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Help:bought Audiophile 2496;sounds just like my SB Live!

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August 16, 2004 2:55:33 AM

Archived from groups: rec.audio.pro (More info?)

I just bought the M-Audio Audiophile 2496, hoping to get better sound
than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
bit, 44.1 khz. In Audobe Audition, I try recording the exact same
thing (a short piano piece) on each card for a back-to-back
comparison, but both recordings sound the same. I have a basic
understanding of bit depth and sample rate, but I don't know how the
bit depth and sample rate of my sound card relates to the BD and SR
that I set my recording session to, and what occurs if I then convert
the BD and SR. So could it have something to do with those settings?
Please let me know, in plain English, when I will be able to hear the
difference between different BD's and SR's, and between these two
sound cards. So much rave about the Audiphile cards, and negativity
towards Sound Blaster, and I can't hear the difference!
Anonymous
August 16, 2004 8:11:16 AM

Archived from groups: rec.audio.pro (More info?)

"Bob" <fiddledd8@yahoo.com> wrote in message
news:b8e50c8f.0408152155.7efdf17f@posting.google.com

> I just bought the M-Audio Audiophile 2496, hoping to get better sound
> than my Sound Blaster Live. Audiophile is 24 bit, 96 kHz. SB is 16
> bit, 44.1 kHz.

Which SB Live! are you talking about? There are at least three major
permutations of this hardware. The first two had definite technical issues
that caused audible coloration but a more recent version called the SB
Live! 5.1 was improved to the point where the audible coloration can be
difficult or impossible to hear under a wider range of real-world
circumstances.

> In Adobe Audition, I try recording the exact same
> thing (a short piano piece) on each card for a back-to-back
> comparison, but both recordings sound the same.

IME this is potentially a valid procedure depending on the associated gear
used to record and play back the piano piece. Key issues are to use
recording and monitoring equipment of sufficient quality, and to time-synch
and level-match the samples being compared. Recordings being compared need
to be played back in quick succession for maximum listener sensitivity. For
maximum sensitivity, a number of different sound sources, and a number of
different instrumental sounds should be used. There should also be some bias
controls, as listener foreknowledge of which alternative is being played can
be distracting and lead to spurious results. A number of programs, mostly
freeware that assist with making close, sensitive, and bias-controlled
listening tests can be found at www.pcabx.com .

A set of recordings made with the second revision of the SB Live! can be
found at http://www.pcabx.com/product/ct4830/index.htm . A number of these
test musical samples are known to produce positive results for audible
differences because of the technical flaws previously mentioned. If you work
with them you will notice that different kinds of music make technical flaws
more or less audible.

One of the dirty little secrets about listening tests is that not all music
reveals all audible differences due to technical flaws. A listening test
with one piece of music can give positive results, and another listening
test with a different piece of music can give negative results, even when
associated recording and playback equipment are of the highest quality, the
listener is properly trained and sensitized to the difference, etc.

For example, consider the SBLive! 4830. The technical flaw that is probably
most audible with this audio interface relates to upper midrange coloration.
While upper midrange content is fairly common in music, various musical
sounds differ in the degree to which they make upper midrange colorations
audible. Musical sounds can have weak upper midrange content, or the
content that is there can be chaotic, and therefore tend to conceal
differences in this critical area. Or, stronger content in other musical
ranges can distract the listener.

> I have a basic understanding of bit depth and sample rate, but I don't
know how the
> bit depth and sample rate of my sound card relates to the BD and SR
> that I set my recording session to, and what occurs if I then convert
> the BD and SR.

Audition has three basic BDs, 8, 16, and 32 bits. The 32 bit format is
floating point and has immense dynamic range. In any case you can't make or
playback recordings with effective BDs that exceed those that the
particular card supports, and you can't exploit high BD recordings with an
audio interfaces that don't support those BDs. Audition tends to make these
differences less obvious from a UI standpoint, as it will automatically
downsample and upsample to mitigate mismatches between the audio interface
and the recorded format.

> So could it have something to do with those settings?

