mirror of
https://github.com/Nheko-Reborn/nheko.git
synced 2024-11-22 11:00:48 +03:00
Move call device handling out of WebRTCSession
This commit is contained in:
parent
375e20462b
commit
f1bc3ba587
8 changed files with 509 additions and 396 deletions
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@ -298,6 +298,7 @@ set(SRC_FILES
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src/AvatarProvider.cpp
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src/BlurhashProvider.cpp
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src/Cache.cpp
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src/CallDevices.cpp
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src/CallManager.cpp
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src/ChatPage.cpp
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src/ColorImageProvider.cpp
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@ -512,6 +513,7 @@ qt5_wrap_cpp(MOC_HEADERS
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src/AvatarProvider.h
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src/BlurhashProvider.h
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src/Cache_p.h
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src/CallDevices.h
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src/CallManager.h
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src/ChatPage.h
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src/CommunitiesList.h
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433
src/CallDevices.cpp
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433
src/CallDevices.cpp
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@ -0,0 +1,433 @@
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#include <cstring>
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#include <optional>
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#include <string_view>
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#include "CallDevices.h"
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#include "ChatPage.h"
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#include "Logging.h"
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#include "UserSettingsPage.h"
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#ifdef GSTREAMER_AVAILABLE
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extern "C"
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{
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#include "gst/gst.h"
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}
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#endif
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CallDevices::CallDevices()
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: QObject()
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{}
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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::vector<AudioSource> audioSources_;
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std::vector<VideoSource> videoSources_;
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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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void
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setDefaultDevice(bool isVideo)
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{
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auto settings = ChatPage::instance()->userSettings();
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if (isVideo && settings->camera().isEmpty()) {
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const VideoSource &camera = videoSources_.front();
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settings->setCamera(QString::fromStdString(camera.name));
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settings->setCameraResolution(
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QString::fromStdString(camera.caps.front().resolution));
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settings->setCameraFrameRate(
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QString::fromStdString(camera.caps.front().frameRates.front()));
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} else if (!isVideo && settings->microphone().isEmpty()) {
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settings->setMicrophone(QString::fromStdString(audioSources_.front().name));
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}
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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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setDefaultDevice(false);
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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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addFrameRate(
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caps.frameRates,
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*getFrameRate(gst_value_get_fraction_range_min(value)));
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addFrameRate(
