mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-23 11:17:09 +08:00
1065 lines
38 KiB
C++
1065 lines
38 KiB
C++
/*
|
||
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
|
||
*
|
||
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
|
||
*
|
||
* Use of this source code is governed by MIT license that can be found in the
|
||
* LICENSE file in the root of the source tree. All contributing project authors
|
||
* may be found in the AUTHORS file in the root of the source tree.
|
||
*/
|
||
|
||
#include "WebRtcTransport.h"
|
||
#include <iostream>
|
||
#include "RtpExt.h"
|
||
#include "Rtcp/Rtcp.h"
|
||
#include "Rtcp/RtcpFCI.h"
|
||
#include "Rtsp/RtpReceiver.h"
|
||
|
||
#define RTP_SSRC_OFFSET 1
|
||
#define RTX_SSRC_OFFSET 2
|
||
#define RTP_CNAME "zlmediakit-rtp"
|
||
#define RTP_LABEL "zlmediakit-label"
|
||
#define RTP_MSLABEL "zlmediakit-mslabel"
|
||
#define RTP_MSID RTP_MSLABEL " " RTP_LABEL
|
||
|
||
using namespace std;
|
||
using namespace mediakit;
|
||
|
||
//RTC配置项目
|
||
namespace RTC {
|
||
#define RTC_FIELD "rtc."
|
||
//rtp和rtcp接受超时时间
|
||
const string kTimeOutSec = RTC_FIELD"timeoutSec";
|
||
//服务器外网ip
|
||
const string kExternIP = RTC_FIELD"externIP";
|
||
//设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质
|
||
const string kRembBitRate = RTC_FIELD"rembBitRate";
|
||
//webrtc单端口udp服务器
|
||
const string kPort = RTC_FIELD"port";
|
||
|
||
static onceToken token([]() {
|
||
mINI::Instance()[kTimeOutSec] = 15;
|
||
mINI::Instance()[kExternIP] = "";
|
||
mINI::Instance()[kRembBitRate] = 0;
|
||
mINI::Instance()[kPort] = 8000;
|
||
});
|
||
|
||
}//namespace RTC
|
||
|
||
static atomic<uint64_t> s_key{0};
|
||
|
||
WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
|
||
_poller = poller;
|
||
_identifier = "zlm_" + to_string(++s_key);
|
||
_packet_pool.setSize(64);
|
||
}
|
||
|
||
void WebRtcTransport::onCreate(){
|
||
_dtls_transport = std::make_shared<RTC::DtlsTransport>(_poller, this);
|
||
_ice_server = std::make_shared<RTC::IceServer>(this, _identifier, makeRandStr(24));
|
||
}
|
||
|
||
void WebRtcTransport::onDestory(){
|
||
_dtls_transport = nullptr;
|
||
_ice_server = nullptr;
|
||
}
|
||
|
||
const EventPoller::Ptr& WebRtcTransport::getPoller() const{
|
||
return _poller;
|
||
}
|
||
|
||
const string &WebRtcTransport::getIdentifier() const {
|
||
return _identifier;
|
||
}
|
||
|
||
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) {
|
||
sendSockData((char *) packet->GetData(), packet->GetSize(), tuple);
|
||
}
|
||
|
||
void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
|
||
InfoL;
|
||
}
|
||
|
||
void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) {
|
||
InfoL;
|
||
}
|
||
|
||
void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) {
|
||
InfoL;
|
||
if (_answer_sdp->media[0].role == DtlsRole::passive) {
|
||
_dtls_transport->Run(RTC::DtlsTransport::Role::SERVER);
|
||
} else {
|
||
_dtls_transport->Run(RTC::DtlsTransport::Role::CLIENT);
|
||
}
|
||
}
|
||
|
||
void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) {
|
||
InfoL;
|
||
}
|
||
|
||
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransport::OnDtlsTransportConnected(
|
||
const RTC::DtlsTransport *dtlsTransport,
|
||
RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
|
||
uint8_t *srtpLocalKey,
|
||
size_t srtpLocalKeyLen,
|
||
uint8_t *srtpRemoteKey,
|
||
size_t srtpRemoteKeyLen,
|
||
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::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
|
||
onStartWebRTC();
|
||
}
|
||
|
||
void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
|
||
sendSockData((char *)data, len, nullptr);
|
||
}
|
||
|
||
void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
|
||
InfoL;
|
||
}
|
||
|
||
void WebRtcTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) {
|
||
InfoL;
|
||
onShutdown(SockException(Err_shutdown, "dtls transport failed"));
|
||
}
|
||
|
||
void WebRtcTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
|
||
InfoL;
|
||
onShutdown(SockException(Err_shutdown, "dtls close notify received"));
|
||
}
|
||
|
||
void WebRtcTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
|
||
