diff --git a/src/voip/WebRTCSession.cpp b/src/voip/WebRTCSession.cpp
new file mode 100644
index 00000000..801a365c
--- /dev/null
+++ b/src/voip/WebRTCSession.cpp
@@ -0,0 +1,1155 @@
+// SPDX-FileCopyrightText: 2021 Nheko Contributors
+//
+// SPDX-License-Identifier: GPL-3.0-or-later
+
+#include <QQmlEngine>
+#include <QQuickItem>
+#include <algorithm>
+#include <cctype>
+#include <chrono>
+#include <cstdlib>
+#include <cstring>
+#include <optional>
+#include <string_view>
+#include <thread>
+#include <utility>
+
+#include "CallDevices.h"
+#include "ChatPage.h"
+#include "Logging.h"
+#include "UserSettingsPage.h"
+#include "WebRTCSession.h"
+
+#ifdef GSTREAMER_AVAILABLE
+extern "C"
+{
+#include "gst/gst.h"
+#include "gst/sdp/sdp.h"
+
+#define GST_USE_UNSTABLE_API
+#include "gst/webrtc/webrtc.h"
+}
+#endif
+
+// https://github.com/vector-im/riot-web/issues/10173
+#define STUN_SERVER "stun://turn.matrix.org:3478"
+
+Q_DECLARE_METATYPE(webrtc::CallType)
+Q_DECLARE_METATYPE(webrtc::State)
+
+using webrtc::CallType;
+using webrtc::State;
+
+WebRTCSession::WebRTCSession()
+ : devices_(CallDevices::instance())
+{
+ qRegisterMetaType<webrtc::CallType>();
+ qmlRegisterUncreatableMetaObject(
+ webrtc::staticMetaObject, "im.nheko", 1, 0, "CallType", "Can't instantiate enum");
+
+ qRegisterMetaType<webrtc::State>();
+ qmlRegisterUncreatableMetaObject(
+ webrtc::staticMetaObject, "im.nheko", 1, 0, "WebRTCState", "Can't instantiate enum");
+
+ connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
+ init();
+}
+
+bool
+WebRTCSession::init(std::string *errorMessage)
+{
+#ifdef GSTREAMER_AVAILABLE
+ if (initialised_)
+ return true;
+
+ GError *error = nullptr;
+ if (!gst_init_check(nullptr, nullptr, &error)) {
+ std::string strError("WebRTC: failed to initialise GStreamer: ");
+ if (error) {
+ strError += error->message;
+ g_error_free(error);
+ }
+ nhlog::ui()->error(strError);
+ if (errorMessage)
+ *errorMessage = strError;
+ return false;
+ }
+
+ initialised_ = true;
+ gchar *version = gst_version_string();
+ nhlog::ui()->info("WebRTC: initialised {}", version);
+ g_free(version);
+ devices_.init();
+ return true;
+#else
+ (void)errorMessage;
+ return false;
+#endif
+}
+
+#ifdef GSTREAMER_AVAILABLE
+namespace {
+
+std::string localsdp_;
+std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
+bool haveAudioStream_ = false;
+bool haveVideoStream_ = false;
+GstPad *localPiPSinkPad_ = nullptr;
+GstPad *remotePiPSinkPad_ = nullptr;
+
+gboolean
+newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
+{
+ 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;
+}
+
+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_sdp_message_free(msg);
+ return nullptr;
+ }
+}
+
+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);
+
+ gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
+ localsdp_ = std::string(sdp);
+ g_free(sdp);
+ gst_webrtc_session_description_free(gstsdp);
+
+ nhlog::ui()->debug(
+ "WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
+}
+
+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);
+}
+
+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);
+}
+
+void
+iceGatheringStateChanged(GstElement *webrtc,
+ GParamSpec *pspec G_GNUC_UNUSED,
+ gpointer user_data G_GNUC_UNUSED)
+{
+ GstWebRTCICEGatheringState newState;
+ g_object_get(webrtc, "ice-gathering-state", &newState, nullptr);
+ if (newState == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
+ nhlog::ui()->debug("WebRTC: GstWebRTCICEGatheringState -> Complete");
+ if (WebRTCSession::instance().isOffering()) {
+ emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
+ emit WebRTCSession::instance().stateChanged(State::OFFERSENT);
+ } else {
+ emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
+ emit WebRTCSession::instance().stateChanged(State::ANSWERSENT);
+ }
+ }
+}
+
+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);
