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

[alsa-devel] [PATCH] asoc/multi-component: fsl: add support for disabled SSI nodes
by Timur Tabi 05 Aug '10

05 Aug '10

Add support for adding "status = disabled" to an SSI node to incidate that it is not wired on the board. This replaces the not-so-intuitive previous method of omitting a codec-handle property. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, would you please ACK the device-tree portion of this patch? I want it to go through the ALSA tree. arch/powerpc/boot/dts/mpc8610_hpcd.dts | 1 + sound/soc/fsl/fsl_ssi.c | 13 ++++++++++--- 2 files changed, 11 insertions(+), 3 deletions(-) diff --git a/arch/powerpc/boot/dts/mpc8610_hpcd.dts b/arch/powerpc/boot/dts/mpc8610_hpcd.dts index 9535ce6..83c3218 100644 --- a/arch/powerpc/boot/dts/mpc8610_hpcd.dts +++ b/arch/powerpc/boot/dts/mpc8610_hpcd.dts @@ -286,6 +286,7 @@ ssi@16100 { compatible = "fsl,mpc8610-ssi"; + status = "disabled"; cell-index = <1>; reg = <0x16100 0x100>; interrupt-parent = <&mpic>; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a0e18b8..45d27b1 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -636,12 +636,19 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, struct resource res; char name[64]; - /* We are only interested in SSIs with a codec phandle in them, so let's - * make sure this SSI has one. + /* SSIs that are not connected on the board should have a + * status = "disabled" + * property in their device tree nodes. */ - if (!of_get_property(np, "codec-handle", NULL)) + if (!of_device_is_available(np)) return -ENODEV; + /* Check for a codec-handle property. */ + if (!of_get_property(np, "codec-handle", NULL)) { + dev_err(&of_dev->dev, "missing codec-handle property\n"); + return -ENODEV; + } + /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); if (!sprop || strcmp(sprop, "i2s-slave")) { -- 1.7.0.1

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[alsa-devel] [PATCH v5 0/1] A codec driver for the TI WL1273 FM Radio.
by Matti J. Aaltonen 05 Aug '10

05 Aug '10

Hello. New in the fifth version: > > > +/* codec private data */ > > > +struct wl1273_priv { > > > + struct snd_soc_codec codec; > > > + enum wl1273_mode mode; > > > + struct wl1273_core *core; > > > + unsigned int channels; > > > +}; > > > > ...why would anything outside the driver be peering into this? > > Well, I see no reason... So I'll move it from the header to the c -file > and then it's fine? OK... that's most probably what you mean. So I've moved the private struct definition from the header to the code file. And following the same logic I also moved the mode enum definition. Cheers, Matti A. Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 552 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 30 +++ 2 files changed, 582 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

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[alsa-devel] [PATCH] ASoC: multi-comp: soc-core: Fix for machine given DAPM widgets
by Peter Ujfalusi 05 Aug '10

05 Aug '10

Move the DAPM initialization after the machine driver has finished it's init (dai_link driver). Since in machine init we can add DAPM widgets we need to take them into account as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello Liam, Mark! With this patch the machine driver given DAPM routing will be functional with the multi-comp. This patch also removes the need to call the snd_soc_dapm_sync from machine driver's init call, since it will be done by the core. Please squash this patch to the core changes. Thanks, Peter sound/soc/soc-core.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a7c9a48..8afa271 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1324,9 +1324,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) return ret; } } - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); - /* mark codec as probed and add to card codec list */ codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); @@ -1375,6 +1372,10 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_sync(codec); + /* register the rtd device */ rtd->dev.init_name = rtd->dai_link->stream_name; rtd->dev.release = rtd_release; -- 1.7.2

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[alsa-devel] arecord: main:552: audio open error: Invalid argument
by Timur Tabi 05 Aug '10

