mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2026-06-17 05:25:57 +08:00
feat: 增加webrtc代理拉流 (#4389)
- 增加客户端模式，支持主动拉流、推流： - addStreamProxy接口新增支持whep主动拉流，拉流地址目前只兼容zlm的whep url。 - addStreamPusherProxy接口新增支持whip主动推流，推流地址目前只兼容zlm的whip url。 - 以上推流url格式为webrtc[s]://server_host:server_port/app/stream_id?key=value, 内部会自动转换为http[s]://server_host:server_port/index/api/[whip/whep]?app=app&stream=stream_id&key=value。 - 增加WebRtc p2p 模式： - 增加 ICE FULL模式。 - 增加STUN/TURN 服务器。 - 增加websocket 信令。 - 增加P2P代理拉流。 --------- Co-authored-by: xia-chu <771730766@qq.com> Co-authored-by: mtdxc <mtdxc@126.com> Co-authored-by: cqm <cqm@97kid.com>
This commit is contained in:
baigao-X
@@ -1,169 +1,232 @@
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/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#include "WebRtcPusher.h"
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#include "Common/config.h"
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#include "Rtsp/RtspMediaSourceImp.h"
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using namespace std;
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namespace mediakit {
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WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const ProtocolOption &option) {
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WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
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ptr->onDestory();
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delete ptr;
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});
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ret->onCreate();
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return ret;
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}
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WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const ProtocolOption &option) : WebRtcTransportImp(poller) {
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_media_info = info;
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_push_src = src;
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_push_src_ownership = ownership;
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_continue_push_ms = option.continue_push_ms;
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CHECK(_push_src);
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}
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bool WebRtcPusher::close(MediaSource &sender) {
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onShutdown(SockException(Err_shutdown, "close media: " + sender.getUrl()));
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// 主动关闭推流，那么不延时注销 [AUTO-TRANSLATED:ee7cc580]
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// Actively close the stream, then do not delay the logout
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_push_src = nullptr;
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return true;
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}
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int WebRtcPusher::totalReaderCount(MediaSource &sender) {
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auto total_count = _push_src ? _push_src->totalReaderCount() : 0;
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if (_simulcast) {
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std::lock_guard<std::recursive_mutex> lock(_mtx);
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for (auto &src : _push_src_sim) {
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total_count += src.second->totalReaderCount();
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}
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}
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return total_count;
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}
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MediaOriginType WebRtcPusher::getOriginType(MediaSource &sender) const {
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return MediaOriginType::rtc_push;
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}
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string WebRtcPusher::getOriginUrl(MediaSource &sender) const {
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return _media_info.full_url;
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}
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std::shared_ptr<SockInfo> WebRtcPusher::getOriginSock(MediaSource &sender) const {
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return static_pointer_cast<SockInfo>(getSession());
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}
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toolkit::EventPoller::Ptr WebRtcPusher::getOwnerPoller(MediaSource &sender) {
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return getPoller();
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}
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void WebRtcPusher::onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
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if (!_simulcast) {
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assert(_push_src);
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_push_src->onWrite(rtp, false);
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return;
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}
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if (rtp->type == TrackAudio) {
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// 音频 [AUTO-TRANSLATED:a577d8e1]
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// Audio
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for (auto &pr : _push_src_sim) {
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pr.second->onWrite(rtp, false);
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}
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} else {
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// 视频 [AUTO-TRANSLATED:904730ac]
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// Video
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std::lock_guard<std::recursive_mutex> lock(_mtx);
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auto &src = _push_src_sim[rid];
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if (!src) {
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const auto& stream = _push_src->getMediaTuple().stream;
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auto src_imp = _push_src->clone(rid.empty() ? stream : stream + '_' + rid);
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_push_src_sim_ownership[rid] = src_imp->getOwnership();
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src_imp->setListener(static_pointer_cast<WebRtcPusher>(shared_from_this()));
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src = src_imp;
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}
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src->onWrite(std::move(rtp), false);
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}
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}
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void WebRtcPusher::onStartWebRTC() {
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WebRtcTransportImp::onStartWebRTC();
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_simulcast = _answer_sdp->supportSimulcast();
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if (canRecvRtp()) {
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_push_src->setSdp(_answer_sdp->toRtspSdp());
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}
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}
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void WebRtcPusher::onDestory() {
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auto duration = getDuration();
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auto bytes_usage = getBytesUsage();
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// 流量统计事件广播 [AUTO-TRANSLATED:6b0b1234]
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// Traffic statistics event broadcast
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GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
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if (getSession()) {
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WarnL << "RTC推流器(" << _media_info.shortUrl() << ")结束推流,耗时(s):" << duration;
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if (bytes_usage >= iFlowThreshold * 1024) {
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NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration, false, *getSession());
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}
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}
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if (_push_src && _continue_push_ms) {
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// 取消所有权 [AUTO-TRANSLATED:4895d8fa]
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// Cancel ownership
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_push_src_ownership = nullptr;
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// 延时10秒注销流 [AUTO-TRANSLATED:e1bb11f9]
