/// <summary>
/// Sets or updates the sources of the audio and/or video streams.
/// </summary>
/// <param name="audioOptions">Optional. If audio is being switched the new source options.
/// Set to null to leave the audio source unchanged.</param>
/// <param name="videoOptions">Optional. If video is being switched the new source options.
/// Set to null to leave the video source unchanged.</param>
public async Task SetSources(AudioOptions audioOptions, VideoOptions videoOptions)
{
if (audioOptions == null)
{
// Do nothing, audio source not being changed.
}
else if (audioOptions.AudioSource == AudioSourcesEnum.None)
{
// Audio source no longer required.
_waveInEvent?.StopRecording();
if (_audioStreamTimer != null)
{
_audioStreamTimer?.Dispose();
// Give any currently executing audio sampling time to complete.
await Task.Delay(AUDIO_SAMPLE_PERIOD_MILLISECONDS * 2).ConfigureAwait(false);
}
_audioStreamReader.Close();
_audioOpts = audioOptions;
}
else
{
SetAudioSource(audioOptions);
_audioOpts = audioOptions;
StartAudio();
}
if (videoOptions == null)
{
// Do nothing, video source not being changed.
}
else if (videoOptions.VideoSource == VideoSourcesEnum.None)
{
// Video source no longer required.
_testPatternVideoSource?.Stop();
_videoOpts = videoOptions;
}
else
{
await SetVideoSource(videoOptions).ConfigureAwait(false);
_videoOpts = videoOptions;
StartVideo();
}
}
/// <summary>
/// Creates a new RTP audio visual session with audio/video capturing and rendering capabilities.
/// </summary>
/// <param name="addrFamily">The address family to create the underlying socket on (IPv4 or IPv6).</param>
/// <param name="audioOptions">Options for the send and receive audio streams on this session.</param>
/// <param name="videoOptions">Options for the send and receive video streams on this session</param>
public RtpAVSession(AddressFamily addrFamily, AudioOptions audioOptions, VideoOptions videoOptions)
: base(addrFamily, false, false, false)
{
_audioOpts = audioOptions ?? DefaultAudioOptions;
_videoOpts = videoOptions ?? DefaultVideoOptions;
// Initialise the video decoding objects. Even if we are not sourcing video
// we need to be ready to receive and render.
_vpxDecoder = new VpxEncoder();
int res = _vpxDecoder.InitDecoder();
if (res != 0)
{
throw new ApplicationException("VPX decoder initialisation failed.");
}
_imgConverter = new ImageConvert();
if (_audioOpts.AudioSource != AudioSourcesEnum.None)
{
var pcmu = new SDPMediaFormat(SDPMediaFormatsEnum.PCMU);
// RTP event support.
int clockRate = pcmu.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat> {
pcmu, rtpEventFormat
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, audioCapabilities);
addTrack(audioTrack);
}
if (_videoOpts.VideoSource != VideoSourcesEnum.None)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
addTrack(videoTrack);
}
// Where the magic (for processing received media) happens.
base.OnRtpPacketReceived += RtpPacketReceived;
}
/// <summary>
/// Initialise the audio capture and render device.
/// </summary>
/// <param name="audioSourceOpts">The options that dictate the type of audio source to use.</param>
/// <param name="sendingFormat">The codec that will be sued to send the audio.</param>
private void SetAudioSource(AudioOptions audioSourceOpts, SDPMediaFormat sendingFormat)
{
uint sampleRate = (uint)SDPMediaFormatInfo.GetClockRate(sendingFormat.FormatCodec);
uint rtpTimestamptRate = (uint)SDPMediaFormatInfo.GetRtpClockRate(sendingFormat.FormatCodec);
_rtpAudioTimestampPeriod = rtpTimestamptRate * AUDIO_SAMPLE_PERIOD_MILLISECONDS / 1000;
WaveFormat waveFormat = new WaveFormat((int)sampleRate, BITS_PER_SAMPLE, CHANNEL_COUNT);
// Render device.
if (_waveOutEvent == null)
{
_waveOutEvent = new WaveOutEvent();
_waveOutEvent.DeviceNumber = (_audioOpts != null) ? _audioOpts.OutputDeviceIndex : AudioOptions.DEFAULT_OUTPUTDEVICE_INDEX;
_waveProvider = new BufferedWaveProvider(waveFormat);
_waveProvider.DiscardOnBufferOverflow = true;
_waveOutEvent.Init(_waveProvider);
}
// Audio source.
