Audio Output Quality is 16 KHz on v2

Products & Technology ReSpeaker
12

Hi!



Using the 48k 6 channels firmware, the audio playback quality is degraded as soon as I start capturing.
[code]
SEEED ReSpeaker 4 Mic Array (UAC1.0) at usb-3f980000.usb-1.1.3, full speed : USB Audio

Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 48000, 48000, 48000

Capture:
Status: Stop
Interface 2
Altset 1
Format: S16_LE
Channels: 6
Endpoint: 2 IN (ASYNC)
Rates: 48000, 48000, 48000
[/code]


Using usbtop, I can confirm that the USB bandwidth is nowhere near its max capacity (< 1MB/s).



Your setup is identical to mine but I am not able to get a clear audio playback during capture.



Here is what I do :



Session 1:
</s>aplay loop-test.wav -Dplughw:CARD=ArrayUAC10 -vv<e>
Stdout:
</s><i> </i>Playing WAVE 'loop-test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 48000 Hz, Stereo Plug PCM: Hardware PCM card 1 'ReSpeaker 4 Mic Array (UAC1.0)' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S24_3LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 24 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 <e>

Audio is clear. Then I start the capture in another session.



Session 2:
</s>arecord -Dhw:CARD=ArrayUAC10 mic.wav -c 6 -r 48000 -f S16_LE -vv<e>
Stdout:
</s><i> </i>Recording WAVE 'mic.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 6 Hardware PCM card 1 'ReSpeaker 4 Mic Array (UAC1.0)' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 <e>

Audio is degraded as soon as arecord starts capturing.



See the Audacity capture: https://we.tl/t-xi2h2gJpZZ



I have no clue why… Hope this helps your software engineers identify the cause. Tell me if you need more info.



Thanks again for your help.