Play or Capture Audio Sound. Send and Receive as Multicast (RTP)

Introduction      

It is not comfortable to use the Sound API in .NET, because there is no support by the class library itself. So I decided to write my own small Audio Assembly WinSound.dll by using PInvoke and the native WMM API (Windows Multi Media). In addition i wrote some Network Assemblies, to stream and play Audio Data over Network by Multicast.

Background 

The WaveOut and WaveIn functions of WMM are very tricky, because you have to know some special features of how to work with the internal buffers and handles. For example you can not stop a recording, without waiting for all buffers to finish. You are not allowed to stay too long in the Callback functions. You have to swap from managed to unmanaged code. The Repeater is the most sensitive one, because you have to balance the recording and playing buffers all the time.

Using the code   

The assemblies which are to be used as DLLs are 

For common sound operations  

• WinSound.dll      

For network operations   

• TCPClient.dll 
• TCPServer.dll
• MulticastReceiver.dll
• MulticastSender.dll 

Example Projects are 

• MulticastPlayer      (Plays a RTP-Multicast to Soundcard)
• MulticastStreamer  (Sends a RTP-Multicast from Soundcard) 
• Player Tester         (Plays or records a wav-File)
• RepeaterTester      (Plays from Soundcard to Soundcard)  

The heart of all this is WinSound.dll. All example projects need this DLL (add as reference). 

Using the Sound Recorder   

Note that buffersize and buffercount must be optimized to your Sounddevice and used SamplesPerSecond.

//Create a Recorder
WinSound.Recorder recorder = new WinSound.Recorder();
//Add DataRecording and RecordingStopped Event Functions
recorder.DataRecorded += new WinSound.Recorder.DelegateDataRecorded(OnDataRecorded);
recorder.RecordingStopped += new WinSound.Recorder.DelegateStopped(OnRecordingStopped);
//Start capturing (44100 SamplesPerSecond, 16 BitsPerSample, 2 Channels, 8 * 4096 Byte Buffers)
recorder.Start("Line 1 (Virtual Audio Cable)", 44100, 16, 2, 8, 4096);
//Check if started
bool isStarted = recorder.Started;
//Stop recording
recorder.Stop();


//Called, when datas are incoming
private void OnDataRecorded(Byte[] data)
{
    
}

//Called, when recording has stopped
private void OnRecordingStopped()
{
    
}

Using the Sound Player  

Note that buffercount must be optimized to your Sounddevice and used SamplesPerSecond.

//Create a SoundPlayer
WinSound.Player player = new WinSound.Player();
//Add Player Stopped and Player Closed Events
player.PlayerStopped += new WinSound.Player.DelegateStopped(OnPlayerStopped);
player.PlayerClosed += new WinSound.Player.DelegateStopped(OnPlayerClosed);
//Open a SoundDevice (44100 SamplesPerSecond, 16 BitsPerSample, 2 Channels, 8 Buffers)
player.Open("Realtek", 44100, 16, 2, 8);
//Check if open
bool isOpen = player.Opened;          
//Get Sound Datas as linear bytes (maybe from Wav-File)
Byte[] data = GetDatas();
//Play linear datas to SoundDevice
player.PlayData(data, false);
//Check if playing
bool isPlaying = player.Playing
//Start pause
player.StartPause();
//Check if paused
bool isPaused = player.Paused;
//End pause
player.EndPause();
//Stop Playing
player.Stop();
//Close Player
player.Stop();
					
//Called, when player is stopped 
private void OnPlayerStopped()
{

}

//Called, when player is closed (WaveOut closed)
private void OnPlayerClosed()
{

}  

Linear Bytes

The linear data bytes you write or read from soundcard, are depending on BitsPerSample and Channel Count. Each "Data Packet" is to interpret as this picture shows.

The Player (Recorder) Tester

This example uses WinSound.dll to play or record a Wav-file. SamplesPerSecond, BitsPerSample and Channels are only used in record mode.

//Create a SoundPlayer
WinSound.Player playerOne = new WinSound.Player();
//Add callback functions for stop and close events (optional)
playerOne.PlayerClosed += new WinSound.Player.DelegateStopped(OnPlayerClosed);
playerOne.PlayerStopped += new WinSound.Player.DelegateStopped(OnPlayerStopped);
//Play a Wav File on a specific Sounddevice
playerOne.PlayFile(@"C:\Record.wav", "Realtek");   
//Close the player
bool hr = playerOne.Close(); 

//Read Wave-Header and payload data from Wav-File
WinSound.WaveFileHeader header = WinSound.WaveFile.Read(@"C:\Record.wav");
//Get the payload data size
uint dataSize = header.DATASize;
//Get the payload data
Byte[] data = header.Payload;

//Write payload data in a new Wav-File 
WinSound.WaveFile.Create(@"C:\Record.wav", 44100, 16, 2, data);
//Add payload data in an existing Wav-File (Append)
WinSound.WaveFile.AppendData(@"C:\Record.wav", data);  

The Repeater Tester

This example is recording Sound-Data from one sound device (WaveIn) to another (WaveOut). Every sound device has its special features, so you have to change the BufferCount and BufferSize for optimal performance. It also differs from operating System Windows-XP, Vista or Windows 7.

