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- improve packet loss reporting + initial DTMF RFC 2833 event reporting

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This commit is contained in:
Dmytro Bogovych
2026-02-25 10:16:24 +03:00
parent 03f662e5ce
commit 06b39dd629
9 changed files with 210 additions and 109 deletions
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src/engine/media/MT_Statistics.h
+37 -30
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@@ -4,16 +4,15 @@
#include <chrono>
#include <map>
#include <optional>
#include <array>
#include "audio/Audio_DataWindow.h"
#include "helper/HL_Optional.hpp"
#include "helper/HL_Statistics.h"
#include "helper/HL_Types.h"
#include "helper/HL_InternetAddress.h"
#include "jrtplib/src/rtptimeutilities.h"
#include "jrtplib/src/rtppacket.h"
using namespace std::chrono_literals;
namespace MT
{
@@ -56,47 +55,55 @@ struct PacketLossEvent
uint32_t mStartSeqno = 0,
mEndSeqno = 0;
int mGap = 0;
std::chrono::microseconds mTimestampStart = 0us,
mTimestampEnd = 0us;
};
struct Dtmf2833Event
{
char mTone;
std::chrono::microseconds mTimestamp;
};
class Statistics
{
public:
size_t mReceived = 0, // Received traffic in bytes
mSent = 0, // Sent traffic in bytes
mReceivedRtp = 0, // Number of received rtp packets
mSentRtp = 0, // Number of sent rtp packets
mReceivedRtcp = 0, // Number of received rtcp packets
mSentRtcp = 0, // Number of sent rtcp packets
mDuplicatedRtp = 0, // Number of received duplicated rtp packets
mOldRtp = 0, // Number of late rtp packets
mPacketLoss = 0, // Number of lost packets
mPacketDropped = 0, // Number of dropped packets (due to time unsync when playing)б
mIllegalRtp = 0; // Number of rtp packets with bad payload type
size_t mReceived = 0, // Received traffic in bytes
mSent = 0, // Sent traffic in bytes
mReceivedRtp = 0, // Number of received rtp packets
mSentRtp = 0, // Number of sent rtp packets
mReceivedRtcp = 0, // Number of received rtcp packets
mSentRtcp = 0, // Number of sent rtcp packets
mDuplicatedRtp = 0, // Number of received duplicated rtp packets
mOldRtp = 0, // Number of late rtp packets
mPacketLoss = 0, // Number of lost packets
mPacketDropped = 0, // Number of dropped packets (due to time unsync when playing)б
mIllegalRtp = 0; // Number of rtp packets with bad payload type
TestResult<float> mDecodingInterval, // Average interval on call to packet decode
mDecodeRequested, // Average amount of requested audio frames to play
mPacketInterval; // Average interval between packet adding to jitter buffer
TestResult<float> mDecodingInterval, // Average interval on call to packet decode
mDecodeRequested, // Average amount of requested audio frames to play
mPacketInterval; // Average interval between packet adding to jitter buffer
std::array<float, 128> mLoss = {0}; // Every item is number of loss of corresping length
size_t mAudioTime = 0; // Decoded/found time in milliseconds
size_t mDecodedSize = 0; // Number of decoded bytes
uint16_t mSsrc = 0; // Last known SSRC ID in a RTP stream
ice::NetworkAddress mRemotePeer; // Last known remote RTP address
std::map<int,int> mLoss; // Every item is number of loss of corresping length
size_t mAudioTime = 0; // Decoded/found time in milliseconds
size_t mDecodedSize = 0; // Number of decoded bytes
uint16_t mSsrc = 0; // Last known SSRC ID in a RTP stream
ice::NetworkAddress mRemotePeer; // Last known remote RTP address
// AMR codec bitrate switch counter
int mBitrateSwitchCounter = 0;
int mCng = 0;
std::string mCodecName;
float mJitter = 0.0f; // Jitter
TestResult<float> mRttDelay; // RTT delay
int mBitrateSwitchCounter = 0;
int mCng = 0;
std::string mCodecName;
float mJitter = 0.0f; // Jitter
TestResult<float> mRttDelay; // RTT delay
// Timestamp when first RTP packet has arrived
std::optional<timepoint_t> mFirstRtpTime;
std::optional<timepoint_t> mFirstRtpTime;
std::map<int, int> mCodecCount; // Stats on used codecs
std::map<int, int> mCodecCount; // Stats on used codecs
std::vector<PacketLossEvent> mPacketLossTimeline; // Packet loss timeline
std::vector<PacketLossEvent> mPacketLossTimeline; // Packet loss timeline
std::vector<Dtmf2833Event> mDtmf2833Timeline;
// It is to calculate network MOS
void calculateBurstr(double* burstr, double* loss) const;