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• When I append a silent audio (mp3) to an existing list of audio it garbles the final audio ?

  6 février 2020, par Marie

  After several hours I have narrowed down the issue with the garbled audio to be the 2-seconds silence audio mp3 I am appending (I think I had produced it once with Wavelab)

  However, I tried using ffmpeg according to a post to produce a similar 2 seconds audio but it too will corrupt/garble/chop voice in the final concatenation of audio files.

  ffmpeg -f lavfi -i anullsrc=r=44100:cl=mono -t 2 -q:a 9 -acodec libmp3lame SILENCE_2sec.MP3

  I typically will have several audio files to concatenate together but for simplicity I have able to narrow it to a couple of files simplifying to the following script. A simple Windows batch file you should be able to use and reproduce the issue at your end.

  rem 
  
  rem  
  
  SET EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"
  
  
  
  SET ROOTPATH=.\
  
  
  
  SET IN_FILE="%ROOTPATH%MyList.txt"
  
  
  
  ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
  
  ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
  
  
  
  SET OPTIONS= -f concat -safe 0 -i  %IN_FILE%  -c copy -y
  
  
  
  SET OUT_FILE="%ROOTPATH%CONCATENATED_AUDIO_2.MP3"
  
  
  
  SET INFO_FILE="INFO.TXT"
  
  
  
  %EXE% %OPTIONS%  %OUT_FILE% 1> %INFO_FILE% 2>&1 
  
  
  
  ECHO ======================== >> %INFO_FILE%
  
  ECHO IN_FILE=%IN_FILE%  >> %INFO_FILE% 
  
  ECHO EXE=%EXE%  >> %INFO_FILE% 
  
  ECHO OPTIONS=%OPTIONS%  >> %INFO_FILE% 
  
  ECHO ======================== >> %INFO_FILE%

  Here is the console info output from the ffmpeg, let me know if you need other output include ones from ffprobe

  ffmpeg version git-2020-01-10-3d894db Copyright (c) 2000-2020 the FFmpeg developers
  
    built with gcc 9.2.1 (GCC) 20191125
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  
    libavutil      56. 38.100 / 56. 38.100
  
    libavcodec     58. 65.103 / 58. 65.103
  
    libavformat    58. 35.101 / 58. 35.101
  
    libavdevice    58.  9.103 / 58.  9.103
  
    libavfilter     7. 70.101 /  7. 70.101
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  [mp3 @ 000000000036af80] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, concat, from '.\MyList.txt':
  
    Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
      Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
  
  Output #0, mp3, to '.\CONCATENATED_AUDIO_2.MP3':
  
    Metadata:
  
      TSSE            : Lavf58.35.101
  
      Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [mp3 @ 0000000000372d00] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 17280 >= 17255
  
  size=      11kB time=00:00:02.73 bitrate=  33.2kbits/s speed=2.73e+03x    
  
  video:0kB audio:11kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.137446%
  
  ======================== 
  
  IN_FILE=".\MyList.txt"   
  
  EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"   
  
  OPTIONS= -f concat -safe 0 -i  ".\MyList.txt"  -c copy -y   
  
  ========================  

  I believe I am running FFmpeg 4.2.1, recently installed (20200112)

  You may produce the HELLO.mp3 by saving the following link

  https://translate.google.com.vn/translate_tts?en=UTF-8&q=Hello+&tl=en&client=tw-ob

  FYI, I am still a novice of ffmpeg and using it more like a black box with the help I received in this very super forum.
  Please be as explicit as you can with command line options on how I can fix this issue.
  Thank you.

  Additional Hints Debugging :

  If I append more files after the silence audio it seems that the silence audio impacts (garbles, chops) the previous audio.
  You may try the following for the list of audio files input.

  ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
  
  ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
  
  ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt
  
  ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt

  I typically add one or more silence file to derive a post silence effect after the actual audio. That’s my current logic. However if you have an alternative to appending a silence in the process of concatenating several audio files or appending x-seconds silence to an existing audio file. I can use that method as well from my coding.

  Thank you.

• Force GStreamer for IOS to use ffmpeg library outside of the framework

  24 avril 2014, par Michelle Cannon

  Gstreamer 1.0 for IOS is delivered in a static framework, the source to build the framework is around 1.2g , this framework is huge and tries to provide for any decoding server scenario you may have. Trouble is it tries to do to much and IMHO not enough thought was put into the IOS port.

  Here’s the problem we have an application that uses the GSTreamer avdec_h264 plugin for displaying an RTP over UDP stream . This works rather well. recently we were required to do some special recording functions so we introduced an api that had its own version of ffmpeg. Gstreamer has Libav compiled into the framework. When we place our api into the application with the gst_IOS_RESTICTED_PLUGINS disabled the code runs fine when we introduce the GStreamer.framework into the application code similar to that shown below fails with a protocol not found error.

  The problem is that the internal version of libav seems to disable all the protocols that ffmpeg supplies. because GSTreamer uses its own custom AVIO callback based on ffmpeg pipe protocol.

