convert audio files to serve them on website

[David Froger]

Hello,

I'm building a website that allow talks to be uploaded (HTTPS POST)
and then be streamed (html5 audio element) or downloaded (HTTPS GET).
I'm also thinking to make them available as podcast. As I have no
experience with audio files, I'm looking for advices.

I would like the upload/download/stream to be as portable as possible:
Currently, the talks may be recorded for example from a voice record, a
smartphone, a tablet. It is then directly uploaded to the website. I've
encountered these formats: (.mp3, .m4a, .3gp) but I would like to
support as much formats as possible. It may then be streamed or
downloaded from one of the major desktop (GNU/Linux, OSX, Windows) or
mobile/tablet (Androïd, IOS) browsers.

So I would like to convert the uploaded file with ffmpeg before serving
it (done by the webserver).

Some questions I have:
- Would .mp3 be the most appropriate format?
- Should I adapt the ffmpeg command according to the input file, or can I use
  an "universal command", something like: ffmpeg -i file.{.m4a,3gp,...} -vn file.mp3
- Should I add "-acodec libmp3lame", or is it implicit from the output
  file .mp3 extension?
- Is there any benefit from adding other options? For example here
  https://stackoverflow.com/a/12952172 I've seen "-ar 44100", but why
  not just keep the input frequency? (same for "-ac 2" and "-b:a 192k",
  why not just keep what is in the input, is it just to optimize the
  size of the output file?) Any other useful option?

Thanks for reading!


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[David Froger]

I'm also planning to use the `-progress` option, to display a progress
bar to the uploader client.

Quoting David Froger (2021-01-20 00:03:20)

> Hello,
>
> I'm building a website that allow talks to be uploaded (HTTPS POST)
> and then be streamed (html5 audio element) or downloaded (HTTPS GET).
> I'm also thinking to make them available as podcast. As I have no
> experience with audio files, I'm looking for advices.
>
> I would like the upload/download/stream to be as portable as possible:
> Currently, the talks may be recorded for example from a voice record, a
> smartphone, a tablet. It is then directly uploaded to the website. I've
> encountered these formats: (.mp3, .m4a, .3gp) but I would like to
> support as much formats as possible. It may then be streamed or
> downloaded from one of the major desktop (GNU/Linux, OSX, Windows) or
> mobile/tablet (Androïd, IOS) browsers.
>
> So I would like to convert the uploaded file with ffmpeg before serving
> it (done by the webserver).
>
> Some questions I have:
> - Would .mp3 be the most appropriate format?
> - Should I adapt the ffmpeg command according to the input file, or can I use
>   an "universal command", something like: ffmpeg -i file.{.m4a,3gp,...} -vn file.mp3
> - Should I add "-acodec libmp3lame", or is it implicit from the output
>   file .mp3 extension?
> - Is there any benefit from adding other options? For example here
>   https://stackoverflow.com/a/12952172 I've seen "-ar 44100", but why
>   not just keep the input frequency? (same for "-ac 2" and "-b:a 192k",
>   why not just keep what is in the input, is it just to optimize the
>   size of the output file?) Any other useful option?
>
> Thanks for reading!

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