if (!_disableExternalAudioSource)
{
if (_waveInEvent == null)
{
if (WaveInEvent.DeviceCount > 0)
{
_waveInEvent = new WaveInEvent();
_waveInEvent.BufferMilliseconds = AUDIO_SAMPLE_PERIOD_MILLISECONDS;
_waveInEvent.NumberOfBuffers = INPUT_BUFFERS;
_waveInEvent.DeviceNumber = 0;
_waveInEvent.WaveFormat = waveFormat;
_waveInEvent.DataAvailable += LocalAudioSampleAvailable;
}
else
{
Log.LogWarning("No audio capture devices are available. No audio stream will be sent.");
}
}
}
}
/// <summary>
/// Sets or updates the sources of the audio and/or video streams.
/// </summary>
/// <param name="audioOptions">Optional. If audio is being switched the new source options.
/// Set to null to leave the audio source unchanged.</param>
/// <param name="videoOptions">Optional. If video is being switched the new source options.
/// Set to null to leave the video source unchanged.</param>
/// <param name="disableExternalAudioSource">If true then no attempt will be made to use an external audio
/// source, e.g. microphone.</param>
public async Task SetSources(AudioOptions audioOptions, VideoOptions videoOptions, bool disableExternalAudioSource = false)
{
_disableExternalAudioSource = disableExternalAudioSource;
// Check whether the underlying media session has changed which dictates whether
// an audio or video source needs to be removed.
if (!HasAudio)
{
// Overrule any application supplied options as the session does not currently support audio.
audioOptions = new AudioOptions {
AudioSource = AudioSourcesEnum.None
};
}
if (!HasVideo)
{
// Overrule any application supplied options as the session does not currently support video.
videoOptions = new VideoOptions {
VideoSource = VideoSourcesEnum.None
};
}
if (audioOptions == null)
{
// Do nothing, audio source not being changed.
}
else if (audioOptions.AudioSource == AudioSourcesEnum.None)
{
// Audio source no longer required.
_waveInEvent?.StopRecording();
if (_audioStreamTimer != null)
{
_audioStreamTimer?.Dispose();
// Give any currently executing audio sampling time to complete.
await Task.Delay(AUDIO_SAMPLE_PERIOD_MILLISECONDS * 2).ConfigureAwait(false);
}
_audioStreamReader?.Close();
_audioOpts = audioOptions;
}
else
{
_sendingAudioFormat = base.GetSendingFormat(SDPMediaTypesEnum.audio);
SetAudioSource(audioOptions, _sendingAudioFormat);
_audioOpts = audioOptions;
StartAudio();
}
if (videoOptions == null)
{
// Do nothing, video source not being changed.
}
else if (videoOptions.VideoSource == VideoSourcesEnum.None)
{
// Video source no longer required.
_testPatternVideoSource?.Stop();
if (_videoOpts.BitmapSource != null)
{
_videoOpts.BitmapSource.OnBitmap -= LocalBitmapAvailable;
}
_videoOpts = videoOptions;
}
else
{
await SetVideoSource(videoOptions).ConfigureAwait(false);
_videoOpts = videoOptions;
StartVideo();
}
}
/// <summary>
/// Creates a new RTP audio visual session with audio/video capturing and rendering capabilities.
/// </summary>
/// <param name="addrFamily">The address family to create the underlying socket on (IPv4 or IPv6).</param>
/// <param name="audioOptions">Options for the send and receive audio streams on this session.</param>
/// <param name="videoOptions">Options for the send and receive video streams on this session</param>
/// <param name="bindAddress">Optional. If specified this address will be used as the bind address for any RTP
/// and control sockets created. Generally this address does not need to be set. The default behaviour
/// is to bind to [::] or 0.0.0.0, depending on system support, which minimises network routing
/// causing connection issues.</param>
/// <param name="disableExternalAudioSource">If true then no attempt will be made to use an external audio
/// source, e.g. microphone.</param>
public RtpAVSession(AudioOptions audioOptions, VideoOptions videoOptions, IPAddress bindAddress = null, bool disableExternalAudioSource = false)
: base(false, false, false, bindAddress)
{
_audioOpts = audioOptions ?? DefaultAudioOptions;
_videoOpts = videoOptions ?? DefaultVideoOptions;
_disableExternalAudioSource = disableExternalAudioSource;
if (_audioOpts != null && _audioOpts.AudioCodecs != null &&
_audioOpts.AudioCodecs.Any(x => !(x == SDPMediaFormatsEnum.PCMU || x == SDPMediaFormatsEnum.PCMA || x == SDPMediaFormatsEnum.G722)))
{
throw new ApplicationException("Only PCMA, PCMU and G722 are supported for audio codec options.");
}
// Initialise the video decoding objects. Even if we are not sourcing video
// we need to be ready to receive and render.
