Hi Chris,
 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:
  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 then go on to
 see if jack_capture solves the issue for you.
 Greetings
 Thomas 
 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-audio-c
 onnection-kit
 Many thanks and best regards
 Chris
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