internal LocalVoiceFramed(VoiceClient voiceClient, IEncoder encoder, byte id, VoiceInfo voiceInfo, int channelId, int frameSize)
: base(voiceClient, encoder, id, voiceInfo, channelId, frameSize)
{
if (frameSize == 0)
{
throw new Exception(LogPrefix + ": non 0 frame size required for framed stream");
}
this.framer = new Framer <T>(FrameSize);
this.bufferFactory = new FactoryPrimitiveArrayPool <T>(DATA_POOL_CAPACITY, Name + " Data", FrameSize);
}
internal LocalVoiceFramed(VoiceClient voiceClient, IEncoder encoder, byte id, VoiceInfo voiceInfo, int channelId, int frameSize)
: base(voiceClient, encoder, id, voiceInfo, channelId, frameSize)
{
#if DUMP_TO_FILE
file = File.Open("dump-" + fileCnt++ + ".raw", FileMode.Create);
#endif
if (frameSize == 0)
{
throw new Exception(LogPrefix + ": non 0 frame size required for framed stream");
}
this.framer = new Framer <T>(FrameSize);
this.bufferFactory = new FactoryPrimitiveArrayPool <T>(DATA_POOL_CAPACITY, Name + " Data", FrameSize);
}
private void InitReverseStream()
{
lock (this)
{
if (!aecInited)
{
if (disposed)
{
return;
}
int size = processFrameSize * reverseSamplingRate / samplingRate * reverseChannels;
reverseFramer = new Framer <float>(size);
reverseBuf = new short[processFrameSize * reverseChannels / channels]; // should match direct stream
if (reverseSamplingRate != samplingRate)
{
logger.LogWarning("WebRTCAudioProcessor AEC: output sampling rate {0} != {1} capture sampling rate. For better AEC, set audio source (microphone) and audio output samping rates to the same value.", reverseSamplingRate, samplingRate);
}
aecInited = true;
}
}
}
private void InitReverseStream()
{
lock (this)
{
if (!aecInited)
{
if (disposed)
{
return;
}
int size = processFrameSize * reverseSamplingRate / samplingRate * reverseChannels;
reverseFramer = new Framer <float>(size);
reverseBufferFactory = new FactoryPrimitiveArrayPool <short>(REVERSE_BUFFER_POOL_CAPACITY, "WebRTCAudioProcessor Reverse Buffers", this.inFrameSize);
logger.LogInfo("[PV] WebRTCAudioProcessor Init reverse stream: frame size {0}, reverseSamplingRate {1}, reverseChannels {2}", size, reverseSamplingRate, reverseChannels);
if (!reverseStreamThreadRunning)
{
#if NETFX_CORE
Windows.System.Threading.ThreadPool.RunAsync((x) =>
{
ReverseStreamThread();
});
#else
var t = new Thread(ReverseStreamThread);
t.Start();
t.Name = "WebRTCAudioProcessor reverse stream";
#endif
}
if (reverseSamplingRate != samplingRate)
{
logger.LogWarning("[PV] WebRTCAudioProcessor AEC: output sampling rate {0} != {1} capture sampling rate. For better AEC, set audio source (microphone) and audio output samping rates to the same value.", reverseSamplingRate, samplingRate);
}
aecInited = true;
}
}
}
internal LocalVoiceFramed(VoiceClient voiceClient, IEncoder encoder, byte id, VoiceInfo voiceInfo, int channelId, int frameSize)
: base(voiceClient, encoder, id, voiceInfo, channelId, frameSize)
{
this.framer = new Framer <T>(FrameSize);
this.bufferFactory = new FactoryPrimitiveArrayPool <T>(DATA_POOL_CAPACITY, Name + " Data", FrameSize);
}
