private void InitializeCaptureSource()
{
if (captureSource == null)
{
mediaElement = new MediaElement();
audioStreamSource = new TestAudioStreamSource(this);
mediaElement.SetSource(audioStreamSource);
// Set the audio properties.
captureSource = new CaptureSource();
captureSource.AudioCaptureDevice = CaptureDeviceConfiguration.GetDefaultAudioCaptureDevice();
if (captureSource.AudioCaptureDevice != null)
{
MediaDeviceConfig.SelectBestAudioFormat(captureSource.AudioCaptureDevice);
if (captureSource.AudioCaptureDevice.DesiredFormat != null)
{
captureSource.AudioCaptureDevice.AudioFrameSize = AudioFormat.Default.MillisecondsPerFrame; // 20 milliseconds
audioSink = new TestAudioSinkAdapter(captureSource);
audioSink.ProcessedFrameAvailable += audioSink_FrameArrived;
ClientLogger.Debug("CaptureSource initialized.");
}
else
{
ClientLogger.Debug("No suitable audio format was found.");
}
}
else
{
// Do something more here eventually, once we figure out what the user experience should be.
ClientLogger.Debug("No audio capture device was found.");
}
}
}
private void InitializeCaptureSource()
{
if (_captureSource != null)
{
return;
}
// Setup the capture source (for recording audio)
_captureSource = new CaptureSource();
_captureSource.AudioCaptureDevice = CaptureDeviceConfiguration.GetDefaultAudioCaptureDevice();
if (_captureSource.AudioCaptureDevice != null)
{
MediaDeviceConfig.SelectBestAudioFormat(_captureSource.AudioCaptureDevice);
if (_captureSource.AudioCaptureDevice.DesiredFormat != null)
{
var mediaStats = new MediaStatistics();
var mediaEnvironment = new MediaEnvironment(mediaStats);
_captureSource.AudioCaptureDevice.AudioFrameSize = AudioFormat.Default.MillisecondsPerFrame; // 20 milliseconds
_audioSinkAdapter = new MultipleControllerAudioSinkAdapter(GetMediaConfig(), _captureSource, 2000);
_mediaStreamSource = new MultipleControllerAudioMediaStreamSource(2000);
ClientLogger.Debug("CaptureSource initialized.");
}
else
{
ClientLogger.Debug("No suitable audio format was found.");
}
}
else
{
// Do something more here eventually, once we figure out what the user experience should be.
ClientLogger.Debug("No audio capture device was found.");
}
}
private void InitializeCaptureSource()
{
if (_captureSource == null)
{
// Setup the capture source (for recording audio)
_captureSource = new CaptureSource();
_captureSource.AudioCaptureDevice = CaptureDeviceConfiguration.GetDefaultAudioCaptureDevice();
if (_captureSource.AudioCaptureDevice != null)
{
MediaDeviceConfig.SelectBestAudioFormat(_captureSource.AudioCaptureDevice);
if (_captureSource.AudioCaptureDevice.DesiredFormat != null)
{
_captureSource.AudioCaptureDevice.AudioFrameSize = AudioFormat.Default.MillisecondsPerFrame; // 20 milliseconds
_audioSink = new AudioSinkAdapter(_captureSource, null, MediaConfig.Default, new TestMediaEnvironment(), AudioFormat.Default);
_recorder = new RecorderBase(_captureSource, _audioSink, speakersAudioVisualizer);
chkSynchronizeRecording.DataContext = _audioSink;
ClientLogger.Debug("CaptureSource initialized.");
}
else
{
ClientLogger.Debug("No suitable audio format was found.");
}
panelMicrophone.DataContext = _captureSource;
}
else
{
// Do something more here eventually, once we figure out what the user experience should be.