Audition will allow you to record and playback a 8 bit recording with a 32
bit audio interface, or record and playback a 32 bit recording with a 16 bit
interface for your convenience, but in the end the rule of the weakest link
will still limit the quality of your recordings and how they sound when you
play them back.

> Please let me know, in plain English, when I will be able to hear the
> difference between different BD's and SR's, and between these two
> sound cards.

It all depends on the various issues that I've raised, and probably some
other ones as well.

Furthermore, some of the benefits of better audio interfaces relate to
safety margins. A card with better dynamic range gives you more wiggle room
for practical suboptimalities. For example one card can have 80 dB dynamic
range and another one can have 120 dB dynamic range. Depending on other
parameters, headroom, quality of associated equipment, the room you record
and listen in, and the basic nature of the actual musical event being
recorded, the dynamic range of the recording might top out at 70 dB. If you
get all the levels optimized, both cards are capable of making a recording
with about 70 dB worth of dynamic range. However, you only have 10 dB worth
of margin with the 80 dB DR card, while you have a whopping 50 dB of margin
with the 120 dB dynamic range card. You are much more likely to exploit the
dynamic range of the basic musical event with the better card, but if you do
everything right, and shave all the margins optimally, the lesser card just
might deliver roughly comparable results.

> So much rave about the Audiophile cards, and negativity towards Sound
Blaster, and I can't hear the difference!

It all depends...
Anonymous
August 16, 2004 10:38:29 AM

Archived from groups: rec.audio.pro (More info?)

"Bob" <fiddledd8@yahoo.com> wrote in message
news:b8e50c8f.0408152155.7efdf17f@posting.google.com...
>I just bought the M-Audio Audiophile 2496, hoping to get better sound
> than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
> bit, 44.1 khz. In Audobe Audition, I try recording the exact same
> thing (a short piano piece) on each card for a back-to-back
> comparison, but both recordings sound the same.


Well, what's your recording chain? What's your monitoring chain? What is
bad about what you're currently hearing? It could be that whatever defects
you're hearing have nothing to do with the sound card.
Anonymous
August 16, 2004 2:06:40 PM

Archived from groups: rec.audio.pro (More info?)

In article <b8e50c8f.0408152155.7efdf17f@posting.google.com> fiddledd8@yahoo.com writes:

> I just bought the M-Audio Audiophile 2496, hoping to get better sound
> than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
> bit, 44.1 khz. In Audobe Audition, I try recording the exact same
> thing (a short piano piece) on each card for a back-to-back
> comparison, but both recordings sound the same.

That's not surprising. A new sound card won't improve your
microphones, preamps, piano, or room acoustics. You've strengthened
one link of your chain and there's nothing wrong with that, but if you
want to improve the sound of your recordings, you have to improve the
sound of what you're sending to the sound card.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 16, 2004 2:10:45 PM

Archived from groups: rec.audio.pro (More info?)

On 15 Aug 2004 22:55:33 -0700, fiddledd8@yahoo.com (Bob) wrote:

>I just bought the M-Audio Audiophile 2496, hoping to get better sound
>than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
>bit, 44.1 khz. In Audobe Audition, I try recording the exact same
>thing (a short piano piece) on each card for a back-to-back
>comparison, but both recordings sound the same. I have a basic
>understanding of bit depth and sample rate, but I don't know how the
>bit depth and sample rate of my sound card relates to the BD and SR
>that I set my recording session to, and what occurs if I then convert
>the BD and SR. So could it have something to do with those settings?
> Please let me know, in plain English, when I will be able to hear the
>difference between different BD's and SR's, and between these two
>sound cards. So much rave about the Audiphile cards, and negativity
>towards Sound Blaster, and I can't hear the difference!

The main advantage of the Audiophile is its low-latency ASIO drivers,
and the ability to run natively at sample rates other than 48KHz (this
affects synchronisation between tracks in a multi-track recording
environment).

The advantage of 24 bits is that it gives you a bigger window in which
to set your recording levels. You don't have to push levels right up
to the 0dB peg, risking overload.

I find 96KHz unnecessary. Others disagree :-) It's "on the chip",
and the marketing guys like it.

You won't notice much difference in basic sound quality. Even cheap
systems now come into the category "pretty good". The better card
measures better, this mat manifest itself in less listening fatigue
over a period.


