From ed36081350d2ca4f692f04c6a2d24d1e3a339da1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:12:52 +0200 Subject: ALSA: hda - Add codec->pcm_format_first flag Introduced a new flag to set up the PCM stream format at first before the stream_id and channel tag. Some codecs (e.g. CA0132) seem preferring this over stream_id -> format order. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++++++++----------------- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 46 insertions(+), 24 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..598b9e2d85e6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); +/* update the stream-id if changed */ +static void update_pcm_stream_id(struct hda_codec *codec, + struct hda_cvt_setup *p, hda_nid_t nid, + u32 stream_tag, int channel_id) +{ + unsigned int oldval, newval; + + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } +} + +/* update the format-id if changed */ +static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p, + hda_nid_t nid, int format) +{ + unsigned int oldval; + + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } +} + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, { struct hda_codec *c; struct hda_cvt_setup *p; - unsigned int oldval, newval; int type; int i; @@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p = get_hda_cvt_setup(codec, nid); if (!p) return; - /* update the stream-id if changed */ - if (p->stream_tag != stream_tag || p->channel_id != channel_id) { - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - p->stream_tag = stream_tag; - p->channel_id = channel_id; - } - /* update the format-id if changed */ - if (p->format_id != format) { - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, 0); - if (oldval != format) { - msleep(1); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - p->format_id = format; - } + + if (codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); + if (!codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + p->active = 1; p->dirty = 0; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d330ca54..7fbc1bcaf1a9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,6 +861,7 @@ struct hda_codec { unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ + unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ -- cgit v1.2.3 From 55cf87fe0e9783e25f442be1e48b8319d86131ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:15:55 +0200 Subject: ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d0d3540e39e7..2685590925ff 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -988,12 +988,12 @@ static void ca0132_config(struct hda_codec *codec) /* Mic-in */ spec->input_pins[0] = 0x12; - spec->input_labels[0] = "Mic-In"; + spec->input_labels[0] = "Mic"; spec->adcs[0] = 0x07; /* Line-In */ spec->input_pins[1] = 0x11; - spec->input_labels[1] = "Line-In"; + spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; } -- cgit v1.2.3 From 27ebeb0b1b5bb26908e485a7e8bd2ec30366ffef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:20:18 +0200 Subject: ALSA: hda - Use the standard PCM ops for CA0132 Now with the workaround using codec->pcm_format_first flag, we can clean up the home-baked codes in patch_ca0132.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 142 +++++++++---------------------------------- 1 file changed, 29 insertions(+), 113 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2685590925ff..31512a0f1d07 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -464,50 +464,17 @@ exit: } /* - * PCM stuffs + * PCM callbacks */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) +static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd("ca0132_setup_stream: " - "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + struct ca0132_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } -/* - * PCM callbacks - */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -515,10 +482,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); } static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -526,92 +491,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dacs[0]); - - return 0; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } /* * Digital out */ -static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); - - return 0; -} - -static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_out); - - return 0; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -/* - * Analog capture - */ -static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); } -static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->adcs[substream->number]); - - return 0; -} - -/* - * Digital capture - */ -static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } -static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_in); - - return 0; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } /* @@ -621,6 +539,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_playback_pcm_open, .prepare = ca0132_playback_pcm_prepare, .cleanup = ca0132_playback_pcm_cleanup }, @@ -630,10 +549,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup - }, }; static struct hda_pcm_stream ca0132_pcm_digital_playback = { @@ -641,6 +556,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_dig_playback_pcm_open, + .close = ca0132_dig_playback_pcm_close, .prepare = ca0132_dig_playback_pcm_prepare, .cleanup = ca0132_dig_playback_pcm_cleanup }, @@ -650,10 +567,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_dig_capture_pcm_prepare, - .cleanup = ca0132_dig_capture_pcm_cleanup - }, }; static int ca0132_build_pcms(struct hda_codec *codec) @@ -961,6 +874,9 @@ static void ca0132_config(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* line-outs */ cfg->line_outs = 1; cfg->line_out_pins[0] = 0x0b; /* front */ -- cgit v1.2.3 From 8e13fc1c5f694a6ae4032c7f94103c137136733f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:26:54 +0200 Subject: ALSA: hda - Add missing SPDIF I/O setup for CA0132 CA0132 driver had some codes to handle the S/PDIF I/O, but the actual setups of pins and converters were missing. Now the pins are added. Also, fixed a few points triggering invalid codec verbs and mixer elements since the digital I/O audio widgets on CA0132 have no amp. