Gfrktyl

I would like to ask why ffmpeg's libfdk_aac encoder is automatically downsampling my audio's bit depth when encoding and I want to know how to stop it.

My encoding is as such (all data is there, except for metadata & file paths):

ffmpeg -i "/Path/To/Input.flac" -c:a libfdk_aac -b:a 192k -ar 48000 -map_metadata -1 -metadata title="Title" -metadata artist="Artist" -metadata date="Date" "/Path/To/Output.m4a"

ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers

  built with Apple LLVM version 10.0.0 (clang-1000.10.44.4)

  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gpl --enable-libmp3lame --enable-libopus --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-opencl --enable-videotoolbox --enable-nonfree

  libavutil      56. 22.100 / 56. 22.100

  libavcodec     58. 35.100 / 58. 35.100

  libavformat    58. 20.100 / 58. 20.100

  libavdevice    58.  5.100 / 58.  5.100

  libavfilter     7. 40.101 /  7. 40.101

  libavresample   4.  0.  0 /  4.  0.  0

  libswscale      5.  3.100 /  5.  3.100

  libswresample   3.  3.100 /  3.  3.100

  libpostproc    55.  3.100 / 55.  3.100

Input #0, flac, from '/Path/To/Input.flac':

  Duration: 00:31:31.71, start: 0.000000, bitrate: 4721 kb/s

    Stream #0:0: Audio: flac, 192000 Hz, stereo, s32 (24 bit)

Stream mapping:

  Stream #0:0 -> #0:0 (flac (native) -> aac (libfdk_aac))

Press [q] to stop, [?] for help

Output #0, ipod, to '/Path/To/Output.m4a':

  Metadata:

    title           : Title

    artist          : Artist

    date            : Date

    encoder         : Lavf58.20.100

    Stream #0:0: Audio: aac (libfdk_aac) (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 192 kb/s

    Metadata:

      encoder         : Lavc58.35.100 libfdk_aac

[NULL @ 0x7fb747801000] sample/frame number mismatch in adjacent frames

size=   44686kB time=00:31:31.71 bitrate= 193.5kbits/s speed=22.2x    

video:0kB audio:44338kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.784360%

No problems with encoding... (I simply encode to AAC, set bitrate to 192kbps, and downsample to 48000 kHz) except that for some reason, the bit depth is downsampled to 16 bit when encoding from a 24 bit source. I know the libfdk_aac encoder supports 24 bit, but for some reason, the encoder auto-downsamples. I have attempted to force 24 bit, with -sample_fmt s32 but that returned with this error:

[libfdk_aac @ 0x7fc58100b200] Specified sample format s32 is invalid or not supported

Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

Conversion failed!

If anyone knows, I'd greatly appreciate a response.

I'm afraid not. The encoded output will decode to floating-point samples but the library only accepts 16-bit samples as input.