Avid Pro Audio Community

Avid Pro Audio Community

How to Join & Post  •  Community Terms of Use  •  Help Us Help You

Knowledge Base Search  •  Community Search  •  Learn & Support


Avid Home Page

Go Back   Avid Pro Audio Community > Pro Tools Software > macOS

Reply
 
Thread Tools Search this Thread Display Modes
  #1  
Old 03-26-2013, 01:03 PM
radardoug radardoug is offline
Member
 
Join Date: Mar 2013
Location: Kerikeri New Zealand
Posts: 7
Default The two questions that Avid don't want to answer

Hi All,
Just thought I would come on and post the two questions that Avid don't want to answer.
First, why does Protools in various versions have so much latency?
Even Yamaha in their early mixers had virtually no latency.
For instance the 02R. It would seem to me that the only reason for latency is as a selling tool, i.e. the cheap interfaces have latency, the expensive one don't.
It takes microseconds to add several digital numbers and spit them out.
Come on Avid, come clean!

Question two. If I copy a sign wave at say - 6dB to 24 tracks, and then sum it out say outputs 1 and 2, it clips. Why?
Because the user selects the number of tracks to an output, Avid could autoscale the individual inputs to correct this problem. Why don't you Avid?

And so users say that mixing in the box sounds worse than sending outputs out separately and summing externally. And it does, because it clips.
Reply With Quote
  #2  
Old 03-26-2013, 01:19 PM
the19thbear the19thbear is offline
Member
 
Join Date: Apr 2011
Location: Denmark
Posts: 223
Default Re: The two questions that Avid don't want to answer

Hehe... Funny guy:)
Reply With Quote
  #3  
Old 03-26-2013, 01:20 PM
crizdee's Avatar
crizdee crizdee is offline
Moderator
 
Join Date: Dec 2004
Location: Brighton, UK
Posts: 10,482
Default Re: The two questions that Avid don't want to answer

Quote:
Originally Posted by radardoug View Post
Hi All,
Just thought I would come on and post the two questions that Avid don't want to answer.
First, why does Protools in various versions have so much latency?
Even Yamaha in their early mixers had virtually no latency.
For instance the 02R. It would seem to me that the only reason for latency is as a selling tool, i.e. the cheap interfaces have latency, the expensive one don't.
It takes microseconds to add several digital numbers and spit them out.
Come on Avid, come clean!

Question two. If I copy a sign wave at say - 6dB to 24 tracks, and then sum it out say outputs 1 and 2, it clips. Why?
Because the user selects the number of tracks to an output, Avid could autoscale the individual inputs to correct this problem. Why don't you Avid?

And so users say that mixing in the box sounds worse than sending outputs out separately and summing externally. And it does, because it clips.
You learn something new every day

I'm selling my world leading Pro Tools system and buying a Yamaha O2

See ya


Chris
__________________
PT MAC Troubleshooting... http://duc.avid.com/showthread.php?t=54888

Producer, Engineer,
UKmastering Mixing & Mastering
Blinders_Columbia top 40 UK album charts
Slow Readers Club_Build A Tower top 20 UK album charts

www.ukmastering.com


PT10.3.10 Mountain Lion HD6 accel Magma PE6R4 Control 24 MacPro 12 Core 3.46ghz UAD-2 Octo x2. Manley Vari-Mu, Manley Massive Passive, SSL VHD, ADL600, Grove Tubes ViPre, Tube-Tech CL-1B. Hafler TRM active monitoring.

Last edited by crizdee; 03-29-2013 at 01:23 AM.
Reply With Quote
  #4  
Old 03-26-2013, 01:20 PM
Craig F Craig F is offline
Member
 
Join Date: Aug 2000
Location: Portland, OR
Posts: 12,591
Default Re: The two questions that Avid don't want to answer

1) there is massive difference between a digital mixer and a DAW (especially with a USB or Firewire interface)
2) Pro Tools mixer works the same way any mixers works, if you feed 24 track of -6 tone into mixers set up at unity it would clip/distort
2b) I would not want to try and mix anything on a mixer that was autoscaling my inputs
__________________
...

