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[alsa-devel] ASoC update for 2.6.33
by Mark Brown 28 Sep '09

28 Sep '09

The following changes since commit 2c9ee33d37a6f3808c4319f0849671fafa8a6f4c: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Graeme Gregory (1): ASoC: pxa-ssp increase max_channels to 8 sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-)

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[alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 28 Sep '09

28 Sep '09

Tested this on my Zotac IONITX motherboard (N330 model). Dmesg output: [ 16.464533] ALSA hda_codec.c:3081: hda_codec: model '3stack-6ch' is selected for config 174b:437b (Zotac IONITX) Patch: --- alsa-kernel/pci/hda/patch_realtek.c.orig 2009-09-24 14:50:45.000000000 -0700 +++ alsa-kernel/pci/hda/patch_realtek.c 2009-09-24 14:38:07.000000000 -0700 @@ -16724,6 +16724,7 @@ static struct snd_pci_quirk alc662_cfg_t ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x174b, 0x437b, "Zotac IONITX", ALC662_3ST_6ch), {} }; Thanks, Gus ------------------------- Message sent via Atmail Open - http://atmail.org/

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[alsa-devel] No sound after returning from suspend-to-ram
by Ozan Çağlayan 28 Sep '09

28 Sep '09

Hi, A user complains that after returning from suspend-to-ram, he isn't able to get any sound from his laptop until rebooting. I delayed reporting this one because I once thought that it may be related to Pulseaudio or some kind of mixer level problem but now I tried to suspect the kernel and the drivers as well. This is 2.6.30.5 with alsa-* 1.0.21a. I have dmesg and alsa-info outputs before and after s2ram: dmesg before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4454 alsa-info before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4455 dmesg after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4456 alsa-info after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4457 Note that a traceback is dumped in dmesg during resume. Maybe it will give a clue about the problem, dunno. And there's something like hda_codec: invalid CONNECT_LIST verb 12[2]:2100 in regular dmesg output, i wonder what does that mean too. Any ideas? Thanks, Ozan Caglayan

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[alsa-devel] Problem with mplayer and OSS on at91sam9g20(kernel 2.6.30)
by Sedji Gaouaou 28 Sep '09

28 Sep '09

Hi, We've been doing some tests here and we noticed a problem when using mplayer that we didn't have previously(on 2.6.27). While playing audio through OSS emulation on 2.6.30 / at91sam9g20ek there seems to be issue with hardware parameters. The sound does not play and a bunch of error messages are comming on the console. Here is the test case on angstrom: -- mplayer output sample ------------------------- mplayer -quiet myfile.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. [..] AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) [AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-( -- kernel output sample ------------------------- Atmel_ssc_dai: Entering atmel_ssc_hw_params() natmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=4a010404 TFMR=001f018f atmel_ssc_dai,hw_params: SSC initialized asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed [..] asoc: machine hw_params failed MPlayer interrupted by signal 2 in module: ao2_init atmel_ssc_shutdown: transmit disabled SSC_SR=0x000004cc atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback -- 2 error messages to note ------------------------- mplayer: "[AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(" kernel: "asoc: machine hw_params failed" => please note also that it is working correctly using the alsa output: mplayer -ao alsa myfile.mp3 Here is the log while using the alsa output (it is working in this case): root@at91sam:~$ mplayer -quiet -ao alsa /media/sda/audio/01.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. Audio file file format detected. Clip info: Title: Angel Artist: Massive Attack Album: Mezzanine Year: 1998 Comment: Ripped by Beto Track: 1 Genre: Rock ========================================================================== Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== atmel_ssc_startup: SSC_SR=0x204 asoc: WM8731 <-> atmel-ssc0 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=19010404 TFMR=001f018f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized transmit enabled SSC_SR=0x000104cf AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... [..] atmel_ssc_shutdown: transmit disabled SSC_SR=0x000104c3 atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown Exiting... (Quit) pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback egards, Sedji

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[alsa-devel] [PATCH 0/1] alsa-lib: pcm_hw: Always use delay ioctl in snd_pcm_delay()
by Kai Vehmanen 28 Sep '09

28 Sep '09

Hello all, this is a updated version of the alsa-lib patch I mentioned in the earlier thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020377.html That was meant to just describe the possible problem, but now I'm more confident that this should be considered for merging. The related alsa-kernel patch is 'ALSA: Add extra delay count in PCM', which added the 'runtime->delay' property. Please review if this patch makes sense. Kai Vehmanen (1): pcm_hw: Always use delay ioctl in snd_pcm_delay() src/pcm/pcm_hw.c | 22 ---------------------- 1 files changed, 0 insertions(+), 22 deletions(-)

