任何声学回声消除（AEC）库能够达到48 kHz？

Question

I'm developing a VoIP application that runs at the sampling rate of 48 kHz. Since it uses Opus, which uses 48 kHz internally, as its codec, and most current Android hardware natively runs at 48 kHz, AEC is the only piece of the puzzle I'm missing now. I've already found the WebRTC implementation but I can't seem to figure out how to make it work. It looks like it corrupts the memory randomly and crashes the whole thing sooner or later. When it doesn't crash, the sound is kinda chunky as if it's quieter for the half of the frame. Here's my code that processes a 20 ms frame:

webrtc::SplittingFilter* splittingFilter;
webrtc::IFChannelBuffer* bufferIn;
webrtc::IFChannelBuffer* bufferOut;
webrtc::IFChannelBuffer* bufferOut2;
// ...
splittingFilter=new webrtc::SplittingFilter(1, 3, 960);
bufferIn=new webrtc::IFChannelBuffer(960, 1, 1);
bufferOut=new webrtc::IFChannelBuffer(960, 1, 3);
bufferOut2=new webrtc::IFChannelBuffer(960, 1, 3);
// ...
int16_t* samples=(int16_t*)data;
float* fsamples[3];
float* foutput[3];
int i;
float* fbuf=bufferIn->fbuf()->bands(0)[0];
// convert the data from 16-bit PCM into float
for(i=0;i<960;i++){
    fbuf[i]=samples[i]/(float)32767;
}
// split it into three "bands" that the AEC needs and for some reason can't do itself
splittingFilter->Analysis(bufferIn, bufferOut);
// split the frame into 6 consecutive 160-sample blocks and perform AEC on them
for(i=0;i<6;i++){
    fsamples[0]=&bufferOut->fbuf()->bands(0)[0][160*i];
    fsamples[1]=&bufferOut->fbuf()->bands(0)[1][160*i];
    fsamples[2]=&bufferOut->fbuf()->bands(0)[2][160*i];
    foutput[0]=&bufferOut2->fbuf()->bands(0)[0][160*i];
    foutput[1]=&bufferOut2->fbuf()->bands(0)[1][160*i];
    foutput[2]=&bufferOut2->fbuf()->bands(0)[2][160*i];
    int32_t res=WebRtcAec_Process(aecState, (const float* const*) fsamples, 3, foutput, 160, 20, 0);
}
// put the "bands" back together
splittingFilter->Synthesis(bufferOut2, bufferIn);
// convert the processed data back into 16-bit PCM
for(i=0;i<960;i++){
    samples[i]=(int16_t) (CLAMP(fbuf[i], -1, 1)*32767);
}

If I comment out the actual echo cancellation and just do the float conversion and band splitting back and forth, it doesn't corrupt the memory, doesn't sound weird and runs indefinitely. (I do pass the farend/speaker signal into AEC, I just didn't want to make the mess of my code by including it in the question)

I've also tried Android's built-in AEC. While it does work, it upsamples the captured signal from 16 kHz.

Answer 1

Unfortunately, there is no free AEC package that support 48khz. So, either move to 32khz or use a commercial AEC package at 48khz.