VoIP example with 16Khz HAL i2s clock: AV_STREAM: audio decoder ringbuf write timeout (AUD-5988)
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AV_STREAM: audio decoder ringbuf write timeoutAV_STREAM: audio decoder ringbuf write timeout (AUD-5988)
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For G711 almost use 8K sample rate, if play tone use 16K and SIP call use 8K, better use following config: av_stream_config_t av_stream_config = {
.algo_mask = ALGORITHM_STREAM_DEFAULT_MASK,
.acodec_samplerate = 8000,
.acodec_type = AV_ACODEC_G711A,
.vcodec_type = AV_VCODEC_NULL,
.hal = {
.audio_samplerate = 16000,
.audio_framesize = PCM_FRAME_SIZE,
},
};
av_stream = av_stream_init(&av_stream_config); |
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@TempoTian Thank you for the response. Yes, this is what I originally did. I set the If you download the zip file I posted, you will see that I changed to I agree that this should work, but it does not. There is a problem somewhere. Have you tried it yourself? Using your own suggestion you should get the same error. |
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I have test the av stream, it works all right if input 8k, output 16k. |
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Just to be clear: the error only occurs when you make a SIP call. Are you saying you made a SIP call using 16k HAL rate without getting errors like I showed? In the zip file I gave, the only real change is So the code I gave: In
And: So in the code I provided, the tone player rate and the |
Environment
Problem Description
Using the
voip_example, settingav_stream_config.hal.audio_samplerate = 16000causes write timeouts and watchdogs.The reason I want to do this is because I have stored audio files recorded at 16kHz and I want to play them at 16kHz. I also need to support VoIP SIP calls running at 8kHz.
In IDF 4.x this worked, the i2s would run at 16kHz and a filter would be used in the SIP pipeline to convert the microphone to 8kHz out to SIP, and incoming SIP audio up to 16kHz to play out the speaker.
I haven't been able to get this setup to work in IDF v5 with the new SIP example that uses the
av_streamcomponent and the newesp_rtc_APIs.Expected Behavior
The
av_streamcomponent has logic to filter between theacodec_samplerateand the.hal.audio_samplerate. So I would expect to be able to play stored audio at 16kHz when not on a SIP call, and when not playing stored audio connect to SIP calls at 8kHz.This works in IDF 4, but there is no direct translation of code to IDF 5 because if the SIP refactoring. I know the hardware can do it, and the
av_streamcode appears to be able to handle filtering to down/up sample, but it doesn't work.Actual Behavior
It does not work.
Steps to Reproduce
3.1 The voip example doesn't work out of the box. The only way I could get the legacy driver stuff sorted out is to remove all the video functions and esp32-camera dependency from
av_stream(wich uses esp32-camera). I only need audio stuff anyway. I did this by making a local copy of the components and editing them.components/av_stream/av_stream_hal/av_stream_hal.h, defineAUDIO_HAL_SAMPLE_RATEas16000(leave codec rate at 8kHz)Code to Reproduce This Issue
See attached:
esp32_adf_voip_16k_crash.zip
Debug Logs
Full logs: voip-sip-16k-wdog.txt
Snippet:
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