+ localcandidates_.push_back({std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
+}
+
+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(State::CONNECTING);
+ break;
+ case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
+ nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
+ emit WebRTCSession::instance().stateChanged(State::ICEFAILED);
+ break;
+ default:
+ break;
+ }
+}
+
+// https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1164
+struct KeyFrameRequestData
+{
+ GstElement *pipe = nullptr;
+ GstElement *decodebin = nullptr;
+ gint packetsLost = 0;
+ guint timerid = 0;
+ std::string statsField;
+} keyFrameRequestData_;
+
+void
+sendKeyFrameRequest()
+{
+ GstPad *sinkpad = gst_element_get_static_pad(keyFrameRequestData_.decodebin, "sink");
+ if (!gst_pad_push_event(sinkpad,
+ gst_event_new_custom(GST_EVENT_CUSTOM_UPSTREAM,
+ gst_structure_new_empty("GstForceKeyUnit"))))
+ nhlog::ui()->error("WebRTC: key frame request failed");
+ else
+ nhlog::ui()->debug("WebRTC: sent key frame request");
+
+ gst_object_unref(sinkpad);
+}
+
+void
+testPacketLoss_(GstPromise *promise, gpointer G_GNUC_UNUSED)
+{
+ const GstStructure *reply = gst_promise_get_reply(promise);
+ gint packetsLost = 0;
+ GstStructure *rtpStats;
+ if (!gst_structure_get(
+ reply, keyFrameRequestData_.statsField.c_str(), GST_TYPE_STRUCTURE, &rtpStats, nullptr)) {
+ nhlog::ui()->error("WebRTC: get-stats: no field: {}", keyFrameRequestData_.statsField);
+ gst_promise_unref(promise);
+ return;
+ }
+ gst_structure_get_int(rtpStats, "packets-lost", &packetsLost);
+ gst_structure_free(rtpStats);
+ gst_promise_unref(promise);
+ if (packetsLost > keyFrameRequestData_.packetsLost) {
+ nhlog::ui()->debug("WebRTC: inbound video lost packet count: {}", packetsLost);
+ keyFrameRequestData_.packetsLost = packetsLost;
+ sendKeyFrameRequest();
+ }
+}
+
+gboolean
+testPacketLoss(gpointer G_GNUC_UNUSED)
+{
+ if (keyFrameRequestData_.pipe) {
+ GstElement *webrtc = gst_bin_get_by_name(GST_BIN(keyFrameRequestData_.pipe), "webrtcbin");
+ GstPromise *promise = gst_promise_new_with_change_func(testPacketLoss_, nullptr, nullptr);
+ g_signal_emit_by_name(webrtc, "get-stats", nullptr, promise);
+ gst_object_unref(webrtc);
+ return TRUE;
+ }
+ return FALSE;
+}
+
+void
+setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointer G_GNUC_UNUSED)
+{
+ if (!std::strcmp(
+ gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(gst_element_get_factory(element))),
+ "rtpvp8depay"))
+ g_object_set(element, "wait-for-keyframe", TRUE, nullptr);
+}
+
+GstElement *
+newAudioSinkChain(GstElement *pipe)
+{
+ 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_link_many(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);
+ return queue;
+}
+
+GstElement *
+newVideoSinkChain(GstElement *pipe)
+{
+ // use compositor for now; acceleration needs investigation
+ GstElement *queue = gst_element_factory_make("queue", nullptr);
+ GstElement *compositor = gst_element_factory_make("compositor", "compositor");
+ GstElement *glupload = gst_element_factory_make("glupload", nullptr);
+ GstElement *glcolorconvert = gst_element_factory_make("glcolorconvert", nullptr);
+ GstElement *qmlglsink = gst_element_factory_make("qmlglsink", nullptr);
+ GstElement *glsinkbin = gst_element_factory_make("glsinkbin", nullptr);
+ g_object_set(compositor, "background", 1, nullptr);
+ g_object_set(qmlglsink, "widget", WebRTCSession::instance().getVideoItem(), nullptr);
+ g_object_set(glsinkbin, "sink", qmlglsink, nullptr);
+ gst_bin_add_many(
+ GST_BIN(pipe), queue, compositor, glupload, glcolorconvert, glsinkbin, nullptr);
+ gst_element_link_many(queue, compositor, glupload, glcolorconvert, glsinkbin, nullptr);
+ gst_element_sync_state_with_parent(queue);
+ gst_element_sync_state_with_parent(compositor);
+ gst_element_sync_state_with_parent(glupload);
