December 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] Recognise and use subdevice 0x3061 found in some HP dv6 notebooks.
by Kunal Gangakhedkar 26 Dec '09

26 Dec '09

On Wednesday 23 Dec 2009 12:02:47 am Kunal Gangakhedkar wrote: > On Monday 14 Dec 2009 4:02:58 pm Takashi Iwai wrote: > > At Mon, 14 Dec 2009 15:53:54 +0530, > > Kunal Gangakhedkar wrote: > > > > > > On Monday 14 Dec 2009 3:37:14 pm Takashi Iwai wrote: > > > > At Mon, 14 Dec 2009 00:29:42 +0530, > > > > Kunal Gangakhedkar wrote: > > > > > > > > > > Obvious patch. > > > > > Recently bought an HP Pavilion dv6-1110ax notebook. The modparm 'model=hp-dv5' > > > > > works, but I thought it was better if it were natively supported. > > > > > > > > > > Also, the 'hp_detect=1' hint > > > > > (echo "hp_detect = 1" > /sys/class/sound/hwC0D0/hints) works nicely to mute the > > > > > speakers when using the headphone jack(s). Is there any way to automatically > > > > > enable this? > > > > > > > > This is weird. hp_detect is basically always set unless you clear > > > > explicitly. I guess it's simply a known IRQ problem on HP dv > > > > laptops. It requires MSI. > > > > > > > > Anyway, could you test the very latest kernel without your patch and > > > > confirm that the change is still needed? > > > > > > > > > > The patch wasn't about hp_detect. It was to get the subdev recognised natively so that > > > you don't need the 'model=hp-dv5' modparm. > > > > Yeah, I know. But the speaker-muting should work even without > > hp_detect=1 hints. The IRQ thingy was only about the auto-muting. > > > > Also, with the latest version, the device might work even without the > > extra model option (as the auto-parser got improved). I'd like to > > avoid unneeded quirk lines as much as possible, so that's why I asked > > to check the latest one. > > > > > I'll try with enable_msi modparm as well without trying the hp_detect hint and let you know > > > about it. Will also try the latest kernel. > > > > Thanks! > > > > > > Takashi > > > > Sorry for the late reply - was busy with office work. > Just built the kernel from your git (sound-2.6.git) master > (as of commit 07e23a669544942a9154b64c583bdef014f66a15). > > This new driver didn't require 'model=hp-dv5' modparm - it did detect and > use my card properly; so, my patch is not needed anymore, I guess :) > > However, the hp_detect=1 problem still persists :(. > I had to use the "echo 'hp_detect=1' > /sys/class/sound/hwC0D0/hints" > method to get the auto-muting to work. > It's also true with enable_msi=1 modparm. > > Anything more you want me to try? > > Thanks, > Kunal > Also, "spec->gpio_led = 0x08" in patch_sigmatel.c seems to enable the colour changing (orange/white) of the mute/unmute button. I got this info from ubuntu's lucid kernel which somehow enabled the behaviour. But, it was missing from the sound-2.6.git master kernel that I built. I'll do more digging to see why it isn't working in the git kernel. This is the code that I copied from lucid kernel into linux-backports-modules that enabled the led behaviour: if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { const struct dmi_device *dev = NULL; while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (strcmp(dev->name, "HP_Mute_LED_1")) { switch (codec->vendor_id) { case 0x111d7608: spec->gpio_led = 0x01; break; case 0x111d7600: case 0x111d7601: case 0x111d7602: case 0x111d7603: /* <-- My codec STAC 92HD75B3X5 */ spec->gpio_led = 0x08; break; } break; } } } Thanks, Kunal

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[alsa-devel] [PATCH] jazz16: refine dma and irq selection
by Krzysztof Helt 25 Dec '09