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caps.frameRates,
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*getFrameRate(gst_value_get_fraction_range_max(value)));
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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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setDefaultDevice(true);
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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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gboolean
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newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data G_GNUC_UNUSED)
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{
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switch (GST_MESSAGE_TYPE(msg)) {
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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 CallDevices::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 CallDevices::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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default:
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break;
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}
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return TRUE;
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}
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#endif
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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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for (const auto &s : sources)
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ret.push_back(s.name);
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// move default device to top of the list
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if (auto it = std::find(ret.begin(), ret.end(), defaultDevice); 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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std::optional<VideoSource>
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getVideoSource(const std::string &cameraName)
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{
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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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return *it;
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}
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return std::nullopt;
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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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}
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void
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CallDevices::init()
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{
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#if GST_CHECK_VERSION(1, 18, 0)
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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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#endif
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}
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void
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CallDevices::refresh()
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{
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#if !GST_CHECK_VERSION(1, 18, 0)
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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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bool
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CallDevices::haveMic() const
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{
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return !audioSources_.empty();
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}
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bool
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CallDevices::haveCamera() const
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{
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return !videoSources_.empty();
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}
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std::vector<std::string>
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CallDevices::names(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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CallDevices::resolutions(const std::string &cameraName) const
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{
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std::vector<std::string> ret;
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if (auto s = getVideoSource(cameraName); s) {
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ret.reserve(s->caps.size());
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for (const auto &c : s->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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CallDevices::frameRates(const std::string &cameraName, const std::string &resolution) const
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{
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if (auto s = getVideoSource(cameraName); s) {
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if (auto it =
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std::find_if(s->caps.cbegin(),
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s->caps.cend(),
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[&](const auto &c) { return c.resolution == resolution; });
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it != s->caps.cend())
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return it->frameRates;
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}
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return {};
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}
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GstDevice *
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CallDevices::audioDevice() const
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{
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std::string name = ChatPage::instance()->userSettings()->microphone().toStdString();
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if (auto it = std::find_if(audioSources_.cbegin(),
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audioSources_.cend(),
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[&name](const auto &s) { return s.name == name; });
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it != audioSources_.cend()) {
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nhlog::ui()->debug("WebRTC: microphone: {}", name);
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return it->device;
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} else {
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nhlog::ui()->error("WebRTC: unknown microphone: {}", name);
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return nullptr;
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}
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}
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GstDevice *
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CallDevices::videoDevice(std::pair<int, int> &resolution, std::pair<int, int> &frameRate) const
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{
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auto settings = ChatPage::instance()->userSettings();
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std::string name = settings->camera().toStdString();
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if (auto s = getVideoSource(name); s) {
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nhlog::ui()->debug("WebRTC: camera: {}", name);
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resolution = tokenise(settings->cameraResolution().toStdString(), 'x');
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frameRate = tokenise(settings->cameraFrameRate().toStdString(), '/');
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nhlog::ui()->debug(
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"WebRTC: camera resolution: {}x{}", resolution.first, resolution.second);
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nhlog::ui()->debug(
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"WebRTC: camera frame rate: {}/{}", frameRate.first, frameRate.second);
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return s->device;
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} else {
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nhlog::ui()->error("WebRTC: unknown camera: {}", name);
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return nullptr;
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}
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}
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#else
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void
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CallDevices::refresh()
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{}
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bool
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CallDevices::haveMic() const
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{
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return false;
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}
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bool
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CallDevices::haveCamera() const
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{
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return false;
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}
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std::vector<std::string>
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CallDevices::names(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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CallDevices::resolutions(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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CallDevices::frameRates(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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45
