feat: use prebuilt webrtc and add comaptibility with the newest webrtc

endpoint/stream/CMakeLists.txt

@@ -1,6 +1,7 @@
 add_subdirectory(stream)
 
 option(PRIVMX_BUILD_WITH_WEBRTC "Build Stream module with additional layer of the native WebRTC library" OFF)
+option(USE_PREBUILT_WEBRTC "Use libwebrtc provided by package manager" ON)
 message(STATUS "Build Stream module with additional layer of the native WebRTC library - PRIVMX_BUILD_WITH_WEBRTC=${PRIVMX_BUILD_WITH_WEBRTC}")
 
 set(LIBS "")

endpoint/stream/webrtc/CMakeLists.txt

@@ -2,11 +2,16 @@ file(GLOB_RECURSE SOURCES ${CMAKE_CURRENT_SOURCE_DIR}/src/*.cpp)
 set(INCLUDE_DIRS ${CMAKE_CURRENT_SOURCE_DIR}/include)
 set(INCLUDE_PUB_DIRS ${CMAKE_CURRENT_SOURCE_DIR}/include_pub)
 
-add_library(libwebrtc SHARED IMPORTED)
-set_target_properties(libwebrtc PROPERTIES
-    IMPORTED_LOCATION "<path_to_libwebrtc>"
-    INTERFACE_INCLUDE_DIRECTORIES "<path_to_libwebrtc>"
-)
+if(USE_PREBUILT_WEBRTC)
+    find_package(libwebrtc REQUIRED CONFIG)
+    add_library(libwebrtc ALIAS libwebrtc::libwebrtc)
+else()
+    add_library(libwebrtc SHARED IMPORTED)
+    set_target_properties(libwebrtc PROPERTIES
+        IMPORTED_LOCATION "<path_to_libwebrtc>"
+        INTERFACE_INCLUDE_DIRECTORIES "<path_to_libwebrtc>"
+    )
+endif()
 
 add_library(privmxendpointstream_webrtc OBJECT ${SOURCES})
 target_include_directories(privmxendpointstream_webrtc PUBLIC ${INCLUDE_DIRS} PUBLIC ${INCLUDE_PUB_DIRS})

endpoint/stream/webrtc/include/privmx/endpoint/stream/PeerConnectionManager.hpp

@@ -36,6 +36,7 @@ enum ConnectionType {
 struct AudioTrackInfo {
     libwebrtc::scoped_refptr<libwebrtc::RTCAudioTrack> track;
     libwebrtc::scoped_refptr<libwebrtc::RTCRtpSender> sender;
+    std::shared_ptr<privmx::webrtc::AudioLevelAnalyzer> audioLevelAnalyzer;
     std::shared_ptr<privmx::webrtc::FrameCryptor> frameCryptor;
 };
 

endpoint/stream/webrtc/include/privmx/endpoint/stream/PmxPeerConnectionObserver.hpp

@@ -43,7 +43,7 @@ class PmxPeerConnectionObserver : public libwebrtc::RTCPeerConnectionObserver {
     void OnIceGatheringState(libwebrtc::RTCIceGatheringState state) override;
     void OnIceConnectionState(libwebrtc::RTCIceConnectionState state) override;
     void OnIceCandidate(libwebrtc::scoped_refptr<libwebrtc::RTCIceCandidate> candidate) override;
-    void OnAddStream(libwebrtc::scoped_refptr<libwebrtc::RTCMediaStream> stream) override;
+    void OnAddStream([[maybe_unused]] libwebrtc::scoped_refptr<libwebrtc::RTCMediaStream> stream) override;
     void OnRemoveStream(libwebrtc::scoped_refptr<libwebrtc::RTCMediaStream> stream) override;
     void OnDataChannel(libwebrtc::scoped_refptr<libwebrtc::RTCDataChannel> data_channel) override;
     void OnRenegotiationNeeded() override;
@@ -70,6 +70,7 @@ class PmxPeerConnectionObserver : public libwebrtc::RTCPeerConnectionObserver {
     privmx::utils::ThreadSaveMap<std::string, std::shared_ptr<RTCVideoRendererImpl>> _RTCVideoRenderers;
     privmx::utils::ThreadSaveMap<std::string, std::shared_ptr<AudioTrackSinkImpl>> _audioTrackSinks;
     privmx::utils::ThreadSaveMap<std::string, std::shared_ptr<DataChannelImpl>> _dataChannels;
+    privmx::utils::ThreadSaveMap<std::string, std::shared_ptr<privmx::webrtc::AudioLevelAnalyzer>> _audioLevelAnalyzers;
     privmx::utils::ThreadSaveMap<std::string, std::shared_ptr<privmx::webrtc::FrameCryptor>> _frameCryptors;
 
     std::optional<std::function<void(libwebrtc::scoped_refptr<libwebrtc::RTCIceCandidate>)>> _onIceCandidate;

