Move call device handling out of WebRTCSession
This commit is contained in:
trilene
parent
375e20462b
commit
f1bc3ba587
8 changed files with 509 additions and 396 deletions
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@ -35,6 +35,7 @@ using webrtc::State;
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WebRTCSession::WebRTCSession()
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: QObject()
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, devices_(CallDevices::instance())
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{
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qRegisterMetaType<webrtc::State>();
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qmlRegisterUncreatableMetaObject(
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@ -68,9 +69,7 @@ WebRTCSession::init(std::string *errorMessage)
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gchar *version = gst_version_string();
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nhlog::ui()->info("WebRTC: initialised {}", version);
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g_free(version);
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#if GST_CHECK_VERSION(1, 18, 0)
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startDeviceMonitor();
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#endif
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devices_.init();
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return true;
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#else
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(void)errorMessage;
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@ -81,195 +80,17 @@ WebRTCSession::init(std::string *errorMessage)
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#ifdef GSTREAMER_AVAILABLE
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namespace {
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struct AudioSource
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{
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std::string name;
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GstDevice *device;
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};
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struct VideoSource
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{
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struct Caps
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{
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std::string resolution;
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std::vector<std::string> frameRates;
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};
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std::string name;
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GstDevice *device;
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std::vector<Caps> caps;
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};
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std::string localsdp_;
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std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
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bool haveAudioStream_;
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bool haveVideoStream_;
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std::vector<AudioSource> audioSources_;
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std::vector<VideoSource> videoSources_;
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GstPad *insetSinkPad_ = nullptr;
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using FrameRate = std::pair<int, int>;
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std::optional<FrameRate>
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getFrameRate(const GValue *value)
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{
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if (GST_VALUE_HOLDS_FRACTION(value)) {
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gint num = gst_value_get_fraction_numerator(value);
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gint den = gst_value_get_fraction_denominator(value);
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return FrameRate{num, den};
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}
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return std::nullopt;
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}
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void
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addFrameRate(std::vector<std::string> &rates, const FrameRate &rate)
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{
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constexpr double minimumFrameRate = 15.0;
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if (static_cast<double>(rate.first) / rate.second >= minimumFrameRate)
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rates.push_back(std::to_string(rate.first) + "/" + std::to_string(rate.second));
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}
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std::pair<int, int>
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tokenise(std::string_view str, char delim)
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{
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std::pair<int, int> ret;
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ret.first = std::atoi(str.data());
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auto pos = str.find_first_of(delim);
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ret.second = std::atoi(str.data() + pos + 1);
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return ret;
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}
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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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return;
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gchar *name = gst_device_get_display_name(device);
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gchar *type = gst_device_get_device_class(device);
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bool isVideo = !std::strncmp(type, "Video", 5);
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g_free(type);
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nhlog::ui()->debug("WebRTC: {} device added: {}", isVideo ? "video" : "audio", name);
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if (!isVideo) {
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audioSources_.push_back({name, device});
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g_free(name);
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return;
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}
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GstCaps *gstcaps = gst_device_get_caps(device);
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if (!gstcaps) {
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nhlog::ui()->debug("WebRTC: unable to get caps for {}", name);
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g_free(name);
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return;
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}
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VideoSource source{name, device, {}};
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g_free(name);
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guint nCaps = gst_caps_get_size(gstcaps);
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for (guint i = 0; i < nCaps; ++i) {
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GstStructure *structure = gst_caps_get_structure(gstcaps, i);
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const gchar *name = gst_structure_get_name(structure);
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if (!std::strcmp(name, "video/x-raw")) {
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gint widthpx, heightpx;
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if (gst_structure_get(structure,
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"width",
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G_TYPE_INT,
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&widthpx,
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"height",
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G_TYPE_INT,
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&heightpx,
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nullptr)) {
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VideoSource::Caps caps;
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caps.resolution =
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std::to_string(widthpx) + "x" + std::to_string(heightpx);
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const GValue *value =
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gst_structure_get_value(structure, "framerate");
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if (auto fr = getFrameRate(value); fr)
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addFrameRate(caps.frameRates, *fr);
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else if (GST_VALUE_HOLDS_FRACTION_RANGE(value)) {
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const GValue *minRate =
