diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index f3fd1bdc..32b67123 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -1,9 +1,10 @@
#include <cctype>
-#include "WebRTCSession.h"
#include "Logging.h"
+#include "WebRTCSession.h"
-extern "C" {
+extern "C"
+{
#include "gst/gst.h"
#include "gst/sdp/sdp.h"
@@ -13,478 +14,498 @@ extern "C" {
Q_DECLARE_METATYPE(WebRTCSession::State)
-namespace {
-bool isoffering_;
-std::string localsdp_;
-std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
-
-gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data);
-GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type);
-void generateOffer(GstElement *webrtc);
-void setLocalDescription(GstPromise *promise, gpointer webrtc);
-void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED);
-gboolean onICEGatheringCompletion(gpointer timerid);
-void iceConnectionStateChanged(GstElement *webrtcbin, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED);
-void createAnswer(GstPromise *promise, gpointer webrtc);
-void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
-void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
-std::string::const_iterator findName(const std::string &sdp, const std::string &name);
-int getPayloadType(const std::string &sdp, const std::string &name);
-}
-
-WebRTCSession::WebRTCSession() : QObject()
+WebRTCSession::WebRTCSession()
+ : QObject()
{
- qRegisterMetaType<WebRTCSession::State>();
- connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
+ qRegisterMetaType<WebRTCSession::State>();
+ connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
}
bool
WebRTCSession::init(std::string *errorMessage)
{
- if (initialised_)
- return true;
+ if (initialised_)
+ return true;
- GError *error = nullptr;
- if (!gst_init_check(nullptr, nullptr, &error)) {
- std::string strError = std::string("WebRTC: failed to initialise GStreamer: ");
- if (error) {
- strError += error->message;
- g_error_free(error);
- }
- nhlog::ui()->error(strError);
- if (errorMessage)
- *errorMessage = strError;
- return false;
- }
+ GError *error = nullptr;
+ if (!gst_init_check(nullptr, nullptr, &error)) {
+ std::string strError = std::string("WebRTC: failed to initialise GStreamer: ");
+ if (error) {
+ strError += error->message;
+ g_error_free(error);
+ }
+ nhlog::ui()->error(strError);
+ if (errorMessage)
+ *errorMessage = strError;
+ return false;
+ }
- gchar *version = gst_version_string();
- std::string gstVersion(version);
- g_free(version);
- nhlog::ui()->info("WebRTC: initialised " + gstVersion);
+ gchar *version = gst_version_string();
+ std::string gstVersion(version);
+ g_free(version);
+ nhlog::ui()->info("WebRTC: initialised " + gstVersion);
- // GStreamer Plugins:
- // Base: audioconvert, audioresample, opus, playback, volume
- // Good: autodetect, rtpmanager
- // 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", "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]);
- if (!plugin) {
- strError += std::string(needed[i]) + " ";
- initialised_ = false;
- continue;
- }
- gst_object_unref(plugin);
- }
+ // GStreamer Plugins:
+ // Base: audioconvert, audioresample, opus, playback, volume
+ // Good: autodetect, rtpmanager
+ // 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",
+ "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]);
+ if (!plugin) {
+ strError += std::string(needed[i]) + " ";
+ initialised_ = false;
+ continue;
+ }
+ gst_object_unref(plugin);
+ }
- if (!initialised_) {
- nhlog::ui()->error(strError);
- if (errorMessage)
- *errorMessage = strError;
- }
- return initialised_;
+ if (!initialised_) {
+ nhlog::ui()->error(strError);
+ if (errorMessage)
+ *errorMessage = strError;
+ }
+ return initialised_;
}
-bool
-WebRTCSession::createOffer()
+namespace {
+
+bool isoffering_;
+std::string localsdp_;
+std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
+
+gboolean
+newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
{
- isoffering_ = true;
- localsdp_.clear();
- localcandidates_.clear();
- return startPipeline(111); // a dynamic opus payload type
+ WebRTCSession *session = static_cast<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);
+ g_clear_error(&error);
+ g_free(debug);
+ session->end();
+ break;
+ default:
+ break;
+ }
+ return TRUE;
}
-bool
-WebRTCSession::acceptOffer(const std::string &sdp)
+GstWebRTCSessionDescription *
+parseSDP(const std::string &sdp, GstWebRTCSDPType type)
{
- nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
- if (state_ != State::DISCONNECTED)
- return false;
+ GstSDPMessage *msg;
+ gst_sdp_message_new(&msg);
+ if (gst_sdp_message_parse_buffer((guint8 *)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
