August 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] Fallback mechanism for pulse plugin
by Takashi Iwai 25 Sep '11

25 Sep '11

Hi, the following are experimental patches for implementing the fallback option of PCM / control pulse plugin. When the connection to PA server fails, the plugin tries to open the fallback name. For achieving this, I added the new standard definition "sysdefault", which is equal with the normal "default" PCM / control definitions. The difference is only the name, i.e. it won't be overridden by other setups. Then two new API functions for opening a fallback PCM / control, and finally a patch for pulse-plugin will follow. All changes are relatively small and easy. Let me know if you have any suggestions or a better idea. thanks, Takashi

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[alsa-devel] [PATCHv2] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 23 Sep '11

23 Sep '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk if the McBSP is already active. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: stable(a)kernel.org --- v2: - Returns -EBUSY from omap_mcbsp_dai_set_dai_sysclk if the input clock rate doesn't match with another stream's configuration and 0 otherwise. For 2.6.37 and onward. This issue is valid to those machine drivers that use the McBSP as a DAI link master and are configuring the CLKS source (like sound/soc/omap/omap3pandora.c in mainline). --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b7723..4b82290 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + if (freq == mcbsp_data->in_freq) + return 0; + else + return -EBUSY; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.4.4

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[alsa-devel] [PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 22 Sep '11

22 Sep '11

With the new/upcoming version of PulseAudio (0.99.x) there is a protocol addition that makes it possible to handle underruns better in the pulse plugin. The attached patch implements that, but it has two flaws I wouldn't mind getting some help with if possible: 1) Since this uses the new API function pa_stream_get_underflow_index, it won't compile with current stable PulseAudio versions, only the upcoming version. 2) So now there are three possibilities for handle_underrun ( 0 = never, 1 = always, 2 = the new improved one that IMO should be used). Since handle_underrun is a bool in the config, it can not be set to "2", only 0 or 1. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH 0/3] ASoC: tpa6130a2: model handling cleanup
by Peter Ujfalusi 21 Sep '11

21 Sep '11

Hello, Small cleanup for the tpa6130a2 driver for model handling: Remove the model_id from platform_data, and use the device name/device_data to distinguish between the supported models of TPA. Regards, Peter --- Peter Ujfalusi (3): ASoC: tpa6130a2: Model support cleanup OMAP: RX51: No need to configure the tpa_model ASoC: tpa6130a2: Remove model_id from platform data arch/arm/mach-omap2/board-rx51-peripherals.c | 1 - include/sound/tpa6130a2-plat.h | 6 ------ sound/soc/codecs/tpa6130a2.c | 12 +++++++++--- 3 files changed, 9 insertions(+), 10 deletions(-) -- 1.7.6.1

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[alsa-devel] [RFC v2 PATCH 1/1] ASoC: soc-core: symmetry checking for each DAIs separately
by Dong Aisheng 21 Sep '11

21 Sep '11

The orginal code does not cover the case that one DAI such as codec may be shared between other two DAIs(CPU). When do symmetry checking, altough the codec DAI requires symmetry, the two CPU DAIs may still be configured to run on different rates. We change to check each DAI's state separately instead of only checking the dai link to prevent this issue. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- Change since v1: Address the comments from Lars-Peter Clausen. * add the dai as an parameter for soc_pcm_apply_symmetry Thanks for the suggestion. --- include/sound/soc-dai.h | 3 +++ include/sound/soc.h | 2 -- sound/soc/soc-pcm.c | 40 +++++++++++++++++++++++++--------------- 3 files changed, 28 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..12d98b4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,6 +242,9 @@ struct snd_soc_dai { void *playback_dma_data; void *capture_dma_data; + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + /* parent platform/codec */ union { struct snd_soc_platform *platform; diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..5449139 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -849,8 +849,6 @@ struct snd_soc_pcm_runtime { unsigned int complete:1; unsigned int dev_registered:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..8eb0f07 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,15 +27,13 @@ #include <sound/soc.h> #include <sound/initval.h> -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, + struct snd_soc_dai *soc_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && + if (!soc_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; @@ -43,19 +41,19 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { - dev_warn(&rtd->dev, + if (!soc_dai->rate) { + dev_warn(soc_dai->dev, "Not enforcing symmetric_rates due to race\n"); return 0; } - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + dev_dbg(soc_dai->dev, "Symmetry forces %dHz rate\n", soc_dai->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); + soc_dai->rate, soc_dai->rate); if (ret < 0) { - dev_err(&rtd->dev, + dev_err(soc_dai->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -185,8 +183,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); + if (cpu_dai->active) { + ret = soc_pcm_apply_symmetry(substream, cpu_dai); + if (ret != 0) + goto config_err; + } + + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -288,8 +292,12 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -447,7 +455,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.0.4

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[alsa-devel] Noise on snd-hda-intel
by Sune Mølgaard 18 Sep '11

