[FFmpeg-user] Convert audio FLAC to ALAC (apple lossless), 24bits 96kHz

[Dagfinn Stangeland]

I have been able to convert normal (16bits at 44,1kHz) FLAC audiofiles to ALAC
using ffmpeg. Searching around I found this little line that has worked
fine:

for i in *.flac; do ffmpeg -i "$i" -acodec alac -map_meta_data 0:0,s0
> "`basename "$i" .flac`.m4a"; done;
>

The above line converts all flac in a dir to alac and preserves tag info.

I was hoping to use ffmpeg to convert HQ FLAC files (24bits at 96kHz) to ALAC
preserving bitdepth and sampling rate.

Here's the output using the above "script":

FFmpeg version git-2611e52, Copyright (c) 2000-2011 the FFmpeg developers
>   built on Feb  6 2011 10:03:23 with gcc 4.5.2 20110127 (prerelease)
>   configuration: --prefix=/usr --enable-gpl --enable-libmp3lame
> --enable-libvorbis --enable-libfaac --enable-libxvid --enable-libx264
> --enable-libvpx --enable-libtheora --enable-postproc --enable-shared
> --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb
> --enable-libschroedinger --enable-libopenjpeg --enable-version3
> --enable-nonfree --enable-runtime-cpudetect --disable-debug
>   libavutil    50. 36. 0 / 50. 36. 0
>   libavcore     0. 16. 1 /  0. 16. 1
>   libavcodec   52.108. 0 / 52.108. 0
>   libavformat  52. 94. 0 / 52. 94. 0
>   libavdevice  52.  2. 3 / 52.  2. 3
>   libavfilter   1. 74. 0 /  1. 74. 0
>   libswscale    0. 12. 0 /  0. 12. 0
>   libpostproc  51.  2. 0 / 51.  2. 0
> [flac @ 0xb98510] max_analyze_duration reached
> Input #0, flac, from '1-Nikolai RimskyKorsakov The S.flac':
>   Metadata:
>     ALBUM ARTIST    : Various Artists
>     ARTIST          : Minnesota Orchestra / Eiji Oue
>     ALBUM           : HDtracks Ultimate Download Experience
>     TITLE           : Nikolai Rimsky-Korsakov: The Snow Maiden - Dance of
> the Tumblers
>     track           : 1
>     GENRE           : Classical / Jazz
>     DATE            : 2009
>     HDTRACKS        : www.hdtracks.com
>   Duration: 00:03:54.64, bitrate: 2775 kb/s
>     Stream #0.0: Audio: flac, 96000 Hz, 2 channels, s32
> [ipod @ 0xb997d0] track 0: output format does not support sample rate
> 96000hz
> Output #0, ipod, to '1-Nikolai RimskyKorsakov The S.m4a':
>   Metadata:
>     encoder         : Lavf52.94.0
>     Stream #0.0: Audio: alac, 96000 Hz, 2 channels, s16, 64 kb/s
> Stream mapping:
>   Stream #0.0 -> #0.0
> Could not write header for output file #0 (incorrect codec parameters ?)
>

Is it so that the alac encoder really does not support such a high sample
rate? Or do I need to pass options to ffmpeg to enable this?
I notice that "ipod" is mentioned in the output stream info, I do not
understand the significance.

Sidenote: I'm not aware of any other tools that supports FLAC to ALAC
conversion. If anyone knows of better suited tools, please let me know.

Regards,

Dagfinn Stangeland