+ return gst_webrtc_session_description_new(type, msg);
+ } else {
+ nhlog::ui()->error("WebRTC: failed to parse remote session description");
+ gst_object_unref(msg);
+ return nullptr;
+ }
+}
- isoffering_ = false;
- localsdp_.clear();
- localcandidates_.clear();
+void
+setLocalDescription(GstPromise *promise, gpointer webrtc)
+{
+ const GstStructure *reply = gst_promise_get_reply(promise);
+ gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
+ GstWebRTCSessionDescription *gstsdp = nullptr;
+ gst_structure_get(reply,
+ isAnswer ? "answer" : "offer",
+ GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
+ &gstsdp,
+ nullptr);
+ gst_promise_unref(promise);
+ g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
- int opusPayloadType = getPayloadType(sdp, "opus");
- if (opusPayloadType == -1)
- return false;
+ gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
+ localsdp_ = std::string(sdp);
+ g_free(sdp);
+ gst_webrtc_session_description_free(gstsdp);
- GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
- if (!offer)
- return false;
+ nhlog::ui()->debug(
+ "WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
+}
- if (!startPipeline(opusPayloadType)) {
- gst_webrtc_session_description_free(offer);
- return false;
- }
+void
+createOffer(GstElement *webrtc)
+{
+ // create-offer first, then set-local-description
+ GstPromise *promise =
+ gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
+ g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
+}
- // set-remote-description first, then create-answer
- GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
- g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
- gst_webrtc_session_description_free(offer);
- return true;
+void
+createAnswer(GstPromise *promise, gpointer webrtc)
+{
+ // create-answer first, then set-local-description
+ gst_promise_unref(promise);
+ promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
+ g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
}
-bool
-WebRTCSession::startPipeline(int opusPayloadType)
+gboolean
+onICEGatheringCompletion(gpointer timerid)
{
- if (state_ != State::DISCONNECTED)
- return false;
+ *(guint *)(timerid) = 0;
+ if (isoffering_) {
+ emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
+ } else {
+ emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
+ }
+ return FALSE;
+}
- emit stateChanged(State::INITIATING);
+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 (!createPipeline(opusPayloadType))
- return false;
+ if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
+ emit WebRTCSession::instance().newICECandidate(
+ {"audio", (uint16_t)mlineIndex, candidate});
+ return;
+ }
- webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
+ localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
- if (!stunServer_.empty()) {
- nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
- g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
- }
+ // GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
+ // GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18. Use a 100ms timeout in
+ // the meantime
+ static guint timerid = 0;
+ if (timerid)
+ g_source_remove(timerid);
- 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));
- }
- if (turnServers_.empty())
- nhlog::ui()->warn("WebRTC: no TURN server provided");
+ timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
+}
- // generate the offer when the pipeline goes to PLAYING
- if (isoffering_)
- g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr);
+void
+iceConnectionStateChanged(GstElement *webrtc,
+ GParamSpec *pspec G_GNUC_UNUSED,
+ gpointer user_data G_GNUC_UNUSED)
+{
+ GstWebRTCICEConnectionState newState;
+ g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
+ switch (newState) {
+ case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
+ nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
+ break;
+ case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
+ nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
+ emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
+ break;
+ default:
+ break;
+ }
+}
- // on-ice-candidate is emitted when a local ICE candidate has been gathered
- g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
+void
+linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
+{
+ GstCaps *caps = gst_pad_get_current_caps(newpad);
+ if (!caps)
+ return;
- // capture ICE failure
- g_signal_connect(webrtc_, "notify::ice-connection-state",
- G_CALLBACK(iceConnectionStateChanged), nullptr);
+ const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
+ gst_caps_unref(caps);
- // incoming streams trigger pad-added
