New Berkeley DAC? This article implies as much but ... ?

[edorr]

1. Use a Lynx or RME card in a desktop PC with a DAC that sends word clock out to the card. The Lynx or RME software is set to accept incoming word clock. The playback software is usually unaware of the configuration. A major pit fall can be manual word clock adjustments when switching sample rates.

 

Lynx advises against using external clock with their card. Not sure if this applies to using clock output from DAC

 

 

Note regarding async v. word clock etc... Based on my experience with many async USB DACs and systems that use external word clock, I think a state of the art externally clocked system can outperform async USB. For example, a dCS Vivaldi & Aurender W20 combination sounds best using AES and external clock. Both systems have state of the art USB implementations for comparison.

 

MSB I2S Pro sends DAC clock to source (slave) over I2S, which is supposed to sound better than using the asynch USB input on the DAC.

[The Computer Audiophile]

Lynx advises against using external clock with their card. Not sure if this applies to using clock output from DAC.

Do you have a source for that? I've talked to Lynx previously and never heard that.

[edorr]

Do you have a source for that? I've talked to Lynx previously and never heard that.

 

I own a lynx card and asked their technical support. Here is the email thread:

 

From: [email protected]

To: Erik Dorr

Subject: RE: AES16 card

Date: Tue, 2 Jul 2013 10:03:23 -0700

 

 

Erik,

 

Good question. Easy answer. No.

 

Any external clock will add jitter to the AES16e card. The low jitter SynchroLock clock on the AES16e is the best way to go.

 

We have done several tests here, clocking our products to external word clocks, including the very high end Antelope unit. In all cases, the jitter was increased.

 

Thanks,

 

Phil Moon

 

Lynx Studio Technology

 

190 McCormick Avenue

Costa Mesa CA 92626

Phone: 714-545-4700 x 204

Fax: 714-545-4777

 

 

From: Erik Dorr

Sent: Tuesday, July 02, 2013 9:30 AM

To: Phil Moon

Subject: RE: AES16 card

 

Phil, I have my AES16e card up and running in a HTPC application, feeding straight into the DACs. My question is do you think SQ would materially benefit from a reasonably priced external word clock, such as the Antelope OCX?

 

Isochrone OCX | Antelope Audio

[Audio_ELF]

It's good to hear an argument that is based on experience of specific equipment rather than just theory...

 

As I've commented before there are many ways to skin a cat ... Or play back your audio!

Hi VandyMan - Good question. Word clock is a strange concept to most computer audiophiles.

 

Two common ways to use word clock with computer audio.

 

1. Use a Lynx or RME card in a desktop PC with a DAC that sends word clock out to the card. The Lynx or RME software is set to accept incoming word clock. The playback software is usually unaware of the configuration. A major pit fall can be manual word clock adjustments when switching sample rates.

 

2. Using an Aurender W20 that has dual AES output and word clock input with a DAC that sends out word clock and accepts dual AES input (dual AES not required though). I just used this configuration at the Magico factory (Aurender W20 with dCS Vivaldi). The Vivaldi (Master) sends word clock to the W20 (Slave). This configuration is all automatic. No manual switching of sample rate required.

 

Note regarding async v. word clock etc... Based on my experience with many async USB DACs and systems that use external word clock, I think a state of the art externally clocked system can outperform async USB. For example, a dCS Vivaldi & Aurender W20 combination sounds best using AES and external clock. Both systems have state of the art USB implementations for comparison.

[The Computer Audiophile]

I own a lynx card and asked their technical support. Here is the email thread:

 

From: [email protected]

To: Erik Dorr

Subject: RE: AES16 card

Date: Tue, 2 Jul 2013 10:03:23 -0700

 

 

Erik,

 

Good question. Easy answer. No.

 

Any external clock will add jitter to the AES16e card. The low jitter SynchroLock clock on the AES16e is the best way to go.

 

We have done several tests here, clocking our products to external word clocks, including the very high end Antelope unit. In all cases, the jitter was increased.

 

Thanks,

 

Phil Moon

 

Lynx Studio Technology

 

190 McCormick Avenue

Costa Mesa CA 92626

Phone: 714-545-4700 x 204

Fax: 714-545-4777

 

 

From: Erik Dorr

Sent: Tuesday, July 02, 2013 9:30 AM

To: Phil Moon

Subject: RE: AES16 card

 

Phil, I have my AES16e card up and running in a HTPC application, feeding straight into the DACs. My question is do you think SQ would materially benefit from a reasonably priced external word clock, such as the Antelope OCX?

 

Isochrone OCX | Antelope Audio

Very interesting. Thanks for proving the info.

 

I won't say I've heard the opposite about SynchroLock from the same person, but a couple years ago I heard the opposite second hand.

[wkhanna]

It's good to hear an argument that is based on experience of specific equipment rather than just theory...

 

As I've commented before there are many ways to skin a cat ... Or play back your audio!

I hope to be providing some observations soon as i am implementing a Mutec Mc 3+ utilizing a 10MHz Rubidium oscillator. This is rather off-topic, though & will be addressed in a future thread.

[Miska]

Problem with word clocks is that it needs frequency multiplier to produce master clocks for modern delta-sigma DACs. And that multiplier tends to also multiply any jitter by the same amount...

 

So better is external master clock from something like M2Tech Evo Clock (super clock & master clock outputs). The provided 22.5792 and 24.576 MHz clocks are native master clocks for many DACs.

 

However, DACs that can use external master clock are really rare compared to pro-gear that can use word clock which is sort of standard. For pro-gear the reason for using word clock is just to sample-synchronize more channels than single ADC or DAC can provide and thus provide a mean to have more channels. 32-128 channels is pretty ordinary studio...

[The Computer Audiophile]

Hi Guys - I received a response from Berkeley Audio Design regarding DSD, PCM, and some items that have been said in this thread. What follows was written completely by Berkeley Audio Design and doesn't necessarily express the opinion of CA.

 

 

 

 

DSD versus PCM and the Berkeley Audio Design approach

12/19/13

 

 

In a recent thread on the Computer Audiophile site, The Multibit DSD debate, an obvious point is stated correctly; it is the sound quality of an overall system that really matters and that sound quality is judged by listening to analog signals. There are many factors that affect sound quality in any specific system, and attributing the differences to only one factor is always an oversimplification.

 

 

There is also a great deal of confusion between DSD and delta-sigma modulation evident in the discussion posts, with DSD being frequently used to refer to delta-sigma. For the purposes of an audiophile discussion, DSD and PCM are delivery / storage formats, and are distinct from the design of data converters and the processing that goes on as part of the conversion process. Delivery / storage formats and the design of data converters should be analyzed separately, since various designs use varying combinations of techniques.

 

 

Direct-Stream Digital (DSD) is the trademark name used by Sony for the raw data output of a 1-bit delta-sigma modulator (DSM), originally coming straight from the A/D, which can then be sent to a 1-bit D/A. This is an idea that is appealing in its simplicity, but in practice it is not so simple, and has its own problems and sonic signature that are different from those of conventional PCM. (More below)

 

 

In evaluating the merits of a delivery / storage format, it is useful to look at its information carrying capacity versus the information rate of the signal that it is delivering - in this case, high quality audio. The information carrying requirement is most easily calculated based on the required dynamic range and frequency extension of the audio signal. In the early days of digital audio, when digital bandwidth was very expensive, many people tackled this question with a goal of picking a minimum information rate that would be considered high fidelity, and the CD was the commercial result. Good, but not really good enough for audiophiles, hence the interim fixes and now hi-res.

 

 

The required dynamic range, as determined by human physiology, to reproduce the full range of audible sound is generally agreed to be about 120 dB, or 20 bit linear PCM resolution. The dynamic range that most transducers can handle well is less, so a delivery format that can provide 120 dB of dynamic range is sufficient. Note that professional formats used for editing, EQing, and processing need to be of higher resolution to produce a good 20 bit result in the final release.

