June 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

2009/6/2 Mark Brown <broonie(a)opensource.wolfsonmicro.com> > On Tue, Jun 02, 2009 at 03:24:11PM +0530, Naveen Krishna Ch wrote: > > > I am done with the entire back porting of ASoC. > > > > I can see the dapm widgets getting ON/OFF and codec registers being > changed. > > > > But, there is no audio out. Am i missing somthing > > > > A direct aplay should play the audio on the headset, right? > > I am not familiar with the TWL4030, you should ask these questions on > the list. I hit reply (Instead of replyall) Sorry, > As a first step I would check that there is data being > transferred over the digital link to the CODEC, then look at the > audio routing to ensure that the signal is routed to the output. Thanks for suggestions.. -- Cheers (: Naveen Krishna Ch :)

2 1

[alsa-devel] Palm27x-asoc: fix it to work without jack gpio specified
by Marek Vasut 02 Jun '09

02 Jun '09

Hi, the patch follows. The problem was with palmz72 which doesn't have the jack detection.

2 3

[alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

Hi Peter, I am using linux 2.6.28 kernel, there is a requirement to support multi (4) channel play back . I guess TWL4030 ASoC driver now has full fledged support for both the DAC1(TX1) and DAC2(TX2) paths in option 1 of codec operation. I am looking forward to back port the ASoC stuff drivers from 2.6.30 rc7 to 2.6.28. Is there any better way of meeting the requirement with out back porting the stuff Suggestions please. Sorry for the english,, -- Cheers (: Naveen Krishna Ch :)

3 4

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..bce2982 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,59 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + /* if there's a reserved word in the card's chosen id, reset it + * to empty. choose_default_id() will then take care to generate + * an id for the card at a later point */ + if (!snd_info_check_reserved_words(card->id)) { + *card->id = '\0'; + return; + } + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..8503a01 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,51 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[alsa-devel] [PATCH] Acer Aspire 8930G support
by Hector Martin 02 Jun '09

02 Jun '09

Getting this one to work has annoyed me to no end. Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek_orig.c 2009-06-02 07:48:47.000000000 +0200 +++ alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek.c 2009-06-02 08:32:29.000000000 +0200 @@ -223,6 +223,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -313,6 +314,8 @@ const struct hda_channel_mode *channel_mode; int num_channel_mode; int need_dac_fix; + int const_channel_count; + int ext_channel_count; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -368,6 +371,7 @@ unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; int need_dac_fix; + int const_channel_count; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -462,7 +466,7 @@ struct alc_spec *spec = codec->spec; return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - spec->multiout.max_channels); + spec->ext_channel_count); } static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, @@ -472,9 +476,12 @@ struct alc_spec *spec = codec->spec; int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; + &spec->ext_channel_count); + if(err >= 0 && spec->const_channel_count == 0) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } return err; } @@ -854,8 +861,13 @@ spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; - spec->multiout.max_channels = spec->channel_mode[0].channels; + if(preset->const_channel_count == 0) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; @@ -1456,6 +1468,44 @@ { } }; +/* + * ALC888 Acer Aspire 8930G model + */ + +static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal Front to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Internal Rear to Rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect Internal CLFE to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect HP out to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable all DACs */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, // DAC DISABLE/MUTE 1? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // bits 1-5, when set, disable DAC nids 0x02-0x06 apparently. Init value 0x38 + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, // DAC DISABLE/MUTE 2? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // some bit here disables the other DACs. Init value 0x4900 +/* Enable amplifiers */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } +}; + static struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { @@ -1508,6 +1558,17 @@ alc_automute_amp(codec); } +static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x1b; + alc_automute_amp(codec); +} + /* * ALC880 3-stack model * @@ -8708,6 +8769,7 @@ [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8746,6 +8808,8 @@ ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC888_ACER_ASPIRE_8930G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -8959,6 +9023,27 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_8930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_8930g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_8930g_init_hook, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer },

2 5

[alsa-devel] SPDIF/AES input sample rate handling?
by Robert Hancock 02 Jun '09

02 Jun '09

A question about how ALSA drivers should handle SPDIF or AES digital audio inputs: I may be eventually be writing an ALSA driver for an audio card which has multiple AES digital audio inputs. The card would support multiple sample rates on input with the provided input data being fed directly to software with no sample rate conversion. This raises the question of how to handle requests for a particular sample rate. If the user app requests recording at a sample rate of say 32KHz and we say "OK", and then they proceed to connect an input running at 44.1KHz sampling rate, the card will produce and the app will receive samples at a faster than expected rate. Presumably this is something the app just has to deal with, but will anything in the ALSA core/library puke because of it? Or is there a better way to deal with this kind of situation?

2 1

Re: [alsa-devel] Macbook[Pro] 5 6ch support
by Takashi Iwai 02 Jun '09

02 Jun '09

At Mon, 1 Jun 2009 20:21:27 +0200, Kacper Szczesniak wrote: > > On Mon, Jun 1, 2009 at 6:54 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 1 Jun 2009 15:36:35 +0200, > > Kacper Szczesniak wrote: > >> > >> On Mon, Jun 1, 2009 at 10:54 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > Hm, this patch changes the channel mapping completely. > >> > Is it intentional? > >> yes, 0xc/0xd/0xe for Front/LFE/Surr > >> > >> >> static struct snd_kcontrol_new alc885_mb5_mixer[] = { > >> >> - HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> >> - HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> >> - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> >> - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), > >> >> HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > >> >> HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), > >> > > >> > > >> > I'm wondering whether this breaks MB5, not MB5.1. If it's MB5.1 > >> > specific, please create another model instead of breaking the old > >> > one. > >> it's confirmed to work on MB5.1 MBP5.1 MBP52 > > > > Hrm, then I wonder why the old mb5 model works as is although the > > mapping is totally different... > > /* Front mixer */ > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - /* LineOut mixer */ > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - /* Front Pin: output 0 (0x0d) */ > + /* Surround mixer */ > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* LFE mixer */ > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* HP mixer */ > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* Front Pin (0x0c) */ > {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, > {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, > - /* HP Pin: output 0 (0x0c) */ > + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, > + /* LFE Pin (0x0e) */ > + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, > + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, > + /* HP Pin (0x0f) */ > {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, > {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, > + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, > > connections are different also... I've changed mixer names and connected PINs to > the right mixer nodes. Fair enough. I merged the patch now to sound git tree. Let me know if you see still any problem. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - fix audio on LG R510
by Ozan Çağlayan 02 Jun '09

02 Jun '09

Currently, LG R510 is only able to produce sound on headphones, the internal speakers are not working. The user tested and confirmed that with model=Dell headphones, internal speakers and the microphone are working flawlessly. Tested-by: Serdar Soytetir <tulliana(a)gmail.com> Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fd258e..a2133ac 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12063,6 +12063,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- 1.6.2

2 2

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Additional check for 4 channel mode
by Peter Ujfalusi 02 Jun '09

02 Jun '09

Hello, the following patch adds additional check in case of four channel mode. The codec has to be configured for TDM (DSP_A) mode when using 4 channel audio. Without the patch boards which does not prepared for the four channel mode could try to play four channel audio, but configure the interface to be in I2S mode, which is not valid. --- Peter Ujfalusi (1): ASoC: TWL4030: Check the interface format for 4 channel mode sound/soc/codecs/twl4030.c | 12 +++++++++--- 1 files changed, 9 insertions(+), 3 deletions(-)

2 2