From b84f08d49188a18d965fab8463c9cb679785eb39 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Feb 2008 12:36:11 +0100 Subject: [ALSA] hda-codec - Fix Master volume on HP dv8000 HP dv8000 laptop has a problem with Master volume. It's due to the connection of the widget 0x13. When it's connected from the analog amp mixer (0x19), it works as expected mysteriously (ALSA bug#3775): https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3775 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f7cd3a804b11..7206b30cbf94 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1230,6 +1230,11 @@ static struct hda_verb cxt5047_toshiba_init_verbs[] = { static struct hda_verb cxt5047_hp_init_verbs[] = { /* pin sensing on HP jack */ {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + /* 0x13 is actually shared by both HP and speaker; + * setting the connection to 0 (=0x19) makes the master volume control + * working mysteriouslly... + */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Record selector: Ext Mic */ {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, -- cgit v1.2.3 From ee47fd12d73706edb2a10efd05d5eed15b4d1e08 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 20 Feb 2008 17:13:15 +0100 Subject: [ALSA] ASoC: Fix TLV320AIC3X PLL divider table for 64 kHz rate Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 710e0287ef8c..569ecaca0e8b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -681,8 +681,8 @@ static const struct aic3x_rate_divs aic3x_divs[] = { {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ - {22579200, 96000, 96000, 0x1, 8, 7075}, - {33868800, 96000, 96000, 0x1, 5, 8049}, + {22579200, 64000, 96000, 0x1, 8, 7075}, + {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, -- cgit v1.2.3 From d513202efd5bb9974545ef1c7f951467b21eb3a5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 Feb 2008 11:01:00 +0100 Subject: [ALSA] usb-audio: add workaround for broken E-Mu frequency feedback Add a workaround for the feedback pipe of E-Mu 0202/0404 USB devices that reports the number of samples per packet instead of the number of samples per microframe. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 38 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 36 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8fa935665702..675672f313be 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -479,6 +479,33 @@ static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, return 0; } +/* + * process after E-Mu 0202/0404 high speed playback sync complete + * + * These devices return the number of samples per packet instead of the number + * of samples per microframe. + */ +static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int f; + unsigned long flags; + + if (urb->iso_frame_desc[0].status == 0 && + urb->iso_frame_desc[0].actual_length == 4) { + f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; + f >>= subs->datainterval; + if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } + } + + return 0; +} + /* determine the number of frames in the next packet */ static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) { @@ -2219,10 +2246,17 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo subs->stream = as; subs->direction = stream; subs->dev = as->chip->dev; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { subs->ops = audio_urb_ops[stream]; - else + } else { subs->ops = audio_urb_ops_high_speed[stream]; + switch (as->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; + break; + } + } snd_pcm_set_ops(as->pcm, stream, stream == SNDRV_PCM_STREAM_PLAYBACK ? &snd_usb_playback_ops : &snd_usb_capture_ops); -- cgit v1.2.3 From 20cde9e8f83711dca532c49605914d50292d9ce5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 Feb 2008 11:04:41 +0100 Subject: [ALSA] sb8: fix SB 1.0 capture DMA programming Fix a wrong version check that would cause an invalid command to be sent to SB 1.0 chips. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/isa/sb/sb8_main.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 6304c3a89ba0..fe03bb820532 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -277,7 +277,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) } else { snd_sbdsp_command(chip, 256 - runtime->rate_den); } - if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->capture_format != SB_DSP_INPUT) { count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); -- cgit v1.2.3 From fb304ce53afbb653bfa67cc81ee9cf06edcbf68e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Feb 2008 15:32:01 +0100 Subject: [ALSA] hda-codec - Fix AD1988 capture elements The some indices of capture elements of AD1988 are wrongly assigned. This patch fixes it. See ALSA bug#3795 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3795 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 19f08846d6fc..c8649282c2cf 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1778,9 +1778,9 @@ static hda_nid_t ad1988_capsrc_nids[3] = { static struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { - { "Front Mic", 0x0 }, - { "Line", 0x1 }, - { "Mic", 0x4 }, + { "Front Mic", 0x1 }, /* port-B */ + { "Line", 0x2 }, /* port-C */ + { "Mic", 0x4 }, /* port-E */ { "CD", 0x5 }, { "Mix", 0x9 }, }, @@ -1789,7 +1789,7 @@ static struct hda_input_mux ad1988_6stack_capture_source = { static struct hda_input_mux ad1988_laptop_capture_source = { .num_items = 3, .items = { - { "Mic/Line", 0x0 }, + { "Mic/Line", 0x1 }, /* port-B */ { "CD", 0x5 }, { "Mix", 0x9 }, }, -- cgit v1.2.3 From 3f1eeaed2c0dc6c787a47ae7a6c774589a04a3a2 Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 25 Feb 2008 16:44:13 +0100 Subject: [ALSA] hda-codec - Add Fujitsu Lifebook E8410 to quirk table Add the proper model entry for Fujitsu Lifebook E8410 with ALC262 codec. From: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 777f8c01ca7a..1534f0866f76 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9238,6 +9238,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), + SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), -- cgit v1.2.3 From b930b9f41d5e9eadd9041f273c4d6d18e7061d05 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 26 Feb 2008 08:40:57 +0100 Subject: [ALSA] oxygen: add owner field I forgot to set the module owner for the HiFier/Xonar models. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 1 + sound/pci/oxygen/virtuoso.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 3ea1f05228a1..666f69a3312e 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -150,6 +150,7 @@ static const struct oxygen_model model_hifier = { .shortname = "C-Media CMI8787", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", + .owner = THIS_MODULE, .init = hifier_init, .control_filter = hifier_control_filter, .mixer_init = hifier_mixer_init, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 40e92f5cd69c..d163397b85cc 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -389,6 +389,7 @@ static const struct oxygen_model model_xonar = { .shortname = "Asus AV200", .longname = "Asus Virtuoso 200", .chip = "AV200", + .owner = THIS_MODULE, .init = xonar_init, .control_filter = xonar_control_filter, .mixer_init = xonar_mixer_init, -- cgit v1.2.3 From 0b167bf456d4af58103e2072bc4bd5733e7e7579 Mon Sep 17 00:00:00 2001 From: Andrew Paprocki Date: Sun, 3 Feb 2008 10:15:44 +0100 Subject: [ALSA] hda_intel - Add model quirk for Albatron KI690-AM2 motherboard This adds a quirk to the Realtek ALC883 table for the Albatron KI690-AM2 motherboard to use the 6stack-dig model. Signed-off-by: Andrew Paprocki Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1534f0866f76..b092bd47e56e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7639,6 +7639,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} -- cgit v1.2.3 From b6a370b6fb3114f9f7fc8a393c3ffc2236d7cbf1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Feb 2008 14:00:53 +0100 Subject: [ALSA] intel8x0 - Add quirk for Acer Travelmate 2310 Added ac97_quirk=hp-only for Acer Travelmate 2310. ALSA bug#3656 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3656 Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 061072c7db03..c5ef12ae3c55 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1738,6 +1738,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "IBM NetVista A30p", /* AD1981B */ .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x1025, + .subdevice = 0x0082, + .name = "Acer Travelmate 2310", + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x1025, .subdevice = 0x0083, -- cgit v1.2.3 From 31bffaa9435f14b35a8e23ed2005925f65ec6d9b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Feb 2008 16:10:44 +0100 Subject: [ALSA] hda-codec - Fix mixer names of realtek codecs to adapt mater controls Some models like eeepc ep20 have invalid mixer names that aren't handled properly by virtual master controls. Rename them to the proper names. Also fixed some typos in the mixer names but they are not compiled in right now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b092bd47e56e..51871c684571 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3973,8 +3973,8 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), { } /* end */ }; @@ -4005,9 +4005,9 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME_MONO("Mono Speaker Playback Volume", 0x0a, 1, 0x0, + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono Speaker Playback Switch", 0x0a, 1, 2, + HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), @@ -8103,7 +8103,7 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */ + HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), @@ -8125,7 +8125,7 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */ + HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), { } /* end */ @@ -13009,8 +13009,8 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { }; static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { - HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), -- cgit v1.2.3 From 338c7ed070bb1e068c3ae8ef14dc577e75d8aecc Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Feb 2008 12:34:48 +0100 Subject: [ALSA] ASoC: Fix DAPM widget function types in pxa machine drivers Add kcontrol argument to functions since the API was changed by the commit 9af6d9562414568ecadf96aaef5b88e7e8b19821. Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai --- sound/soc/pxa/corgi.c | 6 ++++-- sound/soc/pxa/poodle.c | 3 ++- sound/soc/pxa/spitz.c | 3 ++- sound/soc/pxa/tosa.c | 3 ++- 4 files changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 3f34e531bebf..1a70a6ac98ce 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -215,7 +215,8 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol, return 1; } -static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) +static int corgi_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); @@ -225,7 +226,8 @@ static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) return 0; } -static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event) +static int corgi_mic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 5ae59bd309a3..4fbf8bba9627 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -196,7 +196,8 @@ static int poodle_set_spk(struct snd_kcontrol *kcontrol, return 1; } -static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event) +static int poodle_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) locomo_gpio_write(&poodle_locomo_device.dev, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d56709e15435..ecca39033fcc 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -215,7 +215,8 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, return 1; } -static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event) +static int spitz_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) { if (machine_is_borzoi() || machine_is_spitz()) { if (SND_SOC_DAPM_EVENT_ON(event)) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index e4d40b528ca4..7346d7e5d066 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -135,7 +135,8 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol, } /* tosa dapm event handlers */ -static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event) +static int tosa_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE); -- cgit v1.2.3 From 3fffe871b93f957bea443e85f6b221c50bbf9f97 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Feb 2008 12:35:25 +0100 Subject: [ALSA] ASoC: Fix WM9712 mixer_event DAPM widget function type Add kcontrol argument to function since the API was changed by the commit 9af6d9562414568ecadf96aaef5b88e7e8b19821. Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 590baea3c4c3..524f7450804f 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -176,7 +176,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec) * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path. */ -static int mixer_event (struct snd_soc_dapm_widget *w, int event) +static int mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) { u16 l, r, beep, line, phone, mic, pcm, aux; -- cgit v1.2.3 From 008f3599ef97438900d62fe05d75535d114780fc Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Fri, 29 Feb 2008 11:46:32 +0100 Subject: [ALSA] sound: ice1712: unused structs Don't need to declare a struct when defining a structure layout. Both of these structs are unused. sound/pci/ice1712/revo.c:39:3: warning: symbol 'revo51' was not declared. Should it be static? sound/pci/ice1712/phase.c:54:3: warning: symbol 'phase28' was not declared. Should it be static? Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/ice1712/phase.c | 2 +- sound/pci/ice1712/revo.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index 9ab4a9f383cb..5a158b73dcaa 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -51,7 +51,7 @@ struct phase28_spec { unsigned short master[2]; unsigned short vol[8]; -} phase28; +}; /* WM8770 registers */ #define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */ diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index ddd5fc8d4fe1..301bf929acd9 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -36,7 +36,7 @@ struct revo51_spec { struct snd_i2c_device *dev; struct snd_pt2258 *pt2258; -} revo51; +}; static void revo_i2s_mclk_changed(struct snd_ice1712 *ice) { -- cgit v1.2.3 From b4818494edddfe382de4f5d072cb527b60315a46 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sat, 23 Feb 2008 11:34:12 +0100 Subject: [ALSA] hda-codec - Adapt eeepc p701 mixer for virtual master control Fix the line-out volume control of eeepc p701 to be a proper slave of the virtual master control. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 51871c684571..33282f9c01c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12995,8 +12995,8 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), -- cgit v1.2.3 From 0d9ac27afa469dbb20940ad7f25502785af1cbe3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Feb 2008 16:40:18 +0100 Subject: [ALSA] intel8x0 - Add quirk for Compaq Deskpro EN Added the ac97_quirk hp_only for Compaq Deskpro EN. Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c5ef12ae3c55..c52abd0bf22e 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1708,6 +1708,12 @@ static struct ac97_pcm ac97_pcm_defs[] __devinitdata = { }; static struct ac97_quirk ac97_quirks[] __devinitdata = { + { + .subvendor = 0x0e11, + .subdevice = 0x000e, + .name = "Compaq Deskpro EN", /* AD1885 */ + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x0e11, .subdevice = 0x008a, -- cgit v1.2.3