 

 

The required bandwidth, or frequency extension to reproduce the full range of audible sound, is not as well agreed upon as the dynamic range. For steady tones, the physiological limit is around 20 kHz. There is some controversial research indicating the physiology responds to higher frequencies directly, but a more important consideration is that more bandwidth is required to reproduce realistic sounding transients. (Long, complex discussion) Most researchers conclude that a 50-60 kHz bandwidth is enough.

 

 

From the above, it can be concluded that 24-bit 176.4 kHz linear PCM has room to spare in both dynamic range and bandwidth to deliver all perceptible audio content. DSD64x falls a bit short, although it is better than a 16-bit 44.1 kHz CD. DSD128x is capable of hitting the goal with some fancy shaping of the noise floor, which may have audible consequences. (More below)

 

 

Regarding our approach of converting DSD to 24-bit 176.4 kHz linear PCM: We are simply converting one delivery format to another that has greater useful information carrying capacity. It is easy to see that 176/24 has more than enough information bandwidth to carry all of the information in DSD64x. It is possible to put the entire bit stream of a DSD64x signal in 2/3 of the bits of a 176/24 signal, as is done in DoP. It might be argued that because raw DSD128x will not fit in the bits of a 176/24 signal, information is lost. However, when the efficiency of the formats for carrying useful audio information is considered, it becomes clear that 176/24 is still more than sufficient to carry all of the audio information in DSD128x. DSD does not make good use of the available information bandwidth. The conversion process from DSD to PCM can be done with very high precision using digital filtering that is stable and predictable, and especially if done off line, it can preserve all useful audio information so that nothing is lost.

 

 

Since the only valid way that an end user can evaluate a given delivery format is by listening to the end result in an entire system, and since the adequacy of the various delivery formats has been discussed, we will now consider A/D and D/A converters and other system level issues. Here again, there seems to be a great deal of confusion in the discussion posts.

 

 

One of the problems with trying to evaluate digital audio systems, especially DSD, is the huge number of possible variations in implementation. When recordings are specified as 24-bit 176.4 kHz PCM for instance, at least the character of the delivery channel is well known, and the variations in quality from one to another can be attributed to the quality of A/D conversion and the quality of the source itself. The channel is generally understood to have a spectrally flat noise floor that is well below the noise of the converter and the source. The same cannot be said for DSD.

 

 

With DSD, the delta-sigma modulator used to encode the audio in the 1-bit stream has a profound influence on the recording, and the variations in implementation are almost endless. The order of the DSM process is a major factor: the higher the order of the modulator, the greater the dynamic range that can be achieved in a given bandwidth, but at a price. The more one reduces the noise floor in the audio band, the faster the noise rises out of band. It’s like squeezing on a partially inflated balloon – you squeeze in one place and it pops out somewhere else. Also, the in-band noise floor generated by the modulator is normally not flat. Frequently, the noise floor in-band is shaped deliberately to put more dynamic range at frequencies where the ear is most sensitive. This has its own artifacts and sonic signature, most notably in level dependent shifts in instrument timbre. (Does anyone remember Sony’s Super Bit Mapping?)

 

 

Another factor that produces large variations in the system level reproduction of DSD is out-of-band noise. At least some of that noise must be filtered out – the question is how much, and D/A converter designers have a wide range of opinions on that subject. What complicates the decision is the fact that the analog electronics downstream have widely varying tolerances for high levels of high frequencies, and at some point they all become distressed. Typically, high levels of high frequencies cause some part of an amplifier, usually inside a feedback loop, to go into slew-rate limiting, which produces distortion. At onset, it may sound like a softening of the sound, which may be interpreted as euphonic, but it is also a loss of information and addition of distortion.

 

 

In ‘native DSD’ D/A converters, the filtering of the out-of-band noise must be done with an analog filter, and good quality analog filters are expensive and subject to drift with temperature, as well as sometimes requiring tuning during production. They also tend to introduce phase distortion near the filter’s corner frequency and hence also in the audio band. Because of this, ‘native DSD’ D/A converters often produce higher levels of high frequency out-of-band noise to keep the filter simple.

