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[alsa-devel] [PATCH] ALSA: hda - Consider additional capture source/selector in ALC889
by Herton Ronaldo Krzesinski 13 Feb '09

13 Feb '09

Currently code for capture source support in ALC889 only considers capture mixers. This change adds aditional support for ADC+selector present in ALC889, taking into account also the presence of an additional DMIC connection item in the selector. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 105 ++++++++++++++++++++++++++++++----------- 1 files changed, 77 insertions(+), 28 deletions(-) Note: I did this patch without hardware to test on, just based on alsa-info.txt from one machine with ALC889 plus the ALC889 datasheet available from Realtek website, so this is just build tested. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 863ab95..d81cb5e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -238,6 +238,13 @@ enum { ALC883_MODEL_LAST, }; +/* styles of capture selection */ +enum { + CAPT_MUX = 0, /* only mux based */ + CAPT_MIX, /* only mixer based */ + CAPT_1MUX_MIX, /* first mux and other mixers */ +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -276,7 +283,7 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ + int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -294,7 +301,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ @@ -396,7 +403,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->is_mix_capture) { + if (spec->capture_style && + !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -4130,7 +4138,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -4279,7 +4287,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -5487,7 +5495,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -5647,7 +5655,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -7087,7 +7095,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->is_mix_capture = 1; /* matrix-style capture */ + spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -7155,10 +7163,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids + static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -8977,6 +8989,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err = alc880_parse_auto_config(codec); + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; if (err < 0) return err; @@ -8990,6 +9004,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* setup input_mux for ALC889 */ + if (codec->vendor_id == 0x10ec0889) { + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + for (i = 1; i < 3; i++) + memcpy(&spec->private_imux[i], + &spec->private_imux[0], + sizeof(spec->private_imux[0])); + imux->items[imux->num_items].label = "Int DMic"; + imux->items[imux->num_items].index = 0x0b; + imux->num_items++; + spec->num_mux_defs = 3; + spec->input_mux = spec->private_imux; + } + } + return 1; /* config found */ } @@ -9053,14 +9087,36 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC888 Analog"; spec->stream_name_digital = "ALC888 Digital"; } + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; spec->stream_name_digital = "ALC889 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); + spec->adc_nids = alc889_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc889_capsrc_nids; + spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style + capture */ break; default: spec->stream_name_analog = "ALC883 Analog"; spec->stream_name_digital = "ALC883 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -9071,13 +9127,6 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - spec->is_mix_capture = 1; /* matrix-style capture */ if (!spec->cap_mixer) set_capture_mixer(spec); @@ -10512,7 +10561,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -10881,7 +10930,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -11539,7 +11588,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -11657,7 +11706,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -12511,7 +12560,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, */ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; imux->items[imux->num_items].label = "Int Mic"; imux->items[imux->num_items].index = 0x05; imux->num_items++; @@ -12567,7 +12616,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; /* set default input source */ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, @@ -13483,7 +13532,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -13620,7 +13669,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); @@ -14724,7 +14773,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -14803,7 +14852,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; set_capture_mixer(spec); @@ -16397,7 +16446,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -16528,7 +16577,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) @@ -16613,7 +16662,7 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); -- 1.6.1

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[alsa-devel] ASoC update for 2.6.29
by Mark Brown 13 Feb '09

13 Feb '09

The following changes since commit 272edb00493af32c609f43bdf1d75141756fd999: Lopez Cruz, Misael (1): ASoC: Update SDP3430 machine driver for snd_soc_card are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Mark Brown (1): ASoC: Only register AC97 bus if it's not done already sound/soc/soc-core.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] ASoC: Only register AC97 bus if it's not done already
by Mark Brown 13 Feb '09

13 Feb '09

ASoC supports both explicit codec drivers for AC97 devices and a simple driver which uses the standard ALSA AC97 framework for codec support. When used with the generic AC97 codec support that will provide the ad hoc AC97 device for drivers like touchscreens to attach to so the core shouldn't do so. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55fdb4a..ec3f8bb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - if (ac97) { + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); -- 1.5.6.5

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - Consider additional capture source/selector in ALC889
by Takashi Iwai 13 Feb '09

