From e3ad8eeb37a01a952b415bcd05b6682afa5e2a8a Mon Sep 17 00:00:00 2001
From: Daniil Gentili <daniil@daniil.it>
Date: Wed, 29 Jan 2020 20:11:39 +0100
Subject: [PATCH] Save

---
 VoIPController.h                 |  1 +
 controller/audio/OpusDecoder.cpp |  6 ++++--
 controller/net/JitterBuffer.cpp  |  6 +++---
 controller/protocol/Reliable.cpp | 12 ++++++------
 controller/protocol/Tick.cpp     |  6 ++++--
 5 files changed, 18 insertions(+), 13 deletions(-)

diff --git a/VoIPController.h b/VoIPController.h
index 9ff6509..ee6de0c 100755
--- a/VoIPController.h
+++ b/VoIPController.h
@@ -585,6 +585,7 @@ private:
     // Acks now handled in Ack
 
     HistoricBuffer<uint32_t, 10, double> sendLossCountHistory;
+    HistoricBuffer<uint32_t, 10, double> packetCountHistory;
     uint32_t audioTimestampIn = 0;
 
     std::shared_ptr<OpusEncoder> encoder;
diff --git a/controller/audio/OpusDecoder.cpp b/controller/audio/OpusDecoder.cpp
index cfad975..81cb182 100755
--- a/controller/audio/OpusDecoder.cpp
+++ b/controller/audio/OpusDecoder.cpp
@@ -232,8 +232,8 @@ int tgvoip::OpusDecoder::DecodeNextFrame()
 	{
 		fec = true;
 		len = jitterBuffer->HandleOutput(buffer, 8192, 0, false, playbackDuration, isEC);
-		//if(len)
-		//	LOGV("Trying FEC...");
+		/*if (len)
+			LOGV("Trying FEC...");*/
 	}
 	int size;
 	if (len)
@@ -280,11 +280,13 @@ int tgvoip::OpusDecoder::DecodeNextFrame()
 	remainingDataLen = size;
 	if (playbackDuration == 80)
 	{
+		//LOGW("Rescaling to 80");
 		processedBuffer = buffer;
 		audio::Resampler::Rescale60To80((int16_t *)decodeBuffer, (int16_t *)processedBuffer);
 	}
 	else if (playbackDuration == 40)
 	{
+		//LOGW("Rescaling to 40");
 		processedBuffer = buffer;
 		audio::Resampler::Rescale60To40((int16_t *)decodeBuffer, (int16_t *)processedBuffer);
 	}
diff --git a/controller/net/JitterBuffer.cpp b/controller/net/JitterBuffer.cpp
index ddcbec2..18be2b9 100755
--- a/controller/net/JitterBuffer.cpp
+++ b/controller/net/JitterBuffer.cpp
@@ -142,7 +142,7 @@ void JitterBuffer::PutInternal(jitter_packet_t *pkt, bool overwriteExisting)
 	if (expectNextAtTime != 0)
 	{
 		double dev = expectNextAtTime - time;
-		LOGV("packet dev %f", dev);
+		//LOGV("packet dev %f", dev);
 		deviationHistory.Add(dev);
 		expectNextAtTime += step / 1000.0;
 	}
@@ -154,13 +154,13 @@ void JitterBuffer::PutInternal(jitter_packet_t *pkt, bool overwriteExisting)
 	// Late packet check
 	if (pkt->timestamp < nextFetchTimestamp)
 	{
-		LOGW("jitter: would drop packet with timestamp %d because it is late but not hopelessly", pkt->timestamp);
+		//LOGW("jitter: would drop packet with timestamp %d because it is late but not hopelessly", pkt->timestamp);
 		latePacketCount++;
 		lostPackets--;
 	}
 	else if (pkt->timestamp < nextFetchTimestamp - 1)
 	{
-		LOGW("jitter: dropping packet with timestamp %d because it is too late", pkt->timestamp);
+		//LOGW("jitter: dropping packet with timestamp %d because it is too late", pkt->timestamp);
 		latePacketCount++;
 		return;
 	}
diff --git a/controller/protocol/Reliable.cpp b/controller/protocol/Reliable.cpp
index 5a4b02a..10e5422 100644
--- a/controller/protocol/Reliable.cpp
+++ b/controller/protocol/Reliable.cpp
@@ -37,17 +37,17 @@ void VoIPController::UpdateReliablePackets()
     {
         if (qp->timeout > 0 && qp->firstSentTime > 0 && GetCurrentTime() - qp->firstSentTime >= qp->timeout)
         {
-//#ifdef LOG_PACKETS
+#ifdef LOG_PACKETS
             LOGD("Removing queued packet because of timeout");
-//#endif
+#endif
             qp = reliablePackets.erase(qp);
             continue;
         }
         if (!qp->tries--)
         {
-//#ifdef LOG_PACKETS
+#ifdef LOG_PACKETS
             LOGD("Removing queued packet because of no more tries");
-//#endif
+#endif
             qp = reliablePackets.erase(qp);
             continue;
         }
@@ -57,9 +57,9 @@ void VoIPController::UpdateReliablePackets()
             uint32_t seq = packetManager.nextLocalSeq();
             qp->seqs.Add(seq);
             qp->lastSentTime = GetCurrentTime();
-//#ifdef LOG_PACKETS
+#ifdef LOG_PACKETS
             LOGD("Sending reliable queued packet, seq=%u, type=%u, len=%lu", seq, qp->type, qp->data.Length());
-//#endif
+#endif
             Buffer buf(qp->data.Length());
             if (qp->firstSentTime == 0)
                 qp->firstSentTime = qp->lastSentTime;
diff --git a/controller/protocol/Tick.cpp b/controller/protocol/Tick.cpp
index 1e8da6e..8e749f9 100644
--- a/controller/protocol/Tick.cpp
+++ b/controller/protocol/Tick.cpp
@@ -187,11 +187,13 @@ void VoIPController::UpdateCongestion()
         uint32_t sendLossCount = conctl.GetSendLossCount();
         sendLossCountHistory.Add(sendLossCount - prevSendLossCount);
         prevSendLossCount = sendLossCount;
+        
         uint32_t lastSentSeq = getBestPacketManager().getLastSentSeq();
-        double packetsPerSec = 1000 / (lastSentSeq - prevSeq);
+        packetCountHistory.Add(lastSentSeq - prevSeq);
         prevSeq = lastSentSeq;
+
         //double packetsPerSec = 1000 / (double)outgoingStreams[0]->frameDuration;
-        double avgSendLossCount = sendLossCountHistory.Average() / packetsPerSec;
+        double avgSendLossCount = sendLossCountHistory.Average() / packetCountHistory.Average();
         LOGE("avg send loss: %.3f%%", avgSendLossCount*100);
 
         AudioPacketSender *sender = dynamic_cast<AudioPacketSender *>(GetStreamByType(STREAM_TYPE_AUDIO, true)->packetSender.get());