July 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Problem with MX27 ASoC audio
by Randolph Chung 11 Jul '13

11 Jul '13

I have a target that uses iMX27(L) + tlv320aic3x. The kernel tree is HEAD of linux-arm from about a week ago. Simple playback and recording (using aplay and arecord) work fine. However, I've run into a strange problem when both playback and capture streams are opened at the same time. Consider the attached test program. The capture device is setup with a period time of 10ms, and the program reads samples in a loop. The expected result is that I get a sample every 10ms. This works fine if the playback device is *not* opened (more specifically, if the hw_params are not set). However if the playback device is opened (#if 1 code) then I get samples every 20ms, which seems very wrong. I have tried using both DMA and FIQ ssi-pcm interfaces and both give the same strange result. With the FIQ driver, if I instrument the timer callback, the timer triggers every 10ms but there is only enough data every 20ms. The codec is being driven with a 13MHz MCLK (output from MX27) and the BCLK and WCLK come from the codec. They seem to have the correct rates. Any ideas? Can anybody else with MX27 hardware verify if my test program also fails on their target? thanks, randolph /* Test program */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <assert.h> #include <sys/time.h> int main(int argc, char **argv) { int err = -1; char buf[4096]; unsigned long period_size; snd_pcm_t *in_handle = NULL, *out_handle; snd_pcm_hw_params_t *hwparams; if (snd_pcm_open(&in_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) < 0) { printf("Cannot open capture device\n"); return -1; } snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(in_handle, hwparams); snd_pcm_hw_params_set_access(in_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(in_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(in_handle, hwparams, 1); snd_pcm_hw_params_set_rate(in_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(in_handle, hwparams, 10000, 0); snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0); snd_pcm_hw_params(in_handle, hwparams); snd_pcm_hw_params_free(hwparams); if (snd_pcm_open(&out_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) { printf("Cannot open playback device\n"); return -1; } #if 1 snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(out_handle, hwparams); snd_pcm_hw_params_set_access(out_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(out_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(out_handle, hwparams, 1); snd_pcm_hw_params_set_rate(out_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(out_handle, hwparams, 10000, 0); snd_pcm_hw_params(out_handle, hwparams); snd_pcm_hw_params_free(hwparams); #endif printf("period size = %u\n", period_size); printf("Config done, start loop\n"); while (1) { struct timeval tv; int ret = snd_pcm_readi(in_handle, buf, period_size); if (ret < 0) snd_pcm_prepare(in_handle); gettimeofday(&tv, NULL); printf("[%u.%06u] snd_pcm_readi() return %d\n", tv.tv_sec, tv.tv_usec, ret); } return 0; }

5 11

[alsa-devel] CA0110 analog mic capture flatline
by Ki'Sak 11 Jul '13

11 Jul '13

Good day alsa-devel, I have a "Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG" which I have been unable to capture any content from the analog mic jack. I am a Gentoo Linux user with the 3.8.13 kernel / alsa 1.0.26. This issue has been filed with the wrong devs over at ubuntu's bugtracker and opensuse's forum. Also, I found a report that analog capture was working on this card as recently as September 2009. Thank you for your time, Ki'Sak Note: I have already tried to report this issue, this is a resend without urls due to 2 weeks without a response or it reaching the mailing list.

2 3

[alsa-devel] [PATCH 1/4] ASoC: pcm3008: Remove noisy version print
by Mark Brown 10 Jul '13

10 Jul '13

From: Mark Brown <broonie(a)linaro.org> The version number has never been updated and the printk isn't based on any interaction with the device. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f2a6282..32e5a59 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,8 +28,6 @@ #include "pcm3008.h" -#define PCM3008_VERSION "0.2" - #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -64,8 +62,6 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) struct pcm3008_setup_data *setup = codec->dev->platform_data; int ret = 0; - printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- 1.8.3.2

1 3

[alsa-devel] [PATCH] ALSA: usb-audio: fix regression for fixed stream quirk
by Eldad Zack 10 Jul '13

