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@ -11,6 +11,8 @@ extern "C" { |
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#include "gst/webrtc/webrtc.h" |
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} |
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Q_DECLARE_METATYPE(WebRTCSession::State) |
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namespace { |
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bool gisoffer; |
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std::string glocalsdp; |
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@ -29,6 +31,12 @@ std::string::const_iterator findName(const std::string &sdp, const std::string |
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int getPayloadType(const std::string &sdp, const std::string &name); |
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} |
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WebRTCSession::WebRTCSession() : QObject() |
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{ |
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qRegisterMetaType<WebRTCSession::State>(); |
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connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState); |
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} |
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bool |
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WebRTCSession::init(std::string *errorMessage) |
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{ |
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@ -54,14 +62,14 @@ WebRTCSession::init(std::string *errorMessage) |
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nhlog::ui()->info("Initialised " + gstVersion); |
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// GStreamer Plugins:
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// Base: audioconvert, audioresample, opus, playback, videoconvert, volume
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// Base: audioconvert, audioresample, opus, playback, volume
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// Good: autodetect, rtpmanager, vpx
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// Bad: dtls, srtp, webrtc
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// libnice [GLib]: nice
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initialised_ = true; |
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std::string strError = gstVersion + ": Missing plugins: "; |
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const gchar *needed[] = {"audioconvert", "audioresample", "autodetect", "dtls", "nice", |
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"opus", "playback", "rtpmanager", "srtp", "videoconvert", "vpx", "volume", "webrtc", nullptr}; |
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"opus", "playback", "rtpmanager", "srtp", "vpx", "volume", "webrtc", nullptr}; |
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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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@ -91,17 +99,19 @@ WebRTCSession::createOffer() |
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} |
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bool |
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WebRTCSession::acceptOffer(const std::string& sdp) |
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WebRTCSession::acceptOffer(const std::string &sdp) |
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{ |
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nhlog::ui()->debug("Received offer:\n{}", sdp); |
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if (state_ != State::DISCONNECTED) |
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return false; |
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gisoffer = false; |
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glocalsdp.clear(); |
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gcandidates.clear(); |
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int opusPayloadType = getPayloadType(sdp, "opus");
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if (opusPayloadType == -1) { |
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if (opusPayloadType == -1) |
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return false; |
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} |
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GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER); |
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if (!offer) |
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@ -120,9 +130,11 @@ WebRTCSession::acceptOffer(const std::string& sdp) |
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bool |
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WebRTCSession::startPipeline(int opusPayloadType) |
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{ |
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if (isActive()) |
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if (state_ != State::DISCONNECTED) |
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return false; |
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emit stateChanged(State::INITIATING); |
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if (!createPipeline(opusPayloadType)) |
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return false; |
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@ -132,7 +144,12 @@ WebRTCSession::startPipeline(int opusPayloadType) |
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nhlog::ui()->info("WebRTC: Setting STUN server: {}", stunServer_); |
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g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr); |
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} |
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addTurnServers(); |
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for (const auto &uri : turnServers_) { |
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nhlog::ui()->info("WebRTC: Setting TURN server: {}", uri); |
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gboolean udata; |
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g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata)); |
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} |
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// generate the offer when the pipeline goes to PLAYING
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if (gisoffer) |
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@ -152,16 +169,14 @@ WebRTCSession::startPipeline(int opusPayloadType) |
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GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING); |
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if (ret == GST_STATE_CHANGE_FAILURE) { |
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nhlog::ui()->error("WebRTC: unable to start pipeline"); |
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gst_object_unref(pipe_); |
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pipe_ = nullptr; |
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webrtc_ = nullptr; |
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end(); |
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return false; |
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} |
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GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_)); |
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gst_bus_add_watch(bus, newBusMessage, this); |
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gst_object_unref(bus); |
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emit pipelineChanged(true); |
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emit stateChanged(State::INITIATED); |
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return true; |
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} |
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@ -180,10 +195,7 @@ WebRTCSession::createPipeline(int opusPayloadType) |
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if (error) { |
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nhlog::ui()->error("WebRTC: Failed to parse pipeline: {}", error->message); |
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g_error_free(error); |
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if (pipe_) { |
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gst_object_unref(pipe_); |
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pipe_ = nullptr; |
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} |
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end(); |
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return false; |
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} |
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return true; |
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@ -193,7 +205,7 @@ bool |
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WebRTCSession::acceptAnswer(const std::string &sdp) |
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{ |
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nhlog::ui()->debug("WebRTC: Received sdp:\n{}", sdp); |
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if (!isActive()) |
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if (state_ != State::OFFERSENT) |
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return false; |
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GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER); |
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@ -206,18 +218,20 @@ WebRTCSession::acceptAnswer(const std::string &sdp) |
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} |
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void |
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WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate>& candidates) |
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WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates) |
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{ |
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if (isActive()) { |
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for (const auto& c : candidates) |
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if (state_ >= State::INITIATED) { |
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for (const auto &c : candidates) |
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g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str()); |
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} |
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if (state_ < State::CONNECTED) |
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emit stateChanged(State::CONNECTING); |
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} |
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bool |
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WebRTCSession::toggleMuteAudioSrc(bool &isMuted) |
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{ |
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if (!isActive()) |
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if (state_ < State::INITIATED) |
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return false; |
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GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel"); |
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@ -241,20 +255,7 @@ WebRTCSession::end() |
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pipe_ = nullptr; |
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} |
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webrtc_ = nullptr; |
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emit pipelineChanged(false); |
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} |
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void |
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WebRTCSession::addTurnServers() |
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{ |
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if (!webrtc_) |
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return; |
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for (const auto &uri : turnServers_) { |
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nhlog::ui()->info("WebRTC: Setting TURN server: {}", uri); |
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gboolean udata; |
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g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata)); |
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} |
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emit stateChanged(State::DISCONNECTED); |
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} |
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namespace { |
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@ -373,8 +374,10 @@ gboolean |
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onICEGatheringCompletion(gpointer timerid) |
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{ |
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*(guint*)(timerid) = 0; |
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if (gisoffer) |
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if (gisoffer) { |
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emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates); |
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT); |
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} |
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else |
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emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates); |
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@ -445,6 +448,9 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe |
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if (queuepad) { |
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if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad))) |
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nhlog::ui()->error("WebRTC: Unable to link new pad"); |
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else { |
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED); |
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} |
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gst_object_unref(queuepad); |
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} |
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} |
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trilene
4 years ago