CubaseFAQ www.laurencepayne.co.uk/CubaseFAQ.htm
"Possibly the world's least impressive web site": George Perfect
Anonymous
August 16, 2004 8:14:09 PM

Archived from groups: rec.audio.pro (More info?)

"Bob" <fiddledd8@yahoo.com> wrote in message
news:b8e50c8f.0408152155.7efdf17f@posting.google.com...
> I just bought the M-Audio Audiophile 2496, hoping to get better sound
> than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
> bit, 44.1 khz. In Audobe Audition, I try recording the exact same
> thing (a short piano piece) on each card for a back-to-back
> comparison, but both recordings sound the same. I have a basic
> understanding of bit depth and sample rate, but I don't know how the
> bit depth and sample rate of my sound card relates to the BD and SR
> that I set my recording session to, and what occurs if I then convert
> the BD and SR. So could it have something to do with those settings?
> Please let me know, in plain English, when I will be able to hear the
> difference between different BD's and SR's, and between these two
> sound cards. So much rave about the Audiphile cards, and negativity
> towards Sound Blaster, and I can't hear the difference!

Nobody said you would necessarily be able to hear the difference. You will
be able to measure it though. Get a copy of RMAA, which is freeware, and
test both cards. You will see that the AP2496 gives you much greater margin
for recording error.

TonyP.
August 16, 2004 10:35:35 PM

Archived from groups: rec.audio.pro (More info?)

"TonyP" <TonyP@optus.net.com.au> wrote in message news:<412050d3$0$1786$afc38c87@news.optusnet.com.au>...
> "Bob" <fiddledd8@yahoo.com> wrote in message
> news:b8e50c8f.0408152155.7efdf17f@posting.google.com...
> > I just bought the M-Audio Audiophile 2496, hoping to get better sound
> > than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
> > bit, 44.1 khz. In Audobe Audition, I try recording the exact same
> > thing (a short piano piece) on each card for a back-to-back
> > comparison, but both recordings sound the same. I have a basic
> > understanding of bit depth and sample rate, but I don't know how the
> > bit depth and sample rate of my sound card relates to the BD and SR
> > that I set my recording session to, and what occurs if I then convert
> > the BD and SR. So could it have something to do with those settings?
> > Please let me know, in plain English, when I will be able to hear the
> > difference between different BD's and SR's, and between these two
> > sound cards. So much rave about the Audiphile cards, and negativity
> > towards Sound Blaster, and I can't hear the difference!
>
> Nobody said you would necessarily be able to hear the difference. You will
> be able to measure it though. Get a copy of RMAA, which is freeware, and
> test both cards. You will see that the AP2496 gives you much greater margin
> for recording error.
>
> TonyP.

Thanks for the input. I just thought that better sample rate and bit
depth equals higher fidelity, which I thought would be audible. What
do you mean when you say "greater margin for recording error"? What
is recording error? Thanks.
August 16, 2004 11:45:01 PM

Archived from groups: rec.audio.pro (More info?)

"Arny Krueger" <arnyk@hotpop.com> wrote in message news:<3fadnVSF5a0K8b3cRVn-ug@comcast.com>...
> "Bob" <fiddledd8@yahoo.com> wrote in message
> news:b8e50c8f.0408152155.7efdf17f@posting.google.com
>
> > I just bought the M-Audio Audiophile 2496, hoping to get better sound
> > than my Sound Blaster Live. Audiophile is 24 bit, 96 kHz. SB is 16
> > bit, 44.1 kHz.
>

Thanks very much for your detailed reply. Many helpful suggestions.
I have a couple more questions.

Key issues are to use
> recording and monitoring equipment of sufficient quality, and to time-synch
> and level-match the samples being compared.

Don't understand what time-sync and level match mean.

Audition has three basic BDs, 8, 16, and 32 bits. The 32 bit format is
> floating point and has immense dynamic range. In any case you can't make or
> playback recordings with effective BDs that exceed those that the
> particular card supports, and you can't exploit high BD recordings with an
> audio interfaces that don't support those BDs. Audition tends to make these
> differences less obvious from a UI standpoint, as it will automatically
> downsample and upsample to mitigate mismatches between the audio interface
> and the recorded format.

