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Fix required plugin check on gstreamer 1.22

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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
This commit is contained in:
Nicolas Werner 2023-02-08 00:54:02 +01:00
parent 50f5fc6637
commit fc7937c73d
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GPG key ID: C8D75E610773F2D9
3 changed files with 78 additions and 50 deletions
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9
src/voip/CallManager.cpp
View file

@ -193,9 +193,8 @@ CallManager::sendInvite(const QString &roomid, CallType callType, unsigned int w
auto roomInfo = cache::singleRoomInfo(roomid.toStdString());
std::string errorMessage;
if (!session_.havePlugins(false, &errorMessage) ||
((callType == CallType::VIDEO || callType == CallType::SCREEN) &&
!session_.havePlugins(true, &errorMessage))) {
if (!session_.havePlugins(
callType != CallType::VOICE, callType == CallType::SCREEN, &errorMessage)) {
emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
return;
}
@ -469,8 +468,8 @@ CallManager::acceptInvite()
stopRingtone();
std::string errorMessage;
if (!session_.havePlugins(false, &errorMessage) ||
(callType_ == CallType::VIDEO && !session_.havePlugins(true, &errorMessage))) {
if (!session_.havePlugins(
callType_ != CallType::VOICE, callType_ == CallType::SCREEN, &errorMessage)) {
emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
hangUp(CallHangUp::Reason::UserMediaFailed);
return;

92
src/voip/WebRTCSession.cpp
View file

@ -580,53 +580,81 @@ getMediaAttributes(const GstSDPMessage *sdp,
}
bool
WebRTCSession::havePlugins(bool isVideo, std::string *errorMessage)
WebRTCSession::havePlugins(bool isVideo, bool isX11Screenshare, std::string *errorMessage)
{
if (!initialised_ && !init(errorMessage))
return false;
if (!isVideo && haveVoicePlugins_)
return true;
if (isVideo && haveVideoPlugins_)
if (haveVoicePlugins_ && (!isVideo || haveVideoPlugins_) &&
(!isX11Screenshare || haveX11ScreensharePlugins_))
return true;
const gchar *voicePlugins[] = {"audioconvert",
"audioresample",
"autodetect",
"dtls",
"nice",
"opus",
"playback",
"rtpmanager",
"srtp",
"volume",
"webrtc",
nullptr};
static constexpr std::initializer_list<const char *> audio_elements = {
"audioconvert",
"audioresample",
"autoaudiosink",
"capsfilter",
"decodebin",
"opusenc",
"queue",
"rtpopuspay",
"volume",
"webrtcbin",
};
const gchar *videoPlugins[] = {
"compositor", "opengl", "qmlgl", "rtp", "videoconvert", "vpx", nullptr};
static constexpr std::initializer_list<const char *> video_elements = {
"compositor",
"glcolorconvert",
"glsinkbin",
"glupload",
"qmlglsink",
"rtpvp8pay",
"tee",
"videoconvert",
"videoscale",
"vp8enc",
};
static constexpr std::initializer_list<const char *> screenshare_elements = {
"ximagesink",
"ximagesrc",
};
std::string strError("Missing GStreamer plugins: ");
const gchar **needed = isVideo ? videoPlugins : voicePlugins;
bool &havePlugins = isVideo ? haveVideoPlugins_ : haveVoicePlugins_;
havePlugins = true;
std::string strError("Missing GStreamer elements: ");
GstRegistry *registry = gst_registry_get();
for (guint i = 0; i < g_strv_length((gchar **)needed); i++) {
GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
if (!plugin) {
havePlugins = false;
strError += std::string(needed[i]) + " ";
continue;
auto check_plugins = [&strError,
registry](const std::initializer_list<const char *> &elements) {
bool havePlugins = true;
for (const auto &element : elements) {
GstPluginFeature *plugin =
gst_registry_find_feature(registry, element, GST_TYPE_ELEMENT_FACTORY);
if (!plugin) {
havePlugins = false;
strError += std::string(element) + " ";
continue;
}
gst_object_unref(plugin);
}
gst_object_unref(plugin);
}
if (!havePlugins) {
return havePlugins;
};
haveVoicePlugins_ = check_plugins(audio_elements);
// check both elements at once
if (isVideo)
haveVideoPlugins_ = check_plugins(video_elements);
if (isX11Screenshare)
haveX11ScreensharePlugins_ = check_plugins(screenshare_elements);
if (!haveVoicePlugins_ || (isVideo && !haveVideoPlugins_) ||
(isX11Screenshare && !haveX11ScreensharePlugins_)) {
nhlog::ui()->error(strError);
if (errorMessage)
*errorMessage = strError;
return false;
}
if (isVideo) {
if (isVideo || isX11Screenshare) {
// load qmlglsink to register GStreamer's GstGLVideoItem QML type
GstElement *qmlglsink = gst_element_factory_make("qmlglsink", nullptr);
gst_object_unref(qmlglsink);

27
src/voip/WebRTCSession.h
View file

@ -54,7 +54,7 @@ public:
return instance;
}
bool havePlugins(bool isVideo, std::string *errorMessage = nullptr);
bool havePlugins(bool isVideo, bool isX11Screenshare, std::string *errorMessage = nullptr);
webrtc::CallType callType() const { return callType_; }
webrtc::State state() const { return state_; }
bool haveLocalPiP() const;
@ -93,18 +93,19 @@ private:
WebRTCSession();
CallDevices &devices_;
bool initialised_ = false;
bool haveVoicePlugins_ = false;
bool haveVideoPlugins_ = false;
webrtc::CallType callType_ = webrtc::CallType::VOICE;
webrtc::State state_ = webrtc::State::DISCONNECTED;
bool isOffering_ = false;
bool isRemoteVideoRecvOnly_ = false;
bool isRemoteVideoSendOnly_ = false;
QQuickItem *videoItem_ = nullptr;
GstElement *pipe_ = nullptr;
GstElement *webrtc_ = nullptr;
unsigned int busWatchId_ = 0;
bool initialised_ = false;
bool haveVoicePlugins_ = false;
bool haveVideoPlugins_ = false;
bool haveX11ScreensharePlugins_ = false;
webrtc::CallType callType_ = webrtc::CallType::VOICE;
webrtc::State state_ = webrtc::State::DISCONNECTED;
bool isOffering_ = false;
bool isRemoteVideoRecvOnly_ = false;
bool isRemoteVideoSendOnly_ = false;
QQuickItem *videoItem_ = nullptr;
GstElement *pipe_ = nullptr;
GstElement *webrtc_ = nullptr;
unsigned int busWatchId_ = 0;
std::vector<std::string> turnServers_;
uint32_t shareWindowId_ = 0;