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Merge pull request #265 from trilene/voip
Link GStreamer elements before syncing state
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commit
b7b9cee30e
1 changed files with 1 additions and 2 deletions
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@ -283,11 +283,11 @@ linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe
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GstElement *resample = gst_element_factory_make("audioresample", nullptr);
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GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
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gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
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gst_element_link_many(queue, convert, resample, sink, nullptr);
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gst_element_sync_state_with_parent(queue);
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gst_element_sync_state_with_parent(convert);
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gst_element_sync_state_with_parent(resample);
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gst_element_sync_state_with_parent(sink);
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gst_element_link_many(queue, convert, resample, sink, nullptr);
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queuepad = gst_element_get_static_pad(queue, "sink");
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}
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@ -448,7 +448,6 @@ WebRTCSession::startPipeline(int opusPayloadType)
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g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
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}
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for (const auto &uri : turnServers_) {
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nhlog::ui()->info("WebRTC: setting TURN server: {}", uri);
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gboolean udata;
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