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// Delay 10 seconds to log out the stream
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auto push_src = std::move(_push_src);
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getPoller()->doDelayTask(_continue_push_ms, [push_src]() { return 0; });
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}
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WebRtcTransportImp::onDestory();
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}
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void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
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WebRtcTransportImp::onRtcConfigure(configure);
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// 这只是推流 [AUTO-TRANSLATED:f877bf98]
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// This is just pushing the stream
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configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
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}
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float WebRtcPusher::getLossRate(MediaSource &sender,TrackType type) {
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return WebRtcTransportImp::getLossRate(type);
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}
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void WebRtcPusher::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
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// 主动关闭推流，那么不等待重推 [AUTO-TRANSLATED:1ff514d7]
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// Actively close the stream, then do not wait for re-pushing
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_push_src = nullptr;
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WebRtcTransportImp::OnDtlsTransportClosed(dtlsTransport);
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}
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void WebRtcPusher::onRtcpBye() {
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WebRtcTransportImp::onRtcpBye();
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}
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void WebRtcPusher::onShutdown(const SockException &ex) {
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_push_src = nullptr;
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WebRtcTransportImp::onShutdown(ex);
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}
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}// namespace mediakit
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/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#include "WebRtcPusher.h"
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#include "Common/config.h"
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#include "Rtsp/RtspMediaSourceImp.h"
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using namespace std;
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using namespace toolkit;
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namespace mediakit {
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WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const ProtocolOption &option,
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WebRtcTransport::Role role,
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WebRtcTransport::SignalingProtocols signaling_protocols) {
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WebRtcPusher::Ptr pusher(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
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ptr->onDestory();
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delete ptr;
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});
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pusher->setRole(role);
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pusher->setSignalingProtocols(signaling_protocols);
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pusher->onCreate();
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return pusher;
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}
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WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const ProtocolOption &option) : WebRtcTransportImp(poller) {
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_media_info = info;
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_push_src = src;
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_push_src_ownership = ownership;
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_continue_push_ms = option.continue_push_ms;
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CHECK(_push_src);
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}
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bool WebRtcPusher::close(MediaSource &sender) {
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onShutdown(SockException(Err_shutdown, "close media: " + sender.getUrl()));
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// 主动关闭推流，那么不延时注销 [AUTO-TRANSLATED:ee7cc580]
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// Actively close the stream, then do not delay the logout
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_push_src = nullptr;
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return true;
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}
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int WebRtcPusher::totalReaderCount(MediaSource &sender) {
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auto total_count = _push_src ? _push_src->totalReaderCount() : 0;
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if (_simulcast) {
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std::lock_guard<std::recursive_mutex> lock(_mtx);
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for (auto &src : _push_src_sim) {
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total_count += src.second->totalReaderCount();
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}
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}
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return total_count;
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}
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MediaOriginType WebRtcPusher::getOriginType(MediaSource &sender) const {
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return MediaOriginType::rtc_push;
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}
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string WebRtcPusher::getOriginUrl(MediaSource &sender) const {
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return _media_info.full_url;
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}
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std::shared_ptr<SockInfo> WebRtcPusher::getOriginSock(MediaSource &sender) const {
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return static_pointer_cast<SockInfo>(getSession());
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}
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toolkit::EventPoller::Ptr WebRtcPusher::getOwnerPoller(MediaSource &sender) {
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return getPoller();
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}
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void WebRtcPusher::onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
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if (!_simulcast) {
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assert(_push_src);
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_push_src->onWrite(rtp, false);
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return;
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}
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if (rtp->type == TrackAudio) {
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// 音频 [AUTO-TRANSLATED:a577d8e1]
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// Audio
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for (auto &pr : _push_src_sim) {
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pr.second->onWrite(rtp, false);
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}
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} else {
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// 视频 [AUTO-TRANSLATED:904730ac]
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// Video
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std::lock_guard<std::recursive_mutex> lock(_mtx);
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auto &src = _push_src_sim[rid];
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if (!src) {
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const auto& stream = _push_src->getMediaTuple().stream;
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auto src_imp = _push_src->clone(rid.empty() ? stream : stream + '_' + rid);
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_push_src_sim_ownership[rid] = src_imp->getOwnership();
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src_imp->setListener(static_pointer_cast<WebRtcPusher>(shared_from_this()));
