mirror of
https://github.com/Nheko-Reborn/nheko.git
synced 2024-10-31 10:00:46 +03:00
479 lines
15 KiB
C++
479 lines
15 KiB
C++
#include <cctype>
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#include "WebRTCSession.h"
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#include "Logging.h"
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extern "C" {
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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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Q_DECLARE_METATYPE(WebRTCSession::State)
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namespace {
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bool gisoffer;
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std::string glocalsdp;
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std::vector<mtx::events::msg::CallCandidates::Candidate> gcandidates;
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gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data);
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GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type);
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void generateOffer(GstElement *webrtc);
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void setLocalDescription(GstPromise *promise, gpointer webrtc);
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void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED);
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gboolean onICEGatheringCompletion(gpointer timerid);
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void createAnswer(GstPromise *promise, gpointer webrtc);
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void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
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void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
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std::string::const_iterator findName(const std::string &sdp, const std::string &name);
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int getPayloadType(const std::string &sdp, const std::string &name);
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}
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WebRTCSession::WebRTCSession() : QObject()
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{
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qRegisterMetaType<WebRTCSession::State>();
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connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
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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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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 = std::string("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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gchar *version = gst_version_string();
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std::string gstVersion(version);
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g_free(version);
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nhlog::ui()->info("WebRTC: initialised " + gstVersion);
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// GStreamer Plugins:
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// Base: audioconvert, audioresample, opus, playback, volume
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// Good: autodetect, rtpmanager, vpx
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// Bad: dtls, srtp, webrtc
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// libnice [GLib]: nice
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initialised_ = true;
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std::string strError = gstVersion + ": Missing plugins: ";
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const gchar *needed[] = {"audioconvert", "audioresample", "autodetect", "dtls", "nice",
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"opus", "playback", "rtpmanager", "srtp", "vpx", "volume", "webrtc", nullptr};
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GstRegistry *registry = gst_registry_get();
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for (guint i = 0; i < g_strv_length((gchar**)needed); i++) {
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GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
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if (!plugin) {
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strError += needed[i];
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initialised_ = false;
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continue;
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}
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gst_object_unref(plugin);
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}
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if (!initialised_) {
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nhlog::ui()->error(strError);
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if (errorMessage)
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*errorMessage = strError;
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}
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return initialised_;
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}
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bool
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WebRTCSession::createOffer()
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{
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gisoffer = true;
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glocalsdp.clear();
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gcandidates.clear();
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return startPipeline(111); // a dynamic opus payload type
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}
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bool
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WebRTCSession::acceptOffer(const std::string &sdp)
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{
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nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
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if (state_ != State::DISCONNECTED)
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return false;
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gisoffer = false;
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glocalsdp.clear();
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gcandidates.clear();
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int opusPayloadType = getPayloadType(sdp, "opus");
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if (opusPayloadType == -1)
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return false;
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GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
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if (!offer)
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return false;
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if (!startPipeline(opusPayloadType)) {
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gst_webrtc_session_description_free(offer);
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return false;
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}
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// set-remote-description first, then create-answer
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GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
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g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
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gst_webrtc_session_description_free(offer);
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return true;
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}
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bool
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WebRTCSession::startPipeline(int opusPayloadType)
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{
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if (state_ != State::DISCONNECTED)
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return false;
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emit stateChanged(State::INITIATING);
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if (!createPipeline(opusPayloadType))
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return false;
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webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
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if (!stunServer_.empty()) {
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nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
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g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
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}
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for (const auto &uri : turnServers_) {
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nhlog::ui()->info("WebRTC: setting TURN server: {}", uri);
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gboolean udata;
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g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
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}
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// generate the offer when the pipeline goes to PLAYING
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if (gisoffer)
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g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr);
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// on-ice-candidate is emitted when a local ICE candidate has been gathered
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g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
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// incoming streams trigger pad-added
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gst_element_set_state(pipe_, GST_STATE_READY);
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g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
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// webrtcbin lifetime is the same as that of the pipeline
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gst_object_unref(webrtc_);
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// start the pipeline
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GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
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if (ret == GST_STATE_CHANGE_FAILURE) {
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nhlog::ui()->error("WebRTC: unable to start pipeline");
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end();
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return false;
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}
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GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
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gst_bus_add_watch(bus, newBusMessage, this);
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gst_object_unref(bus);
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emit stateChanged(State::INITIATED);
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return true;
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}
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#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
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bool
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WebRTCSession::createPipeline(int opusPayloadType)
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{
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std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
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"autoaudiosrc ! volume name=srclevel ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
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"queue ! " RTP_CAPS_OPUS + std::to_string(opusPayloadType) + " ! webrtcbin.");
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webrtc_ = nullptr;
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GError *error = nullptr;
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pipe_ = gst_parse_launch(pipeline.c_str(), &error);
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if (error) {
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nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
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g_error_free(error);
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end();
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return false;
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}
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return true;
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}
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bool
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WebRTCSession::acceptAnswer(const std::string &sdp)
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{
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nhlog::ui()->debug("WebRTC: received answer:\n{}", sdp);
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if (state_ != State::OFFERSENT)
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return false;
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GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
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if (!answer) {
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end();
