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[alsa-devel] [PATCH 0/3] ASoC updates
by Mark Brown 16 Sep '08

16 Sep '08

The following changes since commit 78bd2cd2124d3a77f7f78af47849ea571a73f2a1: Dmitry Baryshkov (1): ALSA: Separate common pxa2xx-pcm code are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (3): ASoC: Fix dependency on AC97_BUS ASoC: DAPM support for ADC on WM9713 PCM interface ASoC: Add WM8750 SPI support sound/soc/Kconfig | 1 + sound/soc/codecs/wm8750.c | 70 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8750.h | 1 + sound/soc/codecs/wm9713.c | 12 ++++++- 4 files changed, 81 insertions(+), 3 deletions(-)

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[alsa-devel] [PATCH 0/7] HDA patch_via.c update
by Harald Welte 16 Sep '08

16 Sep '08

Hi! As part of my work for VIA I have stumbled upon their current internal HDA codec driver and it's various differences against the current mainline driver. I have spent the better part of two days separating the individual changesets out of the big diff between VIA's driver and current mainline linux-2.6 (which is identical to alsa-kernel.git). After this merge, I will work with product management and R&D inside VIA that all individual changesets (bugfixes / enhancements) will be sent ASAP for mainline inclusion to prevent any large differencees betwee the two trees. If you have any questions regarding VIA's hardware or need access to data sheets or the like, I am happy to assist. -- - Harald Welte <HaraldWelte(a)viatech.com> http://linux.via.com.tw/ ============================================================================ VIA Open Source Liaison

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[alsa-devel] Dai for at91sam9g20.
by Sedji Gaouaou 16 Sep '08

16 Sep '08

Hi, I have written a dai to use a wolfson 8731 with the sam9g20 atmel board. I would like to know what is the proper place to post my patch, is it here? Regards, Sedji

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[alsa-devel] Unable to play aplay
by vasudha rao 16 Sep '08

16 Sep '08

Hi , I want to play and record audio file in my custom atmel 9263 board running with linux(2.6.20). when I tried to play aplay for a wave file, I got $aplay /home/alarm.wav ALSA lib pcm.c:1707:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:446: audio open error: No such file or directory $aplay -l aplay: device_list:187: no soundcards found... $cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0] : control Can you tel me where I am doing wrong..?? Thanks in advance..

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[alsa-devel] [PATCH 0/3] HDA patch_via.c updates
by Harald Welte 15 Sep '08

15 Sep '08

Hi! Unfortunately there is one crash (headphone related), one non-working feature (second S/PDIF out) and one regular bugfix since my seven-patch series from last week. Please review + apply the following patches to your tree. I've tested them against sound-2.6/topic/hda and they apply fine. If you have any questions, please keep Logan Li in the Cc, he is the engineer who actually implemented those changes and knows all the details. Thanks in advance, -- - Harald Welte <HaraldWelte(a)viatech.com> http://linux.via.com.tw/ ============================================================================ VIA Open Source Liaison

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[alsa-devel] [PATCH 0/6] ASoC updates
by Mark Brown 15 Sep '08

15 Sep '08

The following changes since commit 3e3b5651631629482e0e1541f8c4298a435bb100: Mark Brown (1): ALSA: ASoC: Use codec digital mute when stopping playback are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Alan Horstmann (1): ASoC: Fix WM8731 SPI write Dmitry Baryshkov (4): ASoC: Support runtime selection of CPU in pxa2xx-i2s ALSA: Separate common pxa2xx-ac97 code ALSA: pxa2xx-ac97-lib: support building for several CPUs ALSA: Separate common pxa2xx-pcm code Kenneth Kiraly (1): ASoC: Add WM8971 CODEC driver include/sound/pxa2xx-lib.h | 45 ++ sound/arm/Kconfig | 9 + sound/arm/Makefile | 4 + sound/arm/pxa2xx-ac97-lib.c | 384 ++++++++++++++++++ sound/arm/pxa2xx-ac97.c | 247 +----------- sound/arm/pxa2xx-pcm-lib.c | 278 +++++++++++++ sound/arm/pxa2xx-pcm.c | 252 +----------- sound/arm/pxa2xx-pcm.h | 13 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8731.c | 5 +- sound/soc/codecs/wm8971.c | 942 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8971.h | 64 +++ sound/soc/pxa/Kconfig | 3 + sound/soc/pxa/pxa2xx-ac97.c | 272 +------------ sound/soc/pxa/pxa2xx-i2s.c | 20 +- sound/soc/pxa/pxa2xx-pcm.c | 261 +------------ sound/soc/pxa/pxa2xx-pcm.h | 15 - 18 files changed, 1802 insertions(+), 1018 deletions(-) create mode 100644 include/sound/pxa2xx-lib.h create mode 100644 sound/arm/pxa2xx-ac97-lib.c create mode 100644 sound/arm/pxa2xx-pcm-lib.c create mode 100644 sound/soc/codecs/wm8971.c create mode 100644 sound/soc/codecs/wm8971.h