+ gst_element_sync_state_with_parent(glcolorconvert);
+ gst_element_sync_state_with_parent(glsinkbin);
+ return queue;
+}
+
+std::pair<int, int>
+getResolution(GstPad *pad)
+{
+ std::pair<int, int> ret;
+ GstCaps *caps = gst_pad_get_current_caps(pad);
+ const GstStructure *s = gst_caps_get_structure(caps, 0);
+ gst_structure_get_int(s, "width", &ret.first);
+ gst_structure_get_int(s, "height", &ret.second);
+ gst_caps_unref(caps);
+ return ret;
+}
+
+std::pair<int, int>
+getResolution(GstElement *pipe, const gchar *elementName, const gchar *padName)
+{
+ GstElement *element = gst_bin_get_by_name(GST_BIN(pipe), elementName);
+ GstPad *pad = gst_element_get_static_pad(element, padName);
+ auto ret = getResolution(pad);
+ gst_object_unref(pad);
+ gst_object_unref(element);
+ return ret;
+}
+
+std::pair<int, int>
+getPiPDimensions(const std::pair<int, int> &resolution, int fullWidth, double scaleFactor)
+{
+ int pipWidth = fullWidth * scaleFactor;
+ int pipHeight = static_cast<double>(resolution.second) / resolution.first * pipWidth;
+ return {pipWidth, pipHeight};
+}
+
+void
+addLocalPiP(GstElement *pipe, const std::pair<int, int> &videoCallSize)
+{
+ // embed localUser's camera into received video (CallType::VIDEO)
+ // OR embed screen share into received video (CallType::SCREEN)
+ GstElement *tee = gst_bin_get_by_name(GST_BIN(pipe), "videosrctee");
+ if (!tee)
+ return;
+
+ GstElement *queue = gst_element_factory_make("queue", nullptr);
+ gst_bin_add(GST_BIN(pipe), queue);
+ gst_element_link(tee, queue);
+ gst_element_sync_state_with_parent(queue);
+ gst_object_unref(tee);
+
+ GstElement *compositor = gst_bin_get_by_name(GST_BIN(pipe), "compositor");
+ localPiPSinkPad_ = gst_element_get_request_pad(compositor, "sink_%u");
+ g_object_set(localPiPSinkPad_, "zorder", 2, nullptr);
+
+ bool isVideo = WebRTCSession::instance().callType() == CallType::VIDEO;
+ const gchar *element = isVideo ? "camerafilter" : "screenshare";
+ const gchar *pad = isVideo ? "sink" : "src";
+ auto resolution = getResolution(pipe, element, pad);
+ auto pipSize = getPiPDimensions(resolution, videoCallSize.first, 0.25);
+ nhlog::ui()->debug("WebRTC: local picture-in-picture: {}x{}", pipSize.first, pipSize.second);
+ g_object_set(localPiPSinkPad_, "width", pipSize.first, "height", pipSize.second, nullptr);
+ gint offset = videoCallSize.first / 80;
+ g_object_set(localPiPSinkPad_, "xpos", offset, "ypos", offset, nullptr);
+
+ GstPad *srcpad = gst_element_get_static_pad(queue, "src");
+ if (GST_PAD_LINK_FAILED(gst_pad_link(srcpad, localPiPSinkPad_)))
+ nhlog::ui()->error("WebRTC: failed to link local PiP elements");
+ gst_object_unref(srcpad);
+ gst_object_unref(compositor);
+}
+
+void
+addRemotePiP(GstElement *pipe)
+{
+ // embed localUser's camera into screen image being shared
+ if (remotePiPSinkPad_) {
+ auto camRes = getResolution(pipe, "camerafilter", "sink");
+ auto shareRes = getResolution(pipe, "screenshare", "src");
+ auto pipSize = getPiPDimensions(camRes, shareRes.first, 0.2);
+ nhlog::ui()->debug(
+ "WebRTC: screen share picture-in-picture: {}x{}", pipSize.first, pipSize.second);
+
+ gint offset = shareRes.first / 100;
+ g_object_set(remotePiPSinkPad_, "zorder", 2, nullptr);
+ g_object_set(remotePiPSinkPad_, "width", pipSize.first, "height", pipSize.second, nullptr);
+ g_object_set(remotePiPSinkPad_,
+ "xpos",
+ shareRes.first - pipSize.first - offset,
+ "ypos",
+ shareRes.second - pipSize.second - offset,
+ nullptr);
+ }
+}
+
+void
+addLocalVideo(GstElement *pipe)
+{
+ GstElement *queue = newVideoSinkChain(pipe);
+ GstElement *tee = gst_bin_get_by_name(GST_BIN(pipe), "videosrctee");
+ GstPad *srcpad = gst_element_get_request_pad(tee, "src_%u");
+ GstPad *sinkpad = gst_element_get_static_pad(queue, "sink");
+ if (GST_PAD_LINK_FAILED(gst_pad_link(srcpad, sinkpad)))
+ nhlog::ui()->error("WebRTC: failed to link videosrctee -> video sink chain");
