Hi Guys,
Thanks, no problem.
If I use a proxy, it won't ship to me. Not easily anyway. Plus I'll get charged duty in addition to taxes (which I pay anyway). I have Prime, so it would ship free normally. It isn't released yet either, so I have to wait no matter what. If it sells out immediately I'll just have to wait longer, that's all.
What's nice about a pre-order is that the publisher has a sale on now (but you can't pre order directly), and I can forget about it until "a present" shows up at the door (I love those!). At any rate, not the end of the world. Just stupid shite from living outside the USA I guess. I'll check back with Amazon later in hopes I can pre-order.
If anything, Bob can see there is interest and appreciation for his work! lol!
Edit: I just checked the US Amazon site and I can pre-order it there. $133.67 CAN for paperback. If I pre-order from the US Amazon site, I'm not sure the shipping would be free, and they may also charge me duties on top.
Thanks, no problem.
If I use a proxy, it won't ship to me. Not easily anyway. Plus I'll get charged duty in addition to taxes (which I pay anyway). I have Prime, so it would ship free normally. It isn't released yet either, so I have to wait no matter what. If it sells out immediately I'll just have to wait longer, that's all.
What's nice about a pre-order is that the publisher has a sale on now (but you can't pre order directly), and I can forget about it until "a present" shows up at the door (I love those!). At any rate, not the end of the world. Just stupid shite from living outside the USA I guess. I'll check back with Amazon later in hopes I can pre-order.
If anything, Bob can see there is interest and appreciation for his work! lol!
Edit: I just checked the US Amazon site and I can pre-order it there. $133.67 CAN for paperback. If I pre-order from the US Amazon site, I'm not sure the shipping would be free, and they may also charge me duties on top.
You cant order items listed at amazon.com when logged in at amazon.ca.
But nothing is stopping you to create an additional account at amazon.com
I am using five different accounts myself: .nl .de .fr .co.uk and .com
They will all have a different item price and postage cost. Be aware that you might have to pay import duties.
Hey, guess what? If you really poke at the Canadian site, you can find the paperback version for pre-order. I'll get that on my next Credit card cycle as it looks like I am charged immediately (you would think they would give a discount for that!).
Hi OmeEd,
Yes, but I have a US Amazon account as well. I would expect to be dinged for extra shipping and duty doing it this way.
Saved in my wish cart.
Hi OmeEd,
Yes, but I have a US Amazon account as well. I would expect to be dinged for extra shipping and duty doing it this way.
Saved in my wish cart.
Bob, I don't know what's your take on audio compressors in your book. I've always criticized Doug's book (I am sorry for the comparison, but it is bound to happen) when it comes to his chapter covering compressors in a a systems-based or black box approach without any meaningful circuits, and very superficial. Granted, dynamic processors could be the topic of a book all in itself, but I was looking at his new edition, which I received a few days ago, and there doesn't seem to be any improvement whatsoever from the previous editions regarding the audio compressors section, however, I would have to read more before I make any final judgment on it.
A lot about compressors at: https://thatcorp.com/design-notes/
If you want guitar compressor type circuits, I probably have some stuff on that somewhere or other.
If you want guitar compressor type circuits, I probably have some stuff on that somewhere or other.
In audio, compression is often a dirty word, and its abuse can greatly impair the sound. This happens with the loudness wars. In some cases compression (reduction of dynamic range) is used to better suit the intended audience, such as in automobiles or boom boxes. In some cases, less compression is used in the mastering of an SACD versus the CD (or LP) version of an album.
That having been said, there is a very important role for compression in the studio for improving and tailoring the sound of a mix. In many cases, tailored compression may be used on a per-channel basis in making a multitrack recording. Good recording engineers use various kinds of compression to improve the sound, without reducing overall dynamic range in a way that would hurt the recorded end product. Compression is also used in live broadcast situations to make speech more audible. These desired forms of compression, of which there are many, are the ones I mostly focus on in the book. The nature of the circuits for implementing the compression are also covered.
Cheers,
Bob
Two things about DACs worth noting. The first relates to clock jitter and some of its origins. Clock recovery from the Manchester-encoded SPDIF input is often done in the digital audio interface chip (DAI) that precedes the DAC chip with a phase-locked loop (PLL). A PLL is a feedback circuit whose loop gain is usually influenced by the transition density in the input data. The gain of the phase detector is usually dependent on the transition density. If there is a static error, its magnitude is reduced by the amount of loop gain in the PLL. Virtually all digital data streams have data-dependent transition density. This can lead to clock jitter if an error signal is required to move the VCO free-running frequency to the exact frequency of the incoming data. This effect depends on the details of how the DAI chip recovers clock and the quality of its PLL. A sloppy on-chip VCO creates more clock jitter. A precise crystal-controlled VCO (VCXO) with adequate control range can greatly reduce this effect. The quality and behavior of the DAI chip really matters. In some DACs, there are other, more sophisicated ways to handle clocking of the DAC, some of which clock the DAC with an ultra-low jitter fixed-frequency crystal oscillator and use special means to deal with the resulting asynchronisity between the crystal clock and the incoming data frequency.
The second issue worth considering is the matter of "intersample overs". At a source sample rate of 44.1 or 48 kHz, for example, it is possible for the original analog input to have a peak that lies between samples that is higher in amplitude than its two adjacent samples. A full-scale source signal can thus contain a larger amplitude peak than the highest sample value. That will not overload (clip) some of the digital representations in the DAC, but after proper reconstruction, those higher peaks may exist in the digital domain in some parts of the DAC processing. This effect can be seen with a full-scale sine wave at 1/4 the sample rate. This is a subtle effect that does not occur often, but should be considered. A good explanation of this and cures for it can be found on the Benchmark Media website.
Interestingly, intersample overs are usually not a problem in non-oversampled DACs where all of the reconstruction is done by an analog filter.
An ordinary digital peak program meter (PPM) will often just look at the highest digital values. "True-peak meters" are often used to determine the inter-sample peak value to within an acceptable margin of error by looking at the largest digital peak values in a properly-oversampled version of the digital signal, where the oversampling ratio (OSR) is perhaps 8:1 or more.
Cheers,
Bob
Purists hate compression, but musicians love it. Aside from, maybe, classical music, all modern music (ab)uses compression. There are several architectures, FET, VCA, opto, etc... But it seems that most of the modern app notes focus on VCA applications, probably because they want to sell you a VCA and not a FET, and opto cells are many times custom-made, aside from the opto cell of an LA2A or something similar, which you may find on the internet.
I do not necessarily agree with your statement implying that good recording engineers do not abuse compression, although, one could make an argument for what a good audio engineer is in the first place. I can tell you that an audio engineer who refuses to use too much compression in modern music is, most likely, an unemployed engineer. Chris Lord-Alge (and his brother Thom) are notorious for their excessive use of compression, they probably are (definitely were at some point) the most coveted audio mixers of the music industry, mixing thousands of records, winning several Grammys and earning the big bucks. I can guarantee that if you've listened anything from the 2000s till now, you have listened more than once to something these guys made. These are two examples but they are not the only exception. On top of that, mastering engineers completely destroy the sound by excessive compression and limiting, leading to what has been known as the 'Loudness Wars', although since Spotify and other streaming services started normalizing all the audio to -14 LUFS or thereabouts (the user can deactivate normalization but it is on by default, and the majority of regular users won't even know or care about it) this has lead to less compression abuse by mastering engineers, but the problem of excessive compression/limiting still remains. However, 99% of modern music is heavily compressed or has used heavy compression as an effect at some point of the production process.
I don't want to turn this into an argument about whether excessive compression is good or bad, rather, the point I am trying to make is that it is essential. Also, I don't care about the discussion about whether audio compression itself is good or bad, because compression use or abuse is a subjective issue, and I don't care too much about subjective issues. My goal is circuits.
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Hi Bob,
Agreed, consumer compressors are often poorly executed and I cringe working on some dBx products from the 70's and 80's. Although, dBx on tape machines can be very effective as long as the calibration is spot on. Dolby SR was pretty amazing.
In the studio, peak limiters (over-easy type) compressors are needed with digital recording. There is no soft limiting as there is with tape machines and you can't get that "sound" by running in the red. With digital, when you're out of little ones and zeros', you're out and hard. Live dynamic signals easily exceed the dynamic range of recording media. Of you set your zero levels +20 dB, you have lost that resolution down lower even at 32 bit. You'll have to limit those peaks down the chain somewhere anyway.
For automotive and similar environments, reduced dynamic range is preferred from my standpoint. If the car is stopped, off and you're listening in a quiet area, then it would be nice to return to a full dynamic range.
Agreed, consumer compressors are often poorly executed and I cringe working on some dBx products from the 70's and 80's. Although, dBx on tape machines can be very effective as long as the calibration is spot on. Dolby SR was pretty amazing.
In the studio, peak limiters (over-easy type) compressors are needed with digital recording. There is no soft limiting as there is with tape machines and you can't get that "sound" by running in the red. With digital, when you're out of little ones and zeros', you're out and hard. Live dynamic signals easily exceed the dynamic range of recording media. Of you set your zero levels +20 dB, you have lost that resolution down lower even at 32 bit. You'll have to limit those peaks down the chain somewhere anyway.
For automotive and similar environments, reduced dynamic range is preferred from my standpoint. If the car is stopped, off and you're listening in a quiet area, then it would be nice to return to a full dynamic range.
Yeap, however, that dBx 160 (particularly the A or X models from back in the day, not so much the new re-issue) is still a very much coveted compressor by the audio/music community; people like using it on drums and on hip-hop vocals (don't know if they are still using it for this purpose, but at some point it became a favorite amongst some hip-hop artists). I sold mine a few years ago on eBay and the ad didn't last too long before someone grabbed it. Jensen transformers makes a transformer specifically for it, in order to have transformer balanced inputs. I installed the transformer in my unit and it did help improve CMRR.
The 161 from the 70s still sells used for like 2K or more, and I consider it complete garbage....
The 161 from the 70s still sells used for like 2K or more, and I consider it complete garbage....
We know. Anyone here thinking of using simple clock recovery for SPDIF instead of using ASRC or FIFO buffering usually tends to be a beginner. Iancanada has been selling diy FIFO buffer boards for years. Andrea Mori at TheWellAudio also has a very good one, along with SOA clocks. Regarding ASRC with crystal reference, @werewolf wrote an interesting thread explaining all about how they work: https://www.diyaudio.com/community/threads/asynchronous-sample-rate-conversion.28814/post-333645
Also, we know there are a number of free and paid wav file analyzers that show intersample overs and perceptual volume levels according to the ITU standard. Of course, intersample overs are potentially worst at low sample rates at Fs/4.
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Hi AnalogJoe,
Oh, for sure! But we all know that what is popular and deemed "great" has zero bearing on actual performance.
All our music these days has already been stomped on, so maybe an old idea (peak unlimiter) in the digital realm might be a happening thing in the future. Older recordings could use it. It would be nice if they could somehow get rid of early tape hiss, being random broadband noise, it would require AI to do, and a lot of processing power!
Hi Mark,
Yes, but one thing we do not need are ovenized clocks or anything like that. You only need to reduce/eliminate short term frequency shifts. So power supplies (within reason) and stuff like that might be helpful. A standard, decent x-tal clock more than covers things. It is one of the more stable oscillators out there.
Oh, for sure! But we all know that what is popular and deemed "great" has zero bearing on actual performance.
All our music these days has already been stomped on, so maybe an old idea (peak unlimiter) in the digital realm might be a happening thing in the future. Older recordings could use it. It would be nice if they could somehow get rid of early tape hiss, being random broadband noise, it would require AI to do, and a lot of processing power!
Hi Mark,
Yes, but one thing we do not need are ovenized clocks or anything like that. You only need to reduce/eliminate short term frequency shifts. So power supplies (within reason) and stuff like that might be helpful. A standard, decent x-tal clock more than covers things. It is one of the more stable oscillators out there.
Hi Chris, I believe you are sincere. However, I can tell the difference between average and SOA clocks on my system. So can other people. So, I guess its something we may never agree on. Regarding ovenized clocks, we agree they are not needed for audio. Ultra-low close-in phase noise does matter however, IME. It just so happens that my Acko Lab SOA clocks are TCXO though. OTOH my Andrea Mori clocks are not TC.
You are right, the fanatic zeal for all things old in the pro audio/music community is, most of the times, ridiculous. And popular or great means nothing. How many 4-transistor guitar distortion pedals are being sold for over $500? (there was even a documentary IIRC, which covered this issue, it was called "Fuzz: The Sound that Revolutionized the World"). Regarding the 160, I was just making an observation about it, that is all, I am not trying to defend it or anything.
Hi Mark,
I bet we do agree. You see, when you say clock you are referring to the entire clock system. I am referring to the core, or heart of that system. The most spectrally pure clocks we have are crystal clocks, SC cut. Even an AT cut crystal is more than sufficient for your needs. These are incredibly stable by nature, the circuit may not be but the oscillating element is far more stable than you could ever hear no matter what.
What you are hearing can be related to power supply noise and any number of possible clock tuning issues we use to discipline the crystal to the clock we want to lock to. Then there is your FIFO memory and physical implementation.
So many things are at play here you can't comment on one single thing. That and you are talking about what you hear and perceive, so all kinds of other stuff is happening as well. Completely uncontrolled.
I have studied these systems using an HP 5372A. You need to look that thing up and understand what it is. It has the high stability oscillator option as well.
I bet we do agree. You see, when you say clock you are referring to the entire clock system. I am referring to the core, or heart of that system. The most spectrally pure clocks we have are crystal clocks, SC cut. Even an AT cut crystal is more than sufficient for your needs. These are incredibly stable by nature, the circuit may not be but the oscillating element is far more stable than you could ever hear no matter what.
What you are hearing can be related to power supply noise and any number of possible clock tuning issues we use to discipline the crystal to the clock we want to lock to. Then there is your FIFO memory and physical implementation.
So many things are at play here you can't comment on one single thing. That and you are talking about what you hear and perceive, so all kinds of other stuff is happening as well. Completely uncontrolled.
I have studied these systems using an HP 5372A. You need to look that thing up and understand what it is. It has the high stability oscillator option as well.