mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +01:00
372 lines
12 KiB
C++
372 lines
12 KiB
C++
#include "../PrivateDefines.cpp"
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using namespace tgvoip;
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using namespace std;
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#pragma mark - Timer methods
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void VoIPController::SendUdpPings()
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{
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LOGW("Send udp pings");
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ENFORCE_MSG_THREAD;
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for (pair<const int64_t, Endpoint> &e : endpoints)
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{
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if (e.second.type == Endpoint::Type::UDP_RELAY)
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{
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SendUdpPing(e.second);
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}
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}
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if (udpConnectivityState == UDP_UNKNOWN || udpConnectivityState == UDP_PING_PENDING)
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udpConnectivityState = UDP_PING_SENT;
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udpPingCount++;
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if (udpPingCount == 4 || udpPingCount == 10)
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{
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messageThread.CancelSelf();
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udpPingTimeoutID = messageThread.Post(std::bind(&VoIPController::EvaluateUdpPingResults, this), 1.0);
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}
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}
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void VoIPController::EvaluateUdpPingResults()
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{
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double avgPongs = 0;
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int count = 0;
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for (pair<const int64_t, Endpoint> &_e : endpoints)
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{
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Endpoint &e = _e.second;
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if (e.type == Endpoint::Type::UDP_RELAY)
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{
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if (e.udpPongCount > 0)
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{
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avgPongs += (double)e.udpPongCount;
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count++;
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}
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}
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}
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if (count > 0)
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avgPongs /= (double)count;
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else
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avgPongs = 0.0;
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LOGI("UDP ping reply count: %.2f", avgPongs);
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if (avgPongs == 0.0 && proxyProtocol == PROXY_SOCKS5 && udpSocket != realUdpSocket)
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{
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LOGI("Proxy does not let UDP through, closing proxy connection and using UDP directly");
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std::shared_ptr<NetworkSocket> proxySocket = udpSocket;
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proxySocket->Close();
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udpSocket = realUdpSocket;
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selectCanceller->CancelSelect();
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proxySupportsUDP = false;
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ResetUdpAvailability();
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return;
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}
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bool configUseTCP = ServerConfig::GetSharedInstance()->GetBoolean("use_tcp", true);
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if (configUseTCP)
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{
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if (avgPongs == 0.0 || (udpConnectivityState == UDP_BAD && avgPongs < 7.0))
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{
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if (needRateFlags & NEED_RATE_FLAG_UDP_NA)
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needRate = true;
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udpConnectivityState = UDP_NOT_AVAILABLE;
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useTCP = true;
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useUDP = avgPongs > 1.0;
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if (endpoints.at(currentEndpoint).type != Endpoint::Type::TCP_RELAY)
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setCurrentEndpointToTCP = true;
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AddTCPRelays();
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waitingForRelayPeerInfo = false;
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}
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else if (avgPongs < 3.0)
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{
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if (needRateFlags & NEED_RATE_FLAG_UDP_BAD)
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needRate = true;
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udpConnectivityState = UDP_BAD;
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useTCP = true;
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setCurrentEndpointToTCP = true;
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AddTCPRelays();
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udpPingTimeoutID = messageThread.Post(std::bind(&VoIPController::SendUdpPings, this), 0.5, 0.5);
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}
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else
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{
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udpPingTimeoutID = MessageThread::INVALID_ID;
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udpConnectivityState = UDP_AVAILABLE;
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}
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}
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else
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{
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udpPingTimeoutID = MessageThread::INVALID_ID;
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udpConnectivityState = UDP_NOT_AVAILABLE;
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}
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}
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void VoIPController::TickJitterBufferAndCongestionControl()
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{
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// TODO get rid of this and update states of these things internally and retroactively
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for (shared_ptr<Stream> &stm : incomingStreams)
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{
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if (stm->jitterBuffer)
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{
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stm->jitterBuffer->Tick();
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}
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}
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conctl.Tick();
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//MutexGuard m(queuedPacketsMutex);
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double currentTime = GetCurrentTime();
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double rtt = GetAverageRTT();
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double packetLossTimeout = std::max(rtt * 2.0, 0.1);
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for (RecentOutgoingPacket &pkt : recentOutgoingPackets)
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{
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if (pkt.ackTime != 0.0 || pkt.lost)
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continue;
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if (currentTime - pkt.sendTime > packetLossTimeout)
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{
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pkt.lost = true;
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sendLosses++;
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LOGW("Outgoing packet lost: seq=%u, type=%s, size=%u", pkt.seq, GetPacketTypeString(pkt.type).c_str(), (unsigned int)pkt.size);
