Android音频编解码和混音实现
原文链接：http://my.oschina.net/daxia/blog/636074
相关源码:https://github.com/YeDaxia/MusicPlus
认识数字音频:
在实现之前，我们先来了解一下数字音频的有关属性。
采样频率(Sample Rate)：每秒采集声音的数量，它用赫兹(Hz)来表示。(采样率越高越靠近原声音的波形)
采样精度(Bit Depth)：指记录声音的动态范围，它以位(Bit)为单位。(声音的幅度差)
声音通道(Channel)：声道数。比如左声道右声道。
采样量化后的音频最终是一串数字,声音的大小(幅度)会体现在这个每个数字数值大小上；而声音的高低(频率)和声音的音色(Timbre)都和时间维度有关，会体现在数字之间的差异上。
在编码解码之前，我们先来感受一下原始的音频数据究竟是什么样的。我们知道wav文件里面放的就是原始的PCM数据，下面我们通过AudioTrack来直接把这些数据write进去播放出来。下面是某个wav文件的格式，关于wav的格式内容可以看:http://soundfile.sapp.org/doc/WaveFormat/ ，可以通过Binary Viewer等工具去查看一下wav文件的二进制内容。
播放wav文件:
int sampleRateInHz = 44100;
int channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
int bufferSizeInBytes = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, audioFormat);
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, audioFormat, bufferSizeInBytes, AudioTrack.MODE_STREAM);
audioTrack.play();
FileInputStream audioInput = null;
try {
audioInput = new FileInputStream(audioFile);//put your wav file in
audioInput.read(new byte[44]);//skid 44 wav header
byte[] audioData = new byte[512];
while(audioInput.read(audioData)!= -1){
audioTrack.write(audioData, 0, audioData.length); //play raw audio bytes
}
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}finally{
audioTrack.stop();
audioTrack.release();
if(audioInput != null)
try {
audioInput.close();
} catch (IOException e) {
e.printStackTrace();
}
}如果你有试过一下上面的例子，那你应该对音频的源数据有了一个概念了。
音频的解码:
通过上面的介绍，我们不难知道，解码的目的就是让编码后的数据恢复成wav中的源数据。
利用MediaExtractor和MediaCodec来提取编码后的音频数据并解压成音频源数据:
final String encodeFile = "your encode audio file path";
MediaExtractor extractor = new MediaExtractor();
extractor.setDataSource(encodeFile);
MediaFormat mediaFormat = null;
for (int i = 0; i < extractor.getTrackCount(); i++) {
MediaFormat format = extractor.getTrackFormat(i);
String mime = format.getString(MediaFormat.KEY_MIME);
if (mime.startsWith("audio/")) {
extractor.selectTrack(i);
mediaFormat = format;
break;
}
}
if(mediaFormat == null){
DLog.e("not a valid file with audio track..");
extractor.release();
return null;
}
FileOutputStream fosDecoder = new FileOutputStream(outDecodeFile);//your out file path
String mediaMime = mediaFormat.getString(MediaFormat.KEY_MIME);
MediaCodec codec = MediaCodec.createDecoderByType(mediaMime);
codec.configure(mediaFormat, null, null, 0);
codec.start();
ByteBuffer[] codecInputBuffers = codec.getInputBuffers();
ByteBuffer[] codecOutputBuffers = codec.getOutputBuffers();
final long kTimeOutUs = 5000;
MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
boolean sawInputEOS = false;
boolean sawOutputEOS = false;
int totalRawSize = 0;
try{
while (!sawOutputEOS) {
if (!sawInputEOS) {
int inputBufIndex = codec.dequeueInputBuffer(kTimeOutUs);
if (inputBufIndex >= 0) {
ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];
int sampleSize = extractor.readSampleData(dstBuf, 0);
if (sampleSize < 0) {
DLog.i(TAG, "saw input EOS.");
sawInputEOS = true;
codec.queueInputBuffer(inputBufIndex,0,0,0,MediaCodec.BUFFER_FLAG_END_OF_STREAM );
} else {
long presentationTimeUs = extractor.getSampleTime();
codec.queueInputBuffer(inputBufIndex,0,sampleSize,presentationTimeUs,0);
extractor.advance();
}
}
}
int res = codec.dequeueOutputBuffer(info, kTimeOutUs);
if (res >= 0) {
int outputBufIndex = res;
// Simply ignore codec config buffers.