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[alsa-devel] alsa on Apple Mac Pro Tower ( macPro3, 1, quad core, xenon)
by S. Aguinaga 15 Sep '08

15 Sep '08

I downloaded the latest development drivers, lib, and utils (1.0.18rc3) and compiled and installed it again. Next I made sure that the /etc/modprobe.d/sound had the following line: options snd-hda-intel model=imac24 and first I checked the gui tool called "PulseAudio Volume Control" from my "multimedia" section, and it crashes! Next I tried the aplay somesound.wav file from the command line, it my Mac Pro sings!!! I don't know what the PulseAudio issues are, but ... oh well!! Next I checked that Amrok works using the alsa engine, and it does!! If I plug in a headphone set, and I play music again it comes out of the MacPro speaker only, but no from the headphones. Do you have a handle this, if you guys have some links for me to read, please send them my way! Thank you. // Sal Aguinaga // Northwestern University // Evanston, IL ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, August 19, 2008 5:48:47 AM Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 At Mon, 18 Aug 2008 10:26:42 -0700 (PDT), S. Aguinaga wrote: > > ============================================== > Trying to figure out where to add this "model=macpro" > and it looks like it should be in a file called modprobe.conf or an options file under /etc/modprobe.d/ > > but I don't have (with my installation -- fedora {9} ) those files, do you have any input on this? You can create any file, typically, /etc/modprobe.d/sound or so. Put the following line there options snd-hda-intel model=macpro Then reload the module, check the sysfs file /sys/modules/snd_hda_intel/parameters/macpro to see whether the specified option was really passed properly. Takashi > below are the only files I have. > > /etc/modprobe.d/ > > anaconda > blacklist-visor > i915modeset > blacklist > floppy-pnp > modprobe.conf.dist > > > ============================================== > Subject: Re: [alsa-devel] Alsa-devel Digest, Vol 18, Issue 45 > > At Sat, 16 Aug 2008 06:15:46 -0700 (PDT), > S. Aguinaga wrote: > > > > S. Aguinaga wrote: > > > > > > Hello Stan, > > > > > > This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. > > > > > > 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" > > > 2. aplay does not produce sound, period! > > > 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. > > > > > > I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: > > > > > > sudo cat /var/log/messages | grep ALSA > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 > > > Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped > (snip) > > These messages are harmless. It's just information from the driver, > enabled only when you build it with CONFIG_SND_DEBUG_VERBOSE=y. > > > I think your problem is a misidentification of your sound chip. The > > model being used for your 889A appears to be incompatible, thus all > > the errors above. It appears that the macpro ALC885 is being selected > > before it gets to the choice for the ALC889A. Perhaps a developer can > > look into this. > > Well, ALC882, ALC885 and ALC889A are almost compatible. In the > driver, they are all handled in the same routine. > > There are a few models for intel macs with ALC88x. Try to specify > either model=macpro, model=mbp3 or model=imac24 as the module option > of snd-hda-intel driver. > > > Takashi > > > > >

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[alsa-devel] Dev: Alsa sequencer problem (Why ???)
by Gilbux (free) 15 Sep '08

15 Sep '08

Hello, I am trying to write an alsa sequencer. So I read as I can some of your tutorials, and I don't really understand what I am doing wrong, please help me. I want to send scheduled event with the queuing system, but with RTC timer the queue doesn't start, and with the other timer the queue start but no events are sent. I have joined to this email a small code, with the same kind of functions I am using, and some /proc/asound/seq/* outputs. Regards, Gilbux /proc/asound/seq/* before starting the queue : Client info cur clients : 2 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 128 : "TestQSchedEv" [User] Port 0 : "OutputPort" (R-e-) Output pool : Pool size : 500 Cells in use : 16 Peak cells in use : 16 Alloc success : 16 Alloc failures : 0 queue 0: [TQSE_Queue] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 16 timer state : Stopped timer PPQ : 192 current tempo : 500000 current BPM : 120 current time : 0.000000000 s current tick : 0 Timer for queue 0 : RTC timer Period time : 0.000000000 Skew : 65536 / 65536 ----- /proc/asound/seq/* after starting the queue : Client info cur clients : 2 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 128 : "TestQSchedEv" [User] Port 0 : "OutputPort" (R-e-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 16 Alloc success : 16 Alloc failures : 0 queue 0: [TQSE_Queue] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 192 current tempo : 500000 current BPM : 120 current time : 0.000000000 s current tick : 0 Timer for queue 0 : RTC timer Period time : 0.000976562 Skew : 65536 / 65536

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[alsa-devel] [PATCH] ALSA: HDA: patch_analog: Fix SPDIF output on AD1989B
by robbat2＠gentoo.org 15 Sep '08

15 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The SPDIF pins for AD1989 are not enabled by default. Set OUT bit so that they actually work. Also initialize the HDMI SPDIF at the same time. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b037324..42b1a23 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2255,8 +2255,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { - /* SPDIF out pin */ + /* SPDIF-1 out pin */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + /* SPDIF-2/HDMI out pin */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ { } }; -- 1.6.0.1

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[alsa-devel] [PATCH] hda: 92hd73xx fixes
by Matthew Ranostay 15 Sep '08

15 Sep '08

Added support for defining the number of amps and the creation of the private_dimux dynamically for the 92hd73xx codec family. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f41ba17..fff27cc 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -156,13 +156,11 @@ struct sigmatel_spec { unsigned int num_pwrs; unsigned int *pwr_mapping; hda_nid_t *pwr_nids; - hda_nid_t *amp_nids; hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; - unsigned int cur_amux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -201,6 +199,9 @@ struct sigmatel_spec { unsigned int cur_mux[3]; struct hda_input_mux *sinput_mux; unsigned int cur_smux[2]; + unsigned int cur_amux; + hda_nid_t *amp_nids; + unsigned int num_amps; unsigned int powerdown_adcs; /* i/o switches */ @@ -247,8 +248,9 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd73xx_amp_nids[4] = { - 0x0b, 0x0c, 0x0e, 0 +#define DELL_M6_AMP 2 +static hda_nid_t stac92hd73xx_amp_nids[3] = { + 0x0b, 0x0c, 0x0e }; #define STAC92HD73XX_NUM_DMICS 2 @@ -963,29 +965,33 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; +#define DELL_M6_MIXER 6 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - + /* start of config #1 */ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + + /* start of config #2 */ + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + { } /* end */ }; @@ -2956,17 +2962,19 @@ static int stac92xx_auto_create_amp_input_ctls(struct hda_codec *codec) struct hda_input_mux *amp_mux = &spec->private_amp_mux; int i, err; - for (i = 0; i < ARRAY_SIZE(stac92xx_amp_labels); i++) { + for (i = 0; i < spec->num_amps; i++) { amp_mux->items[amp_mux->num_items].label = stac92xx_amp_labels[i]; amp_mux->items[amp_mux->num_items].index = i; amp_mux->num_items++; } - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, - "Amp Selector Capture Switch", 0); - if (err < 0) - return err; + if (spec->num_amps > 1) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, + "Amp Selector Capture Switch", 0); + if (err < 0) + return err; + } return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, "Amp Capture Volume", HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); @@ -3342,12 +3350,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - if (spec->amp_nids) { + if (spec->num_amps) { err = stac92xx_auto_create_amp_input_ctls(codec); if (err < 0) return err; } - if (spec->num_dmics > 0) + if (spec->num_dmics > 0 && !spec->dinput_mux) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; @@ -3375,8 +3383,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) - spec->dinput_mux = &spec->private_dimux; + spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; spec->amp_mux = &spec->private_amp_mux; @@ -3975,9 +3982,9 @@ static struct hda_input_mux stac92hd73xx_dmux = { .num_items = 4, .items = { { "Analog Inputs", 0x0b }, - { "CD", 0x08 }, { "Digital Mic 1", 0x09 }, { "Digital Mic 2", 0x0a }, + { "CD", 0x08 }, } }; @@ -4048,12 +4055,14 @@ again: spec->dmux_nids = stac92hd73xx_dmux_nids; spec->smux_nids = stac92hd73xx_smux_nids; spec->amp_nids = stac92hd73xx_amp_nids; + spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids); spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); - spec->dinput_mux = &stac92hd73xx_dmux; + memcpy(&spec->private_dimux, &stac92hd73xx_dmux, + sizeof(stac92hd73xx_dmux)); + /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; @@ -4061,11 +4070,18 @@ again: switch (spec->board_config) { case STAC_DELL_M6: spec->init = dell_eq_core_init; + spec->num_smuxes = 0; + spec->multiout.hp_nid = + spec->multiout.dac_nids[spec->multiout.num_dacs - 1]; + spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; + spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; + spec->num_amps = 1; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; + spec->private_dimux.num_items = 1; break; case 0x10280271: /* Digital Mics */ case 0x10280272: @@ -4075,18 +4091,22 @@ again: case 0x10280255: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; case 0x10280256: /* Both */ case 0x10280057: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; } break; default: spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); } + spec->dinput_mux = &spec->private_dimux; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids;

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