25 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Narrow the dma and irq selection after the DOS driver. Add ALSA configuration description as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The dma8 must be 8-bit otherwise the 8-bit sound played through 16-bit dma has metalic noise. The 8-bit DMA cannot play 16-bit sound. Documentation/sound/alsa/ALSA-Configuration.txt | 15 +++++++++++++++ sound/isa/sb/jazz16.c | 21 ++++++++++++++++++++- sound/isa/sb/sb8_main.c | 3 ++- 3 files changed, 37 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8923597..92b0975 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1123,6 +1123,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ------------------- + + Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: + MVD1216 + MVA416 + MVA514. + + port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for SB DSP chip (3,5,7,9,10,15) + dma8 - DMA # for SB DSP chip (1,3) + dma16 - DMA # for SB DSP chip (5,7) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (2,3,5,7) + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index d52966b..8d21a3f 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -189,10 +189,29 @@ static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) if (port[dev] == SNDRV_AUTO_PORT) { snd_printk(KERN_ERR "please specify port\n"); return 0; + } else if (port[dev] == 0x200 || (port[dev] & ~0x270)) { + snd_printk(KERN_ERR "incorrect port specified\n"); + return 0; + } + if (dma8[dev] != SNDRV_AUTO_DMA && + dma8[dev] != 1 && dma8[dev] != 3) { + snd_printk(KERN_ERR "dma8 must be 1 or 3\n"); + return 0; } if (dma16[dev] != SNDRV_AUTO_DMA && dma16[dev] != 5 && dma16[dev] != 7) { - snd_printk(KERN_ERR "dma16 must be 5 or 7"); + snd_printk(KERN_ERR "dma16 must be 5 or 7\n"); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x030) != 0x300) { + snd_printk(KERN_ERR "incorrect mpu_port specified\n"); + return 0; + } + if (mpu_irq[dev] != SNDRV_AUTO_DMA && + mpu_irq[dev] != 2 && mpu_irq[dev] != 3 && + mpu_irq[dev] != 5 && mpu_irq[dev] != 7) { + snd_printk(KERN_ERR "mpu_irq must be 2, 3, 5 or 7\n"); return 0; } return 1; diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 3222aed..7d84c9f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -510,7 +510,8 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } switch (chip->hardware) { case SB_HW_JAZZ16: - runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (chip->dma16 == 5 || chip->dma16 == 7) + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; runtime->hw.rate_min = 4000; runtime->hw.rate_max = 50000; -- 1.6.4 ---------------------------------------------------------------------- Sprawdz sekrety swojego biorytmu. Kliknij >> http://link.interia.pl/f2534

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[alsa-devel] [PATCH] hda - HDMI sticky stream tag support
by Wu Fengguang 25 Dec '09

25 Dec '09

When we run the following commands in turn (with CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0), speaker-test -Dhw:0,3 -c2 -twav # HDMI speaker-test -Dhw:0,0 -c2 -twav # Analog The second command will produce sound in the analog lineout _as well as_ HDMI sink. The root cause is, device 0 "reuses" the same stream tag that was used by device 3, and the "intelhdmi - sticky stream id" patch leaves the HDMI codec in a functional state. So the HDMI codec happily accepts the audio samples which reuse its stream tag. The proposed solution is to remember the last device each azx_dev was assigned to, and prefer to 1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used 2) or assign a never-used azx_dev for HDMI With this patch and the above two speaker-test commands, HDMI codec will use stream tag 8 and Analog codec will use 5. The stream tag used by HDMI codec won't be reused by others, as long as we don't run out of the 4 playback azx_dev's. The legacy Analog codec will continue to use stream tag 5 because its device id is 0 (this is a bit tricky). Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_intel.c 2009-12-25 12:17:28.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_intel.c 2009-12-25 12:41:02.000000000 +0800 @@ -356,6 +356,7 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ + int device; /* last device number assigned to */ unsigned int opened :1; unsigned int running :1; @@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure */ /* assign a stream for the PCM */ -static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream) +static inline struct azx_dev * +azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) { int dev, i, nums; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct azx_dev *res = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev = chip->playback_index_offset; nums = chip->playback_streams; } else { @@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign } for (i = 0; i < nums; i++, dev++) if (!chip->azx_dev[dev].opened) { - chip->azx_dev[dev].opened = 1; - return &chip->azx_dev[dev]; + res = &chip->azx_dev[dev]; + if (res->device == substream->pcm->device) + break; } - return NULL; + if (res) { + res->opened = 1; + res->device = substream->pcm->device; + } + return res; } /* release the assigned stream */ @@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_s int err; mutex_lock(&chip->open_mutex); - azx_dev = azx_assign_device(chip, substream->stream); + azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { mutex_unlock(&chip->open_mutex); return -EBUSY;

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Re: [alsa-devel] [PATCH] Fix indentation in pcm_native.c
by Takashi Iwai 25 Dec '09

25 Dec '09

At Thu, 24 Dec 2009 01:17:46 +0100 (CET), Guennadi Liakhovetski wrote: > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Applied now. Thanks. Takashi > --- > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index ab73edf..cf57964 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -1917,13 +1917,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) > > err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, > hw->rate_min, hw->rate_max); > - if (err < 0) > - return err; > + if (err < 0) > + return err; > > err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, > hw->period_bytes_min, hw->period_bytes_max); > - if (err < 0) > - return err; > + if (err < 0) > + return err; > > err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, > hw->periods_min, hw->periods_max); >

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[alsa-devel] Personal invitation from Kunal Narkhede
by Kunal Narkhede 24 Dec '09

