Develop Reference

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. This is audio only. The function runs successfully and outputs a .mp4 of reasonable that is recognized as AAC-encoded. But it doesn't play on Android, web or iOS players, and crashes Audacity. Is there something I am missing? Code is shown below.
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 need to:
Add timestamp information correctly since media muxer need to use it to TAG packet time information.
Add logic to copy data buffer from extractor data buffer (PCM) to mediacodec input buffer, only refer to buffer index will only encode a random data buffer without initial.
Add code to apply input source property such as channels and sample rate to mediacodec. Not sure if you are intent to encode with different channels and sample rate!
Example code as below:
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? And what's the Callback class? Please share the import module! I almost can not pass build with the callback parameter so comment it out!

You seem to be encoding your AAC into LATM format which isn't very popular. Maybe that's the reason players won't play it. Try using some other media type, audio/mp4 or audio/3gpp.
See AAC container formats.

Convert PCM to AAC or MP4 file without MediaMuxer

I need to convert a PCM file to AAC or MP4 file. Until now, I did it with MediaCodec and MediaMuxer, But MediaMuxer is supported from Android 4.3. Is there a method to do the conversion without the use of MediaMuxer?
My code is this:
MediaMuxer mux = null;
try {
File inputFile = new File(filePath + ".pcm");
FileInputStream fis = new FileInputStream(inputFile);
mux = new MediaMuxer(filePath + ".mp4", MediaMuxer.OutputFormat.MUXER_OUTPUT_MPEG_4);
MediaFormat outputFormat = MediaFormat.createAudioFormat(COMPRESSED_AUDIO_FILE_MIME_TYPE,
SAMPLING_RATE, 1);
outputFormat.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
outputFormat.setInteger(MediaFormat.KEY_BIT_RATE, COMPRESSED_AUDIO_FILE_BIT_RATE);
MediaCodec codec = MediaCodec.createEncoderByType(COMPRESSED_AUDIO_FILE_MIME_TYPE);
codec.configure(outputFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
codec.start();
ByteBuffer[] codecInputBuffers = codec.getInputBuffers();
ByteBuffer[] codecOutputBuffers = codec.getOutputBuffers();
MediaCodec.BufferInfo outBuffInfo = new MediaCodec.BufferInfo();
byte[] tempBuffer = new byte[BUFFER_SIZE];
boolean hasMoreData = true;
double presentationTimeUs = 0;
int audioTrackIdx = 0;
int totalBytesRead = 0;
int percentComplete;
do {
int inputBufIndex = 0;
while (inputBufIndex != -1 && hasMoreData) {
inputBufIndex = codec.dequeueInputBuffer(CODEC_TIMEOUT_IN_MS);
if (inputBufIndex >= 0) {
ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];
dstBuf.clear();
int bytesRead = fis.read(tempBuffer, 0, dstBuf.limit());
if (bytesRead == -1) { // -1 implies EOS
hasMoreData = false;
codec.queueInputBuffer(inputBufIndex, 0, 0, (long) presentationTimeUs, MediaCodec.BUFFER_FLAG_END_OF_STREAM);
} else {
totalBytesRead += bytesRead;
dstBuf.put(tempBuffer, 0, bytesRead);
codec.queueInputBuffer(inputBufIndex, 0, bytesRead, (long) presentationTimeUs, 0);
presentationTimeUs = 1000000l * (totalBytesRead / 2) / SAMPLING_RATE;
}
}
}
// Drain audio
int outputBufIndex = 0;
while (outputBufIndex != MediaCodec.INFO_TRY_AGAIN_LATER) {
outputBufIndex = codec.dequeueOutputBuffer(outBuffInfo, CODEC_TIMEOUT_IN_MS);
if (outputBufIndex >= 0) {
ByteBuffer encodedData = codecOutputBuffers[outputBufIndex];
encodedData.position(outBuffInfo.offset);
encodedData.limit(outBuffInfo.offset + outBuffInfo.size);
if ((outBuffInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0 && outBuffInfo.size != 0) {
codec.releaseOutputBuffer(outputBufIndex, false);
} else {
mux.writeSampleData(audioTrackIdx, codecOutputBuffers[outputBufIndex], outBuffInfo);
codec.releaseOutputBuffer(outputBufIndex, false);
}
} else if (outputBufIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
outputFormat = codec.getOutputFormat();
Log.v("AUDIO", "Output format changed - " + outputFormat);
audioTrackIdx = mux.addTrack(outputFormat);
mux.start();
} else if (outputBufIndex == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
Log.e("AUDIO", "Output buffers changed during encode!");
} else if (outputBufIndex != MediaCodec.INFO_TRY_AGAIN_LATER){
Log.e("AUDIO", "Unknown return code from dequeueOutputBuffer - " + outputBufIndex);
}
}
percentComplete = (int) Math.round(((float) totalBytesRead / (float) inputFile.length()) * 100.0);
Log.v("AUDIO", "Conversion % - " + percentComplete);
} while (outBuffInfo.flags != MediaCodec.BUFFER_FLAG_END_OF_STREAM);
fis.close();
mux.stop();
mux.release();