Does this mean that if my card is advertised as having 24-bit 96khz
quality, I won't get any use out of the 32 bit setting in audition?
Or if I use my SB, which is a 16 bit/44.1khz card it won't do me any
good to set the sample rate to 96khz at the beggining of my session?
Also, what is floating point?

Furthermore, some of the benefits of better audio interfaces relate
to
> safety margins. A card with better dynamic range gives you more wiggle room
> for practical suboptimalities. For example one card can have 80 dB dynamic
> range and another one can have 120 dB dynamic range. Depending on other
> parameters, headroom, quality of associated equipment, the room you record
> and listen in, and the basic nature of the actual musical event being
> recorded, the dynamic range of the recording might top out at 70 dB. If you
> get all the levels optimized, both cards are capable of making a recording
> with about 70 dB worth of dynamic range. However, you only have 10 dB worth
> of margin with the 80 dB DR card, while you have a whopping 50 dB of margin
> with the 120 dB dynamic range card. You are much more likely to exploit the
> dynamic range of the basic musical event with the better card, but if you do
> everything right, and shave all the margins optimally, the lesser card just
> might deliver roughly comparable results.

Does this mean that if my card has a bigger dynamic range, my musician
can afford to play louder without causing clipping or distortion in
the recording?
August 17, 2004 12:02:28 AM

Archived from groups: rec.audio.pro (More info?)

"Arny Krueger" <arnyk@hotpop.com> wrote in message news:<3fadnVSF5a0K8b3cRVn-ug@comcast.com>...
> "Bob" <fiddledd8@yahoo.com> wrote in message
> news:b8e50c8f.0408152155.7efdf17f@posting.google.com
>
> > I just bought the M-Audio Audiophile 2496, hoping to get better sound
> > than my Sound Blaster Live. Audiophile is 24 bit, 96 kHz. SB is 16
> > bit, 44.1 kHz.
>

Thanks very much for your detailed reply. Many helpful suggestions.
I have a couple more questions.

Key issues are to use
> recording and monitoring equipment of sufficient quality, and to time-synch
> and level-match the samples being compared.

Don't understand what time-sync and level match mean.

Audition has three basic BDs, 8, 16, and 32 bits. The 32 bit format is
> floating point and has immense dynamic range. In any case you can't make or
> playback recordings with effective BDs that exceed those that the
> particular card supports, and you can't exploit high BD recordings with an
> audio interfaces that don't support those BDs. Audition tends to make these
> differences less obvious from a UI standpoint, as it will automatically
> downsample and upsample to mitigate mismatches between the audio interface
> and the recorded format.

Does this mean that if my card is advertised as having 24-bit 96khz
quality, I won't get any use out of the 32 bit setting in audition?
Or if I use my SB, which is a 16 bit/44.1khz card it won't do me any
good to set the sample rate to 96khz at the beggining of my session?
Also, what is floating point?

Furthermore, some of the benefits of better audio interfaces relate
to
> safety margins. A card with better dynamic range gives you more wiggle room
> for practical suboptimalities. For example one card can have 80 dB dynamic
> range and another one can have 120 dB dynamic range. Depending on other
> parameters, headroom, quality of associated equipment, the room you record
> and listen in, and the basic nature of the actual musical event being
> recorded, the dynamic range of the recording might top out at 70 dB. If you
> get all the levels optimized, both cards are capable of making a recording
> with about 70 dB worth of dynamic range. However, you only have 10 dB worth
> of margin with the 80 dB DR card, while you have a whopping 50 dB of margin
> with the 120 dB dynamic range card. You are much more likely to exploit the
> dynamic range of the basic musical event with the better card, but if you do
> everything right, and shave all the margins optimally, the lesser card just
> might deliver roughly comparable results.

Does this mean that if my card has a bigger dynamic range, my musician
can afford to play louder without causing clipping or distortion in
the recording?
August 17, 2004 12:02:36 AM

Archived from groups: rec.audio.pro (More info?)