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gst_value_get_fraction_range_min(value);
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if (auto fr = getFrameRate(minRate); fr)
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addFrameRate(caps.frameRates, *fr);
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const GValue *maxRate =
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gst_value_get_fraction_range_max(value);
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if (auto fr = getFrameRate(maxRate); fr)
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addFrameRate(caps.frameRates, *fr);
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} else if (GST_VALUE_HOLDS_LIST(value)) {
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guint nRates = gst_value_list_get_size(value);
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for (guint j = 0; j < nRates; ++j) {
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const GValue *rate =
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gst_value_list_get_value(value, j);
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if (auto fr = getFrameRate(rate); fr)
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addFrameRate(caps.frameRates, *fr);
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}
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}
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if (!caps.frameRates.empty())
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source.caps.push_back(std::move(caps));
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}
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}
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}
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gst_caps_unref(gstcaps);
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videoSources_.push_back(std::move(source));
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}
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#if GST_CHECK_VERSION(1, 18, 0)
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template<typename T>
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bool
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removeDevice(T &sources, GstDevice *device, bool changed)
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{
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if (auto it = std::find_if(sources.begin(),
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sources.end(),
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[device](const auto &s) { return s.device == device; });
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it != sources.end()) {
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nhlog::ui()->debug(std::string("WebRTC: device ") +
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(changed ? "changed: " : "removed: ") + "{}",
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it->name);
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gst_object_unref(device);
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sources.erase(it);
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return true;
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}
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return false;
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}
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void
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removeDevice(GstDevice *device, bool changed)
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{
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if (device) {
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if (removeDevice(audioSources_, device, changed) ||
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removeDevice(videoSources_, device, changed))
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return;
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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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emit WebRTCSession::instance().devicesChanged();
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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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emit WebRTCSession::instance().devicesChanged();
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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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@ -724,27 +545,6 @@ getMediaAttributes(const GstSDPMessage *sdp,
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return false;
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}
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template<typename T>
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std::vector<std::string>
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deviceNames(T &sources, const std::string &defaultDevice)
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{
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std::vector<std::string> ret;
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ret.reserve(sources.size());
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std::transform(sources.cbegin(),
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sources.cend(),
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std::back_inserter(ret),
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[](const auto &s) { return s.name; });
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// move default device to top of the list
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if (auto it = std::find_if(ret.begin(),
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ret.end(),
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[&defaultDevice](const auto &s) { return s == defaultDevice; });
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it != ret.end())
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std::swap(ret.front(), *it);
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return ret;
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}
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}
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bool
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@ -995,19 +795,11 @@ WebRTCSession::startPipeline(int opusPayloadType, int vp8PayloadType)
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bool
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WebRTCSession::createPipeline(int opusPayloadType, int vp8PayloadType)
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{
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std::string microphoneSetting =
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ChatPage::instance()->userSettings()->microphone().toStdString();
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auto it =
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std::find_if(audioSources_.cbegin(),
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audioSources_.cend(),
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[µphoneSetting](const auto &s) { return s.name == microphoneSetting; });
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if (it == audioSources_.cend()) {
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nhlog::ui()->error("WebRTC: unknown microphone: {}", microphoneSetting);
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GstDevice *device = devices_.audioDevice();
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if (!device)
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return false;
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}
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nhlog::ui()->debug("WebRTC: microphone: {}", microphoneSetting);
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GstElement *source = gst_device_create_element(it->device, nullptr);
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GstElement *source = gst_device_create_element(device, 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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@ -1070,30 +862,16 @@ bool
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WebRTCSession::addVideoPipeline(int vp8PayloadType)
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{
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// allow incoming video calls despite localUser having no webcam
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if (videoSources_.empty())
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if (!devices_.haveCamera())
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return !isOffering_;
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QSharedPointer<UserSettings> settings = ChatPage::instance()->userSettings();
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std::string cameraSetting = settings->camera().toStdString();
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auto it = std::find_if(videoSources_.cbegin(),
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videoSources_.cend(),
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[&cameraSetting](const auto &s) { return s.name == cameraSetting; });
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if (it == videoSources_.cend()) {
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nhlog::ui()->error("WebRTC: unknown camera: {}", cameraSetting);