As you already pointed out, there's no public system API compatible with older version of Android for this kind of job.
Anyway, you can pursue a custom solution using a native encoder (like FFMPEG). I can suggested you the following: timsu/android-aac-enc
Android AAC Encoder project
Extraction of Android Stagefright VO AAC encoder with a nice Java API.
This project offers an easy Java API for the underlying JNI encoder, and it should be ready to use since the native library is already compiled for generic ARM architectures (please note that I haven't tested it). The whole library is just 500kb so it won't fatten your APK that much.
For a quick test, import in your project the following parts:
Java bindings for the native AAC encorder
Pre-compiled AAC encoder .so library
Example of usage (speech encoding)
You should be able to easily adapt the example to your code.

MediaCodec hasn't any available input buffer

I am using MediaCodec API for encoding video and audio into mp4 file. Data encoded in separate threads. Sometimes on some devices audio encoder stops to return any available input buffer and as result MediaMuxer crashes when trying to stop it. Here is my code:
configuring media codec:
public static final String MIME_TYPE_AUDIO = "audio/mp4a-latm";
public static final int SAMPLE_RATE = 44100;
public static final int CHANNEL_COUNT = 1;
public static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_MONO;
public static final int BIT_RATE_AUDIO = 128000;
public static final int SAMPLES_PER_FRAME = 1024 * 2;
public static final int FRAMES_PER_BUFFER = 24;
public static final int AUDIO_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
public static final int AUDIO_SOURCE = MediaRecorder.AudioSource.MIC;
public static final int MAX_INPUT_SIZE = 16384 * 4;
public static final int MAX_SAMPLE_SIZE = 256 * 1024;
private AudioRecord audioRecord;
private ByteBuffer[] inputBuffers;
private ByteBuffer inputBuffer;
private MediaExtractor mediaExtractor;
private boolean audioSended = false;
private boolean completed = false;
private int sampleCount;
private int iBufferSize;
public AudioEncoderCore(MovieMuxer muxer) throws IOException {
this.muxer = muxer;
bufferInfo = new MediaCodec.BufferInfo();
MediaFormat mediaFormat = null;
mediaFormat = MediaFormat.createAudioFormat(MIME_TYPE_AUDIO, SAMPLE_RATE, CHANNEL_COUNT);
mediaFormat.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
mediaFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, MAX_INPUT_SIZE);
mediaFormat.setInteger(MediaFormat.KEY_BIT_RATE, BIT_RATE_AUDIO);
encoder = MediaCodec.createEncoderByType(MIME_TYPE_AUDIO);
encoder.configure(mediaFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
encoder.start();
iBufferSize = SAMPLES_PER_FRAME * FRAMES_PER_BUFFER;
// Ensure buffer is adequately sized for the AudioRecord
// object to initialize
int iMinBufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE, CHANNEL_CONFIG, AUDIO_FORMAT);
if (iBufferSize < iMinBufferSize)
iBufferSize = ((iMinBufferSize / SAMPLES_PER_FRAME) + 1) * SAMPLES_PER_FRAME * 2;
audioRecord = new AudioRecord(AUDIO_SOURCE, SAMPLE_RATE, CHANNEL_CONFIG, AUDIO_FORMAT, iBufferSize);
audioRecord.startRecording();
}
sending data to encoder:
public void sendDataFromMic(boolean endOfStream) {
if (endOfStream)
Log.d(TAG, "sendDataFromMic end of stream");
long audioPresentationTimeNs;
byte[] mTempBuffer = new byte[SAMPLES_PER_FRAME];
audioPresentationTimeNs = System.nanoTime();
int iReadResult = audioRecord.read(mTempBuffer, 0, mTempBuffer.length);
if (iReadResult == AudioRecord.ERROR_BAD_VALUE || iReadResult == AudioRecord.ERROR_INVALID_OPERATION || iReadResult == 0) {
Log.e(TAG, "audio buffer read error");
} else {
// send current frame data to encoder
try {
if (inputBuffers == null)
inputBuffers = encoder.getInputBuffers();
//Sometimes can't get any available input buffer
int inputBufferIndex = encoder.dequeueInputBuffer(100000);
