Question

我正在尝试创建一个应用程序，该应用程序从麦克风录制音频并通过套接字将其发送到另一部手机上的服务器。为了模拟我目前在localhost上使用ServerSocket的另一部手机。现在我知道，我可能不会在没有疯狂压缩的情况下在服务器上没有波动的44100Hz，立体声，PCM_16BIT音频，但我的插座的传输速度非常低。我大约16kB / s。在localhost上！

这是我的客户端（作为两个线程在服务中运行）：

LinkedBlockingQueue<byte[]> audioQueue = new LinkedBlockingQueue<>(100);
boolean isRecording;

int sample_rate = 44100;
int buff_size = AudioRecord.getMinBufferSize(sample_rate, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT);

void recorderThread_func() throws InterruptedException, IOException{
    int s_read;
    byte[] tmp_buff;
    byte[] buffer = new byte[buff_size];

    AudioRecord recorder = new AudioRecord(
            MediaRecorder.AudioSource.MIC,
            sample_rate,
            AudioFormat.CHANNEL_IN_STEREO,
            AudioFormat.ENCODING_PCM_16BIT,
            buff_size);
    recorder.startRecording();
    isRecording = true;

    while (isRecording) {
        tmp_buff = new byte[buff_size];
        s_read = recorder.read(buffer, 0, buff_size);
        System.arraycopy(buffer, 0, tmp_buff, 0, s_read);

        audioQueue.put(tmp_buff);
    }

    isRecording = false;
    recorder.stop();
}

void socketThread_func() throws IOException, InterruptedException {
    byte[] tmp_buffer;

    Thread.sleep(250);

    Socket audioSocket = new Socket("127.0.0.1", 2004);
    OutputStream out = audioSocket.getOutputStream();
    InputStream in = audioSocket.getInputStream();
    out.flush();

    tmp_buffer = new byte[]{
            (byte) 0xde,
            (byte) ((buff_size >> 24) & 0xff),
            (byte) ((buff_size >> 16) & 0xff),
            (byte) ((buff_size >>  8) & 0xff),
            (byte) ((buff_size      ) & 0xff)};
    out.write(tmp_buffer, 0, 5);
    out.flush();

    long a;
    long b;
    long c = 1;
    long d = 1;
    while (isRecording) {
        tmp_buffer = audioQueue.take();

          a = System.currentTimeMillis();
        out.write(tmp_buffer, 0, buff_size);
        out.flush();
          b = System.currentTimeMillis();
          c += b-a;
          d++;
          System.out.println("speed=" + ((1000*buff_size*(d++))/c));

        in.read(tmp_buffer, 0, 1);
        if (tmp_buffer[0] != (byte)'A')
            break;
    }

    isRecording = false;
    out.flush();
    audioSocket.close();
}

这是我的服务器代码（在两个独立的线程中运行）：

int buff_size = 0;
boolean isPlaying;

LinkedBlockingQueue<byte[]> audioQueue = new LinkedBlockingQueue<>(100);

void socketThread_func() throws IOException, InterruptedException{
    byte[] msg = new byte[5];
    byte[] tmp_buffer;
    int s_read;

    ServerSocket audioSocket = new ServerSocket();
    audioSocket.setReceiveBufferSize(1024 * 1024 * 16);
    audioSocket.bind(new InetSocketAddress("127.0.0.1", 2004));

    Socket connSocket = audioSocket.accept();
    OutputStream out = connSocket.getOutputStream();
    InputStream in = connSocket.getInputStream();
    out.flush();

    in.read(msg, 0, 5);
    if (msg[0] != ((byte)0xde)) {
        return;
    }
    buff_size = ((int)msg[4] & 0xff) + (((int)msg[3] & 0xff) << 8) + (((int)msg[2] & 0xff) << 16) + (((int)msg[1] & 0xff) << 24);
    msg = new byte[buff_size];
    System.out.println("read: " + buff_size);

    isPlaying = true;

    long a;
    long b;
    long c = 1;
    long d = 1;
    while ( isPlaying ) {
          a = System.currentTimeMillis();
        s_read = in.read(msg);
          b = System.currentTimeMillis();
          c += b-a;
          d++;
          System.out.println("speed_sv=" + ((1000*buff_size*(d++))/c));
        if (s_read == -1)
            break;

        tmp_buffer = new byte[buff_size];
        System.arraycopy(msg, 0, tmp_buffer, 0, s_read);
        audioQueue.put(tmp_buffer);

        out.write('A');
        out.flush();
    }
    isPlaying = false;

    connSocket.close();
    audioSocket.close();
}

private static void playerThread_func() throws InterruptedException{
    byte[] tmp_buffer;

    Thread.sleep(750);

    AudioTrack mAudioTrack = new AudioTrack(
            AudioManager.STREAM_MUSIC,
            sample_rate,
            AudioFormat.CHANNEL_IN_STEREO,
            AudioFormat.ENCODING_PCM_16BIT,
            buff_size,
            AudioTrack.MODE_STREAM);
    mAudioTrack.play();

    while (isPlaying) {
        tmp_buffer = audioQueue.take();
        mAudioTrack.write(tmp_buffer, 0, buff_size);
    }

    isPlaying = false;
    mAudioTrack.stop();
}

由于我试图找出为什么这么慢，代码有点乱。这可能是某种愚蠢的错误，但我无法看到它。

编辑：我修改了代码，现在它有不同的线程用于录制/播放和套接字处理。事实证明，代码的发送部分工作正常，高速运行，这足以传输44100Hz音频（通过127.0.0.1）。但是服务器的in.read（）功能似乎正在减慢一切。

速度= 272243612

speed_sv = 459849

Answer 1

无法编写复制循环。您每次读取都要分配两个字节数组。记住这个：

byte[] buffer = new byte[8192];
int count;
while ((count = in.read(buffer)) > 0)
{
    out.write(buffer, 0, count);
}

Android慢速套接字读取

1 个答案: