/// <summary>
/// Initializes the audio renderer.
/// Call the Play Method to start reading samples.
/// </summary>
private void Initialize()
{
Destroy();
// Check the wave format
if (WaveFormat.BitsPerSample != 16 || WaveFormat.Channels != 2)
{
throw new NotSupportedException("Wave Format has to be 16-bit and 2-channel.");
}
// Release the audio device always upon exiting
if (Application.Current is Application app)
{
app.Dispatcher?.BeginInvoke(new Action(() => { app.Exit += OnApplicationExit; }));
}
// Enumerate devices. The default device is the first one so we check
// that we have more than 1 device (other than the default stub)
var hasAudioDevices = MediaElement.RendererOptions.UseLegacyAudioOut ?
LegacyAudioPlayer.EnumerateDevices().Count > 1 :
DirectSoundPlayer.EnumerateDevices().Count > 1;
// Check if we have an audio output device.
if (hasAudioDevices == false)
{
HasFiredAudioDeviceStopped = true;
this.LogWarning(Aspects.AudioRenderer,
"No audio device found for output.");
return;
}
// Initialize the SoundTouch Audio Processor (if available)
AudioProcessor = (SoundTouch.IsAvailable == false) ? null : new SoundTouch();
if (AudioProcessor != null)
{
AudioProcessor.SetChannels(Convert.ToUInt32(WaveFormat.Channels));
AudioProcessor.SetSampleRate(Convert.ToUInt32(WaveFormat.SampleRate));
}
// Initialize the Audio Device
AudioDevice = MediaElement.RendererOptions.UseLegacyAudioOut ?
new LegacyAudioPlayer(this, MediaElement.RendererOptions.LegacyAudioDevice?.DeviceId ?? -1) as IWavePlayer :
new DirectSoundPlayer(this, MediaElement.RendererOptions.DirectSoundDevice?.DeviceId ?? DirectSoundPlayer.DefaultPlaybackDeviceId);
// Create the Audio Buffer
SampleBlockSize = Constants.AudioBytesPerSample * Constants.AudioChannelCount;
var bufferLength = WaveFormat.ConvertMillisToByteSize(2000); // 2-second buffer
AudioBuffer = new CircularBuffer(bufferLength);
AudioDevice.Start();
}
/// <summary>
/// Initializes the audio renderer.
/// Call the Play Method to start reading samples
/// </summary>
private void Initialize()
{
Destroy();
// Release the audio device always upon exiting
if (Application.Current != null)
{
GuiContext.Current.EnqueueInvoke(DispatcherPriority.Render, () =>
Application.Current.Exit += OnApplicationExit);
}
// Enumerate devices. The default device is the first one so we check
// that we have more than 1 device (other than the default stub)
var hasAudioDevices = MediaElement.RendererOptions.UseLegacyAudioOut ?
LegacyAudioPlayer.EnumerateDevices().Count > 1 :
DirectSoundPlayer.EnumerateDevices().Count > 1;
// Check if we have an audio output device.
if (hasAudioDevices == false)
{
WaitForReadyEvent.Complete();
HasFiredAudioDeviceStopped = true;
this.LogWarning(Aspects.AudioRenderer,
"No audio device found for output.");
return;
}
// Initialize the SoundTouch Audio Processor (if available)
AudioProcessor = (SoundTouch.IsAvailable == false) ? null : new SoundTouch();
if (AudioProcessor != null)
{
AudioProcessor.SetChannels(Convert.ToUInt32(WaveFormat.Channels));
AudioProcessor.SetSampleRate(Convert.ToUInt32(WaveFormat.SampleRate));
}
// Initialize the Audio Device
AudioDevice = MediaElement.RendererOptions.UseLegacyAudioOut ?
new LegacyAudioPlayer(this, MediaElement.RendererOptions.LegacyAudioDevice?.DeviceId ?? -1) as IWavePlayer :
new DirectSoundPlayer(this, MediaElement.RendererOptions.DirectSoundDevice?.DeviceId ?? DirectSoundPlayer.DefaultPlaybackDeviceId);
// Create the Audio Buffer
SampleBlockSize = Constants.Audio.BytesPerSample * Constants.Audio.ChannelCount;
var bufferLength = WaveFormat.ConvertMillisToByteSize(2000); // 2-second buffer
AudioBuffer = new CircularBuffer(bufferLength);
AudioDevice.Start();
}
/// <summary>
/// Initializes the audio renderer.
