January 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] capturing data from the microphone
by Riccardo Magliocchetti 14 Jan '10

14 Jan '10

[the same message is waiting in moderator queue, sorry if you receive it two times] Hello, i'm trying to capture audio data from the microphone, but i'm not able to read any frames. The frames are always 0 and delay is fixed to 2730. I've tried calling snd_pcm_avail_delay, snd_pcm_avail_update but does not make a difference. The soundcard is driven by snd_hda_intel. The code is more or less the same as Paul Davis' tutorial on using the alsa api. This is the code for init and the loop that tries to read the data: #define MIC_BUFSIZE 4096 static gpointer snd_pcm_read(gpointer data) { int error; snd_pcm_sframes_t frames = MIC_BUFSIZE; while (TRUE) { if ((error = snd_pcm_wait (pcm_handle, 1000)) < 0) { g_printerr("Failed to poll: %s\n", snd_strerror(error)); continue; } if ((error = snd_pcm_avail(pcm_handle)) < 0) { if (error == -EPIPE) { g_printerr("xrun! %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } else { g_printerr("alsa_pcm_avail_update error %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } } frames = error; if (frames == 0) continue; frames = frames > MIC_BUFSIZE ? MIC_BUFSIZE : frames; g_printerr ("frames: %ld\n", frames); g_static_mutex_lock(&mutex); error = snd_pcm_readi(pcm_handle, Mic_Buffer[Mic_WriteBuf], frames); g_static_mutex_unlock(&mutex); if (error < 0) error = snd_pcm_recover(pcm_handle, error, 0); if (error < 0) { LOG("snd_pcm_readi FAIL!: %s\n", snd_strerror(error)); } } return GINT_TO_POINTER(TRUE); } BOOL Mic_Init() { snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int err; if (Mic_Inited) return TRUE; // Open the default sound card in capture if ((err = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_CAPTURE, /*SND_PCM_NONBLOCK*/ 0)) < 0) { g_printerr("Failed to open device: %s\n", snd_strerror(err)); return FALSE; } // Allocate the snd_pcm_hw_params_t structure and fill it. snd_pcm_hw_params_alloca(&hwparams); if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { g_printerr("Failed to setup hw parameters: %s\n", snd_strerror(err)); return FALSE; } //Set the access if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { g_printerr("Failed to set access: %s\n", snd_strerror(err)); return FALSE; } //dir 0 == exacte (Rate = 16K exacte) if ((err = snd_pcm_hw_params_set_rate(pcm_handle, hwparams, 16000, 0)) < 0) { g_printerr("Failed to set rate: %s\n", snd_strerror(err)); return FALSE; } /* Set sample format */ if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0) { g_printerr("Failed to set format: %s\n", snd_strerror(err)); return FALSE; } // Set one channel (mono) if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1)) < 0) { g_printerr("Failed to set channels: %s\n", snd_strerror(err)); return FALSE; } //Set the params if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { g_printerr("Failed to set hw parameters: %s\n", snd_strerror(err)); return FALSE; } snd_pcm_sw_params_alloca(&swparams); if ((err = snd_pcm_sw_params_current (pcm_handle, swparams)) < 0) { g_printerr("Failed to set current sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_avail_min (pcm_handle, swparams, MIC_BUFSIZE)) < 0) { g_printerr("Failed to set minimum available count: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_start_threshold (pcm_handle, swparams, 0U)) < 0) { g_printerr("Failed to set start mode: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) { g_printerr("Failed to set sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_prepare (pcm_handle)) < 0) { g_printerr("Failed to prepare audio interface to use: %s\n", snd_strerror(err)); return FALSE; } Mic_Inited = TRUE; Mic_Reset(); mic_reader = g_thread_create(snd_pcm_read, NULL, TRUE, NULL); return TRUE; } thanks, Riccardo

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[alsa-devel] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write!
by Konrad Schlichtherle 14 Jan '10

14 Jan '10

Hello, sometimes it occure that my speakers do not play sound although the speakers should play music with VLC. Then i took a look at /var/log/messages and found this: Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. The output of utils_alsa-info.sh is there: http://www.alsa-project.org/db/?f=1767e9bc745724a67c908549d20c137375d0605e I didn't know wether this is a bug or a misconfiguration of my openSUSE 11.2 Konrad

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Re: [alsa-devel] [PATCH] OMAP: Exporting functions doing common register access
by Aggarwal, Anuj 13 Jan '10

