Android 慢速套接字读取

Question

我正在尝试创建一个应用程序，该应用程序记录来自麦克风的音频并通过套接字将其发送到另一部手机上的服务器。为了模拟另一部手机，我目前在 localhost 上使用 ServerSocket。现在我知道了，如果没有一些疯狂的压缩，我可能不会在服务器上获得断断续续的 44100Hz、立体声、PCM_16BIT 音频，但是我的套接字的传输速度非常低。我得到大约 16kB/s。在本地主机上！

这是我的客户端（在服务中作为两个线程运行）：

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（在两个单独的线程中运行）：

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();
}

由于我试图弄清楚为什么它这么慢，所以代码有点乱。这可能是某种愚蠢的错误，但我就是看不到。

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，代码的发送部分在高速下工作得很好，甚至足以传输 44100Hz 的音频（通过 127.0.0.1）。但是服务器的 in.read() 函数似乎让一切都变慢了。

速度=272243612

speed_sv=459849

Answer 1

这不是写复制循环的方法。您每次读取分配两个字节数组。记住这一点：

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