InfoL << hexdump(data, len);
|
||
}
|
||
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple){
|
||
auto pkt = _packet_pool.obtain2();
|
||
pkt->assign(buf, len);
|
||
onSendSockData(std::move(pkt), true, tuple ? tuple : _ice_server->GetSelectedTuple());
|
||
}
|
||
|
||
RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{
|
||
return _ice_server->GetSelectedTuple();
|
||
}
|
||
|
||
void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
|
||
auto remb = FCI_REMB::create({ssrc}, (uint32_t)bit_rate);
|
||
auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size());
|
||
fb->ssrc = htonl(0);
|
||
fb->ssrc_media = htonl(ssrc);
|
||
sendRtcpPacket((char *) fb.get(), fb->getSize(), true);
|
||
}
|
||
|
||
void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
|
||
auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
|
||
pli->ssrc = htonl(0);
|
||
pli->ssrc_media = htonl(ssrc);
|
||
sendRtcpPacket((char *) pli.get(), pli->getSize(), true);
|
||
}
|
||
|
||
string getFingerprint(const string &algorithm_str, const std::shared_ptr<RTC::DtlsTransport> &transport){
|
||
auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str);
|
||
for (auto &finger_prints : transport->GetLocalFingerprints()) {
|
||
if (finger_prints.algorithm == algorithm) {
|
||
return finger_prints.value;
|
||
}
|
||
}
|
||
throw std::invalid_argument(StrPrinter << "不支持的加密算法:" << algorithm_str);
|
||
}
|
||
|
||
void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){
|
||
//设置远端dtls签名
|
||
RTC::DtlsTransport::Fingerprint remote_fingerprint;
|
||
remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm);
|
||
remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash;
|
||
_dtls_transport->SetRemoteFingerprint(remote_fingerprint);
|
||
}
|
||
|
||
void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
|
||
//开启remb后关闭twcc，因为开启twcc后remb无效
|
||
GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
|
||
configure.enableTWCC(!remb_bit_rate);
|
||
}
|
||
|
||
std::string WebRtcTransport::getAnswerSdp(const string &offer){
|
||
try {
|
||
//// 解析offer sdp ////
|
||
_offer_sdp = std::make_shared<RtcSession>();
|
||
_offer_sdp->loadFrom(offer);
|
||
onCheckSdp(SdpType::offer, *_offer_sdp);
|
||
_offer_sdp->checkValid();
|
||
setRemoteDtlsFingerprint(*_offer_sdp);
|
||
|
||
//// sdp 配置 ////
|
||
SdpAttrFingerprint fingerprint;
|
||
fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
|
||
fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
|
||
RtcConfigure configure;
|
||
configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(),
|
||
RtpDirection::sendrecv, fingerprint);
|
||
onRtcConfigure(configure);
|
||
|
||
//// 生成answer sdp ////
|
||
_answer_sdp = configure.createAnswer(*_offer_sdp);
|
||
onCheckSdp(SdpType::answer, *_answer_sdp);
|
||
_answer_sdp->checkValid();
|
||
return _answer_sdp->toString();
|
||
} catch (exception &ex) {
|
||
onShutdown(SockException(Err_shutdown, ex.what()));
|
||
throw;
|
||
}
|
||
}
|
||
|
||
static bool is_dtls(char *buf) {
|
||
return ((*buf > 19) && (*buf < 64));
|
||
}
|
||
|
||
static bool is_rtp(char *buf) {
|
||
RtpHeader *header = (RtpHeader *) buf;
|
||
return ((header->pt < 64) || (header->pt >= 96));
|
||
}
|
||
|
||
static bool is_rtcp(char *buf) {
|
||
RtpHeader *header = (RtpHeader *) buf;
|
||
return ((header->pt >= 64) && (header->pt < 96));
|
||
}
|
||
|
||
static string getPeerAddress(RTC::TransportTuple *tuple){
|
||
return SockUtil::inet_ntoa(((struct sockaddr_in *)tuple)->sin_addr) + ":" + to_string(ntohs(((struct sockaddr_in *)tuple)->sin_port));
|
||
}
|
||
|
||
void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
|
||
if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
|
||
std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *) buf, len));
|
||
if (!packet) {
|
||
WarnL << "parse stun error" << std::endl;
|
||
return;
|
||
}
|
||
_ice_server->ProcessStunPacket(packet.get(), tuple);
|
||
return;
|
||
}
|
||
if (is_dtls(buf)) {
|
||
_dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
|
||
return;
|
||
}
|
||
if (is_rtp(buf)) {
|
||
if (!_srtp_session_recv) {
|
||
WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
|
||
return;
|
||
}
|
