I have been able to convert normal (16bits@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@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 _______________________________________________ libav-user mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-user