13 Feb '09

At Fri, 13 Feb 2009 13:29:38 +0100, Mauro Condarelli wrote: > > Takashi Iwai wrote: > > At Fri, 13 Feb 2009 11:10:10 +0000 (UTC), > > Mauro Condarelli wrote: > > > >> Hi, > >> sorry to disturb. > >> I have an acer Aspire 8930 and I am fighting with sound. > >> I have an up-to-date ubuntu Intrepid distribution. > >> I had NO sound at all. > >> I followed instructions upgrading to latest alsa-driver-1.0.19 added "options > >> snd-hda-intel model=acer" to /etc/modprobe.d/alsa-base and > >> "/usr/local/bin/hda-verb /dev/snd/hwC0D0 0x15 SET_EAPD_BTLENABLE 2" to > >> /etc/rc.local. > >> JOY! I have sound. > >> So far so good. > >> Now my problem is I do not have capture (intrnal or external mic are not > >> working); I kept searching and found this thread. > >> I downloaded the alsa-driver-snapshot-tiwai from ftp.kernel.org, applied the > >> last patch (it seemed not applied in the snapshot) which went in with some line > >> discrepancies. > >> Removed the above settings and insatalled the new driver. > >> Result is now I have a *very* low volume audio and I'm unsure the mic is working > >> at all. > >> I'm obviously missing something. > >> Can someone be so kind to point me in the right direction, please? > >> I'm (obviously!) ready to offer all info required. I simply didn't want to make > >> this post any longer. > >> > > > > Try the latest snapshot below with model=auto option. > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… > > > > If this doesn't work, please run alsa-info.sh with --no-upload option > > and attach the generated file. The script is found at > > http://www.alsa-project.org/alsa-info.sh > > > > > Hi Takashi, > many thanks for the really fast answer! > > > I was already using that snapshot (albeit with further patches). > I recompiled and reinstalled a fresh copy (to be on the safe side), but > the results are the same: *very* low volume even with all mixers up. And it's louder if you pass the model option? If so, please run alsa-info.sh on both states to compare. thanks, Takashi

1 0

[alsa-devel] [PATCH] snd-hda-intel: add quirk for Lenovo X200 laptop dock
by Aristeu Sergio Rozanski Filho 13 Feb '09

13 Feb '09

Currently the HP connector on X200 dock doesn't detect when a HP is connected nor allows sound to be played using it. This patch fixes the problem by adding a quirk for this specific model. It's possible that others have the same NID (0x19) to report when dock HP is connected, but I don't have access to any. Please Cc me in the reply since I'm not subscribed to alsa-devel@. Signed-off-by: Aristeu Rozanski <aris(a)redhat.com> --- sound/pci/hda/patch_conexant.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) --- a/sound/pci/hda/patch_conexant.c 2009-02-12 17:03:19.000000000 -0500 +++ b/sound/pci/hda/patch_conexant.c 2009-02-12 17:33:20.000000000 -0500 @@ -1806,6 +1806,40 @@ static struct hda_verb cxt5051_init_verb { } /* end */ }; +static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Docking HP */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1823,18 +1857,21 @@ static int cxt5051_init(struct hda_codec enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_LENOVO_X200] = "Lenovo X200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), {} }; @@ -1875,6 +1912,8 @@ static int patch_cxt5051(struct hda_code codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; + case CXT5051_LENOVO_X200: + spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; default: case CXT5051_LAPTOP: codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;

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[alsa-devel] [PATCH] ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5 model
by Herton Ronaldo Krzesinski 13 Feb '09

13 Feb '09

Change HP dv7 quirk: although reported to work with hp-m4 model (https://bugzilla.novell.com/show_bug.cgi?id=445321), the original report doesn't contain info about testing of internal microphone. Recently I received a report about internal mic not working (https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be related with the forced line in on pin 0x0e done with hp-m4 model. Thus change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported to be fixed on a provided kernel with this change (https://qa.mandriva.com/show_bug.cgi?id=44855#c196). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3c84817..efc1aff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1811,7 +1811,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, "HP dv5", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, - "HP dv7", STAC_HP_M4), + "HP dv7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7, "HP dv4", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, -- 1.6.1.3

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[alsa-devel] [PATCH] hda: Add STAC_DELL_S14 quirk
by Matthew Ranostay 13 Feb '09

13 Feb '09

Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins in pin_nids for stac92hd83xxx. Also reorganized connection selection code for the respective ports per quirk define. --- sound/pci/hda/patch_sigmatel.c | 52 +++++++++++++++++++++++++++------------- 1 files changed, 35 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 12b3088..ed1c25c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { enum { STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, + STAC_DELL_S14, STAC_92HD83XXX_MODELS }; @@ -480,10 +481,9 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd83xxx_pin_nids[14] = { +static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x1d, 0x1e, 0x1f, 0x20 + 0x0f, 0x10, 0x11, 0x1f, 0x20, }; #define STAC92HD71BXX_NUM_PINS 13 @@ -857,9 +857,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, @@ -1730,21 +1730,28 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd83xxx_pin_configs[14] = { +static unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x01451160, 0x98560170, }; +static unsigned int dell_s14_pin_configs[10] = { + 0x02214030, 0x02211010, 0x02a19020, 0x01014050, + 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160, + 0x40f000f0, 0x40f000f0, +}; + static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, + [STAC_DELL_S14] = dell_s14_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", + [STAC_DELL_S14] = "dell-s14", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1753,6 +1760,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, + "unknown Dell", STAC_DELL_S14), {} /* terminator */ }; @@ -4822,6 +4831,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; int num_dacs; + hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4840,15 +4850,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; - - /* set port 0xe to select the last DAC - */ - num_dacs = snd_hda_get_connections(codec, 0x0e, - conn, STAC92HD83_DAC_COUNT + 1) - 1; - - snd_hda_codec_write_cache(codec, 0xe, 0, - AC_VERB_SET_CONNECT_SEL, num_dacs); - spec->init = stac92hd83xxx_core_init; spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); @@ -4900,6 +4901,23 @@ again: return err; } + switch (spec->board_config) { + case STAC_DELL_S14: + nid = 0xf; + break; + default: + nid = 0xe; + break; + } + + num_dacs = snd_hda_get_connections(codec, nid, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + /* set port X to select the last DAC + */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + codec->patch_ops = stac92xx_patch_ops; codec->proc_widget_hook = stac92hd_proc_hook; -- 1.5.6.3

