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[alsa-devel] 2.6.26.[6|7]-rt11, alsa rawmidi, seq hang
by Fernando Lopez-Lezcano 11 Dec '08

11 Dec '08

(starting a new thread with the hope of being more precise after a lot of testing and rebooting) I'm seeing a realtime patch related hard hang in the kernel alsa subsystem (MIDI input/output). In a nutshell: - alsa rawmidi works (ie: "rawmidi -v -i hw:0" outputs a stream of messages when pointed to a midi capable card that has an external keyboard connected). - the alsa sequencer interface works (ie: aplaymidi connected to aseqdump transfers data just fine). - BOTH combined do NOT work (ie: use aconnect to connect the port that corresponds to the external midi interface to aseqdump: aseqdump hangs forever after transferring the first message and the only way out is a reboot). Just tested this on 2.6.26.7 + rt11, Gina3G soundcard with an external MIDI keyboard sending messages. This is not specific to the Gina card, I have tested also USB midi interfaces and an RME hdsp card with a Multiface interface. 2.6.24.7 + rt21 works fine on the same hardware combinations. The standard Fedora 9 2.6.26.x kernel works fine as well. I'm attaching the kernel configuration for the kernel I'm currently testing plus a dmesg dump of the whole boot process (rt kernel hackers can find there some additional sleeping in wrong context warning messages there), including the output of a "echo t >/proc/sysrq-trigger" that should show where aseqdump currently hangs (or so I think). Hopefully this will narrow the area affected enough so that a solution can be found or more debugging info can be requested. Currently 2.6.26.x-rtx is unusable for musical purposes (at least when you have to interact with external midi controllers). Thanks for any help... -- Fernando

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Re: [alsa-devel] More notes on HD-audio driver
by kls.schlz＠googlemail.com 11 Dec '08

11 Dec '08

If you prepare a Wiki for the HD-Audio, don't forget to merge somehow ALSA-Configuration.txt. IMO this goes together. On Dec 11, 2008 7:24am, Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 10 Dec 2008 14:45:52 -0700, > > stan wrote: > > > > > > Takashi Iwai wrote: > > > > Hi, > > > > > > > > per popular demand, I wrote a brief document more about HD-audio driver. > > > > It's now found in sound git tree as Documentation/sound/alsa/HD-Audio.txt. > > > > The git tree is: > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > > > > > You can get the text from web: > > > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=blob_plain;… > > > > > > > > A PDF file is found at: > > > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf > > > > > > Thank you for this. I found it informative and helpful. > > > > > > Maybe it could be placed on the wiki or sent to the > > > alsa-user list once a month as "how to troubleshoot hd audio"? > > > > Putting Wiki is a good idea. I'm not sure whether it'd be better to > > put the text into Wiki or to put a link to the above URL, though... > > > > > > Takashi > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] Idea about the laptop speakers
by Alexander E. Patrakov 11 Dec '08

11 Dec '08

Hello developers, some time ago, people complained that the sound came out from both the laptop speakers and headphones. While in many cases this is not desirable, I think this effect (when configurable via opening a different PCM) can be useful in the following situation. Suppose that you have a laptop as your main working machine. The headset is always connected, but you are not always near the laptop (e.g., think better if you walk in the room). Then, you configure (by telling it to open different devices) your SIP softphone to play voice through headphones only, but play ringtones through both headphones and speakers. Thus, you can walk in the room without being afraid to miss a call. Is the above proposal implementable? -- Alexander E. Patrakov

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[alsa-devel] [PATCH 0/8] ASoC updates
by Mark Brown 10 Dec '08

10 Dec '08

The following changes since commit f73f2a6a23e34de9cca9672f727694e5af00e6c7: Takashi Iwai (1): ALSA: ASoC - Fix symbol conflicts in omac-mcbsp.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Mark Brown (5): ASoC: Work around warnings from some build environments ASoC: Add codec registration API ASoC: Convert WM8900 to do more work at I2C probe time ASoC: Convert WM8903 driver to register at I2C probe time ASoC: Stop WM8903 SYSCLK when suspending Peter Ujfalusi (3): ASoC: TWL4030: Add missing Carkit output ASoC: TWL4030: Small cleanup ASoC: TWL4030: Change the name for the DACs include/sound/soc.h | 5 + sound/soc/codecs/twl4030.c | 27 +++--- sound/soc/codecs/wm8900.c | 159 +++++++++++++++-------------- sound/soc/codecs/wm8900.h | 7 -- sound/soc/codecs/wm8903.c | 241 +++++++++++++++++++------------------------- sound/soc/codecs/wm8903.h | 5 - sound/soc/soc-core.c | 43 ++++++++ 7 files changed, 247 insertions(+), 240 deletions(-)

