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c sdk 新增webrtc相关函数 (#4473)

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另外调整函数位置，whip、whep请求设置Content-Type为application/sdp
This commit is contained in:
Lidaofu
2025-09-24 17:40:43 +08:00
committed by GitHub
parent a3eb07adfc
commit 493714bc7d
6 changed files with 355 additions and 57 deletions
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77
api/source/mk_common.cpp
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@@ -29,6 +29,7 @@ using namespace mediakit;
static TcpServer::Ptr rtsp_server[2];
static TcpServer::Ptr rtmp_server[2];
static TcpServer::Ptr http_server[2];
static TcpServer::Ptr signaling_server[2];
static TcpServer::Ptr shell_server;
#ifdef ENABLE_RTPPROXY
@@ -37,9 +38,14 @@ static RtpServer::Ptr rtpServer;
#endif
#ifdef ENABLE_WEBRTC
#include "../webrtc/WebRtcSession.h"
#include "webrtc/WebRtcSession.h"
#include "webrtc/IceSession.hpp"
#include "webrtc/WebRtcSignalingSession.h"
#include "webrtc/WebRtcTransport.h"
static UdpServer::Ptr rtcServer_udp;
static TcpServer::Ptr rtcServer_tcp;
static UdpServer::Ptr iceServer_udp;
static TcpServer::Ptr iceServer_tcp;
#endif
#if defined(ENABLE_SRT)
@@ -288,46 +294,45 @@ API_EXPORT uint16_t API_CALL mk_rtc_server_start(uint16_t port) {
#endif
}
#ifdef ENABLE_WEBRTC
class WebRtcArgsUrl : public mediakit::WebRtcArgs {
public:
WebRtcArgsUrl(std::string url) { _url = std::move(url); }
toolkit::variant operator[](const std::string &key) const override {
if (key == "url") {
return _url;
API_EXPORT uint16_t API_CALL mk_signaling_server_start(uint16_t port, int ssl) {
#ifdef ENABLE_WEBRTC
ssl = MAX(0, MIN(ssl, 1));
try {
signaling_server[ssl] = std::make_shared<TcpServer>();
if (ssl) {
signaling_server[ssl]->start<WebRtcWebcosktSignalSslSession>(port);
} else {
signaling_server[ssl]->start<WebRtcWebcosktSignalingSession>(port);
}
return "";
return signaling_server[ssl]->getPort();
} catch (std::exception &ex) {
signaling_server[ssl] = nullptr;
WarnL << ex.what();
return 0;
}
private:
std::string _url;
};
#endif
API_EXPORT void API_CALL mk_webrtc_get_answer_sdp(void *user_data, on_mk_webrtc_get_answer_sdp cb, const char *type,
const char *offer, const char *url) {
mk_webrtc_get_answer_sdp2(user_data, nullptr, cb, type, offer, url);
}
API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data_free user_data_free, on_mk_webrtc_get_answer_sdp cb, const char *type,
const char *offer, const char *url) {
#ifdef ENABLE_WEBRTC
assert(type && offer && url && cb);
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 *) {});
auto args = std::make_shared<WebRtcArgsUrl>(url);
WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
auto &handler = const_cast<WebRtcInterface &>(exchanger);
try {
auto sdp_answer = handler.getAnswerSdp(offer_str);
cb(ptr.get(), sdp_answer.data(), nullptr);
} catch (std::exception &ex) {
cb(ptr.get(), nullptr, ex.what());
}
});
#else
WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
return 0;
#endif
}
API_EXPORT uint16_t API_CALL mk_ice_server_start(uint16_t port){
#ifdef ENABLE_WEBRTC
try {
iceServer_tcp = std::make_shared<TcpServer>();
iceServer_udp = std::make_shared<UdpServer>();
iceServer_udp->start<IceSession>(port);
iceServer_tcp->start<IceSession>(port);
} catch (std::exception &ex) {
iceServer_udp = nullptr;
iceServer_tcp = nullptr;
WarnL << ex.what();
return 0;
}
#else
WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
return 0;
#endif
}