[Eeyore]

RichD wrote:

On Sep 18, Eeyore wrote:
Do you have any experience designing audio amps?

LMAO ! Yes, Kevin had indeed had as have I.
Amps that sold commercially into the
pro-audio market.

Did you use MOSFET on the output stage, and why?

I have used Mosfets in some of my designs. They had very superior HF
characteristics to the readily available bipolars of 1980. Notably they
also don't suffer SOA limitation problems at higher voltages.

They also typically exhibit much lower crossover distortion when
suitably biased than bipolars do.

Graham

[Mr.T]

"Don Klipstein" wrote in message
...
I occaisionally hear artifacts in 16 bit 44.1 KHz, in music.
It is easy to make a test signal turn up severe artifacts with 44.1 KHz
sample - see what happens with a sinewave at a higher audio frequency that
is several Hz off a frequency that the sample frequency is a multiple of.

You should simply use a system that is not "broken" then.

Since I only occaisionally hear artifacts in music with 44.1 KHz 16 bit,
and when I do I usually find them minor, I would expect a sample rate
twice as high as that to be OK.

Of course it is, as is a *competently* designed 44.1 or 48kHz system.
Using either doesn't present much of a problem these days.

MrT.

[JosephKK[_2_]]

On Sun, 21 Sep 2008 13:41:43 -0500, Vladimir Vassilevsky
wrote:



Jan Panteltje wrote:

I think it should be possible [I could] design powered speakers with a WiFi interface.

How would you synchronize the different channels?


Some time stamping and segment overlapping and a restricted RF
environment, plus NTP as needed for clock synchronization. Total
latency and latency variation are the primary issues. Maybe modified
bluetooth is a better idea (it really does support streams).

Each speaker would have its own IP address, or perhaps its own port on one IP,
and from the [new] mixer only digital Ethernet to a wireless access point.
No bandwidth problem I think.
56 Mbits / second, should be enough for a few channels.

The real 802.11G throughput is 2.8MB/s at the best. An uncompressed
audio channel takes roughly 100KB/s.

A bit more than that, the normal time 44.1 kHz 16 bits/sample CDA runs
somewhere around 1.5 Mbits/s. 48 kHz 24 bit audio (semi-pro and pro
level) takes a skosh more, about 1.15 Mbits/s per channel.


You will have power cables to the speaker, but not a lot of audio wiring.
mmm maybe do the mikes too ;-)

The big problem with WiFi for audio is the synchronization between the
different WiFi units while maintaining the reasonable delay. This is
hard (if possible at all) to attain with the WiFi equipment.

AFAIK the solutions for audio via Ethernet (CobraNet and such) used the
special protocol stacks and were not fully compatible with the standard
networking stuff. In the general, Ethernet is not good as the network
for the multimedia; it was not designed for that purpose.


This is very true. If you want well time constrained transport, use
appropriate (usually telephone like or telephone) technology. If you
also want to ship video use SMPTE standards. The bitrates can be
daunting (SD-SDI {standard definition - serial digital interface
[both electrical and optical]} runs at 270 Mb/s) but the standards
guarantee interoperability.


Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com

[JosephKK[_2_]]

On Sun, 21 Sep 2008 21:59:52 GMT, Jan Panteltje
wrote:

On a sunny day (Sun, 21 Sep 2008 16:46:27 -0500) it happened Vladimir
Vassilevsky wrote in
:



Jan Panteltje wrote:
On a sunny day (Sun, 21 Sep 2008 13:41:43 -0500) it happened Vladimir
Vassilevsky wrote in
:



Jan Panteltje wrote:


I think it should be possible [I could] design powered speakers with a WiFi interface.

How would you synchronize the different channels?

Yes, good point, timestamp would be one way, but that does not solve the delay.
the delay would be fatal in a live application.

Here is the idea: using the power frequency as the common timing
reference. In the local WiFi network, the ping time would be at the
order of 1ms, so all channels could be PLLed to the same half period of
the AC power without an ambiguity. With the sufficient amount of
buffering, that should allow streaming multiple synchronized channels.
Sooo simple... I bet somebody already got a patent on that.


Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com

Clever!
How about this: we give each speaker a GPS.
It will also send back its position, and the 'mixer' will
then calculate the optimum sound pattern for 5.1.
GPS also has a very precise clock.

Yes the satellites do have cesium clocks and rubidium clocks, the
system would not work without them. Moreover you do not get good time
solutions without serious long term reception. Your little handheld
does not have such a clock and requires about 45 minutes power on time
to synch up that well.

[JosephKK[_2_]]

On Tue, 23 Sep 2008 01:21:35 +0100, Eeyore
wrote:



krw wrote:

says...
krw wrote:

You're going to have to justify that number. 100MB/s?

He said kBps.

Don't know what I was looking at. ...ahead at WiFi??

I can see 20KB, *maybe*.

44.1k samples / sec x 2 bytes each = 88.2kBps. Allow overhead for collisions etc. Actually if
you transmit the full AES/EBU SPDIF data it's THREE bytes.