- gst_element_set_state(pipe_, GST_STATE_READY);
- g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
+ GstPad *queuepad = nullptr;
+ if (g_str_has_prefix(name, "audio")) {
+ nhlog::ui()->debug("WebRTC: received incoming audio stream");
+ GstElement *queue = gst_element_factory_make("queue", nullptr);
+ GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement *resample = gst_element_factory_make("audioresample", nullptr);
+ GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
+ gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
+ gst_element_sync_state_with_parent(queue);
+ gst_element_sync_state_with_parent(convert);
+ gst_element_sync_state_with_parent(resample);
+ gst_element_sync_state_with_parent(sink);
+ gst_element_link_many(queue, convert, resample, sink, nullptr);
+ queuepad = gst_element_get_static_pad(queue, "sink");
+ }
- // webrtcbin lifetime is the same as that of the pipeline
- gst_object_unref(webrtc_);
+ 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);
+ }
+}
- // start the pipeline
- GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
- if (ret == GST_STATE_CHANGE_FAILURE) {
- nhlog::ui()->error("WebRTC: unable to start pipeline");
- end();
- return false;
- }
+void
+addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
+{
+ if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
+ return;
- GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
- gst_bus_add_watch(bus, newBusMessage, this);
- gst_object_unref(bus);
- emit stateChanged(State::INITIATED);
- return true;
+ 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");
+ gst_object_unref(sinkpad);
}
-#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
-
-bool
-WebRTCSession::createPipeline(int opusPayloadType)
+std::string::const_iterator
+findName(const std::string &sdp, const std::string &name)
{
- std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
- "autoaudiosrc ! volume name=srclevel ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
- "queue ! " RTP_CAPS_OPUS + std::to_string(opusPayloadType) + " ! webrtcbin.");
-
- webrtc_ = nullptr;
- GError *error = nullptr;
- pipe_ = gst_parse_launch(pipeline.c_str(), &error);
- if (error) {
- nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
- g_error_free(error);
- end();
- return false;
- }
- return true;
+ return std::search(
+ sdp.cbegin(),
+ sdp.cend(),
+ name.cbegin(),
+ name.cend(),
+ [](unsigned char c1, unsigned char c2) { return std::tolower(c1) == std::tolower(c2); });
}
-bool
-WebRTCSession::acceptAnswer(const std::string &sdp)
+int
+getPayloadType(const std::string &sdp, const std::string &name)
{
- nhlog::ui()->debug("WebRTC: received answer:\n{}", sdp);
- if (state_ != State::OFFERSENT)
- return false;
+ // eg a=rtpmap:111 opus/48000/2
+ auto e = findName(sdp, name);
+ if (e == sdp.cend()) {
+ nhlog::ui()->error("WebRTC: remote offer - " + name + " attribute missing");
+ return -1;
+ }
- GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
- if (!answer) {
- end();
- return false;
- }
+ if (auto s = sdp.rfind(':', e - sdp.cbegin()); s == std::string::npos) {
+ nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name +
+ " payload type");
+ return -1;
+ } else {
+ ++s;
+ try {
+ return std::stoi(std::string(sdp, s, e - sdp.cbegin() - s));
+ } catch (...) {
+ nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name +
+ " payload type");
+ }
+ }
+ return -1;
+}
- g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
- gst_webrtc_session_description_free(answer);
- return true;
}
-void
-WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
+bool
+WebRTCSession::createOffer()
{
- if (state_ >= State::INITIATED) {
- 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());
- }
- }
+ isoffering_ = true;
+ localsdp_.clear();
+ localcandidates_.clear();
+ return startPipeline(111); // a dynamic opus payload type
}
bool
-WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
+WebRTCSession::acceptOffer(const std::string &sdp)
{
- if (state_ < State::INITIATED)
- return false;
+ nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
+ if (state_ != State::DISCONNECTED)
+ return false;
- GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
- if (!srclevel)
- return false;
+ isoffering_ = false;
+ localsdp_.clear();
+ localcandidates_.clear();
- gboolean muted;
- g_object_get(srclevel, "mute", &muted, nullptr);
- g_object_set(srclevel, "mute", !muted, nullptr);
- gst_object_unref(srclevel);
- isMuted = !muted;
- return true;
-}
+ int opusPayloadType = getPayloadType(sdp, "opus");
+ if (opusPayloadType == -1)
+ return false;
-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;
