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Improve debug messages

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master
trilene 4 years ago
parent 6be21beebd
commit 57d5a3d31f
3 changed files with 54 additions and 33 deletions
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  1. 31
      src/CallManager.cpp
  2. 2
      src/CallManager.h
  3. 54
      src/WebRTCSession.cpp

31
src/CallManager.cpp
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@ -21,7 +21,7 @@ Q_DECLARE_METATYPE(mtx::responses::TurnServer)
using namespace mtx::events; using namespace mtx::events;
using namespace mtx::events::msg; using namespace mtx::events::msg;
// TODO Allow alterative in settings // https://github.com/vector-im/riot-web/issues/10173
#define STUN_SERVER "stun://turn.matrix.org:3478" #define STUN_SERVER "stun://turn.matrix.org:3478"
CallManager::CallManager(QSharedPointer<UserSettings> userSettings) CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
@ -38,14 +38,13 @@ CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
[this](const std::string &sdp, [this](const std::string &sdp,
const std::vector<CallCandidates::Candidate> &candidates) const std::vector<CallCandidates::Candidate> &candidates)
{ {
nhlog::ui()->debug("Offer created with callid_ and roomid_: {} {}", callid_, roomid_.toStdString()); nhlog::ui()->debug("WebRTC: call id: {} - sending offer", callid_);
emit newMessage(roomid_, CallInvite{callid_, sdp, 0, timeoutms_}); emit newMessage(roomid_, CallInvite{callid_, sdp, 0, timeoutms_});
emit newMessage(roomid_, CallCandidates{callid_, candidates, 0}); emit newMessage(roomid_, CallCandidates{callid_, candidates, 0});
QTimer::singleShot(timeoutms_, this, [this](){ QTimer::singleShot(timeoutms_, this, [this](){
if (session_.state() == WebRTCSession::State::OFFERSENT) { if (session_.state() == WebRTCSession::State::OFFERSENT) {
emit newMessage(roomid_, CallHangUp{callid_, 0, CallHangUp::Reason::InviteTimeOut}); hangUp(CallHangUp::Reason::InviteTimeOut);
endCall();
emit ChatPage::instance()->showNotification("The remote side failed to pick up."); emit ChatPage::instance()->showNotification("The remote side failed to pick up.");
} }
}); });
@ -55,7 +54,7 @@ CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
[this](const std::string &sdp, [this](const std::string &sdp,
const std::vector<CallCandidates::Candidate> &candidates) const std::vector<CallCandidates::Candidate> &candidates)
{ {
nhlog::ui()->debug("Answer created with callid_ and roomid_: {} {}", callid_, roomid_.toStdString()); nhlog::ui()->debug("WebRTC: call id: {} - sending answer", callid_);
emit newMessage(roomid_, CallAnswer{callid_, sdp, 0}); emit newMessage(roomid_, CallAnswer{callid_, sdp, 0});
emit newMessage(roomid_, CallCandidates{callid_, candidates, 0}); emit newMessage(roomid_, CallCandidates{callid_, candidates, 0});
}); });
@ -63,7 +62,7 @@ CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
connect(&session_, &WebRTCSession::newICECandidate, this, connect(&session_, &WebRTCSession::newICECandidate, this,
[this](const CallCandidates::Candidate &candidate) [this](const CallCandidates::Candidate &candidate)
{ {
nhlog::ui()->debug("New ICE candidate created with callid_ and roomid_: {} {}", callid_, roomid_.toStdString()); nhlog::ui()->debug("WebRTC: call id: {} - sending ice candidate", callid_);
emit newMessage(roomid_, CallCandidates{callid_, {candidate}, 0}); emit newMessage(roomid_, CallCandidates{callid_, {candidate}, 0});
}); });
@ -121,6 +120,7 @@ CallManager::sendInvite(const QString &roomid)
session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : ""); session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
generateCallID(); generateCallID();
nhlog::ui()->debug("WebRTC: call id: {} - creating invite", callid_);