100kBps sounds about right for one mono channel. Or 150 kBps with flags etc.

Ok, now put three or four of these together and it's still easily
manageable over WiFi.

With Quality of Service and no latency ?

Graham

With total throughput over 15 times what a single unidirectional
channel requires. 4 or 5 channels (per wifi channel, there are 11 in
802.11G) ought to be achievable on a good session, two channels most
of the time; and with adequate latency control.

Just the same i do not think packet orientated protocols deserve to be
used for decent streaming media.

[Jan Panteltje]

On a sunny day (Sun, 21 Sep 2008 13:41:43 -0500) it happened Vladimir
Vassilevsky wrote in
:

Anyways, I repeated the test the other way around, now for a mp3 file:
-rw-r--r-- 1 user user 103760023 2008-09-09 19:03 instrumental.mp3

Reran the test, it seems a card in my network was configured for base
10 ethernet?, system reported 'network congestion'.
So with base 100, or at least the correct configuration, the same test:

On receiving side:
netcat -q 0 -l -p 1234 -u q3.mp3

On transmit side:
-rw-r--r-- 1 user user 103760023 2008-09-09 19:03 instrumental.mp3
/home/user date;cat instrumental.mp3 | netcat -u -q 0 10.0.0.150 1234;date

Wed Sep 24 13:03:32 CEST 2008
Wed Sep 24 13:03:57 CEST 2008

makes 25 sec, 4.150 MB/s = 33.2 Mbps.

-rw-r--r-- 1 root root 103760023 2008-09-24 13:00 instrumental.mp3
-rw-r--r-- 1 root root 103760023 2008-09-24 12:58 q3.mp3
grml: # diff instrumental.mp3 q3.mp3
grml: #
Zero errors at 33.2Mbps, signal strength was 90%, access point Linksys,
channel 7.


Of course UDP based protocols exist, like rtp, I can broadcast rtp
with the dvbstream program, from digital satellite or terrestrial,
It needs an entry in the routing table:
224.0.0.0 * 240.0.0.0 U 0 0 0 eth0

dvbstream -f 12188 -p h -s 27500 -v 166 -a 128
will broadcast RTL2 from sat, while
mplayer -ao oss:/dev/dsp1 -cache 2048 rtp://224.0.1.2:5004/
will play the rtp stream again.
Mixed results via wireless, while TCP over wireless from the same stream works
fine.. no idea why.
There also exists a utility 'dumprtp', it is part of dvbstream.
All this learns us that we can also design our own UDP based protocol,
while sending data you could perhaps run a CRC test over many packets,
and so once in a hundred packets ask for a retransmit... whatever.
I am sure with some careful config that 33.2 Mbps or there about can be used.

Just for fun, for just audio, 8 channels uncompressed sampled at 96 k, and
24 bit wide, makes:
8 x 96000 x 3 x 8 = 18 432 000 is only 18 Mbps :-)

AC3 however, at a bitrate of say 500kbps has 5.1, we can carry 66
of those 5.1 channels.

33Mbps is about the size of a satellite transponder, and can carry
several TV programs, plus audio, plus teletext (ceefax), plus other services
like radio, and subtitles.

So, anyways, wireless is great for that sort of multimedia stuff.
Only limit I see is if you are in a crowded area where everybody uses
it, interference would slow the network way down, and UDP probably could
not be used at all.

I myself will stay with TCP as it, with 2.8MB/s is fast enough for audio and video,
and guarantees zero errors.

The average bandwidth of a sat TV channel here is 4500 to 6000 (wide screen) kbps,
you could get 2 in 22.4 Mbps.

[Arny Krueger]

"Don Klipstein" wrote in message


I occasionally hear artifacts in 16 bit 44.1 KHz, in
music.

Given the false claim that you've posted below, I somehow find that easy to
believe.

It's a common problem among vinylphiles and other digiphobes. They believe
some totally false, but possibly intuitively satisfying (to them) urban
myths about digital, and since they believe them, they hear them. One more
reason why only carefully bias-controlled listening tests can be trusted.


It is easy to make a test signal turn up severe
artifacts with 44.1 KHz sample - see what happens with a
sinewave at a higher audio frequency that is several Hz
off a frequency that the sample frequency is a multiple of.

Absolutely false.

A close relative is the mistaken idea that phase differences that don't
correspond to the sample rate can't be accurately reproduced.

[Eeyore]

JosephKK wrote:

Vladimir Vassilevsky wrote:
Jan Panteltje wrote:

I think it should be possible [I could] design powered speakers with a WiFi interface.

How would you synchronize the different channels?

Some time stamping and segment overlapping and a restricted RF
environment, plus NTP as needed for clock synchronization. Total
latency and latency variation are the primary issues. Maybe modified
bluetooth is a better idea (it really does support streams).

Barely enough reliable bandwidth IIRC.


Each speaker would have its own IP address, or perhaps its own port on one IP,
and from the [new] mixer only digital Ethernet to a wireless access point.
No bandwidth problem I think.
56 Mbits / second, should be enough for a few channels.