src/CallDevices.h
Normal file
45
src/CallDevices.h
Normal file
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#pragma once
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#include <string>
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#include <utility>
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#include <vector>
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#include <QObject>
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typedef struct _GstDevice GstDevice;
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class CallDevices : public QObject
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{
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Q_OBJECT
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public:
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static CallDevices &instance()
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{
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static CallDevices instance;
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return instance;
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}
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void refresh();
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bool haveMic() const;
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bool haveCamera() const;
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std::vector<std::string> names(bool isVideo, const std::string &defaultDevice) const;
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std::vector<std::string> resolutions(const std::string &cameraName) const;
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std::vector<std::string> frameRates(const std::string &cameraName,
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const std::string &resolution) const;
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signals:
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void devicesChanged();
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private:
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CallDevices();
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friend class WebRTCSession;
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void init();
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GstDevice *audioDevice() const;
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GstDevice *videoDevice(std::pair<int, int> &resolution,
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std::pair<int, int> &frameRate) const;
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public:
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CallDevices(CallDevices const &) = delete;
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void operator=(CallDevices const &) = delete;
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};
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@ -7,6 +7,7 @@
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#include <QUrl>
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#include "Cache.h"
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#include "CallDevices.h"
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#include "CallManager.h"
|
||||
#include "ChatPage.h"
|
||||
#include "Logging.h"
|
||||
|
@ -114,21 +115,10 @@ CallManager::CallManager(QObject *parent)
|
|||
emit newCallState();
|
||||
});
|
||||
|
||||
connect(&session_, &WebRTCSession::devicesChanged, this, [this]() {
|
||||
if (ChatPage::instance()->userSettings()->microphone().isEmpty()) {
|
||||
auto mics = session_.getDeviceNames(false, std::string());
|
||||
if (!mics.empty())
|
||||
ChatPage::instance()->userSettings()->setMicrophone(
|
||||
QString::fromStdString(mics.front()));
|
||||
}
|
||||
if (ChatPage::instance()->userSettings()->camera().isEmpty()) {
|
||||
auto cameras = session_.getDeviceNames(true, std::string());
|
||||
if (!cameras.empty())
|
||||
ChatPage::instance()->userSettings()->setCamera(
|
||||
QString::fromStdString(cameras.front()));
|
||||
}
|
||||
emit devicesChanged();
|
||||
});
|
||||
connect(&CallDevices::instance(),
|
||||
&CallDevices::devicesChanged,
|
||||
this,
|
||||
&CallManager::devicesChanged);
|
||||
|
||||
connect(&player_,
|
||||
&QMediaPlayer::mediaStatusChanged,
|
||||
|
@ -292,7 +282,7 @@ CallManager::handleEvent(const RoomEvent<CallInvite> &callInviteEvent)
|
|||
haveCallInvite_ = true;
|
||||
isVideo_ = isVideo;
|
||||
inviteSDP_ = callInviteEvent.content.sdp;
|
||||
session_.refreshDevices();
|
||||
CallDevices::instance().refresh();
|
||||
emit newInviteState();
|
||||
}
|
||||
|
||||
|
@ -409,7 +399,7 @@ CallManager::devices(bool isVideo) const
|
|||
const QString &defaultDevice = isVideo ? ChatPage::instance()->userSettings()->camera()
|
||||
: ChatPage::instance()->userSettings()->microphone();
|
||||
std::vector<std::string> devices =
|
||||
session_.getDeviceNames(isVideo, defaultDevice.toStdString());
|
||||
CallDevices::instance().names(isVideo, defaultDevice.toStdString());
