Play or Capture Audio Sound. Send and Receive as Multicast (RTP)
Play, record and capture Audio sound. Read, write and stream Wav-Files. Send and receive Ulaw RTP-streams. Using the WaveIn WaveOut functions of the WMM API.
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). With the help of this library, you can play or capture audio sound in an easy way. In addition, I wrote some Network Assemblies, to stream and play Audio Data over Network by Multicast.
Optional C++
In addition, I implemented a similar C++ Version for the sound part of this project as a COM Server with name WinSoundServer
. In the project is also included a little WinSoundServerTester.sln for Visual Studio 2012 that shows the recording and playing methods in an example. I did not intend to publish that, but maybe anybody will find it useful anyway. You must first compile the WinSoundServer
project, so that the vc100.tlb and the WinSoundServer.dll will be created. Before starting the test app, you have to register the COM Server with the help of the reg.bat file included in the project directory. This will register the WinSoundServer.dll. After that, the test application must refer to the typelibrary
by an import
statement, so you have access to all interfaces and classes (like adding a reference in C#):
#import "WinSoundServer\\vc100.tlb" no_namespace
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 cannot 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
Assemblies which are to be Used as DLLs
For common sound operations:
- WinSound.dll
For network operations:
- TCPClient.dll
- TCPServer.dll
- MulticastReceiver.dll
- MulticastSender.dll
Example Projects
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();
//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 receiving 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 than or equal to 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 - Using the CSRC Count Information in RTP Header for Multicast Player
- 08.07.2012 - Wav-File Read/Write operations for Player Tester
- 11.08.2012 - Error resolved, when calling
Player.PlayData
in blocking mode =true
- 21.08.2012 - Error resolved. Garbage Collection caused Application crash
- 07.09.2012 - Error resolved. Reading Wave-Files in different formats
- 09.09.2012 - Better performance when playing wav files
- 19.10.2012 - Added Jitter buffer. Added streaming wav-files directly
- 22.10.2013 - Drawing the linear datas as curve
- 12.04.2014 - Added source. WinSound implementation written in C++ as a COM Server (optional)
- 22.04.2014 - Solved possible stability problems