+ gst_object_unref(srcpad);
+}
+
+void
+linkNewPad(GstElement *decodebin, GstPad *newpad, GstElement *pipe)
+{
+ GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
+ GstCaps *sinkcaps = gst_pad_get_current_caps(sinkpad);
+ const GstStructure *structure = gst_caps_get_structure(sinkcaps, 0);
+
+ gchar *mediaType = nullptr;
+ guint ssrc = 0;
+ gst_structure_get(
+ structure, "media", G_TYPE_STRING, &mediaType, "ssrc", G_TYPE_UINT, &ssrc, nullptr);
+ gst_caps_unref(sinkcaps);
+ gst_object_unref(sinkpad);
+
+ WebRTCSession *session = &WebRTCSession::instance();
+ GstElement *queue = nullptr;
+ if (!std::strcmp(mediaType, "audio")) {
+ nhlog::ui()->debug("WebRTC: received incoming audio stream");
+ haveAudioStream_ = true;
+ queue = newAudioSinkChain(pipe);
+ } else if (!std::strcmp(mediaType, "video")) {
+ nhlog::ui()->debug("WebRTC: received incoming video stream");
+ if (!session->getVideoItem()) {
+ g_free(mediaType);
+ nhlog::ui()->error("WebRTC: video call item not set");
+ return;
+ }
+ haveVideoStream_ = true;
+ keyFrameRequestData_.statsField =
+ std::string("rtp-inbound-stream-stats_") + std::to_string(ssrc);
+ queue = newVideoSinkChain(pipe);
+ auto videoCallSize = getResolution(newpad);
+ nhlog::ui()->info(
+ "WebRTC: incoming video resolution: {}x{}", videoCallSize.first, videoCallSize.second);
+ addLocalPiP(pipe, videoCallSize);
+ } else {
+ g_free(mediaType);
+ nhlog::ui()->error("WebRTC: unknown pad type: {}", GST_PAD_NAME(newpad));
+ return;
+ }
+
+ GstPad *queuepad = gst_element_get_static_pad(queue, "sink");
+ if (queuepad) {
+ if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
+ nhlog::ui()->error("WebRTC: unable to link new pad");
+ else {
+ if (session->callType() == CallType::VOICE ||
+ (haveAudioStream_ && (haveVideoStream_ || session->isRemoteVideoRecvOnly()))) {
+ emit session->stateChanged(State::CONNECTED);
+ if (haveVideoStream_) {
+ keyFrameRequestData_.pipe = pipe;
+ keyFrameRequestData_.decodebin = decodebin;
+ keyFrameRequestData_.timerid =
+ g_timeout_add_seconds(3, testPacketLoss, nullptr);
+ }
+ addRemotePiP(pipe);
+ if (session->isRemoteVideoRecvOnly())
+ addLocalVideo(pipe);
+ }
+ }
+ gst_object_unref(queuepad);
+ }
+ g_free(mediaType);
+}
+
+void
+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);
+ // hardware decoding needs investigation; eg rendering fails if vaapi plugin installed
+ g_object_set(decodebin, "force-sw-decoders", TRUE, nullptr);
+ g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
+ g_signal_connect(decodebin, "element-added", G_CALLBACK(setWaitForKeyFrame), nullptr);
+ 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 decodebin");
+ gst_object_unref(sinkpad);
+}
+
+bool
+contains(std::string_view str1, std::string_view str2)
+{
+ return std::search(str1.cbegin(),
+ str1.cend(),
+ str2.cbegin(),
+ str2.cend(),
+ [](unsigned char c1, unsigned char c2) {
+ return std::tolower(c1) == std::tolower(c2);
+ }) != str1.cend();
+}
+
+bool
+getMediaAttributes(const GstSDPMessage *sdp,
+ const char *mediaType,
+ const char *encoding,
+ int &payloadType,
+ bool &recvOnly,
+ bool &sendOnly)
+{
+ payloadType = -1;
+ recvOnly = false;
+ sendOnly = false;
+ for (guint mlineIndex = 0; mlineIndex < gst_sdp_message_medias_len(sdp); ++mlineIndex) {
+ const GstSDPMedia *media = gst_sdp_message_get_media(sdp, mlineIndex);
+ if (!std::strcmp(gst_sdp_media_get_media(media), mediaType)) {
+ recvOnly = gst_sdp_media_get_attribute_val(media, "recvonly") != nullptr;
+ sendOnly = gst_sdp_media_get_attribute_val(media, "sendonly") != nullptr;
+ const gchar *rtpval = nullptr;
+ for (guint n = 0; n == 0 || rtpval; ++n) {
+ rtpval = gst_sdp_media_get_attribute_val_n(media, "rtpmap", n);
+ if (rtpval && contains(rtpval, encoding)) {
+ payloadType = std::atoi(rtpval);
+ break;
+ }
+ }
+ return true;
+ }
+ }