|
||||
ret.reserve(devices.size());
|
||||
std::transform(devices.cbegin(),
|
||||
devices.cend(),
|
||||
|
|
|
@ -8,6 +8,7 @@
|
|||
#include <QString>
|
||||
#include <QTimer>
|
||||
|
||||
#include "CallDevices.h"
|
||||
#include "WebRTCSession.h"
|
||||
#include "mtx/events/collections.hpp"
|
||||
#include "mtx/events/voip.hpp"
|
||||
|
@ -53,7 +54,7 @@ public:
|
|||
public slots:
|
||||
void sendInvite(const QString &roomid, bool isVideo);
|
||||
void syncEvent(const mtx::events::collections::TimelineEvents &event);
|
||||
void refreshDevices() { session_.refreshDevices(); }
|
||||
void refreshDevices() { CallDevices::instance().refresh(); }
|
||||
void toggleMicMute();
|
||||
void toggleCameraView() { session_.toggleCameraView(); }
|
||||
void acceptInvite();
|
||||
|
|
|
@ -39,12 +39,12 @@
|
|||
#include <QtQml>
|
||||
|
||||
#include "Cache.h"
|
||||
#include "CallDevices.h"
|
||||
#include "Config.h"
|
||||
#include "MatrixClient.h"
|
||||
#include "Olm.h"
|
||||
#include "UserSettingsPage.h"
|
||||
#include "Utils.h"
|
||||
#include "WebRTCSession.h"
|
||||
#include "ui/FlatButton.h"
|
||||
#include "ui/ToggleButton.h"
|
||||
|
||||
|
@ -1060,7 +1060,7 @@ UserSettingsPage::UserSettingsPage(QSharedPointer<UserSettings> settings, QWidge
|
|||
[this](const QString &camera) {
|
||||
settings_->setCamera(camera);
|
||||
std::vector<std::string> resolutions =
|
||||
WebRTCSession::instance().getResolutions(camera.toStdString());
|
||||
CallDevices::instance().resolutions(camera.toStdString());
|
||||
cameraResolutionCombo_->clear();
|
||||
for (const auto &resolution : resolutions)
|
||||
cameraResolutionCombo_->addItem(QString::fromStdString(resolution));
|
||||
|
@ -1070,9 +1070,8 @@ UserSettingsPage::UserSettingsPage(QSharedPointer<UserSettings> settings, QWidge
|
|||
static_cast<void (QComboBox::*)(const QString &)>(&QComboBox::currentTextChanged),
|
||||
[this](const QString &resolution) {
|
||||
settings_->setCameraResolution(resolution);
|
||||
std::vector<std::string> frameRates =
|
||||
WebRTCSession::instance().getFrameRates(settings_->camera().toStdString(),
|
||||
resolution.toStdString());
|
||||
std::vector<std::string> frameRates = CallDevices::instance().frameRates(
|
||||
settings_->camera().toStdString(), resolution.toStdString());
|
||||
cameraFrameRateCombo_->clear();
|
||||
for (const auto &frameRate : frameRates)
|
||||
cameraFrameRateCombo_->addItem(QString::fromStdString(frameRate));
|
||||
|
@ -1231,9 +1230,8 @@ UserSettingsPage::showEvent(QShowEvent *)
|
|||
timelineMaxWidthSpin_->setValue(settings_->timelineMaxWidth());
|
||||
privacyScreenTimeout_->setValue(settings_->privacyScreenTimeout());
|
||||
|
||||
WebRTCSession::instance().refreshDevices();
|
||||
auto mics =
|
||||
WebRTCSession::instance().getDeviceNames(false, settings_->microphone().toStdString());
|
||||
CallDevices::instance().refresh();
|
||||
auto mics = CallDevices::instance().names(false, settings_->microphone().toStdString());
|
||||
microphoneCombo_->clear();
|
||||
for (const auto &m : mics)
|
||||
microphoneCombo_->addItem(QString::fromStdString(m));
|
||||
|
@ -1241,8 +1239,7 @@ UserSettingsPage::showEvent(QShowEvent *)
|
|||
auto cameraResolution = settings_->cameraResolution();
|
||||
auto cameraFrameRate = settings_->cameraFrameRate();
|
||||
|
||||
auto cameras =
|
||||
WebRTCSession::instance().getDeviceNames(true, settings_->camera().toStdString());
|
||||
auto cameras = CallDevices::instance().names(true, settings_->camera().toStdString());
|
||||
cameraCombo_->clear();
|
||||
for (const auto &c : cameras)
|
||||
cameraCombo_->addItem(QString::fromStdString(c));
|
||||
|
|
|
@ -35,6 +35,7 @@ using webrtc::State;
|
|||
|
||||
WebRTCSession::WebRTCSession()
|
||||
: QObject()
|
||||
, devices_(CallDevices::instance())
|
||||
{
|
||||
qRegisterMetaType<webrtc::State>();
|
||||
qmlRegisterUncreatableMetaObject(
|
||||
|
@ -68,9 +69,7 @@ WebRTCSession::init(std::string *errorMessage)
|
|||
gchar *version = gst_version_string();
|
||||
nhlog::ui()->info("WebRTC: initialised {}", version);
|
||||
g_free(version);
|
||||
#if GST_CHECK_VERSION(1, 18, 0)
|
||||
startDeviceMonitor();
|
||||
#endif
|
||||
devices_.init();
|
||||
return true;
|
||||
#else
|
||||
(void)errorMessage;
|
||||
|
@ -81,195 +80,17 @@ WebRTCSession::init(std::string *errorMessage)
|
|||
#ifdef GSTREAMER_AVAILABLE
|
||||
namespace {
|
||||
|
||||
struct AudioSource
|
||||
{
|
||||
std::string name;
|
||||
GstDevice *device;
|
||||
};
|
||||
|
||||
struct VideoSource
|
||||
{
|
||||
struct Caps
|
||||
{
|
||||
std::string resolution;
|
||||
std::vector<std::string> frameRates;
|
||||
};
|
||||
std::string name;
|
||||
GstDevice *device;
|
||||
std::vector<Caps> caps;
|
||||
};
|
||||
|
||||
std::string localsdp_;
|
||||
std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
|
||||
bool haveAudioStream_;
|
||||
bool haveVideoStream_;
|
||||
std::vector<AudioSource> audioSources_;
|
||||
std::vector<VideoSource> videoSources_;
|
||||
GstPad *insetSinkPad_ = nullptr;
|
||||
|
||||
