> -----Original Message----- > From: ffmpeg-user [mailto:[email protected]] On > Behalf Of Carl Eugen Hoyos > Sent: Thursday, July 07, 2016 2:44 AM > To: [email protected] > Subject: Re: [FFmpeg-user] rtp source stopped working > upgrading ffmpeg from3.0.0 to 3.1.1 > > Mark Hassman <mark <at> hassman.org> writes: > > > Reading source input from rtp no longer works. > > Please either explain how I can reproduce (explain it as if I > had never heard of rtp) or tell us which change introduced the issue. > > Thank you, Carl Eugen
Hi, Apologies on delayed response.. The rtp stream is generated through an automated workflow. I've created a custom test page capable of generating it: https://dev01.privatecircle.com:8091/ffmpeg/.. add your local ip/port and click start, rtp will be streamed to you. The test page leverages webrtc and works in chrome/firefox. fyi.. i increased my debugging level, issue is locating i-frame in h.264 input.. but, this same rtp stream works fine with ffmpeg v3.0. [h264 @ 0x22c18c0] non-existing PPS 2 referenced [h264 @ 0x22c18c0] nal_unit_type: 1, nal_ref_idc: 3 [h264 @ 0x22c18c0] non-existing PPS 2 referenced [h264 @ 0x22c18c0] decode_slice_header error [h264 @ 0x22c18c0] no frame! Thoughts? Thnx! -Mark _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
