Hi all
Just to give feedback about my experiments. I found it confusing to have
multiple upsampling options and disabled C-3PO. Instead I use the SoX in
piCorePlayer. It seems to me the best settings (in my case) are:
Max sample rate: 88200,96000,176400,192000
Upsample setting:
cfuttrup wrote:
> Hi Marco - thank you for your help and for making the C-3PO plugin. Are
> you a Star Wars fan? :-)
Not really a fan, I like it, but I most like George Lucas raise against
the major's power, this is my tribute to a free man.
cfuttrup wrote:
I just wish to mention here that,
cfuttrup wrote:
> I received nothing but noise out of the above settings.
>
> I then removed the support for low PCM sample rates: 8000 - 16000 -
> 24000 - 32000 ... I have no use for them. I also changed output format
> to uncompressed FLAC (from basic WAV). Now it plays fine.
To output
I received nothing but noise out of the above settings.
I then removed the support for low PCM sample rates: 8000 - 16000 -
24000 - 32000 ... I have no use for them. I also changed output format
to uncompressed FLAC (from basic WAV). Now it plays fine.
/Claus
Here's my settings for C-3PO:
2639826399
/Claus
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marcoc1712 wrote:
> No, you need to investigate the input file (or stream) to determinate
> the sample rate. I implemented this in C-3PO plugin, you could have a
> look at it, is in the third party plugin lis in LMS.
Hi Marco - thank you for your help and for making the C-3PO plugin. Are
you a
schiff1108 wrote:
> I found here some descriptions on upsampling rules using Squeezelite.
> However, there is one thing I am not able to get to work as described
> above.
> A rule that makes Squeezelite to multiply in full numbers, only.
>
> 44.1 -> 176.4
> 96 -> 192
> 192 -> 192 (no
JohnSwenson wrote:
>
>
> The current implementation in Squeezelite does upsampling to the
> highest interger rate your DAC cupports. Thus if your DAC's maximum
> rate is 192, it will upsample to 44.1 to 176.4. If your DAC does not
> support 176.4, you can use the -r option (or max
I know this thread died out a long time ago but I think that there are
still many of us who would like a deeper understanding of the upsampling
options offered in piCorePlayer and I assume other linux-based players.
My linux skills are close to zero. My hope is to help others get the
most from
Not really. Nothing comes up in the search, in the majority of searches
that I've conducted.:(
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View this thread:
Hi John,
I've been following all this and understand where you're coming from.
But what is a CSP?
Murray (N.Z.)
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View this thread:
murrays wrote:
Hi John,
I've been following all this and understand where you're coming from.
But what is a CSP?
'Community Squeeze Project' (http://www.communitysqueeze.org/)
Ralphy
*1*-Touch, *4*-Classics, *2*-Booms, *1*-Reverted UE Radio
*1*-Squeezeslave, *2*-Squeezeplays,
ralphy wrote:
'Community Squeeze Project' (http://www.communitysqueeze.org/)
Ah, thanks. That's something interesting to look into. Strange it
didn't come up in the search.
Murray (N.Z.)
murrays's Profile:
marcoc1712 wrote:
John,
could I ask You witch USB DAC are you using? If is kind of a KIt or a
commercial product, could you point me to the source?
is a kind of the one you'll be using in CSP1?
thx a lot.
p.s.
I could not understand why if you upsample at i.e. 352.8 you still
JohnSwenson wrote:
What you want is the upsampler to make a guess at filling in the
intermediate values between the .5V sample and the next sample.
John,
I've no doubt you know more about the subject than me but I think your
language, namely the appearance of want and guess in the same
bennyboyph wrote:
Cheers - it doesn't want to work at 352.8 or 384kHz, which is what I
need to bypass the digital filters of my PCM5102 chip :-(
OK. Sox maxrate is 192khz afaik. And your Squeezebox environment
(server and client) would have to support 384khz too.
::: ' Touch Toolbox and
Can anyone provide me with covert.conf codes to do this within LMS?