- if (state_ != State::DISCONNECTED)
- emit stateChanged(State::DISCONNECTED);
-}
+ GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
+ if (!offer)
+ return false;
-namespace {
+ if (!startPipeline(opusPayloadType)) {
+ gst_webrtc_session_description_free(offer);
+ return false;
+ }
-std::string::const_iterator findName(const std::string &sdp, const std::string &name)
-{
- return std::search(sdp.cbegin(), sdp.cend(), name.cbegin(), name.cend(),
- [](unsigned char c1, unsigned char c2) {return std::tolower(c1) == std::tolower(c2);});
+ // set-remote-description first, then create-answer
+ GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
+ g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
+ gst_webrtc_session_description_free(offer);
+ return true;
}
-int getPayloadType(const std::string &sdp, const std::string &name)
+bool
+WebRTCSession::acceptAnswer(const std::string &sdp)
{
- // eg a=rtpmap:111 opus/48000/2
- auto e = findName(sdp, name);
- if (e == sdp.cend()) {
- nhlog::ui()->error("WebRTC: remote offer - " + name + " attribute missing");
- return -1;
- }
-
- if (auto s = sdp.rfind(':', e - sdp.cbegin()); s == std::string::npos) {
- nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
- return -1;
- }
- else {
- ++s;
- try {
- return std::stoi(std::string(sdp, s, e - sdp.cbegin() - s));
- }
- catch(...) {
- nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
- }
- }
- return -1;
-}
+ nhlog::ui()->debug("WebRTC: received answer:\n{}", sdp);
+ if (state_ != State::OFFERSENT)
+ return false;
-gboolean
-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);
- g_clear_error(&error);
- g_free(debug);
- session->end();
- break;
- default:
- break;
- }
- return TRUE;
-}
+ GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
+ if (!answer) {
+ end();
+ return false;
+ }
-GstWebRTCSessionDescription*
-parseSDP(const std::string &sdp, GstWebRTCSDPType type)
-{
- GstSDPMessage *msg;
- gst_sdp_message_new(&msg);
- if (gst_sdp_message_parse_buffer((guint8*)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
- return gst_webrtc_session_description_new(type, msg);
- }
- else {
- nhlog::ui()->error("WebRTC: failed to parse remote session description");
- gst_object_unref(msg);
- return nullptr;
- }
+ g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
+ gst_webrtc_session_description_free(answer);
+ return true;
}
void
-generateOffer(GstElement *webrtc)
+WebRTCSession::acceptICECandidates(
+ const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
{
- // create-offer first, then set-local-description
- GstPromise *promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
- g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
+ if (state_ >= State::INITIATED) {
+ 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());
+ }
+ }
}
-void
-setLocalDescription(GstPromise *promise, gpointer webrtc)
+bool
+WebRTCSession::startPipeline(int opusPayloadType)
{
- const GstStructure *reply = gst_promise_get_reply(promise);
- gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
- GstWebRTCSessionDescription *gstsdp = nullptr;
- gst_structure_get(reply, isAnswer ? "answer" : "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &gstsdp, nullptr);
- gst_promise_unref(promise);
- g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
+ if (state_ != State::DISCONNECTED)
+ return false;
- gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
- localsdp_ = std::string(sdp);
- g_free(sdp);
- gst_webrtc_session_description_free(gstsdp);
+ emit stateChanged(State::INITIATING);
- nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
-}
+ if (!createPipeline(opusPayloadType))
+ return false;
-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);
+ webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
- if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
- emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
- return;
- }
+ if (!stunServer_.empty()) {
+ nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
+ g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
+ }
- localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
+ 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));
+ }
+ if (turnServers_.empty())
+ nhlog::ui()->warn("WebRTC: no TURN server provided");
- // GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
- // fixed in v1.18
- // use a 100ms timeout in the meantime
- static guint timerid = 0;
- if (timerid)
- g_source_remove(timerid);
+ // generate the offer when the pipeline goes to PLAYING
+ if (isoffering_)
+ g_signal_connect(
+ webrtc_, "on-negotiation-needed", G_CALLBACK(::createOffer), nullptr);
- timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