05 Aug '10

Liam and Mark, I created a log with some printks that kinda show what's going on. In one process I run speaker-test, and then I start arecord in another process. fsl_ssi_startup:286 substream=ef3e5000 playback=0 capture=0 fsl_ssi_startup:288 rtd=ef3a2c00 rate=0 fsl_ssi_startup:298 fsl_ssi_startup:373 first=(null) second=(null) fsl_ssi_startup:427 soc_pcm_open:381 fsl_dma_open:408 rtd=ef3a2c00 rate=0 soc_pcm_open:391 soc_pcm_open:402 soc_pcm_open:412 soc_pcm_open:463 soc_pcm_open:471 soc_pcm_open:477 soc_pcm_open:484 asoc: cs4270-hifi <-> e0016000.ssi info: asoc: rate mask 0x40001fff asoc: min ch 2 max ch 2 asoc: min rate 5512 max rate 192000 fsl_ssi_hw_params:450 Rate set to 48000Hz (requested 48000Hz) fsl_dma_hw_params:579 soc_pcm_hw_params:748 rtd=ef3a2c00 rate=48000 ... fsl_ssi_startup:286 substream=ef3e5300 playback=1 capture=0 fsl_ssi_startup:288 rtd=ef3a2d10 rate=0 fsl_ssi_startup:373 first=ef3e5000 second=(null) fsl_ssi_startup:411 runtime=ef4ba800 sample_bits=16 fsl_ssi_startup:427 soc_pcm_open:381 fsl_dma_open:408 rtd=ef3a2d10 rate=0 <<------ soc_pcm_open:391 soc_pcm_open:402 soc_pcm_open:412 soc_pcm_open:463 soc_pcm_open:471 soc_pcm_open:477 soc_pcm_open:484 soc_pcm_apply_symmetry:335 rtd=ef3a2d10 rate=0 capture: Symmetry forces 0Hz rate In summary, the problem appears to be that asoc calls soc_pcm_apply_symmetry() in soc_pcm_open(), but this occurs before soc_pcm_hw_params() can set rtd->rate. So rtd->rate is always zero when soc_pcm_apply_symmetry() is called. For some reason, this isn't a problem when arecord is started before speaker-test. In that case, rtd->rate is 48000 when speaker-test is started: fsl_ssi_startup:286 substream=ef3e5300 playback=0 capture=0 fsl_ssi_startup:288 rtd=ef3e6d10 rate=0 fsl_ssi_startup:298 fsl_ssi_startup:373 first=(null) second=(null) fsl_ssi_startup:427 soc_pcm_open:382 fsl_dma_open:408 rtd=ef3e6d10 rate=0 soc_pcm_open:392 soc_pcm_open:403 soc_pcm_open:413 soc_pcm_open:464 soc_pcm_open:472 soc_pcm_open:478 soc_pcm_open:485 asoc: cs4270-hifi <-> e0016000.ssi info: asoc: rate mask 0x40001fff asoc: min ch 2 max ch 2 asoc: min rate 5512 max rate 192000 fsl_ssi_hw_params:450 fsl_dma_hw_params:579 soc_pcm_hw_params:749 rtd=ef3e6d10 rate=48000 pop wq checking: Playback status: inactive waiting: yes ... fsl_ssi_startup:286 substream=ef3e5000 playback=0 capture=1 fsl_ssi_startup:288 rtd=ef3e6c00 rate=48000 fsl_ssi_startup:373 first=ef3e5300 second=(null) fsl_ssi_startup:411 runtime=ef4c6c00 sample_bits=16 fsl_ssi_startup:427 soc_pcm_open:382 fsl_dma_open:408 rtd=ef3e6c00 rate=48000 <<------ soc_pcm_open:392 soc_pcm_open:403 soc_pcm_open:413 soc_pcm_open:464 soc_pcm_open:472 soc_pcm_open:478 soc_pcm_open:485 soc_pcm_apply_symmetry:333 rtd=ef3e6c00 rate=48000 playback: Symmetry forces 48000Hz rate asoc: cs4270-hifi <-> e0016000.ssi info: asoc: rate mask 0x40001fff asoc: min ch 2 max ch 2 asoc: min rate 5512 max rate 192000 fsl_ssi_hw_params:450 Rate set to 48000Hz (requested 48000Hz) fsl_dma_hw_params:579 Buffer soc_pcm_hw_params:749 rtd=ef3e6c00 rate=48000

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[alsa-devel] [PATCH 1/5] sound: serial-u16550: fix uninitialized variable warning
by Bill Pemberton 05 Aug '10

05 Aug '10

Fix sound/drivers/serial-u16550.c:905: warning: 'uart' may be used uninitialized in this function Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> --- sound/drivers/serial-u16550.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index a25fb7b..5f8ae28 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -902,7 +902,7 @@ static int __devinit snd_uart16550_rmidi(struct snd_uart16550 *uart, int device, static int __devinit snd_serial_probe(struct platform_device *devptr) { struct snd_card *card; - struct snd_uart16550 *uart; + struct snd_uart16550 *uninitialized_var(uart); int err; int dev = devptr->id; -- 1.7.2

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[alsa-devel] Muting of the "Front" mixer control
by David Henningsson 05 Aug '10

