Improve debug messages
This commit is contained in:
trilene
parent
6be21beebd
commit
57d5a3d31f
3 changed files with 54 additions and 33 deletions
|
|
@ -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);
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue