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- minor fixes of warnings + counter of dropped packets

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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3
src/engine/agent/Agent_Impl.cpp
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@ -535,6 +535,9 @@ void AgentImpl::processGetMediaStats(JsonCpp::Value& request, JsonCpp::Value& an
#endif
if (result.exists(SessionInfo_PacketLoss))
answer["rtp_lost"] = result[SessionInfo_LostRtp].asInt();
if (result.exists(SessionInfo_DroppedRtp))
answer["rtp_dropped"] = result[SessionInfo_DroppedRtp].asInt();
if (result.exists(SessionInfo_SentRtp))
answer["rtp_sent"] = result[SessionInfo_SentRtp].asInt();
if (result.exists(SessionInfo_ReceivedRtp))

18
src/engine/endpoint/EP_AudioProvider.h
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@ -43,32 +43,32 @@ public:
void sendData(PDatagramSocket s, InternetAddress& destination, const void* dataBuffer, unsigned int datasize);
// Updates SDP offer
void updateSdpOffer(resip::SdpContents::Session::Medium& sdp, SdpDirection direction);
void updateSdpOffer(resip::SdpContents::Session::Medium& sdp, SdpDirection direction) override;
// Called by user agent when session is deleted.
void sessionDeleted();
void sessionDeleted() override;
// Called by user agent when session is terminated.
void sessionTerminated();
void sessionTerminated() override;
// Called by user agent when session is started.
void sessionEstablished(int conntype);
void sessionEstablished(int conntype) override;
// Called by user agent to save media socket for this provider
void setSocket(const RtpPair<PDatagramSocket>& p4, const RtpPair<PDatagramSocket>& p6);
void setSocket(const RtpPair<PDatagramSocket>& p4, const RtpPair<PDatagramSocket>& p6) override;
// Called by user agent to get media socket for this provider
RtpPair<PDatagramSocket>& socket(int family);
RtpPair<PDatagramSocket>& socket(int family) override;
// Called by user agent to process media stream description from remote peer.
// Returns true if description is processed succesfully. Otherwise method returns false.
// myAnswer sets if the answer will be sent after.
bool processSdpOffer(const resip::SdpContents::Session::Medium& media, SdpDirection sdpDirection);
bool processSdpOffer(const resip::SdpContents::Session::Medium& media, SdpDirection sdpDirection) override;
void setState(unsigned state) override;
unsigned state();
MT::Statistics getStatistics();
unsigned state() override;
MT::Statistics getStatistics() override;
MT::PStream activeStream();
void readFile(const Audio::PWavFileReader& stream, MT::Stream::MediaDirection direction);

1
src/engine/endpoint/EP_Session.cpp
View File

@ -478,6 +478,7 @@ void Session::getSessionInfo(Session::InfoOptions options, VariantMap& info)
info[SessionInfo_ReceivedRtp] = static_cast<int>(stat.mReceivedRtp);
info[SessionInfo_ReceivedRtcp] = static_cast<int>(stat.mReceivedRtcp);
info[SessionInfo_LostRtp] = static_cast<int>(stat.mPacketLoss);
info[SessionInfo_DroppedRtp] = static_cast<int>(stat.mPacketDropped);
info[SessionInfo_SentRtp] = static_cast<int>(stat.mSentRtp);
info[SessionInfo_SentRtcp] = static_cast<int>(stat.mSentRtcp);
if (stat.mFirstRtpTime)

3
src/engine/endpoint/EP_Session.h
View File

@ -66,12 +66,13 @@ enum SessionInfo
SessionInfo_ReceivedRtp,
SessionInfo_ReceivedRtcp,
SessionInfo_LostRtp,
SessionInfo_DroppedRtp,
SessionInfo_Duration,
SessionInfo_Jitter,
SessionInfo_Rtt,
SessionInfo_BitrateSwitchCounter, // It is for AMR codecs only
SessionInfo_RemotePeer,
SessionInfo_SSRC
SessionInfo_SSRC,
};

29
src/engine/media/MT_AudioReceiver.cpp
View File

@ -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)

5
src/engine/media/MT_AudioReceiver.h
View File

@ -45,7 +45,7 @@ namespace MT
class Packet
{
public:
Packet(std::shared_ptr<RTPPacket> packet, int timelen, int rate);
Packet(const std::shared_ptr<RTPPacket>& packet, int timelen, int rate);
std::shared_ptr<RTPPacket> rtp() const;
int timelength() const;
int rate() const;
@ -109,7 +109,7 @@ namespace MT
// Returns false when packet is rejected as illegal. codec parameter will show codec which will be used for decoding.
// Lifetime of pointer to codec is limited by lifetime of AudioReceiver (it is container).
bool add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec = nullptr);
bool add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec = nullptr);
// Returns false when there is no rtp data from jitter
enum DecodeOptions
@ -178,6 +178,7 @@ namespace MT
void initPvqa();
void updatePvqa(const void* data, int size);
float calculatePvqaMos(int rate, std::string& report);
std::shared_ptr<Audio::DataWindow> mPvqaBuffer;
#endif