The real 802.11G throughput is 2.8MB/s at the best. An uncompressed
audio channel takes roughly 100KB/s.

A bit more than that, the normal time 44.1 kHz 16 bits/sample CDA runs
somewhere around 1.5 Mbits/s. 48 kHz 24 bit audio (semi-pro and pro
level) takes a skosh more, about 1.15 Mbits/s per channel.


You will have power cables to the speaker, but not a lot of audio wiring.
mmm maybe do the mikes too ;-)

The big problem with WiFi for audio is the synchronization between the
different WiFi units while maintaining the reasonable delay. This is
hard (if possible at all) to attain with the WiFi equipment.

AFAIK the solutions for audio via Ethernet (CobraNet and such) used the
special protocol stacks and were not fully compatible with the standard
networking stuff. In the general, Ethernet is not good as the network
for the multimedia; it was not designed for that purpose.

This is very true. If you want well time constrained transport, use
appropriate (usually telephone like or telephone) technology. If you
also want to ship video use SMPTE standards. The bitrates can be
daunting (SD-SDI {standard definition - serial digital interface
[both electrical and optical]} runs at 270 Mb/s) but the standards
guarantee interoperability.

I spoke to CobraNet some time in the past and they have a time synchronisation element they
call 'the conductor'. I gather it does not integrate well with networkds already carrying
modest traffic and a seperate cable run is advised in such cases IIRC.

Graham.

[Eeyore]

Arny Krueger wrote:

"Don Klipstein" wrote

I occasionally hear artifacts in 16 bit 44.1 KHz, in
music.

Given the false claim that you've posted below, I somehow find that easy to
believe.

It's a common problem among vinylphiles and other digiphobes. They believe
some totally false, but possibly intuitively satisfying (to them) urban
myths about digital, and since they believe them, they hear them. One more
reason why only carefully bias-controlled listening tests can be trusted.

So why do top-end studio use 24 bit 192 kHz like this from my old friends and
colleagues at Prism Sound ? One of the best companies I ever worked for btw.
http://prismsound.com/music_recordin...da8xr_home.php


Graham

[Arny Krueger]

"Eeyore" wrote in
message
krw wrote:

says...

Your phrase "good enough for audio" does not inspire
confidence. I work in the professional / production
area of audio. 'Good enough' usually isn't for us.
Indeed, for a host of reasons I'd also
probably want to transmit 24 bit audio. Some customers
might want 96 kHz sampling too.

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing urban myths like the
crossover distortion that you claim exists in some power amplifiers. Trouble
is that on the test bench, that low level distortion is not to be found.
I've documented it for you in some cases, but AFAIK, no sale.

If you can't hear it, it's good enough.

Some people's ears are sharper than others.

Mostly the differences are in the brain, not the ears. The first problem
I've pointed out, and that is that many people tend to hear what they
believe is there, whether its there or not. Actually, just about everybody
suffers with the error-creating effects of bias, which is why carefully
bias-controlled listening tests are so important.

The other problem is that listener training is very important. Most people
will hear that something is wrong if its wrong enough. In order to get down
to the actual thresholds of detection, most need some coaching. If you want
to run fast you have to train a lot, which amounts to running slower than
record times and building up your strength. If you want to hear the smallest
amounts of distortion that are audible, it is usually very helpful to listen
to that distortion at decreasing levels, starting out pretty high.

I expect the phone would be good enough for you ?

Depends on the phone. If I make a phone up out of a good vocal mic and some
studio monitors, it will be pretty good. These days really good electret
mics cost pennies, while good earphone elements are relatively small and
cheap compared to speakers. Most of the inherent losses in modern phones are
in the communications channel, which is wildly bandwidth-reduced. As
bandwidth becomes cheaper, there is a possibility that good-sounding
telephones will become commonplace.

Why? I could care less about "professional"
audiophoolery. BTW, we
were talking about powered speakers, not "professional"
grade audio. 96kHz? What nonsense.

Many top studios are now mastering at 24 bit 192 kHz.

So what? Many are not.

The enabling technology is audio interfaces that run at 24/192 and cost no
more than the older ones that ran at 24/44. It's all just numbers for the
sake of numbers.

Note that Yamaha used to only make digital consoles that ran up to 24/96.
Most of their production is now consoles that only clock up to 48 KHz. This
is actually a little strange because converters and DSPs that are capable of
running faster are cheaper than ever. I think someone figured it out - the
higher sampling did nothing with practical significance.


Using products like this, reckoned to be the best in the
world and manufactured by another company I used to work
for.
http://prismsound.com/music_recordin...da8xr_home.php

Vast overkill. But, if you're spending other people's money, and you think
you can use the value of your equipment to justify a higher pay rate for
yourself, why not?

[[email protected]]

On Sep 23, 7:00 pm, (Don Klipstein) wrote:
I occaisionally hear artifacts in 16 bit 44.1 KHz, in music.

It is easy to make a test signal turn up severe artifacts
with 44.1 KHz sample - see what happens with a
sinewave at a higher audio frequency that is several
Hz off a frequency that the sample frequency is a
multiple of.

