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[alsa-devel] Via VT82xx VT1708 ICE1724HT and higher sample rate/bit depth files through SPDIF
by Demian Martin 26 Feb '08

26 Feb '08

I have several problems trying to play files higher than 48000 KHz and with bit depth of 24 bits. I submitted this to the users and got a resounding silence. First- sample rate Even though the documents claim that it will play at 176.4 KHz and 192KHz whenever I play a file at these rates it down converts to ½ the rate (88200 and 96000). I tested this with the ICE1724HT (Juli@) which has almost the same core, I never could get 176.4KHz at the SPDIF output. I confirmed that the ICE1724 will play the file at 176.4KHz in Windows so its not a hardware limitation. I tried to do the same with the VT1708 (on a Via CX700 chipset) and the Windows driver wouldnt let me at anything other than 48000. Second, the only way I can get the 24 bit files out in a form that the card can play is to pass them through dmix. But I need to reconfigure dmix for each sample rate. And Im not convinced that it isnt resampling twice, both at the input and the output. Or how to configure the buffers etc. to prevent underruns even when the only conversion is to pass a 24 bit file to the card. Im trying to build a high quality music server that will pass the files to either the internal or an external DAC at the bit depth and sample rate they were recorded at. I have tried to understand the various plugs and settings but it seems they are all focused on mixing various streams. I want to send the files unmixed to the output. I cant figure out what set of plugs and settings in asound.conf would enable this.

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[alsa-devel] [PATCH 1/3] ASoC: Fix TLV320AIC3X PLL divider table for 64 kHz rate
by Mark Brown 26 Feb '08

26 Feb '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 710e028..569ecac 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -681,8 +681,8 @@ static const struct aic3x_rate_divs aic3x_divs[] = { {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ - {22579200, 96000, 96000, 0x1, 8, 7075}, - {33868800, 96000, 96000, 0x1, 5, 8049}, + {22579200, 64000, 96000, 0x1, 8, 7075}, + {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, -- 1.5.4.3

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[alsa-devel] new driver: 30mS interval problem
by William Juul 26 Feb '08

26 Feb '08

Hello I am new to ALSA and trying to write a new driver for a DAC connected with PCM to an AVR32 on a NGW100 reference card provided by Atmel. The sampling rate I am currently using is 11047and the DAC is providing 4 channels of 24bit. The HW interface is using DMA to copy data to RAM. By studying the audio data in hexdump or in Audacity I can verify that the sound looks good in intervals of about 30mS, then all channels are garbled for 30mS. This pattern repeat itself throughout the audio capture. I am not confident I have configured all ALSA parameters properly. How can I go about fixing/debugging this 30mS intverval problem? Below is the command I am using. Best regards William Juul # arecord -r 11047 -c 4 -f S24_LE -s 1 -A 100 -d 5 --buffer-size 16384 -F 21333 -v > test.wav Recording WAVE 'stdin' : Signed 24 bit Little Endian, Rate 11047 Hz, Channels 4 Plug PCM: Hardware PCM card 0 'AVR32 NGW100 external DAC' device 0 subdevice 0 pcm->setup: 1 stream : CAPTURE access : RW_INTERLEAVED format : S24_LE subformat : STD channels : 4 rate : 11047 exact rate : 11047 (11047/1) msbits : 32 buffer_size : 16384 period_size : 490 period_time : 44355 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 490 xfer_align : 490 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824

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Re: [alsa-devel] ASoC - Output widgets
by Mark Brown 26 Feb '08

26 Feb '08

On Tue, Feb 26, 2008 at 10:11:42AM +0100, Alexandre BOUIN wrote: > DAPM WM8900 snd_soc_dapm_stream_event > FM Transmitter: On in 2 out 1 > in static FMOUT_R > in static FMOUT_L > I initialize FM Transmitter like this : > snd_soc_dapm_set_endpoint(codec, "FM Transmitter", 0); > but it still enabled. Why ? According to the above trace your FM transmitter is still reporting as being connected. Have you checked to ensure that your your call to configure the transmitter is happening after your board widgets have been registered?

