Java Android如何接收RTP音频流？

我正在使用Java将rtp音频从麦克风传输到我的android手机应用程序，但我没有听到任何声音。设置AudioGroup和AudioStream时没有错误，因此我假设一切正常。以下是应用程序端的代码：

    AudioStream audioStream;
    AudioGroup audioGroup;
    AudioCodec codec = AudioCodec.PCMU;
    StrictMode.ThreadPolicy policy = new StrictMode.ThreadPolicy.Builder().permitNetwork().build();
    StrictMode.setThreadPolicy(policy);
    AudioManager audio = (AudioManager)getSystemService(AUDIO_SERVICE);
    audio.setMode(AudioManager.MODE_IN_COMMUNICATION);
    audioGroup = new AudioGroup();
    audioGroup.setMode(AudioGroup.MODE_NORMAL);
    InetAddress inetAddress;
    try {
        inetAddress = InetAddress.getByName("163.11.62.208");
        audioStream = new AudioStream(inetAddress);
        audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
        audioStream.setCodec(codec);


        InetAddress inetAddressRemote = InetAddress.getByName("163.11.169.206");
        audioStream.associate(inetAddressRemote, 5004);
        audioStream.join(audioGroup);
    }

我使用的ffmpeg流测试（在vlc Android播放器上工作）是

再一次。。这在VLC android player中有效，但在我的应用程序中无效。没有错误。。只是没有音频

好的，我刚刚在LogCat中注意到，我收到消息，告诉我它正在工作。这是一些日志

stream[59] is configured as RAW 8kHz 32ms mode 0
D/AudioGroup﹕ stream[59] joins group[56]
group[56] switches from mode 0 to 1
stream[54] joins group[56]
getOutputSamplingRate() reading from output desc
V/AudioSystem﹕ getSamplingRate() streamType 0, output 2, sampling rate 48000
V/AudioSystem﹕ getFrameCount() streamType 0, output 2, frameCount 960
V/AudioSystem﹕ getLatency() streamType 0, output 2, latency 160
V/AudioTrack﹕ getMinFrameCount=1280: afFrameCount=960, minBufCount=8, afSampleRate=48000, afLatency=160
D/AudioGroup﹕ reported frame count: output 1280, input 320
D/AudioGroup﹕ adjusted frame count: output 1280, input 512
V/AudioTrack﹕ sampleRate 8000, channelMask 0x1, format 1
V/AudioTrack﹕ streamType 0
V/AudioTrack﹕ set() streamType 0, sampleRate 8000, format 1, frameCount 1280, flags 0000
V/AudioSystem﹕ getLatency() streamType 0, output 2, latency 160
V/AudioSystem﹕ getFrameCount() streamType 0, output 2, frameCount 960
V/AudioSystem﹕ getOutputSamplingRate() reading from output desc
V/AudioSystem﹕ getSamplingRate() streamType 0, output 2, sampling rate 48000
V/AudioTrack﹕ createTrack_l() output 2 afLatency 160
V/AudioTrack﹕ afFrameCount=960, minBufCount=8, afSampleRate=48000, afLatency=160
V/AudioTrack﹕ minFrameCount: 1280, afFrameCount=960, minBufCount=8, sampleRate=8000, afSampleRate=48000, afLatency=160
V/AudioRecord﹕ sampleRate 8000, channelMask 0x10, format 1
V/AudioRecord﹕ inputSource 7
V/AudioRecord﹕ set(): sampleRate 8000, channelMask 0x10, frameCount 512
D/AudioRecord﹕ set(): voiceActivationState 0
D/AudioRecord﹕ Keep input Source type.
V/AudioRecord﹕ AudioRecord::set() minFrameCount = 320
V/AudioRecord﹕ set(): mSessionId 1827
V/AudioSystem﹕ ioConfigChanged() event 3, ioHandle 1828

---

但是：我还注意到，在运行我的应用程序时，我根本没有系统卷。当我改变系统音量时，它通常会发出指示噪音，以显示声音有多大/多安静，但我什么也听不到。我不知道这是为什么。

确保您的清单中有

我打赌你可能已经这样做了，所以请转到

设置->应用->（单击齿轮）->应用权限->麦克风，并确保你的应用已打开

我花了很长时间才弄明白这一点，因为清单中的权限也存在同样的问题。
不要请求严格模式来关闭网络。它在现代设备上根本不起作用，这很可能是问题的原因。如果你需要联网，可以使用线程或异步任务。好的，我已经改变了，但是我仍然没有得到卷。我还将代码移动到一个单独的thread.Bump中。这里也有同样的问题。。。如果我尝试从模式下设置模式，只从模式下接收，只从模式下发送，我会得到非法状态异常：忙（好的，
非法状态异常
已修复，但仍然没有来自服务器的环回声音。@TylerHelmuth解决此问题的运气好吗？
stream[59] is configured as RAW 8kHz 32ms mode 0
D/AudioGroup﹕ stream[59] joins group[56]
group[56] switches from mode 0 to 1
stream[54] joins group[56]
getOutputSamplingRate() reading from output desc
V/AudioSystem﹕ getSamplingRate() streamType 0, output 2, sampling rate 48000
V/AudioSystem﹕ getFrameCount() streamType 0, output 2, frameCount 960
V/AudioSystem﹕ getLatency() streamType 0, output 2, latency 160
V/AudioTrack﹕ getMinFrameCount=1280: afFrameCount=960, minBufCount=8, afSampleRate=48000, afLatency=160
D/AudioGroup﹕ reported frame count: output 1280, input 320
D/AudioGroup﹕ adjusted frame count: output 1280, input 512
V/AudioTrack﹕ sampleRate 8000, channelMask 0x1, format 1
V/AudioTrack﹕ streamType 0
V/AudioTrack﹕ set() streamType 0, sampleRate 8000, format 1, frameCount 1280, flags 0000
V/AudioSystem﹕ getLatency() streamType 0, output 2, latency 160
V/AudioSystem﹕ getFrameCount() streamType 0, output 2, frameCount 960
V/AudioSystem﹕ getOutputSamplingRate() reading from output desc
V/AudioSystem﹕ getSamplingRate() streamType 0, output 2, sampling rate 48000
V/AudioTrack﹕ createTrack_l() output 2 afLatency 160
V/AudioTrack﹕ afFrameCount=960, minBufCount=8, afSampleRate=48000, afLatency=160
V/AudioTrack﹕ minFrameCount: 1280, afFrameCount=960, minBufCount=8, sampleRate=8000, afSampleRate=48000, afLatency=160
V/AudioRecord﹕ sampleRate 8000, channelMask 0x10, format 1
V/AudioRecord﹕ inputSource 7
V/AudioRecord﹕ set(): sampleRate 8000, channelMask 0x10, frameCount 512
D/AudioRecord﹕ set(): voiceActivationState 0
D/AudioRecord﹕ Keep input Source type.
V/AudioRecord﹕ AudioRecord::set() minFrameCount = 320
V/AudioRecord﹕ set(): mSessionId 1827
V/AudioSystem﹕ ioConfigChanged() event 3, ioHandle 1828