diff --git a/src/CallDevices.cpp b/src/CallDevices.cpp
index 0b9809e5..f182c133 100644
--- a/src/CallDevices.cpp
+++ b/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
{
diff --git a/src/CallDevices.h b/src/CallDevices.h
index 2b4129f1..6d9e18c6 100644
--- a/src/CallDevices.h
+++ b/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;
diff --git a/src/CallManager.cpp b/src/CallManager.cpp
index 51bb7b33..b920cfed 100644
--- a/src/CallManager.cpp
+++ b/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();
}
diff --git a/src/CallManager.h b/src/CallManager.h
index ed745b5b..cfc50481 100644
--- a/src/CallManager.h
+++ b/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();
diff --git a/src/UserSettingsPage.cpp b/src/UserSettingsPage.cpp
index 1dcf9f63..7410e35b 100644
--- a/src/UserSettingsPage.cpp
+++ b/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)
diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index 2281145b..c3a28117 100644
--- a/src/WebRTCSession.cpp
+++ b/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_);
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