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author | Nicolas Werner <nicolas.werner@hotmail.de> | 2020-08-14 01:03:27 +0200 |
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committer | Nicolas Werner <nicolas.werner@hotmail.de> | 2020-08-14 01:03:27 +0200 |
commit | f157602a52278519370c45cde4c9a0b12ede8d7e (patch) | |
tree | 26aaa10a71a1f44117605d4248c670a359427bb5 /src/WebRTCSession.cpp | |
parent | Bump flatpak dependency (diff) | |
download | nheko-f157602a52278519370c45cde4c9a0b12ede8d7e.tar.xz |
Disable call support, when GStreamer is unavailable
Integrating that in our CI is currently a bit hard, so disable it for now, if GStreamer isn't found. Just make sure to build against GStreamer for call support!
Diffstat (limited to 'src/WebRTCSession.cpp')
-rw-r--r-- | src/WebRTCSession.cpp | 68 |
1 files changed, 67 insertions, 1 deletions
diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp index 5638c607..dd1dd61f 100644 --- a/src/WebRTCSession.cpp +++ b/src/WebRTCSession.cpp @@ -3,6 +3,7 @@ #include "Logging.h" #include "WebRTCSession.h" +#ifdef GSTREAMER_AVAILABLE extern "C" { #include "gst/gst.h" @@ -11,6 +12,7 @@ extern "C" #define GST_USE_UNSTABLE_API #include "gst/webrtc/webrtc.h" } +#endif Q_DECLARE_METATYPE(WebRTCSession::State) @@ -24,6 +26,7 @@ WebRTCSession::WebRTCSession() bool WebRTCSession::init(std::string *errorMessage) { +#ifdef GSTREAMER_AVAILABLE if (initialised_) return true; @@ -81,10 +84,14 @@ WebRTCSession::init(std::string *errorMessage) *errorMessage = strError; } return initialised_; +#else + (void)errorMessage; + return false; +#endif } +#ifdef GSTREAMER_AVAILABLE namespace { - bool isoffering_; std::string localsdp_; std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_; @@ -631,3 +638,62 @@ WebRTCSession::getAudioSourceNames(const std::string &defaultDevice) } return ret; } +#else + +bool +WebRTCSession::createOffer() +{ + return false; +} + +bool +WebRTCSession::acceptOffer(const std::string &) +{ + return false; +} + +bool +WebRTCSession::acceptAnswer(const std::string &) +{ + return false; +} + +void +WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &) +{} + +bool +WebRTCSession::startPipeline(int) +{ + return false; +} + +bool +WebRTCSession::createPipeline(int) +{ + return false; +} + +bool +WebRTCSession::toggleMuteAudioSrc(bool &) +{ + return false; +} + +void +WebRTCSession::end() +{ +} + +void +WebRTCSession::refreshDevices() +{ +} + +std::vector<std::string> +WebRTCSession::getAudioSourceNames(const std::string &) +{ + return {}; +} + +#endif |