13 Jan '10

> -----Original Message----- > From: Paul Walmsley [mailto:paul@pwsan.com] > Sent: Tuesday, November 17, 2009 2:46 AM > To: Aggarwal, Anuj > Cc: linux-omap(a)vger.kernel.org > Subject: Re: [PATCH] OMAP: Exporting functions doing common register > access > > Hello Anuj, > > On Mon, 16 Nov 2009, Anuj Aggarwal wrote: > > > These functions need to be exported so that drivers (e.g. McBSP) can > > be configured as modules. > > > > McBSP driver gets built as a module when ASoC driver for OMAP3 EVM > > is configured as module. McBSP driver uses functions like > > omap_ctrl_readl/omap_ctrl_writel, which are defined in control.c > > file but not exported. Without that, McBSP driver fails to build as > > a module. > > > > Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> > > This will prevent the McBSP driver from being usable across chips. For > example, if a future OMAP comes out with McBSP clock source switching > implemented in a different way, or if a DaVinci chip comes out with a > McBSP block but no SCM block, this either won't build or won't run. > > A better approach would be to move the OMAP2/3-specific functions from the > McBSP driver into the OMAP McBSP integration code, > arch/arm/*omap*/mcbsp.c, and to define function pointers in > struct platform_data for these functions, and to pass them from the > integration code into the device driver code. That will make these > drivers independent of chip integration details. Sorry to confuse you, I was talking about the platform specific ASoC-McBSP code for OMAP (file - sound/soc/omap/omap-mcbsp.c) platform. The McBSP driver - both platform and machine specific - doesn't use the functions exported by the patch. Including alsa-devel list too... > > > - Paul

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[alsa-devel] Newbie question: Sending untimed MIDI data to ALSA seq
by Josh Lehan 13 Jan '10

13 Jan '10

Hello. Newbie question: I'm wondering about sending untimed MIDI data to the ALSA sequencer subsystem. The application would be for a "beep"-like program, from the command line, that instead of making beeps, would play MIDI instruments to make better-sounding beeps. I have TiMidity hooked up into ALSA, as a softsynth, and it works just fine for playing MIDI data sent to it by ALSA. The "aplaymidi" command works for .MID files, and the "vkeybd" program works for simulating a keyboard and sending in untimed MIDI data. The "aseqview" program also works great for watching the notes being played, as they go by. So, my MIDI software stack appears to be in order. I don't have a hardware synthesizer hooked up at the moment (my Kawai K1 is showing its age, and a few keys are broken). I also looked at "amidi", which features a -S option to send in untimed MIDI bytes, but unfortunately, that addresses an entirely different namespace. Evidently, it only speaks to "RawMIDI" devices, of which TiMidity is not. So, I can't generate untimed MIDI data on my own. I was thinking of writing a small "midibeep" program that would do this, similar to what vkeybd does, but without requiring any user interaction. Does such a program already exist? Thanks! Josh

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[alsa-devel] Mic Boost control problem with snd-hda-intel codec AL662 rev 1
by Anisse Astier 13 Jan '10

13 Jan '10

Hi, The Mic Boost control on a laptop seems to behave incorrectly. The control has the flag "volume" instead of "cvolume", causing it to appear in both the "Playback" and "Capture" tabs of alsamixer. This wouldn't be a problem if it would just work in the Capture tab: it doesn't. When I modify it in Capture, it doesn't retain its value: it's gone after i close alsamixer. But if i modify it in "Playback" it will retain its value and the value will be mirrored by the Mic Boost in Capture (both Mic Boost seem to reflect the same control, except one is RO, the other is RW). It's a minor a annoyance, but I'd like to fix it and have "Mic Boost" appear only as a Capture control and retaining its value. Maybe I should create a new mixer configuration for this hardware in patch_realtek.c, but I don't know where to start. Bios autoprobing works for everything but just what I mentioned. My alsa-info.sh : http://www.alsa-project.org/db/?f=fc17b509e532fc366eb2d5fe5ec9e6730c96c837 Any ideas? Where should I start? Regards, Anisse

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[alsa-devel] [PATCH] ASoC: TWL4030: Replace comma with semicolon in probe function
by Peter Ujfalusi 13 Jan '10

13 Jan '10

The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a27f7b..74f0d65 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2192,7 +2192,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); -- 1.6.6

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[alsa-devel] [PATCH] ALSA: ASoC: ak4104: allow more sample rates
by Daniel Mack 13 Jan '10

13 Jan '10

The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4104.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 3a14c6f..b9ef7e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_32000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- 1.6.5.2

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[alsa-devel] Personal invitation from Kunal Narkhede
by Kunal Narkhede 13 Jan '10

13 Jan '10

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Improved MacBook (Pro) 5, 1 / 5, 2 support
by murray.alex＠gmail.com 13 Jan '10

13 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 files changed, 26 insertions(+), 2 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index f858dcd..bd65e47 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -7201,8 +7201,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7603,6 +7603,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7787,6 +7788,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9233,6 +9255,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- 1.6.3.3

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[alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 13 Jan '10

13 Jan '10

Hi, I am working on android and and in our device we use ALSA. I have enabled dmix support for andoid as by default android doesn't have this plugin enabled and soon will post it in android forum however i have run into some problems.Hoping takashi can help me out on this. Problem is when i am trying to use dmix plugin i am not getting any sound (logs shown below).Is it because i don't have mmap mode support on my device??? If yes then can't dmix work without mmap mode??? Logs shown below: Entered s3c24xx_pcm_hw_params, Entered s3c24xx_pcm_hw_params, line 330 Entered s3c24xx_pcm_mmap Entered s3c24xx_pcm_prepare Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer res_value[0] snd_pcm_update_hw_ptr_pos xun: [0]occured XRUN! Normal playback and recording is fine if i dont use dmix plugin.During my normal playback and recording i don't use mmap as i preallocate DMA memory and use them. Regards, anish

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