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• Could not write header for output file #0 (incorrect codec parameters ?) : Broken pipe

  23 mai 2020, par Laura

  I'm trying to generate audio waveform from an mp4 file as described here : https://github.com/bbc/audiowaveform.

  



  My mp4 file looks like

  



  <?xml version="1.0" encoding="UTF-8"?>


  



  http://www.w3.org/2001/XMLSchema-instance' ;>
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 


  



  <format filename="/cache/1268.mp4" duration="40.112000" size="10610125">&#xA;    <tag key="major_brand" value="isom"></tag>&#xA;    <tag key="minor_version" value="512"></tag>&#xA;    <tag key="compatible_brands" value="isomiso2avc1mp41"></tag>&#xA;    <tag key="encoder" value="Lavf58.20.100"></tag>&#xA;</format>&#xA;

  



  



  I wrote this command line :
/ffmpeg-4.1.1/ffmpeg -i /cache/1268.mp4 -map 0:1 -f wav - | /opt/audiowaveform/audiowaveform —input-format wav —pixels-per-second 25 -b 16

  



  But it fails with the folliwing error :
Could not write header for output file #0 (incorrect codec parameters ?) : Broken pipe
Error initializing output stream 0:0 —
Conversion failed !

  



  Complete log is :

  



          ffmpeg version 4.2.2-static https://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2019 the FFmpeg developers
      built with gcc 8 (Debian 8.3.0-6)
      configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
      libavutil      56. 31.100 / 56. 31.100
      libavcodec     58. 54.100 / 58. 54.100
      libavformat    58. 29.100 / 58. 29.100
      libavdevice    58.  8.100 / 58.  8.100
      libavfilter     7. 57.100 /  7. 57.100
      libswscale      5.  5.100 /  5.  5.100
      libswresample   3.  5.100 /  3.  5.100
      libpostproc    55.  5.100 / 55.  5.100
    Error: unrecognised option '--input-format'
    See '/opt/audiowaveform/audiowaveform --help' for available options
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/cache/1268.mp4':
      Metadata:
        major_brand     : isom
        minor_version   : 512
        compatible_brands: isomiso2avc1mp41
        title           : Big Buck Bunny - test 8
        encoder         : Lavf57.72.101
        comment         : Matroska Validation File 8, secondary audio commentary track, misc subtitle tracks
      Duration: 00:00:46.07, start: 0.000000, bitrate: 2111 kb/s
        Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1024x576 [SAR 1:1 DAR 16:9], 1908 kb/s, 24 fps, 24 tbr, 12288 tbn, 48 tbc (default)
        Metadata:
          handler_name    : VideoHandler
        Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 129 kb/s (default)
        Metadata:
          handler_name    : SoundHandler
        Stream #0:2(eng): Audio: aac (LC) (mp4a / 0x6134706D), 22050 Hz, mono, fltp, 67 kb/s
        Metadata:
          handler_name    : SoundHandler
    Stream mapping:
      Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
    Press [q] to stop, [?] for help
    Could not write header for output file #0 (incorrect codec parameters ?): Broken pipe
    Error initializing output stream 0:0 --
    Conversion failed!


  



  Can anyone help me ?

  



  Note : I read the answer here ffmpeg Could not write header for output file #0 but I need to preserve audio codec.

  


• FFmpeg filter complex add audio file to video at specific points

  4 juin 2020, par steve

  atrim=0:2 starts fine when I play the video it plays the mp3 now how do I add a working end time say finish at 9 seconds ?
  

  



  Summary I want to start at 2 seconds and finish at 9 seconds.
  

  



  ffmpeg -y -i "C:\Users\test\Desktop\vidz\New folder (2)\target\vaastav song .mp4" -i "C:\Users\test\Desktop\vidz\New folder (2)\target\2.mp3" -filter_complex "[0]atrim=0:2[Apre];[0]atrim=5,asetpts=PTS-STARTPTS[Apost];[Apre][1][Apost]concat=n=3:v=0:a=1" -vcodec copy -y "C:\Users\test\Desktop\vidz\New folder (2)\target\output1.mp4"