 

 

In contrast, conversion of the 1-bit DSD stream to multi-bit PCM in the digital domain can be done with digital filters, which are stable and, if well designed, free of most of the problems of analog versions. They can also be easily made selectable, even on a recording by recording basis. This is another argument for doing DSD to PCM conversion off-line. The noise floor of each recording can be reviewed before conversion using a spectrum display, often built into the converter, and then pick the optimal filter for the particular DSD delta-sigma modulator used to make the recording. It is only necessary to do this once. This is one of those things that can satisfy an audiophile who likes to tweak his or her system in a way that was common with analog sources but has largely gone away with digital.

 

 

We stand behind our statement that the vast majority of D/A converters currently on the market, including ours, use multi-bit delta-sigma converters. Originally, monolithic converters went from ladder structures at low oversampling ratios to single bit high oversampling ratios because better performance could be achieved at lower cost for mid-fi consumer CD players. It was not until the problem of element matching was solved that multi-bit high oversampling became practical. It has now taken over the high quality end of the market for both professional and high-end audiophile equipment because multi-bit delta-sigma converters produce very high performance without placing a large burden on the product designer.

 

 

One of the most important advantages of 5-6 bit delta-sigma converters is that the delta-sigma modulator can be low order and still meet dynamic range requirements. The result is that the noise floor rises very slowly as one goes up in frequency above the audio band, and therefore, the analog filter following the converter can be simple.

 

 

The above provides the answer to an intelligent question asked in the CA discussion; why go from single bit DSD to PCM to multi-bit delta-sigma conversion to analog. DSD has large amounts of high frequency noise, which can be easily filtered out digitally in the conversion to high bit precision PCM. The PCM can then be processed normally, including controlling level and up-sampling, and then a 5-6 bit lower order delta-sigma modulator drives the DAC with very slowly rising noise at the output and simple analog filtering.

 

 

Multi-bit delta-sigma audio very definitely is PCM, and represents coarsely quantized whole output sample values being sent to a linear PCM DAC. The fact that it is noise shaped does not negate the fact that it is linear PCM. The assertion that delta-sigma is the difference between adjacent samples of PCM is incorrect: that would be delta modulation, a precursor to delta-sigma modulation. Delta modulation for high quality audio was abandoned decades ago because it has a very serious limitation – it is slew limited by nature. The maximum value of the small difference word must be added repeatedly to the total to make a large level change. As a result, in order to reproduce high level, high frequency signals such as cymbal crashes, the delta modulation coder requires extremely high sample rates. For single bit delta modulation (the only version that was widely used in practice) achieving CD level performance requires multi-gigahertz clocking.

 

 

Further evidence that multi-bit delta-sigma data is PCM is the fact that recovering an audio signal after D/A conversion simply requires a low-pass filter with a flat frequency response in the pass band. If it were delta modulation, as has been claimed, an integrator would be required, which has a 6 dB/octave attenuation slope in the pass band.

 

 

It has been correctly stated that a single bit DAC has perfect linearity. However, eliminating amplitude linearity requirements has a side effect of increasing timing accuracy requirements. The high precision requirement has been moved from the amplitude domain to the time domain. 1-bit DAC’s are very jitter sensitive, and require much better clocking than multi-bit DAC’s. This is a perfect example of the design tradeoffs that designers face.

 

 

 

 

BTW, a misconception posted in the discussion needs to be corrected; Keith Johnson has no connection to Berkeley Audio Design and he was not involved in designing any of our products. He remains a close personal friend of ours, but that is the extent of his involvement.

[esldude]

Berkeley's short paper posted above by Chris leads one to believe good high quality hirez PCM can faithfully transmit any info in DSD. And one would be best served to reproduced all files as PCM, doing conversion from DSD to PCM offline or prior to playback rather than real-time during playback. Which all sounds reasonable to me.

