Author Topic: A/D, D/A and DSP  (Read 2469 times)

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Offline rascal101

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A/D, D/A and DSP
« on: Apr 27, 2006 at 10:09 AM »
The A/D, D/A conversion process does not require programming although the DSP chips are programmable. It is possible to put a DSP circuit block inside the A/D or D/A chip for signal processing - this is the one that can be programmed.

A DSP cannot perform the functions of a filter due to semiconductor materials limitation. Capacitance is achieved by sandwiching two conductive elements with an insulator and inductance is achieved by a zig zag but will result in very low inductances - more like leakage inductance.

Since die sizes are limited in length and height, small values for caps in the pF range and pH, respectively is possible. This is clearly unsuitable for audio use. Also, DSP ICs are limited in the input current and output current it can source or sink.

Offline av_phile1

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Re: A/D, D/A and DSP
« Reply #1 on: Apr 27, 2006 at 10:18 AM »

A DSP cannot perform the functions of a filter due to semiconductor materials limitation. Capacitance is achieved by sandwiching two conductive elements with an insulator and inductance is achieved by a zig zag but will result in very low inductances - more like leakage inductance.

Since die sizes are limited in length and height, small values for caps in the pF range and pH, respectively is possible. This is clearly unsuitable for audio use. Also, DSP ICs are limited in the input current and output current it can source or sink.

How does a flagship receiver perform variable or selectible crossover frequencies or filtering in its preamp section  for the bass management function?  Is it performed outside of the DSP?  Is the bass management module a separate chip or can it also be included in one chip together with other digital sound processing functions as well as DAC? 
« Last Edit: Apr 28, 2006 at 12:28 PM by av_phile1 »

Offline rascal101

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Re: A/D, D/A and DSP
« Reply #2 on: Apr 27, 2006 at 10:31 AM »
I believe this achieved by having several external filters on the ready and the DSP chip accessing them via 3 state semiconductor switches. The DSP chooses which filter it needs.

Offline av_phile1

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Re: A/D, D/A and DSP
« Reply #3 on: Apr 27, 2006 at 10:59 AM »
My understanding is that bass management function is achieved in the digital domain. The filtering is done digitally. 

Offline rascal101

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Re: A/D, D/A and DSP
« Reply #4 on: Apr 27, 2006 at 11:07 AM »
I'm not sure about bass management function being done digitally. To me, it is much easier to have analog filters and have the DSP access them. The ICs I listed in previous thread are old but they are capable of producing complex sounds.
« Last Edit: Apr 27, 2006 at 11:21 AM by rascal101 »

Offline av_phile1

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Re: A/D, D/A and DSP
« Reply #5 on: Apr 27, 2006 at 12:05 PM »
Bass management is typically done in the digital domain.  That is why early HT recievers could not handle the complex algorythms needed to do bass management for SACD when they first came out.  Until now very few receivers can do bass management for SACD.  Except for a few flagship reveivers that must often convert the immense DSD bitstreams into 5-channel PCM before they can bass manage, like what Denon does.   Most bass management on the SACD are being done on the player while the signals are still digital.  Because once they pass thru the player's D/A converter and into the receiver's 5.1 analog inputs (SACD can only be played via the analog 5.1 connection, unless the player and receiver has proprietary D-link or I-link Firewire connection) the receiver assumes that the 5.1 analog signals have already been digitally processed and no further digital processing is done.   

If you use an audio editing software on the PC, like Goldwave, Nero or Soundforge, you'd realize how easy, effective and efficient it is to manipulate audio signals in the digital domain.  And this manipulation is all done courtesy of the software programs I mentioned plus many more.  You can equalize parametrically, filter out frequencies like the voice,  shift keys and pitch, optimize dynamics, simulate stereo out of mono, interchange channels, remove DC offset, subtract Left from Right signals, sum them up, etc, etc all in the digital domain.  Much of these can also be done in receivers.  Those chips in receivers virtually make the modern receiver a computer in itself, using software programs hardcoded into firmwares that may or may not be upgradable, depending on the price points.  Certainly flagship receiver firmware can be upgraded via an RS232 port like in an Onkyo or thru a CD-R playback thru its S/PDIF port.