/// Call the Play Method to start reading samples
/// </summary>
private void Initialize()
{
Destroy();
// Enumerate devices. The default device is the first one so we check
// that we have more than 1 device (other than the default stub)
var hasAudioDevices = MediaElement.RendererOptions.UseLegacyAudioOut ?
LegacyAudioPlayer.EnumerateDevices().Count > 1 :
DirectSoundPlayer.EnumerateDevices().Count > 1;
// Check if we have an audio output device.
if (hasAudioDevices == false)
{
WaitForReadyEvent = null;
MediaCore.Log(MediaLogMessageType.Warning,
$"AUDIO OUT: No audio device found for output.");
return;
}
// Initialize the SoundTouch Audio Processor (if available)
AudioProcessor = (SoundTouch.IsAvailable == false) ? null : new SoundTouch
{
Channels = Convert.ToUInt32(WaveFormat.Channels),
SampleRate = Convert.ToUInt32(WaveFormat.SampleRate)
};
// Initialize the Audio Device
AudioDevice = MediaElement.RendererOptions.UseLegacyAudioOut ?
new LegacyAudioPlayer(this, MediaElement.RendererOptions.LegacyAudioDevice?.DeviceId ?? -1) as IWavePlayer :
new DirectSoundPlayer(this, MediaElement.RendererOptions.DirectSoundDevice?.DeviceId ?? DirectSoundPlayer.DefaultPlaybackDeviceId);
// Create the Audio Buffer
SampleBlockSize = Constants.Audio.BytesPerSample * Constants.Audio.ChannelCount;
var bufferLength = WaveFormat.ConvertMillisToByteSize(2000); // 2-second buffer
AudioBuffer = new CircularBuffer(bufferLength);
AudioDevice.Start();
}
/// <summary>
/// Initializes the audio renderer.
/// Call the Play Method to start reading samples
/// </summary>
private void Initialize()
{
Destroy();
if (SoundTouch.IsAvailable)
{
AudioProcessor = new SoundTouch
{
Channels = Convert.ToUInt32(WaveFormat.Channels),
SampleRate = Convert.ToUInt32(WaveFormat.SampleRate)
};
}
AudioDevice = MediaElement.RendererOptions.UseLegacyAudioOut ?
new LegacyAudioPlayer(this, MediaElement.RendererOptions.LegacyAudioDevice?.DeviceId ?? -1) as IWavePlayer :
new DirectSoundPlayer(this, MediaElement.RendererOptions.DirectSoundDevice?.DeviceId ?? DirectSoundPlayer.DefaultPlaybackDeviceId);
SampleBlockSize = Constants.Audio.BytesPerSample * Constants.Audio.ChannelCount;
var bufferLength = WaveFormat.ConvertMillisToByteSize(AudioDevice.DesiredLatency) * MediaCore.Blocks[MediaType.Audio].Capacity / 2;
AudioBuffer = new CircularBuffer(bufferLength);
AudioDevice.Start();
}
private bool Synchronize(byte[] targetBuffer, int targetBufferOffset, int requestedBytes, double speedRatio)
{
#region Documentation
/*
* Wikipedia says:
* For television applications, audio should lead video by no more than 15 milliseconds and audio should
* lag video by no more than 45 milliseconds. For film, acceptable lip sync is considered to be no more
* than 22 milliseconds in either direction.
*
* The Media and Acoustics Perception Lab says:
* The results of the experiment determined that the average audio leading threshold for a/v sync
* detection was 185.19 ms, with a standard deviation of 42.32 ms
*
* The ATSC says:
* At first glance it seems loose: +90 ms to -185 ms as a Window of Acceptability
* - Undetectable from -100 ms to +25 ms
* - Detectable at -125 ms & +45 ms
* - Becomes unacceptable at -185 ms & +90 ms
*
* NOTE: (- Sound delayed, + Sound advanced)
*/
#endregion
#region Private State
var audioLatencyMs = Latency.TotalMilliseconds;
var isBeyondThreshold = false;
var readableCount = AudioBuffer.ReadableCount;
var rewindableCount = AudioBuffer.RewindableCount;
#endregion
#region Sync Give-up Conditions
if (MediaElement.RendererOptions.AudioDisableSync)
{
return(true);
}
// Determine if we should continue to perform syncs.
