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rtc播放支持nack重传

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This commit is contained in:
xiongziliang
2021-05-11 00:54:33 +08:00
parent e8d3dec050
commit 7ad361b22d
2 changed files with 122 additions and 19 deletions
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58
webrtc/WebRtcTransport.cpp
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@@ -269,21 +269,22 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
}
}
void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, TrackType type) {
void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, void *ctx) {
if (_srtp_session_send) {
CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
memcpy(_srtp_buf, buf, len);
onBeforeEncryptRtp((char *) _srtp_buf, len, type);
onBeforeEncryptRtp((char *) _srtp_buf, len, ctx);
if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
onSendSockData((char *) _srtp_buf, len, flush);
}
}
}
void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){
void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush, void *ctx){
if (_srtp_session_send) {
CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
memcpy(_srtp_buf, buf, len);
onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx);
if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
onSendSockData((char *) _srtp_buf, len, flush);
}
@@ -414,10 +415,13 @@ void WebRtcTransportImp::onStartWebRTC() {
auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type);
//获取offer端rtp的ssrc和pt相关信息
auto &ref = _rtp_info_pt[plan.pt];
_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
ref.plan = &plan;
ref.media = m_with_ssrc;
ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
if (ref.is_common_rtp) {
//rtp
_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
}
ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
ref.receiver = std::make_shared<RtpReceiverImp>([&ref, this](RtpPacket::Ptr rtp) {
@@ -628,7 +632,30 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
}
case RtcpType::RTCP_PSFB:
case RtcpType::RTCP_RTPFB: {
// DebugL << "\r\n" << rtcp->dumpString();
RtcpFB *fb = (RtcpFB *) rtcp;
auto it = _rtp_info_ssrc.find(fb->ssrc);
if (it == _rtp_info_ssrc.end()) {
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
return;
}
if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) {
// DebugL << "\r\n" << rtcp->dumpString();
break;
}
//RTPFB
switch ((RTPFBType) rtcp->report_count) {
case RTPFBType::RTCP_RTPFB_NACK : {
auto &fci = fb->getFci<FCI_NACK>();
it->second->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) {
//rtp重传
onSendRtp(rtp, true, true);
});
break;
}
default:
// DebugL << "\r\n" << rtcp->dumpString();
break;
}
break;
}
default: break;
@@ -703,27 +730,36 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp
info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
auto info = _send_rtp_info[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->nack_list.push_back(rtp);
} else {
WarnL << "重传rtp:" << rtp->getSeq();
}
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info);
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type);
//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);
}
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) {
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) {
RtpPayloadInfo *info = reinterpret_cast<RtpPayloadInfo *>(ctx);
auto header = (RtpHeader *)buf;
auto info = _send_rtp_info[type];
//修改目标pt和ssrc
header->pt = info->plan->pt;
header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc);
changeRtpExtId(header, _rtp_ext_type_to_id);
}
void WebRtcTransportImp::onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) {
}
void WebRtcTransportImp::onShutdown(const SockException &ex){
InfoL << ex.what();
_self = nullptr;