Whatever you can do with analog signals, you can more effectively and efficiently do once you convert analog to digtal first and manipulate the signals in the digital domain.  Thanks to software programming.   Those DD and DTS Decoders, DSP and D/A A/D converters were first done in computers using programs that manipulate the digital bitstreams.  Receivers today simply inherited the advances made in the computer world. 
« Last Edit: Apr 27, 2006 at 12:30 PM by av_phile1 »

Offline rascal101

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Re: A/D, D/A and DSP
« Reply #6 on: Apr 27, 2006 at 12:15 PM »
Ok I'll review a DSP chip and have comments in the next few days ...

Offline Voltraizer

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Re: A/D, D/A and DSP
« Reply #7 on: Apr 27, 2006 at 06:20 PM »
Bass management is typically done in the digital domain.  That is why early HT recievers could not handle the complex algorythms needed to do bass management for SACD when they first came out.  Until now very few receivers can do bass management for SACD.  Except for a few flagship reveivers that must often convert the immense DSD bitstreams into 5-channel PCM before they can bass manage, like what Denon does.   Most bass management on the SACD are being done on the player while the signals are still digital.  Because once they pass thru the player's D/A converter and into the receiver's 5.1 analog inputs (SACD can only be played via the analog 5.1 connection, unless the player and receiver has proprietary D-link or I-link Firewire connection) the receiver assumes that the 5.1 analog signals have already been digitally processed and no further digital processing is done.   

If you use an audio editing software on the PC, like Goldwave, Nero or Soundforge, you'd realize how easy, effective and efficient it is to manipulate audio signals in the digital domain.  And this manipulation is all done courtesy of the software programs I mentioned plus many more.  You can equalize parametrically, filter out frequencies like the voice,  shift keys and pitch, optimize dynamics, simulate stereo out of mono, interchange channels, remove DC offset, subtract Left from Right signals, sum them up, etc, etc all in the digital domain.  Much of these can also be done in receivers.  Those chips in receivers virtually make the modern receiver a computer in itself, using software programs hardcoded into firmwares that may or may not be upgradable, depending on the price points.  Certainly flagship receiver firmware can be upgraded via an RS232 port like in an Onkyo or thru a CD-R playback thru its S/PDIF port.

Whatever you can do with analog signals, you can more effectively and efficiently do once you convert analog to digtal first and manipulate the signals in the digital domain.  Thanks to software programming.   Those DD and DTS Decoders, DSP and D/A A/D converters were first done in computers using programs that manipulate the digital bitstreams.  Receivers today simply inherited the advances made in the computer world. 
I agree. The advancement in technologies and software programming is really influencing the modern designs of
most electronic gadgets.
May I add,      Since there's no dispute that CD is already in digital form. DSP techniques can be applied as compared to let's say analogue signal where you need ADC.
     Before, unwanted electrical noise is removed via filter circuit. With the advent of digital programming
nowadays, the DSP software program takes care of filtering signals to improve signal quality.
   As I said before in another thread, First audio consumer products to take advantage of this Program
Implementation is Celestion DLP-600 whereby sofware program acts as high pass and low pass filter
replacing capacitors, resistors and inductors in loudspeaker cross-over.
  So I believe 0 sampling proponents that Upsampling Theory  isn't really what's being implemented
in 20 bit 192kHz CD players since with Software programs, Digital datas are manipulated and
software programmer will keep their source code in secrecy. "Black or Gray ART" as commented by most audio reviewers. 
At most, this are only marketing hypes made out of jargon such as "24 bit CD player" when in reality the last four digits are marketing bits and in reality it's only "20 bit CD player" and your redbook CD '16 bit'.
Most significant improvement in Upsampling Cd players are as you said Bass Management. Improved low level frequency and improved soundstaging is really noticeable which I suspect
is a result of Sofware Programming  parang the much maligned anti-audiophile digital equalizer effect.
This chips are popular now in usage in cellphone, video cameras and other gadgets .
« Last Edit: Apr 27, 2006 at 06:28 PM by Voltraizer »
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Offline rascal101

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Re: A/D, D/A and DSP
« Reply #8 on: Apr 27, 2006 at 07:01 PM »
With digital data it is possible to remove data that are in excess of 20KHz through a digital filter, yes. I believe this should be fairly easy. However after converting digital data to analog data do you have a guarantee that frequencies greater than 20KHz are no longer present? If you look at circuit topologies today you still have an analog low pass filter. You can't eliminate that.

