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gst-indent
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parent
804c0c2f5e
commit
ca96b6de86
1 changed files with 5 additions and 2 deletions
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@ -186,7 +186,9 @@ create_receiver_entry (SoupWebsocketConnection * connection)
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G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
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error = NULL;
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receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "
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receiver_entry->pipeline =
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gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
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STUN_SERVER " "
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"v4l2src ! videorate ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! "
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"rtph264pay config-interval=-1 name=payloader ! "
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"application/x-rtp,media=video,encoding-name=H264,payload="
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@ -201,7 +203,8 @@ create_receiver_entry (SoupWebsocketConnection * connection)
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gst_bin_get_by_name (GST_BIN (receiver_entry->pipeline), "webrtcbin");
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g_assert (receiver_entry->webrtcbin != NULL);
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g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers", &transceivers);
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g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers",
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&transceivers);
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g_assert (transceivers != NULL && transceivers->len > 0);
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trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
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trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
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