[英]Android: MediaCodec+MediaMuxer encoded audio MP4 won't play
I have the following function which takes a WAV (PCM) file and encodes it to an AAC-encoded MP4 file using Android's MediaCode and MediaMuxer classes. 我有以下函数,该函数需要一个WAV(PCM)文件,并使用Android的MediaCode和MediaMuxer类将其编码为AAC编码的MP4文件。 This is audio only. 这仅是音频。 The function runs successfully and outputs a .mp4 of reasonable that is recognized as AAC-encoded. 该函数成功运行,并输出一个公认的AAC编码的.mp4。 But it doesn't play on Android, web or iOS players, and crashes Audacity. 但是它不能在Android,Web或iOS播放器上播放,并且会使Audacity崩溃。 Is there something I am missing? 我有什么想念的吗? Code is shown below. 代码如下所示。
public void encode(final String from, final String to, final Callback callback) {
new Thread(new Runnable() {
@Override
public void run() {
try {
extractor.setDataSource(from);
int numTracks = extractor.getTrackCount();
for (int i = 0; i < numTracks; ++i) {
MediaFormat format = extractor.getTrackFormat(i);
String mime = format.getString(MediaFormat.KEY_MIME);
Log.d(TAG, "Track " + i + " mime-type: " + mime);
if (true) {
extractor.selectTrack(i);
}
}
MediaCodec codec = MediaCodec.createEncoderByType("audio/mp4a-latm");
MediaFormat format = new MediaFormat();
format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm");
format.setInteger(MediaFormat.KEY_BIT_RATE, 128 * 1024);
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);
final MediaMuxer muxer = new MediaMuxer(to, MediaMuxer.OutputFormat.MUXER_OUTPUT_MPEG_4);
final ByteBuffer byteBuffer = ByteBuffer.allocate(65536);
codec.setCallback(new MediaCodec.Callback() {
@Override
public void onInputBufferAvailable(MediaCodec codec, int bufferIndex) {
ByteBuffer inputBuffer = codec.getInputBuffer(bufferIndex);
if (isEndOfStream) {
return;
}
int sampleCapacity = inputBuffer.capacity() / 8;
if (numAvailable == 0) {
numAvailable = extractor.readSampleData(byteBuffer, 0);
if (numAvailable <= 0) {
codec.queueInputBuffer(bufferIndex, 0, 0, 0, MediaCodec.BUFFER_FLAG_END_OF_STREAM);
isEndOfStream = true;
return;
}
long presentationTimeUs = extractor.getSampleTime();
extractor.advance();
}
if (numAvailable < sampleCapacity) {
codec.queueInputBuffer(bufferIndex, 0, numAvailable * 8, 0, 0);
numAvailable = 0;
} else {
codec.queueInputBuffer(bufferIndex, 0, sampleCapacity * 8, 0, 0);
numAvailable -= sampleCapacity;
}
}
@Override
public void onOutputBufferAvailable(MediaCodec codec, int index, MediaCodec.BufferInfo info) {
ByteBuffer outputBuffer = codec.getOutputBuffer(index);
muxer.writeSampleData(audioTrackIndex,outputBuffer,info);
codec.releaseOutputBuffer(index, true);
if ((info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
Log.d(TAG, "end of encoding!");
codec.stop();
codec.release();
extractor.release();
extractor = null;
muxer.stop();
muxer.release();
callback.run(true);
}
}
@Override
public void onError(MediaCodec codec, MediaCodec.CodecException e) {
Log.e(TAG, "codec error", e);
}
@Override
public void onOutputFormatChanged(MediaCodec codec, MediaFormat format) {
audioTrackIndex = muxer.addTrack(format);
muxer.start();
}
});
codec.start();
} catch (IOException e) {
Log.e(TAG,"Unable to encode",e);
callback.run(false);
}
}
}).run();
You seem to be encoding your AAC into LATM format which isn't very popular. 您似乎将AAC编码为LATM 格式 ,这种格式不太流行。 Maybe that's the reason players won't play it. 也许这就是玩家不玩的原因。 Try using some other media type, audio/mp4
or audio/3gpp
. 尝试使用其他媒体类型, audio/mp4
或audio/3gpp
。
See AAC container formats . 请参阅AAC容器格式 。
You need to: 你需要:
Example code as below: 示例代码如下:
MediaExtractor extractor = null;
int numAvailable = 0;
boolean isEndOfStream = false;
int audioTrackIndex = 0;
long totalen = 0;
int channels = 0;
