1187 lines
46 KiB
C++
1187 lines
46 KiB
C++
// SPDX-FileCopyrightText: 2021 Nheko Contributors
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//
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// SPDX-License-Identifier: GPL-3.0-or-later
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#include <QQmlEngine>
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#include <QQuickItem>
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#include <algorithm>
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#include <cctype>
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#include <chrono>
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#include <cstdlib>
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#include <cstring>
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#include <optional>
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#include <string_view>
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#include <thread>
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#include <utility>
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#include "CallDevices.h"
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#include "ChatPage.h"
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#include "Logging.h"
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#include "UserSettingsPage.h"
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#include "WebRTCSession.h"
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#ifdef GSTREAMER_AVAILABLE
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extern "C"
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{
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#include "gst/gst.h"
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#include "gst/sdp/sdp.h"
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#define GST_USE_UNSTABLE_API
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#include "gst/webrtc/webrtc.h"
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}
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#endif
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// https://github.com/vector-im/riot-web/issues/10173
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#define STUN_SERVER "stun://turn.matrix.org:3478"
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Q_DECLARE_METATYPE(webrtc::CallType)
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Q_DECLARE_METATYPE(webrtc::State)
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using webrtc::CallType;
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using webrtc::State;
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WebRTCSession::WebRTCSession()
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: QObject()
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, devices_(CallDevices::instance())
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{
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qRegisterMetaType<webrtc::CallType>();
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qmlRegisterUncreatableMetaObject(
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webrtc::staticMetaObject, "im.nheko", 1, 0, "CallType", "Can't instantiate enum");
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qRegisterMetaType<webrtc::State>();
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qmlRegisterUncreatableMetaObject(
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webrtc::staticMetaObject, "im.nheko", 1, 0, "WebRTCState", "Can't instantiate enum");
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connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
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init();
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}
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bool
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WebRTCSession::init(std::string *errorMessage)
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{
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#ifdef GSTREAMER_AVAILABLE
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if (initialised_)
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return true;
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GError *error = nullptr;
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if (!gst_init_check(nullptr, nullptr, &error)) {
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std::string strError("WebRTC: failed to initialise GStreamer: ");
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if (error) {
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strError += error->message;
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g_error_free(error);
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}
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nhlog::ui()->error(strError);
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if (errorMessage)
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*errorMessage = strError;
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return false;
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}
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initialised_ = true;
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gchar *version = gst_version_string();
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nhlog::ui()->info("WebRTC: initialised {}", version);
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g_free(version);
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devices_.init();
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return true;
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#else
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(void)errorMessage;
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return false;
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#endif
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}
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#ifdef GSTREAMER_AVAILABLE
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namespace {
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std::string localsdp_;
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std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
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bool haveAudioStream_ = false;
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bool haveVideoStream_ = false;
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GstPad *localPiPSinkPad_ = nullptr;
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GstPad *remotePiPSinkPad_ = nullptr;
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gboolean
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newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
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{
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WebRTCSession *session = static_cast<WebRTCSession *>(user_data);
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switch (GST_MESSAGE_TYPE(msg)) {
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case GST_MESSAGE_EOS:
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nhlog::ui()->error("WebRTC: end of stream");
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session->end();
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break;
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case GST_MESSAGE_ERROR:
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GError *error;
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gchar *debug;
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gst_message_parse_error(msg, &error, &debug);
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nhlog::ui()->error(
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"WebRTC: error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
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g_clear_error(&error);
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g_free(debug);
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session->end();
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break;
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default:
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break;
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}
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return TRUE;
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}
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GstWebRTCSessionDescription *
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parseSDP(const std::string &sdp, GstWebRTCSDPType type)
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{
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GstSDPMessage *msg;
