Bill Unruh wrote:
24 bit quantization error ( being at 135dB below the full signal--
ie completely and utterly inaudible
Cumulative error in the (polyphase or not) digital filter arithmetic
is larger than sampling error. Still small, but how small depends on
how the filters are implemented.
On Sun, 28 Dec 2008, Jamie Lokier wrote:
Bill Unruh wrote:
24 bit quantization error ( being at 135dB below the full signal--
ie completely and utterly inaudible
Cumulative error in the (polyphase or not) digital filter arithmetic
is larger than sampling error. Still small, but how small
On Sun, 21 Dec 2008 03:34:50 +
Jamie Lokier ja...@shareable.org wrote:
Even 2x upsampling introduces distortion.
No, it doesn't.
It can be implemeted as a pure linear transform with constant coefficients,
thus it would be a simple FIR LPF.
Regards,
Sergei.
On Sun, 21 Dec 2008, Jamie Lokier wrote:
But then there's people who say that all analogue output above 20kHz
or so is pointless anyway. :-) (I have no opinion either way on this.
On the one hand, they could be right. The upper auditory limit is
well known. On the other hand, there is a
Sergei Steshenko wrote:
On Sun, 21 Dec 2008 03:34:50 +
Jamie Lokier ja...@shareable.org wrote:
Even 2x upsampling introduces distortion.
No, it doesn't.
It can be implemeted as a pure linear transform with constant coefficients,
thus it would be a simple FIR LPF.
It depends
Bill Unruh wrote:
On Sun, 21 Dec 2008, Jamie Lokier wrote:
But then there's people who say that all analogue output above 20kHz
or so is pointless anyway. :-) (I have no opinion either way on this.
On the one hand, they could be right. The upper auditory limit is
well known. On the
On Sun, 21 Dec 2008, Jamie Lokier wrote:
Sergei Steshenko wrote:
On Sun, 21 Dec 2008 03:34:50 +
Jamie Lokier ja...@shareable.org wrote:
Even 2x upsampling introduces distortion.
No, it doesn't.
It can be implemeted as a pure linear transform with constant coefficients,
thus it
Dominique Michel wrote:
Upsampling does nothing for analog reproduction because you cannot get more
informations that what you get from the DAC output. Upsampling is only a
matter
of cost:
higher the ADC frequency, cheaper is the output filter for approximately the
same analog result at
Sergei Steshenko wrote:
The argument about 192 kHz is true, but why not run the DAC at the
'normal' 96 kHz then? I would prefer a 48 - 96 conversion over a 48 -
110 conversion!.
And how about interference: won't a 96 - 110 conversion give any
interference at 14 kHz? Right in the audible
Paulo Moura Guedes wrote:
BTW, Benchmark DAC1 resamples internally to 110kHz:
On the question of: Why does the DAC1 re-sample to 110 kHz? Here is
why: it is the highest frequency to maintain the full oversampling of
the D-A chip. EVERY D-to-A chip on the market cuts the oversampling rate
in
On Fri, 19 Dec 2008 11:06:06 +0100
Matthijs ten Berge tenberg...@yahoo.com wrote:
Paulo Moura Guedes wrote:
BTW, Benchmark DAC1 resamples internally to 110kHz:
On the question of: Why does the DAC1 re-sample to 110 kHz? Here is
why: it is the highest frequency to maintain the full
Le Fri, 19 Dec 2008 13:22:48 +0200,
Sergei Steshenko steshenko_ser...@list.ru a écrit :
On Fri, 19 Dec 2008 11:06:06 +0100
Matthijs ten Berge tenberg...@yahoo.com wrote:
Paulo Moura Guedes wrote:
BTW, Benchmark DAC1 resamples internally to 110kHz:
On the question of: Why does the
On Fri, 19 Dec 2008 13:16:35 +
Dominique Michel dominique.mic...@vtxnet.ch wrote:
Upsampling does nothing for analog reproduction because you cannot get more
informations that what you get from the DAC output.
I didn't mean upsampling adds audio info, I meant it makes analog smoothing
At Mon, 15 Dec 2008 15:01:08 -0800 (PST),
Bill Unruh wrote:
On Tue, 16 Dec 2008, Sergei Steshenko wrote:
On Mon, 15 Dec 2008 23:40:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 15 Dec 2008 23:17:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On
Bill Unruh wrote:
On Mon, 15 Dec 2008, Paulo Moura Guedes wrote:
[...]
The ASRC, as the name implies, is not syncronized to the clock of
the incoming digital signal. Therefore, its performance is independant of
the
This makes no sense at all. If the incoming signal is a digital signal,
Paulo Moura Guedes wrote:
For my case where I connect to the Benchmark DAC1 via USB (which supports
24bit 96khz), does my sound card have any influence in the process?
No.
If the sound card does 24bit 96khz, and the original sound file is 24bit
96khz, ALSA will not touch/modify the samples.