+ return false;
+}
+}
+
+bool
+WebRTCSession::havePlugins(bool isVideo, std::string *errorMessage)
+{
+ if (!initialised_ && !init(errorMessage))
+ return false;
+ if (!isVideo && haveVoicePlugins_)
+ return true;
+ if (isVideo && haveVideoPlugins_)
+ return true;
+
+ const gchar *voicePlugins[] = {"audioconvert",
+ "audioresample",
+ "autodetect",
+ "dtls",
+ "nice",
+ "opus",
+ "playback",
+ "rtpmanager",
+ "srtp",
+ "volume",
+ "webrtc",
+ nullptr};
+
+ const gchar *videoPlugins[] = {
+ "compositor", "opengl", "qmlgl", "rtp", "videoconvert", "vpx", nullptr};
+
+ std::string strError("Missing GStreamer plugins: ");
+ const gchar **needed = isVideo ? videoPlugins : voicePlugins;
+ bool &havePlugins = isVideo ? haveVideoPlugins_ : haveVoicePlugins_;
+ havePlugins = true;
+ 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) {
+ havePlugins = false;
+ strError += std::string(needed[i]) + " ";
+ continue;
+ }
+ gst_object_unref(plugin);
+ }
+ if (!havePlugins) {
+ nhlog::ui()->error(strError);
+ if (errorMessage)
+ *errorMessage = strError;
+ return false;
+ }
+
+ if (isVideo) {
+ // load qmlglsink to register GStreamer's GstGLVideoItem QML type
+ GstElement *qmlglsink = gst_element_factory_make("qmlglsink", nullptr);
+ gst_object_unref(qmlglsink);
+ }
+ return true;
+}
+
+bool
+WebRTCSession::createOffer(CallType callType, uint32_t shareWindowId)
+{
+ clear();
+ isOffering_ = true;
+ callType_ = callType;
+ shareWindowId_ = shareWindowId;
+
+ // opus and vp8 rtp payload types must be defined dynamically
+ // therefore from the range [96-127]
+ // see for example https://tools.ietf.org/html/rfc7587
+ constexpr int opusPayloadType = 111;
+ constexpr int vp8PayloadType = 96;
+ return startPipeline(opusPayloadType, vp8PayloadType);
+}
+
+bool
+WebRTCSession::acceptOffer(const std::string &sdp)
+{
+ nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
+ if (state_ != State::DISCONNECTED)
+ return false;
+
+ clear();
+ GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
+ if (!offer)
+ return false;
+
+ int opusPayloadType;
+ bool recvOnly;
+ bool sendOnly;
+ if (getMediaAttributes(offer->sdp, "audio", "opus", opusPayloadType, recvOnly, sendOnly)) {
+ if (opusPayloadType == -1) {
+ nhlog::ui()->error("WebRTC: remote audio offer - no opus encoding");
+ gst_webrtc_session_description_free(offer);
+ return false;
+ }
+ } else {
+ nhlog::ui()->error("WebRTC: remote offer - no audio media");
+ gst_webrtc_session_description_free(offer);
+ return false;
+ }
+
+ int vp8PayloadType;
+ bool isVideo = getMediaAttributes(
+ offer->sdp, "video", "vp8", vp8PayloadType, isRemoteVideoRecvOnly_, isRemoteVideoSendOnly_);
+ if (isVideo && vp8PayloadType == -1) {
+ nhlog::ui()->error("WebRTC: remote video offer - no vp8 encoding");
+ gst_webrtc_session_description_free(offer);
+ return false;
+ }
+ callType_ = isVideo ? CallType::VIDEO : CallType::VOICE;
+
+ if (!startPipeline(opusPayloadType, vp8PayloadType)) {
+ gst_webrtc_session_description_free(offer);
+ return false;
+ }
+
+ // avoid a race that sometimes leaves the generated answer without media tracks (a=ssrc
+ // lines)
+ std::this_thread::sleep_for(std::chrono::milliseconds(200));
+
+ // 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;
+}
+
+bool
+WebRTCSession::acceptAnswer(const std::string &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) {
+ end();
+ return false;
+ }
+
+ if (callType_ != CallType::VOICE) {
+ int unused;
+ if (!getMediaAttributes(
+ answer->sdp, "video", "vp8", unused, isRemoteVideoRecvOnly_, isRemoteVideoSendOnly_))
+ isRemoteVideoRecvOnly_ = true;
+ }
+
+ 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)
+{
+ if (state_ >= State::INITIATED) {
+ for (const auto &c : candidates) {
+ nhlog::ui()->debug(
+ "WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
+ if (!c.candidate.empty()) {
+ g_signal_emit_by_name(
+ webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
+ }
+ }
+ }
+}
+
+bool
+WebRTCSession::startPipeline(int opusPayloadType, int vp8PayloadType)
+{
+ if (state_ != State::DISCONNECTED)
+ return false;
+
+ emit stateChanged(State::INITIATING);
+
+ if (!createPipeline(opusPayloadType, vp8PayloadType)) {
+ end();
+ return false;
+ }
+
+ webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
+
+ if (ChatPage::instance()->userSettings()->useStunServer()) {
+ nhlog::ui()->info("WebRTC: setting STUN server: {}", STUN_SERVER);
+ g_object_set(webrtc_, "stun-server", STUN_SERVER, nullptr);
+ }
+
+ 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");
+
+ // generate the offer when the pipeline goes to PLAYING
+ if (isOffering_)
+ g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(::createOffer), nullptr);
+
+ // on-ice-candidate is emitted when a local ICE candidate has been gathered
+ g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
+
+ // capture ICE failure
+ g_signal_connect(
+ webrtc_, "notify::ice-connection-state", G_CALLBACK(iceConnectionStateChanged), nullptr);
+
+ // incoming streams trigger pad-added
+ gst_element_set_state(pipe_, GST_STATE_READY);
+ g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
+
+ // capture ICE gathering completion
+ g_signal_connect(
+ webrtc_, "notify::ice-gathering-state", G_CALLBACK(iceGatheringStateChanged), nullptr);
+
+ // 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_));
+ busWatchId_ = gst_bus_add_watch(bus, newBusMessage, this);
+ gst_object_unref(bus);
+ emit stateChanged(State::INITIATED);
+ return true;
+}
+
+bool
+WebRTCSession::createPipeline(int opusPayloadType, int vp8PayloadType)
+{
+ GstDevice *device = devices_.audioDevice();
+ if (!device)
+ return false;
+
+ GstElement *source = gst_device_create_element(device, nullptr);
+ GstElement *volume = gst_element_factory_make("volume", "srclevel");
+ GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement *resample = gst_element_factory_make("audioresample", nullptr);
+ GstElement *queue1 = gst_element_factory_make("queue", nullptr);
+ GstElement *opusenc = gst_element_factory_make("opusenc", nullptr);
+ GstElement *rtp = gst_element_factory_make("rtpopuspay", nullptr);
+ GstElement *queue2 = gst_element_factory_make("queue", nullptr);
+ GstElement *capsfilter = gst_element_factory_make("capsfilter", nullptr);
+
+ GstCaps *rtpcaps = gst_caps_new_simple("application/x-rtp",
+ "media",
+ G_TYPE_STRING,
+ "audio",
+ "encoding-name",
+ G_TYPE_STRING,
+ "OPUS",
+ "payload",
+ G_TYPE_INT,
+ opusPayloadType,
+ nullptr);
+ g_object_set(capsfilter, "caps", rtpcaps, nullptr);
+ gst_caps_unref(rtpcaps);
+
+ GstElement *webrtcbin = gst_element_factory_make("webrtcbin", "webrtcbin");
+ g_object_set(webrtcbin, "bundle-policy", GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE, nullptr);
+
+ pipe_ = gst_pipeline_new(nullptr);
+ gst_bin_add_many(GST_BIN(pipe_),
+ source,
+ volume,
+ convert,
+ resample,
+ queue1,
+ opusenc,
+ rtp,
+ queue2,
+ capsfilter,
+ webrtcbin,
+ nullptr);
+
+ if (!gst_element_link_many(source,
+ volume,
+ convert,
+ resample,
+ queue1,
+ opusenc,
+ rtp,
+ queue2,
+ capsfilter,
+ webrtcbin,
+ nullptr)) {
+ nhlog::ui()->error("WebRTC: failed to link audio pipeline elements");
+ return false;
+ }
+
+ return callType_ == CallType::VOICE || isRemoteVideoSendOnly_
+ ? true
+ : addVideoPipeline(vp8PayloadType);
+}
+
+bool
+WebRTCSession::addVideoPipeline(int vp8PayloadType)
+{
+ // allow incoming video calls despite localUser having no webcam
+ if (callType_ == CallType::VIDEO && !devices_.haveCamera())
+ return !isOffering_;
+
+ auto settings = ChatPage::instance()->userSettings();
+ GstElement *camerafilter = nullptr;
+ GstElement *videoconvert = gst_element_factory_make("videoconvert", nullptr);
+ GstElement *tee = gst_element_factory_make("tee", "videosrctee");