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std::pair<int, int> resolution;
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std::pair<int, int> frameRate;
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GstDevice *device = devices_.videoDevice(resolution, frameRate);
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if (!device)
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return false;
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}
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std::string resSetting = settings->cameraResolution().toStdString();
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const std::string &res = resSetting.empty() ? it->caps.front().resolution : resSetting;
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std::string frSetting = settings->cameraFrameRate().toStdString();
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const std::string &fr = frSetting.empty() ? it->caps.front().frameRates.front() : frSetting;
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auto resolution = tokenise(res, 'x');
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auto frameRate = tokenise(fr, '/');
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nhlog::ui()->debug("WebRTC: camera: {}", cameraSetting);
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nhlog::ui()->debug("WebRTC: camera resolution: {}x{}", resolution.first, resolution.second);
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nhlog::ui()->debug("WebRTC: camera frame rate: {}/{}", frameRate.first, frameRate.second);
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GstElement *source = gst_device_create_element(it->device, nullptr);
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GstElement *source = gst_device_create_element(device, nullptr);
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GstElement *videoconvert = gst_element_factory_make("videoconvert", nullptr);
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GstElement *capsfilter = gst_element_factory_make("capsfilter", "camerafilter");
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GstCaps *caps = gst_caps_new_simple("video/x-raw",
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@ -1239,111 +1017,6 @@ WebRTCSession::end()
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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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caps = gst_caps_new_empty_simple("video/x-raw");
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gst_device_monitor_add_filter(monitor, "Video/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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#endif
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void
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WebRTCSession::refreshDevices()
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{
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#if GST_CHECK_VERSION(1, 18, 0)
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return;
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#else
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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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caps = gst_caps_new_empty_simple("video/x-raw");
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gst_device_monitor_add_filter(monitor, "Video/Source", caps);
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gst_caps_unref(caps);
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}
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auto clearDevices = [](auto &sources) {
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std::for_each(
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sources.begin(), sources.end(), [](auto &s) { gst_object_unref(s.device); });
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sources.clear();
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};
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clearDevices(audioSources_);
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clearDevices(videoSources_);
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GList *devices = gst_device_monitor_get_devices(monitor);
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if (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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emit devicesChanged();
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#endif
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}
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std::vector<std::string>
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WebRTCSession::getDeviceNames(bool isVideo, const std::string &defaultDevice) const
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{
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return isVideo ? deviceNames(videoSources_, defaultDevice)
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: deviceNames(audioSources_, defaultDevice);
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}
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std::vector<std::string>
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WebRTCSession::getResolutions(const std::string &cameraName) const
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{
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std::vector<std::string> ret;
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if (auto it = std::find_if(videoSources_.cbegin(),
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videoSources_.cend(),
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[&cameraName](const auto &s) { return s.name == cameraName; });
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it != videoSources_.cend()) {
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ret.reserve(it->caps.size());
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for (const auto &c : it->caps)
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ret.push_back(c.resolution);
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}
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return ret;
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}
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std::vector<std::string>
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WebRTCSession::getFrameRates(const std::string &cameraName, const std::string &resolution) const
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{
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if (auto i = std::find_if(videoSources_.cbegin(),
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videoSources_.cend(),
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[&](const auto &s) { return s.name == cameraName; });
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i != videoSources_.cend()) {
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if (auto j =
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std::find_if(i->caps.cbegin(),
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i->caps.cend(),
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[&](const auto &s) { return s.resolution == resolution; });
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j != i->caps.cend())
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return j->frameRates;
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}
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return {};
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}
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#else
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bool
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@ -1400,25 +1073,4 @@ void
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WebRTCSession::end()
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{}
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void
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WebRTCSession::refreshDevices()
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{}
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std::vector<std::string>
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WebRTCSession::getDeviceNames(bool, const std::string &) const
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{
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return {};
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}
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std::vector<std::string>
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WebRTCSession::getResolutions(const std::string &) const
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{
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return {};
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}
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std::vector<std::string>
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WebRTCSession::getFrameRates(const std::string &, const std::string &) const
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{
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return {};
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}
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#endif
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