using FrameRate = std::pair<int, int>;
|
||||
std::optional<FrameRate>
|
||||
getFrameRate(const GValue *value)
|
||||
{
|
||||
if (GST_VALUE_HOLDS_FRACTION(value)) {
|
||||
gint num = gst_value_get_fraction_numerator(value);
|
||||
gint den = gst_value_get_fraction_denominator(value);
|
||||
return FrameRate{num, den};
|
||||
}
|
||||
return std::nullopt;
|
||||
}
|
||||
|
||||
void
|
||||
addFrameRate(std::vector<std::string> &rates, const FrameRate &rate)
|
||||
{
|
||||
constexpr double minimumFrameRate = 15.0;
|
||||
if (static_cast<double>(rate.first) / rate.second >= minimumFrameRate)
|
||||
rates.push_back(std::to_string(rate.first) + "/" + std::to_string(rate.second));
|
||||
}
|
||||
|
||||
std::pair<int, int>
|
||||
tokenise(std::string_view str, char delim)
|
||||
{
|
||||
std::pair<int, int> ret;
|
||||
ret.first = std::atoi(str.data());
|
||||
auto pos = str.find_first_of(delim);
|
||||
ret.second = std::atoi(str.data() + pos + 1);
|
||||
return ret;
|
||||
}
|
||||
|
||||
void
|
||||
addDevice(GstDevice *device)
|
||||
{
|
||||
if (!device)
|
||||
return;
|
||||
|
||||
gchar *name = gst_device_get_display_name(device);
|
||||
gchar *type = gst_device_get_device_class(device);
|
||||
bool isVideo = !std::strncmp(type, "Video", 5);
|
||||
g_free(type);
|
||||
nhlog::ui()->debug("WebRTC: {} device added: {}", isVideo ? "video" : "audio", name);
|
||||
if (!isVideo) {
|
||||
audioSources_.push_back({name, device});
|
||||
g_free(name);
|
||||
return;
|
||||
}
|
||||
|
||||
GstCaps *gstcaps = gst_device_get_caps(device);
|
||||
if (!gstcaps) {
|
||||
nhlog::ui()->debug("WebRTC: unable to get caps for {}", name);
|
||||
g_free(name);
|
||||
return;
|
||||
}
|
||||
|
||||
VideoSource source{name, device, {}};
|
||||
g_free(name);
|
||||
guint nCaps = gst_caps_get_size(gstcaps);
|
||||
for (guint i = 0; i < nCaps; ++i) {
|
||||
GstStructure *structure = gst_caps_get_structure(gstcaps, i);
|
||||
const gchar *name = gst_structure_get_name(structure);
|
||||
if (!std::strcmp(name, "video/x-raw")) {
|
||||
gint widthpx, heightpx;
|
||||
if (gst_structure_get(structure,
|
||||
"width",
|
||||
G_TYPE_INT,
|
||||
&widthpx,
|
||||
"height",
|
||||
G_TYPE_INT,
|
||||
&heightpx,
|
||||
nullptr)) {
|
||||
VideoSource::Caps caps;
|
||||
caps.resolution =
|
||||
std::to_string(widthpx) + "x" + std::to_string(heightpx);
|
||||
const GValue *value =
|
||||
gst_structure_get_value(structure, "framerate");
|
||||
if (auto fr = getFrameRate(value); fr)
|
||||
addFrameRate(caps.frameRates, *fr);
|
||||
else if (GST_VALUE_HOLDS_FRACTION_RANGE(value)) {
|
||||
const GValue *minRate =
|
||||
gst_value_get_fraction_range_min(value);
|
||||
if (auto fr = getFrameRate(minRate); fr)
|
||||
addFrameRate(caps.frameRates, *fr);
|
||||
const GValue *maxRate =
|
||||
gst_value_get_fraction_range_max(value);
|
||||
if (auto fr = getFrameRate(maxRate); fr)
|
||||
addFrameRate(caps.frameRates, *fr);
|
||||
} else if (GST_VALUE_HOLDS_LIST(value)) {
|
||||
guint nRates = gst_value_list_get_size(value);
|
||||
for (guint j = 0; j < nRates; ++j) {
|
||||
const GValue *rate =
|
||||
gst_value_list_get_value(value, j);
|
||||
if (auto fr = getFrameRate(rate); fr)
|
||||
addFrameRate(caps.frameRates, *fr);
|
||||
}
|
||||
}
|
||||
if (!caps.frameRates.empty())
|
||||
source.caps.push_back(std::move(caps));
|
||||
}
|
||||
}
|
||||
}
|
||||
gst_caps_unref(gstcaps);
|
||||
videoSources_.push_back(std::move(source));
|
||||
}
|
||||
|
||||
#if GST_CHECK_VERSION(1, 18, 0)
|
||||
template<typename T>
|
||||
bool
|
||||
removeDevice(T &sources, GstDevice *device, bool changed)
|
||||
{
|
||||
if (auto it = std::find_if(sources.begin(),
|
||||
sources.end(),
|
||||
[device](const auto &s) { return s.device == device; });
|
||||
it != sources.end()) {
|
||||
nhlog::ui()->debug(std::string("WebRTC: device ") +
|
||||
(changed ? "changed: " : "removed: ") + "{}",
|
||||
it->name);
|
||||
gst_object_unref(device);
|
||||
sources.erase(it);
|
||||
return true;
|
||||
}
|
||||
return false;
|
||||
}
|
||||
|
||||
void
|
||||
removeDevice(GstDevice *device, bool changed)
|
||||
{
|
||||
if (device) {
|
||||
if (removeDevice(audioSources_, device, changed) ||
|
||||
removeDevice(videoSources_, device, changed))
|
||||
return;
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
gboolean
|
||||
newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
|
||||
{
|
||||
WebRTCSession *session = static_cast<WebRTCSession *>(user_data);
|
||||
switch (GST_MESSAGE_TYPE(msg)) {
|
||||
#if GST_CHECK_VERSION(1, 18, 0)
|
||||
case GST_MESSAGE_DEVICE_ADDED: {
|
||||
GstDevice *device;
|
||||
gst_message_parse_device_added(msg, &device);
|
||||
addDevice(device);
|
||||
emit WebRTCSession::instance().devicesChanged();
|
||||
break;
|
||||
}
|
||||
case GST_MESSAGE_DEVICE_REMOVED: {
|
||||
GstDevice *device;
|
||||
gst_message_parse_device_removed(msg, &device);
|
||||
removeDevice(device, false);
|
||||
emit WebRTCSession::instance().devicesChanged();
|
||||
break;
|
||||
}
|
||||
case GST_MESSAGE_DEVICE_CHANGED: {
|
||||
GstDevice *device;
|
