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• Should libavcodec handle non-packed/non-byte-aligned mp3 packets ?

  27 août 2015, par Rhythmic Fistman

  I’m transcoding shoutcast streams using libavcodec and I noticed that during the day I get a few 10s’ of packets that it cannot decode without some help :

  1. Leading zero bytes before a packet sync word, e.g. 0x0000fffa

  2. non-byte aligned syncword, e.g 0x82a0fffa

  In the first case I have to toss out the zero bytes and in the second I have to shift the data up by an odd number of nibbles.

  I don’t have a watertight knowledge of the mp3 spec (is there one ?), but my impression was that the packet format was a bit oriented, not byte, so both types of packets look right to me.

  So who is in the wrong here ?

  Is it libavcodec for not parsing valid packets ?

  Or the stream encoder for producing non-standard packets ?

• avformat/framecrcenc : List types and checksums for for side data

  17 mai, par Michael Niedermayer

  avformat/framecrcenc : List types and checksums for for side data
  This allows detecting changes and regressions in side data related code, same as what
  
  framecrc does for before already for packet data itself.
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/framecrcenc.c
  • [DH] tests/ref/fate/autorotate
  • [DH] tests/ref/fate/cover-art-mp3-id3v2-remux
  • [DH] tests/ref/fate/ffmpeg-bsf-input
  • [DH] tests/ref/fate/ffmpeg-spec-disposition
  • [DH] tests/ref/fate/force_key_frames-source
  • [DH] tests/ref/fate/force_key_frames-source-drop
  • [DH] tests/ref/fate/force_key_frames-source-dup
  • [DH] tests/ref/fate/gapless-mp3
  • [DH] tests/ref/fate/h264_redundant_pps-side_data
  • [DH] tests/ref/fate/id3v2-priv-remux
  • [DH] tests/ref/fate/matroska-hdr10-plus-remux
  • [DH] tests/ref/fate/matroska-ogg-opus-remux
  • [DH] tests/ref/fate/matroska-opus-remux
  • [DH] tests/ref/fate/matroska-vp8-alpha-remux
  • [DH] tests/ref/fate/mov-cover-image
  • [DH] tests/ref/fate/segment-mp4-to-ts
  • [DH] tests/ref/fate/shortest
  • [DH] tests/ref/fate/webm-hdr10-plus-remux
  • [DH] tests/ref/fate/webm-webvtt-remux

• Convert RTP (OPUS) stream to HLS (AAC) stream using ffmpeg or gstreamer

  27 décembre 2020, par Gagan Bhat

  I currently have an RTP protocol stream in OPUS codec playing locally on rtp ://127.0.0.1:5006

  


  I would like to convert this stream into an HLS protocol with AAC codec (or others if easier) so that it is more accessible to devices with just a browser.

  


  I know that ffmpeg and gstreamer are capable of this but I'm just lost among the various arguments/parameters.

  


  Currently, I have an SDP file that describes my stream (unsure if this is correct, I wrote it after just googling/reading the spec)

  


  v=0
t=0 0
m=audio 5006 RTP/AVP 98
c=IN IP4 127.0.0.1
a=recvonly
a=rtpmap:98 opus/48000/2
a=fmtp:98 stereo=0; sprop-stereo=0; useinbandfec=1c


  


  Any ideas ?