Digital Technology Collection

Discussion in 'Digital Sources' started by Negotiableterms, Mar 28, 2006.

  1. Negotiableterms

    Negotiableterms Administrator Staff Member Admin Subscriber

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    The purpose of this thread is to collect links, articles, whatever, on how digital sound works. When the collection is big enough, I'll rearrange things into a coherent thread that can be used as a reference by anyone wanting to understand things like quantization, dither, jitter, etc.

    Here's an example of the kind of thing we're looking for, which is a fairly good coparison article on SACD and DVD-A:

    http://www.digit-life.com/articles2/sacd-dvd-a/index.html

    Everyone, please join in. If you have or find something good, POST!
     

     

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  2. Negotiableterms

    Negotiableterms Administrator Staff Member Admin Subscriber

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  3. Negotiableterms

    Negotiableterms Administrator Staff Member Admin Subscriber

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  4. ozmoid

    ozmoid Lunatic Member

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    "Notes on the Troubleshooting and Repair of Compact Disc Players and CDROM Drives"

    http://www.repairfaq.org/sam/cdfaq.htm

    CD players, how they interact with the CD (very long, technical, not too dry), what breaks, what you can fix yourself. LOTS more information than just troubleshooting.
     
  5. guiller

    guiller Toscaninichus Australis

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    Thanks for the sharing!
     
  6. Andyman

    Andyman Scroungus Stereophilus Subscriber

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    Comes up "Forbidden" on my phone just now.
    Dead link?
     

     

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  7. Negotiableterms

    Negotiableterms Administrator Staff Member Admin Subscriber

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    Works fine for me. It's a great post.
     
  8. hbd85

    hbd85 Active Member

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    same here with my PC and google chrome
     
  9. chicks

    chicks Lunatic Member

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  10. Joe Dawson

    Joe Dawson Active Member

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    The video has some problems. First, a scope won't reveal if the sine wave is "exactly" and/or a "perfect" sine wave. For instance, if the fundamental frequency is altered slightly, the scope will never reveal it. Even an amp with 5% or more harmonic distortion will be quite difficult to see on a scope. Now consider an entire orchestra, singing. Evidently, we must believe what he says.

    If other frequencies are introduced, the scope will never reveal it. What do I mean by that? The distortion analyzer shows nothing but the fundamental frequency and harmonics. If the fundamental frequency is slightly altered, it will not be seen.

    When dealing with bit depth, "stair steps" (16 bit, 65,536 values, 24 bit, we have 16,777,216 values), very rarely is the analog signal going to be exactly on a "value"/"step" during the sample period. The signal will be in between, so which value is chosen, an upper or lower value? Whichever value is chosen, the slope/rise time is altered between samples by definition, thus the fundamental frequency is also slightly altered between samples, said instrument won't reveal to us.

    We won't have to worry about the slope exceeding 20khz unless the harmonic is quite high in frequency. Eight bit alters the slope/rise time even more, plus 8 bit lacks dynamics, inner detail even more than 16 bit. The inference that 8 bit, even 16 bit is enough for high quality music is the opposite of what Philips engineers believed, but RCA marketed 16 bit players anyway, and the rest is history.

    Notice the Gibb's effect. It is within 20khz which is to be expected, Notice its amplitude value is high compared to the rectangular wave. Any Intermodulation distortion in the system, whether it be from speakers, electrical components, will cause mixing with this ringing, and cause non musical tones in the audio band.

    It takes two samples to recreate a sine wave. At 10khz, there are only 4 samples per cycle, 5khz only 8 samples. However, music is not a simple sine wave nor a rectangular wave with equal repetitive waveforms. Music is complex with all sorts of phase relationships between instruments and their waveforms. Think it can reproduce the music perfectly when parts tolerances enter the picture and values are altered?

    Ever notice the "digititus" I have heard it called, especially at higher frequencies. A grain, roughness, grittiness etc, cymbals don't sound right, sounds like glass breaking. Attempting to claim analog is 13 bit is not entirely accurate, as the needle follows the grooves in a continuous motion, not just periodically sampling. 44.1k means a cutoff of ~20khz, yet study after study demonstrates that each "ear" can perceive to at least 5us (5 microseconds) rise time differences, in layman's terms comes across as attack times.

