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[alsa-devel] [PATCH] hda-intel: Add Quanta IL1 ALC267 model
by Herton Ronaldo Krzesinski 14 Apr '08

14 Apr '08

This adds support for Quanta IL1 mini-notebook to alsa, defining a new model for it. It comes with an ALC267 codec chip. Some notes about this model: * In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common amp mute, to avoid conflict with mixer switch (mixer and automute use the same nid). * The only connected capture sources in the hardware are the internal mic and external mic jack. So instead of using an input source selector like on other ALC268 models, the mic automute automatically switch between captures. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/Documentation/ALSA-Configuration.txt.orig alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa-kernel/Documentation/ALSA-Configuration.txt.orig 2008-03-25 16:30:07.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-25 16:33:52.000000000 -0300 @@ -829,7 +829,8 @@ Prior to version 0.9.0rc4 options had a basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) - ALC268 + ALC267/268 + quanta-il1 Quanta IL1 mini-notebook 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-25 16:30:07.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-25 16:33:52.000000000 -0300 @@ -105,6 +105,7 @@ enum { /* ALC268 models */ enum { + ALC267_QUANTA_IL1, ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, @@ -9896,6 +9897,64 @@ static void alc268_dell_unsol_event(stru #define alc268_dell_init_hook alc268_dell_automute +static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + { } +}; + +static struct hda_verb alc267_quanta_il1_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc267_quanta_il1_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); +} + +static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, + present ? 0x00 : 0x01); +} + +static void alc267_quanta_il1_automute(struct hda_codec *codec) +{ + alc267_quanta_il1_hp_automute(codec); + alc267_quanta_il1_mic_automute(codec); +} + +static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc267_quanta_il1_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc267_quanta_il1_mic_automute(codec); + break; + } +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -10318,6 +10377,7 @@ static void alc268_auto_init(struct hda_ * configuration and preset */ static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC267_QUANTA_IL1] = "quanta-il1", [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", @@ -10340,11 +10400,27 @@ static struct snd_pci_quirk alc268_cfg_t SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), + SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), {} }; static struct alc_config_preset alc268_presets[] = { + [ALC267_QUANTA_IL1] = { + .mixers = { alc267_quanta_il1_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc267_quanta_il1_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc267_quanta_il1_unsol_event, + .init_hook = alc267_quanta_il1_automute, + }, [ALC268_3ST] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, alc268_beep_mixer }, -- []'s Herton

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Re: [alsa-devel] [PATCH] es1968: fix jitter on some maestro cards
by Rene Herman 14 Apr '08

14 Apr '08

On 30-03-08 00:49, Andreas Mueller wrote: > This patch suppresses jitter on several Maestro cards in stereo mode > (ALSA of course). > > The patch is also incorporated in the *BSD drivers where I "ported" it from. > I am barely used to driver-development so please take a look (esp. the channel > parameters), nevertheless it completely works for me. > > Without this patch most of the stereo audio gets out of sync and really > distorted (oss-emulation with mplayer at 48000khz worked somehow). > > CCed to those listed in the .c file. Added the alsa-devel list (and re-attached the patch). My TerraTec DMX (ESS Canyon3D, ES1970MS-3D) doesn't need this but doesn't seem to suffer from it either. Otherwise no idea about this. Rene. --- sound/pci/es1968.c.old 2008-03-29 19:17:16.771116317 +0100 +++ sound/pci/es1968.c 2008-03-30 00:14:35.363057160 +0100 @@ -1816,7 +1816,24 @@ return 0; } +/* + * suppress jitter on some maestros when playing stereo + */ +static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es) +{ + unsigned int cp1; + unsigned int cp2; + unsigned int diff; + cp1 = __apu_get_register(chip, 0, 5); + cp2 = __apu_get_register(chip, 1, 5); + diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1); + + if (diff > 1) { + __maestro_write(chip, IDR0_DATA_PORT, cp1); + } +} + /* * update pointer */ @@ -1937,8 +1954,11 @@ struct esschan *es; spin_lock(&chip->substream_lock); list_for_each_entry(es, &chip->substream_list, list) { - if (es->running) + if (es->running) { snd_es1968_update_pcm(chip, es); + if (es->fmt & ESS_FMT_STEREO) + snd_es1968_suppress_jitter(chip, es); + } } spin_unlock(&chip->substream_lock); if (chip->in_measurement) {

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[alsa-devel] [PATCH] hda: EAPD power management
by Matthew Ranostay 14 Apr '08

