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ZLMediaKit
Commit ebb947c6 by ziyue
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完善simulcast与ssrc相关处理代码

parent 3862c8f1
正在显示 包含 89 行增加 和 105 行删除
webrtc/WebRtcTransport.cpp
......@@ -255,7 +255,8 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
onRtp(buf, len);
} else {
WarnL;
RtpHeader *rtp = (RtpHeader *) buf;
WarnL << "srtp_unprotect rtp failed, pt:" << (int)rtp->pt;
}
return;
}
......@@ -411,12 +412,10 @@ void WebRtcTransportImp::onStartWebRTC() {
info->rtcp_context_send = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, false);
//send ssrc --> RtpPayloadInfo
_rtp_info_ssrc[info->answer_ssrc_rtp] = std::make_pair(false, info);
_rtp_info_ssrc[info->answer_ssrc_rtx] = std::make_pair(true, info);
_rtp_info_ssrc[info->answer_ssrc_rtp] = info;
//recv ssrc --> RtpPayloadInfo
_rtp_info_ssrc[info->offer_ssrc_rtp] = std::make_pair(false, info);;
_rtp_info_ssrc[info->offer_ssrc_rtx] = std::make_pair(true, info);;
_rtp_info_ssrc[info->offer_ssrc_rtp] = info;
//rtp pt --> RtpPayloadInfo
_rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info));
......@@ -589,9 +588,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
RtcpSR *sr = (RtcpSR *) rtcp;
auto it = _rtp_info_ssrc.find(sr->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
auto &info = it->second;
auto it = info->rtcp_context_recv.find(sr->ssrc);
if (it != info->rtcp_context_recv.end()) {
it->second->onRtcp(sr);
......@@ -600,7 +597,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
}
}
} else {
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
}
......@@ -613,12 +609,9 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
for (auto item : rr->getItemList()) {
auto it = _rtp_info_ssrc.find(item->ssrc);
if (it != _rtp_info_ssrc.end()) {
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
auto &info = it->second;
auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp);
sendRtcpPacket(sr->data(), sr->size(), true);
}
} else {
WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
}
......@@ -653,15 +646,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
auto rtx = it->second.first;
if (!rtx) {
auto &info = it->second.second;
auto &info = it->second;
auto &fci = fb->getFci<FCI_NACK>();
info->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
}
break;
}
default: break;
......@@ -675,7 +665,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx, string *rid_ptr) const{
void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, string *rid_ptr) const{
string rid, repaired_rid;
auto ext_map = RtpExt::getExtValue(header);
for (auto &pr : ext_map) {
if (is_recv) {
......@@ -688,33 +679,9 @@ void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *h
pr.second.setType(it->second);
//重新赋值ext id为 ext type，作为后面处理ext的统一中间类型
pr.second.setExtId((uint8_t) it->second);
switch(it->second){
case RtpExtType::sdes_repaired_rtp_stream_id :
case RtpExtType::sdes_rtp_stream_id : {
auto ssrc = ntohl(header->ssrc);
auto rid = it->second == RtpExtType::sdes_rtp_stream_id ? pr.second.getRtpStreamId() : pr.second.getRepairedRtpStreamId();
//根据rid获取rtp或rtx的ssrc
auto &ssrc_ref = is_rtx ? info.rid_ssrc[rid].second : info.rid_ssrc[rid].first;
if (!ssrc_ref) {
//ssrc未赋值，赋值
ssrc_ref = ssrc;
DebugL << (is_rtx ? "got rid of rtx:" : "got rid:") << rid << ", ssrc:" << ssrc;
}
if (is_rtx) {
//rtx ssrc --> rtp ssrc
auto &rtp_ssrc_ref = info.rtx_ssrc_to_rtp_ssrc[ssrc];
if (!rtp_ssrc_ref && info.rid_ssrc[rid].first) {
//未找到rtx到rtp ssrc的映射关系，且已经获取rtp的ssrc，那么设置映射关系
rtp_ssrc_ref = info.rid_ssrc[rid].first;
DebugL << "got ssrc of rid:" << rid << ", [rtx-rtp]:" << ssrc << "-" << rtp_ssrc_ref;
}
}
if (rid_ptr) {
