Re: [FFmpeg-user] Problem about duration value of converted mp3 file
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 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 > libavformat58. 51.101 / 58. 51.101 > libavdevice58. 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 > libpostproc55. 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-lib
Re: [FFmpeg-user] Problem about duration value of converted mp3 file
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 libavformat58. 51.101 / 58. 51.101 libavdevice58. 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 libpostproc55. 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 libavformat58. 51.101 / 58. 51.101 libavdevice58. 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 libpostproc55. 8.100 / 55. 8.100 Guessed Channel Layout for Input Stream #0.0 : mono Input #0, wav, from
[FFmpeg-user] Problem about duration value of converted mp3 file
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. 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. ___ 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".