||
if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
|
||
onRtp(buf, len, _ticker.createdTime());
|
||
}
|
||
return;
|
||
}
|
||
if (is_rtcp(buf)) {
|
||
if (!_srtp_session_recv) {
|
||
WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
|
||
return;
|
||
}
|
||
if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
|
||
onRtcp(buf, len);
|
||
}
|
||
return;
|
||
}
|
||
}
|
||
|
||
void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
|
||
if (_srtp_session_send) {
|
||
auto pkt = _packet_pool.obtain2();
|
||
//预留rtx加入的两个字节
|
||
pkt->setCapacity((size_t) len + SRTP_MAX_TRAILER_LEN + 2);
|
||
pkt->assign(buf, len);
|
||
onBeforeEncryptRtp(pkt->data(), len, ctx);
|
||
if (_srtp_session_send->EncryptRtp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
|
||
pkt->setSize(len);
|
||
onSendSockData(std::move(pkt), flush);
|
||
}
|
||
}
|
||
}
|
||
|
||
void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx) {
|
||
if (_srtp_session_send) {
|
||
auto pkt = _packet_pool.obtain2();
|
||
//预留rtx加入的两个字节
|
||
pkt->setCapacity((size_t) len + SRTP_MAX_TRAILER_LEN + 2);
|
||
pkt->assign(buf, len);
|
||
onBeforeEncryptRtcp(pkt->data(), len, ctx);
|
||
if (_srtp_session_send->EncryptRtcp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
|
||
pkt->setSize(len);
|
||
onSendSockData(std::move(pkt), flush);
|
||
}
|
||
}
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransportImp::onCreate(){
|
||
WebRtcTransport::onCreate();
|
||
registerSelf();
|
||
|
||
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
|
||
GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
|
||
_timer = std::make_shared<Timer>(timeoutSec / 2, [weak_self]() {
|
||
auto strong_self = weak_self.lock();
|
||
if (!strong_self) {
|
||
return false;
|
||
}
|
||
if (strong_self->_alive_ticker.elapsedTime() > timeoutSec * 1000) {
|
||
strong_self->onShutdown(SockException(Err_timeout, "接受rtp和rtcp超时"));
|
||
}
|
||
return true;
|
||
}, getPoller());
|
||
|
||
_twcc_ctx.setOnSendTwccCB([this](uint32_t ssrc, string fci) {
|
||
onSendTwcc(ssrc, fci);
|
||
});
|
||
}
|
||
|
||
WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
|
||
InfoL << getIdentifier();
|
||
}
|
||
|
||
WebRtcTransportImp::~WebRtcTransportImp() {
|
||
InfoL << getIdentifier();
|
||
}
|
||
|
||
void WebRtcTransportImp::onDestory() {
|
||
WebRtcTransport::onDestory();
|
||
unregisterSelf();
|
||
}
|
||
|
||
void WebRtcTransportImp::onSendSockData(Buffer::Ptr buf, bool flush, RTC::TransportTuple *tuple) {
|
||
if (!_selected_session) {
|
||
WarnL << "send data failed:" << buf->size();
|
||
return;
|
||
}
|
||
//一次性发送一帧的rtp数据，提高网络io性能
|
||
_selected_session->setSendFlushFlag(flush);
|
||
_selected_session->send(std::move(buf));
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
bool WebRtcTransportImp::canSendRtp() const{
|
||
for (auto &m : _answer_sdp->media) {
|
||
if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::sendonly) {
|
||
return true;
|
||
}
|
||
}
|
||
return false;
|
||
}
|
||
|
||
bool WebRtcTransportImp::canRecvRtp() const{
|
||
for (auto &m : _answer_sdp->media) {
|
||
if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::recvonly) {
|
||
return true;
|
||
}
|
||
}
|
||
return false;
|
||
}
|
||
|
||
void WebRtcTransportImp::onStartWebRTC() {
|
||
//获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
|
||
for (auto &m_answer : _answer_sdp->media) {
|
||
auto m_offer = _offer_sdp->getMedia(m_answer.type);
|
||
auto track = std::make_shared<MediaTrack>();
|
||
|
||
track->media = &m_answer;
|
||
track->answer_ssrc_rtp = m_answer.getRtpSSRC();
|
||
track->answer_ssrc_rtx = m_answer.getRtxSSRC();
|
||
track->offer_ssrc_rtp = m_offer->getRtpSSRC();
|
||
track->offer_ssrc_rtx = m_offer->getRtxSSRC();
|
||
track->plan_rtp = &m_answer.plan[0];;
|
||
track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
|
||
track->rtcp_context_send = std::make_shared<RtcpContextForSend>();
|
||
|
||
//rtp track type --> MediaTrack
|
||
if (m_answer.direction == RtpDirection::sendonly || m_answer.direction == RtpDirection::sendrecv) {
|
||
//该类型的track 才支持发送
|
||
_type_to_track[m_answer.type] = track;
|
||
}
|
||
//send ssrc --> MediaTrack