std::vector<RoomMember> members(cache::getMembers(roomid.toStdString())); std::vector<RoomMember> members(cache::getMembers(roomid.toStdString()));
const RoomMember &callee = members.front().user_id == utils::localUser() ? members.back() : members.front(); const RoomMember &callee = members.front().user_id == utils::localUser() ? members.back() : members.front();
emit newCallParty(callee.user_id, callee.display_name, emit newCallParty(callee.user_id, callee.display_name,
@ -133,11 +133,11 @@ CallManager::sendInvite(const QString &roomid)
} }
void void
CallManager::hangUp() CallManager::hangUp(CallHangUp::Reason reason)
{ {
nhlog::ui()->debug("CallManager::hangUp: roomid_: {}", roomid_.toStdString());
if (!callid_.empty()) { if (!callid_.empty()) {
emit newMessage(roomid_, CallHangUp{callid_, 0, CallHangUp::Reason::User}); nhlog::ui()->debug("WebRTC: call id: {} - hanging up", callid_);
emit newMessage(roomid_, CallHangUp{callid_, 0, reason});
endCall(); endCall();
} }
} }
@ -169,7 +169,8 @@ CallManager::handleEvent_(const mtx::events::collections::TimelineEvents &event)
void void
CallManager::handleEvent(const RoomEvent<CallInvite> &callInviteEvent) CallManager::handleEvent(const RoomEvent<CallInvite> &callInviteEvent)
{ {
nhlog::ui()->debug("CallManager::incoming CallInvite from {} with id {}", callInviteEvent.sender, callInviteEvent.content.call_id); nhlog::ui()->debug("WebRTC: call id: {} - incoming CallInvite from {}",
callInviteEvent.content.call_id, callInviteEvent.sender);
if (callInviteEvent.content.call_id.empty()) if (callInviteEvent.content.call_id.empty())
return; return;
@ -238,7 +239,8 @@ CallManager::handleEvent(const RoomEvent<CallCandidates> &callCandidatesEvent)
if (callCandidatesEvent.sender == utils::localUser().toStdString()) if (callCandidatesEvent.sender == utils::localUser().toStdString())
return; return;
nhlog::ui()->debug("CallManager::incoming CallCandidates from {} with id {}", callCandidatesEvent.sender, callCandidatesEvent.content.call_id); nhlog::ui()->debug("WebRTC: call id: {} - incoming CallCandidates from {}",
callCandidatesEvent.content.call_id, callCandidatesEvent.sender);
if (callid_ == callCandidatesEvent.content.call_id) { if (callid_ == callCandidatesEvent.content.call_id) {
if (onActiveCall()) if (onActiveCall())
@ -254,7 +256,8 @@ CallManager::handleEvent(const RoomEvent<CallCandidates> &callCandidatesEvent)
void void
CallManager::handleEvent(const RoomEvent<CallAnswer> &callAnswerEvent) CallManager::handleEvent(const RoomEvent<CallAnswer> &callAnswerEvent)
{ {
nhlog::ui()->debug("CallManager::incoming CallAnswer from {} with id {}", callAnswerEvent.sender, callAnswerEvent.content.call_id); nhlog::ui()->debug("WebRTC: call id: {} - incoming CallAnswer from {}",
callAnswerEvent.content.call_id, callAnswerEvent.sender);
if (!onActiveCall() && callAnswerEvent.sender == utils::localUser().toStdString() && if (!onActiveCall() && callAnswerEvent.sender == utils::localUser().toStdString() &&
callid_ == callAnswerEvent.content.call_id) { callid_ == callAnswerEvent.content.call_id) {
@ -276,7 +279,9 @@ CallManager::handleEvent(const RoomEvent<CallAnswer> &callAnswerEvent)
void void
CallManager::handleEvent(const RoomEvent<CallHangUp> &callHangUpEvent) CallManager::handleEvent(const RoomEvent<CallHangUp> &callHangUpEvent)
{ {
nhlog::ui()->debug("CallManager::incoming CallHangUp from {} with id {}", callHangUpEvent.sender, callHangUpEvent.content.call_id); nhlog::ui()->debug("WebRTC: call id: {} - incoming CallHangUp from {}",
callHangUpEvent.content.call_id, callHangUpEvent.sender);
if (callid_ == callHangUpEvent.content.call_id) { if (callid_ == callHangUpEvent.content.call_id) {
MainWindow::instance()->hideOverlay(); MainWindow::instance()->hideOverlay();
endCall(); endCall();

