问题
I'm trying to decode a mp3 file and stream it to AudioTrack. It all works fine but causes a lot of GC on the Java side. I've made sure to not allocate memory in my play/stream loop and blame ByteBuffer.Get(byte[], int, int) binding of allocating a temp Java array. Anyone can confirm and/or show a better way of feeding data from MediaCodec to AudioTrack? (I know API 21 introduced AudioTrack.write(ByteBuffer, ...)) Thanks
Here is what I do:
byte[] audioBuffer = new byte[...];
...
ByteBuffer codecOutputBuffer = codecOutputBuffers[outputIndex];
// The next line seems to be the source of a lot of GC during playback
codecOutputBuffer.Get(audioBuffer, 0, bufInfo.Size);
audioTrack.Write(audioBuffer, 0, bufInfo.Size);
UPDATE 1: I tried to use Allocation Tracker to confirm the allocation site. What I found out is that the allocated objects are 8kb large byte arrays. Unfortunatelly Allocation Tracker does not show allocation site stacktrace for them:
1 32 org.apache.harmony.dalvik.ddmc.Chunk 6 org.apache.harmony.dalvik.ddmc.DdmServer dispatch
2 16 java.lang.Integer 6 java.lang.Integer valueOf
3 16 byte[] 6
4 8192 byte[] 20
5 8192 byte[] 20
6 8192 byte[] 20
To make sure it was ByteBuffer.Get(byte[], int, int) that allocates the arrays, I rerun the app with:
audioTrack.Write(...) commented out - no change
codecOutputBuffer.Get(audioBuffer, 0, bufInfo.Size) commented out - the allocations are gone
I'm going to rewrite it in Java to check if I get the same results in a native app.
UPDATE 2: I have rewritten the code in Java and now I get a perfectly flat graph in memory monitor - no allocations during playback.
My conclusion/guess is that the ByteBuffer.Get(byte[], int, int) binding from Mono to Java allocates a temp array. Not really sure why it is 8kb large since my audioBuffer only ever gets to slightly above 4kb.
UPDATE 3: My ultimate goal with this is to have a cross platform app (with more complex functionality on top of mp3 player) so I went ahead and created another experiment. I have a java component with audio streaming/decoding/playing functionality that exposes only play(), pause() methods that I consume from C#. This way I don't have allocation problems but still can drive the player from my hopefully reusable c# code. Source code below (this is just research - not production code).
Java:
import android.content.Context;
import android.content.res.AssetFileDescriptor;
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
import android.media.MediaCodec;
import android.media.MediaExtractor;
import android.media.MediaFormat;
import java.nio.ByteBuffer;
public class AudioPlayer {
public void play(Context aContext, final int resourceId){
final Context context = aContext;
new Thread()
{
@Override
public void run() {
try {
AssetFileDescriptor fd = context.getResources().openRawResourceFd(resourceId);
MediaExtractor extractor = new MediaExtractor();
extractor.setDataSource(fd.getFileDescriptor(), fd.getStartOffset(), fd.getLength());
extractor.selectTrack(0);
MediaFormat trackFormat = extractor.getTrackFormat(0);
MediaCodec decoder = MediaCodec.createDecoderByType(trackFormat.getString(MediaFormat.KEY_MIME));
decoder.configure(trackFormat, null, null, 0);
decoder.start();
ByteBuffer[] decoderInputBuffers = decoder.getInputBuffers();
ByteBuffer[] decoderOutputBuffers = decoder.getOutputBuffers();
int inputIndex = decoder.dequeueInputBuffer(-1);
ByteBuffer inputBuffer = decoderInputBuffers[inputIndex];
MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
byte[] audioBuffer = null;
AudioTrack audioTrack = null;
int read = extractor.readSampleData(inputBuffer, 0);
while (read > 0) {
decoder.queueInputBuffer(inputIndex, 0, read, extractor.getSampleTime(), 0);
extractor.advance();
int outputIndex = decoder.dequeueOutputBuffer(bufferInfo, -1);
if (outputIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
trackFormat = decoder.getOutputFormat();
} else if (outputIndex >= 0) {
if (bufferInfo.size > 0) {
ByteBuffer outputBuffer = decoderOutputBuffers[outputIndex];
if (audioBuffer == null || audioBuffer.length < bufferInfo.size) {
audioBuffer = new byte[bufferInfo.size];
}
outputBuffer.rewind();
outputBuffer.get(audioBuffer, 0, bufferInfo.size);
decoder.releaseOutputBuffer(outputIndex, false);
if (audioTrack == null) {
int sampleRateInHz = trackFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE);
int channelCount = trackFormat.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
int channelConfig = channelCount == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO;
audioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
sampleRateInHz,
channelConfig,
AudioFormat.ENCODING_PCM_16BIT,
AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT) * 2,
AudioTrack.MODE_STREAM);
audioTrack.play();
}
audioTrack.write(audioBuffer, 0, bufferInfo.size);
}
}
inputIndex = decoder.dequeueInputBuffer(-1);
inputBuffer = decoderInputBuffers[inputIndex];
read = extractor.readSampleData(inputBuffer, 0);
}
} catch (Exception e) {
}
}
}.start();
}
}
C#
[Activity(Label = "AndroidAudioTest", MainLauncher = true, Icon = "@drawable/icon")]
public class MainActivity : Activity
{
protected override void OnCreate(Bundle bundle)
{
base.OnCreate(bundle);
SetContentView(Resource.Layout.Main);
var play = FindViewById<Button>(Resource.Id.Play);
play.Click += (s, e) =>
{
new AudioPlayer().Play(this, Resource.Raw.PianoInsideMics);
};
}
}
来源：https://stackoverflow.com/questions/28701102/how-to-stream-data-from-mediacodec-to-audiotrack-with-xamarin-for-android