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• FFmpeg stdin "output file is empty, nothing was encoded"

  2 février 2023, par brock

  Just trying to stdin and stdout a simple CAF to MP3 conversion. Output looks exactly the same except using stdin does not encode anything. Windows 10. I'm going bananas here. Please advise.

  


  Using - (stdin)...

  


  >type test.caf | ffmpeg -i - -f mp3 - > test.mp3
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.1.1 (GCC) 20190807
  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-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  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
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, caf, from 'pipe:':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: adpcm_ima_qt (ima4 / 0x34616D69), 48000 Hz, stereo, s16p, 384 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_ima_qt (native) -> mp3 (libmp3lame))
Output #0, mp3, to 'pipe:':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
    TSSE            : Lavf58.29.100
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc58.54.100 libmp3lame
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)


  


  Using -i...

  


  >ffmpeg -i test.caf -f mp3 - > test.mp3
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.1.1 (GCC) 20190807
  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-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  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
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, caf, from 'test.caf':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: adpcm_ima_qt (ima4 / 0x34616D69), 48000 Hz, stereo, s16p, 384 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_ima_qt (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'pipe:':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
    TSSE            : Lavf58.29.100
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc58.54.100 libmp3lame
size=      49kB time=00:00:03.12 bitrate= 129.3kbits/s speed=34.4x
video:0kB audio:49kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.246501%


  


  EDIT : Command syntax is ok. Works as expected with a WAV file. I upgraded FFmpeg now I get errors using stdin. So it is the file that is to blame. However, I do find it odd that using -i is fine, but stdin is not.

  


  >ffmpeg -i - -f mp3 - > test.mp3 < test.caf
ffmpeg version 5.1.2-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12.1.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
[caf @ 0000023dc65b1140] skipping CAF chunk: 00000000 ([0][0][0][0]), size 0
    Last message repeated 324 times
[caf @ 0000023dc65b1140] skipping CAF chunk: 00000000 ([0][0][0][0]), size 431131746560
[caf @ 0000023dc65b1140] skipping CAF chunk: 00000000 ([0][0][0][0]), size 173911953188585728
[caf @ 0000023dc65b1140] skipping CAF chunk: 00FFFF02 ([0][255][255][2]), size -7250317618881344622
pipe:: Invalid data found when processing input


  


• Problem concatenating an mp4 file with an mp4 created by repeating a single image, with the same codec with ffmpeg concat demuxer

  3 février 2023, par ashay

  I have an mp4 video (of a lecture) containing two streams, video and audio. I wanted an introduction for it (title slide of the presentation that accompanied it), so I made an mp4 out of the intro image via ffmpeg -framerate 30 -i lec01_title.jpg -t 3 -c:v libx264 -pix_fmt yuvj420p -vf "scale=1920:1080" lec01_title.mp4 -f lavfi -i anullsrc -c:a aac. When I try to concat the files using the demuxer, it doesn't work. First, I try to verify that the properties (encoding, etc.) of the two videos are the same.

  


  If I run ffmpeg -i lec01.mp4, I get :

  


  ffmpeg version 5.1.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/5.1.1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'dcai_lec01.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    title           : Wide
    encoder         : Lavf58.20.100
  Duration: 00:45:24.01, start: 0.000000, bitrate: 743 kb/s
  Stream #0:0[0x1](und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, stereo, fltp, 129 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]
  Stream #0:1[0x2](und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 606 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
    Metadata:
      handler_name    : VideoHandler
      vendor_id       : [0][0][0][0]


  


  If I run ffmpeg -i lec01_title.mp4, I get :

  


  ffmpeg version 5.1.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/5.1.1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'dcai_lec01_title.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf59.27.100
  Duration: 00:00:03.00, start: 0.000000, bitrate: 150 kb/s
  Stream #0:0[0x1](und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, bt470bg/unknown/unknown, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 12877 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
    Metadata:
      handler_name    : VideoHandler
      vendor_id       : [0][0][0][0]
      encoder         : Lavc59.37.100 libx264
  Stream #0:1[0x2](und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 2 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]


  


  I tried to verify that the properties of the two videos match via ffprobe -select_streams a:0 -show_entries stream=codec_name,channels -of default=nw=1:nk=1 -v 0 lec01.mp4 and ffprobe -select_streams v:0 -show_entries stream=codec_name,width,height,r_frame_rate,pix_fmt -of default=nw=1:nk=1 -v 0 lec01.mp4, and they do. The first command gives me

  


  aac
2


  


  and the second command gives me

  


  h264
1920
1080
yuvj420p
30/1


  


  for both videos.

