diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index a3900b482602127627e96c58c7341f5515b76c19..afa9fc22af286ade696678823b20c06ab95a732c 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -21,6 +21,9 @@ WebRTCSession::WebRTCSession()
{
qRegisterMetaType<WebRTCSession::State>();
connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
+#if GST_CHECK_VERSION(1, 18, 0)
+ init();
+#endif
}
bool
@@ -78,7 +81,11 @@ 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 +102,65 @@ namespace {
bool isoffering_;
std::string localsdp_;
std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
+std::vector<std::pair<std::string, GstDevice *>> 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<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);
+ 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 +564,18 @@ 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 +668,32 @@ 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 +707,42 @@ 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<std::string>
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<std::string> 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 +793,10 @@ void
WebRTCSession::refreshDevices()
{}
+void
+WebRTCSession::startDeviceMonitor()
+{}
+
std::vector<std::string>
WebRTCSession::getAudioSourceNames(const std::string &)
{
diff --git a/src/WebRTCSession.h b/src/WebRTCSession.h
index 8e78812f1004a6c1d23577844b801ff35753e5d2..653ec2cf76103a962df3b73388a5e5d7b275ca78 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<std::string> turnServers_;
- GList *audioSources_ = nullptr;
int audioSourceIndex_ = -1;
bool startPipeline(int opusPayloadType);
bool createPipeline(int opusPayloadType);
void refreshDevices();
+ void startDeviceMonitor();
public:
WebRTCSession(WebRTCSession const &) = delete;