clang-format
This commit is contained in:
trilene
parent
f14d141cb5
commit
e3e7595bab
13 changed files with 918 additions and 852 deletions
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@ -1,9 +1,10 @@
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#include <cctype>
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#include "WebRTCSession.h"
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#include "Logging.h"
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#include "WebRTCSession.h"
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extern "C" {
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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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@ -13,180 +14,445 @@ extern "C" {
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Q_DECLARE_METATYPE(WebRTCSession::State)
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namespace {
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bool isoffering_;
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std::string localsdp_;
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std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
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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 iceConnectionStateChanged(GstElement *webrtcbin, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED);
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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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WebRTCSession::WebRTCSession()
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: 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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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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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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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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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
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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", "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 += std::string(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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// GStreamer Plugins:
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// Base: audioconvert, audioresample, opus, playback, volume
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// Good: autodetect, rtpmanager
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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",
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"audioresample",
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"autodetect",
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"dtls",
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"nice",
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"opus",
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"playback",
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"rtpmanager",
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"srtp",
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"volume",
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"webrtc",
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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 += std::string(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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namespace {
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bool isoffering_;
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std::string localsdp_;
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std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
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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_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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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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gboolean
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onICEGatheringCompletion(gpointer timerid)
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{
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*(guint *)(timerid) = 0;
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if (isoffering_) {
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emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
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emit WebRTCSession::instance().stateChanged(WebRTCSession::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(WebRTCSession::State::ANSWERSENT);
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}
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return FALSE;
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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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if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
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emit WebRTCSession::instance().newICECandidate(
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{"audio", (uint16_t)mlineIndex, candidate});
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return;
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}
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localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
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// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
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// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18. Use a 100ms timeout in
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// 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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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(WebRTCSession::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(WebRTCSession::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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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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if (g_str_has_prefix(name, "audio")) {
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nhlog::ui()->debug("WebRTC: received incoming audio stream");
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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_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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if (queuepad) {
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if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
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nhlog::ui()->error("WebRTC: unable to link new pad");
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else {
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emit WebRTCSession::instance().stateChanged(
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WebRTCSession::State::CONNECTED);
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}
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gst_object_unref(queuepad);
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}
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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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std::string::const_iterator
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findName(const std::string &sdp, const std::string &name)
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{
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return std::search(
