alac_at only converts 16 bit sample

[mertozan]

I'm converting 24-bit DTS file to alac but always converts 16 bit per samples. I also trying "-sample_fmt s32p" not working.
Log file is here:

./ffmpeg -i 5min.mkv -map 0:1 -c:a alac_at 5min.m4a
ffmpeg version N-108816-g874a6f2090-tessus  https://evermeet.cx/ffmpeg/  Copyright (c) 2000-2022 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3 --pkg-config-flags=--static --disable-ffplay
  libavutil      57. 39.101 / 57. 39.101
  libavcodec     59. 51.100 / 59. 51.100
  libavformat    59. 34.101 / 59. 34.101
  libavdevice    59.  8.101 / 59.  8.101
  libavfilter     8. 49.101 /  8. 49.101
  libswscale      6.  8.112 /  6.  8.112
  libswresample   4.  9.100 /  4.  9.100
  libpostproc    56.  7.100 / 56.  7.100
[matroska,webm @ 0x7fdc71c11880] Could not find codec parameters for stream 0 (Video: h264, none(tv, bt709, progressive), 1920x1080): unspecified pixel format
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, matroska,webm, from '5min.mkv':
  Metadata:
    title           : The Dark Knight Rises 2012 IMAX BluRay 1080p DTS AC3 x264-MgB
    ENCODER         : Lavf59.16.100
  Duration: 00:05:00.10, start: 0.005000, bitrate: 8997 kb/s
  Chapters:
    Chapter #0:0: start 0.000000, end 300.000000
      Metadata:
        title           : Chapter 1
  Stream #0:0(eng): Video: h264, none(tv, bt709, progressive), 1920x1080, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 1k tbn (default)
    Metadata:
      title           : The Dark Knight Rises 2012 IMAX BluRay 1080p By MgB
      DURATION        : 00:05:00.096000000
  Stream #0:1(eng): Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (default)
    Metadata:
      title           : DTS 5.1 @ 1509 Kbps
      DURATION        : 00:05:00.004000000
Stream mapping:
  Stream #0:1 -> #0:0 (dts (dca) -> alac (alac_at))
Press [q] to stop, [?] for help
Output #0, ipod, to '5min.m4a':
  Metadata:
    title           : The Dark Knight Rises 2012 IMAX BluRay 1080p DTS AC3 x264-MgB
    encoder         : Lavf59.34.101
  Chapters:
    Chapter #0:0: start 0.000000, end 299.995000
      Metadata:
        title           : Chapter 1
  Stream #0:0(eng): Audio: alac (alac / 0x63616C61), 48000 Hz, 5.1(side), s16, 128 kb/s (default)
    Metadata:
      title           : DTS 5.1 @ 1509 Kbps
      DURATION        : 00:05:00.004000000
      encoder         : Lavc59.51.100 alac_at
size=   67772kB time=00:04:59.94 bitrate=1851.0kbits/s speed=38.8x       
video:0kB audio:67737kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.052030%

[mertozan]

I would be glad if you could provide information. Am I doing a mistake? Or is ffmpeg relevant.

[Balling]

​https://patchwork.ffmpeg.org/project/ffmpeg/patch/20210531150302.359628-1-val.zapod.vz@gmail.com/

Even decoding of 24 bit does not work.

[mertozan]

I’m sorry but why are you changing the priority. This is a bug. Core Audio working 24-bit.

[Balling]

Replying to mertozan:

I’m sorry but why are you changing the priority. This is a bug. Core Audio working 24-bit.

Carl considers this feature request, but that is of course a bug, just like decoding.

[mertozan]

Replying to Balling:

Replying to mertozan:

I’m sorry but why are you changing the priority. This is a bug. Core Audio working 24-bit.

Carl considers this feature request, but that is of course a bug, just like decoding.

You are absolutely right. Where can I download static ffmpeg with this patch used? I think this patch fixes 24-bit encoding also. ​https://patchwork.ffmpeg.org/project/ffmpeg/patch/20210531150302.359628-1-val.zapod.vz@gmail.com/

[Balling]

I think this patch fixes 24-bit encoding also

My patch is only for ffat_create_decoder, it cannot affect encoder.

[mertozan]

Replying to Balling:

I think this patch fixes 24-bit encoding also

My patch is only for ffat_create_decoder, it cannot affect encoder.

Thank you for the answer, Balling. I tried my chance :(

[mertozan2]

I have question then, I use alac instead alac_at (because alac allows 24bit), what I lose. Actually I am wondering this question answer.. ​https://stackoverflow.com/questions/70786595/which-alac-encoder-should-i-use-with-ffmpeg-alac-alac-at
Thank you for read. I am waiting for the precise answers.

[mertozan2]

Any answer

[mertozan2]

I’m wondering if I use alac instead alac_at, I have a lot free space bently no need size. Will I lose quality. what is basic different alac and alac_at. Please explain to me if you know. Thank you.

[mertozan2]

Please somebody help me. I've got some work on hold.

[mertozan2]

Please somebody help me.. I've got some work on hold.

[Elon Musk]

Only wavpack encoder supports encoding lossless float sample format as provided by DTS core only audio.

Other lossless encoders will convert float to other non-float sample format which may ruing quality.

[mertozan2]

Thank you for answer. Is the violation of the Apple License the reason why it cannot be fixed? And last one question alac or alac_at what is difference. Elon Musk

[mertozan2]

Last question please, then I’ll release you Elon Musk :)

[Elon Musk]

This is not support channel, please refrain from misusing this bug tracker for random questions.

[mertozan2]

You’re right for about my last question. Before I’m report a bug, but ballin saying this is not a bug, I don’t understand that. After you coming over here and you saying this is a license violation if this is a true, why part of ffmpeg inside. You’re right this is not a support channel. I just already report a bug before. Encoder not working correctly (decode and encode both). in my opinion if there is a license problem it should either be removed completely or repaired.

[Elon Musk]

alac_at does not support higher sample formats, and I'm not going to research if this is audiotoolbox limitation or not.
alac supports higher sample formats.