Dear FFmpeg user Does anyone know anything about it? I use the 96k option, but a difference of about 1s remains. The file size is large, so I want to use it with more compression, but I cannot apply it.
2020년 9월 3일 (목) 오전 11:23, myounggun jang <jinm...@gmail.com>님이 작성: > Thank you for your interest > > The contents requested for confirmation have been retested and confirmed. > This is the result of recording a wav file on an Android device and > converting it on Windows PC. > > The length of the original file is 26:39, and the result of converting it > to the default option is 25:47, which is displayed in Windows Explorer and > the file size is 4,686KB. > If this is converted using the -b:a 96k option, it has the same length as > the original 26:39 and the file size is 18,740KB. > > I checked and played both the original file and the converted file using > ocenaudio SW, it marked as 26:39 and played. > However, the converted file by default is displayed in the time of 25:47 > in Media Player and played. > > Below is the console output I tested. > I don't find any difference. > > > ========================================================================================================= > defalut converting test > > ========================================================================================================= > > ffmpeg.exe -i 1.wav 1_default.mp3 > ffmpeg version git-2020-08-28-ccc7120 Copyright (c) 2000-2020 the FFmpeg > developers built with gcc 10.2.1 > (GCC) 20200805 > 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 --enable-libgsm > --enable-librav1e --enable-libsvtav1 --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. 58.100 / 56. 58.100 > libavcodec 58.101.100 / 58.101.100 > libavformat 58. 51.101 / 58. 51.101 > libavdevice 58. 11.101 / 58. 11.101 > libavfilter 7. 87.100 / 7. 87.100 > libswscale 5. 8.100 / 5. 8.100 > libswresample 3. 8.100 / 3. 8.100 > libpostproc 55. 8.100 / 55. 8.100 > Guessed Channel Layout for Input Stream #0.0 : mono > Input #0, wav, from '1.wav': > Duration: 00:26:39.00, bitrate: 256 kb/s > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, > s16, 256 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame)) > Press [q] to stop, [?] for help > Output #0, mp3, to '1_default.mp3': > Metadata: > TSSE : Lavf58.51.101 > Stream #0:0: Audio: mp3 (libmp3lame), 16000 Hz, mono, s16p > Metadata: > encoder : Lavc58.101.100 libmp3lame > size= 4685kB time=00:26:39.01 bitrate= 24.0kbits/s speed= 426x > video:0kB audio:4685kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 0.004690% > > > ========================================================================================================= > 96k converting test > > ========================================================================================================= > > ffmpeg.exe -i 1.wav -b:a 96k 1_96k.mp3 > ffmpeg version git-2020-08-28-ccc7120 Copyright (c) 2000-2020 the FFmpeg > developers > built with gcc 10.2.1 (GCC) 20200805 > 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 --enable-libgsm > --enable-librav1e --enable-libsvtav1 --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. 58.100 / 56. 58.100 > libavcodec 58.101.100 / 58.101.100 > libavformat 58. 51.101 / 58. 51.101 > libavdevice 58. 11.101 / 58. 11.101 > libavfilter 7. 87.100 / 7. 87.100 > libswscale 5. 8.100 / 5. 8.100 > libswresample 3. 8.100 / 3. 8.100 > libpostproc 55. 8.100 / 55. 8.100 > Guessed Channel Layout for Input Stream #0.0 : mono > Input #0, wav, from '1.wav': > Duration: 00:26:39.00, bitrate: 256 kb/s > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, > s16, 256 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame)) > Press [q] to stop, [?] for help > Output #0, mp3, to '1_96k.mp3': > Metadata: > TSSE : Lavf58.51.101 > Stream #0:0: Audio: mp3 (libmp3lame), 16000 Hz, mono, s16p, 96 kb/s > Metadata: > encoder : Lavc58.101.100 libmp3lame > size= 18740kB time=00:26:39.01 bitrate= 96.0kbits/s speed= 397x > video:0kB audio:18739kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 0.002486% > > > 2020년 9월 2일 (수) 오전 8:08, Edward Park <kumowoon1...@gmail.com>님이 작성: > >> Hi, >> >> > When converting a wav file to MP3 using the default option, an error >> occurs >> > in the length. >> > Converted using the following command >> > >> > ffmpeg.exe -i 1.wav 1.mp3 >> > >> > The duration of the original wav is 1:09:30, but the length of the >> > converted MP3 is 1:07:16. >> > The length of the file was checked through Windows Explorer and Windows >> > Media Player. >> > However, when checking with ocen audio and other software, it is >> normally >> > displayed as 1:09:30. >> >> That's more of a compromise than an error, encoder is most likely lame, >> and in ffmpeg would use vbr by default with a command like that. So to get >> accurate duration you pretty much need to decode, and it appears windows >> media player estimates instead. If there is a difference when you playback >> with a stopwatch in one hand that would be strange. >> >> > When I tested using the -b:a option, 64k and 96k are converted to the >> same >> > length, but there is a problem with 32k and 48k. >> > >> > ffmpeg.exe -i sample_2.wav -b:a 96k sample_2_96.mp3 >> > >> > In addition, when converting m4a files to MP3, a problem occurs also in >> 96k. >> > Please help me on what to fix or give options. >> >> I think setting the bitrate makes it encode at cbr and that makes it >> possible to determine the duration more accurately but not sure why >> different bitrates gives different results. >> >> Actually how confident are you about the accuracy of the input file >> duration that you are making these comparisons to? >> >> Regards, >> Ted Park >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> To unsubscribe, visit link above, or email >> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". > > _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
