diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index 4ef7a818..5baed72e 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -11,6 +11,8 @@ extern "C" {
#include "gst/webrtc/webrtc.h"
}
+Q_DECLARE_METATYPE(WebRTCSession::State)
+
namespace {
bool gisoffer;
std::string glocalsdp;
@@ -29,6 +31,12 @@ std::string::const_iterator findName(const std::string &sdp, const std::string
int getPayloadType(const std::string &sdp, const std::string &name);
}
+WebRTCSession::WebRTCSession() : QObject()
+{
+ qRegisterMetaType<WebRTCSession::State>();
+ connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
+}
+
bool
WebRTCSession::init(std::string *errorMessage)
{
@@ -54,14 +62,14 @@ WebRTCSession::init(std::string *errorMessage)
nhlog::ui()->info("Initialised " + gstVersion);
// GStreamer Plugins:
- // Base: audioconvert, audioresample, opus, playback, videoconvert, volume
+ // Base: audioconvert, audioresample, opus, playback, volume
// Good: autodetect, rtpmanager, vpx
// Bad: dtls, srtp, webrtc
// libnice [GLib]: nice
initialised_ = true;
std::string strError = gstVersion + ": Missing plugins: ";
const gchar *needed[] = {"audioconvert", "audioresample", "autodetect", "dtls", "nice",
- "opus", "playback", "rtpmanager", "srtp", "videoconvert", "vpx", "volume", "webrtc", nullptr};
+ "opus", "playback", "rtpmanager", "srtp", "vpx", "volume", "webrtc", nullptr};
GstRegistry *registry = gst_registry_get();
for (guint i = 0; i < g_strv_length((gchar**)needed); i++) {
GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
@@ -91,17 +99,19 @@ WebRTCSession::createOffer()
}
bool
-WebRTCSession::acceptOffer(const std::string& sdp)
+WebRTCSession::acceptOffer(const std::string &sdp)
{
nhlog::ui()->debug("Received offer:\n{}", sdp);
+ if (state_ != State::DISCONNECTED)
+ return false;
+
gisoffer = false;
glocalsdp.clear();
gcandidates.clear();
int opusPayloadType = getPayloadType(sdp, "opus");
- if (opusPayloadType == -1) {
+ if (opusPayloadType == -1)
return false;
- }
GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
if (!offer)
@@ -120,9 +130,11 @@ WebRTCSession::acceptOffer(const std::string& sdp)
bool
WebRTCSession::startPipeline(int opusPayloadType)
{
- if (isActive())
+ if (state_ != State::DISCONNECTED)
return false;
+ emit stateChanged(State::INITIATING);
+
if (!createPipeline(opusPayloadType))
return false;
@@ -132,7 +144,12 @@ WebRTCSession::startPipeline(int opusPayloadType)
nhlog::ui()->info("WebRTC: Setting STUN server: {}", stunServer_);
g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
}
- addTurnServers();
+
+ for (const auto &uri : turnServers_) {
+ nhlog::ui()->info("WebRTC: Setting TURN server: {}", uri);
+ gboolean udata;
+ g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
+ }
// generate the offer when the pipeline goes to PLAYING
if (gisoffer)
@@ -152,16 +169,14 @@ WebRTCSession::startPipeline(int opusPayloadType)
GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
nhlog::ui()->error("WebRTC: unable to start pipeline");
- gst_object_unref(pipe_);
- pipe_ = nullptr;
- webrtc_ = nullptr;
+ end();
return false;
}
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
gst_bus_add_watch(bus, newBusMessage, this);
gst_object_unref(bus);
- emit pipelineChanged(true);
+ emit stateChanged(State::INITIATED);
return true;
}
@@ -180,10 +195,7 @@ WebRTCSession::createPipeline(int opusPayloadType)
if (error) {
nhlog::ui()->error("WebRTC: Failed to parse pipeline: {}", error->message);
g_error_free(error);
- if (pipe_) {
- gst_object_unref(pipe_);
- pipe_ = nullptr;
- }
+ end();
return false;
}
return true;
@@ -193,7 +205,7 @@ bool
WebRTCSession::acceptAnswer(const std::string &sdp)
{
nhlog::ui()->debug("WebRTC: Received sdp:\n{}", sdp);
- if (!isActive())
+ if (state_ != State::OFFERSENT)
return false;
GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
@@ -206,18 +218,20 @@ WebRTCSession::acceptAnswer(const std::string &sdp)
}
void
-WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate>& candidates)
+WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
{
- if (isActive()) {
- for (const auto& c : candidates)
+ if (state_ >= State::INITIATED) {
+ for (const auto &c : candidates)
g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
}
+ if (state_ < State::CONNECTED)
+ emit stateChanged(State::CONNECTING);
}
bool
WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
{
- if (!isActive())
+ if (state_ < State::INITIATED)
return false;
GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
@@ -241,20 +255,7 @@ WebRTCSession::end()
pipe_ = nullptr;
}
webrtc_ = nullptr;
- emit pipelineChanged(false);
-}
-
-void
-WebRTCSession::addTurnServers()
-{
- if (!webrtc_)
- return;
-
- for (const auto &uri : turnServers_) {
- nhlog::ui()->info("WebRTC: Setting TURN server: {}", uri);
- gboolean udata;
- g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
- }
+ emit stateChanged(State::DISCONNECTED);
}
namespace {
@@ -373,8 +374,10 @@ gboolean
onICEGatheringCompletion(gpointer timerid)
{
*(guint*)(timerid) = 0;
- if (gisoffer)
+ if (gisoffer) {
emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
+ }
else
emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
@@ -445,6 +448,9 @@ 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");
+ else {
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED);
+ }
gst_object_unref(queuepad);
}
}
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