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View this thread: http://forums.slimdevices.com/showthread.php?t=99088
Try this:
The file needs to be called :
custom-convert.conf
It just has to have these 3 lines
Code:
flc flc * *
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -dcs $START$ $END$ -- $FILE$ | [sox] -D -q -t wav - -t flac -e
signed -C 0 -b 24 - rate -v -I -a
Dear Mr Swenson,
My name is Alex and I am a Master student (Media Management and
Entrepreneurship) at the University Fresenius, Cologne. I have seen you
in this fantastic blog and was really impressed by your ideas and
facts.
My team of three is looking for hardware experts who could help us
soundcheck wrote:
Try this:
The file needs to be called :
custom-convert.conf
The custom-convert.conf overrides convert.conf.
It just has to have these 3 lines
Code:
flc flc * *
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -dcs $START$
John,
could I ask You witch USB DAC are you using? If is kind of a KIt or a
commercial product, could you point me to the source?
is a kind of the one you'll be using in CSP1?
thx a lot.
p.s.
I could not understand why if you upsample at i.e. 352.8 you still need
a software filter, is not
Julf wrote:
Any specific reasons to prefer WAV? I think the general view is that
with lower-end processors (such as those in the squeezeboxes) that don't
have dedicated I/O processors the additional network load caused by the
wasted bits in WAV files causes more CPU load (and thus
Guys. Don't highjack this thread.
My main question still is:
What's your preferred least intrusive and highest quality Sox SRC
setting?
It seems that not anybody is able or willing to come up with a
recommendation!?!?
::: ' Touch Toolbox and more' (http://soundcheck-audio.blogspot.com)
First of all: WAV means: transport twice the number of bits over the
network which causes twice the load in all your network interfaces (NIC,
buffers,...). If you use WiFi it also means: decrypt twice the number of
bits, a process that needs much more CPU cycles than the simple FLAC
decoding.
I
This thread is about resampling qualities resp. differences between
filter settings.
And NOT about file formats.
::: ' Touch Toolbox and more' (http://soundcheck-audio.blogspot.com) :::
by soundcheck
soundcheck's
yep. And one effect of server-side upsampling is a dramatic increase of
bandwidth requirements. Going from 44.1/16 to 192/24 means you increase
the bandwidth required by a factor of 8!
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SBGK wrote:
what are these wasted bits?
The ones that need to be unnecessarily transmitted and received (and
possibly encrypted and decrypted) on the network. FLAC usually achieves
a compression factor close to 2:1.
handling those data packets require CPU power.
I've always found WAVs sound
pippin wrote:
yep. And one effect of server-side upsampling is a dramatic increase of
bandwidth requirements. Going from 44.1/16 to 192/24 means you increase
the bandwidth required by almost a factor of 7!
OK fair enough.
For uncompressed PCM:
16/44.1 = 1,411 Kbps
24/192 = 9,216 Kbps
Your
Don't want to distract this even more, but you are correct, I caught an
additional factor of two for PCM but your video streams are MPEG rates,
I was talking about H.264 which runs at around half the rate.
So it's about the same as a 1080p stream, the ones I looked at here were
all in the 10
I started this because I was using the upsampling built into
Squeezelite(libsoxr), not using SoX in LMS so I can't offer any clues as
to how to get that to work properly. The Squeezelite resampling option
is not the same as in SoX the program arguments, although the underlying
code is the same.
JohnSwenson wrote:
BTW the load on the Wandboard processor is about 8% when using this
setting. When using the default 20 bit setting it is about 4% and when
not doing any upsampling its about 2%.
Cool. Thanks for the info. Do those percentages include FLAC decoding,
or are you feeding
edwardian wrote:
Cool. Thanks for the info. Do those percentages include FLAC decoding,
or are you feeding it PCM?
I'm usually sending flac over the network these days since I don't have
a new enough server to handle 176 and 192 pcm. The above numbers were
for sending flac at 44.1 over
soundcheck wrote:
I tried resampling with SOX and other (reference) tools as discussed at
Audio Asylum and elsewhere several times in the past.
http://soundcheck-audio.blogspot.de/2011/04/tt-resampling.html
I never managed to get it working to my satisfaction. Neither realtime,
edwardian wrote:
I didn't like the idea that it only output FLAC (as all my music is WAV)
Any specific reasons to prefer WAV? I think the general view is that
with lower-end processors (such as those in the squeezeboxes) that don't
have dedicated I/O processors the additional network load
edwardian wrote:
Klaus, I read your document, and unless I missed something, you were
resampling from 16/44.1 to 24/96? Is that correct? Did you ever try
going from 16/44.1 to 24/88.2 or 24/176.4? If so, did you hear any
difference (compared to 24/96)?
