mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
修正代码拼写错误
This commit is contained in:
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68ea5465cc
commit
a12b7c8021
@ -18,8 +18,8 @@ namespace mediakit {
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WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
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WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const RtspMediaSource::Ptr &src,
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const MediaInfo &info,
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const MediaInfo &info,
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bool perferred_tcp) {
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bool preferred_tcp) {
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WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, perferred_tcp), [](WebRtcPlayer *ptr) {
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WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
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ptr->onDestory();
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ptr->onDestory();
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delete ptr;
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delete ptr;
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});
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});
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@ -30,7 +30,7 @@ WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
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WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
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WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const RtspMediaSource::Ptr &src,
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const MediaInfo &info,
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const MediaInfo &info,
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bool perferred_tcp) : WebRtcTransportImp(poller,perferred_tcp) {
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bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
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_media_info = info;
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_media_info = info;
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_play_src = src;
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_play_src = src;
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CHECK(_play_src);
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CHECK(_play_src);
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@ -20,7 +20,7 @@ class WebRtcPlayer : public WebRtcTransportImp {
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public:
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public:
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using Ptr = std::shared_ptr<WebRtcPlayer>;
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using Ptr = std::shared_ptr<WebRtcPlayer>;
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~WebRtcPlayer() override = default;
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~WebRtcPlayer() override = default;
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static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool perferred_tcp = false);
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static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp = false);
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protected:
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protected:
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///////WebRtcTransportImp override///////
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///////WebRtcTransportImp override///////
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@ -30,7 +30,7 @@ protected:
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void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
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void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
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private:
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private:
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WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool perferred_tcp);
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WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp);
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private:
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private:
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//媒体相关元数据
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//媒体相关元数据
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@ -20,8 +20,8 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
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const std::shared_ptr<void> &ownership,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const MediaInfo &info,
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const ProtocolOption &option,
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const ProtocolOption &option,
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bool perferred_tcp) {
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bool preferred_tcp) {
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WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,perferred_tcp), [](WebRtcPusher *ptr) {
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WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,preferred_tcp), [](WebRtcPusher *ptr) {
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ptr->onDestory();
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ptr->onDestory();
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delete ptr;
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delete ptr;
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});
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});
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@ -34,7 +34,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
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const std::shared_ptr<void> &ownership,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const MediaInfo &info,
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const ProtocolOption &option,
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const ProtocolOption &option,
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bool perferred_tcp) : WebRtcTransportImp(poller,perferred_tcp) {
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bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
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_media_info = info;
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_media_info = info;
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_push_src = src;
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_push_src = src;
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_push_src_ownership = ownership;
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_push_src_ownership = ownership;
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@ -21,7 +21,7 @@ public:
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using Ptr = std::shared_ptr<WebRtcPusher>;
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using Ptr = std::shared_ptr<WebRtcPusher>;
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~WebRtcPusher() override = default;
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~WebRtcPusher() override = default;
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static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
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static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
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const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool perferred_tcp = false);
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const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp = false);
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protected:
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protected:
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///////WebRtcTransportImp override///////
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///////WebRtcTransportImp override///////
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@ -52,7 +52,7 @@ protected:
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private:
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private:
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WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
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WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSourceImp::Ptr &src,
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const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool perferred_tcp);
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const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp);
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private:
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private:
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bool _simulcast = false;
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bool _simulcast = false;
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@ -403,8 +403,8 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
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#endif
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#endif
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}
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}
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WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool perferred_tcp)
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WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp)
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: WebRtcTransport(poller), _perferred_tcp(perferred_tcp) {
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: WebRtcTransport(poller), _preferred_tcp(preferred_tcp) {
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InfoL << getIdentifier();
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InfoL << getIdentifier();
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}
