diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index d8497833..8f877c41 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -1,4 +1,5 @@
#include <cctype>
+#include <QQmlEngine>
#include "Logging.h"
#include "WebRTCSession.h"
@@ -14,12 +15,22 @@ 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();
}
@@ -247,11 +258,11 @@ iceGatheringStateChanged(GstElement *webrtc,
if (isoffering_) {
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(
- WebRTCSession::State::OFFERSENT);
+ State::OFFERSENT);
} else {
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(
- WebRTCSession::State::ANSWERSENT);
+ State::ANSWERSENT);
}
}
}
@@ -264,10 +275,10 @@ 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 +296,7 @@ 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 +325,11 @@ 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;
@@ -356,7 +367,7 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
nhlog::ui()->error("WebRTC: unable to link new pad");
else {
emit WebRTCSession::instance().stateChanged(
- WebRTCSession::State::CONNECTED);
+ State::CONNECTED);
}
gst_object_unref(queuepad);
}
@@ -633,21 +644,17 @@ WebRTCSession::createPipeline(int opusPayloadType)
}
bool
-WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
+WebRTCSession::toggleMuteAudioSource()
{
if (state_ < State::INITIATED)
return false;
GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
- if (!srclevel)
- return false;
-
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 +785,7 @@ WebRTCSession::createPipeline(int)
}
bool
-WebRTCSession::toggleMuteAudioSrc(bool &)
+WebRTCSession::toggleMuteAudioSource()
{
return false;
}
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