First, would you care to restructure that into a comprehensible
sentence.

Second, what exactly do you mean by "higher audio
frequency?" I have several consumer-grade, semi-pro
and professional A/D - D/A system here and not a single
one of them show any artifacts that you allude to at any
frequency below 1/2 the sample rate, and that includes
44.1 kHz sample rate systems. I have a couple that do
show problems with signals ABOVE 1/2 the sample rate,
but they are either badly implemented or broken.

Since I only occaisionally hear artifacts in music
with 44.1 KHz 16 bit, and when I do I usually find
them minor, I would expect a sample rate twice as
high as that to be OK.

This is why proper listening and engineering tests are
done with methods that attempt to minimize the effects
of expectation.

[Arny Krueger]

"Eeyore" wrote in
message
Arny Krueger wrote:

"Don Klipstein" wrote

I occasionally hear artifacts in 16 bit 44.1 KHz, in
music.

Given the false claim that you've posted below, I
somehow find that easy to believe.

It is easy to make a test signal turn up severe
artifacts with 44.1 KHz sample - see what happens with a
sinewave at a higher audio frequency that is several Hz
off a frequency that the sample frequency is a multiple of.

Absolutely false.

Note that Graham is uncertain about the falseness of the above urban myth -
he has no comment.

And let me clarify my short comment. I've done my homework and tested
digital audio many ways. I've purposefully avoided using test signals that
are at frequencies that are related to the sample rate, and also used test
signals that were precisely clocked to the sample rate., or very near to it.

The effects claimed above simply don't exist. They might exist to a very
limited degree in undithered systems, but no proper digital audio system is
undithered.

It's a common problem among vinylphiles and other
digiphobes. They believe some totally false, but
possibly intuitively satisfying (to them) urban myths
about digital, and since they believe them, they hear
them. One more reason why only carefully bias-controlled
listening tests can be trusted.

So why do top-end studio use 24 bit 192 kHz like this
from my old friends and colleagues at Prism Sound ?

Same reason why people climb Mount Everest - it is there.

One
of the best companies I ever worked for btw.
http://prismsound.com/music_recordin...da8xr_home.php

One can admire the precision and care of their engineering, regardless of
the lack of practical need.

Note that the ADA8xr converters have only 112 dB dynamic range, which is
equaled by competitive products costing only a fraction of the price. They
are significantly (numerically) surpassed by recent chips from TI that cost
about $10 each.

Sic Transit Gloria

[Phil Allison]

"Eeysore"
Arny Krueger wrote:

It's a common problem among vinylphiles and other digiphobes. They
believe
some totally false, but possibly intuitively satisfying (to them) urban
myths about digital, and since they believe them, they hear them. One
more
reason why only carefully bias-controlled listening tests can be trusted.

So why do top-end studio use 24 bit 192 kHz like this from my old friends
and
colleagues at Prism Sound ?


** Ignoratio elenchi .....

Yaawnnnnnnnn.....

http://en.wikipedia.org/wiki/Ignoratio_elenchi



...... Phil

[Eeyore]

Arny Krueger wrote:

"Eeyore" wrote
krw wrote:
says...

Your phrase "good enough for audio" does not inspire
confidence. I work in the professional / production
area of audio. 'Good enough' usually isn't for us.
Indeed, for a host of reasons I'd also
probably want to transmit 24 bit audio. Some customers
might want 96 kHz sampling too.

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing urban myths like the
crossover distortion that you claim exists in some power amplifiers.

BECAUSE IT'S SO AUDIBLE IT STICKS OUT LIKE A SORE THUMB.

Your precious ABX testing guarantees only a 'lowest common denominator' result.

Anyone that can't hear the distortion of QSC USA or MX series must have severely
damaged hearing.

You were in the forces weren't you ? Explains it all. Hearing damage.

Graham

[Eeyore]

Phil Allison wrote:

"Eeysore"
Arny Krueger wrote:

It's a common problem among vinylphiles and other digiphobes. They
believe some totally false, but possibly intuitively satisfying (to them)
urban
myths about digital, and since they believe them, they hear them. One
more reason why only carefully bias-controlled listening tests can be
trusted.

So why do top-end studio use 24 bit 192 kHz like this from my old friends
and colleagues at Prism Sound ?

** Ignoratio elenchi .....

Yaawnnnnnnnn.....

http://en.wikipedia.org/wiki/Ignoratio_elenchi

How do you KNOW that clock accuracy isn't a factor. I saw some DREADFUL clock
jitter on early digital kit and that's just as bad as amplitude inaccuracy.

Graham

[Phil Allison]

"Eeysore"
Arny Krueger wrote:

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing urban myths like the
crossover distortion that you claim exists in some power amplifiers.

BECAUSE IT'S SO AUDIBLE IT STICKS OUT LIKE A SORE THUMB.


** ********.


Your precious ABX testing guarantees only a 'lowest common denominator'
result.