You still have to account for component noise, layout noise etc. How do you deal with that? These are all unwanted, how does a DSP or digital filter deal with this? DSP or digital filter deal more on the data from the CD or DVD not component or layout noise. You have it wrong when you say that digital filters or DSP can take care of this.

Digital filters existed even before the advent of upsampling DACs and that's not going to change soon however, the analog filters won't be leaving either.

As far as bass management and what have you there were already ICs in the 80s that deal with this. This is nothing new. And mind you, converters were first used in data acquisition systems eg measurement systems.

Offline rascal101

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Re: A/D, D/A and DSP
« Reply #9 on: Apr 28, 2006 at 09:27 AM »
Sir Avphile, Voltraizer,

Would greatly appreciate if you can help me by opening up your receivers and identifying the DSP and D/A chips. Or if possible by posting pictures. This way we can analyze how the receiver functions. TIA.

Offline av_phile1

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Re: A/D, D/A and DSP
« Reply #10 on: Apr 28, 2006 at 12:26 PM »
I have no problem with your request, rascal101.  Would be most happy to oblige.  I don't use a receiver but preamp/processors and will see over the weekend if I can take them out of the rack and take pics.  Then I'll have to get to a photo site.  It seems ages ago when I did.

Offline av_phile1

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Re: A/D, D/A and DSP
« Reply #11 on: Apr 28, 2006 at 03:12 PM »
With digital data it is possible to remove data that are in excess of 20KHz through a digital filter, yes. I believe this should be fairly easy. However after converting digital data to analog data do you have a guarantee that frequencies greater than 20KHz are no longer present? If you look at circuit topologies today you still have an analog low pass filter. You can't eliminate that.

The analog after-DAC filters are meant to clean-up, apart from recover, the 20khz audible passband which contains a lot of digital artifact noise  as a result of the digital processing.  But depending on design, some 0-oversampling players actually omit this stage and let ultrasonic artifact pass through.  Afterall, the human ear has its own brickwall filter.   And some swear it sounds better that way. 

Quote
You still have to account for component noise, layout noise etc. How do you deal with that? These are all unwanted, how does a DSP or digital filter deal with this? DSP or digital filter deal more on the data from the CD or DVD not component or layout noise. You have it wrong when you say that digital filters or DSP can take care of this.

Every electronic component device (resistors, caps, etc.) has its signature noise.  There's really nothing much we can do except to design the entire circuit board in such a way as to reduce these inherent electrical noises to inaudible levels.  Which is really part of intelligent design. 


« Last Edit: Apr 28, 2006 at 03:16 PM by av_phile1 »

Offline Voltraizer

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Re: A/D, D/A and DSP
« Reply #12 on: Apr 28, 2006 at 09:19 PM »
<snp>

The following are not my own , I just cut and paste from a source in the internet:
Advantages of digital filters over analog filters:
The following list gives some of the main advantages of digital over analog filters.

A digital filter is programmable, i.e. its operation is determined by a program stored in the processor's memory. This means the digital filter can easily be changed without affecting the circuitry (hardware). An analog filter can only be changed by redesigning the filter circuit.
Digital filters are easily designed, tested and implemented on a general-purpose computer or workstation.
The characteristics of analog filter circuits (particularly those containing active components) are subject to drift and are dependent on temperature. Digital filters do not suffer from these problems, and so are extremely stable with respect both to time and temperature.
Unlike their analog counterparts, digital filters can handle low frequency signals accurately. As the speed of DSP technology continues to increase, digital filters are being applied to high frequency signals in the RF (radio frequency) domain, which in the past was the exclusive preserve of analog technology.
Digital filters are very much more versatile in their ability to process signals in a variety of ways; this includes the ability of some types of digital filter to adapt to changes in the characteristics of the signal.

