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@ -585,6 +585,7 @@ private:
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// Acks now handled in Ack
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HistoricBuffer<uint32_t, 10, double> sendLossCountHistory;
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HistoricBuffer<uint32_t, 10, double> packetCountHistory;
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uint32_t audioTimestampIn = 0;
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std::shared_ptr<OpusEncoder> encoder;
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@ -232,8 +232,8 @@ int tgvoip::OpusDecoder::DecodeNextFrame()
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{
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fec = true;
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len = jitterBuffer->HandleOutput(buffer, 8192, 0, false, playbackDuration, isEC);
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//if(len)
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// LOGV("Trying FEC...");
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/*if (len)
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LOGV("Trying FEC...");*/
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}
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int size;
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if (len)
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@ -280,11 +280,13 @@ int tgvoip::OpusDecoder::DecodeNextFrame()
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remainingDataLen = size;
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if (playbackDuration == 80)
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{
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//LOGW("Rescaling to 80");
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processedBuffer = buffer;
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audio::Resampler::Rescale60To80((int16_t *)decodeBuffer, (int16_t *)processedBuffer);
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}
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else if (playbackDuration == 40)
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{
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//LOGW("Rescaling to 40");
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processedBuffer = buffer;
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audio::Resampler::Rescale60To40((int16_t *)decodeBuffer, (int16_t *)processedBuffer);
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}
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@ -142,7 +142,7 @@ void JitterBuffer::PutInternal(jitter_packet_t *pkt, bool overwriteExisting)
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if (expectNextAtTime != 0)
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{
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double dev = expectNextAtTime - time;
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LOGV("packet dev %f", dev);
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//LOGV("packet dev %f", dev);
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deviationHistory.Add(dev);
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expectNextAtTime += step / 1000.0;
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}
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@ -154,13 +154,13 @@ void JitterBuffer::PutInternal(jitter_packet_t *pkt, bool overwriteExisting)
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// Late packet check
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if (pkt->timestamp < nextFetchTimestamp)
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{
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LOGW("jitter: would drop packet with timestamp %d because it is late but not hopelessly", pkt->timestamp);
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//LOGW("jitter: would drop packet with timestamp %d because it is late but not hopelessly", pkt->timestamp);
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latePacketCount++;
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lostPackets--;
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}
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else if (pkt->timestamp < nextFetchTimestamp - 1)
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{
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LOGW("jitter: dropping packet with timestamp %d because it is too late", pkt->timestamp);
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//LOGW("jitter: dropping packet with timestamp %d because it is too late", pkt->timestamp);
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latePacketCount++;
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return;
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}
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@ -37,17 +37,17 @@ void VoIPController::UpdateReliablePackets()
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{
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if (qp->timeout > 0 && qp->firstSentTime > 0 && GetCurrentTime() - qp->firstSentTime >= qp->timeout)
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{
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//#ifdef LOG_PACKETS
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#ifdef LOG_PACKETS
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LOGD("Removing queued packet because of timeout");
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//#endif
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#endif
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qp = reliablePackets.erase(qp);
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continue;
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}
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if (!qp->tries--)
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{
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//#ifdef LOG_PACKETS
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#ifdef LOG_PACKETS
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LOGD("Removing queued packet because of no more tries");
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//#endif
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#endif
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qp = reliablePackets.erase(qp);
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continue;
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}
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@ -57,9 +57,9 @@ void VoIPController::UpdateReliablePackets()
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uint32_t seq = packetManager.nextLocalSeq();
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qp->seqs.Add(seq);
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qp->lastSentTime = GetCurrentTime();
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//#ifdef LOG_PACKETS
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#ifdef LOG_PACKETS
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LOGD("Sending reliable queued packet, seq=%u, type=%u, len=%lu", seq, qp->type, qp->data.Length());
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//#endif
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#endif
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Buffer buf(qp->data.Length());
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if (qp->firstSentTime == 0)
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qp->firstSentTime = qp->lastSentTime;
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@ -187,11 +187,13 @@ void VoIPController::UpdateCongestion()
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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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uint32_t lastSentSeq = getBestPacketManager().getLastSentSeq();
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double packetsPerSec = 1000 / (lastSentSeq - prevSeq);
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packetCountHistory.Add(lastSentSeq - prevSeq);
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prevSeq = lastSentSeq;
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//double packetsPerSec = 1000 / (double)outgoingStreams[0]->frameDuration;
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double avgSendLossCount = sendLossCountHistory.Average() / packetsPerSec;
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double avgSendLossCount = sendLossCountHistory.Average() / packetCountHistory.Average();
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LOGE("avg send loss: %.3f%%", avgSendLossCount*100);
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AudioPacketSender *sender = dynamic_cast<AudioPacketSender *>(GetStreamByType(STREAM_TYPE_AUDIO, true)->packetSender.get());
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