||||
GstDevice *oldDevice;
|
||||
gst_message_parse_device_changed(msg, &device, &oldDevice);
|
||||
removeDevice(oldDevice, true);
|
||||
addDevice(device);
|
||||
break;
|
||||
}
|
||||
#endif
|
||||
case GST_MESSAGE_EOS:
|
||||
nhlog::ui()->error("WebRTC: end of stream");
|
||||
session->end();
|
||||
|
@ -724,27 +545,6 @@ getMediaAttributes(const GstSDPMessage *sdp,
|
|||
return false;
|
||||
}
|
||||
|
||||
template<typename T>
|
||||
std::vector<std::string>
|
||||
deviceNames(T &sources, const std::string &defaultDevice)
|
||||
{
|
||||
std::vector<std::string> ret;
|
||||
ret.reserve(sources.size());
|
||||
std::transform(sources.cbegin(),
|
||||
sources.cend(),
|
||||
std::back_inserter(ret),
|
||||
[](const auto &s) { return s.name; });
|
||||
|
||||
// move default device to top of the list
|
||||
if (auto it = std::find_if(ret.begin(),
|
||||
ret.end(),
|
||||
[&defaultDevice](const auto &s) { return s == defaultDevice; });
|
||||
it != ret.end())
|
||||
std::swap(ret.front(), *it);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
bool
|
||||
|
@ -995,19 +795,11 @@ WebRTCSession::startPipeline(int opusPayloadType, int vp8PayloadType)
|
|||
bool
|
||||
WebRTCSession::createPipeline(int opusPayloadType, int vp8PayloadType)
|
||||
{
|
||||
std::string microphoneSetting =
|
||||
ChatPage::instance()->userSettings()->microphone().toStdString();
|
||||
auto it =
|
||||
std::find_if(audioSources_.cbegin(),
|
||||
audioSources_.cend(),
|
||||
[µphoneSetting](const auto &s) { return s.name == microphoneSetting; });
|
||||
if (it == audioSources_.cend()) {
|
||||
nhlog::ui()->error("WebRTC: unknown microphone: {}", microphoneSetting);
|
||||
GstDevice *device = devices_.audioDevice();
|
||||
if (!device)
|
||||
return false;
|
||||
}
|
||||
nhlog::ui()->debug("WebRTC: microphone: {}", microphoneSetting);
|
||||
|
||||
GstElement *source = gst_device_create_element(it->device, nullptr);
|
||||
GstElement *source = gst_device_create_element(device, nullptr);
|
||||
GstElement *volume = gst_element_factory_make("volume", "srclevel");
|
||||
GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
|
||||
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
|
||||
|
@ -1070,30 +862,16 @@ bool
|
|||
WebRTCSession::addVideoPipeline(int vp8PayloadType)
|
||||
{
|
||||
// allow incoming video calls despite localUser having no webcam
|
||||
if (videoSources_.empty())
|
||||
if (!devices_.haveCamera())
|
||||
return !isOffering_;
|
||||
|
||||
QSharedPointer<UserSettings> settings = ChatPage::instance()->userSettings();
|
||||
std::string cameraSetting = settings->camera().toStdString();
|
||||
auto it = std::find_if(videoSources_.cbegin(),
|
||||
videoSources_.cend(),
|
||||
[&cameraSetting](const auto &s) { return s.name == cameraSetting; });
|
||||
if (it == videoSources_.cend()) {
|
||||
nhlog::ui()->error("WebRTC: unknown camera: {}", cameraSetting);
|
||||
std::pair<int, int> resolution;
|
||||
std::pair<int, int> frameRate;
|
||||
GstDevice *device = devices_.videoDevice(resolution, frameRate);
|
||||
if (!device)
|
||||
return false;
|
||||
}
|
||||
|
||||
std::string resSetting = settings->cameraResolution().toStdString();
|
||||
const std::string &res = resSetting.empty() ? it->caps.front().resolution : resSetting;
|
||||
std::string frSetting = settings->cameraFrameRate().toStdString();
|
||||
const std::string &fr = frSetting.empty() ? it->caps.front().frameRates.front() : frSetting;
|
||||
auto resolution = tokenise(res, 'x');
|
||||
auto frameRate = tokenise(fr, '/');
|
||||
nhlog::ui()->debug("WebRTC: camera: {}", cameraSetting);
|
||||
nhlog::ui()->debug("WebRTC: camera resolution: {}x{}", resolution.first, resolution.second);
|
||||
nhlog::ui()->debug("WebRTC: camera frame rate: {}/{}", frameRate.first, frameRate.second);
|
||||
|
||||
GstElement *source = gst_device_create_element(it->device, nullptr);
|
||||
GstElement *source = gst_device_create_element(device, nullptr);
|
||||
GstElement *videoconvert = gst_element_factory_make("videoconvert", nullptr);
|
||||
GstElement *capsfilter = gst_element_factory_make("capsfilter", "camerafilter");
|
||||
GstCaps *caps = gst_caps_new_simple("video/x-raw",
|
||||
|
@ -1239,111 +1017,6 @@ WebRTCSession::end()
|
|||
emit stateChanged(State::DISCONNECTED);
|
||||
}
|
||||
|
||||
#if GST_CHECK_VERSION(1, 18, 0)
|
||||
void
|
||||
WebRTCSession::startDeviceMonitor()
|
||||
{
|
||||
if (!initialised_)
|
||||
return;
|
||||
|
||||
static GstDeviceMonitor *monitor = nullptr;
|
||||
if (!monitor) {
|
||||
monitor = gst_device_monitor_new();
|
||||
GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
|
||||
gst_device_monitor_add_filter(monitor, "Audio/Source", caps);
|
||||
gst_caps_unref(caps);
|
||||
caps = gst_caps_new_empty_simple("video/x-raw");
|
||||
gst_device_monitor_add_filter(monitor, "Video/Source", caps);
|
||||
gst_caps_unref(caps);
|
||||
|
||||
GstBus *bus = gst_device_monitor_get_bus(monitor);
|
||||
gst_bus_add_watch(bus, newBusMessage, nullptr);
|
||||
gst_object_unref(bus);
|
||||
if (!gst_device_monitor_start(monitor)) {