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return false;
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}
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g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
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gst_webrtc_session_description_free(answer);
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return true;
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}
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void
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WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
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{
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if (state_ >= State::INITIATED) {
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for (const auto &c : candidates) {
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nhlog::ui()->debug("WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
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g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
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}
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if (state_ == State::OFFERSENT)
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emit stateChanged(State::CONNECTING);
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}
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}
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bool
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WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
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{
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if (state_ < State::INITIATED)
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return false;
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GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
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if (!srclevel)
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return false;
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gboolean muted;
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g_object_get(srclevel, "mute", &muted, nullptr);
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g_object_set(srclevel, "mute", !muted, nullptr);
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gst_object_unref(srclevel);
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isMuted = !muted;
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return true;
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}
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void
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WebRTCSession::end()
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{
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nhlog::ui()->debug("WebRTC: ending session");
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if (pipe_) {
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gst_element_set_state(pipe_, GST_STATE_NULL);
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gst_object_unref(pipe_);
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pipe_ = nullptr;
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}
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webrtc_ = nullptr;
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if (state_ != State::DISCONNECTED)
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emit stateChanged(State::DISCONNECTED);
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}
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namespace {
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std::string::const_iterator findName(const std::string &sdp, const std::string &name)
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{
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return std::search(sdp.cbegin(), sdp.cend(), name.cbegin(), name.cend(),
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[](unsigned char c1, unsigned char c2) {return std::tolower(c1) == std::tolower(c2);});
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}
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int getPayloadType(const std::string &sdp, const std::string &name)
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{
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// eg a=rtpmap:111 opus/48000/2
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auto e = findName(sdp, name);
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if (e == sdp.cend()) {
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nhlog::ui()->error("WebRTC: remote offer - " + name + " attribute missing");
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return -1;
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}
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if (auto s = sdp.rfind(':', e - sdp.cbegin()); s == std::string::npos) {
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nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
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return -1;
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}
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else {
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++s;
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try {
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return std::stoi(std::string(sdp, s, e - sdp.cbegin() - s));
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}
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catch(...) {
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nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
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}
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}
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return -1;
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}
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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 = (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("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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}
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else {
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nhlog::ui()->error("WebRTC: failed to parse remote session description");
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gst_object_unref(msg);
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return nullptr;
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}
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}
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void
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generateOffer(GstElement *webrtc)
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{
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// create-offer first, then set-local-description
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GstPromise *promise = 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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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, isAnswer ? "answer" : "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &gstsdp, 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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glocalsdp = 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("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", glocalsdp);
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}
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void
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addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, 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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if (WebRTCSession::instance().state() == WebRTCSession::State::CONNECTED) {
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emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
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return;
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}
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gcandidates.push_back({"audio", (uint16_t)mlineIndex, candidate});
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// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
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// fixed in v1.18
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// use a 100ms timeout in the meantime
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static guint timerid = 0;
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if (timerid)
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g_source_remove(timerid);
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timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
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}
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gboolean
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onICEGatheringCompletion(gpointer timerid)
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{
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*(guint*)(timerid) = 0;
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if (gisoffer) {
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emit WebRTCSession::instance().offerCreated(glocalsdp, gcandidates);
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
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}
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else {
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emit WebRTCSession::instance().answerCreated(glocalsdp, gcandidates);
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emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
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}
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return FALSE;
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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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addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
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{
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if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
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return;
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nhlog::ui()->debug("WebRTC: received incoming stream");
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GstElement *decodebin = gst_element_factory_make("decodebin", nullptr);
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g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
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gst_bin_add(GST_BIN(pipe), decodebin);
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gst_element_sync_state_with_parent(decodebin);
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GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
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if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad)))
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nhlog::ui()->error("WebRTC: unable to link new pad");
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gst_object_unref(sinkpad);
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}
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void
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linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
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{
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GstCaps *caps = gst_pad_get_current_caps(newpad);
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if (!caps)
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return;
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const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
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gst_caps_unref(caps);
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GstPad *queuepad = nullptr;
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GstElement *queue = gst_element_factory_make("queue", nullptr);
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if (g_str_has_prefix(name, "audio")) {
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nhlog::ui()->debug("WebRTC: received incoming audio stream");
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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_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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gst_element_link_many(queue, convert, resample, sink, nullptr);
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queuepad = gst_element_get_static_pad(queue, "sink");
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}
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else if (g_str_has_prefix(name, "video")) {
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nhlog::ui()->debug("WebRTC: received incoming video stream");
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GstElement *convert = gst_element_factory_make("videoconvert", nullptr);
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GstElement *sink = gst_element_factory_make("autovideosink", nullptr);
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gst_bin_add_many(GST_BIN(pipe), queue, convert, sink, nullptr);
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gst_element_sync_state_with_parent(queue);
|
|
gst_element_sync_state_with_parent(convert);
|
|
gst_element_sync_state_with_parent(sink);
|
|
gst_element_link_many(queue, convert, sink, nullptr);
|
|
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 {
|
|
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED);
|
|
}
|
|
gst_object_unref(queuepad);
|
|
}
|
|
}
|
|
|
|
}
|