January 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: codecs: wm9090: Fix build warnings due to unused variables
by Fabio Estevam 13 Jan '12

13 Jan '12

Fix the following build warnings: sound/soc/codecs/wm9090.c: In function ‘wm9090_set_bias_level’: sound/soc/codecs/wm9090.c:496: warning: unused variable ‘ret’ sound/soc/codecs/wm9090.c:496: warning: unused variable ‘i’ sound/soc/codecs/wm9090.c:495: warning: unused variable ‘reg_cache’ sound/soc/codecs/wm9090.c: In function ‘wm9090_i2c_remove’: sound/soc/codecs/wm9090.c:658: warning: unused variable ‘wm9090’ Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm9090.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 41ebe0d..acac7d6 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -492,9 +492,6 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) static int wm9090_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 *reg_cache = codec->reg_cache; - int i, ret; - switch (level) { case SND_SOC_BIAS_ON: break; @@ -655,8 +652,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { - struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&i2c->dev); return 0; -- 1.7.1

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Re: [alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

On Thu, 2012-01-12 at 16:12 -0600, Andrew Eikum wrote: > On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > > I have an application that works using 512 sample packets of 22050Hz 16 > > bit mono audio. The 'receiver' takes many audio streams from a network > > via UDP, at the moment it pipes them into pulse. > > > > Can alsa buffer audio. At the moment every time I and set an audio > > buffer size I get a negative response from > > snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the > > units alsa uses ... > > > > You don't want to over-specify your requirements. You require a buffer > size of "at least" 3 * 512 frames. So use set_buffer_size_min(). I cant find any reference to "set_buffer_size_min" in the ALSA API documentation I have or the link you provided ? snd_pcm_hw_params_set_buffer_time_near() is the closest and that seems to take an argument in useconds. ? http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > Otherwise ALSA will try to set exactly that buffer size, which can > fail. > > Check the function signatures for units. Notice that > set_buffer_size*() all take snd_pcm_uframes_t, that is, the number of > frames you want to store. In ALSA terms, a "frame" is a set of a > single sample for every channel. Since you have mono audio, a frame > and a sample are actually the same unit (for 16 bit stereo audio, > 1 frame = 2 samples = 32 bits). > > > What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x > > 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K > > Byte, 512 samples per buffer) in for playback. The packets are arriving > > at roughly the correct rate, I just need a buffer to iron out any > > jitter in network transmit, do I have to do this myself ? > > > > Can somebody help by telling me which numbers I push into which places > > to make it work ? > > > > At the moment I get i keep getting a broken pipe, if I underrun how can > > I make it just wait for me ? > > > > If a packet arrives very late (and one will, eventually), you will > underrun. That's unavoidable. You can check for SND_PCM_STATE_XRUN > from snd_pcm_state(). It's undocumented, but you need to call > snd_pcm_avail_update() first to get an accurate reading from > snd_pcm_state(). When an underrun occurs, recover with > snd_pcm_recover() and then start writing data again. Ok, thanks. > If a packet arrives early, you'll need to check that the ALSA buffer > isn't full (see snd_pcm_avail_update()), and store it within your > application to write later if it is full. > > There's some mostly-accurate information here: > http://0pointer.de/blog/projects/guide-to-sound-apis.html > As with all ALSA documentation, it is confusing and often incorrect, > but it's probably the most helpful document I've found. Many thanks for the help, Jon

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[alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

> If a packet arrives very late (and one will, eventually), you will > underrun. That's unavoidable. You can check for SND_PCM_STATE_XRUN > from snd_pcm_state(). It's undocumented, but you need to call > snd_pcm_avail_update() first to get an accurate reading from > snd_pcm_state(). When an underrun occurs, recover with > snd_pcm_recover() and then start writing data again. > Ok, this my first attempt at recovery, still haven't solved buffering problem so it underruns for each and every write. Does this look roughly correct ? Thanks, Jon #ifdef ALSA // snd_pcm_writen - writes non interleaved (mono) samples // bufs an explicit void* pointer type set to point at real sample buffer "audio" // broken pipe indicates buffer underrun numrecs=sizeof(audio)/2; // 1k buffer, 512 samples //printf("%d \n", snd_pcm_wait(playback_handle[activeslot],1)); err = snd_pcm_writen (playback_handle[activeslot], &bufs, numrecs); if (err<numrecs) { fflush(stdout); if (err>0) { fprintf (stderr, "snd_pcm_writen: wrote only %d\n",err); fflush(stderr); } else { if (err == EPIPE) { errb=snd_pcm_recover(playback_handle[activeslot],err,0); // error passed is resullt of writen if (errb<0) { fprintf(stderr,"Failed to recover from underrun\n"); fflush(stderr); exit(1); } } else { fprintf (stderr, "snd_pcm_writen: write to audio interface failed (%s) (%d)\n", snd_strerror (err),err); fflush(stdout); fflush(stderr); exit (1); } } } #endif

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[alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

