SK-AM62: Audio Codec aix31xx wait bits failed.

Hi, Guy

Can you mesrue the bck and ws clk during runing loopback test？

Hello, 

following are the clock or running but no audio on loopback. 

mclk = 12.02MHz

bclk = 1.4MHz

wclk = 45.1KHz

HI Shenghao, I know this is separate question, can you tell me how to change the audio codec sampling frequenct from default 44.1KHz to 48KHz? please 

in Linux? 

So, the sequence is as follows:

1. Run pulseaudio and the clock mclk,wclk, bclk are on. 

2. Run g-streamer in loopback mode. 

3. get error on consol:

110.296154] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.307741] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.314596] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.321489] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.328458] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.335419] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.346960] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.353838] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.360811] davinci-mcasp 2b10000.mcasp: Receive buffer overflow
[ 110.367923] davinci-mcasp 2b10000.mcasp: Receive buffer

4. log from pulseaudio

...320aic31xx-hifi-0] alsa-util.c: PCM state is XRUN
320aic31xx-hifi-0] alsa-source.c: Starting capture.
C0 is accessible: yes
ks and sources of card alsa_card.platform-soundcard.
0aic31xx-hifi-0] alsa-sink.c: Trying resume...
0aic31xx-hifi-0] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
0aic31xx-hifi-0] alsa-util.c: Maximum hw buffer size is 2972 ms
0aic31xx-hifi-0] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
320aic31xx-hifi-0] alsa-util.c: Got POLLERR from ALSA
320aic31xx-hifi-0] alsa-util.c: Got POLLIN from ALSA
320aic31xx-hifi-0] alsa-util.c: PCM state is XRUN
320aic31xx-hifi-0] alsa-source.c: Starting capture.
0aic31xx-hifi-0] alsa-util.c: Set period size first (to 13505 samples), buffer size second (to 81033 samples).
0aic31xx-hifi-0] alsa-util.c: Device hw:0 doesn't support 48000 Hz, changed to 44100 Hz.
0aic31xx-hifi-0] alsa-util.c: ALSA period wakeups disabled
0aic31xx-hifi-0] alsa-sink.c: Resume failed, couldn't restore original sample settings.
320aic31xx-hifi-0] alsa-util.c: Got POLLERR from ALSA
320aic31xx-hifi-0] alsa-util.c: Got POLLIN from ALSA
320aic31xx-hifi-0] alsa-util.c: PCM state is XRUN
320aic31xx-hifi-0] alsa-source.c: Starting capture.
hannel.
C0 is accessible: yes
ks and sources of card alsa_card.platform-soundcard.
0aic31xx-hifi-0] alsa-sink.c: Trying resume...
0aic31xx-hifi-0] alsa-util.c: Trying to disable ALSA period wakeups, using timers only
0aic31xx-hifi-0] alsa-util.c: Maximum hw buffer size is 2972 ms
0aic31xx-hifi-0] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
320aic31xx-hifi-0] alsa-util.c: Got POLLERR from ALSA
320aic31xx-hifi-0] alsa-util.c: Got POLLIN from ALSA
320aic31xx-hifi-0] alsa-util.c: PCM state is XRUN
320aic31xx-hifi-0] alsa-source.c: Starting capture.
0aic31xx-hifi-0] alsa-util.c: Set period size first (to 13505 samples), buffer size second (to 81033 samples).
0aic31xx-hifi-0] alsa-util.c: Device hw:0 doesn't support 48000 Hz, changed to 44100 Hz.
0aic31xx-hifi-0] alsa-util.c: ALSA period wakeups disabled
0aic31xx-hifi-0] alsa-sink.c: Resume failed, couldn't restore original sample settings.
320aic31xx-hifi-0] alsa-util.c: Got POLLERR from ALSA
320aic31xx-hifi-0] alsa-util.c: Got POLLIN from ALSA
320aic31xx-hifi-0] alsa-util.c: PCM state is XRUN
320aic31xx-hifi-0] alsa-source.c: Starting capture.
C0 is accessible: yes...