Log.d(TAG, "inputBufferIndex = " + inputBufferIndex);
if (inputBufferIndex >= 0) {
inputBuffer = inputBuffers[inputBufferIndex];
inputBuffer.clear();
inputBuffer.put(mTempBuffer, 0, iReadResult);
Log.d(TAG, "sending frame to audio encoder " + iReadResult + " bytes");
encoder.queueInputBuffer(inputBufferIndex, 0, iReadResult, audioPresentationTimeNs / 1000,
endOfStream ? MediaCodec.BUFFER_FLAG_END_OF_STREAM : 0);
}
} catch (Throwable t) {
Log.e(TAG, "sendFrameToAudioEncoder exception");
t.printStackTrace();
}
}
}
drain encoder:
public void drainEncoder() {
ByteBuffer[] encoderOutputBuffers = encoder.getOutputBuffers();
while (true) {
int encoderStatus = encoder.dequeueOutputBuffer(bufferInfo, TIMEOUT_NSECS);
if (encoderStatus == MediaCodec.INFO_TRY_AGAIN_LATER) {
Log.d(TAG, "no output available, spinning to await EOS");
break;
} else if (encoderStatus == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED)
encoderOutputBuffers = encoder.getOutputBuffers();
else if (encoderStatus == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat newFormat = encoder.getOutputFormat();
Log.d(TAG, "encoder format changed: " + newFormat);
trackIndex = muxer.addTrack(newFormat);
} else if (muxer.isMuxerStarted()) {
ByteBuffer encodedData = encoderOutputBuffers[encoderStatus];
if (encodedData == null)
throw new RuntimeException("encoded buffer " + encoderStatus + " was null");
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
Log.d(TAG, "ignoring BUFFER_FLAG_CODEC_CONFIG");
bufferInfo.size = 0;
}
if (bufferInfo.size != 0) {
encodedData.position(bufferInfo.offset);
encodedData.limit(bufferInfo.offset + bufferInfo.size);
muxer.writeSampleData(trackIndex, encodedData, bufferInfo);
Log.d(TAG, "sent " + bufferInfo.size + " bytes to muxer, ts=" +
bufferInfo.presentationTimeUs + " track index=" + trackIndex);
}
encoder.releaseOutputBuffer(encoderStatus, false);
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
Log.d(TAG, "end of stream reached");
completed = true;
break; // out of while
}
}
}
}
Bug is stable reproduced on HTC One, Galaxy S3, but all works fine on Huawei Honor 3C.

After source code investigations I found solution. When I dequeue output buffer, the muxer may be not started yet, in that case buffer not released. So here's working code for drain encoder:
public void drainEncoder() {
ByteBuffer[] encoderOutputBuffers = encoder.getOutputBuffers();
while (true) {
int encoderStatus = encoder.dequeueOutputBuffer(bufferInfo, TIMEOUT_NSECS);
if (encoderStatus == MediaCodec.INFO_TRY_AGAIN_LATER) {
Log.d(TAG, "no output available, spinning to await EOS");
break;
} else if (encoderStatus == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED)
encoderOutputBuffers = encoder.getOutputBuffers();
else if (encoderStatus == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat newFormat = encoder.getOutputFormat();
Log.d(TAG, "encoder format changed: " + newFormat);
trackIndex = muxer.addTrack(newFormat);
} else if (muxer.isMuxerStarted()) {
ByteBuffer encodedData = encoderOutputBuffers[encoderStatus];
if (encodedData == null)
throw new RuntimeException("encoded buffer " + encoderStatus + " was null");
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
Log.d(TAG, "ignoring BUFFER_FLAG_CODEC_CONFIG");
bufferInfo.size = 0;
}
if (bufferInfo.size != 0) {
encodedData.position(bufferInfo.offset);
encodedData.limit(bufferInfo.offset + bufferInfo.size);
muxer.writeSampleData(trackIndex, encodedData, bufferInfo);
Log.d(TAG, "sent " + bufferInfo.size + " bytes to muxer, ts=" +
bufferInfo.presentationTimeUs + " track index=" + trackIndex);
}
encoder.releaseOutputBuffer(encoderStatus, false);
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
Log.d(TAG, "end of stream reached");
completed = true;
break; // out of while
}
}
else{
//Muxer not ready, release buffer
encoder.releaseOutputBuffer(encoderStatus, false);
Log.d(TAG, "muxer not ready, skip data");
}
}
}

