mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-11 16:49:52 +01:00
244 lines
8.4 KiB
C++
244 lines
8.4 KiB
C++
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#include "../../VoIPController.h"
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using namespace tgvoip;
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using namespace std;
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#pragma mark - Audio I/O
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void VoIPController::HandleAudioInput(unsigned char *data, size_t len, unsigned char *secondaryData, size_t secondaryLen)
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{
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if (stopping)
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return;
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// TODO make an AudioPacketSender
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Buffer dataBuf = outgoingAudioBufferPool.Get();
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Buffer secondaryDataBuf = secondaryLen && secondaryData ? outgoingAudioBufferPool.Get() : Buffer();
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dataBuf.CopyFrom(data, 0, len);
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if (secondaryLen && secondaryData)
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{
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secondaryDataBuf.CopyFrom(secondaryData, 0, secondaryLen);
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}
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shared_ptr<Buffer> dataBufPtr = make_shared<Buffer>(move(dataBuf));
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shared_ptr<Buffer> secondaryDataBufPtr = make_shared<Buffer>(move(secondaryDataBuf));
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messageThread.Post([this, dataBufPtr, secondaryDataBufPtr, len, secondaryLen]() {
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unsentStreamPacketsHistory.Add(static_cast<unsigned int>(unsentStreamPackets));
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if (unsentStreamPacketsHistory.Average() >= maxUnsentStreamPackets && !videoPacketSender)
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{
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LOGW("Resetting stalled send queue");
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sendQueue.clear();
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unsentStreamPacketsHistory.Reset();
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unsentStreamPackets = 0;
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}
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if (waitingForAcks || dontSendPackets > 0 || ((unsigned int)unsentStreamPackets >= maxUnsentStreamPackets /*&& endpoints[currentEndpoint].type==Endpoint::Type::TCP_RELAY*/))
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{
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LOGV("waiting for queue, dropping outgoing audio packet, %d %d %d [%d]", (unsigned int)unsentStreamPackets, waitingForAcks, dontSendPackets, maxUnsentStreamPackets);
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return;
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}
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//LOGV("Audio packet size %u", (unsigned int)len);
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if (!receivedInitAck)
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return;
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BufferOutputStream pkt(1500);
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bool hasExtraFEC = peerVersion >= 7 && secondaryLen && shittyInternetMode;
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unsigned char flags = (unsigned char)(len > 255 || hasExtraFEC ? STREAM_DATA_FLAG_LEN16 : 0);
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pkt.WriteByte((unsigned char)(1 | flags)); // streamID + flags
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if (len > 255 || hasExtraFEC)
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{
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int16_t lenAndFlags = static_cast<int16_t>(len);
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if (hasExtraFEC)
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lenAndFlags |= STREAM_DATA_XFLAG_EXTRA_FEC;
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pkt.WriteInt16(lenAndFlags);
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}
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else
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{
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pkt.WriteByte((unsigned char)len);
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}
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pkt.WriteInt32(audioTimestampOut);
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pkt.WriteBytes(*dataBufPtr, 0, len);
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if (hasExtraFEC)
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{
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Buffer ecBuf(secondaryLen);
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ecBuf.CopyFrom(*secondaryDataBufPtr, 0, secondaryLen);
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if (ecAudioPackets.size() == 4)
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{
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ecAudioPackets.pop_front();
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}
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ecAudioPackets.push_back(move(ecBuf));
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uint8_t fecCount = std::min(static_cast<uint8_t>(ecAudioPackets.size()), extraEcLevel);
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pkt.WriteByte(fecCount);
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for (auto ecData = ecAudioPackets.end() - fecCount; ecData != ecAudioPackets.end(); ++ecData)
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{
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pkt.WriteByte((unsigned char)ecData->Length());
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pkt.WriteBytes(*ecData);
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}
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}
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unsentStreamPackets++;
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PendingOutgoingPacket p{
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/*.seq=*/GenerateOutSeq(),
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/*.type=*/PKT_STREAM_DATA,
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/*.len=*/pkt.GetLength(),
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/*.data=*/Buffer(move(pkt)),
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/*.endpoint=*/0,
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};
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conctl.PacketSent(p.seq, p.len);
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SendOrEnqueuePacket(move(p));
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if (peerVersion < 7 && secondaryLen && shittyInternetMode)
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{
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Buffer ecBuf(secondaryLen);
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ecBuf.CopyFrom(*secondaryDataBufPtr, 0, secondaryLen);
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if (ecAudioPackets.size() == 4)
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{
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ecAudioPackets.pop_front();
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}
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ecAudioPackets.push_back(move(ecBuf));
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pkt = BufferOutputStream(1500);
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pkt.WriteByte(outgoingStreams[0]->id);
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pkt.WriteInt32(audioTimestampOut);
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uint8_t fecCount = std::min(static_cast<uint8_t>(ecAudioPackets.size()), extraEcLevel);
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pkt.WriteByte(fecCount);
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for (auto ecData = ecAudioPackets.end() - fecCount; ecData != ecAudioPackets.end(); ++ecData)
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{
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pkt.WriteByte((unsigned char)ecData->Length());
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pkt.WriteBytes(*ecData);
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}
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PendingOutgoingPacket p{
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GenerateOutSeq(),
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PKT_STREAM_EC,
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pkt.GetLength(),
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Buffer(move(pkt)),
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0};
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SendOrEnqueuePacket(move(p));
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}
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audioTimestampOut += outgoingStreams[0]->frameDuration;
