ALSA back end is not working #40
Comments
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I am continuously getting following log when I play a file in Dom U: 06.09.22 16:16:38.071 | XenEvtchn | DBG - Notify event channel, port: 76 Frame and bytes cannot be 0 for every frame, right? Can this be the issue? What could be the reason for this? |
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Hello,
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Dom 0:axx@axx-xxx-RV2:~/jkm/back_end/snd_be/build$ aplay -L axx@axx-xxx-RV2:~/jkm/back_end/snd_be/build$ aplay -l Dom U:axx@axx-HVM-domU:~$ aplay -L axx@axx-HVM-domU:~$ aplay -l
I have played only a 1 sec long wave file. However, log does not stop and towards the end, I can see "Ring buffer overflow"
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Any pointers for this issue? What should I look at for debugging? |
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Could you provide pulseaudio log? It can be obtained by the following document https://fedoraproject.org/wiki/How_to_debug_PulseAudio_problems. Also pulseaudio config would be helpful: |
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Hi, I tried snd_be in another board (which has a different SoC) and followed the same method. In this board audio is coming using alsa back end, but audio is choppy and I keep getting following log: 14.09.22 18:51:15.501 | AlsaPcm | DBG - Write to pcm device: default, size: -32 This issue is similar to issue #34 I have matched the buffer-size as you had suggested, but audio is still choppy. Below is the ALSA output on Dom 0 when same file is played. I have set buffer-size to 96000 (= 24000 * 2 channels * 2 bytes per sample). amd@amd:~/Dom_U1$ sudo aplay -vv ../music.wav Is the buffer-size setting correct? |
Sure, I will provide in the issue #42 |
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Hello, I am able to play audio perfectly from Dom U using ALSA back end now. This is using a different codec than previously used. I had to use "sudo aplay" instead of just "aplay" in Dom U. I am currently checking why sudo is required. Thank you for all your help. |
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Have you tried playing music with multimedia player like Rhythmbox which installed defaultly on your guest (ubuntu) system ? Is it normal? |
Supplementary： |
Hello,
I am trying to play audio from a HVM guest (ubuntu). My Dom 0 is also ubuntu. I am using snd_be for this purpose.
Current status:
Here is my vsnd config:
vsnd = [[ 'card, backend=Domain-0, buffer-size=22050, short-name=VCard, long-name=Virtual sound card, sample-rates=8000;11025;16000;22050;32000;44100;48000, sample-formats=s16_le', 'pcm, name=dev1', 'stream, unique-id=alsa, type=P' ]]
Following are the output of aplya -v and aplay --dump-hw-params respectively. Can you please help me understand what is going wrong here?
================================================================
axx@axx-xxx-RV2:~$ aplay -v ~/SampleAudio/music.wav
Playing WAVE '/home/amd/SampleAudio/music.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
ALSA <-> PulseAudio PCM I/O Plugin
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 22050
period_size : 5512
period_time : 125000
tstamp_mode : NONE
tstamp_type : GETTIMEOFDAY
period_step : 1
avail_min : 5512
period_event : 0
start_threshold : 22050
stop_threshold : 22050
silence_threshold: 0
silence_size : 0
boundary : 6206523236469964800
================================================================
amd@amd-Bilby-RV2:~$ aplay ~/SampleAudio/music.wav --dump-hw-params
Playing WAVE '/home/amd/SampleAudio/music.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
HW Params of device "default":
ACCESS: RW_INTERLEAVED
FORMAT: U8 S16_LE S16_BE S24_LE S24_BE S32_LE S32_BE FLOAT_LE FLOAT_BE MU_LAW A_LAW S24_3LE S24_3BE
SUBFORMAT: STD
SAMPLE_BITS: [8 32]
FRAME_BITS: [8 1024]
CHANNELS: [1 32]
RATE: [1 384000]
PERIOD_TIME: (2 4294967295)
PERIOD_SIZE: [1 1398102)
PERIOD_BYTES: [128 1398102)
PERIODS: [3 1024]
BUFFER_TIME: (7 4294967295]
BUFFER_SIZE: [3 4194304]
BUFFER_BYTES: [384 4194304]
TICK_TIME: ALL
================================================================
Regards,
Jyotirmoy
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