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if (pkt.sender)
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{
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pkt.sender->PacketLost(pkt.seq, pkt.type, pkt.size);
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}
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else if (pkt.type == PKT_STREAM_DATA)
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{
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conctl.PacketLost(pkt.seq);
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}
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}
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}
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}
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void VoIPController::UpdateRTT()
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{
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rttHistory.Add(GetAverageRTT());
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//double v=rttHistory.Average();
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if (rttHistory[0] > 10.0 && rttHistory[8] > 10.0 && (networkType == NET_TYPE_EDGE || networkType == NET_TYPE_GPRS))
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{
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waitingForAcks = true;
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}
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else
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{
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waitingForAcks = false;
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}
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//LOGI("%.3lf/%.3lf, rtt diff %.3lf, waiting=%d, queue=%d", rttHistory[0], rttHistory[8], v, waitingForAcks, sendQueue->Size());
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for (vector<shared_ptr<Stream>>::iterator stm = incomingStreams.begin(); stm != incomingStreams.end(); ++stm)
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{
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if ((*stm)->jitterBuffer)
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{
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int lostCount = (*stm)->jitterBuffer->GetAndResetLostPacketCount();
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if (lostCount > 0 || (lostCount < 0 && recvLossCount > ((uint32_t)-lostCount)))
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recvLossCount += lostCount;
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}
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}
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}
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void VoIPController::UpdateCongestion()
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{
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if (encoder)
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{
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uint32_t sendLossCount = conctl.GetSendLossCount();
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sendLossCountHistory.Add(sendLossCount - prevSendLossCount);
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prevSendLossCount = sendLossCount;
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double packetsPerSec = 1000 / (double)outgoingStreams[0]->frameDuration;
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double avgSendLossCount = sendLossCountHistory.Average() / packetsPerSec;
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//LOGV("avg send loss: %.3f%%", avgSendLossCount*100);
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if (avgSendLossCount > packetLossToEnableExtraEC && networkType != NET_TYPE_GPRS && networkType != NET_TYPE_EDGE)
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{
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if (!shittyInternetMode)
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{
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// Shitty Internet Mode™. Redundant redundancy you can trust.
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shittyInternetMode = true;
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for (shared_ptr<Stream> &s : outgoingStreams)
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{
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if (s->type == STREAM_TYPE_AUDIO)
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{
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s->extraECEnabled = true;
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SendStreamFlags(*s);
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break;
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}
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}
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if (encoder)
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encoder->SetSecondaryEncoderEnabled(true);
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LOGW("Enabling extra EC");
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if (needRateFlags & NEED_RATE_FLAG_SHITTY_INTERNET_MODE)
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needRate = true;
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wasExtraEC = true;
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}
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}
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if (avgSendLossCount > 0.08)
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{
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extraEcLevel = 4;
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}
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else if (avgSendLossCount > 0.05)
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{
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extraEcLevel = 3;
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}
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else if (avgSendLossCount > 0.02)
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{
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extraEcLevel = 2;
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}
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else
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{
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extraEcLevel = 0;
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}
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encoder->SetPacketLoss((int)(avgSendLossCount * 100.0));
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if (avgSendLossCount > rateMaxAcceptableSendLoss)
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needRate = true;
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if ((avgSendLossCount < packetLossToEnableExtraEC || networkType == NET_TYPE_EDGE || networkType == NET_TYPE_GPRS) && shittyInternetMode)
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{
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shittyInternetMode = false;
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for (shared_ptr<Stream> &s : outgoingStreams)
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{
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if (s->type == STREAM_TYPE_AUDIO)
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{
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s->extraECEnabled = false;
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SendStreamFlags(*s);
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break;
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}
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}
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if (encoder)
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encoder->SetSecondaryEncoderEnabled(false);
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LOGW("Disabling extra EC");
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}
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if (!wasEncoderLaggy && encoder->GetComplexity() < 10)
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wasEncoderLaggy = true;
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}
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}
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void VoIPController::UpdateAudioBitrate()
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{
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if (encoder)
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{
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double time = GetCurrentTime();
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if ((audioInput && !audioInput->IsInitialized()) || (audioOutput && !audioOutput->IsInitialized()))
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{
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LOGE("Audio I/O failed");
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lastError = ERROR_AUDIO_IO;
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SetState(STATE_FAILED);
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}
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int act = conctl.GetBandwidthControlAction();
