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Polish voice call UI

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This commit is contained in:
trilene 2020-07-22 21:15:45 -04:00
parent da9995fc3d
commit 88cfa3a8fa
18 changed files with 348 additions and 146 deletions
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76
src/WebRTCSession.cpp
View file

@ -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);
}
}