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• ffmpeg : remove dead call to av_parser_change()

  21 mars 2018, par James Almer

  ffmpeg : remove dead call to av_parser_change()
  It's been a noop for years, and it's been argued that in-band headers
  
  should not be forcedly removed without the user's explicit request.
  Also, as the FIXME line stated, this is a job for a bitstream filter
  
  like extract_extradata, remove_extradata, dump_extradata, and
  
  filter_units.
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] fftools/ffmpeg.c
  • [DH] fftools/ffmpeg.h

• M4a (mp4) audio file encoded with pydub+ffmpeg doesn't play on Android

  6 juillet 2020, par Evelyn

  I have a python script to split up some wav files and export to m4a using pydub. I'm able to get these files to play on several devices, but not on an Android device (using Google Pixel 3). When I try encoding with straight ffmpeg in terminal, that works fine on the Android device.

  


  What is the difference in these two files, and since pydub is using ffmpeg, what do I need to change to make it do exactly the same as the ffmpeg command ?

  


  Not working

  


  from pydub import AudioSegment
>>> audio = AudioSegment.from_wav("input.wav")
>>> slice = audio[1000:3000]
>>> slice.export("pydub_export.m4a", format="mp4", parameters=["-ac", "1", "-c:a", "libfdk_aac", "-profile:a", "aac_he", "-vbr", "2"])


  


  mediainfo output :

  


  General
Complete name                            : pydub_export.m4a
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom (isom/iso2/mp41)
File size                                : 11.7 KiB
Duration                                 : 2 s 115 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 45.1 kb/s
Writing application                      : Lavf58.29.100

Audio
ID                                       : 1
Format                                   : AAC LC SBR
Format/Info                              : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name                          : HE-AAC
Format settings                          : NBC
Codec ID                                 : mp4a-40-5
Duration                                 : 2 s 115 ms
Duration_LastFrame                       : -22 ms
Bit rate mode                            : Constant
Bit rate                                 : 41.4 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 44.1 kHz
Frame rate                               : 21.533 FPS (2048 SPF)
Compression mode                         : Lossy
Stream size                              : 10.7 KiB (92%)
Default                                  : Yes
Alternate group                          : 1


  


  Working

  


  $ffmpeg -i input.wav -acodec copy -ss 1 -to 3 input_slice.wav
$ffmpeg -i input_slice.wav -ac 1 -c:a libfdk_aac -profile:a aac_he -vbr 2 ffmpeg_export.m4a


  


  mediainfo output :

  


  General
Complete name                            : ffmpeg_export.m4a
Format                                   : MPEG-4
Format profile                           : Apple audio with iTunes info
Codec ID                                 : M4A  (isom/iso2)
File size                                : 11.2 KiB
Duration                                 : 2 s 112 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 43.6 kb/s
Writing application                      : Lavf58.29.100

Audio
ID                                       : 1
Format                                   : AAC LC SBR
Format/Info                              : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name                          : HE-AAC
Format settings                          : NBC
Codec ID                                 : mp4a-40-5
Duration                                 : 2 s 112 ms
Duration_LastFrame                       : -25 ms
Bit rate mode                            : Constant
Bit rate                                 : 39.9 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 44.1 kHz
Frame rate                               : 21.533 FPS (2048 SPF)
Compression mode                         : Lossy
Stream size                              : 10.3 KiB (91%)
Default                                  : Yes
Alternate group                          : 1


  


  I already tried moving metadata to the front with -movflags faststart on the broken file and it didn't make a difference.

  


• How can I have ffmpeg receive both video and audio over RTP ?

  23 mai 2018, par KallDrexx

  I am trying to instruct FFMPEG to receive h264 video and aac audio via RTP using out of band session initialization.

  To do that I have the following local SDP :

  v=0
  
  o=sb
  
  s=-
  
  t=0 0
  
  c=IN IP4 127.0.0.1
  
  m=video 12100 RTP/AVP 96
  
  a=rtpmap:96 H264/90000
  
  m=audio 12101 RTP/AVP 97
  
  a=rtpmap:97 MPEG4-GENERIC/44100/2

  When I load ffmpeg with :

  ffmpeg -loglevel debug -protocol_whitelist "file,rtp,udp" -i .\test.sdp -strict -2 test.flv

  I get the following error :

  [udp @ 0000022d7fdafe80] bind failed: Error number -10048 occurred
  
  [AVIOContext @ 0000022d7fd84900] Statistics: 154 bytes read, 0 seeks
  
  .\test.sdp: Invalid data found when processing input

  Confused by that error code I loaded it up on a Linux VM and the bind error I got was Address already in use.

  I tried changing both of those port numbers all around and kept getting that error. Finally I removed one of the media streams from the SDP so it ONLY had video or ONLY had audio and no binding error occurred.

  How can I get ffmpeg to bind to multiple RTP ports for RTP ingestion ?