The two questions that Avid don't want to answer

[radardoug]

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.

[the19thbear]

Hehe... Funny guy:)

[crizdee]

Originally Posted by radardoug

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

[Craig F]

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

[radardoug]

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.

[carlos santana]

ll you have to dl is create a mster fader and use it for monitoring level off all summed traks

[Chief Technician]

Originally Posted by radardoug

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.

[Craig F]

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

[Hugh-H]

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

[radardoug]

Originally Posted by Chief Technician

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]?