Hi Mitesh,

Which part number you used ?

The 48khz should be automatically selected when play 48khz audio, See the previous log, The register status indicates that the ADC & DAC is not shutting down as expected, is there any sound to playback or capture?

Regarding to the log of "Receive buffer overflow" you should increase the minBufferSize.

Thanks

Hi Kevin, we are using TLV320AIC3111. When we do 'pactl list sources' the default rate is 44100 and not 48000. and when I change the config file to use 48000 it says not supported to change the rate. 

Could you please let me know how to increase the buffers please? 

Thanks 

Hi Mitesh,

First of all, Our driver can support 48k. Your log shows the PCM state is XRUN, the XRUN means under run or over run, I think it's related to buffer size, you can add ~/.asoundrc file to user path to config buffersize and sample rate. Here's an example, you can find some resource for learning to config and use ~/.asoundrc file

pcm.!default{
        type asym
        playback.pcm {
                type plug
                slave {
                        pcm "hw:0,2"
                        channels 2  
                        period_size 1024  
                        buffer_size 4096  
                        rate 48000
                        periods 0  
                        period_time 0
                }
        }
        capture.pcm {
                type plug
                slave {
                        pcm "hw:1,0"
                        channels 2  
                        period_size 1024  
                        buffer_size 4096  
                        rate 48000  
                        periods 0  
                        period_time 0
                }
        }

}

Thanks

Thanks Kevin, I shall try this. Cheers. 

Hi Kevin, when I try the following config in asoundrc 

HI Kevin, when I set the asoundrc file to the following 

and try to run loopback g-streamer I get the following error:

I: [alsa-sink-davinci-mcasp.0-tlv320aic31xx-hifi tlv320aic31xx-hifi-0] alsa-util.c: Device hw:0 doesn't support 48000 Hz, changed to 44100 Hz.

Hi Mitesh,

Maybe the file is not working. Make sure the file is placed in ~/, then use command 'arcord -L' to reload the file.

Thanks

root@am62xx-evm:~# arecord -L
null
Discard all samples (playback) or generate zero samples (capture)
sysdefault:CARD=AM62xSKEVM
AM62x-SKEVM, davinci-mcasp.0-tlv320aic31xx-hifi tlv320aic31xx-hifi-0
Default Audio Device

root@am62xx-evm:~# ls -l /home/root/.asoundrc
-rw-r--r-- 1 root root 453 Jul 26 10:21 /home/root/.asoundrc

log from pulseaudio 

D: [pulseaudio] alsa-util.c: snd_pcm_dump():
D: [pulseaudio] alsa-util.c: Hardware PCM card 0 'AM62x-SKEVM' device 0 subdevice 0
D: [pulseaudio] alsa-util.c: Its setup is:
D: [pulseaudio] alsa-util.c: stream : PLAYBACK
D: [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED
D: [pulseaudio] alsa-util.c: format : S16_LE
D: [pulseaudio] alsa-util.c: subformat : STD
D: [pulseaudio] alsa-util.c: channels : 2
D: [pulseaudio] alsa-util.c: rate : 44100
D: [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1)
D: [pulseaudio] alsa-util.c: msbits : 16
D: [pulseaudio] alsa-util.c: buffer_size : 88200
D: [pulseaudio] alsa-util.c: period_size : 14700
D: [pulseaudio] alsa-util.c: period_time : 333333
D: [pulseaudio] alsa-util.c: tstamp_mode : ENABLE
D: [pulseaudio] alsa-util.c: tstamp_type : MONOTONIC
D: [pulseaudio] alsa-util.c: period_step : 1
D: [pulseaudio] alsa-util.c: avail_min : 87319
D: [pulseaudio] alsa-util.c: period_event : 0
D: [pulseaudio] alsa-util.c: start_threshold : -1
D: [pulseaudio] alsa-util.c: stop_threshold : 6206523236469964800
D: [pulseaudio] alsa-util.c: silence_threshold: 0
D: [pulseaudio] alsa-util.c: silence_size : 0
D: [pulseaudio] alsa-util.c: boundary : 6206523236469964800
D: [pulseaudio] alsa-util.c: appl_ptr : 0
D: [pulseaudio] alsa-util.c: hw_ptr : 0

Hi Mitesh,

I think that's because the channel count is stereo, not mono. you should change a wav file, 

Thanks

Hi Mitesh,

could you exit the root? From the log, I didn't see the modification works, 

pcm.rate_convert {
	type rate
	slave {
		pcm "hw:0,0"
		rate 48000
	}
}

Kindly set asoundrc as above and see if it can works.