if ((info.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG)!= 0) {
DLog.i(TAG, "audio encoder: codec config buffer");
codec.releaseOutputBuffer(outputBufIndex, false);
continue;
}
if(info.size != 0){
ByteBuffer outBuf = codecOutputBuffers[outputBufIndex];
outBuf.position(info.offset);
outBuf.limit(info.offset + info.size);
byte[] data = new byte[info.size];
outBuf.get(data);
totalRawSize += data.length;
fosDecoder.write(data);
}
codec.releaseOutputBuffer(outputBufIndex, false);
if ((info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
DLog.i(TAG, "saw output EOS.");
sawOutputEOS = true;
}
} else if (res == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
codecOutputBuffers = codec.getOutputBuffers();
DLog.i(TAG, "output buffers have changed.");
} else if (res == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat oformat = codec.getOutputFormat();
DLog.i(TAG, "output format has changed to " + oformat);
}
}
}finally{
fosDecoder.close();
codec.stop();
codec.release();
extractor.release();
}解压之后，可以用AudioTrack来播放验证一下这些数据是否正确。
音频的混音:
音频混音的原理: 量化的语音信号的叠加等价于空气中声波的叠加。
反应到音频数据上，也就是把同一个声道的数值进行简单的相加，但是这样同时会产生一个问题，那就是相加的结果可能会溢出，当然为了解决这个问题已经有很多方案了，在这里我们采用简单的平均算法(average audio mixing algorithm, 简称V算法)。在下面的演示程序中，我们假设音频文件是的采样率，通道和采样精度都是一样的,这样会便于处理。另外要注意的是，在源音频数据中是按照little-endian的顺序来排放的，PCM值为0表示没声音(振幅为0)。
public void mixAudios(File[] rawAudioFiles){
final int fileSize = rawAudioFiles.length;
FileInputStream[] audioFileStreams = new FileInputStream[fileSize];
File audioFile = null;
FileInputStream inputStream;
byte[][] allAudioBytes = new byte[fileSize][];
boolean[] streamDoneArray = new boolean[fileSize];
byte[] buffer = new byte[512];
int offset;
try {
for (int fileIndex = 0; fileIndex < fileSize; ++fileIndex) {
audioFile = rawAudioFiles[fileIndex];
audioFileStreams[fileIndex] = new FileInputStream(audioFile);
}
while(true){
for(int streamIndex = 0 ; streamIndex < fileSize ; ++streamIndex){
inputStream = audioFileStreams[streamIndex];
if(!streamDoneArray[streamIndex] && (offset = inputStream.read(buffer)) != -1){
allAudioBytes[streamIndex] = Arrays.copyOf(buffer,buffer.length);
}else{
streamDoneArray[streamIndex] = true;
allAudioBytes[streamIndex] = new byte[512];
}
}
byte[] mixBytes = mixRawAudioBytes(allAudioBytes);
//mixBytes 就是混合后的数据
boolean done = true;
for(boolean streamEnd : streamDoneArray){
if(!streamEnd){
done = false;
}
}
if(done){
break;
}
}
} catch (IOException e) {
e.printStackTrace();
if(mOnAudioMixListener != null)
mOnAudioMixListener.onMixError(1);
}finally{
try {
for(FileInputStream in : audioFileStreams){
if(in != null)
in.close();
}
} catch (IOException e) {
e.printStackTrace();
}
}
}
/**
* 每一行是一个音频的数据
*/
byte[] averageMix(byte[][] bMulRoadAudioes) {
if (bMulRoadAudioes == null || bMulRoadAudioes.length == 0)
return null;
byte[] realMixAudio = bMulRoadAudioes[0];
if(bMulRoadAudioes.length == 1)
return realMixAudio;
for(int rw = 0 ; rw < bMulRoadAudioes.length ; ++rw){
if(bMulRoadAudioes[rw].length != realMixAudio.length){
Log.e("app", "column of the road of audio + " + rw +" is diffrent.");
return null;
}
}
int row = bMulRoadAudioes.length;
int coloum = realMixAudio.length / 2;
short[][] sMulRoadAudioes = new short[row][coloum];
for (int r = 0; r < row; ++r) {
for (int c = 0; c < coloum; ++c) {
sMulRoadAudioes[r][c] = (short) ((bMulRoadAudioes[r][c * 2] & 0xff) | (bMulRoadAudioes[r][c * 2 + 1] & 0xff) << 8);
}
}
short[] sMixAudio = new short[coloum];
int mixVal;
int sr = 0;
for (int sc = 0; sc < coloum; ++sc) {
mixVal = 0;
sr = 0;
for (; sr < row; ++sr) {
mixVal += sMulRoadAudioes[sr][sc];
}
sMixAudio[sc] = (short) (mixVal / row);
}
for (sr = 0; sr < coloum; ++sr) {
realMixAudio[sr * 2] = (byte) (sMixAudio[sr] & 0x00FF);
realMixAudio[sr * 2 + 1] = (byte) ((sMixAudio[sr] & 0xFF00) >> 8);
}
return realMixAudio;