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sdp.cbegin(),
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sdp.cend(),
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name.cbegin(),
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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
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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 +
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" payload type");
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return -1;
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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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} catch (...) {
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nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name +
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" payload type");
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}
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}
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return -1;
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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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isoffering_ = true;
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localsdp_.clear();
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localcandidates_.clear();
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return startPipeline(111); // a dynamic opus payload type
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isoffering_ = true;
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localsdp_.clear();
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localcandidates_.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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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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isoffering_ = false;
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localsdp_.clear();
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localcandidates_.clear();
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isoffering_ = false;
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localsdp_.clear();
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localcandidates_.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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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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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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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);
|
||||
g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
|
||||
gst_webrtc_session_description_free(offer);
|
||||
return true;
|
||||
// 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;
|
||||
}
|
||||
|
||||
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);
|
||||
g_signal_emit_by_name(
|
||||
webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
bool
|
||||
WebRTCSession::startPipeline(int opusPayloadType)
|
||||
{
|
||||
if (state_ != State::DISCONNECTED)
|
||||
return false;
|
||||
if (state_ != State::DISCONNECTED)
|
||||
return false;
|
||||
|
||||
emit stateChanged(State::INITIATING);
|
||||
emit stateChanged(State::INITIATING);
|
||||
|
||||
if (!createPipeline(opusPayloadType))
|
||||
return false;
|
||||
if (!createPipeline(opusPayloadType))
|
||||
return false;
|
||||
|
||||
webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
|
||||
webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
|
||||
|
||||
if (!stunServer_.empty()) {
|
||||
nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
|
||||
g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
|
||||
}
|
||||
if (!stunServer_.empty()) {
|
||||
nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
|
||||
g_object_set(webrtc_, "stun-server", stunServer_.c_str(), 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");
|
||||
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(generateOffer), nullptr);
|
||||
// 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);
|
||||
// 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);
|
||||
// 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_);
|
||||
// incoming streams trigger pad-added
|
||||
gst_element_set_state(pipe_, GST_STATE_READY);
|
||||
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
|
||||
|
||||
// webrtcbin lifetime is the same as that of the pipeline
|
||||
gst_object_unref(webrtc_);
|
||||
// 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;
|
||||
}
|
||||
// start the pipeline
|
||||
GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
|
||||
if (ret == GST_STATE_CHANGE_FAILURE) {
|
||||
nhlog::ui()->error("WebRTC: unable to start pipeline");
|
||||
end();
|
||||
return false;
|
||||
}
|
||||
|
||||
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
|
||||
gst_bus_add_watch(bus, newBusMessage, this);
|
||||
gst_object_unref(bus);
|
||||
emit stateChanged(State::INITIATED);
|
||||
return true;
|
||||
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
|
||||
gst_bus_add_watch(bus, newBusMessage, this);
|
||||
gst_object_unref(bus);
|
||||
emit stateChanged(State::INITIATED);
|
||||
return true;
|
||||
}
|
||||
|
||||
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
|
||||
|
|
@ -194,297 +460,52 @@ WebRTCSession::startPipeline(int opusPayloadType)
|
|||
bool
|
||||
WebRTCSession::createPipeline(int opusPayloadType)
|
||||
{
|
||||
std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
|
||||
"autoaudiosrc ! volume name=srclevel ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
|
||||
"queue ! " RTP_CAPS_OPUS + std::to_string(opusPayloadType) + " ! webrtcbin.");
|
||||
std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
|
||||
"autoaudiosrc ! volume name=srclevel ! audioconvert ! "
|
||||
"audioresample ! queue ! opusenc ! rtpopuspay ! "
|
||||
"queue ! " RTP_CAPS_OPUS +
|
||||
std::to_string(opusPayloadType) + " ! webrtcbin.");
|
||||
|
||||
webrtc_ = nullptr;
|
||||
GError *error = nullptr;
|
||||
pipe_ = gst_parse_launch(pipeline.c_str(), &error);
|
||||
if (error) {
|
||||
nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
|
||||
g_error_free(error);
|
||||
end();
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
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;
|
||||
}
|
||||
|
||||
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);
|
||||
g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
|
||||
}
|
||||
}
|
||||
webrtc_ = nullptr;
|
||||
GError *error = nullptr;
|
||||
pipe_ = gst_parse_launch(pipeline.c_str(), &error);
|
||||
if (error) {
|
||||
nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
|
||||
g_error_free(error);
|
||||
end();
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
bool
|
||||
WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
|
||||
{
|
||||
if (state_ < State::INITIATED)
|
||||
return false;
|
||||
if (state_ < State::INITIATED)
|
||||
return false;
|
||||
|
||||
GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
|
||||
if (!srclevel)
|
||||
return false;
|
||||
GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
|
||||
if (!srclevel)
|
||||
return false;
|
||||
|
||||
gboolean muted;
|
||||
g_object_get(srclevel, "mute", &muted, nullptr);
|
||||
g_object_set(srclevel, "mute", !muted, nullptr);
|
||||
gst_object_unref(srclevel);
|
||||
isMuted = !muted;
|
||||
return true;
|
||||
gboolean muted;
|
||||
g_object_get(srclevel, "mute", &muted, nullptr);
|
||||
g_object_set(srclevel, "mute", !muted, nullptr);
|
||||
gst_object_unref(srclevel);
|
||||
isMuted = !muted;
|