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[alsa-devel] [PATCH] ALSA: Add support for mechanical jack insertion
by Mark Brown 10 Dec '08

10 Dec '08

Some systems support both mechanical and electrical jack detection, allowing them to report that a jack is physically present but does not have any functioning connections. Add a new jack type for these, allowing user space to report faulty connections. Thanks to Guillem Jover for the suggestion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/input.h | 1 + include/sound/jack.h | 1 + sound/core/jack.c | 6 ++++++ 3 files changed, 8 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index 1a958cd..9a6355f 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -660,6 +660,7 @@ struct input_absinfo { #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ +#define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 7cb25f4..2e0315c 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -36,6 +36,7 @@ enum snd_jack_types { SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, + SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index 4978db1..7ca905c 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -108,6 +108,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); + if (type & SND_JACK_MECHANICAL) + input_set_capability(jack->input_dev, EV_SW, + SW_JACK_PHYSICAL_INSERT); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -162,6 +165,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); + if (jack->type & SND_JACK_MECHANICAL) + input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, + status & SND_JACK_MECHANICAL); input_sync(jack->input_dev); } -- 1.5.6.5

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[alsa-devel] Improvements on aloop driver
by Takashi Iwai 10 Dec '08

10 Dec '08

Hi, as some people have asked about aloop bugs and developments, I started working on it a bit. The fact is that aloop is buggy per design in two points: - Sharing the same buffer for both playback and capture doesn't work. The playback app may change the "available" buffer area at any time. It can update the last played period very quickly. OTOH, the capture needs to read the last period. So, it obviously conflicts. - We needs to synchronize both playback and capture streams. In the current implementation, both streams are updated individually. For example, the capture can start in the middle of the playback period. This results in the partial available samples. So, I wrote the code and got almost another shape of the driver. The result is found in my alsa-driver-build git tree git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git and the snapshot tarball ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Let me know whether this works for you. Once after confirmed to work well enough, we should push this to the upstream -- or abandon if the driver is conceptually wrong... thanks, Takashi

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[alsa-devel] [PATCH 0/3] ASoC: TWL4030: Fixes and small cleanup
by Peter Ujfalusi 10 Dec '08

10 Dec '08

Hello, The CarkitL/R DAPM output somehow got lost, the first patch adds it. Really small cleanup for line wrap in the mux switch texts. DAC names changed in order to avoid confusion with the mux switch names. --- Peter Ujfalusi (3): ASoC: TWL4030: Add missing Carkit output ASoC: TWL4030: Small cleanup ASoC: TWL4030: Change the name for the DACs sound/soc/codecs/twl4030.c | 27 +++++++++++++-------------- 1 files changed, 13 insertions(+), 14 deletions(-)

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[alsa-devel] Ask a question about WM8976
by peter meng 10 Dec '08

10 Dec '08

Hi, I'm a newbie for alsa and using wm8976 first time . I found the driver of wm8976 comments the contents of wm8976_dapm_widgets . 1. Is this mean all widget in wm8976 does not have power management function ? 2. If wm8976 does not add widget then the process of 'set up audio path map ' will fails , is this correct ? ------------------------------------------------------------------- // TODO Widgets static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = { #if 0 //SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0), //SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0), SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0, &wm8976_speaker_mixer_controls[0], ARRAY_SIZE(wm8976_speaker_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0, &wm8976_mono_mixer_controls[0], ARRAY_SIZE(wm8976_mono_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0), SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, &wm8976_aux_boost_controls, 1), SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, &wm8976_mic_boost_controls, 1), SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, &wm8976_capture_boost_controls), SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTP"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), #endif }; static int wm8976_add_widgets(struct snd_soc_codec *codec) { int i; for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) { snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]); } /* set up audio path map */ for(i = 0; audio_map[i][0] != NULL; i++) { snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); } snd_soc_dapm_new_widgets(codec); return 0; } ----------------------------------------------- Thanks advanced . Best Regards Peter Meng

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Re: [alsa-devel] Warnings in wm8903.c (Re: Merged upstream to topic/asoc branch)
by Mark Brown 10 Dec '08

10 Dec '08

On Wed, Dec 10, 2008 at 08:26:56AM +0100, Takashi Iwai wrote: > Hmm, I got the warnings again on a x86-64 with gcc-4.3.2 randconfig > test build again. > The config is attached below. OK, I've managed to reproduce this with that .config. I'll push a patch later today which adds return statements to silence the warning, meaning that any genuine issues won't be suppressed.

