[arch-proaudio] 'proaudio-settings' package

Jean-Michaël Celerier jeanmichael.celerier at gmail.com
Tue Jul 31 10:08:55 UTC 2018

> Yeah, looking forward to bringing that to [community] actually! :)

wow, thanks ! but it's not ready for prime-time yet, it's still in alpha.
I'll drop you a mail when sufficient battle-testing will have occured,
hopefully by the end of the year.

> An audio device might come with a filter optimized for 48KHz usage.
Okay, that's what I was wondering about. I'm going to rummage a bit through
my soundcards's spec sheets to see if they mention this. Thanks !

Jean-Michaël Celerier

On Tue, Jul 31, 2018 at 11:31 AM, Ralf Mardorf <ralf.mardorf at alice-dsl.net>

> On Tue, 31 Jul 2018 11:01:32 +0200, Jean-Michaël Celerier wrote:
> >Do you have a source for this ?
> Yesno! Actually I'm the source, since I worked as a professional audio
> engineer for e.g. Brauner Microphones, Werner Nekes and other. However,
> perhaps you expect something as the below quoted Wikis:
> "Sampling rate 48,000 Hz
> The standard audio sampling rate used by professional digital video
> equipment such as tape recorders, video servers, vision mixers and so
> on. This rate was chosen because it could reconstruct frequencies up to
> 22 kHz and work with 29.97 frames per second NTSC video - as well as 25
> frame/s, 30 frame/s and 24 frame/s systems. With 29.97 frame/s systems
> it is necessary to handle 1601.6 audio samples per frame delivering an
> integer number of audio samples only every fifth video frame.[9]  Also
> used for sound with consumer video formats like DV, digital TV, DVD,
> and films. The professional Serial Digital Interface (SDI) and
> High-definition Serial Digital Interface (HD-SDI) used to connect
> broadcast television equipment together uses this audio sampling
> frequency. Most professional audio gear uses 48 kHz sampling, including
> mixing consoles, and digital recording devices." -
> https://en.wikipedia.org/wiki/Sampling_(signal_processing)#Sampling_rate
> "To avoid aliasing, the input to an ADC must be low-pass filtered to
> remove frequencies above half the sampling rate. This filter is called
> an anti-aliasing filter, and is essential for a practical ADC system
> that is applied to analog signals with higher frequency content." -
> https://en.wikipedia.org/wiki/Analog-to-digital_converter#Sampling_rate
> An audio device might come with a filter optimized for 48KHz usage.
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