Require GStreamer 1.18 for voip support

CMakeLists.txt

@@ -446,11 +446,11 @@ else()
 endif()
 
 include(FindPkgConfig)
-pkg_check_modules(GSTREAMER IMPORTED_TARGET gstreamer-sdp-1.0>=1.16 gstreamer-webrtc-1.0>=1.16)
+pkg_check_modules(GSTREAMER IMPORTED_TARGET gstreamer-sdp-1.0>=1.18 gstreamer-webrtc-1.0>=1.18)
 if (TARGET PkgConfig::GSTREAMER)
 	add_feature_info(voip ON "GStreamer found. Call support is enabled automatically.")
 else()
-	add_feature_info(voip OFF "GStreamer could not be found on your system.  As a consequence call support has been disabled. If you don't want that, make sure gstreamer-sdp-1.0>=1.16 gstreamer-webrtc-1.0>=1.16 can be found via pkgconfig.")
+	add_feature_info(voip OFF "GStreamer could not be found on your system.  As a consequence call support has been disabled. If you don't want that, make sure gstreamer-sdp-1.0>=1.18 gstreamer-webrtc-1.0>=1.18 can be found via pkgconfig.")
 endif()
 
 # single instance functionality

resources/qml/MessageInput.qml

@@ -44,7 +44,6 @@ Rectangle {
                     } else if (CallManager.isOnCall) {
                         CallManager.hangUp();
                     } else {
-                        CallManager.refreshDevices();
                         var dialog = placeCallDialog.createObject(timelineRoot);
                         dialog.open();
                     }

resources/qml/voip/CallInviteBar.qml

@@ -75,7 +75,6 @@ Rectangle {
             ToolTip.visible: hovered
             ToolTip.text: qsTr("Devices")
             onClicked: {
-                CallManager.refreshDevices();
                 var dialog = devicesDialog.createObject(timelineRoot);
                 dialog.open();
             }

src/CallDevices.cpp

@@ -152,7 +152,6 @@ addDevice(GstDevice *device)
         setDefaultDevice(true);
 }
 
-#if GST_CHECK_VERSION(1, 18, 0)
 template<typename T>
 bool
 removeDevice(T &sources, GstDevice *device, bool changed)
@@ -212,7 +211,6 @@ newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data G_G
         }
         return TRUE;
 }
-#endif
 
 template<typename T>
 std::vector<std::string>
@@ -257,7 +255,6 @@ tokenise(std::string_view str, char delim)
 void
 CallDevices::init()
 {
-#if GST_CHECK_VERSION(1, 18, 0)
         static GstDeviceMonitor *monitor = nullptr;
         if (!monitor) {
                 monitor       = gst_device_monitor_new();
@@ -278,43 +275,6 @@ CallDevices::init()
                         return;
                 }
         }
-#endif
-}
-
-void
-CallDevices::refresh()
-{
-#if !GST_CHECK_VERSION(1, 18, 0)
-
-        static GstDeviceMonitor *monitor = nullptr;
-        if (!monitor) {
-                monitor       = gst_device_monitor_new();
-                GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
-                gst_device_monitor_add_filter(monitor, "Audio/Source", caps);
-                gst_device_monitor_add_filter(monitor, "Audio/Duplex", caps);
-                gst_caps_unref(caps);
-                caps = gst_caps_new_empty_simple("video/x-raw");
-                gst_device_monitor_add_filter(monitor, "Video/Source", caps);
-                gst_device_monitor_add_filter(monitor, "Video/Duplex", caps);
-                gst_caps_unref(caps);
-        }
-
-        auto clearDevices = [](auto &sources) {
-                std::for_each(
-                  sources.begin(), sources.end(), [](auto &s) { gst_object_unref(s.device); });
-                sources.clear();
-        };
-        clearDevices(audioSources_);
-        clearDevices(videoSources_);
-
-        GList *devices = gst_device_monitor_get_devices(monitor);
-        if (devices) {
-                for (GList *l = devices; l != nullptr; l = l->next)
-                        addDevice(GST_DEVICE_CAST(l->data));
-                g_list_free(devices);
-        }
-        emit devicesChanged();
-#endif
 }
 
 bool
@@ -400,10 +360,6 @@ CallDevices::videoDevice(std::pair<int, int> &resolution, std::pair<int, int> &f
 
