24bit/96khz & 192khz sorry for audio noob question

[amrrahmy]

but your talking built up delay, the more is recorded the more delay u get.
in a 10min scene u would get trouble by the end.

 

[Kevin Halverson]

but your talking built up delay, the more is recorded the more delay u get.
in a 10min scene u would get trouble by the end.

You are confusing delay with sample rate drift. These are not the same things at all. The delay is simply latency, it is constant and is therefore irrelevant (it does not accumulate as you have suggested).

What matters is maintaining the sample rate of any external audio recorder (analog or digital) to the sample rate of the video system (or film system in those cases) and the edge or time code meta data. This can be accomplished by either jam syncing the camera and field recorder (locks the meta data) or by a better approach of using a word clock signal (locks the two by sample frames). Word clock systems can be thought of as a similar system to the old pilot tone system used in the past.

Regardless of the method, as long as both sample clocks are accurate, there is nearly no drift (or effectively minimal drift with the occasional resynchronization) when jam synced. When locked via a word clock, the two recording systems (camera video and separate audio) remain perfectly synced no matter what the duration of the recording event. Which a word clock could be 10 minutes, 10 hours or 10,000 years, and the two systems they always remain synchronized.

The presence of the DSP in the signal chain of the recording system that I am describing has no effect on this relationship between the audio and video/film recording systems.

Kevin Halverson

 

[amrrahmy]

the constant process in the middle after recording the sound and before the final sound is recorded has a constant continuous delay, no matter how low it may be and trying to resync in sound pauses (no sound and low data in room sound recording) is not good enough and would mean that u have ur own cpu running to resync, that would be way too eeexxxxppppenssssssssive and not better than realtime 96k(i think)

 

[Kevin Halverson]

the constant process in the middle after recording the sound and before the final sound is recorded has a constant continuous delay, no matter how low it may be and trying to resync in sound pauses (no sound and low data in room sound recording) is not good enough and would mean that u have ur own cpu running to resync, that would be way too eeexxxxppppenssssssssive and not better than realtime 96k(i think)

Nope, completely off the mark. The time through the LP filter (number of clock cycles) is always constant regardless of the sample amplitude. It's simply a set of taps - multipliers and accumulators, it is always the same and is based upon the requirements of the filtration, not based upon the data in any way.

There is already a lot of processing "...in the middle after..." including DSP functions, such as level changes, channel assignment, etc. The additional processing burden of a hardware DSP on the CPU is zero. This LP filter is in the hardware chain after the ADC, not within a post processing CPU. I can assure you that the costs are trivial (in terms of hardware DSP) as these products are already in development (and in some cases, within the ADC Si itself).

Kevin Halverson

 

[amrrahmy]

There is already a lot of processing

no there isn't.
level changes can be manual as well(and more accurate), so no processing is no biggy and u can do a 6ch through 3 duel on a macbook pro(or at least, it's my dream if enough money came from the sky and landed on my lap.(that would be nice))

 

[amrrahmy]

also if the epic came with the same quality of nikon d3, that would be sweet.

 

[Kevin Halverson]

Yes, there is. I have designed a number of ADCs over the years and other than single cycle flash (not appropriate for most audio applications), all have serious hardware processing engines (often in Si).

Nealy every ADC that is employed in digital audio applications have decimation filters (unless we are talking about single bit processes) so there is already a powerful DSP (even if it isn't called a DSP) in the signal path and it is already doing a lot of processing on every sample. I can show you the die area plots that are dedicated to various aspects of the Si (chip) and this will easily confirm the considerable processing that is occurring. Further, dedicated hardware filters are much better at doing this type of processing than some CPU (not unlike why we design visual processing engines with dedicated GPUs).

I think you are still confusing the host computer ("macbook pro" in your example) with the ADC hardware. These are different pieces in the digital audio chain.

As for "manual" level changes, are you referring to scaling the analog signal prior to the quantizing step? If so, this is certainly outside of the processing chain as it is purely and analog function.

Kevin Halverson

 

[amrrahmy]

"There is already a lot of processing", i'm not saying there is no processing, i'm saying the processing can be minimal enough that one laptop can handle 6ch of audio and be a monitor at the same time. also check the signals and the waves, see any imperfections, any distortions before and during the recording, see and limit room noise(i think).

 

[amrrahmy]

seeing the sound on a monitor is very important(to me at least) not the level the actual wave.

 

[Kevin Halverson]

I guess I still am not getting the point across.

If the ADC has an additional and accurate DSP to provide the down sample, the computer will never know that this occurred upstream of it. All the displayed information (waveforms) will be identical to an ADC which lacks this feature. I am fully aware of the necessity for a good UI (user interface), this is simply a way of getting better recordings, not changing the application level software.

The computer doesn't have to do any (not even one clock cycle) more with the data being thrown at it. This is an upstream process. Your DAW application will behave EXACTLY the same.

Kevin Halverson

 

[amrrahmy]

that's exactly what i'm talking about.
u have a full painful hard on equipment process of down sampling that has some fishy delays and some processing with clocks and time comparisons made to resync happening(that would kill any processor and also requires some kind of super rams) before the difficult process that needs a powerful pc and a fast hd, equipment for the down sampling has processing in it that requires at least a couple of ships if not a pc.

 

[amrrahmy]

that is expensive
and not 6ch 1050$ expensive

u r talking specially made ship expensive

 

[amrrahmy]

u 'll need like a ps3(or 2x) wired into process to make it happen realtime without delays.
with rewiring all the gpu power to processing

 

[Kevin Halverson]

Nope, this is a real time process and it isn't "fishy" at all. There is no "resync" as it is naturally always in sync, the memory burden is fully within the X & Y memory space of the ADC's DSP section (just a few k), so no external memory is required. The number of taps is not excessive and the number of clock cycles (all single cycle MAC operations, multiply/accumulate, not a reference to Apple at all) is remarkably few given the parallel nature of the processor.

Best of all it isn't out of the price range of commercial products.

Kevin Halverson

 

[amrrahmy]

what's the price range?and how many channels?

 

[amrrahmy]

and aprx dimensions?

 

[Andrew M.]

I started to use Audition 3 for surround editing and once I installed ASIO compatible sound card in my computer I can do a lot of work real time.

http://en.wikipedia.org/wiki/Audio_Stream_Input/Output

Can you use ASIO compatible device for high end work, or it is just for standard audio editing?

Surround editing in Audition.

 

[amrrahmy]

software seems a bit week compared to logic
and editing would be pain

 

[Kevin Halverson]

Audition CS3 really is a wonderful program, I use it for a lot of standard projects (have been using it long before Adobe bought up the application from Syntrillium) and really like the integration between it and the other Adobe products in the suite.

For high end work we don't really use a DAW at all. It's all command line driven tools running under Irix on a SGI workstation. No GUI, just text, but it runs fast and we know exactly what is being done with the data.

When it comes to the SoundField work in 6.0 (3D surround including a height components) there are only a few tools out there, but I prefer to just process the data as individual fundamental signals (W,X,Y & Z) for editing purposes (treating it as a 4 channel project) and then post process the fundamentals into the delivery channel count.

For more conventional footprints (like 5.x) the commercial tools (like Audition) are more than adequate for the task.

Kevin Halverson

 

[Kevin Halverson]

what's the price range?and how many channels?

The die area is not information that I can disclose, but I can say that a 2 channel ADC fits in an 64 pin QFP. Price for the Si will be well under $80 per channel pair in manufacturing quantities.

Kevin Halverson