2
src/CallManager.h
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@ -24,7 +24,7 @@ public:
CallManager(QSharedPointer<UserSettings>); CallManager(QSharedPointer<UserSettings>);
void sendInvite(const QString &roomid); void sendInvite(const QString &roomid);
void hangUp(); void hangUp(mtx::events::msg::CallHangUp::Reason = mtx::events::msg::CallHangUp::Reason::User);
bool onActiveCall(); bool onActiveCall();
public slots: public slots:

54
src/WebRTCSession.cpp
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@ -45,7 +45,7 @@ WebRTCSession::init(std::string *errorMessage)
GError *error = nullptr; GError *error = nullptr;
if (!gst_init_check(nullptr, nullptr, &error)) { 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) { if (error) {
strError += error->message; strError += error->message;
g_error_free(error); g_error_free(error);
@ -59,7 +59,7 @@ WebRTCSession::init(std::string *errorMessage)
gchar *version = gst_version_string(); gchar *version = gst_version_string();
std::string gstVersion(version); std::string gstVersion(version);
g_free(version); g_free(version);
nhlog::ui()->info("Initialised " + gstVersion); nhlog::ui()->info("WebRTC: initialised " + gstVersion);
// GStreamer Plugins: // GStreamer Plugins:
// Base: audioconvert, audioresample, opus, playback, volume // Base: audioconvert, audioresample, opus, playback, volume
@ -101,7 +101,7 @@ WebRTCSession::createOffer()
bool bool
WebRTCSession::acceptOffer(const std::string &sdp) 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) if (state_ != State::DISCONNECTED)
return false; return false;
@ -117,8 +117,10 @@ WebRTCSession::acceptOffer(const std::string &sdp)
if (!offer) if (!offer)
return false; return false;
if (!startPipeline(opusPayloadType)) if (!startPipeline(opusPayloadType)) {
gst_webrtc_session_description_free(offer);
return false; return false;
}
// set-remote-description first, then create-answer // set-remote-description first, then create-answer
GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr); 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"); webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
if (!stunServer_.empty()) { 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); g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
} }
for (const auto &uri : turnServers_) { for (const auto &uri : turnServers_) {
nhlog::ui()->info("WebRTC: Setting TURN server: {}", uri); nhlog::ui()->info("WebRTC: setting TURN server: {}", uri);
gboolean udata; gboolean udata;
g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&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; GError *error = nullptr;
pipe_ = gst_parse_launch(pipeline.c_str(), &error); pipe_ = gst_parse_launch(pipeline.c_str(), &error);
if (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); g_error_free(error);
end(); end();
return false; return false;
@ -204,13 +206,15 @@ WebRTCSession::createPipeline(int opusPayloadType)
bool bool
WebRTCSession::acceptAnswer(const std::string &sdp) 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) if (state_ != State::OFFERSENT)
return false; return false;
GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER); GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
if (!answer) if (!answer) {
end();
return false; return false;
}
g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr); g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
gst_webrtc_session_description_free(answer); gst_webrtc_session_description_free(answer);
@ -221,11 +225,13 @@ void
WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates) WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
{ {
if (state_ >= State::INITIATED) { 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()); 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 bool
@ -249,13 +255,15 @@ WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
void void
WebRTCSession::end() WebRTCSession::end()
{ {
nhlog::ui()->debug("WebRTC: ending session");
if (pipe_) { if (pipe_) {
gst_element_set_state(pipe_, GST_STATE_NULL); gst_element_set_state(pipe_, GST_STATE_NULL);
gst_object_unref(pipe_); gst_object_unref(pipe_);
pipe_ = nullptr; pipe_ = nullptr;
} }
webrtc_ = nullptr; webrtc_ = nullptr;
emit stateChanged(State::DISCONNECTED); if (state_ != State::DISCONNECTED)
emit stateChanged(State::DISCONNECTED);
} }
namespace { namespace {
@ -297,13 +305,14 @@ newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
WebRTCSession *session = (WebRTCSession*)user_data; WebRTCSession *session = (WebRTCSession*)user_data;
switch (GST_MESSAGE_TYPE(msg)) { switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_EOS: case GST_MESSAGE_EOS:
nhlog::ui()->error("WebRTC: end of stream");
session->end(); session->end();
break; break;
case GST_MESSAGE_ERROR: case GST_MESSAGE_ERROR:
GError *error; GError *error;
gchar *debug; gchar *debug;
gst_message_parse_error(msg, &error, &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_clear_error(&error);
g_free(debug); g_free(debug);
session->end(); session->end();
@ -323,7 +332,7 @@ parseSDP(const std::string &sdp, GstWebRTCSDPType type)
return gst_webrtc_session_description_new(type, msg); return gst_webrtc_session_description_new(type, msg);
} }
else { 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); gst_object_unref(msg);
return nullptr; return nullptr;
} }
@ -352,12 +361,14 @@ setLocalDescription(GstPromise *promise, gpointer webrtc)
g_free(sdp); g_free(sdp);
gst_webrtc_session_description_free(gstsdp); 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 void
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED) 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) { if (WebRTCSession::instance().state() == WebRTCSession::State::CONNECTED) {
emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate}); emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
return; return;
@ -383,8 +394,10 @@ onICEGatheringCompletion(gpointer timerid)
emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates); emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT); emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
} }
else else {
emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates); emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
}
return FALSE; return FALSE;
} }
@ -404,13 +417,14 @@ addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC) if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
return; return;
nhlog::ui()->debug("WebRTC: received incoming stream");
GstElement *decodebin = gst_element_factory_make("decodebin", nullptr); GstElement *decodebin = gst_element_factory_make("decodebin", nullptr);
g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe); g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
gst_bin_add(GST_BIN(pipe), decodebin); gst_bin_add(GST_BIN(pipe), decodebin);
gst_element_sync_state_with_parent(decodebin); gst_element_sync_state_with_parent(decodebin);
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink"); GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad))) 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); 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); GstElement *queue = gst_element_factory_make("queue", nullptr);
if (g_str_has_prefix(name, "audio")) { if (g_str_has_prefix(name, "audio")) {
nhlog::ui()->debug("WebRTC: received incoming audio stream");
GstElement *convert = gst_element_factory_make("audioconvert", nullptr); GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
GstElement *resample = gst_element_factory_make("audioresample", nullptr); GstElement *resample = gst_element_factory_make("audioresample", nullptr);
GstElement *sink = gst_element_factory_make("autoaudiosink", 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"); queuepad = gst_element_get_static_pad(queue, "sink");
} }
else if (g_str_has_prefix(name, "video")) { 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 *convert = gst_element_factory_make("videoconvert", nullptr);
GstElement *sink = gst_element_factory_make("autovideosink", nullptr); GstElement *sink = gst_element_factory_make("autovideosink", nullptr);
gst_bin_add_many(GST_BIN(pipe), queue, convert, sink, 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 (queuepad) {
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, 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 { else {
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED); emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED);
} }

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