on
Called for every 10 ms audio buffer captured from the microphone, before it is handed to the WebRTC engine.
The byteBuffer may be modified in-place to transform the audio before it reaches WebRTC (e.g. gain adjustment, filtering, ring modulation).
The buffer contains interleaved PCM samples in the format and sample rate described by record. For android.media.AudioFormat.ENCODING_PCM_16BIT, each sample is a little-endian Short; use byteBuffer.order(ByteOrder.nativeOrder()).asShortBuffer() to iterate samples conveniently.
Return
true to pass the buffer contents (as-is or modified) to WebRTC; false to discard the buffer and send silence for this frame instead.
Parameters
Recording configuration in use.
Number of bytes populated in byteBuffer.
Buffer containing the raw PCM samples. Safe to read and modify in-place.