And I also tried SOX upsampling in
Julf wrote:
Any specific reasons to prefer WAV? I think the general view is that
with lower-end processors (such as those in the squeezeboxes) that don't
have dedicated I/O processors the additional network load caused by the
wasted bits in WAV files causes more CPU load (and thus
I can confirm that WAV actually causes a LOT more load than FLAC
decoding, at least on an iPhone.
Of course, an iPhone will always uses WiFi but then I believe there have
been similar measurements on the SB Touch as well.
Especially is we talk about upsampled material, we are talking about a
LOT
edwardian wrote:
For me, there's no hassle tagging WAV files. I use dBpoweramp to tag the
WAV files and LMS/Squeezebox Touch has no problem reading them.
Problem is that that might or might not work if you ever switch to some
other software.
To try to judge the real from the false will
Julf wrote:
Problem is that that might or might not work if you ever switch to some
other software.
Understood. Thanks for the information. I've been organizing my music
collection over the past few months and my intention was to have a copy
as WAV and a copy as FLAC to take advantage of
pippin wrote:
Intuitively I still feel just duplicating the samples as normal
oversampling does it should give the best results
This is not how oversampling operates in a DAC (or ADC) - which would
basically achieve nothing. Oversampling operates exactly as described in
the NI link you gave,
Hi Pippin,
to your underlying question: why do the filtering externally rather than
in the DAC chip? The answer is I don't know. When I bypass the
internal filter and do it externally using a basic simple filter it
sounds much better. And it's not just me. I've done this in blind
fashion
Hi there/John.
A lot of writing and reading.
I tried resampling with SOX and other (reference) tools as discussed at
Audio Asylum and elsewhere several times in the past.
http://soundcheck-audio.blogspot.de/2011/04/tt-resampling.html
I never managed to get it working to my satisfaction.
Ah. OK. So how does this interpolation filter work? In the NI link they
show a filter curve that nicely follows the sine wave which will not be
so simple for music.
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learn more about iPeng, the iPhone and iPad remote for the Squeezebox
and
*New: Logitech UE Smart Radio* as well as iPeng
pippin wrote:
Ah. OK. So how does this interpolation filter work? In the NI link they
show a filter curve that nicely follows the sine wave which will not be
so simple for music.
The new samples are inserted and given zero-values. The result is then
low-pass filtered.
Remember, we have a
pippin wrote:
Um. Sorry. Again, I fail to see it.
Again: Why does interpolation filtering on the digital side improve the
analog filtering and how is the interpolation even correlated to your
desired filter response.
I can find a lot of prosa about this on the internet, mainly by makers
I didn't mean to question the sense behind upsampling. I fully
understand why that makes the (analog) filter design easier and the
result better.
But John's argument was that you get a better result by using a NOS DAC
and do the oversampling through sox because the interpolation filter in
sox is
John, I agree with Pippin that it's best to avoid the term stair steps
for digital audio as this is misleading. Samples are valid only for
moments in time.
Darren
Check it, add to it! http://www.dr.loudness-war.info/
darrenyeats wrote:
John, I agree with Pippin that it's best to avoid the term stair steps
for digital audio as this is misleading. Each sample pertains to an
instant ... between those instants is an analogue waveform that we are
sampling or attempting to re-create.
Darren
But what are the
flimflam wrote:
But what are the steps undertaken in re-creating the waveform? Clue:
it's not called a smoothing filter for nothing!
Yea. But that was my point. Unless you know exactly what your waveform
is you are just as likely to AMPLIFY your noise than to SMOOTH your
signal. To really be
flimflam wrote:
But what are the steps undertaken in re-creating the waveform? Clue:
it's not called a smoothing filter for nothing!