Thanks

Hi Kevin, we don't have any other user on the system just root.

What are the steps to make sure we are using the .asoundrc file ?

All I am doing is moving the file to ~/.asoundrc on the system and rebooting the system. 

root@am62xx-evm:~# aplay /usr/share/sounds/alsa/Front_Center.wav
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono

root@am62xx-evm:~# aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=AM62xSKEVM
AM62x-SKEVM, davinci-mcasp.0-tlv320aic31xx-hifi tlv320aic31xx-hifi-0
Default Audio Device
sysdefault:CARD=AM62xSKEVM
AM62x-SKEVM, davinci-mcasp.0-tlv320aic31xx-hifi tlv320aic31xx-hifi-0
Default Audio Device

root@am62xx-evm:~# arecord -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=AM62xSKEVM
AM62x-SKEVM, davinci-mcasp.0-tlv320aic31xx-hifi tlv320aic31xx-hifi-0
Default Audio Device
sysdefault:CARD=AM62xSKEVM
AM62x-SKEVM, davinci-mcasp.0-tlv320aic31xx-hifi tlv320aic31xx-hifi-0
Default Audio Device
root@am62xx-evm:~#

It seems that the error log never happens again, is there a sound playing?

Thanks

Hi Kevin, no really. when I use the pulseaudio and g-streamer, I still get error to say, hardware sound card can only work at 44100 and not 48000.

Thanks 

Hi Mitesh,

If that so, I think the issue is caused by the truly clock, you see the wclk is 45.1KHz, it will cause clock error, could you send out the registers dump during playback? we can locate whether this is an issue caused by the register configuration.

Thanks

Hi Kevin, I shall try to send you the reg dump today, how would you change the buffer size please? 

Hi Mitesh,

Same as the method of changing sample rate, you can add  buffer_size = 4096(or bigger) in .asoundrc, 

Thanks

Hi Kevin, attached is the reg dump after aplay. Thanks 

root@am62xx-evm:~# aplay /usr/share/sounds/alsa/Front_Center.wav
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono

root@am62xx-evm:~# i2cdump -y -f 1 0x18
No size specified (using byte-data access)
0 1 2 3 4 5 6 7 8 9 a b c d e f 0123456789abcdef
00: 00 00 01 56 03 91 08 07 80 00 00 88 82 00 80 80 ..?V?????..??.??
10: 08 00 88 82 80 80 04 00 00 00 01 4c 00 04 88 00 ?.?????...?L.??.
20: 00 00 00 00 80 bb 11 00 00 00 00 00 00 00 00 00 ....???.........
30: 00 00 00 02 32 12 03 02 02 11 10 00 01 04 00 d4 ...?2??????.??.?
40: 0c 00 00 00 6f 38 00 00 00 00 00 ee 10 d8 7e e3 ?...o8.....???~?
50: 00 00 00 00 00 00 a0 fe 50 30 20 00 00 00 00 00 ......??P0 .....
60: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
70: 00 00 00 00 00 04 00 00 00 00 00 00 00 00 00 00 .....?..........
80: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
90: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
a0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
b0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
c0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
d0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
e0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
root@am62xx-evm:~#

Thanks Kevin. I will check that.

But not sure why did the change of sampling rate and adding min buffer size doesn't work when I use the suggested asoundrc settings please. 

Hi Mitesh,

Sorry for late, I think the sample rate depends on whether the hardware can meet the setting, and the buffer size should work if it's within tolerance.

Thanks