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[alsa-devel] alc889
by Stefan Krastanov 22 Jun '09

22 Jun '09

Hi I would like to help (or at least help myself) with the creation of a model config for alc889. I'm not a sound developer. I'm using a laptop with alc889 based sound card. The reported pins are really messed up (it's hard not to use more profane expression). My laptop is Fujitsu-Siemens Amilo xi but the problems are the same as here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/71205 or here: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011295.ht… namely: 1.- no jack sensing (extremely annoying - for two years allready) 2.- misnamed mixers in volume control for gnome or alsamixer (example: front instead of headphone) 3.- the volume-control potentiometer on the side of the laptop is not working 4.- I need to mute two different channel at the same time (center and LFE) to mute one physical output (build-in sub-woofer) 5.- A great deal of other small annoyances destroying the user experience I'm using model 3stack-dig and the optical output is working (at least is lighted up - I do not have the hardware to check it) My questions are: - What I need to do, to create new model option based on 3stack-dig with different pin mapping? - Is it possible that my present configuration (3stack-dig) is using pins that should not be used? I'm asking in search of an explanation of problem 4. - What I need to do to rename the controls in alsamixer? - What to do to repair jack sensing? - that's the most important of all - What to do to repair the potentiometer for volume control? I have seen that hda-analyser can create diff files, but where should they be used. (And why is hda-analyser not working - I was forced to use http://ubuntuforums.org/showpost.php?p=7227626&postcount=33. Is it that much to expect working tools on the official website. I understand that small bug will always exist, but that one is there for at least 10 months) I won't be surprised if many of my questions are answered in the documentation, but I was unable to find anything useful. It would be enough to point me a place to start reading. I know you guys are working hard and you have done great things, but it's really hard to find something in the documentation(which is occasionally quite old) Best regards and thank you for the great work Stefan Krastanov

2 1

[alsa-devel] [PATCH] sound: remove redundant tests on unsigned
by Roel Kluin 22 Jun '09

22 Jun '09

reg and channel_id are unsigned. When negative they are wrapped and caught by the other test. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d1..2deac54 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", &reg, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if ((reg < 0x40) && (val <= 0xffffffff)) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 4d3ad79..957b130 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if ((reg < 0x49) && (val <= 0xffffffff) && (channel_id <= 2)) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f974..6ad3825 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if ((reg < 0xa0) && (val <= 0xffffffff) && (channel_id <= 3)) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5..6f8cd8d 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if ((reg > 9) && (reg != 15)) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; }