On Mon, 15 Dec 2008, James Courtier-Dutton wrote:
Paulo Moura Guedes wrote:
For my case where I connect to the Benchmark DAC1 via USB (which supports
24bit 96khz), does my sound card have any influence in the process?
No.
If the sound card does 24bit 96khz, and the original sound file is
On Mon, 15 Dec 2008 12:56:28 -0800 (PST)
Bill Unruh un...@physics.ubc.ca wrote:
What kind of resampling does ALSA do these days-- linear interpolation ( which
is fast and does not have any delay, but introduces loads and loads of
distortion and noise) or what?
You may choose. There is now
On Mon, 15 Dec 2008 23:40:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 15 Dec 2008 23:17:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 15 Dec 2008 12:56:28 -0800 (PST)
Bill Unruh un...@physics.ubc.ca wrote:
What kind of resampling does
On Mon, 15 Dec 2008 23:17:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 15 Dec 2008 12:56:28 -0800 (PST)
Bill Unruh un...@physics.ubc.ca wrote:
What kind of resampling does ALSA do these days-- linear interpolation (
which
is fast and does not have any delay,
On Tue, 16 Dec 2008, Sergei Steshenko wrote:
On Mon, 15 Dec 2008 23:40:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 15 Dec 2008 23:17:11 +0200
Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 15 Dec 2008 12:56:28 -0800 (PST)
Bill Unruh un...@physics.ubc.ca
On Friday 12 December 2008 22:58:15 James Courtier-Dutton wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
find much info on the web. I'm using ALSA.
Some questions:
- does Linux/ALSA features dynamic sample rates?
- is it
On Friday 12 December 2008 20:34:21 Sergei Steshenko wrote:
On Fri, 12 Dec 2008 20:24:25 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
I don't know about ALSA.
So, ALSA, as well as many Linux applications, have a choice of qualities
for resampling, and I bet you won't hear the
Hi,
On Friday 12 December 2008 21:07:49 stan wrote:
Paulo Moura Guedes wrote:
On Thursday 11 December 2008 01:20:23 stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I
can't find much
info on the web. I'm using ALSA.
I
On Mon, 15 Dec 2008, Paulo Moura Guedes wrote:
On Friday 12 December 2008 20:34:21 Sergei Steshenko wrote:
On Fri, 12 Dec 2008 20:24:25 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
I don't know about ALSA.
So, ALSA, as well as many Linux applications, have a choice of qualities
On Monday 15 December 2008 01:17:23 Bill Unruh wrote:
[...]
I assume you spent $1000 on some oxygen free copper cables as well.
Not really, should I? :P
So, you're the only soul in the world who can see the truth about the
Benchmark DAC1! Congratulations.
Can you recommend any cheaper DAC?
I haven't seen a Benchmark in Australia so have no idea.. But for a good cheap reliable USB soundcard I have found the Behringer UCA-202 is fine.. Again I don't know if it does internal re-sampling so can't answer the bit perfect question here...
On Mon Dec 15 1:48 , Paulo Moura Guedes
Paulo Moura Guedes wrote:
On Thursday 11 December 2008 07:35:25 Clemens Ladisch wrote:
The default device (named default) uses automatic resampling, but the
spdif device does not.
So, I have to manually set the sample-rate, depending on the files I will
play?
No, ALSA automatically
On Thursday 11 December 2008 01:20:23 stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
find much
info on the web. I'm using ALSA.
Some questions:
- does Linux/ALSA features dynamic sample rates?
yes. You can specify
On Thursday 11 December 2008 07:35:25 Clemens Ladisch wrote:
Paulo Moura Guedes wrote:
- does Linux/ALSA features dynamic sample rates?
The default device (named default) uses automatic resampling, but the
spdif device does not.
So, I have to manually set the sample-rate, depending on the
On Thursday 11 December 2008 01:42:12 Bill Unruh wrote:
On Wed, 10 Dec 2008, stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
What is bit perfect output? Sound output is an analog process ( sound
waves in the air are analog). You
BTW, Benchmark has very nice instructions for getting bit-perfect output, but
only for Windows and Mac:
http://extra.benchmarkmedia.com/wiki/index.php/Computer_Audio_Playback_-
_Setup_Guide
Foobar2000 is a great player for Windows, which have complete configuration
options:
On Fri, 12 Dec 2008, Paulo Moura Guedes wrote:
On Thursday 11 December 2008 01:42:12 Bill Unruh wrote:
On Wed, 10 Dec 2008, stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
What is bit perfect output? Sound output is an analog
On Fri, 12 Dec 2008, Paulo Moura Guedes wrote:
On Thursday 11 December 2008 01:20:23 stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
find much
info on the web. I'm using ALSA.
Some questions:
- does Linux/ALSA features dynamic
It's not the first time perfect bits are discussed here.
Last time I managed to stop the fruitless discussion part by reminding that any
CD with 44100Hz sample rate is produced from material sample at much higher
sample rate, and quality resampling is used.
