On Thu, 23 May 2019, Louigi Verona wrote:
> One of the pipelines takes the uploaded file and transcodes it into an mp3. The
> general idea is to convert the original file to wav, resample it to 44100, and
> then finally convert it to mp3 using LAME.
>
> There are several questions here.
>
> 1. Which tool to use for transcoding. Should it be SoX, or FFmpeg, or something
> else? A lot of the info out there seems to favor SoX, but a lot of that info is
> pretty old.
>
> 2. Does it make sense to resample to 44100 or to 48000? If it were opus, the
> answer if simple: 48000, because that's what the opus spec actually recommends.
> There is no such recommendation for mp3 files. Also, upsampling is not an
> innocent procedure and the converter has to be of high quality as well.
For file in to file out why would anyone resample at all? Just keep the
original sample rate for each file and be happy. For file size, 44k1 is
smaller, but not that much.
48k for opus is not just recomended, it is the speed opus works at
internally. The question is if you want to trust opus internal SRC to 48k
or you want to do your own. For opus using 48k in and out, means no SRC.
For 44k1 in and out, it means SRC to 48k going in and then src to 44k1
going out. But I guess the SRC for playing is not something you would have
control of anyway.
From a selling view point, you want the best experience for the
greatest number of your users. This probably means testing for cpu load
and best quality on a low end windows machine... I can't help there as I
can't find any windows machines in the house. What rate do the commercial
music distribution people use? (those that charge per song)
-- Len Ovens www.ovenwerks.net _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org https://lists.linuxaudio.org/listinfo/linux-audio-devReceived on Thu May 23 20:15:02 2019
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