希望你能帮忙 . 我正在录制麦克风的音频并通过网络直播 . 样品质量为11025hz，8位，单声道 . 虽然有一点延迟(1秒)，但效果很好 . 我需要帮助的是我现在尝试实现降噪和压缩，使音频更安静并使用更少的带宽 . 音频样本存储在C＃数组的bytes []中，我使用Socket发送/接收 .

任何人都可以建议如何在C＃中实现压缩和降噪？我不介意使用第三方库，只要它是免费的(LGPL许可证等)，并且可以从C＃中使用 . 但是，我更喜欢实际的工作源代码示例 . 提前感谢您的任何建议 .

UPDATE:

我将位大小从8位音频更改为16位音频，并且噪声问题已修复 . 来自麦克风的8位音频具有太低的信噪比 . 声音听起来很棒，11khz，16位单声道 .

然而，自从我发布这个项目以来，这个项目的要求发生了变化 . 我们现在也在尝试添加视频 . 我有一个回调设置，每100毫秒从网络摄像头接收实时图像 . 我需要对音频和视频进行编码，对它们进行编码，将它们在我的套接字上传输到服务器，服务器将流重新传输到另一个客户端，该客户端接收流，解压缩流并解码音频和视频，显示图片框中的视频，并将音频输出到扬声器 .

我正在寻找ffmpeg以帮助解决(de | en)编码/ [de] muxing，我也将SharpFFmpeg看作是ffmpeg的C＃interop库 .

我找不到这样做的好例子 . 我整个星期都在互联网上搜索，没有真正的运气 . 非常感谢您提供的任何帮助！

这是一些代码，包括我的麦克风录音回叫功能：

private const int AUDIO_FREQ = 11025;

private const int CHANNELS = 1;

private const int BITS = 16;

private const int BYTES_PER_SEC = AUDIO_FREQ * CHANNELS * (BITS / 8);

private const int BLOCKS_PER_SEC = 40;

private const int BUFFER_SECS = 1;

private const int BUF_SIZE = ((int)(BYTES_PER_SEC / BLOCKS_PER_SEC * BUFFER_SECS / 2)) * 2; // rounded to nearest EVEN number

private WaveLib.WaveOutPlayer m_Player;

private WaveLib.WaveInRecorder m_Recorder;

private WaveLib.FifoStream m_Fifo;

WebCam MyWebCam;

public void OnPickupHeadset()

{

stopRingTone();

m_Fifo = new WaveLib.FifoStream();

WaveLib.WaveFormat fmt = new WaveLib.WaveFormat(AUDIO_FREQ, BITS, CHANNELS);

m_Player = new WaveLib.WaveOutPlayer(-1, fmt, BUF_SIZE, BLOCKS_PER_SEC,

new WaveLib.BufferFillEventHandler(PlayerCB));

m_Recorder = new WaveLib.WaveInRecorder(-1, fmt, BUF_SIZE, BLOCKS_PER_SEC,

new WaveLib.BufferDoneEventHandler(RecorderCB));

MyWebCam = null;

try

{

MyWebCam = new WebCam();

MyWebCam.InitializeWebCam(ref pbMyPhoto, pbPhoto.Width, pbPhoto.Height);

MyWebCam.Start();

}

catch { }

}

private byte[] m_PlayBuffer;

private void PlayerCB(IntPtr data, int size)

{

try

{

if (m_PlayBuffer == null || m_PlayBuffer.Length != size)

m_PlayBuffer = new byte[size];

if (m_Fifo.Length >= size)

{

m_Fifo.Read(m_PlayBuffer, 0, size);

}

else

{

// Read what we can

int fifoLength = (int)m_Fifo.Length;

m_Fifo.Read(m_PlayBuffer, 0, fifoLength);

// Zero out rest of buffer

for (int i = fifoLength; i < m_PlayBuffer.Length; i++)

m_PlayBuffer[i] = 0;

}

// Return the play buffer

Marshal.Copy(m_PlayBuffer, 0, data, size);

}

catch { }

}

private byte[] m_RecBuffer;

private void RecorderCB(IntPtr data, int size)

{

try

{

if (m_RecBuffer == null || m_RecBuffer.Length != size)

m_RecBuffer = new byte[size];

Marshal.Copy(data, m_RecBuffer, 0, size);

// HERE'S WHERE I WOULD ENCODE THE AUDIO IF I KNEW HOW

// Send data to server

if (theForm.CallClient != null)

{

SocketAsyncEventArgs args = new SocketAsyncEventArgs();

args.SetBuffer(m_RecBuffer, 0, m_RecBuffer.Length);

theForm.CallClient.SendAsync(args);

}

}

catch { }

}

//Called from network stack when data received from server (other client)

public void PlayBuffer(byte[] buffer, int length)

{

try

{

//HERE'S WHERE I WOULD DECODE THE AUDIO IF I KNEW HOW

m_Fifo.Write(buffer, 0, length);

}

catch { }

}

那么我应该从哪里开始呢？