Also, could it be some settings that I don't have right on my ASIO
drivers, in my program under device properties and settings, or on my
soundcard software, that might make the difference between these cards
more obvious and optimize the sound of the Audiophile?
Anonymous
August 17, 2004 10:36:16 AM

Archived from groups: rec.audio.pro (More info?)

"Bob" <fiddledd8@yahoo.com> wrote in message
news:b8e50c8f.0408161735.7c579577@posting.google.com

> Thanks for the input. I just thought that better sample rate and bit
> depth equals higher fidelity, which I thought would be audible.

Not true. If you do some listening to the files you can download from
www.pcabx.com you will find out that a really good 16 bit card, operating at
optimal levels, does little or nothing to reduce sound quality.

> What do you mean when you say "greater margin for recording error"? What
> is recording error?

Level mismatch. Obviously, if your levels are too high, there will be
clipping and audible distortion. However, the actual levels developed by a
living human musician or vocalist during a performance are subject to
considerable variations. Therefore, recording levels are usually set low and
conservatively. Every 6 dB the levels go below the absolute max, you lose
one bit's worth of resolution. In effect, the 16 bit card becomes a 15 bit
card. Normal 10 dB or so worth of headroom costs you nearly 2 bits worth of
resolution.

If you don't get the levels just right, and have reasonable allowance for
headroom, several bits of resolution are lost. Our nice little 16 bit audio
interface becomes, practically speaking perhaps a 12 bit audio interface.
The point where the loss of resolution becomes audible can be someplace
between 13 and 15 bits. 12 bits might not cut it! But if you are really
careful, and really lucky, you might not end up here.

Contrast this with using a premium audio interface with 19 bits worth of
resolution. If you have a 4 bit resolution loss due to level mismatch and
your allowance for headroom, you are now down to 15 bits, which probably
won't be a problem at all.
Anonymous
August 17, 2004 10:45:19 AM

Archived from groups: rec.audio.pro (More info?)

"Bob" <fiddledd8@yahoo.com> wrote in message
news:b8e50c8f.0408161845.ff79da2@posting.google.com
> "Arny Krueger" <arnyk@hotpop.com> wrote in message
> news:<3fadnVSF5a0K8b3cRVn-ug@comcast.com>...
>> "Bob" <fiddledd8@yahoo.com> wrote in message
>> news:b8e50c8f.0408152155.7efdf17f@posting.google.com
>>
>>> I just bought the M-Audio Audiophile 2496, hoping to get better
>>> sound than my Sound Blaster Live. Audiophile is 24 bit, 96 kHz.
>>> SB is 16 bit, 44.1 kHz.
>>
>
> Thanks very much for your detailed reply. Many helpful suggestions.
> I have a couple more questions.
>
> Key issues are to use
>> recording and monitoring equipment of sufficient quality, and to
>> time-synch and level-match the samples being compared.
>
> Don't understand what time-sync and level match mean.

When you are comparing two recordings, the most sensitive comparison
requires that the only difference be the effect you are listening for, in
this case the difference between two sound cards. Of course the music you
use for each test g must be absolutely identical - a very high quality
recording can provide this kind of source. The basic loudness of the
recordings must be absolutely identical, because you don't want to think
that an audio interface sounds better just because it is a bit louder, which
can easily happen. Also, the musical selections being compared have to start
and end at the same time in the music. You want to be comparing apples and
apples, the only audible difference being due to the inherent charateristics
of the two audio interfaces.

>> Audition has three basic BDs, 8, 16, and 32 bits. The 32 bit format is
>> floating point and has immense dynamic range. In any case you can't
>> make or playback recordings with effective BDs that exceed those
>> that the particular card supports, and you can't exploit high BD
>> recordings with an audio interfaces that don't support those BDs.
>> Audition tends to make these differences less obvious from a UI
>> standpoint, as it will automatically downsample and upsample to
>> mitigate mismatches between the audio interface and the recorded
>> format.

> Does this mean that if my card is advertised as having 24-bit 96khz
> quality, I won't get any use out of the 32 bit setting in audition?

