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ZLMediaKit
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webrtc新增stop软插件，支持http信令调用

This commit is contained in:
Dw9 2023-03-03 21:50:22 +08:00
parent 28f8c1ffec
commit 9b273a33ba
4 changed files with 76 additions and 30 deletions
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2
api/source/mk_common.cpp
View File

@ -305,7 +305,7 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
auto session = std::make_shared<HttpSession>(Socket::createSocket());
std::string offer_str = offer;
std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
WebRtcPluginManager::Instance().getAnswerSdp(*session, type, WebRtcArgsUrl(url),
WebRtcPluginManager::Instance().handleRtcPlugin(*session, type, WebRtcArgsUrl(url),
[offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
try {
auto sdp_answer = const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer_str);

22
server/WebApi.cpp
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@ -1591,10 +1591,28 @@ void installWebApi() {
api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){
CHECK_ARGS("type");
auto type = allArgs["type"];
if(type == "stop") {
WebRtcPluginManager::Instance().handleRtcPlugin(
*(static_cast<Session *>(&sender)), type, WebRtcArgsImp(allArgs, sender.getIdentifier()),
[invoker, val, headerOut](const WebRtcInterface &exchanger) mutable {
try {
std::string tmp = "";
const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(tmp);
} catch (std::exception &ex) {
val["code"] = const_cast<WebRtcInterface &>(exchanger).getErrcode();
val["msg"] = ex.what();
invoker(200, headerOut, val.toStyledString());
}
});
}else{
auto offer = allArgs.getArgs();
CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
WebRtcPluginManager::Instance().getAnswerSdp(*(static_cast<Session *>(&sender)), type,
WebRtcPluginManager::Instance().handleRtcPlugin(*(static_cast<Session *>(&sender)), type,
WebRtcArgsImp(allArgs, sender.getIdentifier()),
[invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
//设置返回类型
@ -1613,7 +1631,7 @@ void installWebApi() {
invoker(200, headerOut, val.toStyledString());
}
});
});
}});
#endif
#if defined(ENABLE_VERSION)

29
webrtc/WebRtcTransport.cpp
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@ -1130,7 +1130,7 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
_map_creator[type] = std::move(cb);
}
void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb) {
void WebRtcPluginManager::handleRtcPlugin(Session &sender, const string &type, const WebRtcArgs &args, const onRtcEvent &cb) {
lock_guard<mutex> lck(_mtx_creator);
auto it = _map_creator.find(type);
if (it == _map_creator.end()) {
@ -1140,11 +1140,31 @@ void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, cons
it->second(sender, args, cb);
}
void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onRtcEvent &cb) {
cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void stop_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onRtcEvent &cb) {
MediaInfo info(args["url"]);
std::string id = args["id"];
bool push_exist = (bool)MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid);
if(!push_exist){
cb(WebRtcException(SockException(Err_other, "cannot find media source")));
return;
}
auto rtc = WebRtcTransportManager::Instance().getItem(id);
if(!rtc){
cb(WebRtcException(SockException(Err_other, "cannot find rtc transport")));
return;
}
rtc->onShutdown(SockException(Err_shutdown, "http(signal) close notify received"));
cb(WebRtcException(SockException(Err_success, "success")));
}
void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onRtcEvent &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
@ -1200,7 +1220,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
}
}
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onRtcEvent &cb) {
MediaInfo info(args["url"]);
bool preferred_tcp = args["preferred_tcp"];
@ -1238,6 +1258,7 @@ static onceToken s_rtc_auto_register([]() {
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
WebRtcPluginManager::Instance().registerPlugin("stop", stop_plugin);
});
}// namespace mediakit

15
webrtc/WebRtcTransport.h
View File

@ -40,6 +40,7 @@ public:
WebRtcInterface() = default;
virtual ~WebRtcInterface() = default;
virtual std::string getAnswerSdp(const std::string &offer) = 0;
virtual int getErrcode(){return 0;}
virtual const std::string &getIdentifier() const = 0;
};
@ -50,6 +51,11 @@ public:
std::string getAnswerSdp(const std::string &offer) override {
throw _ex;
}
int getErrcode() override {
return _ex.getErrCode();
}
const std::string &getIdentifier() const override {
static std::string s_null;
return s_null;
@ -246,6 +252,8 @@ public:
void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
void removeTuple(RTC::TransportTuple* tuple);
void onShutdown(const SockException &ex) override;
protected:
void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
@ -261,7 +269,6 @@ protected:
void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};
void onCreate() override;
void onDestory() override;
void onShutdown(const SockException &ex) override;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
void updateTicker();
float getLossRate(TrackType type);
@ -325,13 +332,13 @@ public:
class WebRtcPluginManager {
public:
using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
using onRtcEvent = std::function<void(const WebRtcInterface &rtc)>;
using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onRtcEvent &cb)>;
static WebRtcPluginManager &Instance();
void registerPlugin(const std::string &type, Plugin cb);
void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
void handleRtcPlugin(Session &sender, const std::string &type, const WebRtcArgs &args, const onRtcEvent &cb);
private:
WebRtcPluginManager() = default;