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[alsa-devel] ASoC ac97: kobject warnings
by Manuel Lauss 13 Feb '09

13 Feb '09

Hi Mark, the asoc generic ac97 codec support throws a warning on 2.6.29: Advanced Linux Sound Architecture Driver Version 1.0.18a. No device for DAI WM8731 No device for DAI au1xpsc_i2s No device for DAI au1xpsc_ac97 AC97 SoC Audio Codec 0.6 asoc: AC97 HiFi <-> au1xpsc_ac97 mapping ok kobject (8f9279ec): tried to init an initialized object, something is seriously wrong. Call Trace: [<80106bcc>] dump_stack+0x8/0x34 [<8022c680>] kobject_init+0x50/0xcc [<8026d914>] device_initialize+0x2c/0x9c [<8026e274>] device_register+0x14/0x28 [<802ff2b8>] snd_soc_init_card+0x18c/0x24c [<803024a4>] ac97_soc_probe+0x12c/0x178 [<802fe76c>] snd_soc_instantiate_cards+0x1d4/0x2f4 [<802fe8fc>] soc_probe+0x70/0xb8 [<8027044c>] driver_probe_device+0xe0/0x1b8 [<8026f884>] bus_for_each_drv+0x60/0xb0 [<80270670>] device_attach+0x60/0x88 [<8026f63c>] bus_attach_device+0x38/0x94 [<8026e0cc>] device_add+0x390/0x524 [<80271a90>] platform_device_add+0x14c/0x1bc [<80462bd4>] db1200_audio_load+0xd8/0x10c [<801090e0>] __kprobes_text_end+0x60/0x1c8 [<804532d0>] kernel_init+0x9c/0x108 [<8010cf90>] kernel_thread_helper+0x10/0x18 ALSA device list: #0: DB1200 AC97 Audio (AC97) The asoc machine code is very similar to sound/soc/au1x/sample-ac97.c (full source here: http://mlau.at/db1200.c). I didn't bisect it; however it started after the first ASoC patches went in mainline after 2.6.28's release. Please have a look! Thanks, Manuel Lauss

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[PATCH] ASoC: Only register AC97 bus if it's not done already
by Mark Brown 12 Feb '09

12 Feb '09

ASoC supports both explicit codec drivers for AC97 devices and a simple driver which uses the standard ALSA AC97 framework for codec support. When used with the generic AC97 codec support that will provide the ad hoc AC97 device for drivers like touchscreens to attach to so the core shouldn't do so. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 580a1a5..ae1d7a9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1382,7 +1382,9 @@ int snd_soc_init_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - if (ac97) { + /* Only instantiate AC97 if not already done (normally only by + * the adaptor for the generic AC97 subsystem). */ + if (ac97 && !codec->ac97) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); -- 1.5.6.5

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Re: [alsa-devel] [PATCH 0/4] Intel HDMI audio fixes
by Takashi Iwai 12 Feb '09

12 Feb '09

At Wed, 11 Feb 2009 14:54:48 -0800, Shane W wrote: > > On Wed, Feb 11, 2009 at 11:42:00PM +0100, Takashi Iwai wrote: > > At Wed, 11 Feb 2009 14:32:41 -0800, > > Shane W wrote: > > > missing. The order in the wav is: > > > front left > > > centre > > > front right > > > rear left > > > rear right > > > sub buz > > > > The order assumed here doesn't match with ALSA mapping. > > In ALSA, > > front left > > front right > > rear left > > rear right > > center > > LFE > > Sorry I meant the spoken order, I've no idea what the > channel mapping of a 6 channel wav is. > > > Better to test with speaker-test program whether the mapping is > > correct. > > Ah didn't know about this program. Ok results with: > speaker-test -D hw:0,3 -s n > where n ranged from 0 to 6 > > -s 1 speaker-test says front left actual is front left > likewise with -s 2 for front right > -s 3 speaker-test says should be rear left actual is lfe > that is the sound came from lfe > -s 4 speaker-test says rear right actual is centre > -s 5 speaker-test says centre actual was rear left > and -s 6 speaker-test says lfe actual is rear right Thanks. Also make sure that you pass -c 6 option. Takashi

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