/// <summary>
/// Cleans up any resources being used.
/// </summary>
public void Dispose()
{
if (m_IsDisposed)
{
return;
}
m_IsDisposed = true;
try{
// If recording, we need to reset wav device first.
waveInReset(m_pWavDevHandle);
// If there are unprepared wav headers, we need to unprepare these.
foreach (BufferItem item in m_pBuffers.Values)
{
item.Dispose();
}
// Close input device.
waveInClose(m_pWavDevHandle);
m_pInDevice = null;
m_pWavDevHandle = IntPtr.Zero;
this.AudioFrameReceived = null;
}
catch {
}
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="device">Input device.</param>
/// <param name="samplesPerSec">Sample rate, in samples per second (hertz). For PCM common values are
/// 8.0 kHz, 11.025 kHz, 22.05 kHz, and 44.1 kHz.</param>
/// <param name="bitsPerSample">Bits per sample. For PCM 8 or 16 are the only valid values.</param>
/// <param name="channels">Number of channels.</param>
/// <param name="bufferSize">Specifies recording buffer size.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>outputDevice</b> is null.</exception>
/// <exception cref="ArgumentException">Is raised when any of the aruments has invalid value.</exception>
public _WaveIn(AudioInDevice device, int samplesPerSec, int bitsPerSample, int channels, int bufferSize)
{
if (device == null)
{
throw new ArgumentNullException("device");
}
if (samplesPerSec < 8000)
{
throw new ArgumentException("Argument 'samplesPerSec' value must be >= 8000.");
}
if (bitsPerSample < 8)
{
throw new ArgumentException("Argument 'bitsPerSample' value must be >= 8.");
}
if (channels < 1)
{
throw new ArgumentException("Argument 'channels' value must be >= 1.");
}
m_pInDevice = device;
m_SamplesPerSec = samplesPerSec;
m_BitsPerSample = bitsPerSample;
m_Channels = channels;
m_BufferSize = bufferSize;
m_BlockSize = m_Channels * (m_BitsPerSample / 8);
m_pBuffers = new Dictionary <long, BufferItem>();
// Try to open wav device.
WAVEFORMATEX format = new WAVEFORMATEX();
format.wFormatTag = WavFormat.PCM;
format.nChannels = (ushort)m_Channels;
format.nSamplesPerSec = (uint)samplesPerSec;
format.nAvgBytesPerSec = (uint)(m_SamplesPerSec * (m_Channels * (m_BitsPerSample / 8)));
format.nBlockAlign = (ushort)m_BlockSize;
format.wBitsPerSample = (ushort)m_BitsPerSample;
format.cbSize = 0;
// We must delegate reference, otherwise GC will collect it.
m_pWaveInProc = new waveInProc(this.OnWaveInProc);
int result = waveInOpen(out m_pWavDevHandle, m_pInDevice.Index, format, m_pWaveInProc, 0, WavConstants.CALLBACK_FUNCTION);
if (result != MMSYSERR.NOERROR)
{
throw new Exception("Failed to open wav device, error: " + result.ToString() + ".");
}
CreateBuffers();
}
/// <summary>
/// Cleans up any resources being used.
/// </summary>
public void Dispose()
{
if (m_IsDisposed)
{
return;
}
Stop();
m_IsDisposed = true;
this.Error = null;
m_pAudioInDevice = null;
m_pAudioCodecs = null;
m_pRTP_Stream.Session.PayloadChanged -= new EventHandler(m_pRTP_Stream_PayloadChanged);
m_pRTP_Stream = null;
m_pActiveCodec = null;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="audioInDevice">Audio-in device to capture.</param>
/// <param name="audioFrameSize">Audio frame size in milliseconds.</param>
/// <param name="codecs">Audio codecs with RTP payload number. For example: 0-PCMU,8-PCMA.</param>
/// <param name="stream">RTP stream to use for audio sending.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>audioInDevice</b>,<b>codecs</b> or <b>stream</b> is null reference.</exception>
public AudioIn_RTP(AudioInDevice audioInDevice, int audioFrameSize, Dictionary <int, AudioCodec> codecs, RTP_SendStream stream)
{
if (audioInDevice == null)
{
throw new ArgumentNullException("audioInDevice");
}
if (codecs == null)
{
throw new ArgumentNullException("codecs");
}
if (stream == null)
{
throw new ArgumentNullException("stream");
}
m_pAudioInDevice = audioInDevice;
m_AudioFrameSize = audioFrameSize;
m_pAudioCodecs = codecs;
m_pRTP_Stream = stream;
m_pRTP_Stream.Session.PayloadChanged += new EventHandler(m_pRTP_Stream_PayloadChanged);
m_pAudioCodecs.TryGetValue(m_pRTP_Stream.Session.Payload, out m_pActiveCodec);
}