mirror of
https://github.com/Nheko-Reborn/nheko.git
synced 2024-11-22 11:00:48 +03:00
Fix required plugin check on gstreamer 1.22
GStreamer 1.22 merged the videoscale plugin into the videoconvertscale plugin. So we should check if the Element is still loadable instead of checking the plugin name. fixes #1352
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parent
50f5fc6637
commit
fc7937c73d
3 changed files with 78 additions and 50 deletions
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@ -193,9 +193,8 @@ CallManager::sendInvite(const QString &roomid, CallType callType, unsigned int w
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auto roomInfo = cache::singleRoomInfo(roomid.toStdString());
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std::string errorMessage;
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if (!session_.havePlugins(false, &errorMessage) ||
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((callType == CallType::VIDEO || callType == CallType::SCREEN) &&
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!session_.havePlugins(true, &errorMessage))) {
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if (!session_.havePlugins(
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callType != CallType::VOICE, callType == CallType::SCREEN, &errorMessage)) {
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emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
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return;
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}
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@ -469,8 +468,8 @@ CallManager::acceptInvite()
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stopRingtone();
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std::string errorMessage;
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if (!session_.havePlugins(false, &errorMessage) ||
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(callType_ == CallType::VIDEO && !session_.havePlugins(true, &errorMessage))) {
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if (!session_.havePlugins(
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callType_ != CallType::VOICE, callType_ == CallType::SCREEN, &errorMessage)) {
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emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
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hangUp(CallHangUp::Reason::UserMediaFailed);
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return;
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@ -580,53 +580,81 @@ getMediaAttributes(const GstSDPMessage *sdp,
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}
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bool
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WebRTCSession::havePlugins(bool isVideo, std::string *errorMessage)
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WebRTCSession::havePlugins(bool isVideo, bool isX11Screenshare, std::string *errorMessage)
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{
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if (!initialised_ && !init(errorMessage))
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return false;
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if (!isVideo && haveVoicePlugins_)
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return true;
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if (isVideo && haveVideoPlugins_)
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if (haveVoicePlugins_ && (!isVideo || haveVideoPlugins_) &&
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(!isX11Screenshare || haveX11ScreensharePlugins_))
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return true;
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const gchar *voicePlugins[] = {"audioconvert",
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"audioresample",
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"autodetect",
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"dtls",
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"nice",
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"opus",
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"playback",
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"rtpmanager",
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"srtp",
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"volume",
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"webrtc",
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nullptr};
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static constexpr std::initializer_list<const char *> audio_elements = {
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"audioconvert",
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"audioresample",
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"autoaudiosink",
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"capsfilter",
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"decodebin",
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"opusenc",
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"queue",
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"rtpopuspay",
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"volume",
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"webrtcbin",
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};
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const gchar *videoPlugins[] = {
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"compositor", "opengl", "qmlgl", "rtp", "videoconvert", "vpx", nullptr};
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static constexpr std::initializer_list<const char *> video_elements = {
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"compositor",
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"glcolorconvert",
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"glsinkbin",
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"glupload",
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"qmlglsink",
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"rtpvp8pay",
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"tee",
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"videoconvert",
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"videoscale",
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"vp8enc",
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};
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static constexpr std::initializer_list<const char *> screenshare_elements = {
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"ximagesink",
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"ximagesrc",
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};
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std::string strError("Missing GStreamer plugins: ");
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const gchar **needed = isVideo ? videoPlugins : voicePlugins;
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bool &havePlugins = isVideo ? haveVideoPlugins_ : haveVoicePlugins_;
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havePlugins = true;
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std::string strError("Missing GStreamer elements: ");
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GstRegistry *registry = gst_registry_get();
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for (guint i = 0; i < g_strv_length((gchar **)needed); i++) {
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GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
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if (!plugin) {
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havePlugins = false;
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strError += std::string(needed[i]) + " ";
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continue;
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auto check_plugins = [&strError,
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registry](const std::initializer_list<const char *> &elements) {
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bool havePlugins = true;
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for (const auto &element : elements) {
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GstPluginFeature *plugin =
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gst_registry_find_feature(registry, element, GST_TYPE_ELEMENT_FACTORY);
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if (!plugin) {
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havePlugins = false;
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strError += std::string(element) + " ";
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continue;
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}
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gst_object_unref(plugin);
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}
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gst_object_unref(plugin);
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}
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if (!havePlugins) {
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return havePlugins;
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};
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haveVoicePlugins_ = check_plugins(audio_elements);
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// check both elements at once
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if (isVideo)
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haveVideoPlugins_ = check_plugins(video_elements);
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if (isX11Screenshare)
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haveX11ScreensharePlugins_ = check_plugins(screenshare_elements);
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if (!haveVoicePlugins_ || (isVideo && !haveVideoPlugins_) ||
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(isX11Screenshare && !haveX11ScreensharePlugins_)) {
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nhlog::ui()->error(strError);
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if (errorMessage)
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*errorMessage = strError;
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return false;
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}
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if (isVideo) {
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if (isVideo || isX11Screenshare) {
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// load qmlglsink to register GStreamer's GstGLVideoItem QML type
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GstElement *qmlglsink = gst_element_factory_make("qmlglsink", nullptr);
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gst_object_unref(qmlglsink);
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@ -54,7 +54,7 @@ public:
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return instance;
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}
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bool havePlugins(bool isVideo, std::string *errorMessage = nullptr);
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bool havePlugins(bool isVideo, bool isX11Screenshare, std::string *errorMessage = nullptr);
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webrtc::CallType callType() const { return callType_; }
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webrtc::State state() const { return state_; }
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bool haveLocalPiP() const;
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@ -93,18 +93,19 @@ private:
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WebRTCSession();
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CallDevices &devices_;
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bool initialised_ = false;
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bool haveVoicePlugins_ = false;
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bool haveVideoPlugins_ = false;
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webrtc::CallType callType_ = webrtc::CallType::VOICE;
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webrtc::State state_ = webrtc::State::DISCONNECTED;
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bool isOffering_ = false;
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bool isRemoteVideoRecvOnly_ = false;
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bool isRemoteVideoSendOnly_ = false;
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QQuickItem *videoItem_ = nullptr;
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GstElement *pipe_ = nullptr;
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GstElement *webrtc_ = nullptr;
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unsigned int busWatchId_ = 0;
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bool initialised_ = false;
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bool haveVoicePlugins_ = false;
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bool haveVideoPlugins_ = false;
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bool haveX11ScreensharePlugins_ = false;
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webrtc::CallType callType_ = webrtc::CallType::VOICE;
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webrtc::State state_ = webrtc::State::DISCONNECTED;
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bool isOffering_ = false;
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bool isRemoteVideoRecvOnly_ = false;
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bool isRemoteVideoSendOnly_ = false;
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QQuickItem *videoItem_ = nullptr;
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GstElement *pipe_ = nullptr;
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GstElement *webrtc_ = nullptr;
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unsigned int busWatchId_ = 0;
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std::vector<std::string> turnServers_;
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uint32_t shareWindowId_ = 0;
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