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Merge branch 'master' of https://github.com/Nheko-Reborn/nheko

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pull/285/head
Jussi Kuokkanen 4 years ago
parent 1f71f7227a 791a01487b
commit 438dcd3c5e
2 changed files with 121 additions and 24 deletions
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  1. 142
      src/WebRTCSession.cpp
  2. 3
      src/WebRTCSession.h

142
src/WebRTCSession.cpp
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@ -21,6 +21,7 @@ WebRTCSession::WebRTCSession()
{
qRegisterMetaType<WebRTCSession::State>();
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<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 +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<std::string>
WebRTCSession::getAudioSourceNames(const std::string &defaultDevice)
{
if (!initialised_)
return {};
#if !GST_CHECK_VERSION(1, 18, 0)
refreshDevices();
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) {
#endif
// move default device to top of the list
std::swap(audioSources_->data, l->data);
std::swap(ret.front(), ret.back());
}
}
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(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<std::string>
WebRTCSession::getAudioSourceNames(const std::string &)
{

3
src/WebRTCSession.h
Unescape View File

@ -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;

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