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初步支持语音双向对讲webrtc插件

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This commit is contained in:
xia-chu
2025-11-25 22:26:20 +08:00
parent 5efe843595
commit 5165ac4f74
5 changed files with 245 additions and 13 deletions
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8
webrtc/WebRtcTransport.cpp
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@@ -28,6 +28,7 @@
#include "WebRtcEchoTest.h"
#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
#include "WebRtcTalk.h"
#include "Rtsp/RtspMediaSourceImp.h"
#define RTP_SSRC_OFFSET 1
@@ -1726,6 +1727,7 @@ void push_plugin(SocketHelper& sender, const WebRtcArgs &args, const onCreateWeb
}
}
template<typename Type>
void play_plugin(SocketHelper &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
MediaInfo info(args["url"]);
@@ -1748,7 +1750,7 @@ void play_plugin(SocketHelper &sender, const WebRtcArgs &args, const onCreateWeb
// 还原成rtc，目的是为了hook时识别哪种播放协议 [AUTO-TRANSLATED:fe8dd2dc]
// Restore to RTC, the purpose is to identify which playback protocol during hooking
info.schema = "rtc";
auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info,
auto rtc = Type::create(EventPollerPool::Instance().getPoller(), src, info,
WebRtcTransport::Role::PEER, WebRtcTransport::SignalingProtocols::WHEP_WHIP);
cb(*rtc);
});
@@ -1831,7 +1833,9 @@ static onceToken s_rtc_auto_register([]() {
WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
#endif
WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
WebRtcPluginManager::Instance().registerPlugin("play", play_plugin<WebRtcPlayer>);
WebRtcPluginManager::Instance().registerPlugin("talk", play_plugin<WebRtcTalk>);
WebRtcPluginManager::Instance().setListener([](SocketHelper& sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
});