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Merge remote-tracking branch 'origin/master' into cross-signing

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This commit is contained in:
Nicolas Werner 2020-10-08 16:57:03 +02:00
commit 99ba1f17d3
21 changed files with 2228 additions and 279 deletions
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54
src/WebRTCSession.cpp
View file

@ -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);
}
@ -632,21 +635,30 @@ 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
@ -777,7 +789,13 @@ WebRTCSession::createPipeline(int)
}
bool
WebRTCSession::toggleMuteAudioSrc(bool &)
WebRTCSession::isMicMuted() const
{
return false;
}
bool
WebRTCSession::toggleMicMute()
{
return false;
}