Merge remote-tracking branch 'origin/master' into cross-signing
This commit is contained in:
Nicolas Werner
commit
4802c34009
43 changed files with 8773 additions and 2200 deletions
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@ -21,6 +21,7 @@ WebRTCSession::WebRTCSession()
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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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init();
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}
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bool
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@ -78,7 +79,11 @@ WebRTCSession::init(std::string *errorMessage)
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gst_object_unref(plugin);
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}
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if (!initialised_) {
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if (initialised_) {
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#if GST_CHECK_VERSION(1, 18, 0)
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startDeviceMonitor();
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#endif
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} else {
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nhlog::ui()->error(strError);
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if (errorMessage)
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*errorMessage = strError;
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@ -95,12 +100,65 @@ 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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std::vector<std::pair<std::string, GstDevice *>> audioSources_;
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void
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addDevice(GstDevice *device)
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{
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if (device) {
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gchar *name = gst_device_get_display_name(device);
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nhlog::ui()->debug("WebRTC: device added: {}", name);
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audioSources_.push_back({name, device});
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g_free(name);
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}
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}
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#if GST_CHECK_VERSION(1, 18, 0)
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void
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removeDevice(GstDevice *device, bool changed)
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{
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if (device) {
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if (auto it = std::find_if(audioSources_.begin(),
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audioSources_.end(),
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[device](const auto &s) { return s.second == device; });
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it != audioSources_.end()) {
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nhlog::ui()->debug(std::string("WebRTC: device ") +
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(changed ? "changed: " : "removed: ") + "{}",
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it->first);
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gst_object_unref(device);
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audioSources_.erase(it);
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}
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}
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}
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#endif
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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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#if GST_CHECK_VERSION(1, 18, 0)
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case GST_MESSAGE_DEVICE_ADDED: {
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GstDevice *device;
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gst_message_parse_device_added(msg, &device);
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addDevice(device);
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break;
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}
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case GST_MESSAGE_DEVICE_REMOVED: {
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GstDevice *device;
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gst_message_parse_device_removed(msg, &device);
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removeDevice(device, false);
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break;
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}
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case GST_MESSAGE_DEVICE_CHANGED: {
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GstDevice *device;
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GstDevice *oldDevice;
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gst_message_parse_device_changed(msg, &device, &oldDevice);
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removeDevice(oldDevice, true);
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addDevice(device);
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break;
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}
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#endif
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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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@ -176,7 +234,7 @@ createAnswer(GstPromise *promise, gpointer webrtc)
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g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
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}
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#if GST_CHECK_VERSION(1, 17, 0)
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#if GST_CHECK_VERSION(1, 18, 0)
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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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@ -223,7 +281,7 @@ addLocalICECandidate(GstElement *webrtc 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 GST_CHECK_VERSION(1, 17, 0)
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#if GST_CHECK_VERSION(1, 18, 0)
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localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
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return;
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#else
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@ -233,8 +291,10 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
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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 v1.17.