ClientLogger.Debug("No audio capture device was found.");
}
}
}
private void InitializeCaptureSource()
{
if (captureSource != null)
{
captureSource.Stop();
}
captureSource = new CaptureSource();
captureSource.AudioCaptureDevice = (AudioCaptureDevice)listBoxAudioSources.SelectedItem;
MediaDeviceConfig.SelectBestAudioFormat(captureSource.AudioCaptureDevice);
captureSource.AudioCaptureDevice.DesiredFormat = captureSource.AudioCaptureDevice.SupportedFormats
.First(format => format.BitsPerSample == AudioConstants.BitsPerSample &&
format.WaveFormat == WaveFormatType.Pcm &&
format.Channels == 1 &&
format.SamplesPerSecond == sampleRate);
captureSource.AudioCaptureDevice.AudioFrameSize = AudioFormat.Default.MillisecondsPerFrame; // 20 milliseconds
audioSink = new TestAudioSinkAdapter(captureSource, new NullAudioController());
audioSink.RawFrameAvailable += audioSink_RawFrameAvailable;
audioSink.ProcessedFrameAvailable += audioSink_FrameArrived;
ClientLogger.Debug("Checking device access.");
if (CaptureDeviceConfiguration.AllowedDeviceAccess || CaptureDeviceConfiguration.RequestDeviceAccess())
{
savedFramesForDebug = new List <byte[]>();
captureSource.Start();
ClientLogger.Debug("CaptureSource started.");
}
}
private void InitializeCaptureSource()
{
ClientLogger.Debug("AudioLoopbackTest:InitializeCaptureSource()");
if (_captureSource != null)
{
_captureSource.Stop();
}
_captureSource = new CaptureSource();
_captureSource.AudioCaptureDevice = (AudioCaptureDevice)lstAudioInputDevices.SelectedItem;
_captureSource.VideoCaptureDevice = (VideoCaptureDevice)lstVideoInputDevices.SelectedItem;
_mediaElement = new MediaElement();
_audioStreamSource = new TestAudioStreamSource(this);
// Set the audio properties.
if (_captureSource.AudioCaptureDevice != null)
{
MediaDeviceConfig.SelectBestAudioFormat(_captureSource.AudioCaptureDevice);
if (_captureSource.AudioCaptureDevice.DesiredFormat != null)
{
_captureSource.AudioCaptureDevice.AudioFrameSize = _audioFormat.MillisecondsPerFrame; // 20 milliseconds
_audioSink = new TestAudioSinkAdapter(_captureSource, new NullAudioController());
_audioSink.RawFrameAvailable += audioSink_RawFrameAvailable;
_audioSink.ProcessedFrameAvailable += audioSink_FrameArrived;
ClientLogger.Debug("CaptureSource initialized.");
}
else
{
ClientLogger.Debug("No suitable audio format was found.");
}
ClientLogger.Debug("Checking device access.");
if (CaptureDeviceConfiguration.AllowedDeviceAccess || CaptureDeviceConfiguration.RequestDeviceAccess())
{
ClientLogger.Debug("AudioLoopbackTest CaptureSource starting with audio device {0}, video device {1}.",
_captureSource.AudioCaptureDevice.FriendlyName, _captureSource.VideoCaptureDevice.FriendlyName);
_captureSource.Start();
ClientLogger.Debug("CaptureSource started; setting media element source.");
_mediaElement.SetSource(_audioStreamSource);
ClientLogger.Debug("MediaElement source set; telling media element to play.");
_mediaElement.Play();
}
}
else
{
// Do something more here eventually, once we figure out what the user experience should be.
ClientLogger.Debug("No audio capture device was found.");
}
}
public void CreateAudioContexts()
{
_captureSource.VideoCaptureDevice = null;
if (_captureSource.AudioCaptureDevice == null)
{
_captureSource.AudioCaptureDevice = CaptureDeviceConfiguration.GetDefaultAudioCaptureDevice();
if (_captureSource.AudioCaptureDevice == null)
{
throw new InvalidOperationException("No suitable audio capture device was found");
}
}
MediaDeviceConfig.SelectBestAudioFormat(_captureSource.AudioCaptureDevice);
_captureSource.AudioCaptureDevice.AudioFrameSize = AudioFormat.Default.MillisecondsPerFrame; // 20 milliseconds
var desiredFormat = _captureSource.AudioCaptureDevice.DesiredFormat;
var rawAudioFormat = new AudioFormat(desiredFormat.SamplesPerSecond, AudioFormat.Default.MillisecondsPerFrame, desiredFormat.Channels, desiredFormat.BitsPerSample);
var playedAudioFormat = new AudioFormat();
var config = MediaConfig.Default;
// Absolutely bare minimum processing - doesn't process sound at all.
var nullAudioFormat = new AudioFormat();
var nullResampler = new ResampleFilter(rawAudioFormat, nullAudioFormat);
nullResampler.InstanceName = "Null resample filter";
var nullEnhancer = new NullEchoCancelFilter(config.ExpectedAudioLatency, config.FilterLength, nullAudioFormat, playedAudioFormat);
nullEnhancer.InstanceName = "Null";
var nullDtx = new NullAudioInplaceFilter();
var nullEncoder = new NullAudioEncoder();
var nullAudioContext = new AudioContext(nullAudioFormat, nullResampler, nullDtx, nullEnhancer, nullEncoder);
nullAudioContext.Description = "Null";
// What we should use when there's only one other person, and CPU is OK:
// 16Khz, Speex, WebRtc at full strength
var directAudioFormat = new AudioFormat();
var directResampler = new ResampleFilter(rawAudioFormat, directAudioFormat);
directResampler.InstanceName = "Direct high quality resample filter";
var directEnhancer = new WebRtcFilter(config.ExpectedAudioLatency, config.FilterLength, directAudioFormat, playedAudioFormat, config.EnableAec, config.EnableDenoise, config.EnableAgc);
directEnhancer.InstanceName = "High";
var directDtx = new DtxFilter(directAudioFormat);
var directEncoder = new SpeexEncoder(directAudioFormat);
var directAudioContext = new AudioContext(directAudioFormat, directResampler, directDtx, directEnhancer, directEncoder);
directAudioContext.Description = "High Quality Direct";
// What we should use when there are multiple people (and hence the audio will need to be decoded and mixed), but CPU is OK:
// 8Khz, G711, WebRtc at full strength
var conferenceAudioFormat = new AudioFormat(AudioConstants.NarrowbandSamplesPerSecond);
var conferenceResampler = new ResampleFilter(rawAudioFormat, conferenceAudioFormat);
conferenceResampler.InstanceName = "Conference high quality resample filter";
var conferenceEnhancer = new WebRtcFilter(config.ExpectedAudioLatency, config.FilterLength, conferenceAudioFormat, playedAudioFormat, config.EnableAec, config.EnableDenoise, config.EnableAgc);
conferenceEnhancer.InstanceName = "Medium";
var conferenceDtx = new DtxFilter(conferenceAudioFormat);
var conferenceEncoder = new G711MuLawEncoder(conferenceAudioFormat);
var conferenceAudioContext = new AudioContext(conferenceAudioFormat, conferenceResampler, conferenceDtx, conferenceEnhancer, conferenceEncoder);
conferenceAudioContext.Description = "High Quality Conference";
// What we should use when one or more remote CPU's isn't keeping up (regardless of how many people are in the room):
// 8Khz, G711, WebRtc at full-strength
var remoteFallbackAudioFormat = new AudioFormat(AudioConstants.NarrowbandSamplesPerSecond);
var remoteFallbackResampler = new ResampleFilter(rawAudioFormat, remoteFallbackAudioFormat);
remoteFallbackResampler.InstanceName = "Fallback remote high cpu resample filter";
var remoteFallbackEnhancer = new WebRtcFilter(config.ExpectedAudioLatency, config.FilterLength, remoteFallbackAudioFormat, playedAudioFormat, config.EnableAec, config.EnableDenoise, config.EnableAgc);
remoteFallbackEnhancer.InstanceName = "Medium";
var remoteFallbackDtx = new DtxFilter(remoteFallbackAudioFormat);
var remoteFallbackEncoder = new G711MuLawEncoder(remoteFallbackAudioFormat);
var remoteFallbackAudioContext = new AudioContext(remoteFallbackAudioFormat, remoteFallbackResampler, remoteFallbackDtx, remoteFallbackEnhancer, remoteFallbackEncoder);
remoteFallbackAudioContext.Description = "Fallback for remote high CPU";
// What we should use when the local CPU isn't keeping up (regardless of how many people are in the room):