}同样，你可以把混音后的数据用AudioTrack播放出来，验证一下混音的效果。
音频的编码:
对音频进行编码的目的用更少的空间来存储和传输，有有损编码和无损编码，其中我们常见的Mp3和ACC格式就是有损编码。在下面的例子中，我们通过MediaCodec来对混音后的数据进行编码，在这里，我们将采用ACC格式来进行。
ACC音频有ADIF和ADTS两种，第一种适用于磁盘，第二种则可以用于流的传输，它是一种帧序列。我们这里用ADTS这种来进行编码，首先要了解一下它的帧序列的构成:
ADTS的帧结构:
| header |
body |
ADTS帧的Header组成:
| Length (bits) | Description |
| 12 | syncword 0xFFF, all bits must be 1 |
| 1 | MPEG Version: 0 for MPEG-4, 1 for MPEG-2 |
| 2 | Layer: always 0 |
| 1 | protection absent, Warning, set to 1 if there is no CRC and 0 if there is CRC |
| 2 | profile, the MPEG-4 Audio Object Type minus 1 |
| 4 | MPEG-4 Sampling Frequency Index (15 is forbidden) |
| 1 | private bit, guaranteed never to be used by MPEG, set to 0 when encoding, ignore when decoding |
| 3 | MPEG-4 Channel Configuration (in the case of 0, the channel configuration is sent via an inband PCE) |
| 1 | originality, set to 0 when encoding, ignore when decoding |
| 1 | home, set to 0 when encoding, ignore when decoding |
| 1 | copyrighted id bit, the next bit of a centrally registered copyright identifier, set to 0 when encoding, ignore when decoding |
| 1 | copyright id start, signals that this frame's copyright id bit is the first bit of the copyright id, set to 0 when encoding, ignore when decoding |
| 13 | frame length, this value must include 7 or 9 bytes of header length: FrameLength = (ProtectionAbsent == 1 ? 7 : 9) + size(AACFrame) |
| 11 | Buffer fullness |
| 2 | Number of AAC frames (RDBs) in ADTS frame minus 1, for maximum compatibility always use 1 AAC frame per ADTS frame |
| 16 | CRC if protection absent is 0 |
我们的思路就很明确了，把编码后的每一帧数据加上header写到文件中，保存后的.acc文件应该是可以被播放器识别播放的。为了简单，我们还是假设之前生成的混音数据源的采样率是44100Hz，通道数是2，采样精度是16Bit。
把音频源数据编码成ACC格式完成源代码:
class AACAudioEncoder{
private final static String TAG = "AACAudioEncoder";
private final static String AUDIO_MIME = "audio/mp4a-latm";
private final static long audioBytesPerSample = 44100*16/8;
private String rawAudioFile；
AACAudioEncoder(String rawAudioFile) {
this.rawAudioFile = rawAudioFile;
}
@Override
public void encodeToFile(String outEncodeFile) {
FileInputStream fisRawAudio = null;
FileOutputStream fosAccAudio = null;
try {
fisRawAudio = new FileInputStream(rawAudioFile);
fosAccAudio = new FileOutputStream(outEncodeFile);
final MediaCodec audioEncoder = createACCAudioDecoder();
audioEncoder.start();
ByteBuffer[] audioInputBuffers = audioEncoder.getInputBuffers();
ByteBuffer[] audioOutputBuffers = audioEncoder.getOutputBuffers();
boolean sawInputEOS = false;
boolean sawOutputEOS = false;
long audioTimeUs = 0 ;
BufferInfo outBufferInfo = new BufferInfo();
boolean readRawAudioEOS = false;
byte[] rawInputBytes = new byte[4096];
int readRawAudioCount = 0;
int rawAudioSize = 0;
long lastAudioPresentationTimeUs = 0;
int inputBufIndex, outputBufIndex;
while(!sawOutputEOS){
if (!sawInputEOS) {
inputBufIndex = audioEncoder.dequeueInputBuffer(10000);
if (inputBufIndex >= 0) {
ByteBuffer inputBuffer = audioInputBuffers[inputBufIndex];
inputBuffer.clear();
int bufferSize = inputBuffer.remaining();
if(bufferSize != rawInputBytes.length){
rawInputBytes = new byte[bufferSize];
}
if(!readRawAudioEOS){
readRawAudioCount = fisRawAudio.read(rawInputBytes);
if(readRawAudioCount == -1){
readRawAudioEOS = true;
}
}
if(readRawAudioEOS){
audioEncoder.queueInputBuffer(inputBufIndex,0 , 0 , 0 ,MediaCodec.BUFFER_FLAG_END_OF_STREAM);
sawInputEOS = true;
}else{
inputBuffer.put(rawInputBytes, 0, readRawAudioCount);
rawAudioSize += readRawAudioCount;
audioEncoder.queueInputBuffer(inputBufIndex, 0, readRawAudioCount, audioTimeUs, 0);
audioTimeUs = (long) (1000000 * (rawAudioSize / 2.0) / audioBytesPerSample);
}
}
}
outputBufIndex = audioEncoder.dequeueOutputBuffer(outBufferInfo, 10000);
if(outputBufIndex >= 0){
// Simply ignore codec config buffers.