  



  log file

  



  ffmpeg version git-2020-04-13-59e3a9a Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200328
  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-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 42.102 / 56. 42.102
  libavcodec     58. 78.102 / 58. 78.102
  libavformat    58. 42.100 / 58. 42.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 77.101 /  7. 77.101
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\test\Desktop\vidz\New folder (2)\target\vaastav song .mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
    creation_time   : 2018-09-02T04:28:46.000000Z
  Duration: 00:05:08.80, start: 0.000000, bitrate: 2289 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 2160 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2018-09-02T04:28:46.000000Z
      handler_name    : ISO Media file produced by Google Inc. Created on: 09/01/2018.
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
    Metadata:
      creation_time   : 2018-09-02T04:28:46.000000Z
      handler_name    : ISO Media file produced by Google Inc. Created on: 09/01/2018.
Input #1, mp3, from 'C:\Users\test\Desktop\vidz\New folder (2)\target\2.mp3':
  Metadata:
    genre           : Electronic;Indie
    title           : A Distorted Noise With A Little Bit Of Sense
    artist          : Lenin Was A Zombie
    encoder         : Lavf56.19.100
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
  Duration: 00:00:54.00, start: 0.025057, bitrate: 192 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
    Metadata:
      encoder         : Lavc56.21
Stream mapping:
  Stream #0:1 (aac) -> atrim
  Stream #0:1 (aac) -> atrim
  Stream #1:0 (mp3float) -> concat:in1:a0
  concat -> Stream #0:0 (aac)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
Output #0, mp4, to 'C:\Users\test\Desktop\vidz\New folder (2)\target\output1.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isommp42
    encoder         : Lavf58.42.100
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc58.78.102 aac
    Stream #0:1(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 2160 kb/s, 25 fps, 25 tbr, 90k tbn, 90k tbc (default)
    Metadata:
      creation_time   : 2018-09-02T04:28:46.000000Z
      handler_name    : ISO Media file produced by Google Inc. Created on: 09/01/2018.
frame=  737 fps=0.0 q=-1.0 size=    7936kB time=00:00:29.44 bitrate=2208.1kbits/s speed=58.9x    
[out_0_0 @ 000000000312c900] 100 buffers queued in out_0_0, something may be wrong.
[out_0_0 @ 000000000312c900] 1000 buffers queued in out_0_0, something may be wrong.
frame= 1401 fps=876 q=-1.0 size=   16640kB time=00:01:46.85 bitrate=1275.7kbits/s speed=66.8x    
frame= 2281 fps=1086 q=-1.0 size=   26112kB time=00:02:22.01 bitrate=1506.3kbits/s speed=67.6x    
frame= 3152 fps=1212 q=-1.0 size=   35584kB time=00:02:56.58 bitrate=1650.8kbits/s speed=67.9x    
frame= 4011 fps=1294 q=-1.0 size=   45824kB time=00:03:31.32 bitrate=1776.4kbits/s speed=68.2x    
frame= 4882 fps=1356 q=-1.0 size=   56320kB time=00:04:06.03 bitrate=1875.2kbits/s speed=68.3x    
frame= 5753 fps=1403 q=-1.0 size=   66560kB time=00:04:40.79 bitrate=1941.8kbits/s speed=68.5x    
frame= 6622 fps=1440 q=-1.0 size=   77056kB time=00:05:15.37 bitrate=2001.6kbits/s speed=68.6x    
frame= 7482 fps=1467 q=-1.0 size=   85248kB time=00:05:50.20 bitrate=1994.1kbits/s speed=68.7x    
frame= 7720 fps=1479 q=-1.0 Lsize=   87248kB time=00:05:59.79 bitrate=1986.5kbits/s speed=68.9x    
video:81451kB audio:5634kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.186449%
[aac @ 0000000003017ac0] Qavg: 648.065


  



  I tried messing about with the line of code but no success so I turn to you for additional support

  


• Algorithm when recording SegmentTimeline

  24 mars 2021, par jgkim0518

  I packaged a media stream by mpeg-dash transcoded video at twice the speed and audio at normal speed.
Here is the command :

  


  ffmpeg -re -stream_loop -1 -i timing_logic.mp4 -c:v hevc_nvenc -filter:v "setpts=2*PTS" -c:a libfdk_aac -map 0:v -map 0:a -f mpegts udp://xxx.xxx.xxx.xxx:xxxx?pkt_size=1316


  


  The source information used here is as follows :

  