The whole point of the sampling theorem is that the samples define ALL
of a complete and continuous analogue waveform of any complexity, as
long as the original
pippin wrote:
Yea. But that was my point. Unless you know exactly what your waveform
is you are just as likely to AMPLIFY your noise than to SMOOTH your
signal. To really be able to smooth you'd have to analyse the whole
signal spectrum and you'd have to do that over some time - ideally the
OK, but that's not smoothing, that's just low-pass filtering, which
ideally removes ALL effects of the stair-steps.
However, here we are not talking about analog processing BEHIND the DAC,
we talk about digital processing BEFORE the DAC.
And here smoothing is not simple and any kind of
pippin wrote:
OK, but that's not smoothing, that's just low-pass filtering
Same thing! Smoothing filter is a specific and recognised term for this
filter - whether you like it or not :-) ! The term is used by many,
Analog Devices refer to it by this name a lot. That the stair steps are
smoothed
flimflam wrote:
Same thing! Smoothing filter is a specific and recognised term for this
filter - whether you like it or not :-) ! The term is used by many,
Analog Devices refer to it by this name a lot. That the stair steps are
smoothed seems obvious, the term does not mean to imply anything
pippin wrote:
So what you are saying is that if you use these kind of DACs with real
(that is: unfiltered) HD material like actual HD recordings you will
have all the aliasing down there in the audible band?
This means you have to actually run real HD material through a filter
to get a
JohnSwenson wrote:
Let me see if I understand your first question. As long as the highest
frequency of the audio data is small relative to the sample rate (say
40KHz maximum signal frequency for a 192 sample rate) you do not need
much if any filtering. There are going to be very little
Yeah sometiomes the aliasing doesn´t matter if it fits the world of the
supporter, the next one will tell you it is evil. The same goes for
ringing, some will hear all kind of problems with ringing at even 192khz
others won´t. Advocats of dsd will never hear the problems caused by
shaping noise
pippin wrote:
But then you've got all these high frequencies still in there on the
analog side. Isn't the biggest problem with the high frequencies that
power amps (especially ones filtering through intermediate frequencies
like Class D amps) tend to create artifacts if you have signal in
Please excuse my ignorance on DAC design and related digital filtering.
While I do believe I understand the signal theory part and the way the
DACs fundamentally work I don't know a lot about how digital filters in
these things are actually implemented.
JohnSwenson wrote:
The amplitude of the
JohnSwenson wrote:
Before getting into details I want to talk about sample rate. The
filters in many DAC chips get simpler the higher up the sample rate is.
For example the chip I'm using in the CSP player has a very simple
filter at 176.4/192 and NO filter at 352.8/384.
So what you
Hi John.
Thank you very much for your detailed information.
We're 3 persons here in Denmark, who now are using upsampling
with the squeezelite and sox library.
Very are all having different audio-gd DACS, and even they all have
a limitation on max 96Khz, I think we can all conclude that it
Wombat wrote:
Many 44.1 material once was at a higher samplerate and there was already
once a choice what filter to apply. How can you know that a chosen
filter doesn´t only do better because it fits more to the filter that
was applied before? I did read to much nonsense about the sound of
As promised here is post number two on external upsampling. This post is
going to focus on the upsampling capability recently included in
Squeezelite which is based on the SoX resampling code. These filters
can also be used offline with SoX, but the command line arguments for
SoX are very
Thanks for sharing John! I look forward to hearing your thoughts on
this. You've certainly got a lot of experience in this area so I'll
certainly be giving the suggestions a try listening (as you noted, this
isn't about bit-perfection but rather subjective experience).
Also, thanks for donating
In recent versions of CSOS Triode and JackOfAll have added upsampling
capabilities to Squeezelite (at my request). I wanted to start a thread
here for discussing sonic differences people hear between no upsampling
and different upsampling parameters.
I'm also going to give some history of my
Many 44.1 material once was at a higher samplerate and there was already
once a choice what filter to apply. How can you know that a chosen
filter doesn´t only do better because it fits more to the filter that
was applied before? I did read to much nonsense about magic filters over
the years.
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