What it means is that when Audition records from a 24 bit interface, using
Audioin's 32 bit mode, the 24 bit resolution limit of the interface rules.
The law of the weakest link is in force, and your recording can't possibly
have more than 24 bit resolution. The good news is that the 24 bits won't be
further corrupted by the 32 bit resolution of Audition.

> Or if I use my SB, which is a 16 bit/44.1khz card it won't do me any
> good to set the sample rate to 96khz at the beggining of my session?

Any recording made through a 16/44 audio interface is limited to the
frequency response limits of 44 KHz sampling, no matter what what follows.

> Also, what is floating point?

Floating point is a way for computers to store data with exceptional dynamic
range of about 1000 dB. Just another way to keep the computer from limiting
the resolution of the recording.

> Furthermore, some of the benefits of better audio interfaces relate to
>> safety margins. A card with better dynamic range gives you more
>> wiggle room for practical suboptimalities. For example one card can
>> have 80 dB dynamic range and another one can have 120 dB dynamic
>> range. Depending on other parameters, headroom, quality of
>> associated equipment, the room you record and listen in, and the
>> basic nature of the actual musical event being recorded, the dynamic
>> range of the recording might top out at 70 dB. If you get all the
>> levels optimized, both cards are capable of making a recording with
>> about 70 dB worth of dynamic range. However, you only have 10 dB
>> worth of margin with the 80 dB DR card, while you have a whopping 50
>> dB of margin with the 120 dB dynamic range card. You are much more
>> likely to exploit the dynamic range of the basic musical event with
>> the better card, but if you do everything right, and shave all the
>> margins optimally, the lesser card just might deliver roughly
>> comparable results.

> Does this mean that if my card has a bigger dynamic range, my musician
> can afford to play louder without causing clipping or distortion in
> the recording?

If you set levels with enough headroom, yes. I just covered this in more
detail in the response to you I just posted.
Anonymous
August 17, 2004 10:46:35 AM

Archived from groups: rec.audio.pro (More info?)

"Bob" <fiddledd8@yahoo.com> wrote in message
news:b8e50c8f.0408161902.5c4a6210@posting.google.com

> Also, could it be some settings that I don't have right on my ASIO
> drivers, in my program under device properties and settings, or on my
> soundcard software, that might make the difference between these cards
> more obvious and optimize the sound of the Audiophile?

At this time Audition doesn't *know* that ASIO exists.
Anonymous
August 17, 2004 1:49:26 PM

Archived from groups: rec.audio.pro (More info?)

On 16 Aug 2004 19:45:01 -0700, fiddledd8@yahoo.com (Bob) wrote:

>
>Key issues are to use
>> recording and monitoring equipment of sufficient quality, and to time-synch
>> and level-match the samples being compared.
>
>Don't understand what time-sync and level match mean.

He's talking about taking a rigorous approach to comparing the two
soundcards. Don't worry about it. if the card sounds good to you -
enjoy :-)


>
>Does this mean that if my card is advertised as having 24-bit 96khz
>quality, I won't get any use out of the 32 bit setting in audition?
>Or if I use my SB, which is a 16 bit/44.1khz card it won't do me any
>good to set the sample rate to 96khz at the beggining of my session?
>Also, what is floating point?

I don't know Audition particularly. But many quality multi-track
programs run internally at a high bit-rate, quite independently of the
chosen input and output sample rate. This gives headroom, making it
almost impossible to run into overload within the mixer. All you
have to do is check input levels and the final output level.

Floating point is a method of mapping audio levels to bytes of data.
You can't do anything about it. Nod sagely at the buzz-word and get
on with your music :-)

Your SB is actually a 48KHz card, with on-the fly re-sampling to other
rates on input and output. This is perhaps its main limitation.
But, yes, I see little point in setting up a 96KHz project if you have
no 96KHz source.


>
>Does this mean that if my card has a bigger dynamic range, my musician
>can afford to play louder without causing clipping or distortion in
>the recording?

Sort of. Actually, it allows you to set levels LOWER, without
falling into the noise floor.


CubaseFAQ www.laurencepayne.co.uk/CubaseFAQ.htm
"Possibly the world's least impressive web site": George Perfect
August 17, 2004 7:02:28 PM

Archived from groups: rec.audio.pro (More info?)