74
src/engine/media/MT_Statistics.cpp
View File

@ -40,7 +40,7 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
int64_t delta = receiveDelta - timestampDelta;
// Update max delta in milliseconds
double delta_in_seconds = fabs(double(delta) / rate);
float delta_in_seconds = float(fabs(double(delta) / rate));
if (delta_in_seconds > mMaxDelta)
mMaxDelta = delta_in_seconds;
@ -56,7 +56,7 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
mReceiveTimestamp = timestamp;
// And mJitter are in seconds again
mJitter.process(mLastJitter.value() / rate);
mJitter.process(mLastJitter.value() / float(rate));
}
}
@ -67,7 +67,7 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
Statistics::Statistics()
:mReceived(0), mSent(0), mReceivedRtp(0), mSentRtp(0),
mReceivedRtcp(0), mSentRtcp(0), mDuplicatedRtp(0), mOldRtp(0), mIllegalRtp(0),
mPacketLoss(0), mJitter(0.0), mAudioTime(0), mSsrc(0)
mPacketLoss(0), mJitter(0.0), mAudioTime(0), mSsrc(0), mPacketDropped(0)
{
mBitrateSwitchCounter = 0;
memset(mLoss, 0, sizeof mLoss);
@ -94,64 +94,11 @@ void Statistics::reset()
mIllegalRtp = 0;
mJitter = 0.0;
mAudioTime = 0;
mPacketDropped = 0;
memset(mLoss, 0, sizeof mLoss);
}
/*
double calculate_mos_g711(double ppl, double burstr, int version) {
double r;
double bpl = 8.47627; //mos = -4.23836 + 0.29873 * r - 0.00416744 * r * r + 0.0000209855 * r * r * r;
double mos;
if(ppl == 0 or burstr == 0) {
return 4.5;
}
if(ppl > 0.5) {
return 1;
}
switch(version) {
case 1:
case 2:
default:
// this mos is calculated for G.711 and PLC
bpl = 17.2647;
r = 93.2062077233 - 95.0 * (ppl*100/(ppl*100/burstr + bpl));
mos = 2.06405 + 0.031738 * r - 0.000356641 * r * r + 2.93143 * pow(10,-6) * r * r * r;
if(mos < 1)
return 1;
if(mos > 4.5)
return 4.5;
}
return mos;
}
double calculate_mos(double ppl, double burstr, int codec, unsigned int received) {
if(codec == PAYLOAD_G729) {
if(opt_mos_g729) {
if(received < 100) {
return 3.92;
}
return (double)mos_g729((long double)ppl, (long double)burstr);
} else {
if(received < 100) {
return 4.5;
}
return calculate_mos_g711(ppl, burstr, 2);
}
} else {
if(received < 100) {
return 4.5;
}
return calculate_mos_g711(ppl, burstr, 2);
}
}
*/
void Statistics::calculateBurstr(double* burstr, double* lossr) const
{
int lost = 0;
@ -230,8 +177,10 @@ Statistics& Statistics::operator += (const Statistics& src)
mDuplicatedRtp += src.mDuplicatedRtp;
mOldRtp += src.mOldRtp;
mPacketLoss += src.mPacketLoss;
mPacketDropped += src.mPacketDropped;
mAudioTime += src.mAudioTime;
for (auto codecStat: src.mCodecCount)
{
if (mCodecCount.find(codecStat.first) == mCodecCount.end())
@ -277,6 +226,8 @@ Statistics& Statistics::operator -= (const Statistics& src)
mDuplicatedRtp -= src.mDuplicatedRtp;
mOldRtp -= src.mOldRtp;
mPacketLoss -= src.mPacketLoss;
mPacketDropped -= src.mPacketDropped;
mAudioTime -= src.mAudioTime;
for (auto codecStat: src.mCodecCount)
{
@ -291,9 +242,10 @@ Statistics& Statistics::operator -= (const Statistics& src)
std::string Statistics::toShortString() const
{
std::ostringstream oss;
oss << "Received: " << mReceivedRtp
<< ", lost: " << mPacketLoss
<< ", sent: " << mSentRtp;
oss << "Received: " << mReceivedRtp
<< ", lost: " << mPacketLoss
<< ", dropped: " << mPacketDropped
<< ", sent: " << mSentRtp;
return oss.str();
}

2
src/engine/media/MT_Statistics.h
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@ -130,7 +130,9 @@ public:
mDuplicatedRtp, // Number of received duplicated rtp packets
mOldRtp, // Number of late rtp packets
mPacketLoss, // Number of lost packets
mPacketDropped, // Number of dropped packets (due to time unsync when playing)
mIllegalRtp; // Number of rtp packets with bad payload type
int mLoss[128]; // Every item is number of loss of corresping length
size_t mAudioTime; // Decoded/found time in milliseconds
uint16_t mSsrc; // Last known SSRC ID in a RTP stream