On Thu, 2012-01-12 at 21:50 +0000, Fons Adriaensen wrote: > On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > > > What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x > > 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K > > Byte, 512 samples per buffer) in for playback. The packets are arriving > > at roughly the correct rate, I just need a buffer to iron out any > > jitter in network transmit, do I have to do this myself ? > > Unless data is arriving on average at *exactly* the sample > rate of the ALSA device your buffering will sooner or later > underrun or overrun. ALSA itself can't handle this use case, > it requires adaptive resampling and it's not easy. I'm assuming alsa is holding 3k bytes for me and im writing 1k byte chunks, so if the audio has not been refreshed by 3 chunks (70ms) then I should throw it away as its no longer 'real time' At the moment i would happy just to throw away packets that would cause an overrun. I just wish i could convince alsa to buffer for me .... ho hum ... with pulse I just did this to set up 'some sort of buffer length' with 'some sort of pre buffering before play' interval. Its a bit hacky at the moment but seems to work. pa_buffer_attr paattr; int buflen=sizeof(audio); paattr.tlength = buflen * 4; // Playback only, target length of the buffer (was *2) paattr.minreq = buflen; paattr.fragsize = buflen; paattr.prebuf = buflen * 6; // How much data pulse holds before it starts playing (was *4) paattr.maxlength = buflen * 8; // playback pipeline N samples deep (was *4) Im looking for alsa equivalent. Thanks, Jon

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[alsa-devel] Trying to understand alsa
by Jonathan Andrews 12 Jan '12

12 Jan '12

Hi people, i'm trying to understand alsa.... I have an application that works using 512 sample packets of 22050Hz 16 bit mono audio. The 'receiver' takes many audio streams from a network via UDP, at the moment it pipes them into pulse. Can alsa buffer audio. At the moment every time I and set an audio buffer size I get a negative response from snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the units alsa uses ... What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K Byte, 512 samples per buffer) in for playback. The packets are arriving at roughly the correct rate, I just need a buffer to iron out any jitter in network transmit, do I have to do this myself ? Can somebody help by telling me which numbers I push into which places to make it work ? At the moment I get i keep getting a broken pipe, if I underrun how can I make it just wait for me ? I only code in plain old C at the moment. Many thanks for any help/advice anyone can offer. Jon

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[alsa-devel] [PATCH RESEND] ASoC: omap: Add .owner to struct snd_soc_card
by Axel Lin 12 Jan '12

12 Jan '12

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/am3517evm.c | 1 + sound/soc/omap/ams-delta.c | 1 + sound/soc/omap/igep0020.c | 1 + sound/soc/omap/n810.c | 1 + sound/soc/omap/omap3evm.c | 1 + sound/soc/omap/omap3pandora.c | 1 + sound/soc/omap/omap4-hdmi-card.c | 1 + sound/soc/omap/osk5912.c | 1 + sound/soc/omap/overo.c | 1 + sound/soc/omap/rx51.c | 1 + sound/soc/omap/sdp3430.c | 1 + sound/soc/omap/sdp4430.c | 1 + sound/soc/omap/zoom2.c | 1 + 13 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index c1cd4a0..add4866 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,6 +107,7 @@ static struct snd_soc_dai_link am3517evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", + .owner = THIS_MODULE, .dai_link = &am3517evm_dai, .num_links = 1, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a04a433..3e523a7 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -597,6 +597,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { /* Audio card driver */ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", + .owner = THIS_MODULE, .dai_link = &ams_delta_dai_link, .num_links = 1, .set_bias_level = ams_delta_set_bias_level, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 591fbf8..ccae58a 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link igep2_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_igep2 = { .name = "igep2", + .owner = THIS_MODULE, .dai_link = &igep2_dai, .num_links = 1, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fc6209b..597be41 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -289,6 +289,7 @@ static struct snd_soc_dai_link n810_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_n810 = { .name = "N810", + .owner = THIS_MODULE, .dai_link = &n810_dai, .num_links = 1, diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 6857895..071fcb0 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -70,6 +70,7 @@ static struct snd_soc_dai_link omap3evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3evm = { .name = "omap3evm", + .owner = THIS_MODULE, .dai_link = &omap3evm_dai, .num_links = 1, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 7605c37..07794bd 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -233,6 +233,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_omap3pandora = { .name = "omap3pandora", + .owner = THIS_MODULE, .dai_link = omap3pandora_dai, .num_links = ARRAY_SIZE(omap3pandora_dai), }; diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c index 52d471c..28d689b 100644 --- a/sound/soc/omap/omap4-hdmi-card.c +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -74,6 +74,7 @@ static struct snd_soc_dai_link omap4_hdmi_dai = { static struct snd_soc_card snd_soc_omap4_hdmi = { .name = "OMAP4HDMI", + .owner = THIS_MODULE, .dai_link = &omap4_hdmi_dai, .num_links = 1, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 351ec9d..d859b59 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link osk_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", + .owner = THIS_MODULE, .dai_link = &osk_dai, .num_links = 1, diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index c3550ae..2ee889c 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link overo_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_overo = { .name = "overo", + .owner = THIS_MODULE, .dai_link = &overo_dai, .num_links = 1, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index ad16db5..fada6ef 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -383,6 +383,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { /* Audio card */ static struct snd_soc_card rx51_sound_card = { .name = "RX-51", + .owner = THIS_MODULE, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e8fbf8e..2c85066 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -213,6 +213,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", + .owner = THIS_MODULE, .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 2735fa0..175ba9a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -226,6 +226,7 @@ static struct snd_soc_dai_link sdp4430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", + .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 7641a7f..981616d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -157,6 +157,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", + .owner = THIS_MODULE, .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), -- 1.7.5.4