// Some live, non-seekable streams will send out-of-sync audio packets
// and after trying too many times we simply give up.
// The Sync conditions are reset in the Update method.
if (MediaCore.State.IsSeekable == false && PlaySyncGaveUp.Value == false)
{
// 1. Determine if a sync is required
if (audioLatencyMs > SyncThresholdLagging ||
audioLatencyMs < SyncThresholdLeading ||
Math.Abs(audioLatencyMs) > SyncThresholdPerfect)
{
PlaySyncCount++;
}
// 2. Compute the variables to determine give-up conditions
var playbackElapsedSeconds = PlaySyncStartTime.HasValue == false ?
0 : DateTime.UtcNow.Subtract(PlaySyncStartTime.Value).TotalSeconds;
var syncsPerSecond = PlaySyncCount / playbackElapsedSeconds;
// 3. Determine if we need to give up
if (playbackElapsedSeconds >= 3 && syncsPerSecond >= 3)
{
this.LogWarning(Aspects.AudioRenderer,
$"SYNC AUDIO: GIVE UP | SECS: {playbackElapsedSeconds:0.00}; " +
$"SYN: {PlaySyncCount}; RATE: {syncsPerSecond:0.00} SYN/s; LAT: {audioLatencyMs} ms.");
PlaySyncGaveUp.Value = true;
}
}
// Detect if we have given up
if (PlaySyncGaveUp == true)
{
return(true);
}
#endregion
#region Large Latency Handling
if (audioLatencyMs > SyncThresholdLagging)
{
isBeyondThreshold = true;
// a positive audio latency means we are rendering audio behind (after) the clock (skip some samples)
// and therefore we need to advance the buffer before we read from it.
if (Math.Abs(speedRatio - 1.0) <= double.Epsilon)
{
this.LogWarning(Aspects.AudioRenderer,
$"SYNC AUDIO: LATENCY: {audioLatencyMs} ms. | SKIP (samples being rendered too late)");
}
// skip some samples from the buffer.
var audioLatencyBytes = WaveFormat.ConvertMillisToByteSize(Convert.ToInt32(Math.Ceiling(audioLatencyMs)));
AudioBuffer.Skip(Math.Min(audioLatencyBytes, readableCount));
}
else if (audioLatencyMs < SyncThresholdLeading)
{
isBeyondThreshold = true;
// Compute the latency in bytes
var audioLatencyBytes = WaveFormat.ConvertMillisToByteSize(Convert.ToInt32(Math.Ceiling(Math.Abs(audioLatencyMs))));
// audioLatencyBytes = requestedBytes; // uncomment this line to enable rewinding.
if (audioLatencyBytes > requestedBytes && audioLatencyBytes < rewindableCount)
{
// This means we have the audio pointer a little too ahead of time and we need to
// rewind it the requested amount of bytes.
AudioBuffer.Rewind(Math.Min(audioLatencyBytes, rewindableCount));
}
else
{
// a negative audio latency means we are rendering audio ahead (before) the clock
// and therefore we need to render some silence until the clock catches up
if (Math.Abs(speedRatio - 1.0) <= double.Epsilon)
{
this.LogWarning(Aspects.AudioRenderer,
$"SYNC AUDIO: LATENCY: {audioLatencyMs} ms. | WAIT (samples being rendered too early)");
}
// render silence for the wait time and return
Array.Clear(targetBuffer, targetBufferOffset, requestedBytes);
return(false);
}
}
#endregion
#region Small Latency Handling
// Check if minor adjustment is necessary
if (MediaCore.State.HasVideo == false || Math.Abs(speedRatio - 1.0) > double.Epsilon ||
isBeyondThreshold || Math.Abs(audioLatencyMs) <= SyncThresholdPerfect)
{
return(true);
}
// Perform minor adjustments until the delay is less than 10ms in either direction
var stepDurationMillis = Convert.ToInt32(Math.Min(SyncThresholdMaxStep, Math.Abs(audioLatencyMs)));
var stepDurationBytes = WaveFormat.ConvertMillisToByteSize(stepDurationMillis);
if (audioLatencyMs > SyncThresholdPerfect)
{
AudioBuffer.Skip(Math.Min(stepDurationBytes, readableCount));
}
else if (audioLatencyMs < -SyncThresholdPerfect)
{
AudioBuffer.Rewind(Math.Min(stepDurationBytes, rewindableCount));
}
#endregion
return(true);
}
private bool Synchronize(byte[] targetBuffer, int targetBufferOffset, int requestedBytes, double speedRatio)
{
#region Documentation
/*
* Wikipedia says:
* For television applications, audio should lead video by no more than 15 milliseconds and audio should
* lag video by no more than 45 milliseconds. For film, acceptable lip sync is considered to be no more
* than 22 milliseconds in either direction.