  According to Gstreamer support that has been somewhat helpful

  ) Add a new recipe with the libav version you want to use and disable the build of the internal libav in gst-libav-1.0 with :
  configure_options = ’—with-system-libav’

  You might need to comment out this part to prevent libav being packaged in the framewiork or make sure that your libav recipe creates these files in the correct place to include them in the framework :

  42 for f in [’libavcodec’, ’libavformat’, ’libavutil’, ’libswscale’] :
  43 for ext in [’.a’, ’.la’] :
  44 path = os.path.join(’lib’, f + ext)
  45 self.files_plugins_codecs_restricted_devel.append(path)

  2) Update libav submodule gst-libav to use the correct version you need.

  https://bugs.freedesktop.org/show_bug.cgi?id=77399

  The first method didn’t work , the recipe kept getting overwritten even after applying a patch for a bug fix that was made as result of this bug report.

  And I have no idea how to do the second method. Which is what I’d like some help with.

  Has anyone with GStreamer 1.0 for iOS

  1) Built the get-libav plugin against an external to to the framework set of ffmpeg static libs (.a)

  2) built the internal libav to allow for RTP , UDP and TCP protocols, or written a custom AVIO callback using the FFPipe protocol.

  3) just managed to somehow get the below code working with GStreamer.

  I don’t ask many questions, I’ve kind of implemented all kinds of encoders/decoders using ffmpeg , lib555 and a few hardware decoders. But this GStreamer issue is causing me more sleepless nights than I’ve had in a long time.

  AVFormatContext * avctx;
  
  avctx = avformat_alloc_context();
  
  
  
  av_register_all();
  
  avformat_network_init();
  
  avcodec_register_all();
  
  avdevice_register_all();
  
  
  
  
  
  
  
  // Set the RTSP Options
  
  AVDictionary *opts = 0;
  
  
  
  av_dict_set(&opts, "rtsp_transport", "udp", 0);
  
  
  
  
  
  int err = 0;
  
  err = avformat_open_input(&avctx, "rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov", NULL, &opts);
  
  av_dict_free(&opts);
  
  if (err) {
  
      NSLog(@"Error: Could not open stream: %d", err);
  
      char errbuf[400];
  
      av_strerror(err,errbuf,400);
  
      NSLog(@"%s failed with error %s","avformat_open_input",errbuf);
  
  
  
  
  
  
  
  }
  
  else {
  
      NSLog(@"Opened stream");
  
  }
  
  
  
  err = avformat_find_stream_info(avctx, NULL);
  
  if( err < 0)
  
  {
  
      char errbuf[400];
  
      av_strerror(err,errbuf,400);
  
      NSLog(@"%s failed with error %s","avformat_find_stream_info",errbuf);
  
      return ;
  
  }else {
  
       NSLog(@"found stream info");
  
  }

  }

• ffmpeg library - why does the Bitrate parameter change on encoding, and how do I force to preserve it ?

  22 avril 2021, par QRrabbit

  Hello forum and all the members of the community !

  


  I have this question on ffmpeg library, why does the bitrate parameter not retained after encoding, even though I explicitly specify the desired rate.

  


  Input file ffprobe :

  


    Duration: 00:00:10.01, start: 0.000000, bitrate: 534719 kb/s
    Stream #0:0(eng): Video: qtrle (rle  / 0x20656C72), argb(progressive), 1920x1080, 533881 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 30k tbn, 30k tbc (default)
    Metadata:
      creation_time   : 2021-04-13T16:35:16.000000Z
      handler_name    : Apple Video Media Handler
      encoder         : Animation
      timecode        : 00:00:00;00


  


  Here's the command I run :

  


  ffmpeg -i input.mov -map 0:a? -map 0:s? -pix_fmt argb -b:v 533881667 -maxrate 533881667 -minrate 533881667 -r 29.97 -top 1 -color_range 1 -colorspace 1 -color_primaries 1 -color_trc bt709 -map_metadata 0 -c:a copy -timecode 00:00:00.00 -c:v qtrle -c:s copy output.mov


  


  Output file ffprobe :

  


    Metadata:
    major_brand     : qt  
    minor_version   : 512
    compatible_brands: qt  
    creation_time   : 2021-04-13T16:35:16.000000Z
  Duration: 00:00:10.01, start: 0.000000, bitrate: 100126 kb/s
    Stream #0:0: Video: qtrle (rle  / 0x20656C72), argb(progressive), 1920x1080, 100133 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 11988 tbn, 11988 tbc (default)
    Metadata:
      creation_time   : 2021-04-13T16:35:16.000000Z
      handler_name    : VideoHandler
      encoder         : Lavf58.58.100
      timecode        : 00:00:00;00


  


  As it is visible from the ffprobe from above, I go from :

  


  The Input's file bitrate : 534719 kb/s
to Output bitrate : 100126 kb/s