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// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. 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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@ -423,8 +483,12 @@ WebRTCSession::acceptICECandidates(
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for (const auto &c : candidates) {
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nhlog::ui()->debug(
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"WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
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g_signal_emit_by_name(
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webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
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if (!c.candidate.empty()) {
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g_signal_emit_by_name(webrtc_,
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"add-ice-candidate",
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c.sdpMLineIndex,
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c.candidate.c_str());
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}
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}
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}
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}
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@ -471,7 +535,7 @@ WebRTCSession::startPipeline(int opusPayloadType)
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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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#if GST_CHECK_VERSION(1, 17, 0)
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#if GST_CHECK_VERSION(1, 18, 0)
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// capture ICE gathering completion
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g_signal_connect(
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webrtc_, "notify::ice-gathering-state", G_CALLBACK(iceGatheringStateChanged), nullptr);
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@ -488,7 +552,7 @@ WebRTCSession::startPipeline(int opusPayloadType)
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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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busWatchId_ = 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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@ -497,19 +561,18 @@ WebRTCSession::startPipeline(int opusPayloadType)
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bool
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WebRTCSession::createPipeline(int opusPayloadType)
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{
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int nSources = audioSources_ ? g_list_length(audioSources_) : 0;
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if (nSources == 0) {
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if (audioSources_.empty()) {
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nhlog::ui()->error("WebRTC: no audio sources");
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return false;
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}
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if (audioSourceIndex_ < 0 || audioSourceIndex_ >= nSources) {
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if (audioSourceIndex_ < 0 || (size_t)audioSourceIndex_ >= audioSources_.size()) {
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nhlog::ui()->error("WebRTC: invalid audio source index");
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return false;
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}
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GstElement *source = gst_device_create_element(
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GST_DEVICE_CAST(g_list_nth_data(audioSources_, audioSourceIndex_)), nullptr);
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GstElement *source =
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gst_device_create_element(audioSources_[audioSourceIndex_].second, nullptr);
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GstElement *volume = gst_element_factory_make("volume", "srclevel");
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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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@ -594,12 +657,40 @@ WebRTCSession::end()
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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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g_source_remove(busWatchId_);
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busWatchId_ = 0;
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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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#if GST_CHECK_VERSION(1, 18, 0)
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void
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WebRTCSession::startDeviceMonitor()
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{
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if (!initialised_)
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return;
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static GstDeviceMonitor *monitor = nullptr;
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if (!monitor) {
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monitor = gst_device_monitor_new();
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GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
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gst_device_monitor_add_filter(monitor, "Audio/Source", caps);
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gst_caps_unref(caps);
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GstBus *bus = gst_device_monitor_get_bus(monitor);
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gst_bus_add_watch(bus, newBusMessage, nullptr);
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gst_object_unref(bus);
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if (!gst_device_monitor_start(monitor)) {
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nhlog::ui()->error("WebRTC: failed to start device monitor");
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return;
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}
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}
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}
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#else
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void
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WebRTCSession::refreshDevices()
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{
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@ -613,31 +704,42 @@ WebRTCSession::refreshDevices()
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gst_device_monitor_add_filter(monitor, "Audio/Source", caps);
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gst_caps_unref(caps);
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}
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g_list_free_full(audioSources_, g_object_unref);
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audioSources_ = gst_device_monitor_get_devices(monitor);
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std::for_each(audioSources_.begin(), audioSources_.end(), [](const auto &s) {
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gst_object_unref(s.second);
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});
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audioSources_.clear();
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GList *devices = gst_device_monitor_get_devices(monitor);
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if (devices) {
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audioSources_.reserve(g_list_length(devices));
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for (GList *l = devices; l != nullptr; l = l->next)
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addDevice(GST_DEVICE_CAST(l->data));
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g_list_free(devices);
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}
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}
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#endif
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std::vector<std::string>
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WebRTCSession::getAudioSourceNames(const std::string &defaultDevice)
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{
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if (!initialised_)
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return {};
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#if !GST_CHECK_VERSION(1, 18, 0)
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refreshDevices();
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#endif
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// move default device to top of the list
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if (auto it = std::find_if(audioSources_.begin(),
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audioSources_.end(),
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[&](const auto &s) { return s.first == defaultDevice; });
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it != audioSources_.end())
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std::swap(audioSources_.front(), *it);
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std::vector<std::string> ret;
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ret.reserve(g_list_length(audioSources_));
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for (GList *l = audioSources_; l != nullptr; l = l->next) {
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gchar *name = gst_device_get_display_name(GST_DEVICE_CAST(l->data));
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ret.emplace_back(name);
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g_free(name);
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if (ret.back() == defaultDevice) {
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// move default device to top of the list
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std::swap(audioSources_->data, l->data);
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std::swap(ret.front(), ret.back());
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}
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}
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ret.reserve(audioSources_.size());
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std::for_each(audioSources_.cbegin(), audioSources_.cend(), [&](const auto &s) {
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ret.push_back(s.first);
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});
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return ret;
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}
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#else
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bool
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@ -688,6 +790,10 @@ void
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WebRTCSession::refreshDevices()
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{}
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void
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WebRTCSession::startDeviceMonitor()
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{}
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std::vector<std::string>
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WebRTCSession::getAudioSourceNames(const std::string &)
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{
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