/// <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); }