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实现rtc转rtsp

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This commit is contained in:
xiongziliang
2021-04-03 09:34:49 +08:00
parent 130a06897f
commit 2abb5078f9
4 changed files with 80 additions and 16 deletions
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26
webrtc/WebRtcTransport.cpp
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@@ -253,6 +253,9 @@ void WebRtcTransportImp::onStartWebRTC() {
_push_src->setSdp(rtsp_sdp);
for (auto &m : getSdp(SdpType::offer).media) {
if (m.type == TrackVideo) {
_recv_video_ssrc = m.rtp_ssrc.ssrc;
}
for (auto &plan : m.plan) {
auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt);
if (!hit_pan) {
@@ -416,6 +419,10 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
//todo 此处应该销毁对象
break;
}
case RtcpType::RTCP_PSFB: {
// InfoL << rtcp->dumpString();
break;
}
default: break;
}
}
@@ -434,18 +441,6 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
}
int makeRtcpPli(char *packet, int len) {
if (packet == NULL || len != 12)
return -1;
memset(packet, 0, len);
RtcpHeader *rtcp = (RtcpHeader *) packet;
rtcp->version = 2;
rtcp->pt = (uint8_t) RtcpType::RTCP_PSFB;
rtcp->report_count = 1;
rtcp->length = htons((len / 4) - 1);
return 12;
}
///////////////////////////////////////////////////////////////////
void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
@@ -457,9 +452,10 @@ void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr
if (_pli_ticker.elapsedTime() > 2000) {
//todo 定期发送pli
_pli_ticker.resetTime();
char rtcpbuf[12];
makeRtcpPli(rtcpbuf, 12);
sendRtcpPacket(rtcpbuf, 12, true);
auto pli = RtcpPli::create();
pli->ssrc = htonl(0);
pli->ssrc_media = htonl(_recv_video_ssrc);
sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true);
InfoL << "send pli";
}
_push_src->onWrite(std::move(rtp), false);