14 Apr '08

Power management support for EAPD enabled laptops, when headphones are sensed it pulls the EAPD GPIO line low to power it down. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 0dd2b6cb1506 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Apr 09 08:16:33 2008 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Apr 11 15:09:39 2008 -0400 @@ -129,6 +129,7 @@ struct sigmatel_spec { unsigned int hp_detect: 1; /* gpio lines */ + unsigned int eapd_mask; unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; @@ -3183,6 +3184,10 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) @@ -3191,6 +3196,10 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data | + spec->eapd_mask); } } @@ -3478,7 +3487,7 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; switch (spec->board_config) { @@ -3584,7 +3593,10 @@ again: spec->aloopback_shift = 0; /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0x1; + spec->gpio_mask = 0x01; + spec->gpio_dir = 0x01; + spec->gpio_mask = 0x01; + spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3770,7 +3782,7 @@ static int patch_stac927x(struct hda_cod case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x01; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3787,7 +3799,7 @@ static int patch_stac927x(struct hda_cod /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x04; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; spec->gpio_data = 0x04; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; @@ -3799,7 +3811,7 @@ static int patch_stac927x(struct hda_cod break; default: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3903,6 +3915,7 @@ static int patch_stac9205(struct hda_cod (AC_USRSP_EN | STAC_HP_EVENT)); spec->gpio_dir = 0x0b; + spec->eapd_mask = 0x01; spec->gpio_mask = 0x1b; spec->gpio_mute = 0x10; /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, @@ -3912,7 +3925,7 @@ static int patch_stac9205(struct hda_cod break; default: /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; break; }

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[alsa-devel] [PATCH] hda: Correct SPDIF out default config
by Matthew Ranostay 14 Apr '08

14 Apr '08

Several laptops have have the SPDIF out defined as 'Digital other out' when it should be 'SPDIF out' in the default config. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 0dd2b6cb1506 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Apr 09 08:16:33 2008 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Apr 11 16:02:57 2008 -0400 @@ -3778,6 +3778,13 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + switch (codec->subsystem_id) { + case 0x10280209: + case 0x1028022e: + /* correct the device field to SPDIF out */ + stac92xx_set_config_reg(codec, 0x21, 0x01442070); + break; + }; /* configure the analog microphone on some laptops */ stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */

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[alsa-devel] [ALC262 patch_realtek.c] Fujitsu Lifebook PC speaker signal
by Tony Vroon 14 Apr '08

14 Apr '08

Good afternoon, Could you please consider the following patch for the 2.6.26 merge window. The legacy PC speaker signal was not routed to outputs. The codec is not prevented from powering down in this patch, although I suppose one could argue that perhaps it should be. Let me know if anyone feels strongly one way or the other. Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin

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[alsa-devel] [PATCH]PCI quirk for laptop LG which use CMI9880
by Jiang zhe 14 Apr '08

14 Apr '08

Hi, Please refer to [0003874] on the alsa mantis. This patch added the pci quirk. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

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[alsa-devel] [PATCH] soc - Include register in DAPM debug output
by Mark Brown 14 Apr '08

14 Apr '08

When logging register changes in DAPM debug output include the register number. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index abac684..e0637d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -226,7 +226,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(POP_TIME); } - dbg("reg old %x new %x change %d\n", old, new, change); + dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; } -- 1.5.4.4

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[alsa-devel] [PATCH]PCI quirk for MSI laptop
by Jiang zhe 14 Apr '08

14 Apr '08

Hi, Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

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[alsa-devel] [PATCH] snd_usb_caiaq: use likely()/unlikely()
by Daniel Mack 13 Apr '08

13 Apr '08

Use likely() and unlikely() in code which is called regularily. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

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[alsa-devel] External IO plugin question
by Taek Joo Kim 12 Apr '08

12 Apr '08

Hi. I am working on ALSA external IO plugin now. Its function is similar to the pulse plugin. A server sends sound to multiple clients. When I play sound with my plugin, the sound is played really fast. It means that it is like playing with x5 or x10 speed. I did not write any code for synchronization between server and client. To try to fix it, I saw the pulse plugin. I guess that it uses some synchronization code (poll_descriptors), but I am not sure. My questions are: 1. Which part should take care of playing sound properly in this architecture, server or client? 2. Do I need to deal with synchronization between server and client? 3. If 2 is yes, is there any hints to do this with multiple clients? Thank you in advance.

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