*rid_ptr = rid;
}
break;
}
switch (it->second) {
case RtpExtType::sdes_rtp_stream_id : rid = pr.second.getRtpStreamId(); break;
case RtpExtType::sdes_repaired_rtp_stream_id : repaired_rid = pr.second.getRepairedRtpStreamId(); break;
default : break;
}
} else {
......@@ -729,74 +696,98 @@ void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *h
pr.second.setExtId(it->second);
}
}
if (!is_recv) {
return;
}
if (rid.empty()) {
rid = repaired_rid;
}
auto ssrc = ntohl(header->ssrc);
if (rid.empty()) {
//获取rid
rid = info.ssrc_to_rid[ssrc];
} else {
//设置rid
info.ssrc_to_rid[ssrc] = rid;
}
if (rid_ptr) {
*rid_ptr = rid;
}
}
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
onRtp_l(buf, len, false);
std::shared_ptr<RtpReceiverImp> WebRtcTransportImp::createRtpReceiver(const string &rid, uint32_t ssrc, bool is_rtx, const RtpPayloadInfo::Ptr &info){
auto ref = std::make_shared<RtpReceiverImp>([info, this, rid](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, rid, std::move(rtp));
});
if (!is_rtx) {
//rtx没nack
info->nack_ctx[rid].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*info, nack, ssrc);
});
//ssrc --> RtpPayloadInfo
_rtp_info_ssrc[ssrc] = info;
}
InfoL << "receive rtp of ssrc:" << ssrc << ", rid:" << rid << ", is rtx:" << is_rtx << ", codec:" << info->plan_rtp->codec;
return ref;
}
void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
if (!rtx) {
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
_bytes_usage += len;
_alive_ticker.resetTime();
}
RtpHeader *rtp = (RtpHeader *) buf;
auto ssrc = ntohl(rtp->ssrc);
//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()) {
WarnL;
WarnL << "unknown rtp pt:" << (int)rtp->pt;
return;
}
bool is_rtx = it->second.first;
auto ssrc = ntohl(rtp->ssrc);
auto &info = it->second.second;
if (!it->second.first) {
//修改ext id至统一
string rid;
changeRtpExtId(*info, rtp, true, &rid);
#if 0
if (rid.empty() && info->media->type == TrackVideo) {
WarnL << "ssrc:" << ssrc << ", rtx:" << is_rtx << ",seq:" << ntohs((uint16_t) rtp->seq);
}
#endif
auto &ref = info->receiver[rid];
if (!ref) {
ref = createRtpReceiver(rid, ssrc, is_rtx, info);
}
if (!is_rtx) {
//这是普通的rtp数据
auto seq = ntohs(rtp->seq);
#if 0
if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) {
if (info->media->type == TrackVideo && seq % 100 == 0) {
//此处模拟接受丢包
DebugL << "recv dropped:" << seq;
return;
}
#endif
auto &ref = info->receiver[ssrc];
if (!rtx) {
//统计rtp接受情况，便于生成nack rtcp包
info->nack_ctx[ssrc].received(seq);
//时间戳转换成毫秒
auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate;
info->nack_ctx[rid].received(seq);
//统计rtp收到的情况，好做rr汇报
auto &cxt_ref = info->rtcp_context_recv[ssrc];
if (!cxt_ref) {
cxt_ref = std::make_shared<RtcpContext>(info->plan_rtp->sample_rate, true);
info->rid_to_ssrc[rid] = ssrc;
}
//时间戳转换成毫秒
auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate;
//统计rtp收到的情况，好做rr汇报
cxt_ref->onRtp(seq, stamp_ms, len);
//修改ext id至统一
string rid;
changeRtpExtId(*info, rtp, true, false, &rid);
if (!ref) {
ref = std::make_shared<RtpReceiverImp>([info, this, rid](RtpPacket::Ptr rtp) mutable {
onSortedRtp(*info, rid, std::move(rtp));
});
info->nack_ctx[ssrc].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable {
onSendNack(*info, nack, ssrc);
});
//recv simulcast ssrc --> RtpPayloadInfo
_rtp_info_ssrc[ssrc] = std::make_pair(false, info);
InfoL << "receive rtp of ssrc:" << ssrc;
}
}
//解析并排序rtp
if (ref) {
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
} else {
WarnL << "rtp dropped, ssrc:" << ssrc << ", is rtx:" << rtx;
}
return;
}
......@@ -808,26 +799,18 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
return;
}
//修改ext id至统一
changeRtpExtId(*info, rtp, true, true);