_vpxDecoder = new VpxEncoder();
int res = _vpxDecoder.InitDecoder();
if (res != 0)
{
throw new ApplicationException("VPX decoder initialisation failed.");
}
_imgConverter = new ImageConvert();
if (_audioOpts.AudioSource != AudioSourcesEnum.None)
{
var pcmu = new SDPMediaFormat(SDPMediaFormatsEnum.PCMU);
//// RTP event support.
//int clockRate = pcmu.GetClockRate();
//SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
//rtpEventFormat.SetFormatAttribute($"{SDP.TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
//rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat>();
if (_audioOpts.AudioCodecs == null || _audioOpts.AudioCodecs.Count == 0)
{
audioCapabilities.Add(pcmu);
}
else
{
foreach (var codec in _audioOpts.AudioCodecs)
{
audioCapabilities.Add(new SDPMediaFormat(codec));
}
}
//audioCapabilities.Add(rtpEventFormat);
if (audioCapabilities.Any(x => x.FormatCodec == SDPMediaFormatsEnum.G722))
{
_g722Encode = new G722Codec();
_g722EncodeState = new G722CodecState(64000, G722Flags.None);
_g722Decode = new G722Codec();
_g722DecodeState = new G722CodecState(64000, G722Flags.None);
}
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, audioCapabilities);
addTrack(audioTrack);
}
if (_videoOpts.VideoSource != VideoSourcesEnum.None)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
addTrack(videoTrack);
}
// Where the magic (for processing received media) happens.
base.OnRtpPacketReceived += RtpPacketReceived;
}
/// <summary>
/// Initialise the audio capture and render device.
/// </summary>
private void SetAudioSource(AudioOptions audioSourceOpts)
{
// Render device.
if (_waveOutEvent == null)
{
_waveOutEvent = new WaveOutEvent();
_waveProvider = new BufferedWaveProvider(_waveFormat);
_waveProvider.DiscardOnBufferOverflow = true;
_waveOutEvent.Init(_waveProvider);
}
// Audio source.
if (audioSourceOpts.AudioSource == AudioSourcesEnum.Microphone)
{
if (_waveInEvent == null)
{
if (WaveInEvent.DeviceCount > 0)
{
_waveInEvent = new WaveInEvent();
_waveInEvent.BufferMilliseconds = AUDIO_SAMPLE_PERIOD_MILLISECONDS;
_waveInEvent.NumberOfBuffers = 1;
_waveInEvent.DeviceNumber = 0;
_waveInEvent.WaveFormat = _waveFormat;
_waveInEvent.DataAvailable += LocalAudioSampleAvailable;
}
else
{
Log.LogWarning("No audio capture devices are available. No audio stream will be sent.");
}
}
}
else if (audioSourceOpts.AudioSource == AudioSourcesEnum.Music)
{
string newAudioFile = audioSourceOpts.SourceFile ?? DEFAULT_AUDIO_SOURCE_FILE;
// Check whether this is the initial load or whether the source file is the same. If it is there's no need to do anything.
if (_audioStreamReader == null || newAudioFile != _audioOpts.SourceFile)
{
if (!File.Exists(newAudioFile))
{
if (File.Exists(DEFAULT_AUDIO_SOURCE_FILE))
{
Log.LogWarning($"The requested audio source file could not be found {newAudioFile}, falling back to default.");
newAudioFile = DEFAULT_AUDIO_SOURCE_FILE;
}
else
{
Log.LogError($"The requested audio source file could not be found {newAudioFile}, no audio source will be initialised.");
newAudioFile = null;
}
}
if (newAudioFile != null)
{
_audioStreamReader = new StreamReader(newAudioFile);
}
}
}
if (_rtpAudioTimestampPeriod == 0)
{
_rtpAudioTimestampPeriod = (uint)(SDPMediaFormatInfo.GetClockRate(SDPMediaFormatsEnum.PCMU) / AUDIO_SAMPLE_PERIOD_MILLISECONDS);
}
}