基本完成rtc转rtsp

This commit is contained in:
xiongziliang
2021-04-02 23:01:58 +08:00
parent dc485c6211
commit 8c460bfcff
4 changed files with 184 additions and 8 deletions

View File

@@ -1,6 +1,11 @@
#include "WebRtcTransport.h"
#include <iostream>
#include "Rtcp/Rtcp.h"
#include "Rtsp/RtpReceiver.h"
#define RTX_SSRC_OFFSET 2
#define RTP_CNAME "zlmediakit-rtp"
#define RTX_CNAME "zlmediakit-rtx"
WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
_dtls_transport = std::make_shared<RTC::DtlsTransport>(poller, this);
@@ -51,7 +56,7 @@ void WebRtcTransport::OnDtlsTransportConnected(
std::string &remoteCert) {
InfoL;
_srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);
_srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
_srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
onStartWebRTC();
}
@@ -209,6 +214,29 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) {
}
void WebRtcTransportImp::onStartWebRTC() {
if (canRecvRtp()) {
_push_src = std::make_shared<RtspMediaSourceImp>(DEFAULT_VHOST, "live", "push");
auto rtsp_sdp = getSdp(SdpType::answer).toRtspSdp();
_push_src->setSdp(rtsp_sdp);
for (auto &m : getSdp(SdpType::offer).media) {
for (auto &plan : m.plan) {
auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt);
if (!hit_pan) {
continue;
}
auto &ref = _rtp_receiver[plan.pt];
ref.plan = &plan;
ref.media = &m;
ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
ref.receiver = std::make_shared<RtpReceiverImp>([&ref, this](RtpPacket::Ptr rtp) {
onSortedRtp(ref, std::move(rtp));
}, [ref, this](const RtpPacket::Ptr &rtp) {
onBeforeSortedRtp(ref, rtp);
});
}
}
}
if (!canSendRtp()) {
return;
}
@@ -244,6 +272,11 @@ bool WebRtcTransportImp::canSendRtp() const{
return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly;
}
bool WebRtcTransportImp::canRecvRtp() const{
auto &sdp = getSdp(SdpType::answer);
return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly;
}
void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
WebRtcTransport::onCheckSdp(type, sdp);
if (type != SdpType::answer || !canSendRtp()) {
@@ -255,7 +288,12 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
continue;
}
m.rtp_ssrc.ssrc = _src->getSsrc(m.type);
m.rtp_ssrc.cname = "zlmediakit-rtc";
m.rtp_ssrc.cname = RTP_CNAME;
//todo 先屏蔽rtx因为chrome报错
if (false && m.getRelatedRtxPlan(m.plan[0].pt)) {
m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc;
m.rtx_ssrc.cname = RTX_CNAME;
}
auto rtsp_media = _rtsp_send_sdp.getMedia(m.type);
if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
_send_rtp_pt[m.type] = m.plan[0].pt;
@@ -309,14 +347,60 @@ SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
return candidate;
}
class RtpReceiverImp : public RtpReceiver {
public:
RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb, function<void(const RtpPacket::Ptr &rtp)> cb_before = nullptr){
_on_sort = std::move(cb);
_on_before_sort = std::move(cb_before);
}
~RtpReceiverImp() override = default;
bool inputRtp(TrackType type, int samplerate, uint8_t *ptr, size_t len){
return handleOneRtp((int) type, type, samplerate, ptr, len);
}
protected:
void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override {
_on_sort(std::move(rtp));
}
void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override {
if (_on_before_sort) {
_on_before_sort(rtp);
}
}
private:
function<void(RtpPacket::Ptr rtp)> _on_sort;
function<void(const RtpPacket::Ptr &rtp)> _on_before_sort;
};
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
RtpHeader *rtp = (RtpHeader *) buf;
auto it = _rtp_receiver.find(rtp->pt);
if (it == _rtp_receiver.end()) {
WarnL;
return;
}
auto &info = it->second;
info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
}
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
RtcpHeader *rtcp = (RtcpHeader *) buf;
}
void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
if(!info.is_common_rtp){
WarnL;
}
_push_src->onWrite(std::move(rtp), true);
}
void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
}
///////////////////////////////////////////////////////////////////