Just a thought;
I tended to find the difference between my old SB Live and my new
M-Audio Delta 1010LT's became more apparent when I started mixing more
and more recorded tracks together. It just seems to have more clarity.

This is probably a combination of a bigger dynamic range, as a few
others have mentioned, and less sonic distortion. I'm no expert, but I
imagine distortion in a particular frequency range will increase when
tracks are summed.

Cheers,

Mark.
--

Bob wrote:
> I just bought the M-Audio Audiophile 2496, hoping to get better sound
> than my Sound Blaster Live. Audiophile is 24 bit, 96 khz. SB is 16
> bit, 44.1 khz. In Audobe Audition, I try recording the exact same
> thing (a short piano piece) on each card for a back-to-back
> comparison, but both recordings sound the same. I have a basic
> understanding of bit depth and sample rate, but I don't know how the
> bit depth and sample rate of my sound card relates to the BD and SR
> that I set my recording session to, and what occurs if I then convert
> the BD and SR. So could it have something to do with those settings?
> Please let me know, in plain English, when I will be able to hear the
> difference between different BD's and SR's, and between these two
> sound cards. So much rave about the Audiphile cards, and negativity
> towards Sound Blaster, and I can't hear the difference!
Anonymous
August 18, 2004 9:02:45 PM

Archived from groups: rec.audio.pro (More info?)

"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:zKKdnU7K7tOAfbzcRVn-uw@comcast.com...
> "Bob" <fiddledd8@yahoo.com> wrote in message
> news:b8e50c8f.0408161735.7c579577@posting.google.com
> > What do you mean when you say "greater margin for recording error"?
What
> > is recording error?
>
> Level mismatch. Obviously, if your levels are too high, there will be
> clipping and audible distortion. However, the actual levels developed by a
> living human musician or vocalist during a performance are subject to
> considerable variations. Therefore, recording levels are usually set low
and
> conservatively. Every 6 dB the levels go below the absolute max, you lose
> one bit's worth of resolution. In effect, the 16 bit card becomes a 15 bit
> card. Normal 10 dB or so worth of headroom costs you nearly 2 bits worth
of
> resolution.

Yes, and since the Live is only good for a real 14 bits to start with, you
can't waste too many more allowing for headroom.
Get it all just right, and the SB live is OK, but there is no margin for
error. The chances of clipping or excessive noise is high.

TonyP.
Anonymous
August 18, 2004 9:02:46 PM

Archived from groups: rec.audio.pro (More info?)

"TonyP" <TonyP@optus.net.com.au> wrote in message
news:4122ff31$0$18669$afc38c87@news.optusnet.com.au
> "Arny Krueger" <arnyk@hotpop.com> wrote in message
> news:zKKdnU7K7tOAfbzcRVn-uw@comcast.com...
>> "Bob" <fiddledd8@yahoo.com> wrote in message
>> news:b8e50c8f.0408161735.7c579577@posting.google.com
>>> What do you mean when you say "greater margin for recording error"?
> What
>>> is recording error?
>>
>> Level mismatch. Obviously, if your levels are too high, there will be
>> clipping and audible distortion. However, the actual levels
>> developed by a living human musician or vocalist during a
>> performance are subject to considerable variations. Therefore,
>> recording levels are usually set low and conservatively. Every 6 dB
>> the levels go below the absolute max, you lose one bit's worth of
>> resolution. In effect, the 16 bit card becomes a 15 bit card. Normal
>> 10 dB or so worth of headroom costs you nearly 2 bits worth of
>> resolution.

> Yes, and since the Live is only good for a real 14 bits to start
> with, you can't waste too many more allowing for headroom.

This is a key point. Audio production is very much prone to slips and
glitches, so safety margin is very important.

To put things into perspective, Good quality line-level audio gear is
capable of dynamic range on the order of 110-120 dB, sometimes more. If
people use practice and procedures that work with that kind of massive
headroom reserves, but this time with sound cards that have only 80 dB
dynamic range, its a recipie for disaster. If you want a sound card with the
same kind of dynamic range as good quality line-level audio gear, you need
to move smartly towards the high end.