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gst_sdp_message_new(&msg);
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if (gst_sdp_message_parse_buffer((guint8 *)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
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return gst_webrtc_session_description_new(type, msg);
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} else {
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nhlog::ui()->error("WebRTC: failed to parse remote session description");
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gst_sdp_message_free(msg);
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return nullptr;
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}
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}
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void
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setLocalDescription(GstPromise *promise, gpointer webrtc)
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{
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const GstStructure *reply = gst_promise_get_reply(promise);
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gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
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GstWebRTCSessionDescription *gstsdp = nullptr;
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gst_structure_get(reply,
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isAnswer ? "answer" : "offer",
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GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
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&gstsdp,
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nullptr);
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gst_promise_unref(promise);
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g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
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gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
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localsdp_ = std::string(sdp);
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g_free(sdp);
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gst_webrtc_session_description_free(gstsdp);
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nhlog::ui()->debug(
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"WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
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}
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void
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createOffer(GstElement *webrtc)
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{
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// create-offer first, then set-local-description
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GstPromise *promise =
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gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
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g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
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}
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void
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createAnswer(GstPromise *promise, gpointer webrtc)
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{
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// create-answer first, then set-local-description
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gst_promise_unref(promise);
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promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
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g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
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}
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void
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iceGatheringStateChanged(GstElement *webrtc,
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GParamSpec *pspec G_GNUC_UNUSED,
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gpointer user_data G_GNUC_UNUSED)
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{
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GstWebRTCICEGatheringState newState;
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g_object_get(webrtc, "ice-gathering-state", &newState, nullptr);
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if (newState == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
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nhlog::ui()->debug("WebRTC: GstWebRTCICEGatheringState -> Complete");
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if (WebRTCSession::instance().isOffering()) {
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emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
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emit WebRTCSession::instance().stateChanged(State::OFFERSENT);
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} else {
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emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
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emit WebRTCSession::instance().stateChanged(State::ANSWERSENT);
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}
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}
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}
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void
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addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
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guint mlineIndex,
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gchar *candidate,
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gpointer G_GNUC_UNUSED)
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{
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nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
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localcandidates_.push_back({std::string() /*max-bundle*/, (uint16_t)mlineIndex, candidate});
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}
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void
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iceConnectionStateChanged(GstElement *webrtc,
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GParamSpec *pspec G_GNUC_UNUSED,
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gpointer user_data G_GNUC_UNUSED)
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{
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GstWebRTCICEConnectionState newState;
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g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
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switch (newState) {
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case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
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nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
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emit WebRTCSession::instance().stateChanged(State::CONNECTING);
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break;
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case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
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nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
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emit WebRTCSession::instance().stateChanged(State::ICEFAILED);
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break;
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default:
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break;
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}
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}
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// https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1164
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struct KeyFrameRequestData
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{
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GstElement *pipe = nullptr;
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GstElement *decodebin = nullptr;
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gint packetsLost = 0;
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guint timerid = 0;
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std::string statsField;
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} keyFrameRequestData_;
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void
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sendKeyFrameRequest()
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{