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src = src_imp;
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}
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src->onWrite(std::move(rtp), false);
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}
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}
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void WebRtcPusher::onStartWebRTC() {
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WebRtcTransportImp::onStartWebRTC();
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_simulcast = _answer_sdp->supportSimulcast();
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if (canRecvRtp()) {
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_push_src->setSdp(_answer_sdp->toRtspSdp());
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}
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}
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void WebRtcPusher::onDestory() {
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auto duration = getDuration();
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auto bytes_usage = getBytesUsage();
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// 流量统计事件广播 [AUTO-TRANSLATED:6b0b1234]
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// Traffic statistics event broadcast
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GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
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if (getSession()) {
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WarnL << "RTC推流器(" << _media_info.shortUrl() << ")结束推流,耗时(s):" << duration;
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if (bytes_usage >= iFlowThreshold * 1024) {
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NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration, false, *getSession());
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}
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}
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if (_push_src && _continue_push_ms) {
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// 取消所有权 [AUTO-TRANSLATED:4895d8fa]
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// Cancel ownership
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_push_src_ownership = nullptr;
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// 延时10秒注销流 [AUTO-TRANSLATED:e1bb11f9]
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// Delay 10 seconds to log out the stream
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auto push_src = std::move(_push_src);
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getPoller()->doDelayTask(_continue_push_ms, [push_src]() { return 0; });
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}
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WebRtcTransportImp::onDestory();
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}
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void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
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WebRtcTransportImp::onRtcConfigure(configure);
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// 这只是推流 [AUTO-TRANSLATED:f877bf98]
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// This is just pushing the stream
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configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
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}
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float WebRtcPusher::getLossRate(MediaSource &sender,TrackType type) {
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return WebRtcTransportImp::getLossRate(type);
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}
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void WebRtcPusher::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
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// 主动关闭推流，那么不等待重推 [AUTO-TRANSLATED:1ff514d7]
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// Actively close the stream, then do not wait for re-pushing
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_push_src = nullptr;
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WebRtcTransportImp::OnDtlsTransportClosed(dtlsTransport);
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}
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void WebRtcPusher::onRtcpBye() {
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WebRtcTransportImp::onRtcpBye();
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}
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void WebRtcPusher::onShutdown(const SockException &ex) {
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_push_src = nullptr;
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WebRtcTransportImp::onShutdown(ex);
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}
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////////////////////////////////////////////////////////////////////////////////////////
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WebRtcPlayerClient::Ptr WebRtcPlayerClient::create(const EventPoller::Ptr &poller, WebRtcTransport::Role role,
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WebRtcTransport::SignalingProtocols signaling_protocols) {
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WebRtcPlayerClient::Ptr pusher(new WebRtcPlayerClient(poller), [](WebRtcPlayerClient *ptr) {
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ptr->onDestory();
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delete ptr;
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});
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pusher->setRole(role);
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pusher->setSignalingProtocols(signaling_protocols);
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pusher->onCreate();
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return pusher;
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}
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WebRtcPlayerClient::WebRtcPlayerClient(const EventPoller::Ptr &poller)
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: WebRtcTransportImp(poller) {
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_demuxer = std::make_shared<RtspDemuxer>();
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}
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void WebRtcPlayerClient::onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
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auto key_pos = _demuxer->inputRtp(rtp);
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if (_push_src) {
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_push_src->onWrite(rtp, key_pos);
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}
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}
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void WebRtcPlayerClient::onStartWebRTC() {
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WebRtcTransportImp::onStartWebRTC();
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CHECK(!_answer_sdp->supportSimulcast());
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auto sdp = _answer_sdp->toRtspSdp();
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if (canRecvRtp()) {
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if (_push_src) {
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_push_src->setSdp(sdp);
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}
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_demuxer->loadSdp(sdp);
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}
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}
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void WebRtcPlayerClient::onRtcConfigure(RtcConfigure &configure) const {
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WebRtcTransportImp::onRtcConfigure(configure);
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// 这只是推流 [AUTO-TRANSLATED:f877bf98]
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// This is just pushing the stream
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configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
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}
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vector<Track::Ptr> WebRtcPlayerClient::getTracks(bool ready) const {
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return _demuxer->getTracks(ready);
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}
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void WebRtcPlayerClient::setMediaSource(RtspMediaSource::Ptr src) {
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_push_src = std::move(src);
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if (_push_src && canRecvRtp()) {
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_push_src->setSdp(_answer_sdp->toRtspSdp());
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}
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}
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}// namespace mediakit
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