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- minor fixes of warnings + counter of dropped packets

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This commit is contained in:
Dmytro Bogovych
2021-07-02 14:08:31 +03:00
parent e95aa2274f
commit cf1f206056
8 changed files with 49 additions and 86 deletions
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29
src/engine/media/MT_AudioReceiver.cpp
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@@ -26,7 +26,7 @@
using namespace MT;
// ----------------- RtpBuffer::Packet --------------
RtpBuffer::Packet::Packet(std::shared_ptr<RTPPacket> packet, int timelength, int rate)
RtpBuffer::Packet::Packet(const std::shared_ptr<RTPPacket>& packet, int timelength, int rate)
:mRtp(packet), mTimelength(timelength), mRate(rate)
{
}
@@ -120,9 +120,8 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
// New sequence number
unsigned newSeqno = packet->GetExtendedSequenceNumber();
for (PacketList::iterator iter = mPacketList.begin(); iter != mPacketList.end(); iter++)
for (Packet& p: mPacketList)
{
Packet& p = *iter;
unsigned seqno = p.rtp()->GetExtendedSequenceNumber();
if (seqno == newSeqno)
@@ -138,6 +137,7 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
minno = seqno;
}
// Get amount of available audio (in milliseconds) in jitter buffer
int available = findTimelength();
if (newSeqno > minno || (available < mHigh))
@@ -188,6 +188,9 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
// Erase from packet list
mPacketList.erase(mPacketList.begin());
// Increase number in statistics
mStat.mPacketDropped++;
}
if (total < mLow)
@@ -351,7 +354,7 @@ AudioReceiver::~AudioReceiver()
}
bool AudioReceiver::add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec)
bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec)
{
// Increase codec counter
mStat.mCodecCount[p->GetPayloadType()]++;
@@ -373,20 +376,20 @@ bool AudioReceiver::add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec)
codecIter->second = mCodecList.codecAt(codecIndex).create();
}
// Return pointer to codec if needed
// Return pointer to codec if needed.get()
if (codec)
*codec = codecIter->second.get();
if (mStat.mCodecName.empty())
mStat.mCodecName = codecIter->second.get()->name();
mStat.mCodecName = codecIter->second->name();
// Estimate time length
int timelen = 0, payloadLength = p->GetPayloadLength(), ptype = p->GetPayloadType();
int time_length = 0, payloadLength = p->GetPayloadLength(), ptype = p->GetPayloadType();
if (!codecIter->second->rtpLength())
timelen = codecIter->second->frameTime();
time_length = codecIter->second->frameTime();
else
timelen = int(double(payloadLength) / codecIter->second->rtpLength() * codecIter->second->frameTime() + 0.5);
time_length = lround(double(payloadLength) / codecIter->second->rtpLength() * codecIter->second->frameTime());
// Process jitter
mJitterStats.process(p.get(), codecIter->second->samplerate());
@@ -394,19 +397,19 @@ bool AudioReceiver::add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec)
// Check if packet is CNG
if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
timelen = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
else
// Check if packet is too short from time length side
if (timelen < 2)
if (time_length < 2)
{
// It will cause statistics to report about bad RTP packet
// I have to replay last packet payload here to avoid report about lost packet
mBuffer.add(p, timelen, codecIter->second->samplerate());
mBuffer.add(p, time_length, codecIter->second->samplerate());
return false;
}
// Queue packet to buffer
return mBuffer.add(p, timelen, codecIter->second->samplerate());
return mBuffer.add(p, time_length, codecIter->second->samplerate());
}
void AudioReceiver::processDecoded(Audio::DataWindow& output, int options)