[Jack-Devel] Capture problems with jack2 on Axia-Alsa device

[Christian Affolter]

Hi Thomas,

On 30.01.2018 14:05, Thomas Brand wrote:
>>> this sounds like a SR mismatch but it's not obvious.
>>
>> Yes, most likely.
>>
>>
>>
>>> before you debug more, can you try to record with
>>> https://github.com/kmatheussen/jack_capture ?
>>>
>>
>> This results in the same issue. I've started jack_capture with the
>> following parameters:
>>
>> jack_capture -b 32 -c 2 -d 10 -fn /tmp/test-jack_capture.wav \ -p
>> system:capture_1 \
>> -p system:capture_2
>>
>>
>> The resulting file:
>> https://filebin.ca/3pykjU8lo2jO/test-jack_capture.wav
>>
>>
>>
>>> A simple test is to connect system:capture to system:playback and see
>>> if you get what's expected on your headphone/speakers.
>>
>> Yes, but unfortunately it's a virtual device running on a headless
>> server without any speakers attached. The playout would go into the
>> Livewire+ network where I would have to re-capture it from another
>> system.
>>
> 
> you can try to run jack with --verbose, maybe it gives another hint.

I did that already, see the pastbin links below:

>>>>> # Starting jackd
>>>>> # Verbose output at: https://pastebin.com/YzHEGSnR
>>>>> jackd -d alsa -d hw:0

>>>>>> # Capture via jackrec
>>>>>> # This results in a too high pitched WAV file
>>>>>> # Verbose output at: https://pastebin.com/PCnymKLA



> you can run jack_bufsize to be absolutely sure it runs at the correct SR.

jack_bufsize
buffer size = 1024  sample rate = 48000


> you can play a file in jack, record that and see if it's ok. if that's the
> case, most probably jack is running ok.

I did that with mpg123, which works correctly (see below)

>>>>> I've also tried to play an mp3 file via mpg123 over jack (without the
>>>>> involvement of the Alsa device) and record it again with jackrec.
>>>>> This works and sounds correct.


> the alsa device could be the issue. you could try to start a dummy jack
> and add the alsa device with alsa_in to see if it makes a difference.
> also make sure that you use a vanilla jack (git clone
> https://github.com/jackaudio/jack2/) to rule out any other side-effect.

OK, I will compile from source later an re-run the captures.