** Gobbledegook plus a massive non-sequitur.


Anyone that can't hear the distortion of QSC USA or MX series must have
severely
damaged hearing.


** Shame how the rest of us do not have any of the defective examples YOU
claim YOU came across that lacked forward bias current in the output
devices.

Shame you were too lazy and dumb to give the bias trim pot a tweak.

******.

You were in the forces weren't you ? Explains it all. Hearing damage.


** So what explains YOUR obvious brain damage then ?

Was it too much LSD or is it simply congenital ASD ???




..... Phil

[Phil Allison]

"Eeysore"
Arny Krueger wrote:

It's a common problem among vinylphiles and other digiphobes. They
believe some totally false, but possibly intuitively satisfying (to
them)
urban
myths about digital, and since they believe them, they hear them. One
more reason why only carefully bias-controlled listening tests can be
trusted.

So why do top-end studio use 24 bit 192 kHz like this from my old
friends
and colleagues at Prism Sound ?

** Ignoratio elenchi .....

Yaawnnnnnnnn.....

http://en.wikipedia.org/wiki/Ignoratio_elenchi

How do you KNOW that clock accuracy isn't a factor.



** More and worse ignoratio elenchi

GIANT yaawnnnnnnnn.....

http://en.wikipedia.org/wiki/Ignoratio_elenchi


Peeeeeuuuuukkeeeeee......



...... Phil

[Arny Krueger]

"Eeyore" wrote in
message
Arny Krueger wrote:

"Eeyore" wrote
krw wrote:
says...

Your phrase "good enough for audio" does not inspire
confidence. I work in the professional / production
area of audio. 'Good enough' usually isn't for us.
Indeed, for a host of reasons I'd also
probably want to transmit 24 bit audio. Some customers
might want 96 kHz sampling too.

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing
urban myths like the crossover distortion that you claim
exists in some power amplifiers.

BECAUSE IT'S SO AUDIBLE IT STICKS OUT LIKE A SORE THUMB.

Everyplace but on test equipment, and in careful listening tests.

Let the record show that I have a number of QSC amps on hand to test, have
done so, and reported the results to Graham on Usenet. He has no such
resources at hand.

Your precious ABX testing guarantees only a 'lowest
common denominator' result.

Horsefeathers.

ABX verifies or improves the known audible thresholds for the detection of
ALL KNOWN forms of distortion.

Anyone that can't hear the distortion of QSC USA or MX
series must have severely damaged hearing.

It just takes a lack of hysteria and making sure that the equipment is not
damaged.

You were in the forces weren't you ?

You engineer live sound in a music bar night after night?

Explains it all.

Speaks to your prejudice against the military Graham, and also your
inability to understand what I have reported again and again:

(1) When I report listening test results, I don't report on findings
obtained with only me as a listener.

(2) Test equipment doesn't lie - if it finds negligible low-level
distortion, no matter how bad the ops hearing is, the distortion isn't
there.

Hearing damage.

In your case Graham it is all about prejudice and hysteria. Of course you
think QSC amps suck - they are a highly competitive product.

[Rich Grise]

On Wed, 24 Sep 2008 08:33:17 -0400, Arny Krueger wrote:
"Eeyore" wrote in
I expect the phone would be good enough for you ?

Depends on the phone. If I make a phone up out of a good vocal mic and some
studio monitors, it will be pretty good. These days really good electret
mics cost pennies, while good earphone elements are relatively small and
cheap compared to speakers. Most of the inherent losses in modern phones are
in the communications channel, which is wildly bandwidth-reduced. As
bandwidth becomes cheaper, there is a possibility that good-sounding
telephones will become commonplace.


If NASA can send broadcast quality video down from the shuttle or ISS,
howcome their audio still sounds like a fast food clown?

Thanks,
Rich

[Eeyore]

Phil Allison wrote:

"Eeysore"
Arny Krueger wrote:

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing urban myths like the
crossover distortion that you claim exists in some power amplifiers.

BECAUSE IT'S SO AUDIBLE IT STICKS OUT LIKE A SORE THUMB.

** ********.

Deaf ****.


Your precious ABX testing guarantees only a 'lowest common denominator'
result.

** Gobbledegook plus a massive non-sequitur.

Read how it works.


Anyone that can't hear the distortion of QSC USA or MX series must have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples YOU
claim YOU came across that lacked forward bias current in the output
devices.

The classic QSC arrangement with grounded collectors has no qiescent current in
the output devices. They are biased 'virtually on the threshold' at about 0.5V
Vbe typically. This is actually quite clever and quite intentional and I've
used the same method myself. It avoid the huge gm jump at crossover by putting
the load current in that area through the DRIVERS alone which are highly
degenerated with about a 10 ohm emitter resistor to make gm look fairly
constant.

Fortunately there are experts here who will understand exactly what I mean by
the above and ignore your huffing and puffing. You're a tech. I'm a designer.

Don ? Kevin ?


Shame you were too lazy and dumb to give the bias trim pot a tweak.

You mean the WRONG tweak.


******.

Careful with your language when you're talking to your betters.