  Input #0, mpegts, from '/home/test/timing_logic/timing_logic.mp4':
Duration: 00:00:30.22, start: 1.400000, bitrate: 16171 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: hevc (Main 10) (HEVC / 0x43564548), yuv420p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc
Stream #0:1[0x101](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 128 kb/s Stream mapping:
Stream #0:0 -> #0:0 (hevc (native) -> hevc (hevc_nvenc))
Stream #0:1 -> #0:1 (mp2 (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
Output #0, mpegts, to 'udp://xxx.xxx.xxx.xxx:xxxx?pkt_size=1316':
Metadata:
encoder : Lavf58.45.100
Stream #0:0: Video: hevc (hevc_nvenc) (Main 10), p010le, 3840x2160 [SAR 1:1 DAR 16:9], q=-1--1, 2000 kb/s, 50 fps, 90k tbn, 50 tbc
Metadata:
encoder : Lavc58.91.100 hevc_nvenc
Side data:
cpb: bitrate max/min/avg: 0/0/2000000 buffer size: 4000000 vbv_delay: N/A
Stream #0:1(eng): Audio: aac (libfdk_aac), 48000 Hz, stereo, s16, 139 kb/s
Metadata:
encoder : Lavc58.91.100 libfdk_aac


  


  My shaka packager command is :

  


  packager \ 'in=udp://xxx.xxx.xxx.xxx:xxxx,stream=video,init_segment=/home/test/timing_logic/package/video/0/video.mp4,segment_template=/home/test/timing_logic/package/video/0/$Time$.m4s' \
'in=udp://xxx.xxx.xxx.xxx:xxxx,stream=audio,init_segment=/home/test/timing_logic/package/audio/0/audio.mp4,segment_template=/home/test/timing_logic/package/audio/0/$Time$.m4a' \
--segment_duration 5 --fragment_duration 5 --minimum_update_period 5 --min_buffer_time 5 \
--preserved_segments_outside_live_window 24 --time_shift_buffer_depth 40 \
--allow_codec_switching --allow_approximate_segment_timeline --log_file_generation_deletion \
--mpd_output $OUTPUT/$output_mpd.mpd &


  


  Looking at the mpd generated as a result of packaging, the duration of the audio is twice that of the video, and the number of segments is half.
The following is the content of mpd :

  


  &lt;?xml version="1.0" encoding="UTF-8"?>&#xA;&#xA;<mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" minbuffertime="PT5S" type="dynamic" publishtime="2021-03-23T10:01:57Z" availabilitystarttime="2021-03-23T09:59:55Z" minimumupdateperiod="PT5S" timeshiftbufferdepth="PT40S">&#xA;<period start="PT0S">&#xA;<adaptationset contenttype="audio" segmentalignment="true">&#xA;<representation bandwidth="140020" codecs="mp4a.40.2" mimetype="audio/mp4" audiosamplingrate="48000">&#xA;<audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>&#xA;<segmenttemplate timescale="90000" initialization="/home/test/timing_logic/package/audio/0/audio.mp4" media="/home/test/timing_logic/package/audio/0/$Time$.m4a" startnumber="16">&#xA;<segmenttimeline>&#xA;<s t="6751436" d="449280"></s>&#xA;<s t="7200716" d="451200"></s>&#xA;<s t="7651916" d="449280"></s>&#xA;<s t="8101196" d="374400"></s>&#xA;<s t="8551196" d="449280"></s>&#xA;<s t="9000476" d="451200"></s>&#xA;<s t="9451674" d="449280"></s>&#xA;<s t="9900956" d="449280"></s>&#xA;<s t="10350236" d="451200"></s>&#xA;<s t="10801434" d="374400"></s>&#xA;</segmenttimeline>&#xA;</segmenttemplate>&#xA;</representation>&#xA;</adaptationset>&#xA;<adaptationset contenttype="video" width="3840" height="2160" framerate="90000/3600" segmentalignment="true" par="16:9">&#xA;<representation bandwidth="1520392" codecs="hvc1.2.4.L153" mimetype="video/mp4" sar="1:1">&#xA;<segmenttemplate timescale="90000" initialization="/home/test/timing_logic/package/video/0/video.mp4" media="/home/test/timing_logic/package/video/0/$Time$.m4s" startnumber="19">&#xA;<segmenttimeline>&#xA;<s t="16954440" d="900000" r="4"></s>&#xA;</segmenttimeline>&#xA;</segmenttemplate>&#xA;</representation>&#xA;</adaptationset>&#xA;</period>&#xA;</mpd>&#xA;

  


  When looking at the MPD, it seems that the shaka packager checks the PTS or DTS of the TS when creating a segment and recording the contents of the SegmentTimeline.
But I couldn't understand even by looking at the MPEG standard documentation and the DASH-IF documentation.

  


  My question is whether the packager refers to PTS or DTS when creating a segment.
How are SegmentTimeline's S@t and S@d recorded ?
What algorithm is the SegmentTimeline recorded with ? Please help me. Thank you.