10 Jul '13

Commit 8f898e92aea2c24c7f379ee265d178f69ebb9c07 removed the redundant reads of bInterfaceProtocol from the descriptors, but introduced a regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT, since fp->protocol is not set in setup process. As a consequence, audio streams would not get initialized, as the following logs show: [ 48.923043] setting usb interface 3:1 [ 48.923056] Creating new capture data endpoint #81 [ 48.923484] 4:3:1: cannot set freq 48000 to ep 0x81 This patch sets fp->protocol in create_fixed_stream_quirk() and resolves the regression. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b01330..1bc45e7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -129,6 +129,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, { struct audioformat *fp; struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; int stream, err; unsigned *rate_table = NULL; @@ -166,6 +167,9 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + altsd = get_iface_desc(alts); + fp->protocol = altsd->bInterfaceProtocol; + if (fp->datainterval == 0) fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) -- 1.8.1.5

2 1

[alsa-devel] [PATCH v2 0/5] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 10 Jul '13

10 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) Tested on at91sam9g35ek. patches applies on next-20130708 [I let the original signed-off from Nicolas and Uwe in place, I don't know if I should replace them by something like "originaly-signed-off-by" since the code has been changed.] [Sorry if it's too late for 3.11] Changes from V1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards Richard Genoud (4): ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: add sound configuration ARM: AT91: DTS: sam9x5ek: enable SSC arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 27 +++++ sound/soc/atmel/Kconfig | 10 ++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 206 +++++++++++++++++++++++++++++++++++ 5 files changed, 248 insertions(+) create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

5 44

[alsa-devel] [PATCH] firewire-speakers: remove not-reused member from structure
by Takashi Sakamoto 09 Jul '13

09 Jul '13

"pcm" member in struct fwspk is used to set pcm operations but is not used again. This commit remove this member and set pcm operations with snd_pcm_set_ops(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/speakers.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 2c63865..fe9e6e2 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -49,7 +49,6 @@ struct fwspk { struct snd_card *card; struct fw_unit *unit; const struct device_info *device_info; - struct snd_pcm_substream *pcm; struct mutex mutex; struct cmp_connection connection; struct amdtp_out_stream stream; @@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk) return err; pcm->private_data = fwspk; strcpy(pcm->name, fwspk->device_info->short_name); - fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - fwspk->pcm->ops = &ops; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); return 0; } -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH v2 1/5] sound: sam9x5_wm8731: machine driver for at91sam9x5 wm8731 boards
by Richard Genoud 09 Jul '13

09 Jul '13

2013/7/8 Mark Brown <broonie(a)kernel.org>: > On Mon, Jul 08, 2013 at 03:29:49PM +0200, Richard Genoud wrote: > >> + * Nicolas Ferre <nicolas.ferre(a)atmel.com> >> + * >> + * Based on sam9g20_wm8731.c by: >> + * Sedji Gaouaou <sedji.gaouaou(a)atmel.com> > > The obvious question here is of course if we can use the same driver for > both of them. I haven't got a g20 to test that, but that's the goal. For now, g20 is still non-DT, so I think it's best to have a DT-only driver like this one for the 9x5 family (9g15, 9g25, 9x25, 9g35, 9x25). When the g20 will move to DT completely, we can drop sam9g20_wm8731.c and adjust sam9x5_wm8731.c (mainly master clock and widgets it seems) By the way, maybe g45 could use that also (and SAMA5 ?) > >> + codec_dai->driver->playback.rates &= SNDRV_PCM_RATE_8000 | >> + SNDRV_PCM_RATE_32000 | >> + SNDRV_PCM_RATE_48000 | >> + SNDRV_PCM_RATE_96000; >> + codec_dai->driver->capture.rates &= SNDRV_PCM_RATE_8000 | >> + SNDRV_PCM_RATE_32000 | >> + SNDRV_PCM_RATE_48000 | >> + SNDRV_PCM_RATE_96000; > > You definitely shouldn't be fiddling with a driver's constant static > data. You want to be using snd_pcm_hw_constraint() APIs to set > additional constraints intead. Ok, I'll change that. Thanks ! -- for me, ck means con kolivas and not calvin klein... does it mean I'm a geek ?

