Recherche avancée

Sur d’autres sites (11486)

• ffmpeg convert hevc to h264 wrong duration

  17 septembre 2021, par Vinh Duy

  I converted an HEVC file to H264 by ffmpeg, but result in wrong duration.
HEVC file duration is 3m32 but H264 file is 9h long. If I skip audio with '-an' parameter, output file duration is matched.
VLC can play but embedded into webpage causes forever loading.

  


  Could anyone help me with this error ?

  


  Check HEVC with ffprobe :

  


  ffprobe -i hevc.mp4
ffprobe version 4.2.4-1ubuntu0.1 Copyright (c) 2007-2020 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
  configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  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
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'hevc.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    title           : Media Presentation
    encoder         : Lavf58.29.100
  Duration: 00:03:32.16, start: 0.000000, bitrate: 1653 kb/s
    Stream #0:0(und): Video: hevc (Main) (hev1 / 0x31766568), yuvj420p(pc, bt709), 1920x1080, 1616 kb/s, 24.99 fps, 25 tbr, 90k tbn, 90k tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 31 kb/s (default)
    Metadata:
      handler_name    : SoundHandler


  


  check H264 file :

  


  ffprobe -i h264.mp4
ffprobe version 4.2.4-1ubuntu0.1 Copyright (c) 2007-2020 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
  configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  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
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'h264.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    title           : Media Presentation
    encoder         : Lavf58.29.100
  Duration: 09:03:59.97, start: 0.000000, bitrate: 12 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 1975 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 16000 Hz, mono, fltp (default)
    Metadata:
      handler_name    : SoundHandler


  


  convert command :

  


  ffmpeg -i hevc.mp4 -c:a copy -c:v libx264 h264.mp4


  


• FFMPEG Video to Audio Conversion Results in Different Durations

  10 juin 2020, par Eric J

  I am trying to covert an MP4 file into a mono WAV file sampled at 16,000 Hz.

  



  When I run below code, the duration goes from 00:09:59.99 (MP4) to 00:09:57.64 (WAV). Its original, longer version goes from 00:48:37.46 (MP4) to 00:48:23.38 (WAV).

  



  ffmpeg -i .mp4 -ac 1 -ar 16000 .wav


  



  I've also tried below code. The result is much worse, going from 00:09:59.99 (MP4) to 00:12:56.29 (AAC).

  



  ffmpeg -I .mp4 -vn -acodec copy .aac


  



  Attaching the log :

  