|
||
_ssrc_to_track[track->answer_ssrc_rtp] = track;
|
||
_ssrc_to_track[track->answer_ssrc_rtx] = track;
|
||
|
||
//recv ssrc --> MediaTrack
|
||
_ssrc_to_track[track->offer_ssrc_rtp] = track;
|
||
_ssrc_to_track[track->offer_ssrc_rtx] = track;
|
||
|
||
//rtp pt --> MediaTrack
|
||
_pt_to_track.emplace(track->plan_rtp->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtpTrack(track, _twcc_ctx, *this)));
|
||
if (track->plan_rtx) {
|
||
//rtx pt --> MediaTrack
|
||
_pt_to_track.emplace(track->plan_rtx->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtxTrack(track)));
|
||
}
|
||
if (m_offer->type != TrackApplication) {
|
||
//记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id
|
||
track->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
|
||
weak_ptr<MediaTrack> weak_track = track;
|
||
track->rtp_ext_ctx->setOnGetRtp([this, weak_track](uint8_t pt, uint32_t ssrc, const string &rid) {
|
||
//ssrc --> MediaTrack
|
||
auto track = weak_track.lock();
|
||
assert(track);
|
||
_ssrc_to_track[ssrc] = std::move(track);
|
||
InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
|
||
});
|
||
|
||
size_t index = 0;
|
||
for (auto &ssrc : m_offer->rtp_ssrc_sim) {
|
||
//记录ssrc对应的MediaTrack
|
||
_ssrc_to_track[ssrc.ssrc] = track;
|
||
if (m_offer->rtp_rids.size() > index) {
|
||
//支持firefox的simulcast, 提前映射好ssrc和rid的关系
|
||
track->rtp_ext_ctx->setRid(ssrc.ssrc, m_offer->rtp_rids[index]);
|
||
}
|
||
++index;
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
|
||
//修改answer sdp的ip、端口信息
|
||
GET_CONFIG(string, extern_ip, RTC::kExternIP);
|
||
for (auto &m : sdp.media) {
|
||
m.addr.reset();
|
||
m.addr.address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
|
||
m.rtcp_addr.reset();
|
||
m.rtcp_addr.address = m.addr.address;
|
||
|
||
GET_CONFIG(uint16_t, local_port, RTC::kPort);
|
||
m.rtcp_addr.port = local_port;
|
||
m.port = m.rtcp_addr.port;
|
||
sdp.origin.address = m.addr.address;
|
||
}
|
||
|
||
if (!canSendRtp()) {
|
||
//设置我们发送的rtp的ssrc
|
||
return;
|
||
}
|
||
|
||
for (auto &m : sdp.media) {
|
||
if (m.type == TrackApplication) {
|
||
continue;
|
||
}
|
||
if (!m.rtp_rtx_ssrc.empty()) {
|
||
//已经生成了ssrc
|
||
continue;
|
||
}
|
||
//添加answer sdp的ssrc信息
|
||
m.rtp_rtx_ssrc.emplace_back();
|
||
auto &ssrc = m.rtp_rtx_ssrc.back();
|
||
//发送的ssrc我们随便定义，因为在发送rtp时会修改为此值
|
||
ssrc.ssrc = m.type + RTP_SSRC_OFFSET;
|
||
ssrc.cname = RTP_CNAME;
|
||
ssrc.label = RTP_LABEL;
|
||
ssrc.mslabel = RTP_MSLABEL;
|
||
ssrc.msid = RTP_MSID;
|
||
|
||
if (m.getRelatedRtxPlan(m.plan[0].pt)) {
|
||
//rtx ssrc
|
||
ssrc.rtx_ssrc = ssrc.ssrc + RTX_SSRC_OFFSET;
|
||
}
|
||
}
|
||
}
|
||
|
||
void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) {
|
||
switch (type) {
|
||
case SdpType::answer: onCheckAnswer(sdp); break;
|
||
case SdpType::offer: break;
|
||
default: /*不可达*/ assert(0); break;
|
||
}
|
||
}
|
||
|
||
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
|
||
WebRtcTransport::onRtcConfigure(configure);
|
||
//添加接收端口candidate信息
|
||
configure.addCandidate(*getIceCandidate());
|
||
}
|
||
|
||
SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
|
||
auto candidate = std::make_shared<SdpAttrCandidate>();
|
||
candidate->foundation = "udpcandidate";
|
||
//rtp端口
|
||
candidate->component = 1;
|
||
candidate->transport = "udp";
|
||
//优先级，单candidate时随便
|
||
candidate->priority = 100;
|
||
GET_CONFIG(string, extern_ip, RTC::kExternIP);
|
||
candidate->address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
|
||
GET_CONFIG(uint16_t, local_port, RTC::kPort);
|
||
candidate->port = local_port;
|
||
candidate->type = "host";
|
||
return candidate;
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {
|
||
public:
|
||
RtpChannel(EventPoller::Ptr poller, RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) {
|
||
_poller = std::move(poller);
|
||
_on_nack = std::move(on_nack);
|
||
setOnSorted(std::move(cb));
|
||
|
||
_nack_ctx.setOnNack([this](const FCI_NACK &nack) {
|
||
onNack(nack);
|
||
});
|
||
}
|
||
|
||
~RtpChannel() override = default;
|
||
|
||
RtpPacket::Ptr inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx) {
|
||
auto rtp = RtpTrack::inputRtp(type, sample_rate, ptr, len);
