之前使用speex做回音消除，不知道是没有用对还是其他什么原因，导致回音消除的效果非常差。
好在在GitHub找到了一个使用WebRTC做回音消除的例子Android-Audio-Processing-Using-WebRTC
。
经过一顿改吧改吧的，终于在自己的项目中使用成功。 回音消除的效果比之前好了不止百倍。
/**
* AudioTrack音频播放,录音
*
* @author Robbie
*/
public class Tracker extends JobHandler {
private static AudioTrack audioTrack;
private Apm _apm;
private ApmViewModel vm;
private AudioRecord audioRecord;
int framSize = 80;
// 音频大小
private int inAudioBufferSize;
// 音频大小
private static int outAudioBufferSize;
// 播放标志
private boolean isPlaying = true;
public Tracker(Handler handler) {
super(handler);
outAudioBufferSize = Constants.inAudioBufferSize;
inAudioBufferSize =Constants.inAudioBufferSize;
this.init();
initAudioTrace(AudioManager.AUDIO_SESSION_ID_GENERATE);
}
public static void initAudioTrace(int sessionId) {
audioTrack = new AudioTrack((new AudioAttributes.Builder())
.setLegacyStreamType(Constants.streamType)
.build(),
(new AudioFormat.Builder())
.setChannelMask(Constants.outputChannelConfig)
.setEncoding(Constants.audioFormat)
.setSampleRate(Constants.sampleRateInHz)
.build(),
outAudioBufferSize,
Constants.trackMode, sessionId);
}
public boolean isPlaying() {
return isPlaying;
}
public void setPlaying(boolean playing) {
isPlaying = playing;
}
public void init() {
// 初始化音频录制
audioRecord = new AudioRecord(Constants.audioSource,
Constants.sampleRateInHz, Constants.inputChannelConfig, Constants.audioFormat, inAudioBufferSize);
}
private void initApm() throws Exception {
int ret;
vm = ApmViewModel.getInstance();
try {
_apm = new Apm(vm.getAecExtendFilter(), vm.getSpeechIntelligibilityEnhance(), vm.getDelayAgnostic(), vm.getBeamForming(),
vm.getNextGenerationAEC(), vm.getExperimentalNS(), vm.getExperimentalAGC());
ret = _apm.HighPassFilter(vm.getHighPassFilter());
if (vm.getAecPC()) {
ret = _apm.AECClockDriftCompensation(false);
ret = _apm.AECSetSuppressionLevel(Apm.AEC_SuppressionLevel.values()[2]);
ret = _apm.AEC(true);
} else if (vm.getAecMobile()) {
ret = _apm.AECMSetSuppressionLevel(Apm.AECM_RoutingMode.values()[2]);
ret = _apm.AECM(true);
}
ret = _apm.NSSetLevel(Apm.NS_Level.values()[1]);
ret = _apm.NS(vm.getNs());
ret = _apm.VAD(vm.getVad());
if (vm.getAgc()) {
ret = _apm.AGCSetAnalogLevelLimits(0, 255);
ret = _apm.AGCSetMode(Apm.AGC_Mode.values()[2]);
ret = _apm.AGCSetTargetLevelDbfs(vm.getAgcTargetLevelInt());
ret = _apm.AGCSetcompressionGainDb(vm.getAgcCompressionGainInt());
ret = _apm.AGCEnableLimiter(true);
ret = _apm.AGC(true);
}
} catch (Exception ex) {
// new AlertDialog.Builder(context).setTitle("System hint")
// .setMessage(ex.getMessage())
// .show();
return;
}
}
@Override
public void run() {
AudioData currentAudioData;
try {
initApm();
} catch (Exception e) {
e.printStackTrace();
}
while ((currentAudioData = MessageQueue.getInstance(MessageQueue.TRACKER_DATA_QUEUE).take()) != null) {
if(currentAudioData.isClose()){
MessageQueue.getInstance(MessageQueue.TRACKER_DATA_QUEUE).clear();
break;
}
if (!IntercomService.connecting) {
break;
}
if (audioTrack == null) {
continue;
}
try {
if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) {
audioTrack.play();
}
short[] bytesPkg = currentAudioData.getRawData();
if (bytesPkg != null) {
for(int i = 0 ; i < bytesPkg.length/framSize ; i++) {
try {
_apm.ProcessRenderStream(bytesPkg, i * framSize);
} catch (Exception e) {
e.printStackTrace();
}
}
audioTrack.write(bytesPkg, 0, bytesPkg.length);
}
this.reacord(bytesPkg);
} catch (Exception e) {
e.printStackTrace();
}
}
}
private void reacord(short[] bytesPkg) throws Exception {
if(audioRecord == null){
this.init();
}
if (audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_STOPPED) {
audioRecord.startRecording();
}
AudioData audioData = new AudioData();
// 实例化音频数据缓冲
short[] rawData = new short[inAudioBufferSize / 2];
int size = audioRecord.read(rawData, 0, inAudioBufferSize / 2);
if (bytesPkg != null) {
int out_analog_level = 200;
for(int i = 0 ; i < rawData.length/framSize ; i++) {
short delay = Short.valueOf(ApmViewModel.getInstance().getAceBufferDelay());
_apm.SetStreamDelay(delay);
try {
if (vm.getAgc()) {
_apm.AGCSetStreamAnalogLevel(out_analog_level);
}
_apm.ProcessCaptureStream(rawData,i*framSize);
if (vm.getAgc()) {
out_analog_level = _apm.AGCStreamAnalogLevel();
Log.i("AGC", out_analog_level + "");
}
if (vm.getVad()) {
if (!_apm.VADHasVoice()) continue;
}
} catch (Exception e) {
e.printStackTrace();
}
}
audioData.setRawData(rawData);
} else {
audioData.setRawData(rawData);
}
audioData.setClose(!IntercomService.connecting);
MessageQueue.getInstance(MessageQueue.ENCODER_DATA_QUEUE).put(audioData);
}
}
其中的 Apm和ApmViewModel照抄Demo的。
但是回音消除的部分使用Demo中的方法，导致声音出现问题。
播放声音之前执行：
int framSize = 80
for(int i = 0 ; i < bytesPkg.length/framSize ; i++) {
try {
_apm.ProcessRenderStream(bytesPkg, i * framSize);
} catch (Exception e) {
e.printStackTrace();
}
}
录音之后执行：
int framSize = 80
int out_analog_level = 200;
for(int i = 0 ; i < rawData.length/framSize ; i++) {
short delay = Short.valueOf(ApmViewModel.getInstance().getAceBufferDelay());
_apm.SetStreamDelay(delay);
try {
if (vm.getAgc()) {
_apm.AGCSetStreamAnalogLevel(out_analog_level);
}
_apm.ProcessCaptureStream(rawData,i*framSize);
if (vm.getAgc()) {
out_analog_level = _apm.AGCStreamAnalogLevel();
Log.i("AGC", out_analog_level + "");
}
if (vm.getVad()) {
if (!_apm.VADHasVoice()) continue;
}
} catch (Exception e) {
e.printStackTrace();
}
}
至于Android中使用到的库文件，可以按照GitHub上的步骤编译。