Thanks again
Chris

>>> On Tue, January 30, 2018 12:35, Christian Affolter wrote:
>>>
>>>> Hi everyone,
>>>>
>>>>
>>>>
>>>> capturing from ALSA capture devices via the Jack2 sound server
>>>> (1.9.12),
>>>> results in too high pitched wav files (they are playing "too fast" and
>>>>  sound like the "chipmunks"). If I run the same capture directly from
>>>> the ALSA devices (without Jack involved), everything sounds as
>>>> expected without any problems.
>>>>
>>>> Audio examples:
>>>> Capture via jackrec: https://filebin.ca/3pyMxBw8cexQ/test-jackrec.wav
>>>> Capture via arecord: https://filebin.ca/3pyOPjcKGym5/test-arecord.wav
>>>>
>>>>
>>>>
>>>> The device in question is a "virtual" Axia-ALSA (Livewire+) audio
>>>> device on CentOS 7 which operates at a sample rate of 48kHz and a bit
>>>> depth of either 16 or 32. As far as I can see, the sample rate and
>>>> format detection on the Jack side looks correct. I'm therefore looking
>>>> for some guidance on how to further debug this, I most certainly
>>>> missed something obvious.
>>>>
>>>> I've also tried to play an mp3 file via mpg123 over jack (without the
>>>>  involvement of the Alsa device) and record it again with jackrec.
>>>> This
>>>> works and sounds correct.
>>>>
>>>> Here is what I've tried and what the environment looks like:
>>>>
>>>>
>>>>
>>>> # Capabilities of the Axia-ALSA device
>>>> arecord -D hw:0 --dump-hw-params
>>>>
>>>> Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono
>>>> HW Params of device "hw:0":
>>>> --------------------
>>>> ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
>>>> FORMAT:  S16_LE S32_LE
>>>> SUBFORMAT:  STD
>>>> SAMPLE_BITS: [16 32]
>>>> FRAME_BITS: [16 256]
>>>> CHANNELS: [1 8]
>>>> RATE: 48000
>>>> PERIOD_TIME: (41 1365334)
>>>> PERIOD_SIZE: [2 65536]
>>>> PERIOD_BYTES: [64 131072]
>>>> PERIODS: [1 1024]
>>>> BUFFER_TIME: (41 1365334)
>>>> BUFFER_SIZE: [2 65536]
>>>> BUFFER_BYTES: [64 131072]
>>>> TICK_TIME: ALL
>>>> --------------------
>>>> arecord: set_params:1299: Sample format non available
>>>> Available formats:
>>>> - S16_LE
>>>> - S32_LE
>>>>
>>>>
>>>>
>>>>
>>>> # Capture via arecord directly from the ALSA device (without jackd)
>>>> # This works as expected and the WAV file sounds fine
>>>> arecord -D hw:0 -c 2 -d 10 -r 48000 -f S32_LE -v /tmp/test-arecord.wav
>>>>
>>>>
>>>> Recording WAVE '/tmp/test-arecord.wav' : Signed 32 bit Little Endian,
>>>>  Rate 48000 Hz, Stereo
>>>> Hardware PCM card 0 'Axia' device 0 subdevice 0
>>>> Its setup is:
>>>> stream       : CAPTURE access       : RW_INTERLEAVED format       :
>>>> S32_LE
>>>> subformat    : STD channels     : 2 rate         : 48000 exact rate
>>>> :
>>>> 48000 (48000/1)
>>>> msbits       : 32 buffer_size  : 16384 period_size  : 4096 period_time
>>>> :
>>>> 85333
>>>> tstamp_mode  : NONE tstamp_type  : MONOTONIC period_step  : 1
>>>> avail_min
>>> :
>>>
>>>> 4096
>>>> period_event : 0 start_threshold  : 1 stop_threshold   : 16384
>>>> silence_threshold: 0
>>>> silence_size : 0 boundary     : 4611686018427387904 appl_ptr     : 0
>>>> hw_ptr : 0
>>>>
>>>>
>>>>
>>>>
>>>> # playing arecord wav (via my local notebook's HDA Intel PCH
>>>> # device), sounds correct
>>>> aplay test-arecord.wav
>>>>
>>>> Playing WAVE 'test-arecord.wav' : Signed 32 bit Little Endian, Rate
>>>> 48000 Hz, Stereo
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> # Starting jackd
>>>> # Verbose output at: https://pastebin.com/YzHEGSnR
>>>> jackd -d alsa -d hw:0
>>>>
>>>> jackdmp 1.9.12 Copyright 2001-2005 Paul Davis and others. Copyright
>>>> 2004-2016 Grame.
>>>> Copyright 2016-2017 Filipe Coelho.
>>>> jackdmp comes with ABSOLUTELY NO WARRANTY This is free software, and
>>>> you are welcome to redistribute it under certain conditions; see the
>>>> file COPYING for details
>>>> no message buffer overruns no message buffer overruns no message
>>>> buffer overruns JACK server starting in realtime mode with priority 20
>>>>  self-connect-mode is "Don't restrict self connect requests"
>>>> audio_reservation_init Acquire audio card Audio0 creating alsa driver
>>>> ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
>>>> configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2
>>>> periods ALSA: final selected sample format for capture: 32bit integer
>>>>  little-endian ALSA: use 2 periods for capture ALSA: final selected
>>>> sample format for playback: 32bit integer little-endian ALSA: use 2
>>>> periods for playback
>>>>
>>>>
>>>>
>>>> # Capture via jackrec
>>>> # This results in a too high pitched WAV file
>>>> # Verbose output at: https://pastebin.com/PCnymKLA
>>>> jackrec -f /tmp/test-jackrec.wav -d 10 -b 32 system:capture_1
>>>> system:capture_2
>>>>
>>>>
>>>>
>>>> # playing jackrec wav (via my local notebook's HDA Intel PCH
>>>> # device), sounds incorrect
>>>> aplay test-jackrec.wav
>>>>
>>>> Playing WAVE 'test-jackrec.wav' : Signed 32 bit Little Endian, Rate
>>>> 48000 Hz, Stereo
>>>>
>>>>
>>>>
>>>>
>>>> System environment:
>>>> Distribution: CentOS 7.4.1708
>>>> Kernel:       3.10.0-693.17.1.el7.x86_64
>>>> ALSA Utils:   1.1.3
>>>> Jackd:        1.9.12
>>>>
>>>>
>>>>
>>>> The jackd was rebuilt from Fedora source RPM to be able to test with
>>>> the latest version:
>>>> https://build.opensuse.org/package/show/home:radiorabe:audio/jack-aud
>>>> io-c onnection-kit
>>>>
>>>>
>>>> Many thanks and best regards
>>>> Chris
>>>> _______________________________________________
>>>> Jack-Devel mailing list
>>>> Jack-Devel at lists.jackaudio.org
>>>> http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org
>>>>
>>>>
>>>>
>>>
>>>
>>
>>
> 
>