diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp index a3900b4..d849783 100644 --- a/src/WebRTCSession.cpp +++ b/src/WebRTCSession.cpp @@ -21,6 +21,7 @@ WebRTCSession::WebRTCSession() { qRegisterMetaType(); connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState); + init(); } bool @@ -78,7 +79,11 @@ WebRTCSession::init(std::string *errorMessage) gst_object_unref(plugin); } - if (!initialised_) { + if (initialised_) { +#if GST_CHECK_VERSION(1, 18, 0) + startDeviceMonitor(); +#endif + } else { nhlog::ui()->error(strError); if (errorMessage) *errorMessage = strError; @@ -95,12 +100,65 @@ namespace { bool isoffering_; std::string localsdp_; std::vector localcandidates_; +std::vector> audioSources_; + +void +addDevice(GstDevice *device) +{ + if (device) { + gchar *name = gst_device_get_display_name(device); + nhlog::ui()->debug("WebRTC: device added: {}", name); + audioSources_.push_back({name, device}); + g_free(name); + } +} + +#if GST_CHECK_VERSION(1, 18, 0) +void +removeDevice(GstDevice *device, bool changed) +{ + if (device) { + if (auto it = std::find_if(audioSources_.begin(), + audioSources_.end(), + [device](const auto &s) { return s.second == device; }); + it != audioSources_.end()) { + nhlog::ui()->debug(std::string("WebRTC: device ") + + (changed ? "changed: " : "removed: ") + "{}", + it->first); + gst_object_unref(device); + audioSources_.erase(it); + } + } +} +#endif gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data) { WebRTCSession *session = static_cast(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); + break; + } + case GST_MESSAGE_DEVICE_REMOVED: { + GstDevice *device; + gst_message_parse_device_removed(msg, &device); + removeDevice(device, false); + 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(); @@ -504,19 +562,18 @@ WebRTCSession::startPipeline(int opusPayloadType) bool WebRTCSession::createPipeline(int opusPayloadType) { - int nSources = audioSources_ ? g_list_length(audioSources_) : 0; - if (nSources == 0) { + if (audioSources_.empty()) { nhlog::ui()->error("WebRTC: no audio sources"); return false; } - if (audioSourceIndex_ < 0 || audioSourceIndex_ >= nSources) { + if (audioSourceIndex_ < 0 || (size_t)audioSourceIndex_ >= audioSources_.size()) { nhlog::ui()->error("WebRTC: invalid audio source index"); return false; } - GstElement *source = gst_device_create_element( - GST_DEVICE_CAST(g_list_nth_data(audioSources_, audioSourceIndex_)), nullptr); + GstElement *source = + gst_device_create_element(audioSources_[audioSourceIndex_].second, 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); @@ -609,6 +666,32 @@ 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); + + 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; + } + } +} + +#else + void WebRTCSession::refreshDevices() { @@ -622,31 +705,42 @@ WebRTCSession::refreshDevices() gst_device_monitor_add_filter(monitor, "Audio/Source", caps); gst_caps_unref(caps); } - g_list_free_full(audioSources_, g_object_unref); - audioSources_ = gst_device_monitor_get_devices(monitor); + + std::for_each(audioSources_.begin(), audioSources_.end(), [](const auto &s) { + gst_object_unref(s.second); + }); + audioSources_.clear(); + GList *devices = gst_device_monitor_get_devices(monitor); + if (devices) { + audioSources_.reserve(g_list_length(devices)); + for (GList *l = devices; l != nullptr; l = l->next) + addDevice(GST_DEVICE_CAST(l->data)); + g_list_free(devices); + } } +#endif std::vector WebRTCSession::getAudioSourceNames(const std::string &defaultDevice) { - if (!initialised_) - return {}; - +#if !GST_CHECK_VERSION(1, 18, 0) refreshDevices(); +#endif + // move default device to top of the list + if (auto it = std::find_if(audioSources_.begin(), + audioSources_.end(), + [&](const auto &s) { return s.first == defaultDevice; }); + it != audioSources_.end()) + std::swap(audioSources_.front(), *it); + std::vector ret; - ret.reserve(g_list_length(audioSources_)); - for (GList *l = audioSources_; l != nullptr; l = l->next) { - gchar *name = gst_device_get_display_name(GST_DEVICE_CAST(l->data)); - ret.emplace_back(name); - g_free(name); - if (ret.back() == defaultDevice) { - // move default device to top of the list - std::swap(audioSources_->data, l->data); - std::swap(ret.front(), ret.back()); - } - } + ret.reserve(audioSources_.size()); + std::for_each(audioSources_.cbegin(), audioSources_.cend(), [&](const auto &s) { + ret.push_back(s.first); + }); return ret; } + #else bool @@ -697,6 +791,10 @@ void WebRTCSession::refreshDevices() {} +void +WebRTCSession::startDeviceMonitor() +{} + std::vector WebRTCSession::getAudioSourceNames(const std::string &) { diff --git a/src/WebRTCSession.h b/src/WebRTCSession.h index 8e78812..653ec2c 100644 --- a/src/WebRTCSession.h +++ b/src/WebRTCSession.h @@ -7,7 +7,6 @@ #include "mtx/events/voip.hpp" -typedef struct _GList GList; typedef struct _GstElement GstElement; class WebRTCSession : public QObject @@ -71,12 +70,12 @@ private: unsigned int busWatchId_ = 0; std::string stunServer_; std::vector turnServers_; - GList *audioSources_ = nullptr; int audioSourceIndex_ = -1; bool startPipeline(int opusPayloadType); bool createPipeline(int opusPayloadType); void refreshDevices(); + void startDeviceMonitor(); public: WebRTCSession(WebRTCSession const &) = delete;