Any of you know how to do this ...
Tried a few tricks but no good thus far
Thanx
shan
Any of you know how to do this ...
Tried a few tricks but no good thus far
Thanx
shan
Linux is Latin for off-the-beaten-track
what I like MOST about our Ubuntu ... The Community ie 50 brains are better than one
Playing with Slackware too now ...
ShanArt
You can use "File"->"Export Audio"->"Export" and then select "(external program)" as format. As the command you should enter
sourceCode:ffmpeg -i - -acodec alac "%f"
Holger
Thank u very much Holger was trying to do that through "Custom FFmpeg Export" and was getting nowhere fast
Last edited by shantiq; May 17th, 2020 at 04:39 PM.
Linux is Latin for off-the-beaten-track
what I like MOST about our Ubuntu ... The Community ie 50 brains are better than one
Playing with Slackware too now ...
ShanArt
really wanted a 24-bit file so after much horsing around this does it >>
Code:ffmpeg -i - -c:a alac -sample_fmt s32p "%f"
LOG:
Code:ffmpeg -i - -c:a alac -sample_fmt s32p "/home/shan/Desktop/NAMEOFFILE.m4a" ffmpeg version N-91586-g90dc584 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 7 (Ubuntu 7.3.0-16ubuntu3) configuration: --prefix=/home/shan/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/shan/ffmpeg_build/include --extra-ldflags=-L/home/shan/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/shan/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-openssl --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree libavutil 56. 18.102 / 56. 18.102 libavcodec 58. 22.101 / 58. 22.101 libavformat 58. 17.101 / 58. 17.101 libavdevice 58. 4.101 / 58. 4.101 libavfilter 7. 26.100 / 7. 26.100 libswscale 5. 2.100 / 5. 2.100 libswresample 3. 2.100 / 3. 2.100 libpostproc 55. 2.100 / 55. 2.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'pipe:': Duration: N/A, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> alac (native)) [alac @ 0x55ab92dc1e00] encoding as 24 bits-per-sample Output #0, ipod, to '/home/shan/Desktop/NAMEOFFILE.m4a': Metadata: encoder : Lavf58.17.101 Stream #0:0: Audio: alac (alac / 0x63616C61), 48000 Hz, stereo, s32p (24 bit), 128 kb/s Metadata: encoder : Lavc58.22.101 alac
Linux is Latin for off-the-beaten-track
what I like MOST about our Ubuntu ... The Community ie 50 brains are better than one
Playing with Slackware too now ...
ShanArt
128k? That seems strange for a lossless codec. But I'm not too familiar with ALAC, so maybe I'm misunderstanding something.Stream #0:0: Audio: alac (alac / 0x63616C61), 48000 Hz, stereo, s32p (24 bit), 128 kb/s
yes since the truth is more in the 1400/1800k range but this is what ffmpeg displays when high conversions take place; it is as if it thinks it is so high we will just say the default for an mp3 conversion 128k
I can assure you the result is what it should be; the reading is just odd :]
Linux is Latin for off-the-beaten-track
what I like MOST about our Ubuntu ... The Community ie 50 brains are better than one
Playing with Slackware too now ...
ShanArt
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