+ gst_bin_add_many(GST_BIN(pipe_), videoconvert, tee, nullptr);
+ if (callType_ == CallType::VIDEO || (settings->screenSharePiP() && devices_.haveCamera())) {
+ std::pair<int, int> resolution;
+ std::pair<int, int> frameRate;
+ GstDevice *device = devices_.videoDevice(resolution, frameRate);
+ if (!device)
+ return false;
+
+ GstElement *camera = gst_device_create_element(device, nullptr);
+ GstCaps *caps = gst_caps_new_simple("video/x-raw",
+ "width",
+ G_TYPE_INT,
+ resolution.first,
+ "height",
+ G_TYPE_INT,
+ resolution.second,
+ "framerate",
+ GST_TYPE_FRACTION,
+ frameRate.first,
+ frameRate.second,
+ nullptr);
+ camerafilter = gst_element_factory_make("capsfilter", "camerafilter");
+ g_object_set(camerafilter, "caps", caps, nullptr);
+ gst_caps_unref(caps);
+
+ gst_bin_add_many(GST_BIN(pipe_), camera, camerafilter, nullptr);
+ if (!gst_element_link_many(camera, videoconvert, camerafilter, nullptr)) {
+ nhlog::ui()->error("WebRTC: failed to link camera elements");
+ return false;
+ }
+ if (callType_ == CallType::VIDEO && !gst_element_link(camerafilter, tee)) {
+ nhlog::ui()->error("WebRTC: failed to link camerafilter -> tee");
+ return false;
+ }
+ }
+
+ if (callType_ == CallType::SCREEN) {
+ nhlog::ui()->debug("WebRTC: screen share frame rate: {} fps",
+ settings->screenShareFrameRate());
+ nhlog::ui()->debug("WebRTC: screen share picture-in-picture: {}",
+ settings->screenSharePiP());
+ nhlog::ui()->debug("WebRTC: screen share request remote camera: {}",
+ settings->screenShareRemoteVideo());
+ nhlog::ui()->debug("WebRTC: screen share hide mouse cursor: {}",
+ settings->screenShareHideCursor());
+
+ GstElement *ximagesrc = gst_element_factory_make("ximagesrc", "screenshare");
+ if (!ximagesrc) {
+ nhlog::ui()->error("WebRTC: failed to create ximagesrc");
+ return false;
+ }
+ g_object_set(ximagesrc, "use-damage", FALSE, nullptr);
+ g_object_set(ximagesrc, "xid", shareWindowId_, nullptr);
+ g_object_set(ximagesrc, "show-pointer", !settings->screenShareHideCursor(), nullptr);
+
+ GstCaps *caps = gst_caps_new_simple("video/x-raw",
+ "framerate",
+ GST_TYPE_FRACTION,
+ settings->screenShareFrameRate(),
+ 1,
+ nullptr);
+ GstElement *capsfilter = gst_element_factory_make("capsfilter", nullptr);
+ g_object_set(capsfilter, "caps", caps, nullptr);
+ gst_caps_unref(caps);
+ gst_bin_add_many(GST_BIN(pipe_), ximagesrc, capsfilter, nullptr);
+
+ if (settings->screenSharePiP() && devices_.haveCamera()) {
+ GstElement *compositor = gst_element_factory_make("compositor", nullptr);
+ g_object_set(compositor, "background", 1, nullptr);
+ gst_bin_add(GST_BIN(pipe_), compositor);
+ if (!gst_element_link_many(ximagesrc, compositor, capsfilter, tee, nullptr)) {
+ nhlog::ui()->error("WebRTC: failed to link screen share elements");
+ return false;
+ }
+
+ GstPad *srcpad = gst_element_get_static_pad(camerafilter, "src");
+ remotePiPSinkPad_ = gst_element_get_request_pad(compositor, "sink_%u");
+ if (GST_PAD_LINK_FAILED(gst_pad_link(srcpad, remotePiPSinkPad_))) {
+ nhlog::ui()->error("WebRTC: failed to link camerafilter -> compositor");
+ gst_object_unref(srcpad);
+ return false;
+ }
+ gst_object_unref(srcpad);
+ } else if (!gst_element_link_many(ximagesrc, videoconvert, capsfilter, tee, nullptr)) {
+ nhlog::ui()->error("WebRTC: failed to link screen share elements");
+ return false;
+ }
+ }
+
+ GstElement *queue = gst_element_factory_make("queue", nullptr);
+ GstElement *vp8enc = gst_element_factory_make("vp8enc", nullptr);
+ g_object_set(vp8enc, "deadline", 1, nullptr);
+ g_object_set(vp8enc, "error-resilient", 1, nullptr);
+ GstElement *rtpvp8pay = gst_element_factory_make("rtpvp8pay", nullptr);
+ GstElement *rtpqueue = gst_element_factory_make("queue", nullptr);
+ GstElement *rtpcapsfilter = gst_element_factory_make("capsfilter", nullptr);
+ GstCaps *rtpcaps = gst_caps_new_simple("application/x-rtp",
