diff options
Diffstat (limited to 'src/WebRTCSession.cpp')
| -rw-r--r-- | src/WebRTCSession.cpp | 65 |
1 files changed, 44 insertions, 21 deletions
diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp index ff9ec661..95a9041e 100644 --- a/src/WebRTCSession.cpp +++ b/src/WebRTCSession.cpp @@ -14,9 +14,9 @@ extern "C" { Q_DECLARE_METATYPE(WebRTCSession::State) namespace { -bool gisoffer; -std::string glocalsdp; -std::vector<mtx::events::msg::CallCandidates::Candidate> gcandidates; +bool isoffering_; +std::string localsdp_; +std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_; gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data); GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type); @@ -24,6 +24,7 @@ void generateOffer(GstElement *webrtc); void setLocalDescription(GstPromise *promise, gpointer webrtc); void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED); gboolean onICEGatheringCompletion(gpointer timerid); +void iceConnectionStateChanged(GstElement *webrtcbin, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED); void createAnswer(GstPromise *promise, gpointer webrtc); void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe); void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe); @@ -92,9 +93,9 @@ WebRTCSession::init(std::string *errorMessage) bool WebRTCSession::createOffer() { - gisoffer = true; - glocalsdp.clear(); - gcandidates.clear(); + isoffering_ = true; + localsdp_.clear(); + localcandidates_.clear(); return startPipeline(111); // a dynamic opus payload type } @@ -105,9 +106,9 @@ WebRTCSession::acceptOffer(const std::string &sdp) if (state_ != State::DISCONNECTED) return false; - gisoffer = false; - glocalsdp.clear(); - gcandidates.clear(); + isoffering_ = false; + localsdp_.clear(); + localcandidates_.clear(); int opusPayloadType = getPayloadType(sdp, "opus"); if (opusPayloadType == -1) @@ -152,14 +153,20 @@ WebRTCSession::startPipeline(int opusPayloadType) gboolean udata; g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata)); } + if (turnServers_.empty()) + nhlog::ui()->warn("WebRTC: no TURN server provided"); // generate the offer when the pipeline goes to PLAYING - if (gisoffer) + if (isoffering_) g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr); // on-ice-candidate is emitted when a local ICE candidate has been gathered g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr); + // capture ICE failure + g_signal_connect(webrtc_, "notify::ice-connection-state", + G_CALLBACK(iceConnectionStateChanged), nullptr); + // incoming streams trigger pad-added gst_element_set_state(pipe_, GST_STATE_READY); g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_); @@ -229,8 +236,6 @@ WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandi nhlog::ui()->debug("WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate); g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str()); } - if (state_ == State::OFFERSENT) - emit stateChanged(State::CONNECTING); } } @@ -357,11 +362,11 @@ setLocalDescription(GstPromise *promise, gpointer webrtc) g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr); gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp); - glocalsdp = std::string(sdp); + localsdp_ = std::string(sdp); g_free(sdp); gst_webrtc_session_description_free(gstsdp); - nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", glocalsdp); + nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_); } void @@ -369,12 +374,12 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar * { nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate); - if (WebRTCSession::instance().state() == WebRTCSession::State::CONNECTED) { + if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) { emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate}); return; } - gcandidates.push_back({"audio", (uint16_t)mlineIndex, candidate}); + localcandidates_.push_back({"audio", (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 @@ -390,19 +395,37 @@ gboolean onICEGatheringCompletion(gpointer timerid) { *(guint*)(timerid) = 0; - if (gisoffer) { - emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates); + if (isoffering_) { + emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_); emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT); } else { - emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates); - emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING); + emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_); + emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT); } - return FALSE; } void +iceConnectionStateChanged(GstElement *webrtc, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED) +{ + GstWebRTCICEConnectionState newState; + g_object_get(webrtc, "ice-connection-state", &newState, nullptr); + switch (newState) { + case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: + nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking"); + emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING); + break; + case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: + nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed"); + emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED); + break; + default: + break; + } +} + +void createAnswer(GstPromise *promise, gpointer webrtc) { // create-answer first, then set-local-description |