3 3

Re: [alsa-devel] [PATCH] ASoC: ak4554: add DT support
by Lars-Peter Clausen 08 Jul '13

08 Jul '13

On 07/06/2013 08:58 PM, Sylwester Nawrocki wrote: > On 07/05/2013 07:13 PM, Stephen Warren wrote: >> On 07/05/2013 11:04 AM, Lars-Peter Clausen wrote: >>> On 07/05/2013 06:55 PM, Stephen Warren wrote: >>>> On 07/05/2013 03:37 AM, Lars-Peter Clausen wrote: >>>>>> + - compatible : "asahi-kasei,ak4554" >>>>> >>>>> The vendor prefix for Asahi Kasei is documented as "ak" in >>>>> Documentation/devicetree/bindings/vendor-prefixes.txt. But it doesn;t seem >>>>> to be used in any upstream drivers, Stephen add that prefix maybe he knows >>>>> if it is used. If it is unused you should probably send a patch updating the >>>>> Documentation. >>>> >>>> It's used at: >>>> >>>> arch/arm/boot/dts/tegra20-seaboard.dts:499: compatible = "ak,ak8975"; >>>> >>>> ... although I see that somebody has incorrectly modified >>>> ./drivers/iio/magnetometer/ak8975.c to support the "asahi-kasei," vendor >>>> prefix:-( >>> >>> It looks as if the ak prefix was never in an upstream kernel and the commit >>> date that added the dt table predates the commit date for your devicetree. >> >> The usage I quoted above is in the mainline kernel, along with the entry >> in vendor-prefixes.txt. >> >> When I fixed vendor-prefixes.txt to define a prefix for AK, and updated >> the Tegra DT to use them[1], I guess I didn't even look at the driver, >> since all I2C device matching uses the i2c_device_id table rather than >> the of_device_id, and the i2c_device_id table doesn't have a concept of >> a vendor prefix. Hence, I didn't notice the conflict. >> >> So, even though the asahi-kasei prefix is in the of_device_id table in >> the driver, I'd say the ak prefix trumps it, since it's actually in-use >> in device trees, and is registered in vendor-prefixes.txt. > > I'm going to update this [2] patch to use "ak,ak8975" compatible string > then. Still there are some audio codec drivers that use "asahi-kasei" > vendor prefix: > > Documentation/devicetree/bindings/sound/ak4104.txt: - compatible : > "asahi-kasei,ak4104" > Documentation/devicetree/bindings/sound/ak4104.txt: compatible = > "asahi-kasei,ak4104"; > Documentation/devicetree/bindings/sound/ak4642.txt: - compatible : > "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" > Documentation/devicetree/bindings/sound/ak4642.txt: compatible = > "asahi-kasei,ak4642"; > Documentation/devicetree/bindings/sound/ak5386.txt: - compatible : > "asahi-kasei,ak5386" > Documentation/devicetree/bindings/sound/ak5386.txt: compatible = > "asahi-kasei,ak5386"; > ... > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = > &soc_codec_dev_ak4642}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = > &soc_codec_dev_ak4642}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = > &soc_codec_dev_ak4648}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > It's not in any dts file though, I'm not sure if that makes any > difference. > Well strictly speaking this is all public ABI, so I guess the best is to add a ak prefixed compatible string besides the asahi-kasei one and stick to ak only fore future drivers. - Lars

2 1

Re: [alsa-devel] No sound with Sony VAIO VPCZ1 (ALC889)
by Raymond Yau 08 Jul '13

