + chip->capture_streams = ULI_NUM_CAPTURE; + chip->playback_index_offset = ULI_PLAYBACK_INDEX; + chip->capture_index_offset = ULI_CAPTURE_INDEX; + break; + default: + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + chip->playback_index_offset = ICH6_PLAYBACK_INDEX; + chip->capture_index_offset = ICH6_CAPTURE_INDEX; + break; + } + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); + if (! chip->azx_dev) { + snd_printk(KERN_ERR "cannot malloc azx_dev\n"); + goto errout; + } + /* allocate memory for the BDL for each stream */ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - PAGE_SIZE, &chip->bdl)) < 0) { + BDL_SIZE, &chip->bdl)) < 0) { snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); goto errout; } if (chip->position_fix == POS_FIX_POSBUF) { /* allocate memory for the position buffer */ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - MAX_ICH6_DEV * 8, &chip->posbuf)) < 0) { + chip->num_streams * 8, &chip->posbuf)) < 0) { snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); goto errout; } @@ -1386,6 +1469,10 @@ static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, goto errout; } + strcpy(card->driver, "HDA-Intel"); + strcpy(card->shortname, driver_short_names[chip->driver_type]); + sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->addr, chip->irq); + *rchip = chip; return 0; @@ -1414,15 +1501,12 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id * return -ENOMEM; } - if ((err = azx_create(card, pci, position_fix[dev], &chip)) < 0) { + if ((err = azx_create(card, pci, position_fix[dev], pci_id->driver_data, + &chip)) < 0) { snd_card_free(card); return err; } - strcpy(card->driver, "HDA-Intel"); - strcpy(card->shortname, "HDA Intel"); - sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->addr, chip->irq); - /* create codec instances */ if ((err = azx_codec_create(chip, model[dev])) < 0) { snd_card_free(card); @@ -1463,13 +1547,13 @@ static void __devexit azx_remove(struct pci_dev *pci) /* PCI IDs */ static struct pci_device_id azx_ids[] = { - { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH6 */ - { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH7 */ - { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ESB2 */ - { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ATI SB450 */ - { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* VIA VT8251/VT8237A */ - { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SIS966 */ - { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ALI 5461? */ + { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH6 */ + { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH7 */ + { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ESB2 */ + { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ + { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ + { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ + { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- cgit From a278655ff5d0c9d5eb34cf99f3a4c20da09eb09e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Aug 2005 17:57:25 +0200 Subject: [ALSA] Fix missing spin_unlock au88x0 driver,Common EMU synth Fixed missing spin_unlock. Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index de95bd6d1ee1..38bd2b5dd434 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -220,8 +220,10 @@ snd_vortex_pcm_hw_params(snd_pcm_substream_t * substream, vortex_adb_allocroute(chip, -1, params_channels(hw_params), substream->stream, type); - if (dma < 0) + if (dma < 0) { + spin_unlock_irq(&chip->lock); return dma; + } stream = substream->runtime->private_data = &chip->dma_adb[dma]; stream->substream = substream; /* Setup Buffers. */ -- cgit From e0c93cf311b3eaa04bfb5769cff38861d4e748c8 Mon Sep 17 00:00:00 2001 From: Daniel Horchner Date: Thu, 25 Aug 2005 13:02:24 +0200 Subject: [ALSA] intel8x0 - add AC97_TUNE_MUTE_LED quirk for HP nx8220 laptop 103c:0934 Intel8x0 driver Signed-off-by: Jaroslav Kysela --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index e38984dc4d95..05c7994a8d5e 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1824,6 +1824,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP nc6000", .type = AC97_TUNE_MUTE_LED }, + { + .subvendor = 0x103c, + .subdevice = 0x0934, + .name = "HP nx8220", + .type = AC97_TUNE_MUTE_LED + }, { .subvendor = 0x103c, .subdevice = 0x099c, -- cgit From 9970dce56686d7b71310388025d8925d3d29e6ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Aug 2005 20:30:27 +0200 Subject: [ALSA] intel8x0 - Add quirk for FSC 4010 Intel8x0 driver Added ac97_quirk for FSC 4010. Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 05c7994a8d5e..7b548416dcef 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1890,6 +1890,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "Fujitsu S6210", /* STAC9750/51 */ .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x10cf, + .subdevice = 0x12ec, + .name = "Fujitsu-Siemens 4010", + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x10f1, .subdevice = 0x2665, -- cgit From 6a293552ba0d104ef9d35b48972a01540fbd181b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 26 Aug 2005 17:26:40 +0200 Subject: [ALSA] emu10k1 - Add missing ac97 support on SBLive! Player 5.1 EMU10K1/EMU10K2 driver Added the missing ac97 support on SBLive! Player 5.1. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index c0b67b70e345..83597c694596 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -752,6 +752,7 @@ static emu_chip_details_t emu_chip_details[] = { .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]", .id = "Live", .emu10k1_chip = 1, + .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", -- cgit From c6c0b841dd85c878fe9794862b10ce8eb4e90761 Mon Sep 17 00:00:00 2001 From: Lee Revell Date: Mon, 29 Aug 2005 17:42:00 +0200 Subject: [ALSA] Add identification for Live 5.1 [SB0220] EMU10K1/EMU10K2 driver Add identification for Live 5.1 [SB0220] Signed-off-by: Lee Revell Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 83597c694596..e69d5b739e80 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -741,6 +741,13 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , + /* Tested by Thomas Zehetbauer 27th Aug 2005 */ + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102, + .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, .driver = "EMU10K1", .name = "SB Live 5.1", .id = "Live", -- cgit From 829ca9a30a2ddb727981d80fabdbff2ea86bc9ea Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Sat, 3 Sep 2005 15:56:56 -0700 Subject: [PATCH] swsusp: fix remaining u32 vs. pm_message_t confusion Fix remaining bits of u32 vs. pm_message confusion. Should not break anything. Signed-off-by: Pavel Machek Signed-off-by: Rafael J. Wysocki Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/pci/atiixp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 904d17394e1c..188df085b7ee 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1427,7 +1427,7 @@ static int snd_atiixp_suspend(snd_card_t *card, pm_message_t state) snd_atiixp_aclink_down(chip); snd_atiixp_chip_stop(chip); - pci_set_power_state(chip->pci, 3); + pci_set_power_state(chip->pci, PCI_D3hot); pci_disable_device(chip->pci); return 0; } @@ -1438,7 +1438,7 @@ static int snd_atiixp_resume(snd_card_t *card) int i; pci_enable_device(chip->pci); - pci_set_power_state(chip->pci, 0); + pci_set_power_state(chip->pci, PCI_D0); pci_set_master(chip->pci); snd_atiixp_aclink_reset(chip); -- cgit From 8db08ea7e6527eff82d8e45507468003e3cefba3 Mon Sep 17 00:00:00 2001 From: Pekka Enberg Date: Tue, 6 Sep 2005 15:18:36 -0700 Subject: [PATCH] ALSA: convert kcalloc to kzalloc This patch introduces a memory-leak tracking version of kzalloc for ALSA. Signed-off-by: Pekka Enberg Cc: Jaroslav Kysela Signed-off-by: Jiri Slaby Signed-off-by: Andrew Morton Signed-off-by: Linus Torvalds --- sound/pci/ali5451/ali5451.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index ce6c9fadb594..4943299cf137 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2249,7 +2249,7 @@ static int __devinit snd_ali_create(snd_card_t * card, return -ENXIO; } - if ((codec = kcalloc(1, sizeof(*codec), GFP_KERNEL)) == NULL) { + if ((codec = kzalloc(sizeof(*codec), GFP_KERNEL)) == NULL) { pci_disable_device(pci); return -ENOMEM; } -- cgit