Are we there Yet !!

this thread is entertaing at least.

 
mudgel

jyeager11

mudgel

No. What i'm saying is that once you uncheck that box and Don't share drivers with other programs you will be able to play something in Sonar while at the same time play something completely different in another program; though why you'd want to do that I don't know.

While that makes absolutely no sense to me (logic dictates that the whole point of having a "Share Driver With" option is to allow you to "Share Driver With" when it's checked, not unchecked) -- the entire argument is moot because whether checked or not, I can't get Sonar X1b to let me play anything else as long as it's running. It doesn't even need to be playing anything, or even have the window focus. As long as it's simply PRESENT, nothing else will play. Not WMP, not IE, not FF, nothing.

Alternatively, if another application is already playing a sound (such as FF playing a YouTube video, for instance) and THEN I load Sonar X1b, then I'm told by Sonar that the drivers are unavailable and am offered the option to disable or use them anyway. As you might imagine, using them anyway produces no sound from Sonar X1b.

But if I'm on YouTube and the video is stopped, and then I load Sonar X1b, then Sonar hijacks the driver for as long as it's loaded. The YouTube video will not produce sound, and Sonar X1b will. Until I shut down Sonar.

Conclusion :
The "Share Driver With" option in Sonar X1b preferences has absolutely ZERO effect on my Echo Gina 3G. Whichever application is playing sound first is the one hijacking the driver. In the case of Sonar X1b, it doesn't even need to be playing the sound first to hijack the audio driver - it just needs to have been launched while no other applications were emitting any sound.

Any of this make sense to anyone else?

Using ASIO.

Sorry you find what I said is illogical. I don't want to argue with you. Nevertheless it is a fact not my opinion.. Nothing I can do about that. You have come to an erroneous conclusion as there are other factors at play here besides the Share Drivers option
 
 
The Sharing drivers option is included for that very reason ie. Whichever program has focus gets the audio device. that makes sense as it stops 2 programs from simulataneously playing different audio streams through the sound device.
 
As I said there is also something else going on causing your issues.
 
If you're using ASIO for SONAR it explains the matter. Windows and media player are not using the same drivers. Windows doesn't work with ASIO drivers. So there's another reason. Widows will be grabbing the MME or Windows sound Mapper(WDM/KS or WASPI driver for whatever programs are running in Windows but if SONAR is set to ASIO then there'll be a conflict which you are experienceing.
 
It's just not possible to address a bit of hardware with 2 different driver models. Every bit of software has to al least be using the same driver to have a chance.
 
Usually what we do is to use onboard sound device for Windows sounds and any associated audio programs and leave SONAR to use the Pro/Semi Pro device exclusively.
 
if you really want all these things to be able to play simultaneously then you'll have to use the same driver for all the programs and as Wiondows doesn't use ASIO drivers you'll have to choose WDM for SONAR and Windows, then you'll be able to uncheck Share Drivers and notice the difference and away you go.
 
that's the way it is - You have your answer.

mudgel - I'm sorry but some of what you say here is incorrect.  I currently have Soanr X1 open and Windows Media Player open both at the same time.  Sonar is using ASIO drivers for the MOTU.  In my MOTU panel I have "use wavert drivers for windows" checked.
 
both WMP and Sonar will play at the same time.  it's cacophony, but they both play at the same time.  There are 2 reasons I can think of why windows can't use the soundcard at the same time that sonar is open.  1! is that the drivers aren't written (for the Gina in jyeager11's case) so that both driver modes can be accessed at the same time.  that would be the manufacturer of the drivers to change that.
 
2) I have noticed that if the sonar project I have open has a sample rate of anything other than 44.1, then WMP will not play or Sonar will not play one or the other.  it depends on which one I opened first.  the first one gets the drivers the other does not if the sample rates are mismatched.

Loptec

Here’s what you wrote in Swedish:

“It is not my intention to confront, but find you all these details important. Your analogy is incorrect since Sweden in English as he. The analogy isn’t correct it self. Not just the word you use.”

If this really is what you wanted to say.. Well.. Good for you then.. :)

Heh. That's rather disappointing of Google translate...

UnderTow

How does bit depth (dynamic range) and sample rate (frequency range) in the digital domain compare to the analog domain? I've been reading that professional analog tape has a frequency range of 10Hz ~ 30kHz(+) so to be able to emulate analog (which is what we're all striving for isn't it?), wouldn't you need to be recording in the digital domain at at least 88.2/24?

Here's some good reading on Analog Vs. Digital and 96kHz. <- This information is leached from another page that is linked at the bottom. This page has an interesting analog to digital reference chart at the bottom that's not on the page the original information came from.