 #else
 
-void
-CallDevices::refresh()
-{}
-
 bool
 CallDevices::haveMic() const
 {

src/CallDevices.h

@@ -19,7 +19,6 @@ public:
                 return instance;
         }
 
-        void refresh();
         bool haveMic() const;
         bool haveCamera() const;
         std::vector<std::string> names(bool isVideo, const std::string &defaultDevice) const;

src/CallManager.cpp

@@ -290,7 +290,6 @@ CallManager::handleEvent(const RoomEvent<CallInvite> &callInviteEvent)
         haveCallInvite_ = true;
         callType_       = isVideo ? CallType::VIDEO : CallType::VOICE;
         inviteSDP_      = callInviteEvent.content.sdp;
-        CallDevices::instance().refresh();
         emit newInviteState();
 }
 

src/CallManager.h

@@ -59,7 +59,6 @@ public:
 public slots:
         void sendInvite(const QString &roomid, webrtc::CallType);
         void syncEvent(const mtx::events::collections::TimelineEvents &event);
-        void refreshDevices() { CallDevices::instance().refresh(); }
         void toggleMicMute();
         void toggleCameraView() { session_.toggleCameraView(); }
         void acceptInvite();

src/UserSettingsPage.cpp

@@ -1288,7 +1288,6 @@ UserSettingsPage::showEvent(QShowEvent *)
         timelineMaxWidthSpin_->setValue(settings_->timelineMaxWidth());
         privacyScreenTimeout_->setValue(settings_->privacyScreenTimeout());
 
-        CallDevices::instance().refresh();
         auto mics = CallDevices::instance().names(false, settings_->microphone().toStdString());
         microphoneCombo_->clear();
         for (const auto &m : mics)

src/WebRTCSession.cpp

@@ -174,7 +174,6 @@ createAnswer(GstPromise *promise, gpointer webrtc)
         g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
 }
 
-#if GST_CHECK_VERSION(1, 18, 0)
 void
 iceGatheringStateChanged(GstElement *webrtc,
                          GParamSpec *pspec G_GNUC_UNUSED,
@@ -194,23 +193,6 @@ iceGatheringStateChanged(GstElement *webrtc,
         }
 }
 
-#else
-
-gboolean
-onICEGatheringCompletion(gpointer timerid)
-{
-        *(guint *)(timerid) = 0;
-        if (WebRTCSession::instance().isOffering()) {
-                emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
-                emit WebRTCSession::instance().stateChanged(State::OFFERSENT);
-        } else {
-                emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
-                emit WebRTCSession::instance().stateChanged(State::ANSWERSENT);
-        }
-        return FALSE;
-}
-#endif
-
 void
 addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
                      guint mlineIndex,
@@ -218,28 +200,7 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
                      gpointer G_GNUC_UNUSED)
 {
         nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
-
-#if GST_CHECK_VERSION(1, 18, 0)
         localcandidates_.push_back({std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
-        return;
-#else
-        if (WebRTCSession::instance().state() >= State::OFFERSENT) {
-                emit WebRTCSession::instance().newICECandidate(
-                  {std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
-                return;
-        }
-
-        localcandidates_.push_back({std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
-
-        // GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
-        // GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18.
-        // Use a 1s timeout in the meantime
-        static guint timerid = 0;
-        if (timerid)
-                g_source_remove(timerid);
-
-        timerid = g_timeout_add(1000, onICEGatheringCompletion, &timerid);
-#endif
 }
 
 void
@@ -328,7 +289,6 @@ testPacketLoss(gpointer G_GNUC_UNUSED)
         return FALSE;
 }
 
-#if GST_CHECK_VERSION(1, 18, 0)
 void
 setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointer G_GNUC_UNUSED)
 {
@@ -337,7 +297,6 @@ setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointe
               "rtpvp8depay"))
                 g_object_set(element, "wait-for-keyframe", TRUE, nullptr);
 }
-#endif
 
 GstElement *
 newAudioSinkChain(GstElement *pipe)
@@ -537,9 +496,7 @@ addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
         // hardware decoding needs investigation; eg rendering fails if vaapi plugin installed
         g_object_set(decodebin, "force-sw-decoders", TRUE, nullptr);
         g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
-#if GST_CHECK_VERSION(1, 18, 0)
         g_signal_connect(decodebin, "element-added", G_CALLBACK(setWaitForKeyFrame), nullptr);
-#endif
         gst_bin_add(GST_BIN(pipe), decodebin);
         gst_element_sync_state_with_parent(decodebin);
         GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
@@ -810,11 +767,10 @@ WebRTCSession::startPipeline(int opusPayloadType, int vp8PayloadType)
         gst_element_set_state(pipe_, GST_STATE_READY);
         g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
 
-#if GST_CHECK_VERSION(1, 18, 0)
         // capture ICE gathering completion
         g_signal_connect(
           webrtc_, "notify::ice-gathering-state", G_CALLBACK(iceGatheringStateChanged), nullptr);
-#endif
+
         // webrtcbin lifetime is the same as that of the pipeline
         gst_object_unref(webrtc_);