-}
+ // on-ice-candidate is emitted when a local ICE candidate has been gathered
+ g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
-gboolean
-onICEGatheringCompletion(gpointer timerid)
-{
- *(guint*)(timerid) = 0;
- if (isoffering_) {
- emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
- emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
- }
- else {
- emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
- emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
- }
- return FALSE;
-}
+ // capture ICE failure
+ g_signal_connect(
+ webrtc_, "notify::ice-connection-state", G_CALLBACK(iceConnectionStateChanged), nullptr);
-void
-iceConnectionStateChanged(GstElement *webrtc, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED)
-{
- GstWebRTCICEConnectionState newState;
- g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
- switch (newState) {
- case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
- nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
- emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
- break;
- case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
- nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
- emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
- break;
- default:
- break;
- }
-}
+ // incoming streams trigger pad-added
+ gst_element_set_state(pipe_, GST_STATE_READY);
+ g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
-void
-createAnswer(GstPromise *promise, gpointer webrtc)
-{
- // create-answer first, then set-local-description
- gst_promise_unref(promise);
- promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
- g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
+ // webrtcbin lifetime is the same as that of the pipeline
+ gst_object_unref(webrtc_);
+
+ // start the pipeline
+ GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
+ if (ret == GST_STATE_CHANGE_FAILURE) {
+ nhlog::ui()->error("WebRTC: unable to start pipeline");
+ end();
+ return false;
+ }
+
+ GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
+ gst_bus_add_watch(bus, newBusMessage, this);
+ gst_object_unref(bus);
+ emit stateChanged(State::INITIATED);
+ return true;
}
-void
-addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
+#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
+
+bool
+WebRTCSession::createPipeline(int opusPayloadType)
{
- if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
- return;
+ std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
+ "autoaudiosrc ! volume name=srclevel ! audioconvert ! "
+ "audioresample ! queue ! opusenc ! rtpopuspay ! "
+ "queue ! " RTP_CAPS_OPUS +
+ std::to_string(opusPayloadType) + " ! webrtcbin.");
- 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");
- gst_object_unref(sinkpad);
+ webrtc_ = nullptr;
+ GError *error = nullptr;
+ pipe_ = gst_parse_launch(pipeline.c_str(), &error);
+ if (error) {
+ nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
+ g_error_free(error);
+ end();
+ return false;
+ }
+ return true;
}
-void
-linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
+bool
+WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
{
- GstCaps *caps = gst_pad_get_current_caps(newpad);
- if (!caps)
- return;
-
- const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
- gst_caps_unref(caps);
+ if (state_ < State::INITIATED)
+ return false;
- GstPad *queuepad = nullptr;
- if (g_str_has_prefix(name, "audio")) {
- nhlog::ui()->debug("WebRTC: received incoming audio stream");
- GstElement *queue = gst_element_factory_make("queue", nullptr);
- GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
- GstElement *resample = gst_element_factory_make("audioresample", nullptr);
- GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
- gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
- gst_element_sync_state_with_parent(queue);
- gst_element_sync_state_with_parent(convert);
- gst_element_sync_state_with_parent(resample);
- gst_element_sync_state_with_parent(sink);
- gst_element_link_many(queue, convert, resample, sink, nullptr);
- queuepad = gst_element_get_static_pad(queue, "sink");
- }
+ GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
+ if (!srclevel)
+ return false;
- 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);
- }
+ gboolean muted;
+ g_object_get(srclevel, "mute", &muted, nullptr);
+ g_object_set(srclevel, "mute", !muted, nullptr);
+ gst_object_unref(srclevel);
+ isMuted = !muted;
+ return true;
}
+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;
+ if (state_ != State::DISCONNECTED)
+ emit stateChanged(State::DISCONNECTED);
}
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