24 Dec '09

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Re: [alsa-devel] [PATCH] ASoC: fix params_rate() macro use in several codecs
by Mark Brown 24 Dec '09

24 Dec '09

On Thu, Dec 24, 2009 at 01:13:51AM +0100, Guennadi Liakhovetski wrote: > Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro > returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical > sampling rate. Fix them. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: fix params_rate() macro use in several codecs
by Liam Girdwood 24 Dec '09

24 Dec '09

On Thu, 2009-12-24 at 01:13 +0100, Guennadi Liakhovetski wrote: > Sevelar ASoC codec drivers wrongly assume, that the params_rate() macro > returns one of SNDRV_PCM_RATE_* defines instead of the actual numerical > sampling rate. Fix them. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > --- Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk>

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[alsa-devel] need help with io plugin programming: how to add delay ?
by Stefan Schoenleitner 23 Dec '09

23 Dec '09

Hi, I would like to write a simple alsa io plugin that takes some amount of PCM samples and sends them to a socket in a timely manner (this would be playback). Later on it should work the other way round as well (recording). (Well, actually I have been working on this for a month already but so far I still did not manage to get a working solution.) However, if the PCM stream comes from something prerecorded (i.e. a wav file) there needs to be some kind of delay. If for example the wav file would have a sampling rate of 8 kHz, 16 bit resolution and one period should consist of 160 samples, it would be necessary to write on period exactly each 20ms. For this reason delay needs to be added somewhere so that each 20ms exactly one period is begin sent to the socket. Otherwise all the PCM samples would be just immediately written to the socket without any delay. I read a lot of other alsa plugin code but still I have found no way to properly solve the delay problem. * Where and how do I need to add delay ?? My first approach was to add delay by sleeping in the transfer callback function. After I had implemented that solution (and it did not work that well) I was told that this is not the proper way to do it. * What is the proper way then ? I was told that the alsa plugin framework is poll based and I should come up with a poll based solution. However, I can not just let the alsa io plugin poll for POLLOUT on the socket as this would not add any delay at all. (It would just wait until new data can be written to the socket.) After looking at the bluetooth alsa plugin I came up with another idea (and my second approach): Basically it should be possible to create an independent timing thread in the alsa plugin that writes dummy data (e.g. a single byte) to a pipe each period (thus in my case each 20ms). The plugin would then poll for POLLIN on the above mentioned pipe and poll for POLLOUT on the socket at the same time. This way the transfer callback function would be called if and only if the following two conditions are met: * 20 ms period time has passed * the socket can accept new data * Would this solution be the proper way ? Another solution (my third one already) would be to introduce *no* delay in the alsa plugin itself, but do everything in the application that is listening on the socket. For this reason the application would read one period, process the data and then sleep for the rest of the time until the time for one period (i.e. 20ms) is over. After that the next period is read from the socket and so forth. The alsa plugin would then "automatically" write new data to the socket each time the application listening to that socket reads data and there is free space in the socket send buffer. * Once again, would this be a proper solution ? * If so, how do I know when to increment the hardware pointer ? (- After the send() call in the transfer callback function ? - Using an independent thread in the plugin that just increments the hardware pointer each 20ms ?) - something even more complex ?) My guess is that the third solution (delay is added in the application that listens to the socket) might work well as the delay is added at the end of the "audio processing chain" which makes it less sensitive to delays that are introduced in between. * Is this correct ? I would appreciate any help as I have invested way too much time in this already. cheers, stefan

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[alsa-devel] ASoC update for 2.6.33
by Mark Brown 23 Dec '09

23 Dec '09

The following changes since commit 1628af5adf64cc2960bce81009f119de822f876e: Guennadi Liakhovetski (1): ASoC: add missing parameter to mx27vis_hifi_hw_free() are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Eric Millbrandt (1): ASoC: Do not write to invalid registers on the wm9712. sound/soc/codecs/wm9712.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

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Re: [alsa-devel] Mic not working on Dell Vostro 1700 by default
by Takashi Iwai 23 Dec '09

23 Dec '09

At Wed, 23 Dec 2009 17:47:21 +0100, Anisse Astier wrote: > > On Wed, 23 Dec 2009 17:28:45 +0100, Anisse Astier <anisse(a)astier.eu> wrote : > > > > > I was confused previously, and i didn't know a pci device could have a > > subsystem vendor/device with id's. > > > > I cleared my head, and did this patch. It's on top of alsa-kernel tree. > > > > Compiling now, I don't see why it wouldn't work, I'll give you an update > > ASAP. > > I works as expected. OK, applied now. Thanks. Takashi

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