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GstPad *sinkpad = gst_element_get_static_pad(keyFrameRequestData_.decodebin, "sink");
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if (!gst_pad_push_event(sinkpad,
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gst_event_new_custom(GST_EVENT_CUSTOM_UPSTREAM,
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gst_structure_new_empty("GstForceKeyUnit"))))
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nhlog::ui()->error("WebRTC: key frame request failed");
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else
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nhlog::ui()->debug("WebRTC: sent key frame request");
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gst_object_unref(sinkpad);
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}
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void
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testPacketLoss_(GstPromise *promise, gpointer G_GNUC_UNUSED)
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{
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const GstStructure *reply = gst_promise_get_reply(promise);
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gint packetsLost = 0;
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GstStructure *rtpStats;
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if (!gst_structure_get(reply,
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keyFrameRequestData_.statsField.c_str(),
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GST_TYPE_STRUCTURE,
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&rtpStats,
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nullptr)) {
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nhlog::ui()->error("WebRTC: get-stats: no field: {}",
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keyFrameRequestData_.statsField);
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gst_promise_unref(promise);
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return;
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}
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gst_structure_get_int(rtpStats, "packets-lost", &packetsLost);
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gst_structure_free(rtpStats);
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gst_promise_unref(promise);
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if (packetsLost > keyFrameRequestData_.packetsLost) {
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nhlog::ui()->debug("WebRTC: inbound video lost packet count: {}", packetsLost);
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keyFrameRequestData_.packetsLost = packetsLost;
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sendKeyFrameRequest();
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}
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}
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gboolean
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testPacketLoss(gpointer G_GNUC_UNUSED)
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{
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if (keyFrameRequestData_.pipe) {
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GstElement *webrtc =
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gst_bin_get_by_name(GST_BIN(keyFrameRequestData_.pipe), "webrtcbin");
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GstPromise *promise =
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gst_promise_new_with_change_func(testPacketLoss_, nullptr, nullptr);
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g_signal_emit_by_name(webrtc, "get-stats", nullptr, promise);
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gst_object_unref(webrtc);
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return TRUE;
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}
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return FALSE;
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}
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void
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setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointer G_GNUC_UNUSED)
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{
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if (!std::strcmp(
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gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(gst_element_get_factory(element))),
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"rtpvp8depay"))
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g_object_set(element, "wait-for-keyframe", TRUE, nullptr);
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}
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GstElement *
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newAudioSinkChain(GstElement *pipe)
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{
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GstElement *queue = gst_element_factory_make("queue", nullptr);
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GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
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GstElement *resample = gst_element_factory_make("audioresample", nullptr);
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GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
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gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
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gst_element_link_many(queue, convert, resample, sink, nullptr);
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gst_element_sync_state_with_parent(queue);
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gst_element_sync_state_with_parent(convert);
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gst_element_sync_state_with_parent(resample);
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gst_element_sync_state_with_parent(sink);
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return queue;
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}
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GstElement *
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newVideoSinkChain(GstElement *pipe)
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{
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// use compositor for now; acceleration needs investigation
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GstElement *queue = gst_element_factory_make("queue", nullptr);
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GstElement *compositor = gst_element_factory_make("compositor", "compositor");
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GstElement *glupload = gst_element_factory_make("glupload", nullptr);
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GstElement *glcolorconvert = gst_element_factory_make("glcolorconvert", nullptr);
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GstElement *qmlglsink = gst_element_factory_make("qmlglsink", nullptr);
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GstElement *glsinkbin = gst_element_factory_make("glsinkbin", nullptr);
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g_object_set(compositor, "background", 1, nullptr);
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g_object_set(qmlglsink, "widget", WebRTCSession::instance().getVideoItem(), nullptr);
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g_object_set(glsinkbin, "sink", qmlglsink, nullptr);
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gst_bin_add_many(
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GST_BIN(pipe), queue, compositor, glupload, glcolorconvert, glsinkbin, nullptr);
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gst_element_link_many(queue, compositor, glupload, glcolorconvert, glsinkbin, nullptr);
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gst_element_sync_state_with_parent(queue);
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gst_element_sync_state_with_parent(compositor);
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gst_element_sync_state_with_parent(glupload);
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gst_element_sync_state_with_parent(glcolorconvert);
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gst_element_sync_state_with_parent(glsinkbin);
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return queue;
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}
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std::pair<int, int>
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getResolution(GstPad *pad)
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{
