The command I used was :
C:\cmd\ffmpeg\bin\ffmpeg.exe -i "C:\Users\user\Dropbox\Workspace - C#\SnowflakeGenerator\out\out%d.png" -c:v libvpx -b:v 1M "C:\Users\user\Dropbox\Workspace - C#\SnowflakeGenerator\video.webm" -y
The output was:
ffmpeg version N-82889-g54931fd Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 43.100 / 55. 43.100
libavcodec 57. 68.100 / 57. 68.100
libavformat 57. 61.100 / 57. 61.100
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 68.100 / 6. 68.100
libswscale 4. 3.101 / 4. 3.101
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Input #0, image2, from 'C:\Users\yb5243yv\Dropbox\Workspace - C#\SnowflakeGenerator\out\out%d.png':
Duration: 00:00:21.04, start: 0.000000, bitrate: N/A
Stream #0:0: Video: png, rgba(pc), 1000x1000 [SAR 3779:3779 DAR 1:1], 25 fps, 25 tbr, 25 tbn, 25 tbc
[libvpx # 0000000000be89c0] v1.6.0
[libvpx # 0000000000be89c0] Transparency encoding with auto_alt_ref does not work
Stream mapping:
Stream #0:0 -> #0:0 (png (native) -> vp8 (libvpx))
Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
The error is:
[libvpx # 0000000000be89c0] Transparency encoding with auto_alt_ref does not work
and
Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
I'm not sure exactly what is causing the issue, I followed this tutorial, however they did not mention the error. I should mention that I am not using audio, so I left the audio encoder out (I did try it with the audio encoder to make sure, and that was not the problem)
Like the error msg says, Transparency encoding with auto_alt_ref does not work, so
ffmpeg.exe -i "out%d.png" -c:v libvpx -b:v 1M -auto-alt-ref 0 "video.webm" -y
Related
I am new to ffmpeg and trying to figure out how to convert .MOD files to any generally readable format.
So far I was trying to
ffmpeg -i MOV00A.MOD -vcodec libx264 -acodec aac out.mp4
Result video is readable and has correct ratio aspect is correct but with artefacts like those horizontal lines:
My guess is that I might be using wrong codec but I do not understand which one to choose ?
Running ffmprobe on .MOD results:
ffprobe version 5.1.2 Copyright (c) 2007-2022 the FFmpeg developers
built with Apple clang version 14.0.0 (clang-1400.0.29.202)
configuration: --prefix=/usr/local/Cellar/ffmpeg/5.1.2_4 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libsvtav1 --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, mpeg, from 'MOV00A.MOD':
Duration: 00:00:41.28, start: 0.262222, bitrate: 5285 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, bt470bg, top first), 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn
Side data:
cpb: bitrate max/min/avg: 9286000/0/0 buffer size: 1835008 vbv_delay: N/A
Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 256 kb/s
I am recording screen capture of my Electron app to a file, as suggested here: Save captured video to file in Electron
It works great, but the file is a “transport stream”. I can play it in Chrome browser, but can’t adjust the time slider.
The suggestion was to use ffmpeg to post-process the file. The simplest, straight-forward command I found is:
ffmpeg -fflags +genpts -i in.webm out.webm
I thought that this simply generates timestamps. However, the out.webm file is 15 times smaller than in.webm! I don’t see any change in the quality. The downside is - the processing takes about the same time as the duration of the video.
Two questions:
Is there any way I can speedup this “timestamping” process?
Can I record a video that is already 15 times smaller in size?
I specify maxFrameRate: 30 when calling webkitGetUserMedia().