You were in the forces weren't you ? Explains it all. Hearing damage.

** So what explains YOUR obvious brain damage then ?

Was it too much LSD or is it simply congenital ASD ???

Clearly you're stupid as well as deaf.

Graham

[Vladimir Vassilevsky]

Rich Grise wrote:


If NASA can send broadcast quality video down from the shuttle or ISS,
howcome their audio still sounds like a fast food clown?

This is for the presence effect; otherwise you will think of ISS or
Shuttle as if it is something routine.

Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com

[Eeyore]

Arny Krueger wrote:

"Eeyore" wrote
Arny Krueger wrote:
"Eeyore" wrote
krw wrote:
says...

Your phrase "good enough for audio" does not inspire
confidence. I work in the professional / production
area of audio. 'Good enough' usually isn't for us.
Indeed, for a host of reasons I'd also
probably want to transmit 24 bit audio. Some customers
might want 96 kHz sampling too.

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing
urban myths like the crossover distortion that you claim
exists in some power amplifiers.

BECAUSE IT'S SO AUDIBLE IT STICKS OUT LIKE A SORE THUMB.

Everyplace but on test equipment, and in careful listening tests.

It stands out like a sore thumb on test equipment too. It's HIGHLY visible and
occurs typically at critical listening levels (around the 100mW - 1W area).


Let the record show that I have a number of QSC amps on hand to test, have
done so, and reported the results to Graham on Usenet. He has no such
resources at hand.

Uh ? I don't have access to it at the moment but my measurements have been
made on such amplifiers using AP test Equipment - the industry standard and
capable of incredible resolution. I may also shortly get my hands on a Prism
Sound dScope 3.
http://ap.com/products/index.html
I personally love the Portable One to bits since it's so intuitive and fast to
use and totally self contained. With System Ones you often had to turn off the
PC monitor to get accurate results.
http://www.prismsound.com/test_measu...scope_home.php


Your precious ABX testing guarantees only a 'lowest
common denominator' result.

Horsefeathers.

As I read how ABX testing works, if say, you had a group 30 listeners and 3
were consistently about to determine A from B and the other 27 couldn't, then
you would say the products were indistinguishable, i.e. discard the results of
the 3 that could determine a difference.

That's what I mean by 'lowest common denominator'. All it proves is that most
people have crap hearing.

What the test should do is affirm there IS a difference on the basis of the 3
who can hear a difference.

If I have misunderstood these principles, my apologies.

However I WILL NOT engage in futile discussion about things *I* know I *can*
hear. Some of my colleagues recently used me as the 'professional ears' to
track down a curious hum that a studio mix engineer had reported but they
couldn't hear themselves. I eventually found it. It was a slightly humming
wall wart in the back of an equipment rack. If it was much louder than 10
phons I'd be surprised.

Graham

[Eeyore]

Arny Krueger wrote:

"Eeyore" wrote

You were in the forces weren't you ?

You engineer live sound in a music bar night after night?

No. And when I did engineer regularly it was probably only 2-3 times a well
and I was very careful not to expose myself to excessive sound levels.
Besides, for a couple of hours, and certainly not every day, it's almost
irrelevant.

After the period when I had been mixing live a lot, I happened to have an ear
problem. I was tested in an acoustic booth and my hearing was as the nurse
said "the best I've ever seen". The results were bang on the 0 phon line.


Explains it all.

Speaks to your prejudice against the military Graham, and also your
inability to understand what I have reported again and again:

No prejudice whatever. My mother's side of the family contained many RAF
officers, My Grandfather being a Wing Commander and my best girlfriend ever's
father who was a superb guy was an ex RN Lieutenant Commander BUT the military
are reknowned for big *impulsive* bangs that are they most damaging type to
hearing.

Impulsive noise is vastly more damaging that continuous noise like music.
Hence all the sound level recommendations that were based on steel mills with
their anvils and the like are utterly irrelevent to music (except for the
drummer who typically goes deaf first if anyone is going to because it's
impulsive noise from the kit).

Graham

[Eeyore]

Arny Krueger wrote:

"Eeyore" wrote

(2) Test equipment doesn't lie - if it finds negligible low-level
distortion, no matter how bad the ops hearing is, the distortion isn't
there.

Too damn right it doesn't. It sees x'over distortion with a clarity clearly
unmatched by your hearing.

Have you noticed QSC have now abandoned the grounded collector arrangement in
newer and high-end models.

If it was so good why would they do that ?


Hearing damage.

In your case Graham it is all about prejudice and hysteria. Of course you
think QSC amps suck - they are a highly competitive product.

One of our OEMs offered us a QSC 'clone'. I managed to make it work better !
As in lower THD AND noise by a simple pcb re-layout and front-end tweak.

Don't forget I've met the guy who really designed the RMXs, Tim Lau. We
exchanged some very interesting ideas on amp design whilst I was in China
visiting them..

Graham

[Eeyore]

Rich Grise wrote:

On Wed, 24 Sep 2008 08:33:17 -0400, Arny Krueger wrote:
"Eeyore" wrote in
I expect the phone would be good enough for you ?

