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RtcpContext修改时间戳单位、整理WebRTC相关代码

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This commit is contained in:
ziyue
2021-06-25 14:59:27 +08:00
parent 6c01cf336e
commit 964cf39145
8 changed files with 49 additions and 52 deletions
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64
webrtc/WebRtcTransport.cpp
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@@ -408,19 +408,19 @@ void WebRtcTransportImp::onStartWebRTC() {
info->offer_ssrc_rtx = m_offer->getRtxSSRC();
info->plan_rtp = &m_answer.plan[0];;
info->plan_rtx = m_answer.getRelatedRtxPlan(info->plan_rtp->pt);
info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
info->rtcp_context_send = std::make_shared<RtcpContext>(false);
//send ssrc --> MediaTrack
_rtp_info_ssrc[info->answer_ssrc_rtp] = info;
_ssrc_to_track[info->answer_ssrc_rtp] = info;
//recv ssrc --> MediaTrack
_rtp_info_ssrc[info->offer_ssrc_rtp] = info;
_ssrc_to_track[info->offer_ssrc_rtp] = info;
//rtp pt --> MediaTrack
_rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info));
_pt_to_track.emplace(info->plan_rtp->pt, std::make_pair(false, info));
if (info->plan_rtx) {
//rtx pt --> MediaTrack
_rtp_info_pt.emplace(info->plan_rtx->pt, std::make_pair(true, info));
_pt_to_track.emplace(info->plan_rtx->pt, std::make_pair(true, info));
}
if (m_offer->type != TrackApplication) {
//记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id
@@ -464,10 +464,10 @@ void WebRtcTransportImp::onStartWebRTC() {
}
auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
auto it = _rtp_info_pt.find(m.plan[0].pt);
CHECK(it != _rtp_info_pt.end());
auto it = _pt_to_track.find(m.plan[0].pt);
CHECK(it != _pt_to_track.end());
//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
_send_rtp_info[m.type] = it->second.second;
_type_to_track[m.type] = it->second.second;
}
}
}
@@ -558,8 +558,7 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
class RtpChannel : public RtpReceiver {
public:
uint32_t ssrc;
RtcpContext::Ptr rtcp_context;
uint32_t rtp_ssrc;
public:
RtpChannel(function<void(RtpPacket::Ptr rtp)> on_rtp, function<void(const FCI_NACK &nack)> on_nack) {
@@ -576,11 +575,16 @@ public:
//统计rtp接受情况，便于生成nack rtcp包
nack_ctx.received(seq);
//统计rtp收到的情况，好做rr汇报
rtcp_context->onRtp(seq, ntohl(rtp->stamp) * uint64_t(1000) / sample_rate, len);
rtcp_context.onRtp(seq, ntohl(rtp->stamp), len);
}
return handleOneRtp((int) type, type, sample_rate, ptr, len);
}
Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
rtcp_context.onRtcp(sr);
return rtcp_context.createRtcpRR(ssrc, rtp_ssrc);
}
protected:
void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override {
_on_sort(std::move(rtp));
@@ -588,6 +592,7 @@ protected:
private:
NackContext nack_ctx;
RtcpContext rtcp_context{true};
function<void(RtpPacket::Ptr rtp)> _on_sort;
};
@@ -611,15 +616,14 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
case RtcpType::RTCP_SR : {
//对方汇报rtp发送情况
RtcpSR *sr = (RtcpSR *) rtcp;
auto it = _rtp_info_ssrc.find(sr->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto it = _ssrc_to_track.find(sr->ssrc);
if (it != _ssrc_to_track.end()) {
auto &info = it->second;
auto rtp_chn = info->getRtpChannel(sr->ssrc);
if(!rtp_chn){
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
} else {
rtp_chn->rtcp_context->onRtcp(sr);
auto rr = rtp_chn->rtcp_context->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc);
auto rr = rtp_chn->createRtcpRR(sr, info->answer_ssrc_rtp);
sendRtcpPacket(rr->data(), rr->size(), true);
}
} else {
@@ -632,8 +636,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
//对方汇报rtp接收情况
RtcpRR *rr = (RtcpRR *) rtcp;
for (auto item : rr->getItemList()) {
auto it = _rtp_info_ssrc.find(item->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto it = _ssrc_to_track.find(item->ssrc);
if (it != _ssrc_to_track.end()) {
auto &info = it->second;
auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp);
sendRtcpPacket(sr->data(), sr->size(), true);
@@ -647,12 +651,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
//对方汇报停止发送rtp
RtcpBye *bye = (RtcpBye *) rtcp;
for (auto ssrc : bye->getSSRC()) {
auto it = _rtp_info_ssrc.find(*ssrc);
if (it == _rtp_info_ssrc.end()) {
auto it = _ssrc_to_track.find(*ssrc);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
continue;
}
_rtp_info_ssrc.erase(it);
_ssrc_to_track.erase(it);
}
onShutdown(SockException(Err_eof, "rtcp bye message received"));
break;
@@ -666,8 +670,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
switch ((RTPFBType) rtcp->report_count) {
case RTPFBType::RTCP_RTPFB_NACK : {
RtcpFB *fb = (RtcpFB *) rtcp;
auto it = _rtp_info_ssrc.find(fb->ssrc_media);
if (it == _rtp_info_ssrc.end()) {
auto it = _ssrc_to_track.find(fb->ssrc_media);
if (it == _ssrc_to_track.end()) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
@@ -752,11 +756,9 @@ void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, cons
onSendNack(*info, nack, ssrc);
});
//rid --> rtp ssrc
ref->ssrc = ssrc;
//rtp ssrc --> RtcpContext
ref->rtcp_context = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
ref->rtp_ssrc = ssrc;
//rtp ssrc --> MediaTrack
_rtp_info_ssrc[ssrc] = info;
_ssrc_to_track[ssrc] = info;
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << info->plan_rtp->codec;
}
@@ -766,8 +768,8 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
RtpHeader *rtp = (RtpHeader *) buf;
//根据接收到的rtp的pt信息，找到该流的信息
auto it = _rtp_info_pt.find(rtp->pt);
if (it == _rtp_info_pt.end()) {
auto it = _pt_to_track.find(rtp->pt);
if (it == _pt_to_track.end()) {
WarnL << "unknown rtp pt:" << (int)rtp->pt;
return;
}
@@ -822,7 +824,7 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
//rtx 转换为 rtp
rtp->pt = info->plan_rtp->pt;
rtp->seq = htons(origin_seq);
rtp->ssrc = htonl(ref->ssrc);
rtp->ssrc = htonl(ref->rtp_ssrc);
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
@@ -878,14 +880,14 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &info, const string &rid, RtpPac
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
auto &info = _send_rtp_info[rtp->type];
auto &info = _type_to_track[rtp->type];
if (!info) {
//忽略，对方不支持该编码类型
return;
}
if (!rtx) {
//统计rtp发送情况，好做sr汇报
info->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
info->rtcp_context_send->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
info->nack_list.push_back(rtp);
#if 0
//此处模拟发送丢包