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WebRTC: Fixes to packet timing and loss statistics

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This commit is contained in:
Thulinma 2020-09-24 12:46:32 +02:00
parent 3ba7723b10
commit 2a5a808107
4 changed files with 78 additions and 32 deletions
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3
lib/rtp.h
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@ -149,7 +149,6 @@ namespace RTP{
public:
uint64_t trackId;
uint64_t bootMsOffset;
double multiplier; ///< Multiplier to convert from millis to RTP time
std::string codec; ///< Codec of this track
std::string type; ///< Type of this track
@ -161,7 +160,7 @@ namespace RTP{
bool recentWrap; ///< True if a wraparound happened recently.
uint32_t prevTime;
uint64_t firstTime;
int32_t milliSync;
int64_t milliSync;
void (*cbPack)(const DTSC::Packet &pkt);
void (*cbInit)(const uint64_t track, const std::string &initData);
// Codec-specific handlers