08 Jul '13

>>> >>> Actually... there should be three mic channels: >>> The built-in (internal) microphone >>> + >>> One mic jack beside the headphone jack >>> + >>> One other hidden within the headphone jack: >>> (a) the Notebook came with Noise-cancelling headsets, but they are small in-ear plugs so there is no place for noise-cancelling logic in the plugs >>> (b) I think I remember to have read somewhere that this thingy actually needs the windows driver installed to be noise-cancelling! >>> (c) the PLUG of these headphones have another channel (visible through another ring on the jack) >>> >>> This explains the 2nd mic channel :) >> >> >> Try hda-jack-sense-test to find out which node is used by the headset mic jack (TRRS) and mic jack. > > > Thanks for the hint. > > Found and installed: snd-hda-tools_0.20111207_amd64.deb > But it gave me: > > /usr/src % hda-jack-sense-test $1 #28540 me@seven pts/4 > Traceback (most recent call last): > File "/usr/bin/hda-jack-sense-test", line 44, in <module> > codec = HDACodec(cardindex, codecindex) > File "/usr/share/snd-hda-tools/hda_codec.py", line 908, in __init__ > self.fd = os.open("/dev/snd/hwC%sD%s" % (card, device), os.O_RDWR) > OSError: [Errno 13] Permission denied: '/dev/snd/hwC0D0' You need root privilege and need 0.20130207 hda-tools > > /usr/src % sudo hda-jack-sense-test $1 #28541 me@seven pts/4 > Traceback (most recent call last): > File "/usr/bin/hda-jack-sense-test", line 44, in <module> > codec = HDACodec(cardindex, codecindex) > File "/usr/share/snd-hda-tools/hda_codec.py", line 921, in __init__ > self.parse_proc() > File "/usr/share/snd-hda-tools/hda_codec.py", line 1003, in parse_proc > self.proc_codec = HDACodecProc(self.card, self.device, file) > File "/usr/share/snd-hda-tools/hda_proc.py", line 474, in __init__ > self.parse(proc_file) > File "/usr/share/snd-hda-tools/hda_proc.py", line 647, in parse > line, nid = self.decodeintw(line, 'Node ') > File "/usr/share/snd-hda-tools/hda_proc.py", line 140, in decodeintw > self.wrongfile('integer decode %s (%s)' % (repr(str), repr(prefix))) > File "/usr/share/snd-hda-tools/hda_proc.py", line 143, in wrongfile > raise ValueError, "wrong proc file format (%s)" % msg > ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node ')) > http://git.alsa-project.org/?p=alsa.git;a=commitdiff;h=7859cb050f2cbeba54ce…

1 0

[alsa-devel] [PATCH] ASoC: rt5640: add headset detection function
by bardliao＠realtek.com 08 Jul '13

08 Jul '13

From: Bard Liao <bardliao(a)realtek.com> Add headset detection function to rt5640 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ 2 files changed, 62 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..657efb7 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1911,6 +1911,66 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define rt5640_resume NULL #endif +/** + * rt5640_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +int rt5640_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int jack_type; + int sclk_src = 0; + int reg63, reg64; + + if (jack_insert) { + reg63 = snd_soc_read(codec, RT5640_PWR_ANLG1); + reg64 = snd_soc_read(codec, RT5640_PWR_ANLG2); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_write(codec, RT5640_PWR_ANLG1, 0xa814); + sclk_src = snd_soc_read(codec, RT5640_GLB_CLK) & + RT5640_SCLK_SRC_MASK; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, + 0x2 << RT5640_SCLK_SRC_SFT); + } + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LDO2, RT5640_PWR_LDO2); + snd_soc_update_bits(codec, RT5640_PWR_ANLG2, + RT5640_PWR_MB1, RT5640_PWR_MB1); + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK | RT5640_MIC1_OVTH_MASK | + RT5640_PWR_CLK25M_MASK | RT5640_PWR_MB_MASK, + RT5640_MIC1_OVCD_EN | RT5640_MIC1_OVTH_600UA | + RT5640_PWR_MB_PU | RT5640_PWR_CLK25M_PU); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0x1); + msleep(100); + if (snd_soc_read(codec, RT5640_IRQ_CTRL2) & 0x8) + jack_type = RT5640_HEADPHO_DET; + else + jack_type = RT5640_HEADSET_DET; + snd_soc_update_bits(codec, RT5640_IRQ_CTRL2, + RT5640_MB1_OC_CLR, 0); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, sclk_src); + snd_soc_write(codec, RT5640_PWR_ANLG1, reg63); + snd_soc_write(codec, RT5640_PWR_ANLG2, reg64); + } else { + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK, + RT5640_MIC1_OVCD_DIS); + + jack_type = RT5640_NO_JACK; + } + + return jack_type; +} +EXPORT_SYMBOL_GPL(rt5640_headset_detect); + #define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..c3856c8 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2089,4 +2089,6 @@ struct rt5640_priv { int dmic_en; }; +int rt5640_headset_detect(struct snd_soc_codec *codec, int jack_insert); + #endif -- 1.8.1.1.439.g50a6b54

3 3