I am making an Android-to-Android VoIP (loudspeaker) app using its AudioRecord and AudioTrack class, along with Speex via NDK to do echo cancellation. I was able to successfully pass into and retrieve data from Speex's speex_echo_cancellation() function, but the echo remains.
Here is the relevant android thread code that is recording/sending and receiving/playing audio:
//constructor
public MyThread(DatagramSocket socket, int frameSize, int filterLength){
this.socket = socket;
nativeMethod_initEchoState(frameSize, filterLength);
}
public void run(){
short[] audioShorts, recvShorts, recordedShorts, filteredShorts;
byte[] audioBytes, recvBytes;
int shortsRead;
DatagramPacket packet;
//initialize recorder and player
int samplingRate = 8000;
int managerBufferSize = 2000;
AudioTrack player = new AudioTrack(AudioManager.STREAM_MUSIC, samplingRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, managerBufferSize, AudioTrack.MODE_STREAM);
recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, samplingRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, managerBufferSize);
recorder.startRecording();
player.play();
//record first packet
audioShorts = new short[1000];
shortsRead = recorder.read(audioShorts, 0, audioShorts.length);
//convert shorts to bytes to send
audioBytes = new byte[shortsRead*2];
ByteBuffer.wrap(audioBytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().put(audioShorts);
//send bytes
packet = new DatagramPacket(audioBytes, audioBytes.length);
socket.send(packet);
while (!this.isInterrupted()){
//recieve packet/bytes (received audio data should have echo cancelled already)
recvBytes = new byte[2000];
packet = new DatagramPacket(recvBytes, recvBytes.length);
socket.receive(packet);
//convert bytes to shorts
recvShorts = new short[packet.getLength()/2];
ByteBuffer.wrap(packet.getData(), 0, packet.getLength()).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(recvShorts);
//play shorts
player.write(recvShorts, 0, recvShorts.length);
//record shorts
recordedShorts = new short[1000];
shortsRead = recorder.read(recordedShorts, 0, recordedShorts.length);
//send played and recorded shorts into speex,
//returning audio data with the echo removed
filteredShorts = nativeMethod_speexEchoCancel(recordedShorts, recvShorts);
//convert filtered shorts to bytes
audioBytes = new byte[shortsRead*2];
ByteBuffer.wrap(audioBytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().put(filteredShorts);
//send off bytes
packet = new DatagramPacket(audioBytes, audioBytes.length);
socket.send(packet);
}//end of while loop
}
Here is the relevant NDK / JNI code:
void nativeMethod_initEchoState(JNIEnv *env, jobject jobj, jint frameSize, jint filterLength){
echo_state = speex_echo_state_init(frameSize, filterLength);
}
jshortArray nativeMethod_speexEchoCancel(JNIEnv *env, jobject jObj, jshortArray input_frame, jshortArray echo_frame){
//create native shorts from java shorts
jshort *native_input_frame = (*env)->GetShortArrayElements(env, input_frame, NULL);
jshort *native_echo_frame = (*env)->GetShortArrayElements(env, echo_frame, NULL);
//allocate memory for output data
jint length = (*env)->GetArrayLength(env, input_frame);
jshortArray temp = (*env)->NewShortArray(env, length);
jshort *native_output_frame = (*env)->GetShortArrayElements(env, temp, 0);
//call echo cancellation
speex_echo_cancellation(echo_state, native_input_frame, native_echo_frame, native_output_frame);
//convert native output to java layer output
jshortArray output_shorts = (*env)->NewShortArray(env, length);
(*env)->SetShortArrayRegion(env, output_shorts, 0, length, native_output_frame);
//cleanup and return
(*env)->ReleaseShortArrayElements(env, input_frame, native_input_frame, 0);
(*env)->ReleaseShortArrayElements(env, echo_frame, native_echo_frame, 0);
(*env)->ReleaseShortArrayElements(env, temp, native_output_frame, 0);
return output_shorts;
}
These code runs fine and audio data is definitely being sent/received/processed/played from android-to-android. Given audio sample rate of 8000 Hz and packet size of 2000bytes/1000shorts, I've found that a frameSize of 1000 is needed in order for the played audio to be smooth. Most value of filterLength (aka tail length according to Speex doc) will run, but seems to have no effect on the echo removal.
Does anyone understand enough AEC as to provide me some pointers on implementing or configuring Speex? Thanks for reading.
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