BTW ... it references 'resolution' several times.

How does bit depth (dynamic range) and sample rate (frequency range) in the digital domain compare to the analog domain? I've been reading that professional analog tape has a frequency range of 10Hz ~ 30kHz(+) so to be able to emulate analog (which is what we're all striving for isn't it?), wouldn't you need to be recording in the digital domain at at least 88.2/24?

Where this logic goes off track is in the presumption that recording inaudible frequencies is actually beneficial.

They are all going to be filtered out at some point, whether it's during MP3 encoding, SRC to 44.1 for CDs, or simply due to the limitations of your playback system. And even if you do manage to turn 30KHz content into acoustical energy by playing 88.2KHz files through speakers equipped with ultrasonic transducers, you still won't be able to physically hear it!

There is just one thing that higher sample rates can accomplish: they relax the need for anti-aliasing measures within the signal chain. But there are other ways to do that which are far more efficient, such as oversampling within plugins that are prone to aliasing (e.g. limiters).

John

I've been reading that professional analog tape has a frequency range of 10Hz ~ 30kHz(+)

What does that mean? That tape has that frequency band or that a tape recorder has that frequency band including the tape? At what dB -/+?

What I read referred to the tape. "High quality open-reel tape frequency response can extend from 10 Hz to well above 20 kHz." It also goes on to say that some LP's had frequencies up to 50kHz encoded on them. Click here for the full article.

Resolution is used in graphics when discussing at what point two things can be distinguished from one another meaning resolved. This is used in optics but has no real meaning in audio. One needs to understand what the "circle of confusion" is as it relates to fine detail in an optical system. This has nothing to do with audio.

I gave you a link where other folks use the term resolution when explaining properties of digital audio. I'm not sure where they are getting their information. Here's a quote from a second article, "So far, actual jitter in consumer products seems to be too small to be detected at least for reproduction of music signals. It is not clear, however, if detection thresholds obtained in the present study would really represent the limit of auditory resolution or it would be limited by resolution of equipment."

I personally don't care about the use of the term, I'm just trying to show you that others do use it in regard to audio.

Thanks,

Bub.

bitflipper

How does bit depth (dynamic range) and sample rate (frequency range) in the digital domain compare to the analog domain? I've been reading that professional analog tape has a frequency range of 10Hz ~ 30kHz(+) so to be able to emulate analog (which is what we're all striving for isn't it?), wouldn't you need to be recording in the digital domain at at least 88.2/24?

Where this logic goes off track is in the presumption that recording inaudible frequencies is actually beneficial.

They are all going to be filtered out at some point, whether it's during MP3 encoding, SRC to 44.1 for CDs, or simply due to the limitations of your playback system. And even if you do manage to turn 30KHz content into acoustical energy by playing 88.2KHz files through speakers equipped with ultrasonic transducers, you still won't be able to physically hear it!

There is just one thing that higher sample rates can accomplish: they relax the need for anti-aliasing measures within the signal chain. But there are other ways to do that which are far more efficient, such as oversampling within plugins that are prone to aliasing (e.g. limiters).

It is important to note that the nyquist theorem applies to signals that are sampled for infinite time and any time-limited signal cannot be perfectly bandlimited.

bitflipper

SvenArne: Comparisons to images/video are confusing at best. They do not help anyone better understand how digital audio works.

jyeager11: Only to people who have no inclination of how images/video work. To those of us that do, Loptec's metaphor was spot-on, and would help many newbies coming in here with more knowledge of image than audio.

I just wanted to repeat these quotes for the benefit of any actual newbies who might be trying to follow along. The former paragraph is correct. The latter is absolutely bogus and a great disservice to struggling noobs.

Well it seems that I came back here to write again, after all.. :)
(Whiii! More popcorn to the people!) 

For some people the only important thing is to compare the technical aspects of a digital image and digital audio. If it’s ANYTHING in this world that gets confusing for a newbie, it’s this kind of talk!

My metaphor had only one purpose and that was to give an easy-to-understand description of why it’s better to use higher quality.

To best get to understand something that you haven’t been in contact with before you sometimes have to make comparisons to things where, even though they’re totally different, you still have to think in similar ways to get the best result.

Like in my example, when working with digital visual art and with digital audio: Use high quality because if you do you have more control over what you’re working with.

To even start talking about the technical things just make things sooo pathetic.. This isn't what it's about!

