Why is WebRtc AEC3 giving Flat out MicOutPut after Echo Cancellation?

[Kazi Mehedi Hasan]

Hello All,

Let me begin this topic by stating that I am a total newbie to WebRtc and if any I mention anything half-witty, please bear with me in

a forgivable manner.

I am writing an app, that does echo cancellation performance comparison between Speex and Web RTC AEC3.
[WebRtc AEC3 code base: https://webrtc.googlesource.com/src/+/branch-heads/72]

The app reads WAV files and feeds the samples to the AEC module, and a WAV writer saves the output of Echo cancellation,

I have 2 inputs:

1) Speaker Input or Rendered Signal or FarEnd Signal 

2) MicInput or Captured Signal or NearEnd Signal

And one Output:
1) MicOutput- Which is the result of Echo cancellation.

Now for Speex modules, I see a well behaved manner. Please have a look at the attached file, its doing a good job in cancelling the rendered signal from

Captured Signal.

However, when I am passing the same files with WebRtc Aec3, I am getting a flat out signal. Attached is also the result of AEC3.

It seems like it is also cancelling out the original mic signal too.

I am using the following parameters(extracted from Wav file reader):
Sample rate = 8000

Channel = 1

Bits/Sample = 16

NumOfSamples = 270399

Samples Fed to AEC at a time = (10*SampleRate)/1000 = 80

This is the initialization:

    

    m_streamConfig.set_sample_rate_hz(sampleRate);

    m_streamConfig.set_num_channels(CHANNEL_COUNT);

    // Create a temporary buffer to convert our RTOP input audio data into the webRTC required AudioBuffer. 

    m_tempBuffer[0] = static_cast<float*> (malloc(sizeof(float) * m_samplesPerBlock));

    // Create AEC3. 

    m_echoCanceller3.reset(new EchoCanceller3(m_echoCanceller3Config, sampleRate, true)); //use high pass filter is true

    // Create noise suppression.

    m_noiseSuppression.reset(new NoiseSuppressionImpl(&m_criticalSection));

    m_noiseSuppression->Initialize(CHANNEL_COUNT, sampleRate);

And this is how I am calling the APIs:

//auto capturedAudioBuffer = CreateAudioBuffer(&micSamples[index]);

        auto renderAudioBuffer = CreateAudioBuffer(spkSamples);

        auto capturedAudioBuffer = CreateAudioBuffer(micSamples);

        // Analyze capture buffer

        m_echoCanceller3->AnalyzeCapture(capturedAudioBuffer.get());

        // Analyze render buffer

        m_echoCanceller3->AnalyzeRender(renderAudioBuffer.get());

        // Cancel echo

        m_echoCanceller3->ProcessCapture(

            capturedAudioBuffer.get(), false); // Assuming the analog level is not changed.  If we want to detect change, need to use gain controller and remember the previously rendered audio's analog level

        // Copy the Captured audio out 

        capturedAudioBuffer->CopyTo(m_streamConfig, m_tempBuffer);

        //arrayCopy_32f(m_tempBuffer[0], &micOut[index], m_samplesPerBlock);

        arrayCopy_32f(m_tempBuffer[0], micOut, m_samplesPerBlock);

And also regarding the parameters (delay, echoModel, reverb, noisefloor etc.) , I am using all default values.

Can anyone tell me what I am doing wrong? Or how can I make it better by adjusting the appropriate parameters?

Thanks.

-Kazi

[Kazi Mehedi Hasan]

UpDate:(02/22/2019)

Figured out why is Echo Output muted. Seems like Webrtc AEC3 cannot process 8k and 16k sampling rate, although in source code there are indication they support 4 different sampling rate: 8k, 16k, 32k and 48k.

I got an echo cancelled output after I gave input of 32k and 48k Samples. However, I do not see any echo cancellation. It just spits out the exact samples as it was fed in for NearEnd/Mic/Captured input. So yeah, probably I am missing key parameter settings. Still looking for help.

[Vitaly Ivanov]

I'd try plugging your data into one of AEC3 unit tests instead of writing your own app. Maybe just for a start, to get the thing working

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[Kazi Mehedi Hasan]

Okay, thanks for the feedback Vitaly. I would try it. Thanks.