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添加流量统计事件

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This commit is contained in:
xia-chu
2021-04-07 18:35:38 +08:00
parent 0c5fa36e4d
commit a22a6bafb7
3 changed files with 38 additions and 4 deletions
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32
webrtc/WebRtcTransport.cpp
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@@ -285,10 +285,37 @@ WebRtcTransportImp::~WebRtcTransportImp() {
void WebRtcTransportImp::onDestory() {
WebRtcTransport::onDestory();
uint64_t duration = _alive_ticker.createdTime() / 1000;
//流量统计事件广播
GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
if (_play_src) {
WarnP(_socket) << "RTC播放器("
<< _media_info._vhost << "/"
<< _media_info._app << "/"
<< _media_info._streamid
<< ")结束播放,耗时(s):" << duration;
if (_bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast<SockInfo &>(*_socket));
}
}
if (_push_src) {
WarnP(_socket) << "RTC推流器("
<< _media_info._vhost << "/"
<< _media_info._app << "/"
<< _media_info._streamid
<< ")结束推流,耗时(s):" << duration;
if (_bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast<SockInfo &>(*_socket));
}
}
}
void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, bool is_play) {
void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, const MediaInfo &info, bool is_play) {
assert(src);
_media_info = info;
if (is_play) {
_play_src = src;
} else {
@@ -455,6 +482,7 @@ private:
};
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
_bytes_usage += len;
auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len);
for (auto rtcp : rtcps) {
switch ((RtcpType) rtcp->pt) {
@@ -504,6 +532,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
}
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
_bytes_usage += len;
_alive_ticker.resetTime();
RtpHeader *rtp = (RtpHeader *) buf;
//根据接收到的rtp的pt信息，找到该流的信息
@@ -549,6 +578,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
//忽略，对方不支持该编码类型
return;
}
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
//统计rtp发送情况，好做sr汇报
_rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);