Question

我正在尝试使用MediaExtractor / MediaCodec播放mp3流。由于延迟和长缓冲区大小，MediaPlayer是不可能的。

我找到的唯一示例代码是：http://dpsm.wordpress.com/category/android/

代码示例只是parcial（？）并使用File而不是stream。

我一直在努力调整这个例子来播放音频流，但我无法理解它应该如何工作。像往常一样，Android文档没有任何帮助。

据我所知，首先我们获取有关流的信息，可能是使用此信息设置AudioTrack（代码示例包括AudioTrack初始化？）然后打开输入缓冲区和输出缓冲区。

我为此重新创建了代码，我可以猜到的是缺少的部分，但没有音频出来。

有人能指出我正确的方向，以了解这应该如何工作？

public final String LOG_TAG = "mediadecoderexample";
private static int TIMEOUT_US = -1;
MediaCodec codec;
MediaExtractor extractor;

MediaFormat format;
ByteBuffer[] codecInputBuffers;
ByteBuffer[] codecOutputBuffers;
Boolean sawInputEOS = false;
Boolean sawOutputEOS = false;
AudioTrack mAudioTrack;
BufferInfo info;

@Override
protected void onCreate(Bundle savedInstanceState) {
    super.onCreate(savedInstanceState);
    setContentView(R.layout.activity_main);

    String url = "http://82.201.100.9:8000/RADIO538_WEB_MP3";
    extractor = new MediaExtractor();

    try {
        extractor.setDataSource(url);
    } catch (IOException e) {
    }

    format = extractor.getTrackFormat(0);
    String mime = format.getString(MediaFormat.KEY_MIME);
    int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);

    Log.i(LOG_TAG, "===========================");
    Log.i(LOG_TAG, "url "+url);
    Log.i(LOG_TAG, "mime type : "+mime);
    Log.i(LOG_TAG, "sample rate : "+sampleRate);
    Log.i(LOG_TAG, "===========================");

    codec = MediaCodec.createDecoderByType(mime);
    codec.configure(format, null , null , 0);
    codec.start();

    codecInputBuffers = codec.getInputBuffers();
    codecOutputBuffers = codec.getOutputBuffers();

    extractor.selectTrack(0); 

    mAudioTrack = new AudioTrack(
            AudioManager.STREAM_MUSIC, 
            sampleRate, 
            AudioFormat.CHANNEL_OUT_STEREO, 
            AudioFormat.ENCODING_PCM_16BIT, 
            AudioTrack.getMinBufferSize (
                    sampleRate, 
                    AudioFormat.CHANNEL_OUT_STEREO, 
                    AudioFormat.ENCODING_PCM_16BIT
                    ), 
            AudioTrack.MODE_STREAM
            );

    info = new BufferInfo();


    input();
    output();


}

private void output()
{
    final int res = codec.dequeueOutputBuffer(info, TIMEOUT_US);
    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

        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(LOG_TAG, "Output format has changed to " + oformat);
        mAudioTrack.setPlaybackRate(oformat.getInteger(MediaFormat.KEY_SAMPLE_RATE));
    }

}

private void input()
{
    Log.i(LOG_TAG, "inputLoop()");
    int inputBufIndex = codec.dequeueInputBuffer(TIMEOUT_US);
    Log.i(LOG_TAG, "inputBufIndex : "+inputBufIndex);

    if (inputBufIndex >= 0) {   
        ByteBuffer dstBuf = codecInputBuffers[inputBufIndex];

        int sampleSize = extractor.readSampleData(dstBuf, 0);
        Log.i(LOG_TAG, "sampleSize : "+sampleSize);
        long presentationTimeUs = 0;
        if (sampleSize < 0) {
            Log.i(LOG_TAG, "Saw input end of stream!");
            sawInputEOS = true;
            sampleSize = 0;
        } else {
            presentationTimeUs = extractor.getSampleTime();
            Log.i(LOG_TAG, "presentationTimeUs "+presentationTimeUs);
        }

        codec.queueInputBuffer(inputBufIndex,
                               0, //offset
                               sampleSize,
                               presentationTimeUs,
                               sawInputEOS ? MediaCodec.BUFFER_FLAG_END_OF_STREAM : 0);
        if (!sawInputEOS) {
            Log.i(LOG_TAG, "extractor.advance()");
            extractor.advance();