"Fly High Freeee click psst tic tic tic click Bird Yeah!" - dave911

PT11 System Requirements link http://avid.force.com/pkb/articles/C...m-Requirements


Thank you,

Craig
Reply With Quote
  #5  
Old 03-26-2013, 01:27 PM
radardoug radardoug is offline
Member
 
Join Date: Mar 2013
Location: Kerikeri New Zealand
Posts: 7
Default Re: The two questions that Avid don't want to answer

There is a massive hardware difference, but both products are doing the same job. In terms of DSP code, there is not that much difference.
Bothe Firewire and USB are quoted as high speed interfaces, are you saying they are not?
When I say autoscaling, I mean that this is a one time thing based on the selection of tracks, It's not going to be changing gain dynamically.
There is another way to do it of course, use greater bit numbers and hence dynamic range, and allow for the need for headroom.
Good analog mixers will not distort under these conditions, because the designer is aware of the combination factor.
Secondary to this, why do Protools not provide a way of monitoring the bus level so that users can see if it is clipping. That wouldn't be difficult.
Reply With Quote
  #6  
Old 03-26-2013, 01:41 PM
carlos santana carlos santana is offline
Member
 
Join Date: Feb 2005
Location: Canada
Posts: 300
Default Re: The two questions that Avid don't want to answer

ll you have to dl is create a mster fader and use it for monitoring level off all summed traks
Reply With Quote
  #7  
Old 03-26-2013, 01:58 PM
Chief Technician's Avatar
Chief Technician Chief Technician is offline
Moderator
 
Join Date: Dec 2001
Location: NYC
Posts: 6,981
Post Re: The two questions that Avid don't want to answer

Quote:
Originally Posted by radardoug View Post
Question two. If I copy a sign wave at say - 6dB to 24 tracks, and then sum it out say outputs 1 and 2, it clips. Why?
When you sum two signals whose frequency content, amplitude, and phase are identical, you will get a +6dB gain at the sum of those two signals. So if you have two sine waves whose frequency is 1kHz, whose signal level is -6dBFS, and whose phase is coherent with each other and then add them together, you are going to have a 0dBFS (aka full scale) 1kHz sine wave. You have already met your headroom with only two tracks. Not expecting clipping at the output when summing 24 tracks of 1kHz sine waves at the same amplitude level is absurd. Even an analog mixer is going to clip under those circumstances.
__________________
Jonathan S. Abrams, CEA, CEV, CBNT
Apple Certified - Technical Coordinator (v10.5), Support Professional (v10.6 through v10.10)
Reply With Quote
  #8  
Old 03-26-2013, 01:59 PM
Craig F Craig F is offline
Member
 
Join Date: Aug 2000
Location: Portland, OR
Posts: 12,591
Default Re: The two questions that Avid don't want to answer

USB and Firewire are slow relative to other connection systems / a digital mixers is a contained system
DAW software has to get the data from the converter over USB or Firewire from the OS, many more steps than a digital mixers custom software tuned to minimize latency
try a HD system, latency is similar to a digital mixer

even a 32 bit data path has to go to a 24 bit DAC
__________________
...

"Fly High Freeee click psst tic tic tic click Bird Yeah!" - dave911

PT11 System Requirements link http://avid.force.com/pkb/articles/C...m-Requirements


Thank you,

Craig
Reply With Quote
  #9  
Old 03-26-2013, 02:07 PM
Hugh-H Hugh-H is offline
Member
 
Join Date: Jun 2011
Location: Los Angeles
Posts: 192
Default Re: The two questions that Avid don't want to answer

Hello,

I'm sort of hoping he's kidding, but I'll bite anyway -

Hello radardoug,

Actually the dsp in a hardware mixer and the dsp in your daw are different architecturally. They accomplish the same thing but do it differently. The processing in your daw requires the higher latency in order to buffer the I/O or it can't keep up, whereas the dsp in a hardware mixer does not need this (to a first approximation) because it has dedicated tdm (NOT Digi TDM!) timeslots for processing and I/O, mostly negating the need for buffering although it still exists to a smaller degree.

