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libtgvoip/controller/audio/AudioPacketSender.cpp

167 lines
6.0 KiB
C++

#include "AudioPacketSender.h"
#include "../PrivateDefines.h"
using namespace tgvoip;
AudioPacketSender::AudioPacketSender(VoIPController *controller, const std::shared_ptr<OpusEncoder> &encoder, const std::shared_ptr<VoIPController::Stream> &stream) : PacketSender(controller), encoder(encoder), stream(stream)
{
SetSource(encoder);
}
AudioPacketSender::~AudioPacketSender()
{
}
void AudioPacketSender::SetSource(const std::shared_ptr<OpusEncoder> &encoder)
{
if (this->encoder == encoder || !encoder)
return;
this->encoder = encoder;
encoder->SetCallback(bind(&AudioPacketSender::SendFrame, this, placeholders::_1, placeholders::_2, placeholders::_3, placeholders::_4));
}
void AudioPacketSender::SendFrame(unsigned char *data, size_t len, unsigned char *secondaryData, size_t secondaryLen)
{
if (IsStopping())
return;
Buffer dataBuf = outgoingAudioBufferPool.Get();
Buffer secondaryDataBuf = secondaryLen && secondaryData ? outgoingAudioBufferPool.Get() : Buffer();
dataBuf.CopyFrom(data, 0, len);
if (secondaryLen && secondaryData)
{
secondaryDataBuf.CopyFrom(secondaryData, 0, secondaryLen);
}
shared_ptr<Buffer> dataBufPtr = make_shared<Buffer>(move(dataBuf));
shared_ptr<Buffer> secondaryDataBufPtr = make_shared<Buffer>(move(secondaryDataBuf));
GetMessageThread().Post([this, dataBufPtr, secondaryDataBufPtr, len, secondaryLen]() {
/*
unsentStreamPacketsHistory.Add(static_cast<unsigned int>(unsentStreamPackets));
if (unsentStreamPacketsHistory.Average() >= maxUnsentStreamPackets && !videoPacketSender)
{
LOGW("Resetting stalled send queue");
sendQueue.clear();
unsentStreamPacketsHistory.Reset();
unsentStreamPackets = 0;
}
//if (waitingForAcks || dontSendPackets > 0 || ((unsigned int)unsentStreamPackets >= maxUnsentStreamPackets))
/*{
LOGV("waiting for queue, dropping outgoing audio packet, %d %d %d [%d]", (unsigned int)unsentStreamPackets, waitingForAcks, dontSendPackets, maxUnsentStreamPackets);
return;
}*/
//LOGV("Audio packet size %u", (unsigned int)len);
if (!ReceivedInitAck())
return;
BufferOutputStream pkt(1500);
bool hasExtraFEC = PeerVersion() >= 7 && secondaryLen && shittyInternetMode;
unsigned char flags = (unsigned char)(len > 255 || hasExtraFEC ? STREAM_DATA_FLAG_LEN16 : 0);
pkt.WriteByte((unsigned char)(1 | flags)); // streamID + flags
if (len > 255 || hasExtraFEC)
{
int16_t lenAndFlags = static_cast<int16_t>(len);
if (hasExtraFEC)
lenAndFlags |= STREAM_DATA_XFLAG_EXTRA_FEC;
pkt.WriteInt16(lenAndFlags);
}
else
{
pkt.WriteByte((unsigned char)len);
}
pkt.WriteInt32(audioTimestampOut);
pkt.WriteBytes(*dataBufPtr, 0, len);
if (hasExtraFEC)
{
Buffer ecBuf(secondaryLen);
ecBuf.CopyFrom(*secondaryDataBufPtr, 0, secondaryLen);
if (ecAudioPackets.size() == 4)
{
ecAudioPackets.pop_front();
}
ecAudioPackets.push_back(move(ecBuf));
uint8_t fecCount = std::min(static_cast<uint8_t>(ecAudioPackets.size()), extraEcLevel);
pkt.WriteByte(fecCount);
for (auto ecData = ecAudioPackets.end() - fecCount; ecData != ecAudioPackets.end(); ++ecData)
{
pkt.WriteByte((unsigned char)ecData->Length());
pkt.WriteBytes(*ecData);
}
}
//unsentStreamPackets++;
//PendingOutgoingPacket p{
// /*.seq=*/nextLocalSeq(),
// /*.type=*/PKT_STREAM_DATA,
// /*.len=*/pkt.GetLength(),
// /*.data=*/Buffer(move(pkt)),
// /*.endpoint=*/0,
//};
//conctl.PacketSent(p.seq, p.len);
//shared_ptr<VoIPController::Stream> outgoingAudioStream = GetStreamByType(STREAM_TYPE_AUDIO, false);
if (PeerVersion() < PROTOCOL_RELIABLE)
{
double rtt = LastRtt();
rtt = !rtt || rtt > 0.3 ? 0.5 : rtt; // Tweak this (a lot) later
double timeout = 0; //(outgoingAudioStream && outgoingAudioStream->jitterBuffer ? outgoingAudioStream->jitterBuffer->GetTimeoutWindow() : 0) - rtt;
LOGE("TIMEOUT %lf", timeout + rtt);
timeout = timeout <= 0 ? rtt : timeout;
SendPacketReliably(PKT_STREAM_DATA, pkt.GetBuffer(), pkt.GetLength(), rtt, timeout, 10); // Todo Optimize RTT
}
else
{
}
//SendOrEnqueuePacket(move(p));
if (PeerVersion() < 7 && secondaryLen && shittyInternetMode)
{
Buffer ecBuf(secondaryLen);
ecBuf.CopyFrom(*secondaryDataBufPtr, 0, secondaryLen);
if (ecAudioPackets.size() == 4)
{
ecAudioPackets.pop_front();
}
ecAudioPackets.push_back(move(ecBuf));
pkt = BufferOutputStream(1500);
pkt.WriteByte(stream->id);
pkt.WriteInt32(audioTimestampOut);
uint8_t fecCount = std::min(static_cast<uint8_t>(ecAudioPackets.size()), extraEcLevel);
pkt.WriteByte(fecCount);
for (auto ecData = ecAudioPackets.end() - fecCount; ecData != ecAudioPackets.end(); ++ecData)
{
pkt.WriteByte((unsigned char)ecData->Length());
pkt.WriteBytes(*ecData);
}
PendingOutgoingPacket p{
0,
PKT_STREAM_EC,
pkt.GetLength(),
Buffer(move(pkt)),
0};
SendPacket(std::move(p));
}
audioTimestampOut += stream->frameDuration;
});
#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
if (audioPreprocDataCallback)
{
int size = opus_decode(preprocDecoder.get(), data, len, preprocBuffer, 4096, 0);
audioPreprocDataCallback(preprocBuffer, size);
}
#endif
}