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Add audio input device selector

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This commit is contained in:
trilene 2020-08-05 17:56:44 -04:00
parent 02dfc8039f
commit df65093374
10 changed files with 231 additions and 22 deletions
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116
src/WebRTCSession.cpp
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@ -487,23 +487,74 @@ WebRTCSession::startPipeline(int opusPayloadType)
return true;
}
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
bool
WebRTCSession::createPipeline(int opusPayloadType)
{
std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
"autoaudiosrc ! volume name=srclevel ! audioconvert ! "
"audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS +
std::to_string(opusPayloadType) + " ! webrtcbin.");
int nSources = audioSources_ ? g_list_length(audioSources_) : 0;
if (nSources == 0) {
nhlog::ui()->error("WebRTC: no audio sources");
return false;
}
webrtc_ = nullptr;
GError *error = nullptr;
pipe_ = gst_parse_launch(pipeline.c_str(), &error);
if (error) {
nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
g_error_free(error);
if (audioSourceIndex_ < 0 || audioSourceIndex_ >= nSources) {
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 *volume = gst_element_factory_make("volume", "srclevel");
GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
GstElement *queue1 = gst_element_factory_make("queue", nullptr);
GstElement *opusenc = gst_element_factory_make("opusenc", nullptr);
GstElement *rtp = gst_element_factory_make("rtpopuspay", nullptr);
GstElement *queue2 = gst_element_factory_make("queue", nullptr);
GstElement *capsfilter = gst_element_factory_make("capsfilter", nullptr);
GstCaps *rtpcaps = gst_caps_new_simple("application/x-rtp",
"media",
G_TYPE_STRING,
"audio",
"encoding-name",
G_TYPE_STRING,
"OPUS",
"payload",
G_TYPE_INT,
opusPayloadType,
nullptr);
g_object_set(capsfilter, "caps", rtpcaps, nullptr);
gst_caps_unref(rtpcaps);
GstElement *webrtcbin = gst_element_factory_make("webrtcbin", "webrtcbin");
g_object_set(webrtcbin, "bundle-policy", GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE, nullptr);
pipe_ = gst_pipeline_new(nullptr);
gst_bin_add_many(GST_BIN(pipe_),
source,
volume,
convert,
resample,
queue1,
opusenc,
rtp,
queue2,
capsfilter,
webrtcbin,
nullptr);
if (!gst_element_link_many(source,
volume,
convert,
resample,
queue1,
opusenc,
rtp,
queue2,
capsfilter,
webrtcbin,
nullptr)) {
nhlog::ui()->error("WebRTC: failed to link pipeline elements");
end();
return false;
}
@ -541,3 +592,42 @@ WebRTCSession::end()
if (state_ != State::DISCONNECTED)
emit stateChanged(State::DISCONNECTED);
}
void
WebRTCSession::refreshDevices()
{
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);
}
g_list_free_full(audioSources_, g_object_unref);
audioSources_ = gst_device_monitor_get_devices(monitor);
}
std::vector<std::string>
WebRTCSession::getAudioSourceNames(const std::string &defaultDevice)
{
if (!initialised_)
return {};
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) {
// move default device to top of the list
std::swap(audioSources_->data, l->data);
std::swap(ret.front(), ret.back());
}
}
return ret;
}