//前两个字节是原始的rtp的seq
auto origin_seq = payload[0] << 8 | payload[1];
//rtx的seq转换为rtp的seq
rtp->seq = htons(origin_seq);
if (info->offer_ssrc_rtp) {
//非simulcast或音频
rtp->ssrc = htonl(info->offer_ssrc_rtp);
TraceL << "received rtx rtp,ssrc: " << ssrc << ", seq:" << origin_seq << ", pt:" << (int)rtp->pt;
} else {
//todo simulcast下，辅码流通过rtx传输？
//simulcast情况下，根据rtx的ssrc查找rtp的ssrc
rtp->ssrc = htonl(info->rtx_ssrc_to_rtp_ssrc[ntohl(rtp->ssrc)]);
}
//rtx的ssrc转换为rtp的ssrc
rtp->ssrc = htonl(info->rid_to_ssrc[rid]);
//rtx的pt转换为rtp的pt
rtp->pt = info->plan_rtp->pt;
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
buf += 2;
len -= 2;
onRtp_l(buf, len, true);
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
}
void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc) {
......@@ -908,12 +891,12 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *
if (!pr->first || !pr->second->plan_rtx) {
//普通的rtp,或者不支持rtx, 修改目标pt和ssrc
changeRtpExtId(*pr->second, header, false, false);
changeRtpExtId(*pr->second, header, false);
header->pt = pr->second->plan_rtp->pt;
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
} else {
//重传的rtp, rtx
changeRtpExtId(*pr->second, header, false, true);
changeRtpExtId(*pr->second, header, false);
header->pt = pr->second->plan_rtx->pt;
if (pr->second->answer_ssrc_rtx) {
//有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc
......
webrtc/WebRtcTransport.h
......@@ -298,8 +298,6 @@ protected:
void onRtcConfigure(RtcConfigure &configure) const override;
void onRtp(const char *buf, size_t len) override;
void onRtp_l(const char *buf, size_t len, bool rtx);
void onRtcp(const char *buf, size_t len) override;
void onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) override;
void onBeforeEncryptRtcp(const char *buf, size_t &len, void *ctx) override {};
......@@ -351,16 +349,19 @@ private:
const RtcMedia *media;
NackList nack_list;
RtcpContext::Ptr rtcp_context_send;
unordered_map<string/*rid*/, std::pair<uint32_t/*rtp ssrc*/, uint32_t/*rtx ssrc*/> > rid_ssrc;
unordered_map<uint32_t/*rtx ssrc*/, uint32_t/*rtp ssrc*/> rtx_ssrc_to_rtp_ssrc;
unordered_map<uint32_t/*simulcast ssrc*/, NackContext> nack_ctx;
//只生成rtp的nack，rtx的不生成
unordered_map<string/*rid*/, NackContext> nack_ctx;
unordered_map<string/*rid*/, uint32_t/*rtp ssrc*/> rid_to_ssrc;
unordered_map<string/*rid*/, std::shared_ptr<RtpReceiverImp> > receiver;
unordered_map<uint32_t/*simulcast ssrc*/, string/*rid*/> ssrc_to_rid;
//只统计rtp的接收情况，rtx的不统计
unordered_map<uint32_t/*simulcast ssrc*/, RtcpContext::Ptr> rtcp_context_recv;
unordered_map<uint32_t/*simulcast ssrc*/, std::shared_ptr<RtpReceiverImp> > receiver;
};
void onSortedRtp(RtpPayloadInfo &info, const string &rid, RtpPacket::Ptr rtp);
void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc);
void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx = false, string *rid_ptr = nullptr) const;
void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, string *rid_ptr = nullptr) const;
std::shared_ptr<RtpReceiverImp> createRtpReceiver(const string &rid, uint32_t ssrc, bool is_rtx, const RtpPayloadInfo::Ptr &info);
private:
uint16_t _rtx_seq[2] = {0, 0};
......@@ -389,8 +390,8 @@ private:
RtpPayloadInfo::Ptr _send_rtp_info[2];
//根据接收rtp的pt获取相关信息
unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_pt;
//根据rtcp的ssrc获取相关信息
unordered_map<uint32_t/*ssrc*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_ssrc;
//根据rtcp的ssrc获取相关信息，只记录rtp的ssrc，rtx的ssrc不记录
unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo::Ptr> _rtp_info_ssrc;
//发送rtp时需要修改rtp ext id
map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
//接收rtp时需要修改rtp ext id
......
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