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C++からWebRTC (DataChannel)を旋喘する

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v望 卅儲
v望 卅儲

This document discusses using C++ and the WebRTC Native APIs to build real-time communications applications. It provides an overview of the WebRTC architecture and APIs, describes how to set up the native development environment by checking out the WebRTC code from Chromium, and gives examples of using key WebRTC classes like PeerConnection, DataChannel, and observers. It also covers building for different platforms like Linux and macOS and integrating WebRTC with other libraries and protocols.

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C++ WebRTC (DataChannel) / @llamerada_jp WebRTC Meetup Tokyo #20
SE / / GitHub : llamerada-jp facebook : ito.yuuji twitter : @llamerada_jp blog : http://llamerad-jp.hatenablog.com/
/ PROCESSWARP Web /
/ www.oinari.app libvein Server
/ libvein Web / 2 PubSub KVS Seed / Server
Documents / https://webrtc.org/native-code/development/ & API code reading WebRTC ( ) ?https://gist.github.com/szktty/ a47213f0077294c64ea58621c2dcfaf2 android iOS CUI WebRTC https://github.com/llamerada-jp/webrtc-cpp-sample WebRTC https://github.com/llamerada-jp/libwebrtc
Native APIs
WebRTC Native APIs ? WebView / Electron WebSocket iOS / Android (C++ ) WebRTC Native APIs API API
WebRTC API ( ) connect SDP ICE send data disconnect event change status recv data raise error
libuv libuv main Thread rtc::Runnable Thread webrtc:: CreatePeerConnectionFactory rtc::Runnable::Run uv_loop uv_async_send Event!
libvein JavaScritp WebAssembly WebRTC/WebSocket API SDP / ICE (Server / Seed ) libvein core (C++) WebSocket Wrapper WebRTC Wrapper Native WebRTC Browser WebRTC Browser WebSocket WebSocket++ C++ API C API Python JavaScript
https://webrtc.org/native-code/development/ http://commondatastorage.googleapis.com/chrome-infra-docs/?at/depot_tools/ docs/html/depot_tools_tutorial.html
$ cd <workdir> $ git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git $ export PATH=$PATH:`pwd`/depot_tools $ cd <workdir> $ mkdir webrtc-checkout $ cd webrtc-checkout $ fetch --nohooks webrtc $ gclient sync $ gn gen out/Default --args=｀is_debug=false¨ $ ninja -C out/Default
Chrome WebRTC
@ WebRTC Σ( ω )
chromium URL https://omahaproxy.appspot.com/all OS stable branch_commit HASH of chromium
$ cd <workdir> $ git clone https://chromium.googlesource.com/chromium/src chromium $ cd chromium $ git show <HASH of chromium>:DEPS < > ｀src/third_party/webrtc': Var('webrtc_git') + '/src.git' + '@' + ｀784fccbd71c0130b32345d81459b5d0cb07ff6e5', < > HASH of webrtc
< gclient sync > $ git fetch $ git checkout <HASH of webrtc> $ gclient sync $ gn gen out/Default [--args='is_debug=false'] $ ninja -C out/Default [<target>]
@ Linux static library(.a ) .o .a ar 1 .a src/out/Default/obj/examples/peerconnection_client.ninja build <target>: link <obj1> <obj2> ´ build ./peerconnection_client: link obj/examples/ peerconnection_client/conductor.o obj/examples/ peerconnection_client/defaults.o obj/examples/ peerconnection_client/peer_connection_client.o obj/ examples/peerconnection_client/main.o obj/examples/ peerconnection_client/main_wnd.o obj/api/video/ video_frame_i420/i420_buffer.o obj/rtc_base/checks/ checks.o obj/rtc_base/stringutils/string_to_number.o
@ macOS obj/sdk/mac_framework_objc_shared_library/WebRTC obj/api/video_codecs/libbuiltin_video_encoder_factory.a obj/api/video_codecs/libbuiltin_video_decoder_factory.a src/out/Default/obj/sdk/mac_framework_objc_shared_library.ninja src/obj/api/video_codecs/libbuiltin_video_encoder_factory.a src/obj/api/video_codecs/libbuiltin_video_decoder_factory.a
nm https://github.com/llamerada-jp/search- symbol chromium