So, rather than being afaraid of
On Fr, 12.12.08 15:53 Paulo Moura Guedes mo...@kdewebdev.org wrote:
- is it possible to set the bit-depth? (in my case to 24 bit)
Yes, if the hardware supports it.
It's a Benchmark DAC1 which supports 24 bit. Where can I set it?
The DAC doesn't matter for ALSA and its settings, but
On Friday 12 December 2008 18:20:32 Thomas Kuther wrote:
On Fr, 12.12.08 15:53 Paulo Moura Guedes mo...@kdewebdev.org wrote:
- is it possible to set the bit-depth? (in my case to 24 bit)
Yes, if the hardware supports it.
It's a Benchmark DAC1 which supports 24 bit. Where can I set
On Friday 12 December 2008 17:31:43 Bill Unruh wrote:
On Fri, 12 Dec 2008, Paulo Moura Guedes wrote:
On Thursday 11 December 2008 01:42:12 Bill Unruh wrote:
On Wed, 10 Dec 2008, stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
On Friday 12 December 2008 17:34:18 Bill Unruh wrote:
- does Linux/ALSA features dynamic sample rates?
yes. You can specify whatever rate you want and it will
happen. eg 57,325 frames/second.
I meant dynamic, e.g., if my FLAC file has a sample rate of 96KHz, that
will be used, but
On Friday 12 December 2008 17:52:29 Sergei Steshenko wrote:
It's not the first time perfect bits are discussed here.
Last time I managed to stop the fruitless discussion part by reminding that
any CD with 44100Hz sample rate is produced from material sample at much
higher sample rate, and
On Friday 12 December 2008 19:27:51 Sergei Steshenko wrote:
On Fri, 12 Dec 2008 18:58:37 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
On Friday 12 December 2008 17:52:29 Sergei Steshenko wrote:
It's not the first time perfect bits are discussed here.
Last time I managed to
On Fri, 12 Dec 2008 19:45:00 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
On Friday 12 December 2008 19:27:51 Sergei Steshenko wrote:
On Fri, 12 Dec 2008 18:58:37 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
On Friday 12 December 2008 17:52:29 Sergei Steshenko wrote:
On Friday 12 December 2008 19:50:31 Sergei Steshenko wrote:
On Fri, 12 Dec 2008 19:45:00 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
On Friday 12 December 2008 19:27:51 Sergei Steshenko wrote:
On Fri, 12 Dec 2008 18:58:37 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
On Fri, 12 Dec 2008 20:24:25 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
I don't know about ALSA.
So, ALSA, as well as many Linux applications, have a choice of qualities for
resampling, and I bet you won't hear the difference between middle and high
quality.
There is no OS
On Fri, 12 Dec 2008 20:24:25 +
Paulo Moura Guedes mo...@kdewebdev.org wrote:
Anyway, this seems the typical response when something is not possible to do:
your computer doesn't work with this operating system? Buy another. :)
Huh ?
People have already answered you how to avoid
Paulo Moura Guedes wrote:
On Thursday 11 December 2008 01:20:23 stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
find much
info on the web. I'm using ALSA.
I think you are somewhat confused. ALSA is a low level
interface
On Fri, 12 Dec 2008 21:36:02 +0100
Florian Faber fa...@faberman.de wrote:
Paulo,
I would like that all the resampling is made by the DAC because I can trust
on its quality.
Bwahahahaha!
Flo
For example, here: http://www.futurlec.com/ICSFOthers.shtml one can see
that DACs are
On Fri, 12 Dec 2008 14:07:49 -0700
stan ghjeold_i_m...@cox.net wrote:
The long and short of it is that alsa is as bit perfect as
you ask it to be and as the device and digitial input it is
using is. The method I outlined above, or close facsimile
thereof, should be able to prove it to
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't find
much
info on the web. I'm using ALSA.
Some questions:
- does Linux/ALSA features dynamic sample rates?
- is it possible to set the bit-depth? (in my case to 24 bit)
- what other
I'm trying to get bit perfect output out of my linux box, but I can't find much
info on the web. I'm using ALSA.
Some questions:
- does Linux/ALSA features dynamic sample rates?
- is it possible to set the bit-depth? (in my case to 24 bit)
- what other variables do i have to consider in order
I'll answer what I can.
1. Not sure of your question.. here so I'll leave this along
2. This is hardware dependant.. many/most onboard digital outputs are fixed at 16 bit 48 khz devices like the M-Audio delta series are flexible and can be set via software application (i.e.
On Wed, 10 Dec 2008, stan wrote:
Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't
What is bit perfect output? Sound output is an analog process ( sound waves
in the air are analog). You do know that you can just copy the sound file
whereever you
Paulo Moura Guedes wrote:
- does Linux/ALSA features dynamic sample rates?
The default device (named default) uses automatic resampling, but the
spdif device does not.
- is it possible to set the bit-depth? (in my case to 24 bit)
Yes, if the hardware supports it.
- what other variables do i
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