Create MP4 container for .ts segment

I want to play static HLS content (not live video content) on my app in Android. What I currently do is download all the segments from the .m3u8 file and merge it into one file. When I play this file, I can see this the video being played, but it is not seekable. As per this link, .ts files are not seekable on Android.
I cannot risk running ffmpeg on phone for converting the file to MP4 format. I have studied MP4 format and its atom structure. What I want to know is, if there is an easy way to create MP4 container (atoms hierarchy) which would simply refer to the .ts segment (the merged segment that was created from sub-segments) in its data atom (mdat)
I would really appreciate any help/suggestions.

Not possible without a copy. TS is uses 188 byte packets with headers. These headers create breaks in the middle of frames. In mp4, frames must be contiguous.

Android provides support libraries such as MediaCodec and MediaExtractor that provides access to low level media encoding/decoding. It is fast and efficient as it uses hardware acceleration.
Here's how I believe one is suppose to do it on Android unless you are ok with using ffmpeg which of course if resource intensive operation.
1) Use MediaExtractor to extract data from the file.
2) Pass the extracted data to MediaCodec.
3) Use MediaCodec to render output to a surface (in case of video) and AudioTrack (in case of audio).
4) This is the most difficult step: Synchronize audio/video. I haven't implemented this yet. But this would require keeping track of time sync between audio and video. Audio would be played normally and you might have to drop some frames in case of video to keep it in sync with audio playback.
Here's code for decoding audio/video and playing them respectively using AudioTrack and Surface.
In case of video decoding, there's a sleep to slowdown the frame rendering.
public void decodeVideo(Surface surface) throws IOException {
MediaExtractor extractor = new MediaExtractor();
MediaCodec codec;
ByteBuffer[] codecInputBuffers;
ByteBuffer[] codecOutputBuffers;
extractor.setDataSource(file);
Log.d(TAG, "No of tracks = " + extractor.getTrackCount());
MediaFormat format = extractor.getTrackFormat(0);
String mime = format.getString(MediaFormat.KEY_MIME);
Log.d(TAG, "mime = " + mime);
Log.d(TAG, "format = " + format);
codec = MediaCodec.createDecoderByType(mime);
codec.configure(format, surface, null, 0);
codec.start();
codecInputBuffers = codec.getInputBuffers();
codecOutputBuffers = codec.getOutputBuffers();
extractor.selectTrack(0);
final long timeout_in_Us = 5000;
MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
boolean sawInputEOS = false;
boolean sawOutputEOS = false;
int noOutputCounter = 0;
long startMs = System.currentTimeMillis();
while(!sawOutputEOS && noOutputCounter < 50) {
noOutputCounter++;
if(!sawInputEOS) {
int inputBufIndex = codec.dequeueInputBuffer(timeout_in_Us);
if(inputBufIndex >= 0) {
ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];
int sampleSize = extractor.readSampleData(dstBuf, 0);
long presentationTimeUs = 0;
if(sampleSize < 0) {
Log.d(TAG, "saw input EOS.");
sawInputEOS = true;
sampleSize = 0;
} else {
presentationTimeUs = extractor.getSampleTime();
}
codec.queueInputBuffer(inputBufIndex, 0, sampleSize, presentationTimeUs, sawInputEOS ? MediaCodec.BUFFER_FLAG_END_OF_STREAM : 0);
if(!sawInputEOS) {
extractor.advance();
}
}
}
int res = codec.dequeueOutputBuffer(info, timeout_in_Us);
if(res >= 0) {
if(info.size > 0) {
noOutputCounter = 0;
}
int outputBufIndex = res;
while(info.presentationTimeUs/1000 > System.currentTimeMillis() - startMs) {
try {
Thread.sleep(5);
} catch (Exception e) {
break;
}
}
codec.releaseOutputBuffer(outputBufIndex, true);
if((info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) == MediaCodec.BUFFER_FLAG_END_OF_STREAM) {
Log.d(TAG, "saw output EOS.");
sawOutputEOS = true;
}
} else if(res == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