It’s like taking your 3 year old child to a soccer field and start lecturing him about all rules and strategies of the game... What he’s allowed to do and what he’s not... “Nonono!!! No hands, child! No hands! ..Now you’ll have to sit outside the field for 2 mins. and watch dad play! Well, off you go then!!! NO CRYING!! These are the rules of the game!!”
..When all he wants to do is try and kick the ball..

For god’s sake, just let him kick the ball and develop from there in his own f**** pase..

And also... How can someone complain about a metaphor regarding digital audio and a digital image and then later in the thread compare Swedish with Dutch?! Should I start complaining about this comparison now? Should I tell everyone over and over how totally different these two languages are?  I could but I won’t, since 1) it’s details that won’t be interesting, since it’s not what we’re actually talking about and 2) I’m no jerk

Audio signals that we work with do not meet the criteria for application of the theorem therefore there is no reason to believe that the theorem's expected results would apply.

You are kidding right? The only problem with that is we see it doing just that every time we play digital audio or record digital audio.

"the can be perfectly reconstructed part of the theorem"

Maybe maybe not. It can be quickly tested by created a 20 kHz sine wave.

Just because most of us can not quantify the difference with current playback technology does not mean there is not a difference that has value.

I have no idea what this means.

You are kidding right?

absolutely not the theorem is clear on the type of signals that it applies to.

Audio signals that we work with do not meet the criteria for application of the theorem therefore there is no reason to believe that the theorem's expected results would apply.

Huh?

Just because most of us can not quantify the difference with current playback technology does not mean there is not a difference that has value.

Double huh?

I'm not questioning whether or not there is a valid point in there somewhere, but whatever it is it zipped right past my head!

Loptec

To even start talking about the technical things just make things sooo pathetic.. This isn't what it's about!

If you find it pathetic, don't join in a technical discussion. But don't just call it pathetic just because you have been shown to be wrong and can't accept that you are wrong.

It’s like taking your 3 year old child to a soccer field and start lecturing him about all rules and strategies of the game...

I don't assume the people that read this forum are three year olds. I assume they are all fully capable of understanding what this is about if they are interested. Maybe not directly if they are missing some bits of information but anyone that wants to know more can always ask for a clarification or pointers to more information. Telling people something that is technically wrong is not going to help anyone.

Btw, I found an interesting article about sampling theory and Video where they are also saying that there is no point in going over certain sampling rates and numbers of pixels etc. It seems that it is not only in audio that people get this wrong. Not surprising considering that it is is not directly intuitive. See here: http://www.cognitech.com/pdfs/samplingTheory.pdf

And also... How can someone complain about a metaphor regarding digital audio and a digital image and then later in the thread compare Swedish with Dutch?! Should I start complaining about this comparison now?

1) This is not a forum about language but one about audio and Sonar. 2) Of course they can be compared! Linguists compare languages all the time! 3) Everything I said about English, Dutch and Swedish was 100% correct for the simple and obvious reason that I said I was guessing. I know my limitations. I know when I don't know enough about a subject not to present things in absolute terms.

Everyone has already acknowledged that it is best to record at 24 bits. That does not mean that your analogy was correct. Can you really not live with that?

UnderTow

rabeach

John

It is important to note that the nyquist theorem applies to signals that are sampled for infinite time and any time-limited signal cannot be perfectly bandlimited.

What does that mean?

Audio signals that we work with do not meet the criteria for application of the theorem therefore there is no reason to believe that the theorem's expected results would apply. Sampling at the nyquist frequency may not offer the accurate signal sampling "the can be perfectly reconstructed part of the theorem" that people choose to believe it does. Over sampling could have benefits in these regards and of course drawbacks. Just because most of us can not quantify the difference with current playback technology does not mean there is not a difference that has value.

That depends on how you define value. As regards to audio destined for human consumption, I would say that if no difference can be perceived by humans, the difference has no value.

UnderTow

UnderTow

rabeach

John

It is important to note that the nyquist theorem applies to signals that are sampled for infinite time and any time-limited signal cannot be perfectly bandlimited.

What does that mean?

Audio signals that we work with do not meet the criteria for application of the theorem therefore there is no reason to believe that the theorem's expected results would apply. Sampling at the nyquist frequency may not offer the accurate signal sampling "the can be perfectly reconstructed part of the theorem" that people choose to believe it does. Over sampling could have benefits in these regards and of course drawbacks. Just because most of us can not quantify the difference with current playback technology does not mean there is not a difference that has value.

That depends on how you define value. As regards to audio destined for human consumption, I would say that if no difference can be perceived by humans, the difference has no value.

UnderTow

Can't disagree with that; the point is I guess that humans don't perceive it due to the technology they use to acquire it. The value could be undiscovered.

Were not there yet ..are we...