        }
     }

}
}

编辑：添加logcat输出以获得额外的想法。

03-10 16:47:54.115: I/mediadecoderexample(24643): ===========================
03-10 16:47:54.115: I/mediadecoderexample(24643): url ....
03-10 16:47:54.115: I/mediadecoderexample(24643): mime type : audio/mpeg
03-10 16:47:54.115: I/mediadecoderexample(24643): sample rate : 32000
03-10 16:47:54.115: I/mediadecoderexample(24643): ===========================
03-10 16:47:54.120: I/OMXClient(24643): Using client-side OMX mux.
03-10 16:47:54.150: I/Reverb(24643):  getpid() 24643, IPCThreadState::self()->getCallingPid() 24643
03-10 16:47:54.150: I/mediadecoderexample(24643): inputLoop()
03-10 16:47:54.155: I/mediadecoderexample(24643): inputBufIndex : 0
03-10 16:47:54.155: I/mediadecoderexample(24643): sampleSize : 432
03-10 16:47:54.155: I/mediadecoderexample(24643): presentationTimeUs 0
03-10 16:47:54.155: I/mediadecoderexample(24643): extractor.advance()
03-10 16:47:59.085: D/HTTPBase(24643): [2] Network BandWidth = 187 Kbps
03-10 16:47:59.085: D/NuCachedSource2(24643): Remaining (64K), HighWaterThreshold (20480K)
03-10 16:48:04.635: D/HTTPBase(24643): [3] Network BandWidth = 141 Kbps
03-10 16:48:04.635: D/NuCachedSource2(24643): Remaining (128K), HighWaterThreshold (20480K)
03-10 16:48:09.930: D/HTTPBase(24643): [4] Network BandWidth = 127 Kbps
03-10 16:48:09.930: D/NuCachedSource2(24643): Remaining (192K), HighWaterThreshold (20480K)
03-10 16:48:15.255: D/HTTPBase(24643): [5] Network BandWidth = 120 Kbps
03-10 16:48:15.255: D/NuCachedSource2(24643): Remaining (256K), HighWaterThreshold (20480K)
03-10 16:48:20.775: D/HTTPBase(24643): [6] Network BandWidth = 115 Kbps
03-10 16:48:20.775: D/NuCachedSource2(24643): Remaining (320K), HighWaterThreshold (20480K)
03-10 16:48:26.510: D/HTTPBase(24643): [7] Network BandWidth = 111 Kbps
03-10 16:48:26.510: D/NuCachedSource2(24643): Remaining (384K), HighWaterThreshold (20480K)
03-10 16:48:31.740: D/HTTPBase(24643): [8] Network BandWidth = 109 Kbps
03-10 16:48:31.740: D/NuCachedSource2(24643): Remaining (448K), HighWaterThreshold (20480K)
03-10 16:48:37.260: D/HTTPBase(24643): [9] Network BandWidth = 107 Kbps
03-10 16:48:37.260: D/NuCachedSource2(24643): Remaining (512K), HighWaterThreshold (20480K)
03-10 16:48:42.620: D/HTTPBase(24643): [10] Network BandWidth = 106 Kbps
03-10 16:48:42.620: D/NuCachedSource2(24643): Remaining (576K), HighWaterThreshold (20480K)
03-10 16:48:48.295: D/HTTPBase(24643): [11] Network BandWidth = 105 Kbps
03-10 16:48:48.295: D/NuCachedSource2(24643): Remaining (640K), HighWaterThreshold (20480K)
03-10 16:48:53.735: D/HTTPBase(24643): [12] Network BandWidth = 104 Kbps
03-10 16:48:53.735: D/NuCachedSource2(24643): Remaining (704K), HighWaterThreshold (20480K)
03-10 16:48:59.115: D/HTTPBase(24643): [13] Network BandWidth = 103 Kbps
03-10 16:48:59.115: D/NuCachedSource2(24643): Remaining (768K), HighWaterThreshold (20480K)
03-10 16:49:04.480: D/HTTPBase(24643): [14] Network BandWidth = 103 Kbps
03-10 16:49:04.480: D/NuCachedSource2(24643): Remaining (832K), HighWaterThreshold (20480K)
03-10 16:49:09.955: D/HTTPBase(24643): [15] Network BandWidth = 102 Kbps

Answer 1

对于仍在寻找可靠播放流式音频问题的答案的人，您可能想看看这个项目（基于MediaCodec API）

https://code.google.com/p/android-openmxplayer/

Answer 2

onCreate()中的代码表明您对MediaCodec的工作方式存在误解。您的代码目前是：

onCreate() {
    ...setup...
    input();
    output();
}

MediaCodec对访问单元进行操作。对于视频，每次调用输入/输出都会为您提供一帧视频。我还没有使用音频，但我的理解是它的行为类似。您不会将整个文件加载到输入缓冲区中，也不会为您播放该流;你拿一小块文件，把它交给解码器，然后回传解码数据（例如YUV视频缓冲器或PCM音频数据）。然后，您可以执行播放该数据所需的任何操作。

因此，您的示例充其量只能解码一小段音频。您需要在循环中执行submit-input-get-output并正确处理流末尾。您可以在各种bigflake示例中看到为视频完成此操作。看起来你的代码有必要的部分。

您正在使用超时-1（无限），因此您将提供一个输入缓冲区并永远等待输出缓冲区。在视频中，这不起作用 - 我测试过的解码器似乎想要输出四个输入缓冲区，然后它们才能产生任何输出 - 但我再也没有使用过音频，所以我不确定这是否有用。由于您的代码悬而未决，我猜测它不是。将超时更改为（例如）10000并查看挂起是否消失可能很有用。

我假设这是一个实验，你并不是真的会在onCreate()中完成所有这些。： - ）

Answer 3

上述代码存在两个问题。首先，如接受的答案所述，从输入流中只进行一次读取。但是，其次，.play()需要拨打AudioTrack。

此修改修复了OPs代码：

mAudioTrack.play();

do {
    input();
    output();
} while (!sawInputEOS);

Android MediaExtractor和mp3流

3 个答案: