Android MediaExtractor和mp3流-Java 学习之路

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 .

我一直在尝试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 .

我已经为此重新创建了代码，我可以猜到这将是缺少的部分，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

3 回答

4
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 . 我没有't worked with audio, but my understanding is that it behaves similarly. You don' t将整个文件加载到输入缓冲区中，并且它不会为您播放流;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部分音频 . 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-1（无限）的超时，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需要大约四个缓冲区的输入才能产生任何输出 - 但我再也没有用过音频，所以我不确定这是不是预计会奏效 . 由于您的代码挂起，我猜它不是 . 将超时更改为（例如）10000并查看挂起是否消失可能很有用 .

我'm assuming this is an experiment and you'在 onCreate() 中并没有真正做到这一切 . :-)
回复于 2024-05-11T00:06:22+08:00
1

对于仍在寻找可靠地播放音频流问题的答案的人，您可能想看一下这个项目（基于MediaCodec API）

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

回复于 2024-05-11T00:06:22+08:00
2
上面的代码有两个问题 . 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， AudioTrack 需要调用 .play() .

此修改修复了OP代码：
```
mAudioTrack.play();

do {
    input();
    output();
} while (!sawInputEOS);
```
回复于 2024-05-11T00:06:22+08:00

Android MediaExtractor和mp3流

3 回答

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