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[alsa-devel] [PATCH 00/18] ASoC updates
by Mark Brown 10 Dec '08

10 Dec '08

The following changes since commit 28a1d869560a49d960ba2a3b450ec965712e5560: Daniel Mack (1): ASoC: tlv320aic3x: control additions and cleanups are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (9): ASoC: Add card registration API ASoC: Add DAI registration API ASoC: Register platform DAIs ASoC: Add platform registration API ASoC: Register platform drivers ASoC: Register non-AC97 codec DAIs ASoC: Initial framework for dynamic card instantiation ASoC: Wait for non-AC97 codec DAIs before instantiating ASoC: Convert WM8900 to allow registration by machine code Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down include/sound/soc-dai.h | 8 + include/sound/soc.h | 10 ++ sound/soc/atmel/atmel-pcm.c | 12 ++ sound/soc/atmel/atmel_ssc_dai.c | 11 ++ sound/soc/au1x/dbdma2.c | 3 +- sound/soc/au1x/psc-ac97.c | 3 +- sound/soc/au1x/psc-i2s.c | 3 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 ++ sound/soc/blackfin/bf5xx-ac97.c | 12 ++ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 ++ sound/soc/blackfin/bf5xx-i2s.c | 12 ++ sound/soc/codecs/ad73311.c | 12 ++ sound/soc/codecs/ak4535.c | 12 ++ sound/soc/codecs/cs4270.c | 12 ++ sound/soc/codecs/pcm3008.c | 12 ++ sound/soc/codecs/ssm2602.c | 12 ++ sound/soc/codecs/tlv320aic23.c | 12 ++ sound/soc/codecs/tlv320aic26.c | 15 ++- sound/soc/codecs/tlv320aic3x.c | 12 ++ sound/soc/codecs/twl4030.c | 277 +++++++++++++++++++++++++++++++-- sound/soc/codecs/uda134x.c | 12 ++ sound/soc/codecs/uda1380.c | 12 ++ sound/soc/codecs/wm8510.c | 12 ++ sound/soc/codecs/wm8580.c | 12 ++ sound/soc/codecs/wm8728.c | 12 ++ sound/soc/codecs/wm8731.c | 12 ++ sound/soc/codecs/wm8750.c | 12 ++ sound/soc/codecs/wm8753.c | 12 ++ sound/soc/codecs/wm8900.c | 105 ++++--------- sound/soc/codecs/wm8903.c | 12 ++ sound/soc/codecs/wm8971.c | 12 ++ sound/soc/codecs/wm8990.c | 12 ++ sound/soc/davinci/davinci-i2s.c | 12 ++ sound/soc/davinci/davinci-pcm.c | 12 ++ sound/soc/fsl/fsl_dma.c | 12 ++ sound/soc/fsl/fsl_ssi.c | 10 ++ sound/soc/fsl/mpc5200_psc_i2s.c | 4 + sound/soc/omap/omap-mcbsp.c | 13 ++ sound/soc/omap/omap-pcm.c | 12 ++ sound/soc/pxa/pxa-ssp.c | 12 ++ sound/soc/pxa/pxa2xx-ac97.c | 12 ++ sound/soc/pxa/pxa2xx-i2s.c | 13 ++- sound/soc/pxa/pxa2xx-pcm.c | 12 ++ sound/soc/s3c24xx/s3c2412-i2s.c | 13 ++ sound/soc/s3c24xx/s3c2443-ac97.c | 13 ++ sound/soc/s3c24xx/s3c24xx-i2s.c | 12 ++ sound/soc/s3c24xx/s3c24xx-pcm.c | 12 ++ sound/soc/sh/dma-sh7760.c | 12 ++ sound/soc/sh/hac.c | 12 ++ sound/soc/sh/ssi.c | 12 ++ sound/soc/soc-core.c | 291 +++++++++++++++++++++++++++++++++-- 51 files changed, 1102 insertions(+), 110 deletions(-)

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