I'm using ffmpeg to convert audio from FLAC to AAC and I'd like to use the aac_mf encoder, which uses Microsoft Media Foundation. If I use the native AAC encoder built into ffmpeg it works with the following command:
ffmpeg -i input.flac -acodec aac -vn output.m4a
However, the following command gives an error:
> ffmpeg -i input.flac -acodec aac_mf -vn output.m4a
ffmpeg version 5.0.1-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 11.2.0 (Rev7, 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-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --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. 17.100 / 57. 17.100
libavcodec 59. 18.100 / 59. 18.100
libavformat 59. 16.100 / 59. 16.100
libavdevice 59. 4.100 / 59. 4.100
libavfilter 8. 24.100 / 8. 24.100
libswscale 6. 4.100 / 6. 4.100
libswresample 4. 3.100 / 4. 3.100
libpostproc 56. 3.100 / 56. 3.100
Input #0, flac, from '.\input.flac':
Duration: 00:04:09.17, start: 0.000000, bitrate: 833 kb/s
Stream #0:0: Audio: flac, 44100 Hz, stereo, s16
Stream mapping:
Stream #0:0 -> #0:0 (flac (native) -> aac (aac_mf))
Press [q] to stop, [?] for help
[aac_mf @ 000001bad81c6400] MFT name: 'Microsoft AAC Audio Encoder MFT'
[ipod @ 000001bad81fb640] track 0: codec frame size is not set
Output #0, ipod, to 'output.m4a':
Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 128 kb/s
Metadata:
encoder : Lavc59.18.100 aac_mf
[aac_mf @ 000001bad81c6400] nb_samples (4096) != frame_size (0)
Audio encoding failed
Conversion failed!
This is a bug. For now, add -frame_size. Will be fixed in 5.1 and soon in git master.
ffmpeg -i input.flac -acodec aac -vn -frame_size 1024 output.m4a