    16 bit may be good enough for one's average system, but some want better, with both higher sampling rate and higher bit, wider bandwidth. So why be against higher quality, memory is easily available and cheap.

    Now I like digital and use it often, convenient, but I use 20 bit or higher if possible.

    Continue on with your discussion.

    keep on truckin
    joe
     
    Last edited: Nov 25, 2017
    cpt_paranoia likes this.
  11. chicks

    chicks Lunatic Member

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    Um, no. You need to educate yourself. Did you even pay attention to the video?

    Much of the technology used for lossless compression and streaming was invented by Monty. I assure you he knows his stuff FAR better than you do. Ever hear of FLAC? Monty.
     

     

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  12. Joe Dawson

    Joe Dawson Active Member

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    Your reply demonstrates your lack of understanding of the science stated in my post. As such, we see no civilized argument in your defense.

    Continue on sir.

    keep on truckin
    joe
     
    Last edited: Oct 27, 2017
  13. cpt_paranoia

    cpt_paranoia Well-Known Member

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    His discussion of the stepless output is based on pure sampling theory, where each sample is a zero-width Dirac delta function (his 'lollipop diagram').

    That's correct in theory, but in the real world, a DAC does implement the zero-order hold he mentions, and the output of the DAC chip DOES have steps; it does not produce zero-width delta function outputs. A DAC chip is therefore followed by a Nyquist reconstruction filter, to remove the steps.

    A real-world ADC will also generate a sampling (or quantisation) error, as, unless you have infinite bits, there will be an error between the real sample, and the nearest ADC quantising value. His lollipops are assumed to be perfect samples, with zero quantisation error.

    Sampling theory assumes perfect, 'brick-wall' Nyquist filters. In the real world, these do not exist. Real filters have problems like roll-off rates, and ripple in passband amplitude & phase.

    I was once asked to look at the design of a digital radio transceiver. They were having trouble with harmonics in the transmitted RF, violating the spectrum mask. It turned out to be the the use of a 4-bit, 12 sample per cycle sine wave generator used in the digital IF mixer (implemented in an FPGA). The quantisation error caused by the 4 bit samples was causing the harmonic distortion. I increased the size of the samples, which allowed us to reduce the harmonics to an acceptable level. The analysis of the problem needed nothing more than a look at the VHDL code to understand the sample scheme, and creating an excel spreadsheet to create a set of repeated cycles, on which I got excel to compute an FFT, which showed the harmonics.

    It would have been illuminating for him to have changed the precision of his samples, from the 16 bits he used, to 8 bits, or even 4 bits. Using 16 bits, the quantisation error will be below the noise floor of the analyser he was using. It's not a good idea to try to claim an effect doesn't exist because you can't measure it.

    Much of the effort expended in the design of DACs has been to try to eliminate that zero-order hold step problem; to smooth the transitions in some way. A first order interpolator would draw a straight line between the samples, and higher order interpolators would try to draw a smooth curve. The most common approach is a digital oversampling filter, which generates additional samples to fill in the gaps, with a higher precision DAC; e.g. a 4x oversampling DAC, generating 176.4kSa/s, 18-bit samples. Then there are the 'noise-shaping' DACs, such as Bitstream and MASH, that use an entirely different approach to reconstruction, with a stream of very high rate pulses; they push the noise way above the Nyquist frequency, allowing simpler, more linear filters to be used.

    I started my career working on the development of the GSM standard, and the first network and handsets. In particular, the frequency synthesis and modulation. We used a technique called Digiphase, a type of fractional-N synthesiser. It did direct digital modulation by constantly changing the synthesiser frequency. It used a third-order interpolator, combined with digital predistortion to meet the modulation and spectral mask requirements. Essentially, a noise-shaping DAC.
     
    Last edited: Nov 25, 2017

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