// 8Khz, G711, WebRtc at half-strength
var fallbackAudioFormat = new AudioFormat(AudioConstants.NarrowbandSamplesPerSecond);
var fallbackResampler = new ResampleFilter(rawAudioFormat, fallbackAudioFormat);
fallbackResampler.InstanceName = "Fallback resample filter";
var fallbackEnhancer = new WebRtcFilter(config.ExpectedAudioLatencyFallback, config.FilterLengthFallback, fallbackAudioFormat, playedAudioFormat, config.EnableAec, false, false);
fallbackEnhancer.InstanceName = "Low";
var fallbackDtx = new DtxFilter(fallbackAudioFormat);
var fallbackEncoder = new G711MuLawEncoder(fallbackAudioFormat);
var fallbackAudioContext = new AudioContext(fallbackAudioFormat, fallbackResampler, fallbackDtx, fallbackEnhancer, fallbackEncoder);
fallbackAudioContext.Description = "Fallback for local high CPU";
AudioContextCollection.Clear();
AudioContextCollection.Add(nullAudioContext);
AudioContextCollection.Add(directAudioContext);
AudioContextCollection.Add(conferenceAudioContext);
AudioContextCollection.Add(remoteFallbackAudioContext);
AudioContextCollection.Add(fallbackAudioContext);
CurrentAudioContext = nullAudioContext;
}
public void StartSendingAudioToRoom(string ownerUserTag, string host, List <byte[]> testFrames, bool sendLive, OperationCallback callback)
{
// What we should use when there's only one other person, and CPU is OK:
// 16Khz, Speex, WebRtc at full strength
var config = MediaConfig.Default;
config.LocalSsrcId = (ushort)rnd.Next(ushort.MinValue, ushort.MaxValue);
config.AudioContextSelection = AudioContextSelection.HighQualityDirect;
config.MediaServerHost = host;
// Create the media controller
var playedAudioFormat = new AudioFormat();
var mediaStatistics = new TimingMediaStatistics();
var mediaEnvironment = new TestMediaEnvironment();
var mediaConnection = new RtpMediaConnection(config, mediaStatistics);
var vqc = new VideoQualityController(MediaConfig.Default.LocalSsrcId);
_mediaController = new MediaController(MediaConfig.Default, playedAudioFormat, mediaStatistics, mediaEnvironment, mediaConnection, vqc);
// Create the audio sink adapter.
_captureSource = new CaptureSource();
_captureSource.VideoCaptureDevice = null;
if (_captureSource.AudioCaptureDevice == null)
{
_captureSource.AudioCaptureDevice = CaptureDeviceConfiguration.GetDefaultAudioCaptureDevice();
if (_captureSource.AudioCaptureDevice == null)
{
throw new InvalidOperationException("No suitable audio capture device was found");
}
}
MediaDeviceConfig.SelectBestAudioFormat(_captureSource.AudioCaptureDevice);
_captureSource.AudioCaptureDevice.AudioFrameSize = AudioFormat.Default.MillisecondsPerFrame; // 20 milliseconds
_audioSinkAdapter = sendLive
? new AudioSinkAdapter(_captureSource, _mediaController, config, mediaEnvironment, playedAudioFormat)
: new FromFileAudioSinkAdapter(_captureSource, _mediaController, config, mediaEnvironment, playedAudioFormat, testFrames);
var roomService = new RoomServiceAdapter();
roomService.CreateClient();
roomService.GetRoomId(Constants.DefaultCompanyTag, Constants.DefaultAuthenticationGroupTag, ownerUserTag, Constants.DefaultRoomName, (getRoomError, result) =>
{
if (getRoomError != null)
{
callback(getRoomError);
}
else
{
// Connect.
_mediaController.Connect(result.RoomId.ToString(), connectError => Deployment.Current.Dispatcher.BeginInvoke(() =>
{
if (connectError == null)
{
ClientLogger.Debug("MacTestViewModel connected to MediaController");
_captureSource.Start();
}
_mediaController.RegisterRemoteSession((ushort)(config.LocalSsrcId + 1));
callback(connectError);
}));
}
});
}