|
||||
nhlog::ui()->error("WebRTC: failed to start device monitor");
|
||||
return;
|
||||
}
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
void
|
||||
WebRTCSession::refreshDevices()
|
||||
{
|
||||
#if GST_CHECK_VERSION(1, 18, 0)
|
||||
return;
|
||||
#else
|
||||
if (!initialised_)
|
||||
return;
|
||||
|
||||
static GstDeviceMonitor *monitor = nullptr;
|
||||
if (!monitor) {
|
||||
monitor = gst_device_monitor_new();
|
||||
GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
|
||||
gst_device_monitor_add_filter(monitor, "Audio/Source", caps);
|
||||
gst_caps_unref(caps);
|
||||
caps = gst_caps_new_empty_simple("video/x-raw");
|
||||
gst_device_monitor_add_filter(monitor, "Video/Source", caps);
|
||||
gst_caps_unref(caps);
|
||||
}
|
||||
|
||||
auto clearDevices = [](auto &sources) {
|
||||
std::for_each(
|
||||
sources.begin(), sources.end(), [](auto &s) { gst_object_unref(s.device); });
|
||||
sources.clear();
|
||||
};
|
||||
clearDevices(audioSources_);
|
||||
clearDevices(videoSources_);
|
||||
|
||||
GList *devices = gst_device_monitor_get_devices(monitor);
|
||||
if (devices) {
|
||||
for (GList *l = devices; l != nullptr; l = l->next)
|
||||
addDevice(GST_DEVICE_CAST(l->data));
|
||||
g_list_free(devices);
|
||||
}
|
||||
emit devicesChanged();
|
||||
#endif
|
||||
}
|
||||
|
||||
std::vector<std::string>
|
||||
WebRTCSession::getDeviceNames(bool isVideo, const std::string &defaultDevice) const
|
||||
{
|
||||
return isVideo ? deviceNames(videoSources_, defaultDevice)
|
||||
: deviceNames(audioSources_, defaultDevice);
|
||||
}
|
||||
|
||||
std::vector<std::string>
|
||||
WebRTCSession::getResolutions(const std::string &cameraName) const
|
||||
{
|
||||
std::vector<std::string> ret;
|
||||
if (auto it = std::find_if(videoSources_.cbegin(),
|
||||
videoSources_.cend(),
|
||||
[&cameraName](const auto &s) { return s.name == cameraName; });
|
||||
it != videoSources_.cend()) {
|
||||
ret.reserve(it->caps.size());
|
||||
for (const auto &c : it->caps)
|
||||
ret.push_back(c.resolution);
|
||||
}
|
||||
return ret;
|
||||
}
|
||||
|
||||
std::vector<std::string>
|
||||
WebRTCSession::getFrameRates(const std::string &cameraName, const std::string &resolution) const
|
||||
{
|
||||
if (auto i = std::find_if(videoSources_.cbegin(),
|
||||
videoSources_.cend(),
|
||||
[&](const auto &s) { return s.name == cameraName; });
|
||||
i != videoSources_.cend()) {
|
||||
if (auto j =
|
||||
std::find_if(i->caps.cbegin(),
|
||||
i->caps.cend(),
|
||||
[&](const auto &s) { return s.resolution == resolution; });
|
||||
j != i->caps.cend())
|
||||
return j->frameRates;
|
||||
}
|
||||
return {};
|
||||
}
|
||||
|
||||
#else
|
||||
|
||||
bool
|
||||
|
@ -1400,25 +1073,4 @@ void
|
|||
WebRTCSession::end()
|
||||
{}
|
||||
|
||||
void
|
||||
WebRTCSession::refreshDevices()
|
||||
{}
|
||||
|
||||
std::vector<std::string>
|
||||
WebRTCSession::getDeviceNames(bool, const std::string &) const
|
||||
{
|
||||
return {};
|
||||
}
|
||||
|
||||
std::vector<std::string>
|
||||
WebRTCSession::getResolutions(const std::string &) const
|
||||
{
|
||||
return {};
|
||||
}
|
||||
|
||||
std::vector<std::string>
|
||||
WebRTCSession::getFrameRates(const std::string &, const std::string &) const
|
||||
{
|
||||
return {};
|
||||
}
|
||||
#endif
|
||||
|
|
|
@ -5,6 +5,7 @@
|
|||
|
||||
#include <QObject>
|
||||
|
||||
#include "CallDevices.h"
|
||||
#include "mtx/events/voip.hpp"
|
||||
|
||||
typedef struct _GstElement GstElement;
|
||||
|
@ -59,13 +60,6 @@ public:
|
|||
|
||||
void setTurnServers(const std::vector<std::string> &uris) { turnServers_ = uris; }
|
||||
|
||||
void refreshDevices();
|
||||
std::vector<std::string> getDeviceNames(bool isVideo,
|
||||
const std::string &defaultDevice) const;
|
||||
std::vector<std::string> getResolutions(const std::string &cameraName) const;
|
||||
std::vector<std::string> getFrameRates(const std::string &cameraName,
|
||||
const std::string &resolution) const;
|
||||
|
||||
void setVideoItem(QQuickItem *item) { videoItem_ = item; }
|
||||
QQuickItem *getVideoItem() const { return videoItem_; }
|
||||
|
||||
|
@ -76,7 +70,6 @@ signals:
|
|||
const std::vector<mtx::events::msg::CallCandidates::Candidate> &);
|
||||
void newICECandidate(const mtx::events::msg::CallCandidates::Candidate &);
|
||||
void stateChanged(webrtc::State);
|
||||
void devicesChanged();
|
||||
|
||||
private slots:
|
||||
void setState(webrtc::State state) { state_ = state; }
|
||||
|
@ -84,6 +77,7 @@ private slots:
|
|||
private:
|
||||
WebRTCSession();
|
||||
|
||||
CallDevices &devices_;
|
||||
bool initialised_ = false;
|
||||
bool haveVoicePlugins_ = false;
|
||||
bool haveVideoPlugins_ = false;
|
||||
|
@ -101,7 +95,6 @@ private:
|
|||
bool startPipeline(int opusPayloadType, int vp8PayloadType);
|
||||
bool createPipeline(int opusPayloadType, int vp8PayloadType);
|
||||
bool addVideoPipeline(int vp8PayloadType);
|
||||
void startDeviceMonitor();
|
||||
|
||||
public:
|
||||
WebRTCSession(WebRTCSession const &) = delete;
|
||||
|
|
Loading…
Reference in a new issue