2011/3/16 Justin Ruggles <[email protected]> > Hi, > > On 03/16/2011 06:04 AM, Dagfinn Stangeland wrote: > > > 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. > > The ALAC encoder does not support 24-bit. And the MP4 muxer does not > support the high sample rate. > > > 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. > > > Maybe dbpoweramp? It has an ALAC encoder, but I don't know what it > supports. > > -Justin > > _______________________________________________ > libav-user mailing list > [email protected] > https://lists.libav.org/mailman/listinfo/libav-user >
Ok, too bad. I can use XLD on my mac. Will have to do without the muscle of my ffmpeg box. D _______________________________________________ libav-user mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-user