I feel pretty comfortable with audio interfaces with performance in the
95-103 dB range.

> Get it all just right, and the SB live is OK, but there is no margin
> for error. The chances of clipping or excessive noise is high.

Noise builds up as you add processing and tracks.
August 21, 2004 5:55:14 PM

Archived from groups: rec.audio.pro (More info?)

a few things to help you out:

1)i thought soundblaster was 48k, not 44.1...you sure about that?
2) make sure your software is set to 24 bit recording. many software
apps default to 16 bit.
3) if you're using cheap computer speakers, it can be hard to tell the
differences, epecially if you have background computer fan noise
clouding your listening
4) the audiophile doesn't sound that amazing. it's "ok". what it
does have is excellent drivers. the thing just "works" with midi,
ASIO, GSIF, etc.

I think of it as kind of an ugly deep sea fish. It's not pretty, but
it's tough and it gets you there.
Anonymous
August 21, 2004 11:13:48 PM

Archived from groups: rec.audio.pro (More info?)

"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:b6udnQdE3bfft77cRVn-oQ@comcast.com...
> "TonyP" <TonyP@optus.net.com.au> wrote in message
> > Get it all just right, and the SB live is OK, but there is no margin
> > for error. The chances of clipping or excessive noise is high.
>
> Noise builds up as you add processing and tracks.

Not at all. The processing can be done at 32 bit FP (or even 64 bit!), and
no need for track bouncing with digital.

TonyP.
Anonymous
August 21, 2004 11:13:49 PM

Archived from groups: rec.audio.pro (More info?)

"TonyP" <TonyP@optus.net.com.au> wrote in message
news:41271267$0$6809$afc38c87@news.optusnet.com.au

> "Arny Krueger" <arnyk@hotpop.com> wrote in message
> news:b6udnQdE3bfft77cRVn-oQ@comcast.com...

>> "TonyP" <TonyP@optus.net.com.au> wrote in message

>>> Get it all just right, and the SB live is OK, but there is no margin
>>> for error. The chances of clipping or excessive noise is high.

>> Noise builds up as you add processing and tracks.

> Not at all. The processing can be done at 32 bit FP (or even 64 bit!),

That can be true, but the noise still builds up as you add processing. It's
just that with 32 bit FP, it adds up at such a level that it is irrelevant.
However, I was speaking in the context of the general case, which includes
16 bit processing. If you're making the point that you can track at 16 bits
and then use 32 bit processingm, that point is well-taken.

>and no need for track bouncing with digital.

I wasn't speaking of track bouncing. If you mix a lot of tracks, each one
brings in its noise floor. Hopefully, each track's noise floor will be
random and be attenuated during the mixing process. However, any discrete
noise that is common to a number of tracks will add with simple arithmetic
instead of adding geometrically like random signals. Things add up much
faster with simple arithmetic.
Anonymous
August 23, 2004 5:36:13 AM

Archived from groups: rec.audio.pro (More info?)

"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:X4edna_5YaSFvbrcRVn-qA@comcast.com...
> That can be true, but the noise still builds up as you add processing.
It's
> just that with 32 bit FP, it adds up at such a level that it is
irrelevant.

If you're going to pick nits, it should be easy to use a NR algorythm that
will recover the fraction of a dB lost, without being too audible :-)

> However, I was speaking in the context of the general case, which includes
> 16 bit processing. If you're making the point that you can track at 16
bits
> and then use 32 bit processingm, that point is well-taken.

Of course I was. You don't even have to choose with most programs.

> >and no need for track bouncing with digital.
> I wasn't speaking of track bouncing. If you mix a lot of tracks, each one
> brings in its noise floor. Hopefully, each track's noise floor will be
> random and be attenuated during the mixing process. However, any discrete
> noise that is common to a number of tracks will add with simple arithmetic
> instead of adding geometrically like random signals. Things add up much
> faster with simple arithmetic.

Not true, this occurs only if you don't gate unused tracks. Otherwise the
real signals all add as well as the noise. In fact the signal will be more
correlated than the noise in most cases.

TonyP.
!