  Report written to "ffmpeg-20200610-093115.log"
Command line:
ffmpeg -i short.mp4 -ac 1 -ar 16000 short.wav -report
ffmpeg version 4.1.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/openjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/openjdk-11.0.2.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
Splitting the commandline.
Reading option '-i' ... matched as input url with argument 'short.mp4'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
Reading option 'short.wav' ... matched as output url.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url short.mp4.
Successfully parsed a group of options.
Opening an input file: short.mp4.
[NULL @ 0x7f98a3008200] Opening 'short.mp4' for reading
[file @ 0x7f98a2904440] Setting default whitelist 'file,crypto'
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] ISO: File Type Major Brand: mp42
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Processing st: 0, edit list 0 - media time: 0, duration: 7679872
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Processing st: 1, edit list 0 - media time: 1024, duration: 26459559
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] drop a frame at curr_cts: 0 @ 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Before avformat_find_stream_info() pos: 11213917 bytes read:318782 seeks:1 nb_streams:2
[h264 @ 0x7f98a3808800] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x7f98a3808800] nal_unit_type: 8(PPS), nal_ref_idc: 3
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] demuxer injecting skip 1024 / discard 0
[aac @ 0x7f98a1008c00] skip 1024 / discard 0 samples due to side data
[h264 @ 0x7f98a3808800] nal_unit_type: 6(SEI), nal_ref_idc: 0
[h264 @ 0x7f98a3808800] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 0x7f98a3808800] Format yuv420p chosen by get_format().
[h264 @ 0x7f98a3808800] Reinit context to 640x368, pix_fmt: yuv420p
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] All info found
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] After avformat_find_stream_info() pos: 21961 bytes read:351550 seeks:2 frames:46
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'short.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 1
    compatible_brands: isommp41mp42
    creation_time   : 2020-06-10T16:12:17.000000Z
  Duration: 00:09:59.99, start: 0.000000, bitrate: 149 kb/s
    Stream #0:0(eng), 1, 1/12800: Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 47 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2020-06-10T16:12:17.000000Z
      handler_name    : Core Media Video
    Stream #0:1(eng), 45, 1/44100: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 98 kb/s (default)
    Metadata:
      creation_time   : 2020-06-10T16:12:17.000000Z
      handler_name    : Core Media Audio
Successfully opened the file.
Parsing a group of options: output url short.wav.
Applying option ac (set number of audio channels) with argument 1.
Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
Successfully parsed a group of options.
Opening an output file: short.wav.
[file @ 0x7f98a0c1db40] Setting default whitelist 'file,crypto'
Successfully opened the file.
Stream mapping:
  Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[aac @ 0x7f98a100de00] skip 1024 / discard 0 samples due to side data
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
detected 12 logical cores
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'time_base' to value '1/44100'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'sample_rate' to value '44100'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'channel_layout' to value '0x4'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] tb:1/44100 samplefmt:fltp samplerate:44100 chlayout:0x4
[format_out_0_0 @ 0x7f98a0e2cb80] Setting 'sample_fmts' to value 's16'
[format_out_0_0 @ 0x7f98a0e2cb80] Setting 'sample_rates' to value '16000'
[format_out_0_0 @ 0x7f98a0e2cb80] Setting 'channel_layouts' to value '0x4'
[format_out_0_0 @ 0x7f98a0e2cb80] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 0x7f98a0c16ac0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 0x7f98a0e2d540] [SWR @ 0x7f98a28e1000] Using fltp internally between filters
[auto_resampler_0 @ 0x7f98a0e2d540] ch:1 chl:mono fmt:fltp r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
Output #0, wav, to 'short.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 1
    compatible_brands: isommp41mp42
    ISFT            : Lavf58.20.100
    Stream #0:0(eng), 0, 1/16000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s (default)
    Metadata:
      creation_time   : 2020-06-10T16:12:17.000000Z
      handler_name    : Core Media Audio
      encoder         : Lavc58.35.100 pcm_s16le
size=   17152kB time=00:09:16.63 bitrate= 252.4kbits/s speed=1.11e+03x    
[out_0_0 @ 0x7f98a0e2c700] EOF on sink link out_0_0:default.
No more output streams to write to, finishing.
size=   18676kB time=00:09:59.99 bitrate= 255.0kbits/s speed=1.11e+03x    
video:0kB audio:18676kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000408%
Input file #0 (short.mp4):
  Input stream #0:0 (video): 1 packets read (3689 bytes); 
  Input stream #0:1 (audio): 25739 packets read (7375414 bytes); 25738 frames decoded (26355712 samples); 
  Total: 25740 packets (7379103 bytes) demuxed
Output file #0 (short.wav):
  Output stream #0:0 (audio): 25739 frames encoded (9562163 samples); 25739 packets muxed (19124326 bytes); 
  Total: 25739 packets (19124326 bytes) muxed
25738 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x7f98a0c1dc40] Statistics: 4 seeks, 76 writeouts
[AVIOContext @ 0x7f98a29045c0] Statistics: 10902846 bytes read, 29 seeks


  


• ffmpeg : How to limit/decrease the size of PCM files ?

  7 août 2021, par a_hayler

  I am playing around with the Shazam API and trying to let it identify a song from a chosen sound-bite. For this I need as the body of the request the following :

  


  "Encoded base64 string of byte[] that generated from raw data less than 500KB (3-5 seconds sample are good enough for detection). The raw sound data must be 44100Hz, 1 channel (Mono), signed 16 bit PCM little endian."

  


  I extracted a ten-second interval in the file slice.mp3 and (hopefully) converted it to the right format by using :

  


  ffmpeg -i song_mono.mp3 -f s16le -ac 1 -ar 44100 -b:a 128k result.raw

  


  The problem now is that the resulting file is about 21MB, just 20.5MB over the API's limit. I am sure that there has to be a way to decrease the size of the audio file to under 500KB. The first thing that I have noticed is that the bitrate of the output file is again at 700+ kb/s even though I changed it in the slicing process to 128kb/s. Additionally adding -b:a 128k doesn't seem to do anything.

  


  Thus I am asking myself (and you now) : How do I bring the size of the file under control (in this case 500KB) whilst still maintaining the specified requirements.

  


  Any help is greatly appreciated !

  


  Here is the output of the following commands :

  


  ffmpeg -i slice.mp3
ffmpeg -i song_mono.mp3 -f s16le -ac 1 -ar 44100 -b:a 128k result.raw


  


  ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Input #0, mp3, from 'slice.mp3':
  Metadata:
    encoder         : Lavf57.83.100
  Duration: 00:00:10.06, start: 0.023021, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, mono, s16p, 128 kb/s
At least one output file must be specified
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Input #0, mp3, from 'song_mono.mp3':
  Metadata:
    encoder         : Lavf57.83.100
  Duration: 00:04:04.87, start: 0.023021, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, mono, s16p, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, s16le, to 'result.raw':
  Metadata:
    encoder         : Lavf57.83.100
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
    Metadata:
      encoder         : Lavc57.107.100 pcm_s16le
size=   21089kB time=00:04:04.83 bitrate= 705.6kbits/s speed= 460x    
video:0kB audio:21089kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%