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

[Kjetil Matheussen]

I see a posssible bug in jackrec. Don't know if this is the cause though:

diff --git a/example-clients/capture_client.c
b/example-clients/capture_client.c
index d3360b4..da11fd8 100644
--- a/example-clients/capture_client.c
+++ b/example-clients/capture_client.c
@@ -167,7 +167,7 @@ jack_shutdown (void *arg)
 static void
 setup_disk_thread (jack_thread_info_t *info)
 {
-   SF_INFO sf_info;
+ SF_INFO sf_info = {0}
        int short_mask;


On Tue, Jan 30, 2018 at 12:35 PM, Christian Affolter <
c.affolter at purplehaze.ch> 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-audio-connection-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
>
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