Java applet VOIP audiostream合并,发出不稳定的声音

时间:2012-11-27 09:35:34

标签: java applet voip audio-streaming

我正在尝试编写一个VOIP会议Java Applet。一切顺利,音质良好,直到第三个用户进入房间。我正在尝试合并几个音频流,但是当它完成时声音真的很不稳定。每个流都以不同的线程进入。 登录applet的用户越多,质量损失就越大。

Audioformat:采样率:8000,比特:16

这是一个单独的音频流进来的地方:

while (true) {
                DatagramPacket receive = new DatagramPacket(mBuffer, mBuffer.length);
                datagramSocket.receive(receive);
                System.out.println("Received from: " + receive.getAddress() + ":" + receive.getPort());
                short[] shortBuffer = convert(mBuffer); // Convert the buffer to a 2byte short

                double volume = calcVolume(calcDistance(Client.getUser().getXPosition(), Client.getUser().getYPosition(), mUser.getXPosition(), mUser.getYPosition())); // Used to calculate sound for proximity effect
                for (int i = 0; i < shortBuffer.length; i++) {
                    shortBuffer[i] *= volume;
                }
                mBuffer = convert(shortBuffer);

                position = 0; // Set the short position in the buffer to 0
                //resetPosition();
            }

这是音频流聚集在一起并合并的地方:

    class PlayThread extends Thread {
    private short[] previousBuffer; // Used to prevent from duplicated data
    private short[] buffer = new short[1];

    @Override
    public void run() {
        while (true) {

            previousBuffer = Arrays.copyOf(buffer, buffer.length);

            for (int i = 0; i < buffer.length; i++) {
                int sum = 0;
                int total = 0;

                synchronized (mReceiveThreads) {
                    for (ReceiveThread receiveThread : mReceiveThreads) {
                        short[] sBuffer = convert(receiveThread.mBuffer);

                        buffer = smoothArray(sBuffer, 1.6); // Smooth the current buffer
                        if (receiveThread.position < sBuffer.length) {
                            sum += sBuffer[receiveThread.position];
                            receiveThread.position++;
                            total++;
                        }

                    }
                }

                if (total > 0) {
                    buffer[i] = (short) (sum / total);
                }

            }

            if(!Arrays.equals(buffer, previousBuffer)) {

                //lowerVolume();
                if(ClientPanel.getAudioselected()) {
                    sourceDataLine.write(convert(buffer), 0, buffer.length * 2);
                }
            }
            //System.out.println("Took " + (System.nanoTime() - start) + "ns.");
        }
    }

正如你所看到的,我循环遍历所有的接收线,并从所有线程中获得1个短路。我得到平均声音并将输出写入sourcedataline。

任何想法如何防止声音如此波动?

0 个答案:

没有答案