Depends on the phone. If I make a phone up out of a good vocal mic and some
studio monitors, it will be pretty good. These days really good electret
mics cost pennies, while good earphone elements are relatively small and
cheap compared to speakers. Most of the inherent losses in modern phones are
in the communications channel, which is wildly bandwidth-reduced. As
bandwidth becomes cheaper, there is a possibility that good-sounding
telephones will become commonplace.


If NASA can send broadcast quality video down from the shuttle or ISS,
howcome their audio still sounds like a fast food clown?

No-one cares.

Try listening in on VHF ATC messages to aircraft. It's a bloody disgrace. No
wonder they have accidents.

Graham

[Eeyore]

Vladimir Vassilevsky wrote:

Rich Grise wrote:

If NASA can send broadcast quality video down from the shuttle or ISS,
howcome their audio still sounds like a fast food clown?

This is for the presence effect; otherwise you will think of ISS or
Shuttle as if it is something routine.

That happened fast enough to the Apollo missions too !

Graham

[John Larkin]

On Wed, 24 Sep 2008 10:18:39 -0400, "Arny Krueger"
wrote:

"Eeyore" wrote in
message
Arny Krueger wrote:

"Eeyore" wrote
krw wrote:
says...

Your phrase "good enough for audio" does not inspire
confidence. I work in the professional / production
area of audio. 'Good enough' usually isn't for us.
Indeed, for a host of reasons I'd also
probably want to transmit 24 bit audio. Some customers
might want 96 kHz sampling too.

Yes, I know you're a "professional" audiophool.

Just a professional.

But one with a pretty fair track record for hearing
urban myths like the crossover distortion that you claim
exists in some power amplifiers.

BECAUSE IT'S SO AUDIBLE IT STICKS OUT LIKE A SORE THUMB.

Everyplace but on test equipment, and in careful listening tests.

Let the record show that I have a number of QSC amps on hand to test, have
done so, and reported the results to Graham on Usenet. He has no such
resources at hand.

Your precious ABX testing guarantees only a 'lowest
common denominator' result.

Horsefeathers.

ABX verifies or improves the known audible thresholds for the detection of
ALL KNOWN forms of distortion.

Anyone that can't hear the distortion of QSC USA or MX
series must have severely damaged hearing.

It just takes a lack of hysteria and making sure that the equipment is not
damaged.

You were in the forces weren't you ?

You engineer live sound in a music bar night after night?

Explains it all.

Speaks to your prejudice against the military Graham, and also your
inability to understand what I have reported again and again:

(1) When I report listening test results, I don't report on findings
obtained with only me as a listener.

(2) Test equipment doesn't lie - if it finds negligible low-level
distortion, no matter how bad the ops hearing is, the distortion isn't
there.

Hearing damage.

In your case Graham it is all about prejudice and hysteria. Of course you
think QSC amps suck - they are a highly competitive product.



What is it about audio that makes people so angry and profane? I
thought music was supposed to make people happy. Peace Train, All you
Need is Love, stuff like that.

What happened to "musick hath powers to calme ye savage beaste?" It
sure doesn't seem to be making you guys very calm.

Is it because most of you guys can't afford decent test equipment, so
everything becomes subjective?

John

[Eeyore]

John Larkin wrote:

What is it about audio that makes people so angry and profane? I
thought music was supposed to make people happy. Peace Train, All you
Need is Love, stuff like that.

God only knows.

If I can find a cute new way of putting an extra zero behind the decimal point
in a THD figure why should I lambasted for it ?

And as for hearing the difference between amps of 0.1% specified THD rating
and 0.01%, I discoverd about 30 years ago that it was like chalk and cheese.

Once you get to 0.00003% like some of Nat Semi's op-amps the argument is
hypothetical anyway. The ear would be incapable of resolving such a low figure
so why not use them anyway ? What harm could they do ?

Arny seems to be almost as bad as the tube afficionados who LIKE lots of
distortion.

Graham

[Kris Krieger]

Rich Grise wrote in
news
On Wed, 24 Sep 2008 08:33:17 -0400, Arny Krueger wrote:
"Eeyore" wrote in
I expect the phone would be good enough for you ?

Depends on the phone. If I make a phone up out of a good vocal mic and
some studio monitors, it will be pretty good. These days really good
electret mics cost pennies, while good earphone elements are relatively
small and cheap compared to speakers. Most of the inherent losses in
modern phones are in the communications channel, which is wildly
bandwidth-reduced. As bandwidth becomes cheaper, there is a possibility
that good-sounding telephones will become commonplace.


If NASA can send broadcast quality video down from the shuttle or ISS,
howcome their audio still sounds like a fast food clown?

Thanks,
Rich


I wonder, tho', whether their secure/encrypted comms are as bad. I'd suspect
not - I'd imagine that a secure channel woudl be able to carry mroe data, and
therefore be "cleaner". But that's just a guess...