|
||
if (!rtp) {
|
||
return rtp;
|
||
}
|
||
auto seq = rtp->getSeq();
|
||
_nack_ctx.received(seq, is_rtx);
|
||
if (!is_rtx) {
|
||
//统计rtp接受情况，便于生成nack rtcp包
|
||
_rtcp_context.onRtp(seq, rtp->getStamp(), rtp->ntp_stamp, sample_rate, len);
|
||
}
|
||
return rtp;
|
||
}
|
||
|
||
Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
|
||
_rtcp_context.onRtcp(sr);
|
||
return _rtcp_context.createRtcpRR(ssrc, getSSRC());
|
||
}
|
||
|
||
int getLossRate() {
|
||
return _rtcp_context.geLostInterval() * 100 / _rtcp_context.getExpectedPacketsInterval();
|
||
}
|
||
|
||
private:
|
||
void starNackTimer(){
|
||
if (_delay_task) {
|
||
return;
|
||
}
|
||
weak_ptr<RtpChannel> weak_self = shared_from_this();
|
||
_delay_task = _poller->doDelayTask(10, [weak_self]() -> uint64_t {
|
||
auto strong_self = weak_self.lock();
|
||
if (!strong_self) {
|
||
return 0;
|
||
}
|
||
auto ret = strong_self->_nack_ctx.reSendNack();
|
||
if (!ret) {
|
||
strong_self->_delay_task = nullptr;
|
||
}
|
||
return ret;
|
||
});
|
||
}
|
||
|
||
void onNack(const FCI_NACK &nack) {
|
||
_on_nack(nack);
|
||
starNackTimer();
|
||
}
|
||
|
||
private:
|
||
NackContext _nack_ctx;
|
||
RtcpContextForRecv _rtcp_context;
|
||
EventPoller::Ptr _poller;
|
||
EventPoller::DelayTask::Ptr _delay_task;
|
||
function<void(const FCI_NACK &nack)> _on_nack;
|
||
};
|
||
|
||
std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const{
|
||
auto it_chn = rtp_channel.find(rtp_ext_ctx->getRid(ssrc));
|
||
if (it_chn == rtp_channel.end()) {
|
||
return nullptr;
|
||
}
|
||
return it_chn->second;
|
||
}
|
||
|
||
void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
|
||
_bytes_usage += len;
|
||
auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len);
|
||
for (auto rtcp : rtcps) {
|
||
switch ((RtcpType) rtcp->pt) {
|
||
case RtcpType::RTCP_SR : {
|
||
//对方汇报rtp发送情况
|
||
RtcpSR *sr = (RtcpSR *) rtcp;
|
||
auto it = _ssrc_to_track.find(sr->ssrc);
|
||
if (it != _ssrc_to_track.end()) {
|
||
auto &track = it->second;
|
||
auto rtp_chn = track->getRtpChannel(sr->ssrc);
|
||
if(!rtp_chn){
|
||
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
|
||
} else {
|
||
//InfoL << "接收丢包率,ssrc:" << sr->ssrc << ",loss rate(%):" << rtp_chn->getLossRate();
|
||
//设置rtp时间戳与ntp时间戳的对应关系
|
||
rtp_chn->setNtpStamp(sr->rtpts, sr->getNtpUnixStampMS());
|
||
auto rr = rtp_chn->createRtcpRR(sr, track->answer_ssrc_rtp);
|
||
sendRtcpPacket(rr->data(), rr->size(), true);
|
||
}
|
||
} else {
|
||
WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
|
||
}
|
||
break;
|
||
}
|
||
case RtcpType::RTCP_RR : {
|
||
_alive_ticker.resetTime();
|
||
//对方汇报rtp接收情况
|
||
RtcpRR *rr = (RtcpRR *) rtcp;
|
||
for (auto item : rr->getItemList()) {
|
||
auto it = _ssrc_to_track.find(item->ssrc);
|
||
if (it != _ssrc_to_track.end()) {
|
||
auto &track = it->second;
|
||
track->rtcp_context_send->onRtcp(rtcp);
|
||
auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
|
||
sendRtcpPacket(sr->data(), sr->size(), true);
|
||
} else {
|
||
WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
|
||
}
|
||
}
|
||
break;
|
||
}
|
||
case RtcpType::RTCP_BYE : {
|
||
//对方汇报停止发送rtp
|
||
RtcpBye *bye = (RtcpBye *) rtcp;
|
||
for (auto ssrc : bye->getSSRC()) {
|
||
auto it = _ssrc_to_track.find(*ssrc);
|
||
if (it == _ssrc_to_track.end()) {
|
||
WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
|
||
continue;
|
||
}
|
||
_ssrc_to_track.erase(it);
|
||
}
|
||
onShutdown(SockException(Err_eof, "rtcp bye message received"));
|
||
break;
|
||
}
|
||
case RtcpType::RTCP_PSFB:
|
||
case RtcpType::RTCP_RTPFB: {
|
||
if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) {
|
||
break;
|
||
}
|
||
//RTPFB
|
||
switch ((RTPFBType) rtcp->report_count) {
|
||
case RTPFBType::RTCP_RTPFB_NACK : {
|
||
RtcpFB *fb = (RtcpFB *) rtcp;
|
||
auto it = _ssrc_to_track.find(fb->ssrc_media);
|
||
if (it == _ssrc_to_track.end()) {
|
||
WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
|
||
return;
|
||
}
|
||
auto &track = it->second;
|
||
auto &fci = fb->getFci<FCI_NACK>();
|
||
track->nack_list.forEach(fci, [&](const RtpPacket::Ptr &rtp) {
|
||
//rtp重传
|
||