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});
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#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
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if (audioPreprocDataCallback)
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{
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int size = opus_decode(preprocDecoder.get(), data, len, preprocBuffer, 4096, 0);
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audioPreprocDataCallback(preprocBuffer, size);
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}
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#endif
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}
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void VoIPController::InitializeAudio()
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{
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double t = GetCurrentTime();
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shared_ptr<Stream> outgoingAudioStream = GetStreamByType(STREAM_TYPE_AUDIO, true);
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LOGI("before create audio io");
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audioIO = audio::AudioIO::Create(currentAudioInput, currentAudioOutput);
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audioInput = audioIO->GetInput();
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audioOutput = audioIO->GetOutput();
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#ifdef __ANDROID__
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audio::AudioInputAndroid *androidInput = dynamic_cast<audio::AudioInputAndroid *>(audioInput.get());
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if (androidInput)
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{
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unsigned int effects = androidInput->GetEnabledEffects();
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if (!(effects & audio::AudioInputAndroid::EFFECT_AEC))
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{
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config.enableAEC = true;
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LOGI("Forcing software AEC because built-in is not good");
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}
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if (!(effects & audio::AudioInputAndroid::EFFECT_NS))
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{
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config.enableNS = true;
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LOGI("Forcing software NS because built-in is not good");
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}
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}
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#elif defined(__APPLE__) && TARGET_OS_OSX
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SetAudioOutputDuckingEnabled(macAudioDuckingEnabled);
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#endif
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LOGI("AEC: %d NS: %d AGC: %d", config.enableAEC, config.enableNS, config.enableAGC);
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echoCanceller.reset(new EchoCanceller(config.enableAEC, config.enableNS, config.enableAGC));
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encoder.reset(new OpusEncoder(audioInput, true));
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encoder->SetCallback(bind(&VoIPController::HandleAudioInput, this, placeholders::_1, placeholders::_2, placeholders::_3, placeholders::_4));
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encoder->SetOutputFrameDuration(outgoingAudioStream->frameDuration);
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encoder->SetEchoCanceller(echoCanceller);
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encoder->SetSecondaryEncoderEnabled(false);
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if (config.enableVolumeControl)
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{
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encoder->AddAudioEffect(inputVolume);
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}
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#if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
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dynamic_cast<audio::AudioInputCallback *>(audioInput.get())->SetDataCallback(audioInputDataCallback);
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dynamic_cast<audio::AudioOutputCallback *>(audioOutput.get())->SetDataCallback(audioOutputDataCallback);
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#endif
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if (!audioOutput->IsInitialized())
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{
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LOGE("Error initializing audio playback");
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lastError = ERROR_AUDIO_IO;
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SetState(STATE_FAILED);
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return;
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}
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UpdateAudioBitrateLimit();
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LOGI("Audio initialization took %f seconds", GetCurrentTime() - t);
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}
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void VoIPController::StartAudio()
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{
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OnAudioOutputReady();
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encoder->Start();
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if (!micMuted)
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{
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audioInput->Start();
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if (!audioInput->IsInitialized())
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{
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LOGE("Error initializing audio capture");
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lastError = ERROR_AUDIO_IO;
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SetState(STATE_FAILED);
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return;
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}
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}
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}
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void VoIPController::OnAudioOutputReady()
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{
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LOGI("Audio I/O ready");
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auto &stm = incomingStreams[0];
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stm->decoder = make_shared<OpusDecoder>(audioOutput, true, peerVersion >= 6);
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stm->decoder->SetEchoCanceller(echoCanceller);
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if (config.enableVolumeControl)
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{
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stm->decoder->AddAudioEffect(outputVolume);
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}
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stm->decoder->SetJitterBuffer(stm->jitterBuffer);
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stm->decoder->SetFrameDuration(stm->frameDuration);
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stm->decoder->Start();
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}
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void VoIPController::UpdateAudioOutputState()
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{
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bool areAnyAudioStreamsEnabled = false;
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for (auto s = incomingStreams.begin(); s != incomingStreams.end(); ++s)
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{
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if ((*s)->type == STREAM_TYPE_AUDIO && (*s)->enabled)
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areAnyAudioStreamsEnabled = true;
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}
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if (audioOutput)
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{
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LOGV("New audio output state: %d", areAnyAudioStreamsEnabled);
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if (audioOutput->IsPlaying() != areAnyAudioStreamsEnabled)
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{
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if (areAnyAudioStreamsEnabled)
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audioOutput->Start();
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else
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audioOutput->Stop();
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}
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}
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}
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