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 28 Sep '09

28 Sep '09

Hi Peter, > Yes it is indicating so. I read through this thread I didn't find any > simple reason why it's not working. Your setup is similar with the > Pandora, i.e. McBSP functional clock is coming the codec and codec is > slave. Yes, I pretty much used the pandora as an example to write mine. > The McBSP is operating if the FSX and CLKX are toggling in this setup. > Is it so that you don't get any DMA interrupts or just few of them? I don't get any DMA int's: ~# cat /proc/interrupts CPU0 11: 0 INTC prcm 12: 0 INTC DMA 25: 0 INTC dispc 37: 5003 INTC gp timer 56: 3425 INTC i2c_omap 62: 0 INTC McBSP 63: 0 INTC McBSP > Your CLKGDV divisor value 256 means too low bit clock and sample rate > for both external 12.288 MHz and internal 96 MHz but still the DMA > should be running (if there is no bug with this divisor value). > Can you try divisor value 8 does it work then? For 48 kHz sample rate > with I2S you need a bit clock of 48 kHz*2*16 = 1.536 MHz and this you > get by dividing the 12.288 MHz with 8. > Yes, I had tried 8 before and just tried it now. Same result. It really seems like there is a DMA issue. Can you think of anything else to try before I dive into the DMA code? Rick

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Re: [alsa-devel] Problems with Intel HDA driver
by Munzir Taha (منذر طه) 27 Sep '09

27 Sep '09

Hi Takashi, Sorry, the correct link to alsa-info report is http://www.alsa-project.org/db/?f=2e50beab11390b9687490feae899217710e81afa On Yaum al-Ithnain 24 Ramadan 1430 4:23:05 pm Takashi Iwai wrote: > At Mon, 14 Sep 2009 15:15:53 +0200, > > I wrote: > > At Mon, 14 Sep 2009 16:15:02 +0300, > > > > Ozan Çağlayan wrote: > > > Takashi Iwai wrote On 14-09-2009 15:36: > > > > Yeah, I think we can add EAPD handling as default for IDT92HD71xx > > > > codecs. I've never seen machines that can't live with GPIO0 bit up. > > > > > > enable_msi doesn't seem to affect the auto-muting problem. I played > > > with every single node in hda-analyzer but can't get auto-mute work. I > > > can manually mute the HP output and the internal speakers by > > > checking/unchecking OUT checkboxes on relevant NIDs. There's a 8xn > > > matrix for GPIO settings, I played with them as well to set unsol but > > > nope. > > > > > > I want to remember that dell-m4-1 makes automute work but breaks the > > > mixer channels: > > > > > > http://www.spinics.net/lists/alsa-devel/msg26714.html > > > > > > Err, I know that I didn't answer the last question in the thread as I'm > > > a little bit lost in the NID pool and didn't have time to investigate > > > further. > > > > Could you check whether this still valid? > > > > Basically the model option doesn't change the auto-mute behavior. > > It just changes which pin to be detected and which pin to mute. > > So, if model=dell-m4-1 makes the auto-muting working, it means > > that the BIOS setup is broken. > > I looked through alsa-info output now, and found the culprit. > NID 0x0f is set as a line-out, and currently the driver doesn't > don't do auto-muting for line-out unless a headphone is present. > Changing it to a headphone should fix. Or you can give hp_detect=1 > hint via sysfs and reconfigure. > > > Takashi -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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[alsa-devel] Problems with Intel HDA driver
by Andrius Štikonas 27 Sep '09

27 Sep '09

On my new laptop there were no sound with snd-hda-intel driver, so I tried to change model parameter. It seems that with model=hp-m4 my soundcard mostly works (except that main speakers still play when I plug headphones). I've attached my alsa-info.txt script. -- Andrius Štikonas

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[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model modified: Documentation/HD-Audio-Models.txt modified: pci/hda/patch_analog.c
by Miguel de Barros 27 Sep '09

27 Sep '09

Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..7df8a08 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,143 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1984a_touchsmart_capture_source = { + .num_items = 3, + .items = { + { "Mix", 0x3 }, /* Mixer */ + { "Mic", 0x4 }, /* port-E */ + { "Internal Mic", 0x5 }, /* Digital Mic */ + }, +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4176,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4185,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4198,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4282,22 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984a_touchsmart_capture_source; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-w/QltLfzo6NHdpL3XZCU--

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[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
by Miguel de Barros 27 Sep '09

27 Sep '09

Reference: ALSA bug #0004614 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 port-A (0x11) - front hp-out port-D (0x12) - rear line out port-E (0x1c) - front mic-in port-F (0x16) - Internal speakers digital-mic (0x17) - Internal mic init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..2d603f6 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4160,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-J56U1/hvhPtbcxv8DR5c--

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