18 Sep '11

Plugging almost any headset into the front headset jacks on my machine results in noise in the headphones as well as in Skype conversations and/or recording. Said noise is, however not head in the regular speakers (except when playing back recordings, of course. lspci -vvnn gives 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: ASUSTeK Computer Inc. Device [1043:82ea] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 256 bytes Interrupt: pin A routed to IRQ 70 Region 0: Memory at f7ff8000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel and aplay -l gives **** List of PLAYBACK Hardware Devices **** card 0: Device [USB Multimedia Audio Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 I have tried with no modules options as well as options snd_hda_intel model=6stack and options snd_hda_intel model=6stack-dig Best regards, Sune Mølgaard -- You buy Windows, and you put it on your machine-you can't do a whole lot with it. - Bob Muglia, Microsoft Senior Vice President

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[alsa-devel] Dell XPS 13 (Codec: IDT 92HD73C1X5) and 3 stereo jack output
by Ruben Van Boxem 14 Sep '11

14 Sep '11

Hi, I have been trying to get my Dell XPS 13 to output 5.1 sound through the stereo jacks (because the NVIDIA HDMI only does 2.1). Attached is alsa-info output. I can't seem to be able to set the Line/Mic In to output. I have tried several things, going from fiddling with Kmix to fiddling with the modprobe "model=" parameter, and even looking through the hda_analyzer.py output. This last one lifted a tiny corner of the solution, but I don't think a vanilla Alsa install will get me through this. In Windows, I can use the three jacks to output 2 channels each, so I am sure the hardware is capable of it. In HDA analyzer I can see that there are several "Line Out" entries, but since they all have the node cap "Unsolicited Capabilities", I'm pretty sure the problem is alsa not knowing about the, well, capabilities. I searched the web and documentation for anything sane to try, but nothing worked. The default modprobe "model=" is the best (I can get 4.0 output from the jacks, and everything works, except the third jack). If you need more input or information, please ask, I'm very willing to get this solved! Thanks, Ruben

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[alsa-devel] [PATCH 1/1] HDA - Add Headphone Playback Volume control for ad1988/ad1989
by Raymond Yau 12 Sep '11

12 Sep '11

HDA - Add Headphone Playback Volume control for ad1988/ad1989 - use DAC0 instead of DAC1 for Port-A Headphone - assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp"

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Re: [alsa-devel] [PATCH v2] sound: usb: mixer: increase control transfer timeout
by Jassi Brar 12 Sep '11

12 Sep '11

On Mon, Aug 29, 2011 at 6:01 PM, Felipe Balbi <balbi(a)ti.com> wrote: > 100ms might not be enough for slower devices. Increasing > timeout to 1000ms doesn't seem like overkill. > > Signed-off-by: Felipe Balbi <balbi(a)ti.com> > --- > > Changes since v1: > . Increase all 100ms timeout to 1000ms > > sound/usb/mixer.c | 4 ++-- > sound/usb/mixer_quirks.c | 10 +++++----- > 2 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > index cdd19d7..1c9d443 100644 > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -296,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v > if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), > - buf, val_len, 100) >= val_len) { > + buf, val_len, 1000) >= val_len) { > *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); > snd_usb_autosuspend(cval->mixer->chip); > return 0; > @@ -445,7 +445,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, > usb_sndctrlpipe(chip->dev, 0), request, > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), > - buf, val_len, 100) >= 0) { > + buf, val_len, 1000) >= 0) { > snd_usb_autosuspend(chip); > return 0; > } > diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c > index 3d0f487..7c41399 100644 > --- a/sound/usb/mixer_quirks.c > +++ b/sound/usb/mixer_quirks.c > @@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - !value, 0, NULL, 0, 100); > + !value, 0, NULL, 0, 1000); > /* USB X-Fi S51 Pro */ > if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - !value, 0, NULL, 0, 100); > + !value, 0, NULL, 0, 1000); > else > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - value, index + 2, NULL, 0, 100); > + value, index + 2, NULL, 0, 1000); > if (err < 0) > return err; > mixer->audigy2nx_leds[index] = value; > @@ -299,7 +299,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, > usb_rcvctrlpipe(mixer->chip->dev, 0), > UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | > USB_RECIP_INTERFACE, 0, > - jacks[i].unitid << 8, buf, 3, 100); > + jacks[i].unitid << 8, buf, 3, 1000); > if (err == 3 && (buf[0] == 3 || buf[0] == 6)) > snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); > else > @@ -332,7 +332,7 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - 50, 0, &new_status, 1, 100); > + 50, 0, &new_status, 1, 1000); How about also converting to use a local define for timeout value ?

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[alsa-devel] [PATCH 1/4] sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree
by Julia Lawall 12 Sep '11

12 Sep '11

From: Julia Lawall <julia(a)diku.dk> Adjust the goto to jump to the error handling code that includes kfree. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index a33fc51..8f16cd3 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error; + goto error_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K);

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