Here is the output of that ffmpeg command:
vlad$ ffmpeg -fflags +genpts -i in.webm out.webm
ffmpeg version 4.3.2 Copyright (c) 2000-2021 the FFmpeg developers
built with Apple clang version 12.0.0 (clang-1200.0.32.29)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --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-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --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-librtmp --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
libavutil 56. 51.100 / 56. 51.100
libavcodec 58. 91.100 / 58. 91.100
libavformat 58. 45.100 / 58. 45.100
libavdevice 58. 10.100 / 58. 10.100
libavfilter 7. 85.100 / 7. 85.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 7.100 / 5. 7.100
libswresample 3. 7.100 / 3. 7.100
libpostproc 55. 7.100 / 55. 7.100
Input #0, matroska,webm, from 'in.webm':
Metadata:
encoder : Chrome
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv), 2560x1416, SAR 1:1 DAR 320:177, 1k tbr, 1k tbn, 1k tbc (default)
Metadata:
alpha_mode : 1
Stream mapping:
Stream #0:0 -> #0:0 (vp9 (native) -> vp9 (libvpx-vp9))
Press [q] to stop, [?] for help
[libvpx-vp9 # 0x7f85f2012600] v1.9.0
[libvpx-vp9 # 0x7f85f2012600] Neither bitrate nor constrained quality specified, using default CRF of 32
Output #0, webm, to 'out.webm':
Metadata:
encoder : Lavf58.45.100
Stream #0:0(eng): Video: vp9 (libvpx-vp9), yuv420p, 2560x1416 [SAR 1:1 DAR 320:177], q=-1--1, 1k fps, 1k tbn, 1k tbc (default)
Metadata:
alpha_mode : 1
encoder : Lavc58.91.100 libvpx-vp9
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
frame= 209 fps=5.8 q=0.0 Lsize= 881kB time=00:00:17.81 bitrate= 405.0kbits/s speed=0.494x
video:879kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.231567%
And here is the information on both files from ffprobe:
vlad$ ffprobe in.webm
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
built with Apple clang version 12.0.0 (clang-1200.0.32.29)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --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-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --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-librtmp --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
libavutil 56. 51.100 / 56. 51.100
libavcodec 58. 91.100 / 58. 91.100
libavformat 58. 45.100 / 58. 45.100
libavdevice 58. 10.100 / 58. 10.100
libavfilter 7. 85.100 / 7. 85.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 7.100 / 5. 7.100
libswresample 3. 7.100 / 3. 7.100
libpostproc 55. 7.100 / 55. 7.100
Input #0, matroska,webm, from 'in.webm':
Metadata:
encoder : Chrome
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv), 2560x1416, SAR 1:1 DAR 320:177, 1k tbr, 1k tbn, 1k tbc (default)
Metadata:
alpha_mode : 1
and
vlad$ ffprobe out.webm
ffprobe version 4.3.2 Copyright (c) 2007-2021 the FFmpeg developers
built with Apple clang version 12.0.0 (clang-1200.0.32.29)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --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-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --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-librtmp --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox
libavutil 56. 51.100 / 56. 51.100
libavcodec 58. 91.100 / 58. 91.100
libavformat 58. 45.100 / 58. 45.100
libavdevice 58. 10.100 / 58. 10.100
libavfilter 7. 85.100 / 7. 85.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 7.100 / 5. 7.100
libswresample 3. 7.100 / 3. 7.100
libpostproc 55. 7.100 / 55. 7.100
Input #0, matroska,webm, from 'out.webm':
Metadata:
ENCODER : Lavf58.45.100
Duration: 00:00:17.82, start: 0.000000, bitrate: 405 kb/s
Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv), 2560x1416, SAR 1:1 DAR 320:177, 1k tbr, 1k tbn, 1k tbc (default)
Metadata:
ALPHA_MODE : 1
ENCODER : Lavc58.91.100 libvpx-vp9
DURATION : 00:00:17.815000000
I am fairly sure your FFmpeg processing isn't exactly what you think it is -- with your command line, particularly with absence of certain explicit switches, FFmpeg will transcode your video and audio at its own discretion, which is why your output file is much smaller -- the data have been re-compressed, with potentially loss of quality.
