将g711数据包转换为波形文件时声音不稳定

Question

我正在使用Sharppcap库从SIP调用中获取数据包。到现在为止还好。但是，当我加入这些数据包(G.711 Alaw数据包在一个MemoryStream中)，并将其与我从https://www.codeproject.com/Articles/14237/Using-the-G711-standard获得的Wav动态链接库进行转换，并使用解码的数组字节编写Wav文件时，我可以收听录音，但它听起来不稳定和缓慢。我刚接触音频编程，所以我没有太多的经验。下面是我为实现这一点而编写的代码片段：

private static void Device_OnPacketArrival(object sender, CaptureEventArgs e)
    {
        var time = e.Packet.Timeval.Date.AddHours(-3);
        var len = e.Packet.Data.Length;
        


        var packet = PacketDotNet.Packet.ParsePacket(e.Packet.LinkLayerType, e.Packet.Data);
        var device = sender as ICaptureDevice;
        var tcpPacket = packet.Extract<PacketDotNet.TcpPacket>();
        var udpPacket = packet.Extract<PacketDotNet.UdpPacket>();

        if (udpPacket != null)
        {
            var ipPacket = (PacketDotNet.IPPacket)udpPacket.ParentPacket;
            System.Net.IPAddress srcIp = ipPacket.SourceAddress;
            System.Net.IPAddress dstIp = ipPacket.DestinationAddress;
            int srcPort = udpPacket.SourcePort;
            int dstPort = udpPacket.DestinationPort;
            byte[] udpHeaderData = udpPacket.HeaderData;
            byte[] udpPayloadData = udpPacket.PayloadData;
            string decodedUdpPayloadData = Encoding.UTF8.GetString(udpPayloadData);

            if (decodedUdpPayloadData.Contains("m=audio"))
            {
                FindRTPAudioPort(device, decodedUdpPayloadData);
            }
            else if (device.Filter != "udp port 5060")
            {
                RtpPacketsToWave(device, udpPayloadData);
            }
            else
            {
                Console.WriteLine("{0}:{1}:{2},{3} Len={4} {5}:{6} -> {7}:{8} UDP Packet " +
                "\n {9} \n Hex DUMP: {10} \n",
                time.Hour, time.Minute, time.Second, time.Millisecond, len,
                srcIp, srcPort, dstIp, dstPort,
                decodedUdpPayloadData,
                BitConverter.ToString(udpPayloadData));
            }
        }
        else if (tcpPacket != null)
        {
            var ipPacket = (PacketDotNet.IPPacket)tcpPacket.ParentPacket;
            System.Net.IPAddress srcIp = ipPacket.SourceAddress;
            System.Net.IPAddress dstIp = ipPacket.DestinationAddress;
            int srcPort = tcpPacket.SourcePort;
            int dstPort = tcpPacket.DestinationPort;


            Console.WriteLine("{0}:{1}:{2},{3} Len={4} {5}:{6} -> {7}:{8}",
                time.Hour, time.Minute, time.Second, time.Millisecond, len,
                srcIp, srcPort, dstIp, dstPort);
        }
        else
        {
            Console.WriteLine("\n");
        }
    }

    private static void RtpPacketsToWave(ICaptureDevice dev, byte[] payloadData)
    {
        try
        {

            MemoryStreamSingleton memoryStreamSingleton = MemoryStreamSingleton.GetInstance();
            MemoryStream memStream;
            byte[] headlessPayloadData = new byte[160];
            if (payloadData.Length == 172)
            {
                //Skips first 12 bytes containing the packet header
                headlessPayloadData = payloadData.Skip(12).ToArray();
                memStream = new MemoryStream(headlessPayloadData);
                memStream.CopyTo(memoryStreamSingleton);
            }
            Console.WriteLine("Payload length: {0}", headlessPayloadData.Length);
            Console.WriteLine(memoryStreamSingleton.Length);
            if(memoryStreamSingleton.Length > 600000)
            {
                WaveFileGenerator(memoryStreamSingleton.ToArray());
                dev.StopCapture();
            }
        }
        catch (Exception ex)
        {

            Console.WriteLine(ex.ToString());
        }
        
    }

    private static void WaveFileGenerator(byte[] buffer)
    {
        try
        {
          
            Console.WriteLine("Device closed, generating audio file..");
            WaveFormat waveFormat = new WaveFormat(8000, 16, 1);

            short[] pcm16bit = ALawDecoder.ALawDecode(buffer);
            byte[] result1 = new byte[pcm16bit.Length * sizeof(short)];
            Buffer.BlockCopy(pcm16bit, 0, result1, 0, result1.Length);
            

            var outputWave = new WaveFileWriter(@"tmp/test.wav", waveFormat);
            outputWave.Write(result1, 0, result1.Length);
            outputWave.Close();
            var waveFileProvider = new WaveFileReader(@"tmp/test.wav");
            MonoToStereoProvider16 toStereo = new MonoToStereoProvider16(waveFileProvider);
            WaveFileWriter.CreateWaveFile("test.wav", toStereo);
            waveFileProvider.Dispose();
            File.Delete(@"tmp/test.wav");
        }
        catch (Exception ex)
        {
            Console.WriteLine(ex.ToString());
            File.WriteAllText("log.txt", ex.ToString());
        }
    }

我不明白我错过了什么..。

Answer 1

由于在接收函数中花费的时间过长，可能会丢失数据包，从而导致数据包溢出内部缓冲区。

在有大量数据包被捕获的情况下，接收例程中的任何额外处理(包括打印)都可能导致数据包丢失。

为了帮助减少在该例程中花费的时间并最大限度地提高数据包速率，您可以将数据包排队以在后台线程中进行处理。

在SharpPcap示例中，有一个如何通过后台线程排队和处理数据包的示例，您可以在https://github.com/chmorgan/sharppcap/blob/master/Examples/QueuingPacketsForBackgroundProcessing/Program.cs上找到该示例

使用队列方法，接收例程非常快地退出，使得能够无丢失地处理非常高的分组速率。

此外，您还可以添加一个队列大小检查，以确认后台线程是否跟上了传入数据包的速率。

降低传入数据包速率的一种方法是使用网络数据包筛选器(在操作系统或驱动程序层中运行)，并且仅包括潜在的SIP数据包。我不熟悉SIP，但是如果你有更多的信息，我可以试着帮助你。您的代码似乎正在使用过滤器，但不清楚在发现SIP流的情况下会发生什么，在这种情况下，您是否正在过滤以仅包含该端口？

让我知道这是否有助于减少或消除起伏不定的声音。