*
* The Media and Acoustics Perception Lab says:
* The results of the experiment determined that the average audio leading threshold for a/v sync
* detection was 185.19 ms, with a standard deviation of 42.32 ms
*
* The ATSC says:
* At first glance it seems loose: +90 ms to -185 ms as a Window of Acceptability
* - Undetectable from -100 ms to +25 ms
* - Detectable at -125 ms & +45 ms
* - Becomes unacceptable at -185 ms & +90 ms
*
* NOTE: (- Sound delayed, + Sound advanced)
*/
#endregion
var hardwareLatencyMs = WaveFormat.ConvertByteSizeToDuration(requestedBytes).TotalMilliseconds;
var bufferLatencyMs = BufferLatency.TotalMilliseconds; // we want the buffer latency to be the negative of the device latency
var minAcceptableLeadMs = -1.5 * hardwareLatencyMs; // less than this and we need to rewind samples
var maxAcceptableLagMs = -0.5 * hardwareLatencyMs; // more than this and we need to skip samples
var isLoggingEnabled = Math.Abs(speedRatio - 1.0) <= double.Epsilon;
var operationName = string.Empty;
try
{
RealTimeLatency = default;
// we don't want to perform AV sync if the latency is huge
// or if we have simply disabled it
if (MediaElement.RendererOptions.AudioDisableSync)
{
return(true);
}
// The ideal target latency is the negative of the audio device's desired latency.
// this is approximately -40ms (i.e. have the buffer position 40ms ahead (negative lag) of the playback clock
// so that samples are rendered right on time.)
if (bufferLatencyMs >= minAcceptableLeadMs && bufferLatencyMs <= maxAcceptableLagMs)
{
return(true);
}
if (bufferLatencyMs > maxAcceptableLagMs)
{
// this is the case where the buffer latency is too positive (i.e. buffer is lagging by too much)
// the goal is to skip some samples to make the buffer latency approximately that of the hardware latency
// so that the buffer leads by the hardware lag and we get sync-perferct results.
var audioLatencyBytes = WaveFormat.ConvertMillisToByteSize(bufferLatencyMs + hardwareLatencyMs);
if (AudioBuffer.ReadableCount > audioLatencyBytes)
{
operationName = "SKIP OK ";
AudioBuffer.Skip(audioLatencyBytes);
return(true);
}
// render silence and return
operationName = "SKIP ERR";
Array.Clear(targetBuffer, targetBufferOffset, requestedBytes);
return(false);
}
else if (bufferLatencyMs < minAcceptableLeadMs)
{
// this is the case where the buffer latency is too negative (i.e. buffer is leading by too much)
// the goal is to rewind some samples to make the buffer latency approximately that of the hardware latency
// so that the buffer leads by the hardware lag and we get sync-perferct results.
var audioLatencyBytes = WaveFormat.ConvertMillisToByteSize(Math.Abs(bufferLatencyMs) + hardwareLatencyMs);
if (AudioBuffer.RewindableCount > audioLatencyBytes)
{
operationName = "RWND OK ";
AudioBuffer.Rewind(audioLatencyBytes);
return(true);
}
// render silence and return
operationName = "RWND ERR";
Array.Clear(targetBuffer, targetBufferOffset, requestedBytes);
return(false);
}
}
finally
{
RealTimeLatency = BufferLatency + TimeSpan.FromMilliseconds(hardwareLatencyMs);
if (isLoggingEnabled && !string.IsNullOrWhiteSpace(operationName))
{
this.LogWarning(Aspects.AudioRenderer,
$"SYNC AUDIO: {operationName} | Initial: {bufferLatencyMs:0} ms. Current: {BufferLatency.TotalMilliseconds:0} ms. Device: {hardwareLatencyMs:0} ms.");
}
}
return(true);
}