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[alsa-devel] Dell N5110 HDA External Speaker is mute
by Cefn Hoile 12 Jan '12

12 Jan '12

I'd like to report an issue experienced with the latest Oneiric, Dell N5110, and IDT STAC92HD87 which has been explored through quite a few steps already, and seems to have bottomed out to a set of pins/controls/flags which are not being properly managed in order to toggle speaker activation when a headphone is inserted. So far, I haven't heard sound from my speakers at all, although within Alsamixer, the Speaker volume is visibly toggled from 100% to MM (mute) on headphone insertion. I've tried every model against snd-hda-intel, many different other options, and started to explore the HDA Analyzer, which includes controls named like this... Node 0x13 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Device: name="STAC92xx Analog", type="Audio", device=0 Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=63 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Amp-Out vals: [0x7f 0x7f] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples ...but toggling these checkboxes within the HDA Analyzer gui doesn't trigger any sound from the speakers either. It could be a rich combination of the very many checkboxes and sliders which are revealed by the Analyzer, but I don't really know where to start. The bug information downstream is here... https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/914290 I'd really welcome suggestions for what else to try and happy to invest time exploring options to try and make ALSA better still. Cefn http://cefn.com

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Re: [alsa-devel] [PATCH 2/4] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 12 Jan '12

12 Jan '12

Sorry, somehow my message got corrupted in my last post. I will search the problem later and reply again. Thanks, Torsten

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Re: [alsa-devel] [PATCH 2/4] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 12 Jan '12

12 Jan '12

On Thu, 12 Jan 2012 07:42:19 +0100Takashi Iwai <tiwai(a)suse.de> wrote: > At Thu, 12 Jan 2012 00:15:56 +0100,> Torsten Schenk wrote:> > > > /*> > - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$> > - * this is done because the linear values cause rapid degredation> > - * of volume in the uppermost region.> > - */> > -static const u8 log_volume_table[128] = {> > - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34,> > - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43,> > - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e,> > - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56,> > - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c,> > - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62,> > - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68,> > - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c,> > - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71,> > - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75,> > - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79,> > - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c,> > - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f };> > So, with the new implementation, you don't need the linear -> dB> conversion?I decided to change the volume slider behaviour because of theadditional information I got on the DAC. On the other hand it seemsreasonable to do that in a separate patch. I could do so in my nextsubmit. > > > .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,> > - .info = usb6fire_control_master_vol_info,> > - .get = usb6fire_control_master_vol_get,> > - .put = usb6fire_control_master_vol_put> > + .info = usb6fire_control_output_vol_info,> > + .get = usb6fire_control_output_vol_get,> > + .put = usb6fire_control_output_vol_put,> > + .private_value = 1> > },> > {> > .iface = SNDRV_CTL_ELEM_IFACE_MIXER,> > + .name = "Output 5/6 Playback Volume",> > + .index = 0,> > + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,> > + .info = usb6fire_control_output_vol_info,> > + .get = usb6fire_control_output_vol_get,> > + .put = usb6fire_control_output_vol_put,> > + .private_value = 2> > + }> > +};> > Add the terminator in this patch. I know you fixed in the last patch> but, In general, try not to introduce a bug in each commit. Otherwise> it'll make bisection difficult.That's right, I've overseen this problem in the first place andstumbled upon it after patch 2 and 3 have already been sent. Thanks,Torsten

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[alsa-devel] overrun in case of record
by Rajeev kumar 12 Jan '12

12 Jan '12

Hello Mark I am testing my audio driver on spear platform.During recording I am getting two issues. 1. Overrun issue 2. pcm_read error I am using arecord for capturing data. #arecord -f dat > t overrun!!! (at least 32.949 ms long) Overrun issue: The issue is only reproducible if we try to record for more than 40 sec.The strange part is that Instead of overrun condition generated, it will record complete data for 10sec.(This is verified with the number of bytes recorded). So could you please let me know whether this is an error condition or just a warning. pcm_read error: For this I have seen a bug is locked in the bugzila(*Bug 472469* <show_bug.cgi?id=472469> -arecord: input/output error ). and the status is closed saying insufficient data. Could you please let me know what is the meaning of this bug? Is it like, data is not available in the buffer and pcm_read from the user space fails? But why this situation will arise? Best Regards Rajeev

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