下面给出的代码在模拟器上工作正常,但不适用于设备。我发现以下几行对我来说很可疑:
V / MediaExtractor(5030):自动检测媒体内容为' audio / mpeg'有信心0.20 V / ChromiumHTTPDataSource(5030):mContentSize未定义或网络可能已断开连接 V / ChromiumHTTPDataSource(5030):mContentSize未定义或网络可能已断开连接 D / com.example.mediacodectest(5030):MIME TYPE:audio / mpeg
我正在寻找提示/建议。提前谢谢......
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());
}
}
}
答案 0 :(得分:1)
我没有使用音频,但我想我可能会看到问题。你挂在dequeueOutputBuffer()
,因为编解码器正在等待更多的输入。
有些视频编解码器在完成初始化(for example)之前需要大约4个输入缓冲区。我希望一些音频编解码器的行为方式可能相同。编解码器的实现因设备而异,因此模拟器上运行的内容的行为差别不大。
将超时从-1(永远等待)更改为适度的(例如,1000微秒)。