||||
return true;
|
||||
}
|
||||
|
||||
void
|
||||
WebRTCSession::end()
|
||||
{
|
||||
nhlog::ui()->debug("WebRTC: ending session");
|
||||
if (pipe_) {
|
||||
gst_element_set_state(pipe_, GST_STATE_NULL);
|
||||
gst_object_unref(pipe_);
|
||||
pipe_ = nullptr;
|
||||
}
|
||||
webrtc_ = nullptr;
|
||||
if (state_ != State::DISCONNECTED)
|
||||
emit stateChanged(State::DISCONNECTED);
|
||||
}
|
||||
|
||||
namespace {
|
||||
|
||||
std::string::const_iterator findName(const std::string &sdp, const std::string &name)
|
||||
{
|
||||
return std::search(sdp.cbegin(), sdp.cend(), name.cbegin(), name.cend(),
|
||||
[](unsigned char c1, unsigned char c2) {return std::tolower(c1) == std::tolower(c2);});
|
||||
}
|
||||
|
||||
int getPayloadType(const std::string &sdp, const std::string &name)
|
||||
{
|
||||
// eg a=rtpmap:111 opus/48000/2
|
||||
auto e = findName(sdp, name);
|
||||
if (e == sdp.cend()) {
|
||||
nhlog::ui()->error("WebRTC: remote offer - " + name + " attribute missing");
|
||||
return -1;
|
||||
}
|
||||
|
||||
if (auto s = sdp.rfind(':', e - sdp.cbegin()); s == std::string::npos) {
|
||||
nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
|
||||
return -1;
|
||||
}
|
||||
else {
|
||||
++s;
|
||||
try {
|
||||
return std::stoi(std::string(sdp, s, e - sdp.cbegin() - s));
|
||||
}
|
||||
catch(...) {
|
||||
nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
|
||||
}
|
||||
}
|
||||
return -1;
|
||||
}
|
||||
|
||||
gboolean
|
||||
newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
|
||||
{
|
||||
WebRTCSession *session = (WebRTCSession*)user_data;
|
||||
switch (GST_MESSAGE_TYPE(msg)) {
|
||||
case GST_MESSAGE_EOS:
|
||||
nhlog::ui()->error("WebRTC: end of stream");
|
||||
session->end();
|
||||
break;
|
||||
case GST_MESSAGE_ERROR:
|
||||
GError *error;
|
||||
gchar *debug;
|
||||
gst_message_parse_error(msg, &error, &debug);
|
||||
nhlog::ui()->error("WebRTC: error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
|
||||
g_clear_error(&error);
|
||||
g_free(debug);
|
||||
session->end();
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
GstWebRTCSessionDescription*
|
||||
parseSDP(const std::string &sdp, GstWebRTCSDPType type)
|
||||
{
|
||||
GstSDPMessage *msg;
|
||||
gst_sdp_message_new(&msg);
|
||||
if (gst_sdp_message_parse_buffer((guint8*)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
|
||||
return gst_webrtc_session_description_new(type, msg);
|
||||
}
|
||||
else {
|
||||
nhlog::ui()->error("WebRTC: failed to parse remote session description");
|
||||
gst_object_unref(msg);
|
||||
return nullptr;
|
||||
}
|
||||
}
|
||||
|
||||
void
|
||||
generateOffer(GstElement *webrtc)
|
||||
{
|
||||
// create-offer first, then set-local-description
|
||||
GstPromise *promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
|
||||
g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
|
||||
}
|
||||
|
||||
void
|
||||
setLocalDescription(GstPromise *promise, gpointer webrtc)
|
||||
{
|
||||
const GstStructure *reply = gst_promise_get_reply(promise);
|
||||
gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
|
||||
GstWebRTCSessionDescription *gstsdp = nullptr;
|
||||
gst_structure_get(reply, isAnswer ? "answer" : "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &gstsdp, nullptr);
|
||||
gst_promise_unref(promise);
|
||||
g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
|
||||
|
||||
gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
|
||||
localsdp_ = std::string(sdp);
|
||||
g_free(sdp);
|
||||
gst_webrtc_session_description_free(gstsdp);
|
||||
|
||||
nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
|
||||
}
|
||||
|
||||
void
|
||||
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED)
|
||||
{
|
||||
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
|
||||
|
||||
if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
|
||||
emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
|
||||
return;
|
||||
}
|
||||
|
||||
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
|
||||
|
||||
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
|
||||
// fixed in v1.18
|
||||
// use a 100ms timeout in the meantime
|
||||
static guint timerid = 0;
|
||||
if (timerid)
|
||||
g_source_remove(timerid);
|
||||
|
||||
timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
|
||||
}
|
||||
|
||||
gboolean
|
||||
onICEGatheringCompletion(gpointer timerid)
|
||||
{
|
||||
*(guint*)(timerid) = 0;
|
||||
if (isoffering_) {
|
||||
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
|
||||
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
|
||||
}
|
||||
else {
|
||||
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
|
||||
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
|
||||
}
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
void
|
||||
iceConnectionStateChanged(GstElement *webrtc, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED)
|
||||
{
|
||||
GstWebRTCICEConnectionState newState;
|
||||
g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
|
||||
switch (newState) {
|
||||
case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
|
||||
nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
|
||||
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
|
||||
break;
|
||||
case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
|
||||
nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
|
||||
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
void
|
||||
createAnswer(GstPromise *promise, gpointer webrtc)
|
||||
{
|
||||
// create-answer first, then set-local-description
|
||||
gst_promise_unref(promise);
|
||||
promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
|
||||
g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
|
||||
}
|
||||
|
||||
void
|
||||
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);
|
||||
g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
|
||||
gst_bin_add(GST_BIN(pipe), decodebin);
|
||||
gst_element_sync_state_with_parent(decodebin);
|
||||
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
|
||||
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad)))
|
||||
nhlog::ui()->error("WebRTC: unable to link new pad");
|
||||
gst_object_unref(sinkpad);
|
||||
}
|
||||
|
||||
void
|
||||
linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
|
||||
{
|
||||
GstCaps *caps = gst_pad_get_current_caps(newpad);
|
||||
if (!caps)
|
||||
return;
|
||||
|
||||
const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
|
||||
gst_caps_unref(caps);
|
||||
|
||||
GstPad *queuepad = nullptr;
|
||||
if (g_str_has_prefix(name, "audio")) {
|
||||
nhlog::ui()->debug("WebRTC: received incoming audio stream");
|
||||
GstElement *queue = gst_element_factory_make("queue", nullptr);
|
||||
GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
|
||||
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
|
||||
GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
|
||||
gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
|
||||
gst_element_sync_state_with_parent(queue);
|
||||
gst_element_sync_state_with_parent(convert);
|
||||
gst_element_sync_state_with_parent(resample);
|
||||
gst_element_sync_state_with_parent(sink);
|
||||
gst_element_link_many(queue, convert, resample, sink, nullptr);
|
||||
queuepad = gst_element_get_static_pad(queue, "sink");
|
||||
}
|
||||
|
||||
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);
|
||||
}
|
||||
}
|
||||
|
||||
nhlog::ui()->debug("WebRTC: ending session");
|
||||
if (pipe_) {
|
||||
gst_element_set_state(pipe_, GST_STATE_NULL);
|
||||
gst_object_unref(pipe_);
|
||||
pipe_ = nullptr;
|
||||
}
|
||||
webrtc_ = nullptr;
|
||||
if (state_ != State::DISCONNECTED)
|
||||
emit stateChanged(State::DISCONNECTED);
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue