提取webrtc推流、播放代码为单独的派生类

2026-06-13 03:31:45 +08:00 · 2021-10-15 16:27:17 +08:00
parent 8531b5e1cb
commit 7f3f47abbb
7 changed files with 379 additions and 229 deletions
--- a/webrtc/WebRtcPusher.cpp
+++ b/webrtc/WebRtcPusher.cpp
@@ -0,0 +1,131 @@
+/*
+ * 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 "WebRtcPusher.h"
+
+WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
+                                       const RtspMediaSource::Ptr &src,
+                                       const MediaInfo &info) {
+    WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, info), [](WebRtcPusher *ptr) {
+        ptr->onDestory();
+        delete ptr;
+    });
+    ret->onCreate();
+    return ret;
+}
+
+WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
+                           const RtspMediaSource::Ptr &src,
+                           const MediaInfo &info) : WebRtcTransportImp(poller) {
+    InfoL << this;
+    _media_info = info;
+    _push_src = src;
+    CHECK(_push_src);
+}
+
+bool WebRtcPusher::close(MediaSource &sender, bool force) {
+    //此回调在其他线程触发
+    if (!force && totalReaderCount(sender)) {
+        return false;
+    }
+    string err = StrPrinter << "close media:" << sender.getSchema() << "/" << sender.getVhost() << "/"
+                            << sender.getApp() << "/" << sender.getId() << " " << force;
+    weak_ptr<WebRtcPusher> weak_self = static_pointer_cast<WebRtcPusher>(shared_from_this());
+    getPoller()->async([weak_self, err]() {
+        auto strong_self = weak_self.lock();
+        if (strong_self) {
+            strong_self->onShutdown(SockException(Err_shutdown, err));
+        }
+    });
+    return true;
+}
+
+int WebRtcPusher::totalReaderCount(MediaSource &sender) {
+    auto total_count = 0;
+    for (auto &src : _push_src_simulcast) {
+        total_count += src.second->totalReaderCount();
+    }
+    return total_count + _push_src->totalReaderCount();
+}
+
+MediaOriginType WebRtcPusher::getOriginType(MediaSource &sender) const {
+    return MediaOriginType::rtc_push;
+}
+
+string WebRtcPusher::getOriginUrl(MediaSource &sender) const {
+    return _media_info._full_url;
+}
+
+std::shared_ptr<SockInfo> WebRtcPusher::getOriginSock(MediaSource &sender) const {
+    return static_pointer_cast<SockInfo>(getSession());
+}
+
+void WebRtcPusher::onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
+    if (!_simulcast) {
+        assert(_push_src);
+        _push_src->onWrite(rtp, false);
+        return;
+    }
+
+    if (rtp->type == TrackAudio) {
+        //音频
+        for (auto &pr : _push_src_simulcast) {
+            pr.second->onWrite(rtp, false);
+        }
+    } else {
+        //视频
+        auto &src = _push_src_simulcast[rid];
+        if (!src) {
+            auto stream_id = rid.empty() ? _push_src->getId() : _push_src->getId() + "_" + rid;
+            auto src_imp = std::make_shared<RtspMediaSourceImp>(_push_src->getVhost(), _push_src->getApp(), stream_id);
+            src_imp->setSdp(_push_src->getSdp());
+            src_imp->setProtocolTranslation(_push_src->isRecording(Recorder::type_hls),
+                                            _push_src->isRecording(Recorder::type_mp4));
+            src_imp->setListener(static_pointer_cast<WebRtcPusher>(shared_from_this()));
+            src = src_imp;
+        }
+        src->onWrite(std::move(rtp), false);
+    }
+}
+
+void WebRtcPusher::onStartWebRTC() {
+    WebRtcTransportImp::onStartWebRTC();
+    _simulcast = _answer_sdp->supportSimulcast();
+    if (canRecvRtp()) {
+        _push_src->setSdp(_answer_sdp->toRtspSdp());
+    }
+}
+
+void WebRtcPusher::onDestory() {
+    WebRtcTransportImp::onDestory();
+
+    auto duration = getDuration();
+    auto bytes_usage = getBytesUsage();
+    //流量统计事件广播
+    GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
+
+    if (getSession()) {
+        WarnL << "RTC推流器("
+              << _media_info._vhost << "/"
+              << _media_info._app << "/"
+              << _media_info._streamid
+              << ")结束推流,耗时(s):" << duration;
+        if (bytes_usage >= iFlowThreshold * 1024) {
+            NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration,
+                                               false, static_cast<SockInfo &>(*getSession()));
+        }
+    }
+}
+
+void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
+    WebRtcTransportImp::onRtcConfigure(configure);
+    //这只是推流
+    configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
+}