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 31512a0f1d07..9c0ec0a55bef 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); } - if (dac) + if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); } @@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } - if (adc) + if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } @@ -841,18 +841,16 @@ static int ca0132_build_controls(struct hda_codec *codec) spec->dig_out); if (err < 0) return err; - err = add_out_volume(codec, spec->dig_out, "IEC958"); + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); if (err < 0) return err; + /* spec->multiout.share_spdif = 1; */ } if (spec->dig_in) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); if (err < 0) return err; - err = add_in_volume(codec, spec->dig_in, "IEC958"); - if (err < 0) - return err; } return 0; } @@ -912,6 +910,16 @@ static void ca0132_config(struct hda_codec *codec) spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static void ca0132_init_chip(struct hda_codec *codec) -- cgit v1.2.3 From 144dad99ef6ad10c8c8ebe787d08157c4a94201f Mon Sep 17 00:00:00 2001 From: James Ralston Date: Thu, 9 Aug 2012 09:38:59 -0700 Subject: ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH Signed-off-by: James Ralston Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced182fd1..60882c62f180 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, LPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c21), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From 14bc9c6dc694e2d7930802f7afd275de25ef8394 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 10 Aug 2012 13:29:32 +0200 Subject: ALSA: hda - Fix panned "Beep Playback Switch" When "Beep Playback Switch" had a different value on left and right channels (such as muting left but not right, or vice versa), this could result in the right channel being ignored. This patch enables beep to be sounding from right channel only, and also give correct result back to userspace (e g amixer). Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315b181d..d26ae65b43b7 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -237,10 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) { + if (beep && !beep->enabled) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = 0; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +251,18 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) - snd_hda_enable_beep_device(codec, - *ucontrol->value.integer.value); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v1.2.3 From e037cb4a54e26b5f55f856e0e7445cfcfb2f3d31 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 10 Aug 2012 14:11:58 +0200 Subject: ALSA : hda - bug fix on checking the supported power states of a codec The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..629131ad7b8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; -- cgit v1.2.3 From 088c820b732dbfd515fc66d459d5f5777f79b406 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 13 Aug 2012 14:11:10 +0800 Subject: ALSA: hda - fix Copyright debug message As spec said, 1 indicates no copyright is asserted. Signed-off-by: Wang Xingchao Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258fc700..6894ec66258c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_EMPHASIS) snd_iprintf(buffer, " Preemphasis"); if (digi1 & AC_DIG1_COPYRIGHT) - snd_iprintf(buffer, " Copyright"); + snd_iprintf(buffer, " Non-Copyright"); if (digi1 & AC_DIG1_NONAUDIO) snd_iprintf(buffer, " Non-Audio"); if (digi1 & AC_DIG1_PROFESSIONAL) -- cgit v1.2.3 From 265d931a9e9a7e290faa5e2145f4b2ebf38ea84c Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 13 Aug 2012 17:10:46 +0200 Subject: ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch Some Conexant devices (e g CX20590) have no mute capability on their Beep widgets. This patch makes sure we don't try setting mutes on those widgets. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d26ae65b43b7..0849aac449f2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,15 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); +static bool ctl_has_mute(struct snd_kcontrol *kcontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return query_amp_caps(codec, get_amp_nid(kcontrol), + get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE; +} + /* get/put callbacks for beep mute mixer switches */ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep && !beep->enabled) { + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = 0; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -263,6 +270,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, enable |= *valp; snd_hda_enable_beep_device(codec, enable); } + if (!ctl_has_mute(kcontrol)) + return 0; return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v1.2.3 From 5e68fb3cab23b327e9f15803607e697d7eea1966 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Aug 2012 14:11:09 +0200 Subject: ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040ccf8e8f..ea5775a1a7db 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- cgit v1.2.3 From c41999a23929f30808bae6009d8065052d4d73fd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Aug 2012 11:17:00 +0200 Subject: ALSA: hda - don't create dysfunctional mixer controls for ca0132 It's possible that these amps are settable somehow, e g through secret codec verbs, but for now, don't create the controls (as they won't be working anyway, and cause errors in amixer). Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/1038651 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9c0ec0a55bef..49750a96d649 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) { + snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) { + snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } -- cgit v1.2.3 From 535b6c51fe8293c88ce919cdfc4390c67a1cb6d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Aug 2012 21:25:22 +0200 Subject: ALSA: hda - Fix leftover codec->power_transition When the codec turn-on operation is canceled by the immediate power-on, the driver left the power_transition flag as is. This caused the persistent avoidance of power-save behavior. Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci/hda') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c3077d5dec6e..f560051a949e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4454,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; } -- cgit v1.2.3