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Re: [alsa-devel] ASoC - Output widgets
by Mark Brown 26 Feb '08

26 Feb '08

On Mon, Feb 25, 2008 at 10:39:15AM +0100, Alexandre BOUIN wrote: > - 1 SND_SOC_DAPM_MIXER is connected to 2 SND_SOC_DAPM_PGA with power > control. > - each PGA is connected to an SND_SOC_DAPM_OUTPUT > - according to tosa example : > - first OUTPUT is connected to a SND_SOC_DAPM_HP defined as endpoint > - second OUTPUT is connected to a SND_SOC_DAPM_SPK defined as endpoint > this SPK is set disconnected in my tosa_ac97_init -like function Could you please post dmesg output with DAPM_DEBUG turned on which shows this happening?

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Re: [alsa-devel] [Alsa-devel] Questions about sound/isa/sb/sb8_main.c
by Clemens Ladisch 26 Feb '08

26 Feb '08

Rask Ingemann Lambertsen wrote: > I'm looking at sound/isa/sb/sb8_main.c from linux 2.6.20. In > snd_sb8_capture_prepare(), a line reads: > > if (chip->capture_format != SB_DSP_OUTPUT) { > > Shouldn't this be SB_DSP_INPUT instead? Yes. Now fixed in the ALSA tree. Thanks! Clemens

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[alsa-devel] POLLIN and POLLOUT event in pcm io plugins
by Pharaoh . 26 Feb '08

26 Feb '08

Hi, The structure snd_pcm_ioplug has following members: int poll_fd; /**< poll file descriptor */ unsigned int poll_events; /**< poll events */ and the associated events are POLLIN and POLLOUT, I suppose. I am unable to understand how these fields are getting used. There are many other plugins which dont use this. What is the significance of these fields? -pharaoh.

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[alsa-devel] Fwd: HW decoding abstracted in ALSA
by Harsha priya gupta 25 Feb '08

25 Feb '08

Hi, Thanks for your immediate reply. My hardware supports MP3 decoding in HW (in addition to few others). I am looking to write a driver such that when an MP3 file is played, the file is decoded using the HW decoder and played. Is there a possibility in current ALSA framework to do so? Thanks, Harsha. On Mon, Feb 25, 2008 at 5:33 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Mon, 25 Feb 2008 12:27:00 +0530, > Harsha priya gupta wrote: > > > > Hi, > > > > I am new to ALSA and audio stack in Linux. I have a sound card that can > do > > HW decoding as well. I am looking at writing an ALSA driver such that > > whatever may be the player or MMF above, it should use the HW decoder > and > > not the inherit SW decoder that is available with the player. > > > > Can someone help me with some pointers or existing like-drivers? > > Which format is in mind? In theoery, you can implement the h/w > decoder on the ALSA framework, but there are some restrictions: > > - ALSA PCM assumes the constant bitrate > - The format bit must be defined > - The specific app recognizes and supports that format > > In the case of AC3 or DTS, it's embedded in SPDIF frames so the PCM > core stuff doesn't take care of the above. But, in the case of native > (non-PCM) formats, the support can be difficult as is. > > > Takashi > -- -Harsha -- -Harsha

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[alsa-devel] Patch for alsa-info.sh - AC97 including regs
by Travis Place 25 Feb '08

25 Feb '08

Ok, ive added the +regs file aswell, but decided to store its output in a seperate tmp file (before being added to the complete output). This way, we can change how its output if we'd like, with less stuffing around (and all tmp files are cleaned up after each run, anyway). -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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Re: [alsa-devel] hdsp rpm support
by kgrill＠chello.at 25 Feb '08

25 Feb '08

I started to work on this about a year ago, but had to stop because my son took the card to Istanbul where he spent a year studying. I'll get back to it in the next days, and depending on how much has changed in the meantime I may get something ready by the end of January. regards kg

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