? Do you have any references for me to investigate, I am trying to understand
how noise is reduced by introducing higher sampling rates. (I tried to search,
but maybe it is so obvious that nobody even explains it)
This is not very obvious. It requires you to understand basic signal processing
theory. I will give some pointers below.
Any physical signal (e.g. audio coming out of speaker, current driving the
speaker) is analog and contiguous in nature. On other hand, all codecs (and
everything in DSP) works on discrete signals (samples).
When you convert from discrete samples to analog signal, you need to apply an
ideal filter, which has a frequency response of 1.0 (0 dB attenuation) in it?s
passband and 0.0 (?infinite dB) in its stopband. The most important thing to
understand here is that such an ideal filter does not exist in reality. What we
do (most DACs do this) is to approximate such a filter using IIR?s or an FIR
with some delay (sufficient delay that the truncation of impulse response does
not matter much).
With higher sampling rates, it gives us flexibility to design this filter such
that the stopband attenuation is large enough (although not ?infinite dB) and
the noise is filtered to inaudible levels.
From: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] On Behalf Of
Edwin van den Oetelaar
Sent: Saturday, June 07, 2014 5:22 PM
To: Andrew Lentvorski
Cc: opus at xiph.org; Jean-Marc Valin
Subject: Re: [opus] High Sampling Rates
On Sat, Jun 7, 2014 at 10:58 AM, Andrew Lentvorski <bsder at
allcaps.org<mailto:bsder at allcaps.org>> wrote:
On 6/7/14, 1:55 AM, Jean-Marc Valin wrote:> Actually... no! 24-bit can indeed be useful as extra margin and Opus
> can actually represent even more dynamic range than 24-bit PCM. That's
> not the case for 192 kHz. There's no "margin" that 192 kHz
buys you
> over 48 kHz. You can do as much linear filtering as you like, the
> stuff above 20 kHz isn't going to help you.
But lots of effects are not linear--simulating a tube guitar amplifier,
for example.
Even something as straightforward as resampling a signal to 44.1KHz is
going to benefit from starting at 192KHz rather than 48KHz.
There may not be more signal information but there will be less noise.
Hello Andrew,
Do you have any references for me to investigate, I am trying to understand how
noise is reduced by introducing higher sampling rates. (I tried to search, but
maybe it is so obvious that nobody even explains it)
In resampling - as far as I understand it - you first limit the bandwidth to the
1/2 the sampling rate to prevent aliasing problems.
If your input signal contains no energy outside the resampled bandwidth to start
with, how is this going to increase the signal to noise ratio?
Thanks for your time,
Edwin van den Oetelaar
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