codecOutputBuffers = codec.getOutputBuffers();
Log.d(TAG, "output buffers have changed.");
} else if(res == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat format1 = codec.getOutputFormat();
Log.d(TAG, "output format has changed to " + format1);
} else if(res == MediaCodec.INFO_TRY_AGAIN_LATER) {
Log.d(TAG, "Codec try again returned" + res);
}
}
codec.stop();
codec.release();
}
private int audioSessionId = -1;
private AudioTrack createAudioTrack(MediaFormat format) {
int channelConfiguration = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT) == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO;
int bufferSize = AudioTrack.getMinBufferSize(format.getInteger(MediaFormat.KEY_SAMPLE_RATE), channelConfiguration, AudioFormat.ENCODING_PCM_16BIT) * 8;
AudioTrack audioTrack;
if(audioSessionId == -1) {
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, format.getInteger(MediaFormat.KEY_SAMPLE_RATE), channelConfiguration,
AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
} else {
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, format.getInteger(MediaFormat.KEY_SAMPLE_RATE), channelConfiguration,
AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM, audioSessionId);
}
audioTrack.play();
audioSessionId = audioTrack.getAudioSessionId();
return audioTrack;
}
public void decodeAudio() throws IOException {
MediaExtractor extractor = new MediaExtractor();
MediaCodec codec;
ByteBuffer[] codecInputBuffers;
ByteBuffer[] codecOutputBuffers;
extractor.setDataSource(file);
Log.d(TAG, "No of tracks = " + extractor.getTrackCount());
MediaFormat format = extractor.getTrackFormat(1);
String mime = format.getString(MediaFormat.KEY_MIME);
Log.d(TAG, "mime = " + mime);
Log.d(TAG, "format = " + format);
codec = MediaCodec.createDecoderByType(mime);
codec.configure(format, null, null, 0);
codec.start();
codecInputBuffers = codec.getInputBuffers();
codecOutputBuffers = codec.getOutputBuffers();
extractor.selectTrack(1);
AudioTrack audioTrack = createAudioTrack(format);
final long timeout_in_Us = 5000;
MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
boolean sawInputEOS = false;
boolean sawOutputEOS = false;
int noOutputCounter = 0;
while(!sawOutputEOS && noOutputCounter < 50) {
noOutputCounter++;
if(!sawInputEOS) {
int inputBufIndex = codec.dequeueInputBuffer(timeout_in_Us);
if(inputBufIndex >= 0) {
ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];
int sampleSize = extractor.readSampleData(dstBuf, 0);
long presentationTimeUs = 0;
if(sampleSize < 0) {
Log.d(TAG, "saw input EOS.");
sawInputEOS = true;
sampleSize = 0;
} else {
presentationTimeUs = extractor.getSampleTime();
}
codec.queueInputBuffer(inputBufIndex, 0, sampleSize, presentationTimeUs, sawInputEOS ? MediaCodec.BUFFER_FLAG_END_OF_STREAM : 0);
if(!sawInputEOS) {
extractor.advance();
}
}
}
int res = codec.dequeueOutputBuffer(info, timeout_in_Us);
if(res >= 0) {
if(info.size > 0) {
noOutputCounter = 0;
}
int outputBufIndex = res;
//Possibly store the decoded buffer
ByteBuffer buf = codecOutputBuffers[outputBufIndex];
final byte[] chunk = new byte[info.size];
buf.get(chunk);
buf.clear();
if(chunk.length > 0) {
audioTrack.write(chunk, 0 ,chunk.length);
}
codec.releaseOutputBuffer(outputBufIndex, false);
if((info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) == MediaCodec.BUFFER_FLAG_END_OF_STREAM) {
Log.d(TAG, "saw output EOS.");
sawOutputEOS = true;
}
} else if(res == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
codecOutputBuffers = codec.getOutputBuffers();
Log.d(TAG, "output buffers have changed.");
} else if(res == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat format1 = codec.getOutputFormat();
Log.d(TAG, "output format has changed to " + format1);
audioTrack.stop();
audioTrack = createAudioTrack(codec.getOutputFormat());
} else if(res == MediaCodec.INFO_TRY_AGAIN_LATER) {
Log.d(TAG, "Codec try again returned" + res);
}
}
codec.stop();
codec.release();
}