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std::pair<int, int> ret;
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GstCaps *caps = gst_pad_get_current_caps(pad);
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const GstStructure *s = gst_caps_get_structure(caps, 0);
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gst_structure_get_int(s, "width", &ret.first);
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gst_structure_get_int(s, "height", &ret.second);
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gst_caps_unref(caps);
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return ret;
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}
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std::pair<int, int>
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getResolution(GstElement *pipe, const gchar *elementName, const gchar *padName)
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{
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GstElement *element = gst_bin_get_by_name(GST_BIN(pipe), elementName);
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GstPad *pad = gst_element_get_static_pad(element, padName);
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auto ret = getResolution(pad);
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gst_object_unref(pad);
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gst_object_unref(element);
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return ret;
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}
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std::pair<int, int>
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getPiPDimensions(const std::pair<int, int> &resolution, int fullWidth, double scaleFactor)
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{
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int pipWidth = fullWidth * scaleFactor;
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int pipHeight = static_cast<double>(resolution.second) / resolution.first * pipWidth;
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return {pipWidth, pipHeight};
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}
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void
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addLocalPiP(GstElement *pipe, const std::pair<int, int> &videoCallSize)
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{
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// embed localUser's camera into received video (CallType::VIDEO)
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// OR embed screen share into received video (CallType::SCREEN)
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GstElement *tee = gst_bin_get_by_name(GST_BIN(pipe), "videosrctee");
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if (!tee)
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return;
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GstElement *queue = gst_element_factory_make("queue", nullptr);
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gst_bin_add(GST_BIN(pipe), queue);
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gst_element_link(tee, queue);
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gst_element_sync_state_with_parent(queue);
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gst_object_unref(tee);
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GstElement *compositor = gst_bin_get_by_name(GST_BIN(pipe), "compositor");
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localPiPSinkPad_ = gst_element_get_request_pad(compositor, "sink_%u");
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g_object_set(localPiPSinkPad_, "zorder", 2, nullptr);
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bool isVideo = WebRTCSession::instance().callType() == CallType::VIDEO;
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const gchar *element = isVideo ? "camerafilter" : "screenshare";
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const gchar *pad = isVideo ? "sink" : "src";
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auto resolution = getResolution(pipe, element, pad);
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auto pipSize = getPiPDimensions(resolution, videoCallSize.first, 0.25);
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nhlog::ui()->debug(
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"WebRTC: local picture-in-picture: {}x{}", pipSize.first, pipSize.second);
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g_object_set(localPiPSinkPad_, "width", pipSize.first, "height", pipSize.second, nullptr);
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gint offset = videoCallSize.first / 80;
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g_object_set(localPiPSinkPad_, "xpos", offset, "ypos", offset, nullptr);
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GstPad *srcpad = gst_element_get_static_pad(queue, "src");
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if (GST_PAD_LINK_FAILED(gst_pad_link(srcpad, localPiPSinkPad_)))
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nhlog::ui()->error("WebRTC: failed to link local PiP elements");
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gst_object_unref(srcpad);
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gst_object_unref(compositor);
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}
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void
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addRemotePiP(GstElement *pipe)
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{
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// embed localUser's camera into screen image being shared
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if (remotePiPSinkPad_) {
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auto camRes = getResolution(pipe, "camerafilter", "sink");
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auto shareRes = getResolution(pipe, "screenshare", "src");
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auto pipSize = getPiPDimensions(camRes, shareRes.first, 0.2);
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nhlog::ui()->debug(
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"WebRTC: screen share picture-in-picture: {}x{}", pipSize.first, pipSize.second);
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gint offset = shareRes.first / 100;
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g_object_set(remotePiPSinkPad_, "zorder", 2, nullptr);
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g_object_set(
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remotePiPSinkPad_, "width", pipSize.first, "height", pipSize.second, nullptr);
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g_object_set(remotePiPSinkPad_,
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"xpos",
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shareRes.first - pipSize.first - offset,
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"ypos",
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shareRes.second - pipSize.second - offset,
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nullptr);
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}
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}
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void
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addLocalVideo(GstElement *pipe)
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{
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GstElement *queue = newVideoSinkChain(pipe);
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GstElement *tee = gst_bin_get_by_name(GST_BIN(pipe), "videosrctee");
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GstPad *srcpad = gst_element_get_request_pad(tee, "src_%u");
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GstPad *sinkpad = gst_element_get_static_pad(queue, "sink");
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if (GST_PAD_LINK_FAILED(gst_pad_link(srcpad, sinkpad)))
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nhlog::ui()->error("WebRTC: failed to link videosrctee -> video sink chain");
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gst_object_unref(srcpad);
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}
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void
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linkNewPad(GstElement *decodebin, GstPad *newpad, GstElement *pipe)
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{
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GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
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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
|