Dedicated daw dsp (Digi, Scope, Mixtreme, et.al.) and hardware mixer dsp have an architecture that is dsp-based, meaning it is time-division-multiplexed (the TDM acronym in Digi's TDM systems). Each signal has a timeslot and it keeps running along at a fixed pace. In a computer cpu it only does one thing at a time (for simplification sake I'll call it only one) and it has to swap out fast, so the I/O is buffered to give the cpu time to spend on processing before it must grab some more.

In our latest desk purchases (Calrec Apollo) Calrec could not acquire a dsp solution that had enough capability at short enough processing delay times, so they've taken another approach that's even faster - dedicated fpga that they custom burn to give them an enormous amount of massively parallel processing on a single chip.

Another desk manufacturer faced with the same limitations on their dsp-based solution, is considering going the other way - several Intel cpus in a custom OS that would leverage their extreme power outside of a standard OS, effectively turning them into dsp-like solutions. We'll see if they succeed.

If you genuinely think a native daw and dsp are the same and are interested in learning I would suggest some research on dsp architecture vs. computer cpu architecture. Quite fascinating to me anyway.

As to an autoscaling functionality, well, we're all entitled to an opinion. A master fader would be a good place to start.

Hugh
Reply With Quote
  #10  
Old 03-26-2013, 02:08 PM
radardoug radardoug is offline
Member
 
Join Date: Mar 2013
Location: Kerikeri New Zealand
Posts: 7
Default Re: The two questions that Avid don't want to answer

Quote:
Originally Posted by Chief Technician View Post
When you sum two signals whose frequency content, amplitude, and phase are identical, you will get a +6dB gain at the sum of those two signals. So if you have two sine waves whose frequency is 1kHz, whose signal level is -6dBFS, and whose phase is coherent with each other and then add them together, you are going to have a 0dBFS (aka full scale) 1kHz sine wave. You have already met your headroom with only two tracks. Not expecting clipping at the output when summing 24 tracks of 1kHz sine waves at the same amplitude level is absurd. Even an analog mixer is going to clip under those circumstances.
Exactly my point! In a normal ITB Protools session you are going to be mixing a number of tracks, and their level could actually be up to 0dBFS. The mixer requires headroom. Actually as I have previously stated, a well designed analog mixer will not clip under these conditions. Because the designers have allowed for the fact that you are mixing a large number of tracks.
Why has Protools not allowed for this?
Is this why 90% of recorded music these days sounds like [bleep][bleep][bleep][bleep]?
Reply With Quote
Reply

Thread Tools Search this Thread
Search this Thread:

Advanced Search
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off

Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
Dear Avid customer support! Would you please answer my both avid support cases Ale Pro Tools HDX & HD Native Systems (Mac) 0 11-24-2011 01:56 PM
Avid Answer Our 5 Questions On Pro Tools 10 & HDX - Is This Enough? zedhed Pro Tools 10 132 11-06-2011 01:39 AM
Answer these questions three RobMacki Pro Tools TDM Systems (Mac) 1 09-19-2003 11:41 PM
Please Answer Some Questions B4 I Buy JPGAndR Tips & Tricks 1 07-19-2001 09:52 AM
Three questions about OSX for anyone who can answer dave23 003, Mbox 2, Digi 002, original Mbox, Digi 001 (Mac) 6 02-16-2001 11:48 AM


All times are GMT -7. The time now is 04:15 PM.


Powered by: vBulletin, Copyright ©2000 - 2008, Jelsoft Enterprises Limited. Forum Hosted By: URLJet.com