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• ffmpeg radio live transcoding mp3/aac to g722

  13 mai 2017, par p3tter

  I have tried to transcode a radio stream with ffmpeg to g722.
  I get the stream to work and im able to listen to the stream.
  The problem is that the output stream have faster speed than the input stream.
  so the result is not good. I have tried to slow down the speed with atempo without any luck.

  like :

  size=     241kB time=00:00:28.67 bitrate=  68.8kbits/s speed= 1.4x

  this varies from 1.x to 15.x

  Console output :

  c:\ffmpeg\bin>ffmpeg -i http://lyd.nrk.no/nrk_radio_mp3_mp3_l -ac 1 -acodec g722 -f rtp -ab 64k -ar 16k rtp://192.168.0.99:555
  
  ffmpeg version N-85750-ga75ef15 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 6.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  
    libavutil      55. 61.100 / 55. 61.100
  
    libavcodec     57. 93.100 / 57. 93.100
  
    libavformat    57. 72.101 / 57. 72.101
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 88.100 /  6. 88.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
    libpostproc    54.  6.100 / 54.  6.100
  
  Input #0, mp3, from 'http://lyd.nrk.no/nrk_radio_mp3_mp3_l':
  
    Metadata:
  
      icy-name        : NRK mP3
  
      icy-pub         : 1
  
    Duration: N/A, start: 0.000000, bitrate: 96 kb/s
  
      Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 96 kb/s
  
  [udp @ 000000000244bec0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
  
  [udp @ 00000000024781a0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> adpcm_g722 (g722))
  
  Press [q] to stop, [?] for help
  
  Output #0, rtp, to 'rtp://192.168.0.99:555':
  
    Metadata:
  
      icy-name        : NRK mP3
  
      icy-pub         : 1
  
      encoder         : Lavf57.72.101
  
      Stream #0:0: Audio: adpcm_g722 (g722), 16000 Hz, mono, s16, 64 kb/s
  
      Metadata:
  
        encoder         : Lavc57.93.100 g722
  
  SDP:
  
  v=0
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=No Name
  
  c=IN IP4 192.168.0.99
  
  t=0 0
  
  a=tool:libavformat 57.72.101
  
  m=audio 555 RTP/AVP 9
  
  b=AS:64
  
  
  
  size=     413kB time=00:00:49.17 bitrate=  68.8kbits/s speed=1.43x

  Does anyone know what im doing wrong ?
  Thanks

• Emulate radio stream metadata

  17 juin 2018, par IllidanS4

  I work with an application that can connect to an audio stream via HTTP and play the audio. It can also connect to SHOUTcast and Icecast radios and play the streams, but for some radios, when the format is supported, it can read special metadata as well, like the name of the current song.

  I would like to use ffmpeg to produce an MP3 stream that contains the specific metadata this application understands. I know it is probably something specific to SHOUTcast or Icecast and not to the format itself, but I don’t know what exactly it is that the application can decode.

  If ffmpeg cannot produce such a stream (possibly because it is part of the HTTP metadata), I would welcome any details on how the metadata is encoded and how can I generate it (in PHP for example).

• Avoiding ffmpeg audio drift for live dash output

  22 mai 2021, par mdale

  Have people experienced playback drift with ffmpeg based dash segment generation ? For example :

  


  ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i AVSyncTest.mp4 \
-af "aresample=async=1:min_hard_comp=0.100000:first_pts=0" \
-c:v:0 libx264 \
-pix_fmt:0 yuv420p \
-preset:0 medium \
-a53cc:0 1 \
-nal-hrd:0 cbr \
-x264opts:0 scenecut=-1:rc_lookahead=0 -b:v:0 5000k -bufsize:0 500k \
-force_key_frames:0 "expr:gte(t,n_forced*2)" \
-bf:0 8 \
-r 30 \
-c:a:0 aac  -ar 48000 \
-b:a:1 96k  \
-f tee -map 0:v \
-map 0:a "[f=dash:media_seg_name='chunk-stream_\$RepresentationID\$-\$Number%05d\$.mp4':init_seg_name='init-stream_\$RepresentationID\$.mp4':seg_duration=2:utc_timing_url=https\\\://time.akamai.com\\?iso:remove_at_exit=0:use_timeline=0:http_user_agent=ffmpeg_encoder.0:streaming=1:index_correction=1:timeout=1:dash_segment_type=mp4:method=PUT:http_persistent=1:adaptation_sets='id=0,streams=v\:0 id=1,streams=a' ]http://localhost:5000/out.mpd "


  


  After a bit of playback it starts audio starts to drift. Analysis of the moof (with tools like mp4box) shows 1024 sample size with 93 Moofs with Moof->traf->default_sample_duration=1024 at 48000 sample rate giving segment duration of 1.984 seconds with every 3rd segment or so with 92 Moofs with duration of 1.962 … While each segment starts aligned ( if you reset playback you get alignment) it drifts against continuous playback.

  


  Other examples like Akamai’s public stream. show 94 Moofs or 2.005 duration with every 3rd segment or so at the 1.984 duration. (what one would want)

  


  Akamai’s (presumably ffmepg based) stream is averaging out to the 2s target where the above ffmpeg command is not …