If you just want to generate the index necessary for the kind of seeking most players do, and avoid the undesired transcoding, the following command line will suffice:
ffmpeg -i <input-file> -codec copy <output-file>
On a minor note, do take into account that without an additional explicit switch to instruct it otherwise, FFmpeg infers the output container format from the output file name extension. I assume both your input and output are WebM, so there isn't any issue in your case. But even if they weren't, re-muxing media into another format of container doesn't have to result in quality loss.
As you can surmise from the command line above, just running FFmpeg without any explicit switch pertaining to "generation of frame time index", causes it to build the index anyway. It will do so with or without transcoding taking place.
P.S. There are some players, like [the now abandoned] MPC-HC, which are able to seek at least some transport streams. I expect it to be able to seek your kind of WebM media. I don't know the exact details on how it actually does the seeking, but if I would guess I'd say it simply does a linear search from some appropriate location in the media (or the very beginning) for a frame matching the time of seek. Much like looking for a row in a database table which doesn't have an index to help with the search predicate.
I am extracting a clip from an mpg video using the following command.
ffmpeg.exe -i 271b.mpg -ss 00:00:05.000 -t 00:00:03.200 -c copy out_271b.mpg
I am using the copy parameter to avoid reencoding the clip.
The resulting mpg file only has the audio track, no video.
The following is the FFMpeg screen dump.
C:\ffmpeg-20170615\bin>ffmpeg.exe -i 271b.mpg -ss 00:00:05.000 -t 00:00:03.200 -c copy out_271b.mpg
ffmpeg version N-86482-gbc40674 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 7.1.0 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
libavutil 55. 66.100 / 55. 66.100
libavcodec 57. 99.100 / 57. 99.100
libavformat 57. 73.100 / 57. 73.100
libavdevice 57. 7.100 / 57. 7.100
libavfilter 6. 92.100 / 6. 92.100
libswscale 4. 7.101 / 4. 7.101
libswresample 2. 8.100 / 2. 8.100
libpostproc 54. 6.100 / 54. 6.100
Input #0, mpeg, from '271b.mpg':
Duration: 00:01:11.44, start: 0.460000, bitrate: 2154 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, bt470bg, progressive), 640x480 [SAR 1:1 DAR 4:3], 2000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0:1[0x1c0]: Audio: mp2, 44100 Hz, mono, s16p, 128 kb/s
[mpeg # 0000000000666aa0] VBV buffer size not set, using default size of 130KB
If you want the mpeg file to be compliant to some specification
Like DVD, VCD or others, make sure you set the correct buffer size
Output #0, mpeg, to 'out_271b.mpg':
Metadata:
encoder : Lavf57.73.100
Stream #0:0: Video: mpeg2video (Main), yuv420p(tv, bt470bg, progressive), 640x480 [SAR 1:1 DAR 4:3], q=2-31, 2000 kb/s, 25 fps, 25 tbr, 90k tbn, 25 tbc
Stream #0:1: Audio: mp2, 44100 Hz, mono, s16p, 128 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 74 fps=0.0 q=-1.0 Lsize= 786kB time=00:00:03.17 bitrate=2027.2kbits/s speed= 198x
video:730kB audio:50kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.843343%
Thanks
Addition: I tried to open the generated clip file with ffplay and got the following results. Note that ffplay was able to open and play the original mpg file.