Android MediaCodec API - Music Plays on Emulator, Not on the Device

The code given below works fine on the emulator, but not the device. I found the following lines that looked suspicious to me:
V/MediaExtractor(5030): Autodetected media content as 'audio/mpeg' with confidence 0.20
V/ChromiumHTTPDataSource(5030): mContentSize is undefined or network might be disconnected
V/ChromiumHTTPDataSource(5030): mContentSize is undefined or network might be disconnected
D/com.example.mediacodectest(5030): MIME TYPE: audio/mpeg
I am looking for hints/suggestions. Thanks in advance...
private class PlayerThread extends Thread {
#Override
public void run() {
MediaExtractor extractor;
MediaCodec codec;
ByteBuffer[] codecInputBuffers;
ByteBuffer[] codecOutputBuffers;
AudioTrack mAudioTrack;
mAudioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
44100,
AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
8192 * 2,
AudioTrack.MODE_STREAM);
extractor = new MediaExtractor();
try
{
extractor.setDataSource("http://anmp3streamingsource.com/stream");
MediaFormat format = extractor.getTrackFormat(0);
String mime = format.getString(MediaFormat.KEY_MIME);
Log.d(TAG, String.format("MIME TYPE: %s", mime));
codec = MediaCodec.createDecoderByType(mime);
codec.configure(
format,
null /* surface */,
null /* crypto */,
0 /* flags */ );
codec.start();
codecInputBuffers = codec.getInputBuffers();
codecOutputBuffers = codec.getOutputBuffers();
extractor.selectTrack(0); // <= You must select a track. You will read samples from the media from this track!
boolean sawInputEOS = false;
boolean sawOutputEOS = false;
for (;;) {
int inputBufIndex = codec.dequeueInputBuffer(-1);
if (inputBufIndex >= 0) {
ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];
int sampleSize = extractor.readSampleData(dstBuf, 0);
long presentationTimeUs = 0;
if (sampleSize < 0) {
sawInputEOS = true;
sampleSize = 0;
} else {
presentationTimeUs = extractor.getSampleTime();
}
codec.queueInputBuffer(inputBufIndex,
0, //offset
sampleSize,
presentationTimeUs,
sawInputEOS ? MediaCodec.BUFFER_FLAG_END_OF_STREAM : 0);
if (!sawInputEOS) {
extractor.advance();
}
MediaCodec.BufferInfo info = new BufferInfo();
final int res = codec.dequeueOutputBuffer(info, -1);
if (res >= 0) {
int outputBufIndex = res;
ByteBuffer buf = codecOutputBuffers[outputBufIndex];
final byte[] chunk = new byte[info.size];
buf.get(chunk); // Read the buffer all at once
buf.clear(); // ** MUST DO!!! OTHERWISE THE NEXT TIME YOU GET THIS SAME BUFFER BAD THINGS WILL HAPPEN
mAudioTrack.play();
if (chunk.length > 0) {
mAudioTrack.write(chunk, 0, chunk.length);
}
codec.releaseOutputBuffer(outputBufIndex, false /* render */);
if ((info.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
sawOutputEOS = true;
}
}
else if (res == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED)
{
codecOutputBuffers = codec.getOutputBuffers();
}
else if (res == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED)
{
final MediaFormat oformat = codec.getOutputFormat();
Log.d(TAG, "Output format has changed to " + oformat);
mAudioTrack.setPlaybackRate(oformat.getInteger(MediaFormat.KEY_SAMPLE_RATE));
}
}
}
}
catch (IOException e)
{
Log.e(TAG, e.getMessage());
}
}
}

I haven't worked with audio, but I think I may see the problem. You're hanging in dequeueOutputBuffer() because the codec is waiting for more input.
Some of the video codecs want ~4 buffers of input before they'll even finish initialization (for example). I expect some audio codecs may behave the same way. Codec implementations vary from device to device, so it's not surprising that what runs on the emulator behaves much differently.
Change the timeouts from -1 (wait forever) to something modest (say, 1000 microseconds).