onSendRtp(rtp, true, true);
|
||
});
|
||
break;
|
||
}
|
||
default: break;
|
||
}
|
||
break;
|
||
}
|
||
default: break;
|
||
}
|
||
}
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track) {
|
||
//rid --> RtpReceiverImp
|
||
auto &ref = track.rtp_channel[rid];
|
||
weak_ptr<WebRtcTransportImp> weak_self = dynamic_pointer_cast<WebRtcTransportImp>(shared_from_this());
|
||
ref = std::make_shared<RtpChannel>(getPoller(), [&track, this, rid](RtpPacket::Ptr rtp) mutable {
|
||
onSortedRtp(track, rid, std::move(rtp));
|
||
}, [&track, weak_self, ssrc](const FCI_NACK &nack) mutable {
|
||
//nack发送可能由定时器异步触发
|
||
auto strong_self = weak_self.lock();
|
||
if (strong_self) {
|
||
strong_self->onSendNack(track, nack, ssrc);
|
||
}
|
||
});
|
||
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << track.plan_rtp->codec;
|
||
}
|
||
|
||
void WebRtcTransportImp::updateTicker() {
|
||
_alive_ticker.resetTime();
|
||
}
|
||
|
||
void WebRtcTransportImp::onRtp(const char *buf, size_t len, uint64_t stamp_ms) {
|
||
_bytes_usage += len;
|
||
_alive_ticker.resetTime();
|
||
|
||
RtpHeader *rtp = (RtpHeader *) buf;
|
||
//根据接收到的rtp的pt信息，找到该流的信息
|
||
auto it = _pt_to_track.find(rtp->pt);
|
||
if (it == _pt_to_track.end()) {
|
||
WarnL << "unknown rtp pt:" << (int)rtp->pt;
|
||
return;
|
||
}
|
||
it->second->inputRtp(buf, len, stamp_ms, rtp);
|
||
}
|
||
|
||
void WrappedRtpTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
|
||
#if 0
|
||
auto seq = ntohs(rtp->seq);
|
||
if (track->media->type == TrackVideo && seq % 100 == 0) {
|
||
//此处模拟接受丢包
|
||
return;
|
||
}
|
||
#endif
|
||
|
||
auto ssrc = ntohl(rtp->ssrc);
|
||
|
||
//修改ext id至统一
|
||
string rid;
|
||
auto twcc_ext = track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
|
||
|
||
if (twcc_ext) {
|
||
_twcc_ctx.onRtp(ssrc, twcc_ext.getTransportCCSeq(), stamp_ms);
|
||
}
|
||
|
||
auto &ref = track->rtp_channel[rid];
|
||
if (!ref) {
|
||
_transport.createRtpChannel(rid, ssrc, *track);
|
||
}
|
||
|
||
//解析并排序rtp
|
||
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, false);
|
||
}
|
||
|
||
void WrappedRtxTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
|
||
//修改ext id至统一
|
||
string rid;
|
||
track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
|
||
|
||
auto &ref = track->rtp_channel[rid];
|
||
if (!ref) {
|
||
//再接收到对应的rtp前，丢弃rtx包
|
||
WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ntohl(rtp->ssrc) << ", codec:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
|
||
return;
|
||
}
|
||
|
||
//这里是rtx重传包
|
||
// https://datatracker.ietf.org/doc/html/rfc4588#section-4
|
||
auto payload = rtp->getPayloadData();
|
||
auto size = rtp->getPayloadSize(len);
|
||
if (size < 2) {
|
||
return;
|
||
}
|
||
|
||
//前两个字节是原始的rtp的seq
|
||
auto origin_seq = payload[0] << 8 | payload[1];
|
||
// rtx 转换为 rtp
|
||
rtp->pt = track->plan_rtp->pt;
|
||
rtp->seq = htons(origin_seq);
|
||
rtp->ssrc = htonl(ref->getSSRC());
|
||
|
||
memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
|
||
buf += 2;
|
||
len -= 2;
|
||
ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, true);
|
||
}
|
||
|
||
void WebRtcTransportImp::onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc) {
|
||
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
|
||
rtcp->ssrc = htonl(track.answer_ssrc_rtp);
|
||
rtcp->ssrc_media = htonl(ssrc);
|
||
sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
|
||
}
|
||
|
||
void WebRtcTransportImp::onSendTwcc(uint32_t ssrc, const string &twcc_fci) {
|
||
auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_TWCC, twcc_fci.data(), twcc_fci.size());
|
||
rtcp->ssrc = htonl(0);
|
||
rtcp->ssrc_media = htonl(ssrc);
|
||
sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
|
||
if (track.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
|
||
//定期发送pli请求关键帧，方便非rtc等协议
|
||
_pli_ticker.resetTime();
|
||
sendRtcpPli(rtp->getSSRC());
|
||
|
||
//开启remb，则发送remb包调节比特率
|
||
GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
|
||
if (remb_bit_rate && _answer_sdp->supportRtcpFb(SdpConst::kRembRtcpFb)) {
|
||
sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
|
||
}
|
||
}
|
||
|
||
onRecvRtp(track, rid, std::move(rtp));