[One and a half]

I'm sorry, I don't understand any of the technical reasons that Berkeley Design has made to explain either case for PCM or DSD other than a container issue, like DoP (OK, with inefficiencies, so what).

 

The text is far beyond my comprehension of the subtleties of digital manipulation, other than it appears there are more ways to skin a cat (o'nine tails) to achieve an analog out that sounds decent.

 

If a DAC needs a conversion process in a computer to play a DSD, there's more processing in the computer, which is not all that great an idea, just creates far more noise which is difficult to remove than DSD out of band noise could ever create.

 

The text hasn't persuaded me to be interested in the BADA for purchase, it's "passed through to the keeper".

 

In the meantime, I am going to enjoy all the DSD files with piece of hardware than can decode them in one go.

[esldude]

I'm sorry, I don't understand any of the technical reasons that Berkeley Design has made to explain either case for PCM or DSD other than a container issue, like DoP (OK, with inefficiencies, so what).

 

The text is far beyond my comprehension of the subtleties of digital manipulation, other than it appears there are more ways to skin a cat (o'nine tails) to achieve an analog out that sounds decent.

 

If a DAC needs a conversion process in a computer to play a DSD, there's more processing in the computer, which is not all that great an idea, just creates far more noise which is difficult to remove than DSD out of band noise could ever create.

 

The text hasn't persuaded me to be interested in the BADA for purchase, it's "passed through to the keeper".

 

In the meantime, I am going to enjoy all the DSD files with piece of hardware than can decode them in one go.

 

Actually they suggest it is best to do the conversion of DSD into PCM with software offline. So during playback the computer would not be burdened by any extra processing. Then in their opinion doing it real time with their software is still preferred vs doing it in hardware.

[Miska]

Multi-bit delta-sigma audio very definitely is PCM, and represents coarsely quantized whole output sample values being sent to a linear PCM DAC.

 

OK, so they just demonstrated that they don't understand how multi-bit delta-sigma DACs or multi-element 1-bit delta-sigma DACs are constructed...

 

I'm becoming to conclusion that I don't buy DACs anymore from people who use DAC chips, unless the price is very low (then I can accept it for price reasons).

 

IIRC, Charles said once that people who use op-amps don't understand how to design an analog circuit. Now I feel that people who use DAC chips don't understand how to design a converter...

[The Computer Audiophile]

This is not true. The threshold of pain in in the 120 to 125 db range. Standing a few feet away from a jackhammer is 95 db, next to your power saw is 110 db. Knock that 20 bits down to 16, and you get the 96 db of standard CD, more than enough.

What isn't true?

[The Computer Audiophile]

OK, so they just demonstrated that they don't understand how multi-bit delta-sigma DACs or multi-element 1-bit delta-sigma DACs are constructed...

 

I'm becoming to conclusion that I don't buy DACs anymore from people who use DAC chips, unless the price is very low (then I can accept it for price reasons).

 

IIRC, Charles said once that people who use op-amps don't understand how to design an analog circuit. Now I feel that people who use DAC chips don't understand how to design a converter...

Miska - Your bold statements are starting to make me think you don't understand what you're talking about. Suggesting Berkeley Audio Design doesn't under stand this is preposterous. Please explain what lead you to this conclusion.

[esldude]

Miska - Your bold statements are starting to make me think you don't understand what you're talking about. Suggesting Berkeley Audio Design doesn't under stand this is preposterous. Please explain what lead you to this conclusion.

 

Having heard one for a few hours, B.A.D. certainly knows how to make one BAD sounding DAC (Bad as in evilly good). Just wish they weren't so dear.

[Allan F]

Hi Guys - I received a response from Berkeley Audio Design regarding DSD, PCM, and some items that have been said in this thread. What follows was written completely by Berkeley Audio Design and doesn't necessarily express the opinion of CA.

 

Wouldn't you love to hear what people like George Klissarov or Andreas Koch might have to say about BADA's response?

[The Computer Audiophile]

Wouldn't you love to hear what people like George Klissarov or Andreas Koch might have to say about BADA's response?

I encourage everyone to chime in here. If some people, such as designers or engineers, want to work through me they can send me an email and I'll post it.

[Miska]

Miska - Your bold statements are starting to make me think you don't understand what you're talking about. Suggesting Berkeley Audio Design doesn't under stand this is preposterous. Please explain what lead you to this conclusion.

 

I have explained this so many times that I'm just getting tired and may soon give up and just stop caring...

 

Because multi-bit delta-sigma DAC output stages have equally weighted bits controlled by unary coded sample values, compared to PCM DACs that use binary coded (two's complement) encoding of 2^x weighted bits. (Wolson uses 14 non-equally weighed 1-bit delta-sigma modulated elements and differs in this respect and thus ends up having total of 78 elements)

 

So you can think multi-bit delta-sigma DAC as array of 1-bit DACs. For example Sabre has 64 pieces of 1-bit DACs run in parallel. Multi-level output is constructed from MASH-like structure of cascaded low-order modulators. Then the output is scrambled through the unary thermometer-coding to dynamic element matching where same bit position is every time allocated to different equally weighted output element.

 

Big difference already between SDM and PCM is that SDM typically has odd number of output levels which may be a non-power-of-two number like 25 (dCS). While PCM has always even number of power-of-two output levels.

 

I have also earlier shown in other threads how to use such array of 1-bit DACs to form a linear phase analog filter (that doesn't have any phase distortion) for use with DSD decoding and won't repeat it here.

 

P.S. I'm getting extremely disappointed by the quality of responses I see, especially compared to all the books, scientific papers and lectures over several decades on delta-sigma converter designs. (including bunch of AES papers on the topic)

[Hiro]

Speaking of downsampling DSD to PCM, I once mentioned a SHARP Delta-Sigma Digital amplifier operating at 5.6MHz here on the forum, and was immediately faced with skepticism of other forum members suggesting that it's probably just marketing and the digital amp in all likelihood downsamples the signal to a much lower speed (+300kHz?) somewhere internally.

 

I'm disappointed that some think it's OK to downsample 2.8MHz~5.6MHz DSD to 174kHz PCM and call it a day. These days, when delta sigma converters constitute the vast majority of the market, and the newer ones are capable of handling DSD at its native rate, there's really a no-no for such a practice.

 

BTW, has anybody seen any TTs on the market that convert the analog signal to PCM because their manufacturers insist that PCM is enough and you won't mind?

[Allan F]

BTW, has anybody seen any TTs on the market that convert the analog signal to PCM because their manufacturers insist that PCM is enough and you won't mind?

 

No, but that's probably because it's really hard to find a low compliance cartridge that will play PCM.

[Hiro]

No, but that's probably because it's really hard to find a low compliance cartridge that will play PCM.

 

Well, they could still convert the analog signal to PCM afterwards internally. I'm curious how many vinyl fans would like to listen to their analog records on such TT...

[Paul R]

I thought all those USB turntables did just that? $99 for perfect sound, every time. ($69 on sale on Amazon!)

-Paul

 

 

No, but that's probably because it's really hard to find a low compliance cartridge that will play PCM.

[Allan F]

No, but that's probably because it's really hard to find a low compliance cartridge that will play PCM.

Well, they could still convert the analog signal to PCM afterwards internally. I'm curious how many vinyl fans would like to listen to their analog records on such TT...

 

Sorry, it was intended as a joke.

[Hiro]

I thought all those USB turntables did just that? $99 for perfect sound, every time. ($69 on sale on Amazon!)

 

Typical PCM advertising. "24bit" - you won't need anything else. Best TT in the world

[The Computer Audiophile]

BTW, has anybody seen any TTs on the market that convert the analog signal to PCM because their manufacturers insist that PCM is enough and you won't mind?

People could just stick with the PCM version that was used to create the vinyl for most vinyl releases today :~)