2 1

Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Mark Brown 22 Jun '09

22 Jun '09

On Fri, Jun 19, 2009 at 05:28:15PM +0800, Barry Song wrote: > 1. add AD1938 codec driver (codec) > 2. add blackfin SPORT-TDM DAI and PCM driver (platform) > 3. add bf5xx board with AD1938 driver (machine) As Liam said you really need to submit this as a patch series rather than as a single big patch - as your commit log here indicates you've got several different things going on here. > +++ b/include/sound/soc-dai.h > @@ -30,6 +30,7 @@ struct snd_pcm_substream; > #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ > #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ > #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ > +#define SND_SOC_DAIFMT_SPORT_TDM 6 /* SPORT TDM for ADI parts */ If you're going to add a new DAI format that really needs more explanation than this explaining what the DAI format is. It'd be very surprising to see hardware needing a new format. Looking at the datasheet for the ad1938 it appears that the actual format here is just normal I2S with TDM. This does not need a new DAI format or new CPU DAI, you just need to add suport for TDM to the Blackfin I2S driver. The format is fairly standard and implemented by a number of other devices. See set_tdm_slot() for setting up the higher channel counts - there's some ongoing revisions to that API so you'll want to also ensure that the code is set up so that it can cope with specification of the sample width for each slot in set_tdm_slot(). Given this I've only looked at the CODEC driver below. > diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c > new file mode 100644 > index 0000000..9aa78e1 > --- /dev/null > +++ b/sound/soc/codecs/ad1938.c > + * > + * Revision history > + * 4 June 2009 Initial version. Don't include this, git provides code history for us. > +struct snd_soc_device *ad1938_socdev; > + > +/* dac de-emphasis enum control */ > +static const char *ad1938_deemp[] = {"flat", "48kHz", "44.1kHz", "32kHz"}; For consistency with other drivers "flat" should be "None". > +/* AD1938 volume/mute/de-emphasis etc. controls */ > +static const struct snd_kcontrol_new ad1938_snd_controls[] = { > + /* DAC volume control */ > + SOC_SINGLE("DAC L1 Volume", AD1938_DAC_L1_VOL, 0, 0xFF, 1), > + SOC_SINGLE("DAC R1 Volume", AD1938_DAC_R1_VOL, 0, 0xFF, 1), These (and the other stereo pairs below) should be SOC_DOUBLE_R(). This allows ALSA to represent them as stereo controls to applications rather than as two separate controls. You should also provide TLV information so actually SOC_DOUBLE_R_TLV() if possible. > + /* DAC mute control */ > + SOC_SINGLE("DAC L1 Switch", AD1938_DAC_CHNL_MUTE, 0, 1, 1), > + SOC_SINGLE("DAC R1 Switch", AD1938_DAC_CHNL_MUTE, 1, 1, 1), These should be stereo controls too - SOC_DOUBLE() since they're in the same register. > + /* ADC mute control */ > + SOC_SINGLE("ADC L1 Switch", AD1938_ADC_CTRL0, ADC0_MUTE, 1, 1), > + SOC_SINGLE("ADC R1 Switch", AD1938_ADC_CTRL0, ADC1_MUTE, 1, 1), These too. > + /* DAC de-emphasis */ > + SOC_ENUM("Playback Deemphasis", ad1938_enum[0]), Don't put your enums in an array, use named variables for them. This makes drivers easier to maintian when you get a lot of enums. > +static int ad1938_add_controls(struct snd_soc_codec *codec) > +{ > + int err, i; > + > + for (i = 0; i < ARRAY_SIZE(ad1938_snd_controls); i++) { > + err = snd_ctl_add(codec->card, > + snd_soc_cnew(&ad1938_snd_controls[i], codec, NULL)); Use snd_soc_add_controls() here - you can replace the entire function with a call to that. > +/* dac/adc/pll poweron/off functions */ > +static int ad1938_dac_powerctrl(struct snd_soc_codec *codec, int cmd) > +{ > + int reg; > + > + reg = codec->read(codec, AD1938_DAC_CTRL0); > + if (cmd) > + reg &= ~DAC_POWERDOWN; > + else > + reg |= DAC_POWERDOWN; > + codec->write(codec, AD1938_DAC_CTRL0, reg); This should be handled by DAPM - either have a single DAC widget representing all the channels (since you don't appear to have independant control anyway) or have a bunch of dummy DAC widgets and a supply widget representing the actual power control. The same thing applies to the ADCs. > +static int ad1938_set_pll(struct snd_soc_dai *codec_dai, > + int pll_id, unsigned int freq_in, unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + > + if (freq_out) > + ad1938_pll_powerctrl(codec, 1); > + else { > + /* playing while recording, framework will poweroff-poweron pll redundantly */ > + if ((!codec_dai->capture.active) && (!codec_dai->playback.active)) > + ad1938_pll_powerctrl(codec, 0); > + } Hrm. This appears to completely ignore the frequencies supplied for the PLL and just provide power control. I suspect that you can just handle the PLL as a SND_SOC_DAPM_SUPPLY(), there seems to be no need to expose the set_pll() operation and make machine drivers call it given that there isn't any frequency configuration going on. > +static int ad1938_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + > + if (!mute) > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0); > + else > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0xff); > + > + return 0; > +} This isn't going to play well with the explicit mute controls you've got above - it's writing to the same register bits without any coordination. One or the other set of controls ought to be removed. > +static int ad1938_tdm_set(struct snd_soc_codec *codec) > +{ > + codec->write(codec, AD1938_DAC_CTRL0, (codec->read(codec, AD1938_DAC_CTRL0) & > + (~DAC_SERFMT_MASK)) | DAC_SERFMT_TDM); > + codec->write(codec, AD1938_DAC_CTRL1, 0x84); /* invert bclk, 256bclk/frame, latch in mid */ > + codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ > + codec->write(codec, AD1938_ADC_CTRL2, 0x6F); /* left high, driver on rising edge */ > + > + return 0; > +} If you use set_tdm_slot() then the BCLK/frame ratio will be set by that. Inversion of BCLK (and any other clocks) should be handled by the set_dai_fmt() operation based on the machine driver request rather than done unconditionally. > + /* bit size */ > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S16_LE: > + word_len = 3; > + break; Once you implement set_tdm_slot() you should allow the word length to be configured there if it's called or otherwise keep this code here - see Daniel Ribeiro's patche "change set_tdm_slot api to allow slot_width override" posted to the ALSA list this week. > +static int __devinit ad1938_spi_probe(struct spi_device *spi) > +{ > + spi->dev.power.power_state = PMSG_ON; > + ad1938_socdev->card->codec->control_data = spi; > + > + return 0; > +} > + > +static int __devexit ad1938_spi_remove(struct spi_device *spi) > +{ > + return 0; > +} Your device probing should all be restructured so that the SPI device for the CODEC is registered as any other SPI device rather than being set up as part of probing the ASoC device. See the wm8731 driver for an example of doing this for a SPI device. This will require that the arch code for any systems with the ad1938 do the setup of the device. > + .name = "AD1938", > + .playback = { > + .stream_name = "Playback", > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_48000, > + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, Please keep your lines to under 80 columns. > +#define AD1938_PLL_CLK_CTRL0 0 > +#define PLL_POWERDOWN 0x01 > +#define AD1938_PLL_CLK_CTRL1 1 > +#define AD1938_DAC_CTRL0 2 > +#define DAC_POWERDOWN 0x01 > +#define DAC_SERFMT_MASK 0xC0 > +#define DAC_SERFMT_STEREO (0 << 6) > +#define DAC_SERFMT_TDM (1 << 6) These defines need namespacing if they're going to appear in the headers - everything should have the AD1938_ prefix.

