February 2008 - Alsa-devel - mailman.alsa-project.org

[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 . 25 Feb '08

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

25 Feb '08

Sorry for the frequent patch interval.. I think this will do for a while.. Adds AC97 codec output if an AC97 card is in the system. -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] How does ALSA detect underrun in OSS emulation?
by Timur Tabi 25 Feb '08

25 Feb '08

Can someone explain to me the exact method that ALSA uses to detect underrun with OSS emulation? I have a driver that reports underrun on almost every period when I use OSS emulation and I'm playing with a non-supported sample rate. So I'm assuming that maybe my hardware is playing the audio too fast or too slow, and the driver is returning periods before ALSA expects them. What I don't understand is: how does ALSA know that a period was finished too early. Does it use a timer, or is it purely application driven? -- Timur Tabi Linux kernel developer at Freescale

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