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This repository has been archived by the owner on Mar 30, 2024. It is now read-only.
In order to use the native FFmpeg AAC encoder, an additional argument must be passed to
the command line of FFmpeg when transcoding the raw audio file to .m4a file format.
The command line would have to look something like this: ffmpeg -y -ac 1 -f s16le -i RAW_TEMP_FILE -strict experimental -ab 48k OUTPUT.m4a
This is an issue with the audiosprite utility in your repository. But I can't open new issues there.
I haven't tested if that has any negative side-effects when not using the native AAC encoder because
that's not available to me.
In order to use the native FFmpeg AAC encoder, an additional argument must be passed to
the command line of FFmpeg when transcoding the raw audio file to .m4a file format.
The command line would have to look something like this:
ffmpeg -y -ac 1 -f s16le -i RAW_TEMP_FILE -strict experimental -ab 48k OUTPUT.m4a
This is an issue with the audiosprite utility in your repository. But I can't open new issues there.
I haven't tested if that has any negative side-effects when not using the native AAC encoder because
that's not available to me.
source: https://trac.ffmpeg.org/wiki/Encode/AAC
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