Commit bbed24d9 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵
Browse files

webrtc: Change H264 examples to use aggregate-mode=zero-latency for best compatibility

The default changed back to none because it broke existing code.

See gst-plugins-good#749

Part-of: <!22>
parent 6378337a
Pipeline #187701 waiting for manual action with stages
in 44 seconds
......@@ -78,7 +78,7 @@ from gi.repository import GstSdp
if DO_VP8:
( encoder, payloader, rtp_encoding) = ( "vp8enc target-bitrate=100000 overshoot=25 undershoot=100 deadline=33000 keyframe-max-dist=1", "rtpvp8pay picture-id-mode=2", "VP8" )
else:
( encoder, payloader, rtp_encoding) = ( "x264enc", "rtph264pay", "H264" )
( encoder, payloader, rtp_encoding) = ( "x264enc", "rtph264pay aggregate-mode=zero-latency", "H264" )
PIPELINE_DESC = '''
webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302
......
......@@ -66,7 +66,7 @@ const VP8: VideoParameter = VideoParameter {
const H264: VideoParameter = VideoParameter {
encoder: "x264enc tune=zerolatency",
encoding_name: "H264",
payloader: "rtph264pay",
payloader: "rtph264pay aggregate-mode=zero-latency",
};
impl std::str::FromStr for VideoParameter {
......
......@@ -179,7 +179,7 @@ create_receiver_entry (SoupWebsocketConnection * connection)
gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
"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 ! "
"rtph264pay config-interval=-1 name=payloader ! "
"rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
"application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);
if (error != NULL) {
......
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