  


  Now, if I have a file called lec01.txt containing :

  


  file 'lec01_title.mp4'
file 'lec01.mp4'


  


  when I run ffmpeg -f concat -i lec01.txt -c copy output.mp4, the resulting video is of length 04:44:04 (four hours, when my two input videos were 3 seconds and 45:24 minutes respectively), and it only shows the title slide for that entire duration.

  


  Furthermore, when I run this concat command, I get the following message repeated many times : [mp4 @ 0x7f82b9013b80] Non-monotonous DTS in output stream 0:1; previous: 133290, current: 133120; changing to 133291. This may result in incorrect timestamps in the output file.

  


  I'm missing something. When I look up this error, it seems to be something related to the decoding time stamps (DTS) or presentation time stamps. Anyone know what I'm doing wrong and how to fix ? Thank you for your help !

  


  Edit :
It appears to work if I re-encode via ffmpeg -i lec01_title.mp4 -i lec01.mp4 -filter_complex "[0:v:0][0:a:0][1:v:0][1:a:0]concat=n=2:v=1:a=1[outv][outa]" -map "[outv]" -map "[outa]" output.mp4, but I'd like to avoid doing this since I have tons of videos I need to do this for.

  


• Latency and DAF in RTP transmissions

  24 février 2023, par jfernandz

  I'm trying to perform some tests for audio RTP transmissions to know their technical limitations. The idea is to prevent DAF effect in this kind of transmissions, I'm assuming a latency lower than 50ms will prevent it. But there is another handicap in my analysis, the RTP transmission must be over WiFi.

  


  For this tests I'm trying to transmit raw audio (not sure if skipping the encoding stage will improve latency) through ffmpeg between two different laptops, so I'm running ffmpeg in the first laptop (172.20.1.2) as :

  


  $ ffmpeg -f pulse -i 56 -c copy -f rtp rtp://172.20.1.5:10000

  


  which produces the following output :

  


  ffmpeg version n5.1.2 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-opencl --enable-opengl --enable-shared --enable-version3 --enable-vulkan
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from '56':
  Duration: N/A, start: 1677234050.938677, bitrate: 1536 kb/s
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Output #0, rtp, to 'rtp://172.20.1.5:10000':
  Metadata:
    encoder         : Lavf59.27.100
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:1536

Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=     322kB time=00:00:01.67 bitrate=1573.6kbits/s speed=1.06x


  


  I'm assuming the shown SDP is a valid one :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:1536


  


  So I saved it in a file called ccopy.sdp on the second laptop (172.20.1.5). However, when I run ffplay in this other laptop as :

  


  $ ffplay -protocol_whitelist file,rtp,udp -i ccopy.sdp

  


  I can see there is problems with this SDP :

  


  ffplay version n5.1.2 Copyright (c) 2003-2022 the FFmpeg developers
  built with gcc 12.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-opencl --enable-opengl --enable-shared --enable-version3 --enable-vulkan
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
[sdp @ 0x7f8eec000c80] Could not find codec parameters for stream 0 (Audio: none, 0 channels): unknown codec
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, sdp, from 'ccopy.sdp':
  Metadata:
    title           : No Name
  Duration: N/A, bitrate: N/A
  Stream #0:0: Audio: none, 0 channels
Failed to open file 'ccopy.sdp' or configure filtergraph
    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   


  


  Not sure if I'm doing something wrong or this is because of I cannot actually use pcm_s16le for an RTP transmission. Moreover ... Is there some argument for ffmpeg that I can use to improve this RTP transmission and reduce latency under 50ms.

  


  Thank you all :-)

  


  PS : When I don't use -c copy argument for ffmpeg and therefore I have this SDP

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:768
a=rtpmap:97 PCMU/48000/2


  


  The RTP transmission works as I expect, but with a significant DAF.