4 10

[PATCH] ASoC: Add a shutdown callback
by Mark Brown 22 Jun '09

22 Jun '09

Ensure that the audio subsystem is powered down cleanly when the system shuts down by providing a shutdown operation. This ensures that all the components have been returned to an off state cleanly which should avoid audio issues from partially charged capacitors or noise on digital inputs if the system is restarted quickly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f..35814ce 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4414117..55d45c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1020,6 +1020,21 @@ static int soc_remove(struct platform_device *pdev) return 0; } +static void soc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + if (!card->instantiated) + return; + + /* Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. */ + run_delayed_work(&card->delayed_work); + + snd_soc_dapm_shutdown(socdev); +} + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { @@ -1030,6 +1045,7 @@ static struct platform_driver soc_driver = { .remove = soc_remove, .suspend = soc_suspend, .resume = soc_resume, + .shutdown = soc_shutdown, }; /* create a new pcm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6534359..b9129ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2032,6 +2032,35 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_dapm_widget *w; + LIST_HEAD(down_list); + int powerdown = 0; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (w->power) { + dapm_seq_insert(w, &down_list, dapm_down_seq); + powerdown = 1; + } + } + + /* If there were no widgets to power down we're already in + * standby. + */ + if (powerdown) { + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + dapm_seq_run(codec, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + } + + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); +} + /* Module information */ MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); -- 1.6.3.1

1 0

[alsa-devel] [PATCH 2/3] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 22 Jun '09

22 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both audio and voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..fd5b30f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "Audio high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

3 3

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 22 Jun '09

22 Jun '09

The following changes since commit ef39412622b6e8f09c383de9565b07e93553fc27: Takashi Iwai (1): ASoC: Kill BUS_ID_SIZE are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Barry Song (1): ASoC: Blackfin: update the bf5xx_i2s_resume parameters Cliff Cai (1): ASoC: Blackfin: keep better track of SPORT configuration state sound/soc/blackfin/bf5xx-i2s.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH] via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
by Simon Arlott 22 Jun '09

22 Jun '09

There's a large 500ms delay in snd_via82xx_codec_wait() that, at least on my hardware, appears to be unnecessary. The rest of the init of the card works without logging any warnings or errors and both audio and mixer settings work. This adds an "nodelay" parameter to disable this (undocumented in the code) large delay improving bootup time by 489-500ms. [ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usecs vs. [ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usecs Signed-off-by: Simon Arlott <simon(a)fire.lp0.eu> --- sound/pci/via82xx.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1ef58c5..949fcaf 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -85,6 +85,7 @@ static int joystick; static int ac97_clock = 48000; static char *ac97_quirk; static int dxs_support; +static int nodelay; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); @@ -102,6 +103,8 @@ module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); module_param(dxs_support, int, 0444); MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)"); +module_param(nodelay, int, 0444); +MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); /* just for backward compatibility */ static int enable; @@ -549,7 +552,8 @@ static void snd_via82xx_codec_wait(struct snd_ac97 *ac97) int err; err = snd_via82xx_codec_ready(chip, ac97->num); /* here we need to wait fairly for long time.. */ - msleep(500); + if (!nodelay) + msleep(500); } static void snd_via82xx_codec_write(struct snd_ac97 *ac97, -- 1.6.3.1 -- Simon Arlott

2 1

[alsa-devel] Need help on MMAP
by Christopher Tsui 22 Jun '09

22 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

1 0

[alsa-devel] Acer 7730G support for ALSA.
by Denis Kup 21 Jun '09

21 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, > .num_dacs = ARRAY_SIZE(alc883_dac_nids), > .dac_nids = alc883_dac_nids, > .dig_out_nid = ALC883_DIGOUT_NID, > .dig_in_nid = ALC883_DIGIN_NID, > .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

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[alsa-devel] [PATCH] Acer Inspire 6530G model for Realtek ALC888
by Tony Vroon 21 Jun '09

21 Jun '09

The selected 4930G model seemed to keep the subwoofer 'tuba' function from operating correctly. Removing the existing PCI ID match made this work again, but it was mapped to 'Side' instead of to LFE as one would expect. This attempts to enable all functionality and keep the amount of available mixer sliders low. Any slider that had no audible effect on the output audio has been removed, and as such EAPD is not currently enabled. Signed-off-by: Tony Vroon <tony(a)linx.net> --- linux-2.6.30-git14/sound/pci/hda/patch_realtek.c.orig 2009-06-20 21:55:53.000000000 +0100 +++ linux-2.6.30-git14/sound/pci/hda/patch_realtek.c 2009-06-21 00:19:53.000000000 +0100 @@ -224,6 +224,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -1471,6 +1472,25 @@ }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Bias voltage on for external mic port */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Enable speaker output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, +/* Enable headphone output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1564,25 @@ } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 2, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -8153,6 +8192,19 @@ { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9021,7 +9073,7 @@ SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9255,6 +9307,24 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_6530G] = { + .mixers = { alc888_acer_aspire_6530_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_6530g_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_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_acer_aspire_6530_sources, + .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 },

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