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}
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@ -630,13 +630,13 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
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if (extern_ips.empty()) {
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if (extern_ips.empty()) {
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std::string local_ip = SockUtil::get_local_ip();
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std::string local_ip = SockUtil::get_local_ip();
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if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
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if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
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if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _perferred_tcp ? 125 : 115, "tcp")); }
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if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
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} else {
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} else {
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const uint32_t delta = 10;
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const uint32_t delta = 10;
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uint32_t priority = 100 + delta * extern_ips.size();
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uint32_t priority = 100 + delta * extern_ips.size();
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for (auto ip : extern_ips) {
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for (auto ip : extern_ips) {
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if (local_udp_port) { configure.addCandidate(*makeIceCandidate(ip, local_udp_port, priority, "udp")); }
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if (local_udp_port) { configure.addCandidate(*makeIceCandidate(ip, local_udp_port, priority, "udp")); }
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if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority - (_perferred_tcp ? -5 : 5), "tcp")); }
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if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(ip, local_tcp_port, priority - (_preferred_tcp ? -5 : 5), "tcp")); }
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priority -= delta;
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priority -= delta;
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}
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}
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}
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}
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@ -1154,9 +1154,9 @@ void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
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void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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MediaInfo info(args["url"]);
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MediaInfo info(args["url"]);
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bool perferred_tcp = args["perferred_tcp"];
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bool preferred_tcp = args["preferred_tcp"];
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Broadcast::PublishAuthInvoker invoker = [cb, info, perferred_tcp](const string &err, const ProtocolOption &option) mutable {
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Broadcast::PublishAuthInvoker invoker = [cb, info, preferred_tcp](const string &err, const ProtocolOption &option) mutable {
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if (!err.empty()) {
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if (!err.empty()) {
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cb(WebRtcException(SockException(Err_other, err)));
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cb(WebRtcException(SockException(Err_other, err)));
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return;
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return;
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@ -1195,7 +1195,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
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push_src_ownership = push_src->getOwnership();
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push_src_ownership = push_src->getOwnership();
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push_src->setProtocolOption(option);
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push_src->setProtocolOption(option);
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}
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}
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auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, perferred_tcp);
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auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, preferred_tcp);
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push_src->setListener(rtc);
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push_src->setListener(rtc);
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cb(*rtc);
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cb(*rtc);
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};
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};
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@ -1210,10 +1210,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
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void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
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MediaInfo info(args["url"]);
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MediaInfo info(args["url"]);
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bool perferred_tcp = args["perferred_tcp"];
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bool preferred_tcp = args["preferred_tcp"];
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auto session_ptr = sender.shared_from_this();
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auto session_ptr = sender.shared_from_this();
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Broadcast::AuthInvoker invoker = [cb, info, session_ptr, perferred_tcp](const string &err) mutable {
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Broadcast::AuthInvoker invoker = [cb, info, session_ptr, preferred_tcp](const string &err) mutable {
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if (!err.empty()) {
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if (!err.empty()) {
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cb(WebRtcException(SockException(Err_other, err)));
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cb(WebRtcException(SockException(Err_other, err)));
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return;
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return;
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@ -1229,7 +1229,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
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}
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}
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// 还原成rtc,目的是为了hook时识别哪种播放协议
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// 还原成rtc,目的是为了hook时识别哪种播放协议
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info._schema = RTC_SCHEMA;
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info._schema = RTC_SCHEMA;
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auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, perferred_tcp);
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auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, preferred_tcp);
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cb(*rtc);
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cb(*rtc);
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});
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});
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};
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};
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@ -249,7 +249,7 @@ public:
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void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
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void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
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protected:
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protected:
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WebRtcTransportImp(const EventPoller::Ptr &poller,bool perferred_tcp = false);
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WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
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void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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void onStartWebRTC() override;
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void onStartWebRTC() override;
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void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
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void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
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@ -279,7 +279,7 @@ private:
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void onCheckAnswer(RtcSession &sdp);
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void onCheckAnswer(RtcSession &sdp);
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private:
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private:
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bool _perferred_tcp;
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bool _preferred_tcp;
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uint16_t _rtx_seq[2] = {0, 0};
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uint16_t _rtx_seq[2] = {0, 0};
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//用掉的总流量
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//用掉的总流量
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uint64_t _bytes_usage = 0;
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uint64_t _bytes_usage = 0;
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