Webrtc Aecm for Android

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I'm developing a Sip client for android and i want to implement the Aecm module from webrtc but i don't understand how to do properly. I've 2 runnable class, one for AndroidRecord (microphone) and one for AudioTrack (speaker), so i don't know how and where i've to call that api :

int32_t WebRtcAec_BufferFarend(void *aecInst, const int16_t *farend, int16_t nrOfSamples);
int32_t WebRtcAec_Process(void *aecInst, const int16_t *nearend, const int16_t *nearendH, int16_t *out, int16_t *outH, int16_t nrOfSamples, int16_t msInSndCardBuf, int32_t skew);

BufferFarend is the AudioTrack buffer? and the buffer that will be written on AudioRecord have to pass into "WebRtcAec_Process" ?

asked 59 secs ago

user2418939

1

Webrtc Aecm for Android