Merge pull request #289 from trilene/master

Port ActiveCallBar to Qml

1 files changed, 36 insertions, 18 deletions

diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp

index d8497833..9c8c032d 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -1,3 +1,4 @@
+#include <QQmlEngine>
 #include <cctype>
 
 #include "Logging.h"
@@ -14,12 +15,17 @@ extern "C"
 }
 #endif
 
-Q_DECLARE_METATYPE(WebRTCSession::State)
+Q_DECLARE_METATYPE(webrtc::State)
+
+using webrtc::State;
 
 WebRTCSession::WebRTCSession()
   : QObject()
 {
-        qRegisterMetaType<WebRTCSession::State>();
+        qRegisterMetaType<webrtc::State>();
+        qmlRegisterUncreatableMetaObject(
+          webrtc::staticMetaObject, "im.nheko", 1, 0, "WebRTCState", "Can't instantiate enum");
+
         connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
         init();
 }
@@ -246,12 +252,10 @@ iceGatheringStateChanged(GstElement *webrtc,
                 nhlog::ui()->debug("WebRTC: GstWebRTCICEGatheringState -> Complete");
                 if (isoffering_) {
                         emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
-                        emit WebRTCSession::instance().stateChanged(
-                          WebRTCSession::State::OFFERSENT);
+                        emit WebRTCSession::instance().stateChanged(State::OFFERSENT);
                 } else {
                         emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
-                        emit WebRTCSession::instance().stateChanged(
-                          WebRTCSession::State::ANSWERSENT);
+                        emit WebRTCSession::instance().stateChanged(State::ANSWERSENT);
                 }
         }
 }
@@ -264,10 +268,10 @@ onICEGatheringCompletion(gpointer timerid)
         *(guint *)(timerid) = 0;
         if (isoffering_) {
                 emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
-                emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
+                emit WebRTCSession::instance().stateChanged(State::OFFERSENT);
         } else {
                 emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
-                emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
+                emit WebRTCSession::instance().stateChanged(State::ANSWERSENT);
         }
         return FALSE;
 }
@@ -285,7 +289,7 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
         localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
         return;
 #else
-        if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
+        if (WebRTCSession::instance().state() >= State::OFFERSENT) {
                 emit WebRTCSession::instance().newICECandidate(
                   {"audio", (uint16_t)mlineIndex, candidate});
                 return;
@@ -314,11 +318,11 @@ iceConnectionStateChanged(GstElement *webrtc,
         switch (newState) {
         case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
                 nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
-                emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
+                emit WebRTCSession::instance().stateChanged(State::CONNECTING);
                 break;
         case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
                 nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
-                emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
+                emit WebRTCSession::instance().stateChanged(State::ICEFAILED);
                 break;
         default:
                 break;
@@ -355,8 +359,7 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
                 if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
                         nhlog::ui()->error("WebRTC: unable to link new pad");
                 else {
-                        emit WebRTCSession::instance().stateChanged(
-                          WebRTCSession::State::CONNECTED);
+                        emit WebRTCSession::instance().stateChanged(State::CONNECTED);
                 }
                 gst_object_unref(queuepad);
         }
@@ -633,21 +636,30 @@ WebRTCSession::createPipeline(int opusPayloadType)
 }
 
 bool
-WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
+WebRTCSession::isMicMuted() const
 {
         if (state_ < State::INITIATED)
                 return false;
 
         GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
-        if (!srclevel)
+        gboolean muted;
+        g_object_get(srclevel, "mute", &muted, nullptr);
+        gst_object_unref(srclevel);
+        return muted;
+}
+
+bool
+WebRTCSession::toggleMicMute()
+{
+        if (state_ < State::INITIATED)
                 return false;
 
+        GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
         gboolean muted;
         g_object_get(srclevel, "mute", &muted, nullptr);
         g_object_set(srclevel, "mute", !muted, nullptr);
         gst_object_unref(srclevel);
-        isMuted = !muted;
-        return true;
+        return !muted;
 }
 
 void
@@ -778,7 +790,13 @@ WebRTCSession::createPipeline(int)
 }
 
 bool
-WebRTCSession::toggleMuteAudioSrc(bool &)
+WebRTCSession::isMicMuted() const
+{
+        return false;
+}
+
+bool
+WebRTCSession::toggleMicMute()
 {
         return false;
 }