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Android: Add 25% FEC to the video stream

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This commit is contained in:
Jan Schmidt 2019-09-14 19:12:10 +10:00 committed by Matthew Waters
parent 68f30a2431
commit 46ea108b5e
1 changed files with 14 additions and 5 deletions
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19
webrtc/android/app/src/main/jni/webrtc.c
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@ -310,6 +310,19 @@ on_negotiation_needed (GstElement * element, WebRTC * webrtc)
g_signal_emit_by_name (webrtc->webrtcbin, "create-offer", NULL, promise); g_signal_emit_by_name (webrtc->webrtcbin, "create-offer", NULL, promise);
} }
static void
add_fec_to_offer (GstElement * webrtc)
{
GstWebRTCRTPTransceiver *trans = NULL;
/* A transceiver has already been created when a sink pad was
* requested on the sending webrtcbin */
g_signal_emit_by_name (webrtc, "get-transceiver", 0, &trans);
g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED,
"fec-percentage", 25, "do-nack", FALSE, NULL);
}
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload=100" #define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload=100"
#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload=101" #define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload=101"
@ -336,11 +349,7 @@ start_pipeline (WebRTC * webrtc)
webrtc->webrtcbin = gst_bin_get_by_name (GST_BIN (webrtc->pipe), "sendrecv"); webrtc->webrtcbin = gst_bin_get_by_name (GST_BIN (webrtc->pipe), "sendrecv");
g_assert (webrtc->webrtcbin != NULL); g_assert (webrtc->webrtcbin != NULL);
add_fec_to_offer (webrtc->webrtcbin);
pad = gst_element_get_static_pad (webrtc->webrtcbin, "sink_0");
gst_util_set_object_arg (G_OBJECT (pad), "fec-type", "ulp-red");
g_object_set (pad, "do-nack", FALSE, NULL);
gst_object_unref (pad);
/* This is the gstwebrtc entry point where we create the offer and so on. It /* This is the gstwebrtc entry point where we create the offer and so on. It
* will be called when the pipeline goes to PLAYING. */ * will be called when the pipeline goes to PLAYING. */