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx) {
|
||
auto &track = _type_to_track[rtp->type];
|
||
if (!track) {
|
||
//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
|
||
track->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
|
||
track->nack_list.pushBack(rtp);
|
||
#if 0
|
||
//此处模拟发送丢包
|
||
if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
|
||
return;
|
||
}
|
||
#endif
|
||
} else {
|
||
//发送rtx重传包
|
||
//TraceL << "send rtx rtp:" << rtp->getSeq();
|
||
}
|
||
pair<bool/*rtx*/, MediaTrack *> ctx{rtx, track.get()};
|
||
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
|
||
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
|
||
}
|
||
|
||
void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx) {
|
||
auto pr = (pair<bool/*rtx*/, MediaTrack *> *) ctx;
|
||
auto header = (RtpHeader *) buf;
|
||
|
||
if (!pr->first || !pr->second->plan_rtx) {
|
||
//普通的rtp,或者不支持rtx, 修改目标pt和ssrc
|
||
pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
|
||
header->pt = pr->second->plan_rtp->pt;
|
||
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
|
||
} else {
|
||
//重传的rtp, rtx
|
||
pr->second->rtp_ext_ctx->changeRtpExtId(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
|
||
header->ssrc = htonl(pr->second->answer_ssrc_rtx);
|
||
} else {
|
||
//未单独指定rtx的ssrc，那么使用rtp的ssrc
|
||
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
|
||
}
|
||
|
||
auto origin_seq = ntohs(header->seq);
|
||
//seq跟原来的不一样
|
||
header->seq = htons(_rtx_seq[pr->second->media->type]);
|
||
++_rtx_seq[pr->second->media->type];
|
||
|
||
auto payload = header->getPayloadData();
|
||
auto payload_size = header->getPayloadSize(len);
|
||
if (payload_size) {
|
||
//rtp负载后移两个字节，这两个字节用于存放osn
|
||
//https://datatracker.ietf.org/doc/html/rfc4588#section-4
|
||
memmove(payload + 2, payload, payload_size);
|
||
}
|
||
payload[0] = origin_seq >> 8;
|
||
payload[1] = origin_seq & 0xFF;
|
||
len += 2;
|
||
}
|
||
}
|
||
|
||
void WebRtcTransportImp::onShutdown(const SockException &ex){
|
||
WarnL << ex.what();
|
||
unrefSelf();
|
||
for (auto &pr : _history_sessions) {
|
||
auto session = pr.second.lock();
|
||
if (session) {
|
||
session->shutdown(ex);
|
||
}
|
||
}
|
||
}
|
||
|
||
void WebRtcTransportImp::setSession(Session::Ptr session) {
|
||
_history_sessions.emplace(session.get(), session);
|
||
if (_selected_session) {
|
||
InfoL << "rtc network changed: " << _selected_session->get_peer_ip() << ":" << _selected_session->get_peer_port()
|
||
<< " -> " << session->get_peer_ip() << ":" << session->get_peer_port() << ", id:" << getIdentifier();
|
||
}
|
||
_selected_session = std::move(session);
|
||
unrefSelf();
|
||
}
|
||
|
||
const Session::Ptr &WebRtcTransportImp::getSession() const {
|
||
return _selected_session;
|
||
}
|
||
|
||
uint64_t WebRtcTransportImp::getBytesUsage() const{
|
||
return _bytes_usage;
|
||
}
|
||
|
||
uint64_t WebRtcTransportImp::getDuration() const{
|
||
return _alive_ticker.createdTime() / 1000;
|
||
}
|
||
|
||
/////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
void WebRtcTransportImp::registerSelf() {
|
||
_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
|
||
WebRtcTransportManager::Instance().addItem(getIdentifier(), _self);
|
||
}
|
||
|
||
void WebRtcTransportImp::unrefSelf() {
|
||
_self = nullptr;
|
||
}
|
||
|
||
void WebRtcTransportImp::unregisterSelf() {
|
||
unrefSelf();
|
||
WebRtcTransportManager::Instance().removeItem(getIdentifier());
|
||
}
|
||
|
||
WebRtcTransportManager &WebRtcTransportManager::Instance() {
|
||
static WebRtcTransportManager s_instance;
|
||
return s_instance;
|
||
}
|
||
|
||
void WebRtcTransportManager::addItem(const string &key, const WebRtcTransportImp::Ptr &ptr) {
|
||
lock_guard<mutex> lck(_mtx);
|
||
_map[key] = ptr;
|
||
}
|
||
|
||
WebRtcTransportImp::Ptr WebRtcTransportManager::getItem(const string &key) {
|
||
if (key.empty()) {
|
||
return nullptr;
|
||
}
|
||
lock_guard<mutex> lck(_mtx);
|
||
auto it = _map.find(key);
|
||
if (it == _map.end()) {
|
||
return nullptr;
|
||
}
|
||
return it->second.lock();
|
||
}
|
||
|
||
void WebRtcTransportManager::removeItem(const string &key) {
|
||