ffplay.exe out_271b.mpg
ffplay version N-86482-gbc40674 Copyright (c) 2003-2017 the FFmpeg developers
built with gcc 7.1.0 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
libavutil 55. 66.100 / 55. 66.100
libavcodec 57. 99.100 / 57. 99.100
libavformat 57. 73.100 / 57. 73.100
libavdevice 57. 7.100 / 57. 7.100
libavfilter 6. 92.100 / 6. 92.100
libswscale 4. 7.101 / 4. 7.101
libswresample 2. 8.100 / 2. 8.100
libpostproc 54. 6.100 / 54. 6.100
[mpeg # 00000000023cde80] probed stream 2 failedB sq= 0B f=0/0
[mpeg # 00000000023cde80] start time for stream 0 is not set in estimate_timings_from_pts
[mpeg # 00000000023cde80] Could not find codec parameters for stream 2 (Video: none, none): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpeg, from 'out_271bb.mpg':
Duration: 00:00:03.22, start: 0.500000, bitrate: 2068 kb/s
Stream #0:0[0x1bf]: Data: dvd_nav_packet 0KB sq= 0B f=0/0
Stream #0:1[0x1c0]: Audio: mp2, 44100 Hz, mono, s16p, 128 kb/s
Stream #0:2[0x1e0]: Video: none, none, 25 fps, 25 tbr, 90k tbn
SDL_OpenAudio (1 channels, 44100 Hz): DirectSoundCreate: No audio device found
No more combinations to try, audio open failed
No codec could be found with id 0
Failed to open file 'out_271b.mpg' or configure filtergraph
Here is the error I am getting, I took a 7.1 AAC file, and remixed one of the tracks, and now I am trying to export back to the original format.
ffmpeg -i - -strict experimental -c:a aac -ac 8 -b:a 240k "F:\Something.aac"
ffmpeg version 2.2.2 Copyright (c) 2000-2014 the FFmpeg developers
built on May 22 2014 19:56:44 with gcc 4.8.2 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-
libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
libavutil 52. 66.100 / 52. 66.100
libavcodec 55. 52.102 / 55. 52.102
libavformat 55. 33.100 / 55. 33.100
libavdevice 55. 10.100 / 55. 10.100
libavfilter 4. 2.100 / 4. 2.100
libswscale 2. 5.102 / 2. 5.102
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
Guessed Channel Layout for Input Stream #0.0 : 7.1
Input #0, wav, from 'pipe:':
Duration: N/A, bitrate: 5644 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 7.1, s16, 5644 kb/s
[aac # 02567320] Unsupported number of channels: 8
Output #0, adts, to 'F:\Something.aac':
Stream #0:0: Audio: aac, 44100 Hz, 7.1, fltp, 240 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le -> aac)
Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!
Your ffmpeg is too old.
You can get a new build for Windows at Zeranoe. Users of other OSs can find links to builds at the FFmpeg Download page.
I'm trying to convert a .tak file to .flac, but ffmpeg stops converting before the end (with no error message).
>ffmpeg -i CDImage.tak CDImage.flac
ffmpeg version N-69659-gc0367f7 Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.9.2 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib
modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw
b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge
r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en
able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-
libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enab
le-zlib
libavutil 54. 18.100 / 54. 18.100
libavcodec 56. 21.102 / 56. 21.102
libavformat 56. 19.100 / 56. 19.100
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 9.104 / 5. 9.104
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, tak, from 'CDImage.tak':
Duration: 00:17:35.95, start: 0.000000, bitrate: 1013 kb/s
Stream #0:0: Audio: tak, 44100 Hz, 2 channels, s16p
Output #0, flac, to 'CDImage.flac':
Metadata:
encoder : Lavf56.19.100
Stream #0:0: Audio: flac, 44100 Hz, stereo, s16, 128 kb/s
Metadata:
encoder : Lavc56.21.102 flac
Stream mapping:
Stream #0:0 -> #0:0 (tak (native) -> flac (native))
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size= 83540kB time=00:10:54.19 bitrate=1046.1kbits/s
video:0kB audio:83532kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.009689%
As you can see, the input file is 17:35.95, but ffmpeg is only converting 10:54.19 of it. Does anyone know what's going on here?
edit: I tried it with a different tak file and got the same result. Only 27 minutes of a 50 minute tak was converted to flac. I get the exact same output file times when converting to wav, so I'm guessing it must either be something with the tak files I have, or with ffmpeg's tak decoder.