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Android: Add 25% FEC to the video stream
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1 changed files with 14 additions and 5 deletions
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@ -310,6 +310,19 @@ on_negotiation_needed (GstElement * element, WebRTC * webrtc)
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g_signal_emit_by_name (webrtc->webrtcbin, "create-offer", NULL, promise);
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}
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static void
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add_fec_to_offer (GstElement * webrtc)
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{
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GstWebRTCRTPTransceiver *trans = NULL;
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/* A transceiver has already been created when a sink pad was
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* requested on the sending webrtcbin */
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g_signal_emit_by_name (webrtc, "get-transceiver", 0, &trans);
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g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED,
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"fec-percentage", 25, "do-nack", FALSE, NULL);
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}
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#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload=100"
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#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload=101"
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@ -336,11 +349,7 @@ start_pipeline (WebRTC * webrtc)
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webrtc->webrtcbin = gst_bin_get_by_name (GST_BIN (webrtc->pipe), "sendrecv");
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g_assert (webrtc->webrtcbin != NULL);
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pad = gst_element_get_static_pad (webrtc->webrtcbin, "sink_0");
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gst_util_set_object_arg (G_OBJECT (pad), "fec-type", "ulp-red");
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g_object_set (pad, "do-nack", FALSE, NULL);
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gst_object_unref (pad);
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add_fec_to_offer (webrtc->webrtcbin);
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/* This is the gstwebrtc entry point where we create the offer and so on. It
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* will be called when the pipeline goes to PLAYING. */
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