int sampleRate = 0;
public void encode(final String from, final String to) {
new Thread(new Runnable() {
@Override
public void run() {
try {
extractor = new MediaExtractor();
extractor.setDataSource(from);
int numTracks = extractor.getTrackCount();
for (int i = 0; i < numTracks; ++i) {
MediaFormat format = extractor.getTrackFormat(i);
String mime = format.getString(MediaFormat.KEY_MIME);
Log.d(TAG, "Track " + i + " mime-type: " + mime);
if (true) {
extractor.selectTrack(i);
channels = extractor.getTrackFormat(i).getInteger(MediaFormat.KEY_CHANNEL_COUNT);
sampleRate = extractor.getTrackFormat(i).getInteger(MediaFormat.KEY_SAMPLE_RATE);
Log.e(TAG,"sampleRate:" + sampleRate + " channels:" + channels);
}
}
String mimeType = "audio/mp4a-latm";
MediaCodec codec = MediaCodec.createEncoderByType(mimeType);
MediaFormat format = new MediaFormat();
format.setString(MediaFormat.KEY_MIME, mimeType);
format.setInteger(MediaFormat.KEY_BIT_RATE, 128 * 1024);
format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, channels);
format.setInteger(MediaFormat.KEY_SAMPLE_RATE, sampleRate);
format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
codec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
final MediaMuxer muxer = new MediaMuxer(to, MediaMuxer.OutputFormat.MUXER_OUTPUT_MPEG_4);
final ByteBuffer byteBuffer = ByteBuffer.allocate(65536);
codec.setCallback(new MediaCodec.Callback() {
@Override
public void onInputBufferAvailable(MediaCodec codec, int bufferIndex) {
ByteBuffer inputBuffer = codec.getInputBuffer(bufferIndex);
inputBuffer.clear();
if (isEndOfStream) {
return;
}
int sampleCapacity = inputBuffer.capacity();
if (numAvailable == 0) {
numAvailable = extractor.readSampleData(byteBuffer, 0);
if (numAvailable <= 0) {
codec.queueInputBuffer(bufferIndex, 0, 0, 0, MediaCodec.BUFFER_FLAG_END_OF_STREAM);
isEndOfStream = true;
return;
}
extractor.advance();
}
long timestampUs = 1000000l * totalen / (2 * channels * sampleRate);
if (numAvailable < sampleCapacity) {
byte[] byteArray = new byte[numAvailable];
byteBuffer.get(byteArray);
inputBuffer.put(byteArray, 0, (int)numAvailable);
totalen += numAvailable;
codec.queueInputBuffer(bufferIndex, 0, numAvailable, timestampUs, 0);
numAvailable = 0;
} else {
byte[] byteArray = new byte[sampleCapacity];
byteBuffer.get(byteArray);
inputBuffer.put(byteArray, 0, (int)sampleCapacity);
totalen += sampleCapacity;
codec.queueInputBuffer(bufferIndex, 0, sampleCapacity, timestampUs, 0);
numAvailable -= sampleCapacity;
}
}
@Override
public void onOutputBufferAvailable(MediaCodec codec, int index, MediaCodec.BufferInfo info) {
ByteBuffer outputBuffer = codec.getOutputBuffer(index);
muxer.writeSampleData(audioTrackIndex,outputBuffer,info);
codec.releaseOutputBuffer(index, true);
if ((info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
Log.d(TAG, "end of encoding!");
codec.stop();
codec.release();
extractor.release();
extractor = null;
muxer.stop();
muxer.release();
//callback.run(true);
}
}
@Override
public void onError(MediaCodec codec, MediaCodec.CodecException e) {
Log.e(TAG, "codec error", e);
}
@Override
public void onOutputFormatChanged(MediaCodec codec, MediaFormat format) {
audioTrackIndex = muxer.addTrack(format);
muxer.start();
}
});
codec.start();
} catch (IOException e) {
Log.e(TAG,"Unable to encode",e);
//callback.run(false);
}
}
}).run();
}
BTW,Why you need to divide 8 with the buffer length? 顺便说一句,为什么您需要将8除以缓冲区长度? And what's the Callback class? 什么是回调类? Please share the import module! 请共享导入模块! I almost can not pass build with the callback parameter so comment it out! 我几乎无法通过带有回调参数的构建,因此将其注释掉!
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