+ "media",
+ G_TYPE_STRING,
+ "video",
+ "encoding-name",
+ G_TYPE_STRING,
+ "VP8",
+ "payload",
+ G_TYPE_INT,
+ vp8PayloadType,
+ nullptr);
+ g_object_set(rtpcapsfilter, "caps", rtpcaps, nullptr);
+ gst_caps_unref(rtpcaps);
+
+ gst_bin_add_many(GST_BIN(pipe_), queue, vp8enc, rtpvp8pay, rtpqueue, rtpcapsfilter, nullptr);
+
+ GstElement *webrtcbin = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
+ if (!gst_element_link_many(
+ tee, queue, vp8enc, rtpvp8pay, rtpqueue, rtpcapsfilter, webrtcbin, nullptr)) {
+ nhlog::ui()->error("WebRTC: failed to link rtp video elements");
+ gst_object_unref(webrtcbin);
+ return false;
+ }
+
+ if (callType_ == CallType::SCREEN &&
+ !ChatPage::instance()->userSettings()->screenShareRemoteVideo()) {
+ GArray *transceivers;
+ g_signal_emit_by_name(webrtcbin, "get-transceivers", &transceivers);
+ GstWebRTCRTPTransceiver *transceiver =
+ g_array_index(transceivers, GstWebRTCRTPTransceiver *, 1);
+ transceiver->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
+ g_array_unref(transceivers);
+ }
+
+ gst_object_unref(webrtcbin);
+ return true;
+}
+
+bool
+WebRTCSession::haveLocalPiP() const
+{
+ if (state_ >= State::INITIATED) {
+ if (callType_ == CallType::VOICE || isRemoteVideoRecvOnly_)
+ return false;
+ else if (callType_ == CallType::SCREEN)
+ return true;
+ else {
+ GstElement *tee = gst_bin_get_by_name(GST_BIN(pipe_), "videosrctee");
+ if (tee) {
+ gst_object_unref(tee);
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+bool
+WebRTCSession::isMicMuted() const
+{
+ if (state_ < State::INITIATED)
+ return false;
+
+ GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
+ gboolean muted;
+ g_object_get(srclevel, "mute", &muted, nullptr);
+ gst_object_unref(srclevel);
+ return muted;
+}
+
+bool
+WebRTCSession::toggleMicMute()
+{
+ if (state_ < State::INITIATED)
+ return false;
+
+ GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
+ gboolean muted;
+ g_object_get(srclevel, "mute", &muted, nullptr);
+ g_object_set(srclevel, "mute", !muted, nullptr);
+ gst_object_unref(srclevel);
+ return !muted;
+}
+
+void
+WebRTCSession::toggleLocalPiP()
+{
+ if (localPiPSinkPad_) {
+ guint zorder;
+ g_object_get(localPiPSinkPad_, "zorder", &zorder, nullptr);
+ g_object_set(localPiPSinkPad_, "zorder", zorder ? 0 : 2, nullptr);
+ }
+}
+
+void
+WebRTCSession::clear()
+{
+ callType_ = webrtc::CallType::VOICE;
+ isOffering_ = false;
+ isRemoteVideoRecvOnly_ = false;
+ isRemoteVideoSendOnly_ = false;
+ videoItem_ = nullptr;
+ pipe_ = nullptr;
+ webrtc_ = nullptr;
+ busWatchId_ = 0;
+ shareWindowId_ = 0;
+ haveAudioStream_ = false;
+ haveVideoStream_ = false;
+ localPiPSinkPad_ = nullptr;
+ remotePiPSinkPad_ = nullptr;
+ localsdp_.clear();
+ localcandidates_.clear();
+}
+
+void
+WebRTCSession::end()
+{
+ nhlog::ui()->debug("WebRTC: ending session");
+ keyFrameRequestData_ = KeyFrameRequestData{};
+ if (pipe_) {
+ gst_element_set_state(pipe_, GST_STATE_NULL);
+ gst_object_unref(pipe_);
+ pipe_ = nullptr;
+ if (busWatchId_) {
+ g_source_remove(busWatchId_);
+ busWatchId_ = 0;
+ }
+ }
+
+ clear();
+ if (state_ != State::DISCONNECTED)
+ emit stateChanged(State::DISCONNECTED);
+}
+
+#else
+
+bool
+WebRTCSession::havePlugins(bool, std::string *)
+{
+ return false;
+}
+
+bool
+WebRTCSession::haveLocalPiP() const
+{
+ return false;
+}
+
+bool WebRTCSession::createOffer(webrtc::CallType, uint32_t) { return false; }
+
+bool
+WebRTCSession::acceptOffer(const std::string &)
+{
+ return false;
+}
+
+bool
+WebRTCSession::acceptAnswer(const std::string &)
+{
+ return false;
+}
+
+void
+WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &)
+{}
+
+bool
+WebRTCSession::isMicMuted() const
+{
+ return false;
+}
+
+bool
+WebRTCSession::toggleMicMute()
+{
+ return false;
+}
+
+void
+WebRTCSession::toggleLocalPiP()
+{}
+
+void
+WebRTCSession::end()
+{}
+
+#endif
|