Fast DSP processors can handle complex combinations of filters in parallel or cascade (series), making the hardware requirements relatively simple and compact in comparison with the equivalent analog circuitry.



« Last Edit: Dec 22, 2010 at 09:56 AM by Voltraizer »
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Offline rascal101

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Re: A/D, D/A and DSP
« Reply #13 on: May 01, 2006 at 02:53 AM »
True. Digital filters have advantages over their analog counterparts and there's nothing like doing things in the digital domain.

About drift, the article did not mention that integrated circuits are also subject to drift. Probably the stored data may not be subject to drift however, the performance of the chip is. It still has to source or sink current and subject to the stability of the power supply. Regarding life, chips will mostly outdo electrolytic capacitors. However, it is reasonable to assume that the other passive components eg resistors and inductors will outdo chips. But were talking of >20 years or so. Clearly, beyond the typical life of an appliance. Look up the reliability bathtub curve.

I have no problems with both technologies and will implement any one depending on the application. I would go on the agree that yes, we will probably see more and more of DSP and digital filters as we progress in time. Even with crossovers, it is becoming cheaper by the minute. Just didn't realize it earlier.

Offline Voltraizer

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Re: A/D, D/A and DSP
« Reply #14 on: May 01, 2006 at 03:09 AM »
I agree.
Hey _rascal101. First time we're in agreement.   :) ;)
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Offline eRaSeR

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Re: A/D, D/A and DSP
« Reply #15 on: May 01, 2006 at 05:20 AM »
 ;D ;D ;D ;D ;D

Offline rascal101

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Re: A/D, D/A and DSP
« Reply #16 on: May 02, 2006 at 10:01 AM »
Oo nga. Nalimutan ko kasi iyung digital filters. Tagal na siya pero di ko pinapansin ...

Offline av_phile1

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Re: A/D, D/A and DSP
« Reply #17 on: May 02, 2006 at 10:22 AM »
Av_phille, (now that you mentioned it) Do you believe the 0 sampling protaganist when they said that filtering
is best left to the acoustic transducers: loudspeakers and the ear because the noise 
is well beyond what we can hear or what most speakers can reproduce? That's why
I  ordered an obsolete CS8412 digital audio interface receiver IC and TDA 1543 dual
from www.fururlec.com para ipabubuo ko na cloned 47 lab digital filterless 0 sampling
DAC to experience what the sound is.



There'a a practical design consideration why analog filters are employed to clean and recover the 20khz passband.  High frequencies in the input are said to cause some if not majority of amplifiers to become unstable.  So rather than risk it, and since they are after the mass market, CD designers employ analog filters after D/A conversion.  However, because of the steepness of such filters, requiring at least a 90db attention between the narrow band between 20khz and the 22.05khz Nyquist limit, some phase shifts and smearing occurs.  Oversampling solves this problem neatly and requires only a gentle first order analog filter after the DAC.

But because the human ear has its own built-in brickwall filter after 20khz (older people after 16khz) and that most commercial speakers also cannot reproduce beyond 22khz anyway, an analog filter may be dispensed with.  But not all amplifers are said to welcome this.  For those with amps that do, some sonic improvements are noticed.

Offline rascal101

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Re: A/D, D/A and DSP
« Reply #18 on: May 02, 2006 at 11:12 AM »
I believe you are referring to compensation. For stability, this is required specially if you are using SS with lots of feedback. And, referring to upsampling rather than oversampling.

You can't produce frequencies > 20KHz yes but in the time domain frequencies > 20KHz ride through low frequencies thus altering the signal waveforms. The digital filters do not function after the gain stage at the analog section that's why noise is still an issue. The low pass filters do not completely filter out noise. In fact, there are frequencies up to about 100KHz. Though low in magnitude they are enough to make your listening 'fatiguing'. This is probably due to layout and noise contributed via your amp. Nowadays the earth for your CDP is not the same as the amp. Unlike those tt days where you connect the audio source and amp. And, there's no guarantee that amp ground is of the same magnitude as CDP ground.