05 Aug '10

Here's an interesting problem which I'm not sure whether it should be fixed in PulseAudio or the driver, but I'm starting here. I have a laptop with a HDA VIA VT1708 chip and would like auto-muting. VIA advertises auto-muting by changing the "Front Playback Switch" (see sound/pci/hda/patch_via.c:2023, via_hp_automute). Now PulseAudio's default mixer path, "analog-output", treats "Front" as part of the volume merge chain, which in this case is incorrect, and so when the switch gets muted, PA thinks that the chain is muted and gets confused (as in: no sound output, and PA advertises to g-v-c and others that it is now muted). It seems to me like from PA's perspective the name should have been "Speaker Playback Switch" instead, but VIA does not seem to use "Speaker" at all, and BIOS seems to have set up my caps for Speaker as "Line out", so I'm not sure if this is the right way to do it. For reference, here's my alsa-info: http://launchpadlibrarian.net/53047763/alsa-info.txt.JUhgxH8rDZ -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH v4 0/1] A codec driver for the TI WL1273 FM Radio.
by Matti J. Aaltonen 05 Aug '10

05 Aug '10

Hello, and thank you for the comments... Now I've created the patch against the multi component branch. I haven't been able to test the codec but I thought that it makes sense to send the code for review anyway. Liam wrote: > Btw, now that we have a naming convention for components how about this > having a "-radio" or "-fm" device suffix ? Now the name is set to "wl1273-fm". Thanks, Matti Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 542 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 40 ++++ 2 files changed, 582 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

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Re: [alsa-devel] [LAD] announcing envy24control, mudita (*) edition.
by Niels Mayer 05 Aug '10

05 Aug '10

On Sat, Jul 31, 2010 at 12:06 PM, Tim E. Real <termtech(a)rogers.com> wrote: > Niels do you have a new release or repo I can work with rather than > the original release, which may be too outdated to patch against by now? http://nielsmayer.com/envy24control/mudita24-1.0.1.tar.gz (source directory, follow README). Please give it a try and report on how this "release candidate" works for you. This 1.0.1 release should fix most of the problems/gripes/bugs/errors reported on linux-audio-devel. New command-line options "-g 1" allow multi-channel folk to get rid of the stereo grouping; those annoyed by the "detent" on the sliders or the extra space taken up by the markings, use "-n" to turn off this feature.. The main thing I haven't been able to resolve is Fons' complaint regarding the DAC Volumes in "Analog Volume" panel not all being the same height, due to a -10/+4 radiobox that only appears in first DAC column for the Terratec EWS88MT; this radiobox causes the associated slider to be shorter than the other channels. Shouldn't this radiobox appear in all DAC columns?. Is this a regression in "mudita24" or a feature request for standard "envy24control"? I didn't change this aspect of the code, as far as I can tell. -- Niels. http://nielsmayer.com

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Re: [alsa-devel] Intel HDMI audio stops after 15 minutes
by Wu Fengguang 05 Aug '10

05 Aug '10

On Thu, Aug 05, 2010 at 01:35:09AM +0800, Shane W wrote: > Hey all, > > On my Intel DG45 mainboard (Yamaha RX-v1800 receiver) I am > now only getting audio for about 15 minutes before it stops > and I have to restart the X server. Playing a movie with > audio works fine but playing an mp3 has the cutout. Is > there a new power saving mode or the like that has crept > in? Did the screen go off at the same time? I was also fighting with it with a bunch of options Section "Monitor" Identifier "Configured Monitor" option "DPMS" "false" EndSection Section "ServerFlags" # minutes Option "BlankTime" "100" Option "StandbyTime" "300" Option "SuspendTime" "400" Option "OffTime" "500" EndSection Thanks, Fengguang

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[alsa-devel] als4000: enable burst mode
by Ondrej Zary 04 Aug '10

04 Aug '10

Enable burst mode to prevent dropouts during high PCI bus usage. The card is useless in X without this because of dropouts when anything moves on the screen (at least with PCI VGA card). Enabling this is also recommended by the datasheet (page 48). Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.35-rc2/sound/pci/als4000.c 2010-06-06 05:43:24.000000000 +0200 +++ linux-2.6.35-rc3/sound/pci/als4000.c 2010-08-03 23:50:08.000000000 +0200 @@ -763,9 +763,9 @@ static void snd_als4000_configure(struct /* SPECS_PAGE: 39 */ for (i = ALS4K_GCR91_DMA0_ADDR; i <= ALS4K_GCR96_DMA3_MODE_COUNT; ++i) snd_als4k_gcr_write(chip, i, 0); - + /* enable burst mode to prevent dropouts during high PCI bus usage */ snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL, - snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL)); + snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL) | 0x04); spin_unlock_irq(&chip->reg_lock); } -- Ondrej Zary

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