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if (shittyInternetMode)
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{
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encoder->SetBitrate(8000);
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}
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else if (act == TGVOIP_CONCTL_ACT_DECREASE)
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{
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uint32_t bitrate = encoder->GetBitrate();
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if (bitrate > 8000)
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encoder->SetBitrate(bitrate < (minAudioBitrate + audioBitrateStepDecr) ? minAudioBitrate : (bitrate - audioBitrateStepDecr));
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}
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else if (act == TGVOIP_CONCTL_ACT_INCREASE)
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{
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uint32_t bitrate = encoder->GetBitrate();
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if (bitrate < maxBitrate)
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encoder->SetBitrate(bitrate + audioBitrateStepIncr);
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}
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if (state == STATE_ESTABLISHED && time - lastRecvPacketTime >= reconnectingTimeout)
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{
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SetState(STATE_RECONNECTING);
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if (needRateFlags & NEED_RATE_FLAG_RECONNECTING)
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needRate = true;
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wasReconnecting = true;
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ResetUdpAvailability();
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}
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if (state == STATE_ESTABLISHED || state == STATE_RECONNECTING)
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{
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if (time - lastRecvPacketTime >= config.recvTimeout)
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{
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const Endpoint &_currentEndpoint = endpoints.at(currentEndpoint);
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if (_currentEndpoint.type != Endpoint::Type::UDP_RELAY && _currentEndpoint.type != Endpoint::Type::TCP_RELAY)
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{
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LOGW("Packet receive timeout, switching to relay");
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currentEndpoint = preferredRelay;
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for (pair<const int64_t, Endpoint> &_e : endpoints)
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{
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Endpoint &e = _e.second;
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if (e.IsP2P())
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{
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e.averageRTT = 0;
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e.rtts.Reset();
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}
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}
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if (allowP2p)
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{
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SendPublicEndpointsRequest();
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}
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UpdateDataSavingState();
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UpdateAudioBitrateLimit();
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BufferOutputStream s(4);
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s.WriteInt32(dataSavingMode ? INIT_FLAG_DATA_SAVING_ENABLED : 0);
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if (peerVersion < 6)
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{
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SendPacketReliably(PKT_NETWORK_CHANGED, s.GetBuffer(), s.GetLength(), 1, 20);
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}
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else
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{
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Buffer buf(move(s));
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SendExtra(buf, EXTRA_TYPE_NETWORK_CHANGED);
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}
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lastRecvPacketTime = time;
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}
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else
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{
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LOGW("Packet receive timeout, disconnecting");
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lastError = ERROR_TIMEOUT;
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SetState(STATE_FAILED);
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}
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}
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}
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}
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}
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void VoIPController::UpdateSignalBars()
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{
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int prevSignalBarCount = GetSignalBarsCount();
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double packetsPerSec = 1000 / (double)outgoingStreams[0]->frameDuration;
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double avgSendLossCount = sendLossCountHistory.Average() / packetsPerSec;
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int signalBarCount = 4;
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if (state == STATE_RECONNECTING || waitingForAcks)
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signalBarCount = 1;
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if (endpoints.at(currentEndpoint).type == Endpoint::Type::TCP_RELAY)
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{
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signalBarCount = std::min(signalBarCount, 3);
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}
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if (avgSendLossCount > 0.1)
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{
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signalBarCount = 1;
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}
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else if (avgSendLossCount > 0.0625)
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{
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signalBarCount = std::min(signalBarCount, 2);
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}
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else if (avgSendLossCount > 0.025)
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{
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signalBarCount = std::min(signalBarCount, 3);
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}
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for (shared_ptr<Stream> &stm : incomingStreams)
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{
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if (stm->jitterBuffer)
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{
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double avgLateCount[3];
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stm->jitterBuffer->GetAverageLateCount(avgLateCount);
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if (avgLateCount[2] >= 0.2)
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signalBarCount = 1;
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else if (avgLateCount[2] >= 0.1)
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signalBarCount = std::min(signalBarCount, 2);
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}
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}
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signalBarsHistory.Add(static_cast<unsigned char>(signalBarCount));
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//LOGV("Signal bar count history %08X", *reinterpret_cast<uint32_t *>(&signalBarsHistory));
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int _signalBarCount = GetSignalBarsCount();
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if (_signalBarCount != prevSignalBarCount)
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{
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LOGD("SIGNAL BAR COUNT CHANGED: %d", _signalBarCount);
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if (callbacks.signalBarCountChanged)
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callbacks.signalBarCountChanged(this, _signalBarCount);
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}
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} |