if ((outBufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG)!= 0) {
DLog.i(TAG, "audio encoder: codec config buffer");
audioEncoder.releaseOutputBuffer(outputBufIndex, false);
continue;
}
if(outBufferInfo.size != 0){
ByteBuffer outBuffer = audioOutputBuffers[outputBufIndex];
outBuffer.position(outBufferInfo.offset);
outBuffer.limit(outBufferInfo.offset + outBufferInfo.size);
DLog.i(TAG, String.format(" writing audio sample : size=%s , presentationTimeUs=%s", outBufferInfo.size, outBufferInfo.presentationTimeUs));
if(lastAudioPresentationTimeUs < outBufferInfo.presentationTimeUs){
lastAudioPresentationTimeUs = outBufferInfo.presentationTimeUs;
int outBufSize = outBufferInfo.size;
int outPacketSize = outBufSize + 7;
outBuffer.position(outBufferInfo.offset);
outBuffer.limit(outBufferInfo.offset + outBufSize);
byte[] outData = new byte[outBufSize + 7];
addADTStoPacket(outData, outPacketSize);
outBuffer.get(outData, 7, outBufSize);
fosAccAudio.write(outData, 0, outData.length);
DLog.i(TAG, outData.length + " bytes written.");
}else{
DLog.e(TAG, "error sample! its presentationTimeUs should not lower than before.");
}
}
audioEncoder.releaseOutputBuffer(outputBufIndex, false);
if ((outBufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
sawOutputEOS = true;
}
}else if (outputBufIndex == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
audioOutputBuffers = audioEncoder.getOutputBuffers();
} else if (outputBufIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat audioFormat = audioEncoder.getOutputFormat();
DLog.i(TAG, "format change : "+ audioFormat);
}
}
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
} finally {
try {
if (fisRawAudio != null)
fisRawAudio.close();
if(fosAccAudio != null)
fosAccAudio.close();
} catch (IOException e) {
e.printStackTrace();
}
}
}
private MediaCodec createACCAudioDecoder() throws IOException {
MediaCodec codec = MediaCodec.createEncoderByType(AUDIO_MIME);
MediaFormat format = new MediaFormat();
format.setString(MediaFormat.KEY_MIME, AUDIO_MIME);
format.setInteger(MediaFormat.KEY_BIT_RATE, 128000);
format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 2);
format.setInteger(MediaFormat.KEY_SAMPLE_RATE, 44100);
format.setInteger(MediaFormat.KEY_AAC_PROFILE,MediaCodecInfo.CodecProfileLevel.AACObjectLC);
codec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
return codec;
}
/**
* Add ADTS header at the beginning of each and every AAC packet.
* This is needed as MediaCodec encoder generates a packet of raw
* AAC data.
*
* Note the packetLen must count in the ADTS header itself.
**/
private void addADTStoPacket(byte[] packet, int packetLen) {
int profile = 2; //AAC LC
//39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;
int freqIdx = 4; //44.1KHz
int chanCfg = 2; //CPE
// fill in ADTS data
packet[0] = (byte)0xFF;
packet[1] = (byte)0xF9;
packet[2] = (byte)(((profile-1)<<6) + (freqIdx<<2) +(chanCfg>>2));
packet[3] = (byte)(((chanCfg&3)<<6) + (packetLen>>11));
packet[4] = (byte)((packetLen&0x7FF) >> 3);
packet[5] = (byte)(((packetLen&7)<<5) + 0x1F);
packet[6] = (byte)0xFC;
}
}参考资料:
数字音频: http://en.flossmanuals.net/pure-data/ch003_what-is-digital-audio/
WAV文件格式: http://soundfile.sapp.org/doc/WaveFormat/
ACC文件格式: http://www.cnblogs.com/caosiyang/archive/2012/07/16/2594029.html
有关Android Media编程的一些CTS:https://android.googlesource.com/platform/cts/+/jb-mr2-release/tests/tests/media/src/android/media/cts
WAV转ACC相关问题: http://stackoverflow.com/questions/18862715/how-to-generate-the-aac-adts-elementary-stream-with-android-mediacodec