//Create the Repeater
WinSound.Repeater repeaterOne = new WinSound.Repeater(); 
//Start the Repeater
repeaterOne.Start("Line 6", "Realtek", samplesPerSecond, bitsPerSample, channels, bufferCount, bufferSize);
//Stop the Repeater 
repeaterOne.Stop();

The Multicast Streamer

This example Streams ULAW RTP-Multicast data from a sound device or a Wav-File. When using sound device streaming, it is possible to synchronise it with the "Time Sync" checkbox. Buffer Count is the amout of buffers, that are used by capturing from sound-device (Use 8 as standard). You can use the Multicast Player to play this Multicast. Also you can use the VLC Media Player for testing (Use Samples Per Second = 8000 and BitsPerSample = 16 with VLC).

//Create a  Multicast Sender and a Sound-Device Recorder
NF.MulticastSender m_MulticastSender = new NF.MulticastSender(Address, Port, TTL);
WinSound.Recorder m_Recorder = new WinSound.Recorder();
//Define a callback function for reveiving datas from soundcard
m_Recorder.DataRecorded += new WinSound.Recorder.DelegateDataRecorded(OnDataReceivedFromSoundcard);
//Start Recorder 
m_Recorder.Start(SoundDeviceName, SamplesPerSecond, BitsPerSample, ChannelCount, BufferCount, BufferSize) 
//In callback function, get the linear datas from soundcard and translate to MuLaw
Byte[] mulawData = WinSound.Utils.LinearToMulaw(linearData, Config.BitsPerSample, Config.Channels);
//Create the RTP Header 
Byte[] rtpBytes = RTPHeader.ToBytes();
//Combine RTP Header and mulawData
Byte[] bytes = new Byte[mulawData.Length + WinSound.RTPPacket.MinHeaderLength];
Array.Copy(rtpBytes, 0, bytes, 0, WinSound.RTPPacket.MinHeaderLength);
Array.Copy(mulawData, 0, bytes, WinSound.RTPPacket.MinHeaderLength, mulawData.Length);
//Send Bytes to Multicast
m_MulticastSender.SendBytes(bytes);
 

The Multicast Player

This example plays a ULAW RTP-Multicast stream to a sound device. You can use the Multicast Streamer for this purpose. Note, the packet size of the Multicast Player must be greater or equal than the packet size of the multicast streamer (Use 16000 Bytes as standard) . A Jitter Buffer is used, when entering a minimum value of "2" as jitter count. Jitter count means the amount of RTP-Packets that will be queued as maximum. Values of "0" or "1" means, there is no Jitter Buffer in use. (Use 20 as standard)

//Create a Multicast Receiver and a SoundPlayer 
NF.MulticastReceiver m_Receiver = new NF.MulticastReceiver(PacketSize);
WinSound.Player m_Player = new WinSound.Player(); 
//Define a callback function for receiving datas from multicast
m_Receiver.DataReceived += new NF.MulticastReceiver.DelegateDataReceived(OnDataReceived);
//Connect the Multicast and open the Player
m_Receiver.Connect(MulticasAddress, MulticastPort); 
m_Player.Open(SoundDeviceName, SamplesPerSecond, BitsPerSample, Channels, BufferCount); 
//In the callback function get the rtp header of the multicast packet
WinSound.RTPPacket rtp = new WinSound.RTPPacket(bytes);
//Translate MuLaw to linear
Byte[] linearBytes = WinSound.Utils.MuLawToLinear(rtp.Data, Config.BitsPerSample, Config.Channels);
//Play the linear datas to soundcard
m_Player.PlayData(linearBytes, false);
//Close the Player and the Receiver
m_Receiver.Disconnect();
m_Player.Close(); 

History  

• 31.05.2012 - Added.
• 30.06.2012 - Updated. Using the CSRC Count Information in RTP Header for Multicast Player.
• 08.07.2012 - Updated. Wav-File Read/Write operations for Player Tester.  
• 11.08.2012 - Updated. Error resolved, when calling Player.PlayData in blocking mode = true
• 21.08.2012 - Updated. Error resolved. Garbage Collection caused Application crash.
• 07.09.2012 - Updated. Error resolved. Reading Wave-Files in different formats.
• 09.09.2012 - Updated. Better performance when playing wav files.
• 19.10.2012 - Updated. Added Jitter buffer. Added streaming wav-files directly.