[John Larkin]

On Wed, 24 Sep 2008 21:13:56 +0100, Eeyore
wrote:



John Larkin wrote:

What is it about audio that makes people so angry and profane? I
thought music was supposed to make people happy. Peace Train, All you
Need is Love, stuff like that.

God only knows.

If I can find a cute new way of putting an extra zero behind the decimal point
in a THD figure why should I lambasted for it ?

And as for hearing the difference between amps of 0.1% specified THD rating
and 0.01%, I discoverd about 30 years ago that it was like chalk and cheese.

Once you get to 0.00003% like some of Nat Semi's op-amps the argument is
hypothetical anyway. The ear would be incapable of resolving such a low figure
so why not use them anyway ? What harm could they do ?

Arny seems to be almost as bad as the tube afficionados who LIKE lots of
distortion.

Graham

Soft distortion ain't bad, and lots of people like it. Crossover
distortion is ghastly, but tubes sort of inherently don't do that when
biased in the usual (ie, ancient traditional) ways.

The crossover thing is a non-trivial and often subtle issue with
anything but Class A semiconductor amps. There's some *really bad*
biasing going on out there.

But all that can be measured.

I've been hanging out with a lot of artists lately, and some of them
tend to be snooty based on their "vision" that dumb mortals just don't
have. Similar delusions... and similar economics.

John

[Phil Allison]

"Eeysore the Lunatic "


** Gobbledegook plus a massive non-sequitur.

Read how it works.


** Insane false logic.



Anyone that can't hear the distortion of QSC USA or MX series must have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples
YOU
claim YOU came across that lacked forward bias current in the output
devices.

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.



** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.




....... Phil

[Eeyore]

Phil Allison wrote:

"Eeysore the Lunatic "

** Gobbledegook plus a massive non-sequitur.

Read how it works.

** Insane false logic.

Anyone that can't hear the distortion of QSC USA or MX series must have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples
YOU claim YOU came across that lacked forward bias current in the output
devices.

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.

You have it 100% incorrect.

But then for a toaster repair man it's not too surprising.

If Kevin or Don Pearce were to re-enter the thread, they'd know EXACTLY what I'm
driving at. Actually, Larkin probably does too but maybe he's just trying to
stir it up ?

Graham

[Eeyore]

Phil Allison wrote:

"Eeysore the Lunatic "

** Gobbledegook plus a massive non-sequitur.

Read how it works.

** Insane false logic.

Anyone that can't hear the distortion of QSC USA or MX series must have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples
YOU
claim YOU came across that lacked forward bias current in the output
devices.

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.

And you know nothing about amplifier design.

Graham

[Phil Allison]

"Eeysore the Lunatic "

Anyone that can't hear the distortion of QSC USA or MX series must
have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples
YOU claim YOU came across that lacked forward bias current in the
output
devices.

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.

You have it 100% incorrect.


** Five QSC MX series amps I have here RIGHT NOW prove you WRONG.

All have about 20 mA per output BJT at idle - more when hot.

You have no proof whatever of you MAD & WRONG assertion.




...... Phil

[Phil Allison]

"Eeysore the Lunatic "


The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.


** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.




....... Phil

[Eeyore]

Phil Allison wrote:

"Eeysore the Lunatic "

Anyone that can't hear the distortion of QSC USA or MX series must
have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples
YOU claim YOU came across that lacked forward bias current in the
output
devices.

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.

You have it 100% incorrect.

** Five QSC MX series amps I have here RIGHT NOW prove you WRONG.

All have about 20 mA per output BJT at idle - more when hot.

I suppose you set them that way.


You have no proof whatever of you MAD & WRONG assertion.

Well lucky you. You simply don't understand electronic design and never will.

How many tens or hundreds of thousands of products of YOUR design have you sold
?

Graham

[Eeyore]

Phil Allison wrote:

"Eeysore the Lunatic "


The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

FACT. Straight out of the factory.

Mind you, their Q/C might be sloppy enough to have misled you.

Graham

[Eeyore]

Phil Allison wrote:

"Eeysore the Lunatic "

Anyone that can't hear the distortion of QSC USA or MX series must
have
severely damaged hearing.

** Shame how the rest of us do not have any of the defective examples
YOU claim YOU came across that lacked forward bias current in the
output
devices.

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.

You have it 100% incorrect.

** Five QSC MX series amps I have here RIGHT NOW prove you WRONG.

All have about 20 mA per output BJT at idle - more when hot.

You have no proof whatever of you MAD & WRONG assertion.

BORING

[Eeyore]

Phil Allison wrote:

"Eeysore the Lunatic "

The classic QSC arrangement with grounded collectors has no qiescent
current in the output devices.

** Massive BLATANT LIE !!

You are ridiculous, stupid damn LIAR Stevenson.

That's what you always say when you're caught out.

Graham

[Eeyore]

Eeyore wrote:

How many tens or hundreds of thousands of products of YOUR design have you sold ?

Many hundreds of thousands and I still get congratulatory emails as to how their 23
your old unit is still working fine.

The only trouble is, the paint tend to wear off.

Graham