lock_guard<mutex> lck(_mtx);
|
||
_map.erase(key);
|
||
}
|
||
|
||
//////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
WebRtcPluginManager &WebRtcPluginManager::Instance() {
|
||
static WebRtcPluginManager s_instance;
|
||
return s_instance;
|
||
}
|
||
|
||
void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
|
||
lock_guard<mutex> lck(_mtx_creator);
|
||
_map_creator[type] = std::move(cb);
|
||
}
|
||
|
||
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const string &offer, const WebRtcArgs &args,
|
||
const onCreateRtc &cb) {
|
||
lock_guard<mutex> lck(_mtx_creator);
|
||
auto it = _map_creator.find(type);
|
||
if (it == _map_creator.end()) {
|
||
cb(WebRtcException(SockException(Err_other, "the type can not supported")));
|
||
return;
|
||
}
|
||
it->second(sender, offer, args, cb);
|
||
}
|
||
|
||
#include "WebRtcPlayer.h"
|
||
#include "WebRtcPusher.h"
|
||
#include "WebRtcEchoTest.h"
|
||
|
||
void echo_plugin(Session &sender, const string &offer, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
|
||
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
|
||
}
|
||
|
||
void push_plugin(Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
|
||
MediaInfo info(args["url"]);
|
||
Broadcast::PublishAuthInvoker invoker = [cb, offer_sdp, info](const string &err, bool enable_hls, bool enable_mp4) mutable {
|
||
if (!err.empty()) {
|
||
cb(WebRtcException(SockException(Err_other, err)));
|
||
return;
|
||
}
|
||
|
||
RtspMediaSourceImp::Ptr push_src;
|
||
std::shared_ptr<void> push_src_ownership;
|
||
auto src = MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid);
|
||
auto push_failed = (bool)src;
|
||
|
||
while (src) {
|
||
//尝试断连后继续推流
|
||
auto rtsp_src = dynamic_pointer_cast<RtspMediaSourceImp>(src);
|
||
if (!rtsp_src) {
|
||
//源不是rtsp推流产生的
|
||
break;
|
||
}
|
||
auto ownership = rtsp_src->getOwnership();
|
||
if (!ownership) {
|
||
//获取推流源所有权失败
|
||
break;
|
||
}
|
||
push_src = std::move(rtsp_src);
|
||
push_src_ownership = std::move(ownership);
|
||
push_failed = false;
|
||
break;
|
||
}
|
||
|
||
if (push_failed) {
|
||
cb(WebRtcException(SockException(Err_other, "already publishing")));
|
||
return;
|
||
}
|
||
|
||
if (!push_src) {
|
||
push_src = std::make_shared<RtspMediaSourceImp>(info._vhost, info._app, info._streamid);
|
||
push_src_ownership = push_src->getOwnership();
|
||
push_src->setProtocolTranslation(enable_hls, enable_mp4);
|
||
}
|
||
auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info);
|
||
push_src->setListener(rtc);
|
||
cb(*rtc);
|
||
};
|
||
|
||
//rtsp推流需要鉴权
|
||
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, info, invoker, static_cast<SockInfo &>(sender));
|
||
if (!flag) {
|
||
//该事件无人监听,默认不鉴权
|
||
GET_CONFIG(bool, to_hls, General::kPublishToHls);
|
||
GET_CONFIG(bool, to_mp4, General::kPublishToMP4);
|
||
invoker("", to_hls, to_mp4);
|
||
}
|
||
}
|
||
|
||
void play_plugin(Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
|
||
MediaInfo info(args["url"]);
|
||
auto session_ptr = sender.shared_from_this();
|
||
Broadcast::AuthInvoker invoker = [cb, offer_sdp, info, session_ptr](const string &err) mutable {
|
||
if (!err.empty()) {
|
||
cb(WebRtcException(SockException(Err_other, err)));
|
||
return;
|
||
}
|
||
|
||
//webrtc播放的是rtsp的源
|
||
info._schema = RTSP_SCHEMA;
|
||
MediaSource::findAsync(info, session_ptr, [=](const MediaSource::Ptr &src_in) mutable {
|
||
auto src = dynamic_pointer_cast<RtspMediaSource>(src_in);
|
||
if (!src) {
|
||
cb(WebRtcException(SockException(Err_other, "stream not found")));
|
||
return;
|
||
}
|
||
//还原成rtc，目的是为了hook时识别哪种播放协议
|
||
info._schema = RTC_SCHEMA;
|
||
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
|
||
cb(*rtc);
|
||
});
|
||
};
|
||
|
||
//广播通用播放url鉴权事件
|
||
auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, info, invoker, static_cast<SockInfo &>(sender));
|
||
if (!flag) {
|
||
//该事件无人监听,默认不鉴权
|
||
invoker("");
|
||
}
|
||
}
|
||
|
||
static onceToken s_rtc_auto_register([](){
|
||
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
|
||
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
|
||
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
|
||
});
|