mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-10-06 12:13:49 +08:00
rtc推流支持nack/rtx重传
This commit is contained in:
xia-chu
parent
0d61de758e
commit
cb280e1eeb
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@ -421,6 +421,10 @@ void WebRtcTransportImp::onStartWebRTC() {
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if (ref.is_common_rtp) {
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//rtp
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_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
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} else {
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//rtx
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auto apt = atoi(plan.getFmtp("apt").data());
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ref.plan_apt = m_with_ssrc->getPlan(apt);
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}
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ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
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ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
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@ -642,7 +646,6 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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return;
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}
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if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) {
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// DebugL << "\r\n" << rtcp->dumpString();
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break;
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}
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//RTPFB
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@ -655,9 +658,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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});
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break;
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}
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default:
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// DebugL << "\r\n" << rtcp->dumpString();
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break;
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default: break;
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}
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break;
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}
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@ -666,73 +667,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
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}
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}
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void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
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_bytes_usage += len;
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_alive_ticker.resetTime();
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RtpHeader *rtp = (RtpHeader *) buf;
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//根据接收到的rtp的pt信息，找到该流的信息
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auto it = _rtp_info_pt.find(rtp->pt);
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if (it == _rtp_info_pt.end()) {
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WarnL;
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return;
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}
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auto &info = it->second;
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#if 1
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auto header = (RtpHeader *) buf;
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auto seq = ntohs(header->seq);
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if (info.is_common_rtp) {
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//此处模拟接受丢包
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if (info.media->type == TrackVideo && seq % 10 == 0) {
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//丢包
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DebugL << "模拟接受丢包:" << seq;
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return;
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} else {
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info.nack_ctx.received(seq);
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}
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} else {
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//收到重传包
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header->ssrc = info.media->rtp_rtx_ssrc[0].ssrc;
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InfoL << "收到重传包:" << seq;
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}
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#endif
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//解析并排序rtp
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info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
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}
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void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) {
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auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
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rtcp->ssrc = htons(0);
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rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc);
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InfoL << rtcp->RtcpHeader::dumpString();
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sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
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if (!info.is_common_rtp) {
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WarnL;
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return;
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}
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if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
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//定期发送pli请求关键帧，方便非rtc等协议
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_pli_ticker.resetTime();
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sendRtcpPli(rtp->getSSRC());
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//开启remb，则发送remb包调节比特率
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GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
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if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) {
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sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
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}
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}
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if (_push_src) {
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_push_src->onWrite(std::move(rtp), false);
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}
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}
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static void setExtType(RtpExt &ext, uint8_t tp) {}
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static void setExtType(RtpExt &ext, RtpExtType tp) {
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ext.setType(tp);
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@ -750,17 +686,98 @@ static void changeRtpExtId(const RtpHeader *header, const Type &map) {
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}
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setExtType(pr.second, it->first);
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setExtType(pr.second, it->second);
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// DebugL << pr.second.dumpString();
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pr.second.setExtId((uint8_t) it->second);
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}
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}
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void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
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onRtp_l(buf, len, false);
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}
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void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) {
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_bytes_usage += len;
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_alive_ticker.resetTime();
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RtpHeader *rtp = (RtpHeader *) buf;
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//根据接收到的rtp的pt信息，找到该流的信息
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auto it = _rtp_info_pt.find(rtp->pt);
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if (it == _rtp_info_pt.end()) {
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WarnL;
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return;
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}
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auto &info = it->second;
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if (info.is_common_rtp) {
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//这是普通的rtp数据
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auto seq = ntohs(rtp->seq);
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#if 0
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if (!rtx && info.media->type == TrackVideo && seq % 100 == 0) {
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//此处模拟接受丢包
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DebugL << "recv dropped:" << seq;
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return;
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}
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#endif
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if (!rtx) {
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//统计rtp接受情况，便于生成nack rtcp包
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info.nack_ctx.received(seq);
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}
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//解析并排序rtp
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info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
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return;
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}
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//这里是rtx重传包
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//https://datatracker.ietf.org/doc/html/rfc4588#section-4
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auto payload = rtp->getPayloadData();
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auto size = rtp->getPayloadSize(len);
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if (size < 2) {
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return;
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}
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//前两个字节是原始的rtp的seq
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auto origin_seq = payload[0] << 8 | payload[1];
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InfoL << "received rtx rtp: " << origin_seq;
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rtp->seq = htons(origin_seq);
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rtp->ssrc = htonl(info.media->rtp_rtx_ssrc[0].ssrc);
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rtp->pt = info.plan_apt->pt;
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memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
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buf += 2;
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len -= 2;
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onRtp_l(buf, len, true);
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}
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void WebRtcTransportImp::onNack(RtpPayloadInfo &info, const FCI_NACK &nack) {
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auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
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rtcp->ssrc = htons(0);
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rtcp->ssrc_media = htonl(info.media->rtp_rtx_ssrc[0].ssrc);
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sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onBeforeSortedRtp(RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
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changeRtpExtId(rtp->getHeader(), _rtp_ext_id_to_type);
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//统计rtp收到的情况，好做rr汇报
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info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
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}
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void WebRtcTransportImp::onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
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if (info.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
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//定期发送pli请求关键帧，方便非rtc等协议
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_pli_ticker.resetTime();
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sendRtcpPli(rtp->getSSRC());
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//开启remb，则发送remb包调节比特率
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GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
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if (remb_bit_rate && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) {
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sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
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}
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}
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if (_push_src) {
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_push_src->onWrite(std::move(rtp), false);
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}
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}
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///////////////////////////////////////////////////////////////////
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void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx){
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auto info = _send_rtp_info[rtp->type];
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if (!info) {
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@ -773,13 +790,13 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
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info->nack_list.push_back(rtp);
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#if 0
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//此处模拟发送丢包
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if(rtp->getSeq() % 10 == 0){
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DebugL << "模拟发送丢包:" << rtp->getSeq();
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if(rtp->getSeq() % 100 == 0){
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DebugL << "send droped:" << rtp->getSeq();
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return;
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}
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#endif
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} else {
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WarnL << "rtp发送重传:" << rtp->getSeq();
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WarnL << "send rtx rtp:" << rtp->getSeq();
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}
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sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, info);
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_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
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@ -298,6 +298,8 @@ protected:
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void onRtcConfigure(RtcConfigure &configure) const override;
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void onRtp(const char *buf, size_t len) override;
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void onRtp_l(const char *buf, size_t len, bool rtx);
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void onRtcp(const char *buf, size_t len) override;
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void onBeforeEncryptRtp(const char *buf, size_t len, void *ctx) override;
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void onBeforeEncryptRtcp(const char *buf, size_t len, void *ctx) override;
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@ -342,6 +344,7 @@ private:
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public:
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bool is_common_rtp;
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const RtcCodecPlan *plan;
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const RtcCodecPlan *plan_apt;
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const RtcMedia *media;
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std::shared_ptr<RtpReceiverImp> receiver;
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RtcpContext::Ptr rtcp_context_recv;
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