peerconnection_client https://github.com/llamerada-jp/webrtc-cpp-sample API JavaScript JavaScript C++ API
PeerConnectionObserver class PCO : public webrtc::PeerConnectionObserver { void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state) override; void OnAddStream(rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override; void OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override; // DataChannel ! void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override; void OnRenegotiationNeeded() override; void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state) override; void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state) override; // ICE void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; };
DataChannelObserver class DCO : public webrtc::DataChannelObserver { // void OnStateChange() override; // void OnMessage(const webrtc::DataBuffer& buffer) override; void OnBufferedAmountChange(uint64_t previous_amount) override; };
SessionDescriptionObserver class CSDO : public webrtc::CreateSessionDescriptionObserver { void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; void OnFailure(const std::string& error) override; }; class SSDO : public webrtc::SetSessionDescriptionObserver { void OnSuccess() override; void OnFailure(const std::string& error) override; }; rtc::scoped_refptr<CSDO> csdo (new rtc::RefCountedObject<CSDO>()); rtc::scoped_refptr<SSDO> ssdo (new rtc::RefCountedObject<SSDO>());
& CreatePeerConnectionFactory class CustomRunnable : public rtc::Runnable { public: void Run(rtc::Thread* subthread) override { peer_connection_factory = webrtc::CreatePeerConnectionFactory( nullptr /* network_thread */, nullptr /* worker_thread */, nullptr /* signaling_thread */, nullptr /* default_adm */, webrtc::CreateBuiltinAudioEncoderFactory(), webrtc::CreateBuiltinAudioDecoderFactory(), webrtc::CreateBuiltinVideoEncoderFactory(), webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */, nullptr /* audio_processing */); if (peer_connection_factory.get() == nullptr) { // Error } subthread->Run(); } };
rtc::PhysicalSocketServer socket_server; thread.reset(new rtc::Thread(&socket_server)); rtc::InitializeSSL(); CustomRunnable runnable; thread->Start(&runnable); < peer_connection_factory > peer_connection = peer_connection_factory->CreatePeerConnection(con?guration, nullptr, nullptr, &pco); webrtc::DataChannelInit con?g; // DataChannel data_channel = peer_connection->CreateDataChannel("data_channel", &con?g); data_channel->RegisterObserver(&dco); peer_connection->CreateOffer(csdo, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
std::string message(＾hello!!￣,); webrtc::DataBuffer buffer( rtc::CopyOnWriteBuffer( message.c_str(), message.size()), true); data_channel->Send(buffer);
class DCO : public webrtc::DataChannelObserver { // void OnMessage(const webrtc::DataBuffer& buffer) override { std::cout << std::string(buffer.data.data<char>(), buffer.data.size()) << std::endl; } };
// Close peer_connection->Close(); peer_connection = nullptr; data_channel = nullptr; peer_connection_factory = nullptr; thread->Quit(); thread.reset(); rtc::CleanupSSL();
macOS : WEBRTC_MAC=1 WEBRTC_POSIX=1 Linux : WEBRTC_LINUX=1 WEBRTC_POSIX=1 clang (libc++ ) -stdlib=libc++ C++ RTTI -fno-rtti typeinfo dynamic_cast --start-group --end-group
$ clang++ -I<workdir/webrtc-checkout/src> -I<workdir/webrtc-checkout/src/ third_party/abseil-cpp> -DWEBRTC_MAC=1 -DWEBRTC_POSIX=1 -std=c++11 -W -Wall - Wno-unused-parameter -O0 -g -fno-rtti -o <output> -o <source> (github ) build ./build.sh $ cd <path to build> $ make clean $ make VERBOSE=1
OpenSSL BoringSSL API HTTP WebSocket BoringSSL WebRTC Third party src/third_party
WebRTC BSD src/third_party OSS WebRTC
WebRTC C++

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C++からWebRTC (DataChannel)を旋喘する