diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index c3f5341a..ff9ec661 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -45,7 +45,7 @@ WebRTCSession::init(std::string *errorMessage)
GError *error = nullptr;
if (!gst_init_check(nullptr, nullptr, &error)) {
- std::string strError = std::string("Failed to initialise GStreamer: ");
+ std::string strError = std::string("WebRTC: failed to initialise GStreamer: ");
if (error) {
strError += error->message;
g_error_free(error);
@@ -59,7 +59,7 @@ WebRTCSession::init(std::string *errorMessage)
gchar *version = gst_version_string();
std::string gstVersion(version);
g_free(version);
- nhlog::ui()->info("Initialised " + gstVersion);
+ nhlog::ui()->info("WebRTC: initialised " + gstVersion);
// GStreamer Plugins:
// Base: audioconvert, audioresample, opus, playback, volume
@@ -101,7 +101,7 @@ WebRTCSession::createOffer()
bool
WebRTCSession::acceptOffer(const std::string &sdp)
{
- nhlog::ui()->debug("Received offer:\n{}", sdp);
+ nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
if (state_ != State::DISCONNECTED)
return false;
@@ -117,8 +117,10 @@ WebRTCSession::acceptOffer(const std::string &sdp)
if (!offer)
return false;
- if (!startPipeline(opusPayloadType))
+ if (!startPipeline(opusPayloadType)) {
+ gst_webrtc_session_description_free(offer);
return false;
+ }
// set-remote-description first, then create-answer
GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
@@ -141,12 +143,12 @@ WebRTCSession::startPipeline(int opusPayloadType)
webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
if (!stunServer_.empty()) {
- nhlog::ui()->info("WebRTC: Setting STUN server: {}", stunServer_);
+ nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
}
for (const auto &uri : turnServers_) {
- nhlog::ui()->info("WebRTC: Setting TURN server: {}", uri);
+ nhlog::ui()->info("WebRTC: setting TURN server: {}", uri);
gboolean udata;
g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
}
@@ -193,7 +195,7 @@ WebRTCSession::createPipeline(int opusPayloadType)
GError *error = nullptr;
pipe_ = gst_parse_launch(pipeline.c_str(), &error);
if (error) {
- nhlog::ui()->error("WebRTC: Failed to parse pipeline: {}", error->message);
+ nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
g_error_free(error);
end();
return false;
@@ -204,13 +206,15 @@ WebRTCSession::createPipeline(int opusPayloadType)
bool
WebRTCSession::acceptAnswer(const std::string &sdp)
{
- nhlog::ui()->debug("WebRTC: Received sdp:\n{}", sdp);
+ nhlog::ui()->debug("WebRTC: received answer:\n{}", sdp);
if (state_ != State::OFFERSENT)
return false;
GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
- if (!answer)
+ if (!answer) {
+ end();
return false;
+ }
g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
gst_webrtc_session_description_free(answer);
@@ -221,11 +225,13 @@ void
WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
{
if (state_ >= State::INITIATED) {
- for (const auto &c : candidates)
+ for (const auto &c : candidates) {
+ 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);
}
- if (state_ < State::CONNECTED)
- emit stateChanged(State::CONNECTING);
}
bool
@@ -249,13 +255,15 @@ WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
void
WebRTCSession::end()
{
+ nhlog::ui()->debug("WebRTC: ending session");
if (pipe_) {
gst_element_set_state(pipe_, GST_STATE_NULL);
gst_object_unref(pipe_);
pipe_ = nullptr;
}
webrtc_ = nullptr;
- emit stateChanged(State::DISCONNECTED);
+ if (state_ != State::DISCONNECTED)
+ emit stateChanged(State::DISCONNECTED);
}
namespace {
@@ -297,13 +305,14 @@ newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
WebRTCSession *session = (WebRTCSession*)user_data;
switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_EOS:
+ nhlog::ui()->error("WebRTC: end of stream");
session->end();
break;
case GST_MESSAGE_ERROR:
GError *error;
gchar *debug;
gst_message_parse_error(msg, &error, &debug);
- nhlog::ui()->error("WebRTC: Error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
+ nhlog::ui()->error("WebRTC: error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
g_clear_error(&error);
g_free(debug);
session->end();
@@ -323,7 +332,7 @@ parseSDP(const std::string &sdp, GstWebRTCSDPType type)
return gst_webrtc_session_description_new(type, msg);
}
else {
- nhlog::ui()->error("WebRTC: Failed to parse remote session description");
+ nhlog::ui()->error("WebRTC: failed to parse remote session description");
gst_object_unref(msg);
return nullptr;
}
@@ -352,12 +361,14 @@ setLocalDescription(GstPromise *promise, gpointer webrtc)
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", glocalsdp);
}
void
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED)
{
+ nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
+
if (WebRTCSession::instance().state() == WebRTCSession::State::CONNECTED) {
emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
return;
@@ -383,8 +394,10 @@ onICEGatheringCompletion(gpointer timerid)
emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
}
- else
+ else {
emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
+ }
return FALSE;
}
@@ -404,13 +417,14 @@ addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
return;
+ nhlog::ui()->debug("WebRTC: received incoming stream");
GstElement *decodebin = gst_element_factory_make("decodebin", nullptr);
g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
gst_bin_add(GST_BIN(pipe), decodebin);
gst_element_sync_state_with_parent(decodebin);
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad)))
- nhlog::ui()->error("WebRTC: Unable to link new pad");
+ nhlog::ui()->error("WebRTC: unable to link new pad");
gst_object_unref(sinkpad);
}
@@ -428,6 +442,7 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
GstElement *queue = gst_element_factory_make("queue", nullptr);
if (g_str_has_prefix(name, "audio")) {
+ nhlog::ui()->debug("WebRTC: received incoming audio stream");
GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
@@ -440,6 +455,7 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
queuepad = gst_element_get_static_pad(queue, "sink");
}
else if (g_str_has_prefix(name, "video")) {
+ nhlog::ui()->debug("WebRTC: received incoming video stream");
GstElement *convert = gst_element_factory_make("videoconvert", nullptr);
GstElement *sink = gst_element_factory_make("autovideosink", nullptr);
gst_bin_add_many(GST_BIN(pipe), queue, convert, sink, nullptr);
@@ -452,7 +468,7 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
if (queuepad) {
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
- nhlog::ui()->error("WebRTC: Unable to link new pad");
+ nhlog::ui()->error("WebRTC: unable to link new pad");
else {
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED);
}
|