endpoint/stream/webrtc/src/PmxPeerConnectionObserver.cpp

@@ -78,7 +78,7 @@ void PmxPeerConnectionObserver::OnIceCandidate([[maybe_unused]] libwebrtc::scope
     LOG_DEBUG("STREAMS ", "API ", _streamRoomId + ": ON ICE CANDIDATE")
     if(_onIceCandidate.has_value()) _onIceCandidate.value()(candidate);
 }
-void PmxPeerConnectionObserver::OnAddStream(libwebrtc::scoped_refptr<libwebrtc::RTCMediaStream> stream) {
+void PmxPeerConnectionObserver::OnAddStream([[maybe_unused]] libwebrtc::scoped_refptr<libwebrtc::RTCMediaStream> stream) {
     LOG_DEBUG("STREAMS ", "API ", _streamRoomId + ": STREAM ADDED")
     LOG_DEBUG("STREAMS ", "API ", "stream->video_tracks().size() -> " + std::to_string(stream->video_tracks().size()))
     LOG_DEBUG("STREAMS ", "API ", "stream->audio_tracks().size() -> " + std::to_string(stream->audio_tracks().size()))
@@ -107,9 +107,21 @@ void PmxPeerConnectionObserver::OnTrack([[maybe_unused]] libwebrtc::scoped_refpt
 void PmxPeerConnectionObserver::OnAddTrack([[maybe_unused]] libwebrtc::vector<libwebrtc::scoped_refptr<libwebrtc::RTCMediaStream>> streams, [[maybe_unused]] libwebrtc::scoped_refptr<libwebrtc::RTCRtpReceiver> receiver) {
     LOG_DEBUG("STREAMS ", "API ", _streamRoomId + ": ON ADD TRACK")    
     // set frame crypto to decrypt track
+    auto audioLevelAnalyzer = receiver->track()->kind().std_string() == "audio"
+        ? privmx::webrtc::FrameCryptorFactory::audioLevelAnalyzer()
+        : nullptr;
+    if (audioLevelAnalyzer) {
+        _audioLevelAnalyzers.set(receiver->track()->id().std_string(), audioLevelAnalyzer);
+    }
     _frameCryptors.set(
         receiver->track()->id().std_string(), 
-        privmx::webrtc::FrameCryptorFactory::frameCryptorFromRtpReceiver(_peerConnectionFactory ,receiver, _currentKeys, _options)
+        privmx::webrtc::FrameCryptorFactory::frameCryptorFromRtpReceiver(
+            _peerConnectionFactory,
+            receiver,
+            _currentKeys,
+            audioLevelAnalyzer,
+            _options
+        )
     );
 
     DataType dataType = receiver->track()->kind().std_string() == "video" ? DataType::VIDEO : DataType::AUDIO;
@@ -171,6 +183,8 @@ void PmxPeerConnectionObserver::OnRemoveTrack([[maybe_unused]] libwebrtc::scoped
             tmp.value()->OnRemoteTrack(Track{dataType, streamIds, track->id().std_string(), !track->enabled(), [track](bool mute) {return track->set_enabled(!mute);}}, TrackAction::ADDED);
         }
     }
+    if (dataType == DataType::AUDIO) _audioLevelAnalyzers.erase(receiver->track()->id().std_string());
+    _frameCryptors.erase(receiver->track()->id().std_string());
     if(dataType == DataType::AUDIO) _audioTrackSinks.erase(receiver->track()->id().std_string());
     if(dataType == DataType::VIDEO) _RTCVideoRenderers.erase(receiver->track()->id().std_string());
 }

endpoint/stream/webrtc/src/WebRTCImpl.cpp

@@ -280,13 +280,15 @@ void WebRTCImpl::updatePeerConnectionWithLocalStream(
 void WebRTCImpl::AddAudioTrack(std::shared_ptr<privmx::endpoint::stream::JanusConnection> jc, libwebrtc::scoped_refptr<libwebrtc::RTCAudioTrack> audioTrack, std::string id) {
     jc->peerConnection->mediaStream->AddTrack(audioTrack);
     auto sender = jc->peerConnection->pc->AddTrack(audioTrack, libwebrtc::vector<libwebrtc::string>{std::vector<libwebrtc::string>{jc->peerConnection->mediaStream->id()}});
+    auto audioLevelAnalyzer = privmx::webrtc::FrameCryptorFactory::audioLevelAnalyzer();
     std::shared_ptr<privmx::webrtc::FrameCryptor> frameCryptor;
     {
         std::shared_lock<std::shared_mutex> lock(jc->peerConnection->trackMutex);
         frameCryptor = privmx::webrtc::FrameCryptorFactory::frameCryptorFromRtpSender(
             _peerConnectionFactory,
             sender, 
             jc->peerConnection->keys,
+            audioLevelAnalyzer,
             _frameCryptorOptions
         );
     }
@@ -297,6 +299,7 @@ void WebRTCImpl::AddAudioTrack(std::shared_ptr<privmx::endpoint::stream::JanusCo
             AudioTrackInfo{
                 .track=audioTrack, 
                 .sender=sender, 
+                .audioLevelAnalyzer=audioLevelAnalyzer,
                 .frameCryptor=frameCryptor
             }
         ));
@@ -313,6 +316,7 @@ void WebRTCImpl::AddVideoTrack(std::shared_ptr<privmx::endpoint::stream::JanusCo
             _peerConnectionFactory,
             sender, 
             jc->peerConnection->keys,
+            nullptr,
             _frameCryptorOptions
         );
     }