Audio Codec configuration for 44100Hz sample rate on STM32MP1 - difficulties with Bluetooth Audio Streaming
Dear community,
I once again need your help. :)
I'm using a TAS5760L on a custom board with STM32MP157 MPU and Yocto-Linux, Dunfell.
Since the codec is not part of the available Linux kernel audio codecs, a simple spdif-codec (inux,spdif-dit) with I2S signal is used to send data to the TAS5760L.
Audio is working quite well when playing wav-files with aplay. But I recognised, that the Frame Sync (FS, or LRCLK) is always at 48kHz, no matter what is configured in the device tree or in /etc/alsa.conf.
Background:
For aplay this doesn't matter much, but if audio is streamed via Bluetooth (at 44.1kHz) there is a difference in sample rates, which results in the following error:
"An underrun has occurred". This error occurs cyclically about every 7 seconds and a glitch/stutter can be heard then in the audio.
I use bluez-alsa (bluealsa/bluealsa-aplay) to send audio stream from for example a smartphone to the board. Here is some more information:
- Bluetooth module has CYW4343W chipset (same as the STM32MP157-DK2)
- BT audio stream comes via UART/A2DP
- If the BT audio stream is captured with arecord (arecord -D bluealsa capture.wav), the resulting wav-file is glitch-free.
- Bluetooth audio streaming works glitch-free on the STM32MP157-DK2 with the same configuration/frameworks (bluez-alsa, UART/AD2P, Linux Kernel: 5.10)
My question now:
I think the "underrun" is caused by the differences of the sample rates. How can I set LRCLK of my audio codec to 44.1kHz? Or asked differently: Why do I always measure 48kHz there, no matter how I configure?
I would be very grateful for any help or tips regarding this issue.
Appendix
Device Tree snippet:
/* Sound interface */
sound {
#address-cells = <1>;
#size-cells = <0>;
compatible = "simple-audio-card";
simple-audio-card,name = "TAS5760L";
simple-audio-card,dai-link@0 {
reg = <0>;
format = "i2s";
bitclock-master = <&sound1_cpu>;
frame-master = <&sound1_cpu>;
mclk-fs = <256>; /*factor x for mclk = x*fs (framesynchro clock)*/
sound1_cpu: cpu {
sound-dai = <&sai3b>;
};
codec {
sound-dai = <&codec_out>;
};
};
};
codec_out: spdif-out {
#address-cells = <0>;
#size-cells = <0>;
#sound-dai-cells = <0>;
compatible = "linux,spdif-dit";
status = "okay";
};
};Asound.conf;
defaults.pcm.card 0
defaults.ctl.card 0
defaults.pcm.rate_converter "linear"
pcm.tas5760l {
type hw
card 0
device 0
}
pcm.dmixer {
type dmix
ipc_key 4096
ipc_perm 0666
slave.pcm "tas5760l"
slave {
period_time 1000000
period_size 2048
buffer_size 5000000
format S16_LE
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
pcm.softvol {
type softvol
slave.pcm "dmixer"
control {
name "Softvol"
card 0
}
min_dB -70.2
max_dB -0.0
}
pcm.bluetooth_softvol {
type softvol
slave.pcm "dmixer"
control {
name "Bluetooth_Vol"
card 0
}
min_dB -70.2
max_dB -0.0
}
pcm.!default {
type plug
slave.pcm "softvol"
}
pcm.bluetooth {
type plug
slave.pcm "bluetooth_softvol"
}
defaults.bluealsa.interface "hci0"
defaults.bluealsa.profile "a2dp"Block diagram of the setup:
