SleepyOne
Headphoneus Supremus
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Chord Hugo
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Rob Watts
Member of the Trade: Chord Electronics
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I have had a number of posts asking me about DSD, how it is done within Hugo etc. A constant idea is that DSD should remain native and hence "pure" and that this would be best. Now I sympathise with this idea, as if it were analogue then one would want to keep everything as simple and direct as possible, as each analogue component you add, you degrade transparency. But this rule simply does not apply in the digital domain, as it is possible to have processing that is perfectly transparent (just to stop there - although a digital module can be made perfectly transparent, it is categorically not easy to do so).
So I need to explain why native DSD is a bad idea inside the DAC. Firstly, Hugo needs to see the original file that you can get hold off, so if it was originally PCM, use that, if it was DSD, feed that to Hugo via DoP. So my comments about native DSD apply about inside the DAC, something people do not see.
Firstly, some history. I first started getting involved in designing DAC's in 1989, when I heard Phillips Bitstream DAC the SAA7320. Compared to multi-bit DAC's at the time, it was a revelation - digital was starting to sound smooth and refined. Now these DACs were PDM types - that is 1 bit with 256 times oversampling - technically exactly the same as DSD but running at 256 times not 64. Now I started with these DAC's, made improvements, and I realised that the noise shapers were limiting resolution, so I started using multiple chips each with their own dither, to improve resolution. Noise shapers convert PCM to lower resolution data like 1 bit DSD. Also I found that the out of band noise from these noise shapers were overloading the analogue sections, giving noise floor modulation, making it sound harder. Also the DAC's were innately very sensitive to clock jitter. To try to solve these problems I designed the PDM1024, which had multiple noise shapers (improve resolution) and digital filtering (delay and add) to help with the jitter sensitivity and the out of band noise problems. Now the PDM1024 (early 90's now) gave a big step forward, but I could not resolve all of these problems. So I started developing Pulse Array, which was a multi-bit noise shape technology. To solve the noise shaper resolution problems, it runs at 2048 FS and is 5th order or better. This theoretically approaches 90dB more noise shaper resolution than PDM at 256 FS, and 150 dB more resolution than DSD 64. The Pulse Array modulation scheme also has the benefit in that it has much lower master clock jitter sensitivity than native DSD/PDM and, more importantly, has no jitter induced noise that is signal dependent as it is a constant clocking scheme - so it has no innate noise floor modulation. Also, by running at 2048 FS, the noise shaper noise at 1MHz is much lower - about 1000 times lower noise than usual DAC's. This means a simple analogue single stage with minimal filtering, so you get much more transparency. Also, the analogue active section has a much easier time, as RF induced noise floor modulation is fundamentally easier.
Now this happened in 1995. At the same time, silicon DAC designers were on a similar path - moving performance DAC's to multi-level noise shaping, away from single bit. At this time DSD started, which was moving in the opposite direction - instead of 256 FS it had reduced to 64 FS, simply because of data rate limitations on optical disks. Now as I have talked about in earlier posts, DSD has a major benefit - it does not have the big timing problems of PCM - but it suffers from much poorer resolution, and creates more distortion and noise than PCM. Using the WTA filter addresses (I won't say eliminates the timing issue because I think we need more taps than today to do that) the timing problems of PCM, giving you the potential of better resolution from PCM and overall better sound.
Now when DSD was first presented, it was claimed that processing could be maintained natively, that is if you wanted to add volume control or freq EQ, you could modify the bit-stream directly and re-noise shape the OP. But people quickly found out that this was not possible. When you re noise shape DSD 64 it very quickly degrades in terms of noise performance - you simply cannot connect 3 or 4 noise shaped stages together, unless you want awful performance. With regular PCM this is not the case, you can add as many stages together and it won't significantly change the measured performance. This is why DSD is initially recorded with PCM at 352.4 kHz - the DXD standard. Then it can be mixed, EQ etc, with minimal losses. Then finally it gets converted from DXD to DSD. And if you can get hold of DXD master recordings and the DSD you can hear the transparency losses of DSD (try 2L website they have free samples).
Now with Qute I did have a choice - I could easily use the DSD data and delay it, then feed it into the 4E DAC. But this would be a very bad thing to do, as DSD is -20dB at 100 kHz, and this noise is distorted, signal correlated, and would cause noise floor modulation in the OP stage. Also, it would be very master clock jitter sensitive, so you would get jitter induced noise floor modulation. The result would be back to the 90's, sub 100 dB dynamic range, distortion, noise floor modulation, whistles, pops and gurgle noises... together with poorer sound stage, poorer detail resolution and a hard aggressive sound quality.
So, in Qute it is digitally filtered, upsampled, filtered again. At this point you still have identical performance, as these steps can be done transparently, that is the audio spectrum is identical to the original signal. It is next fed into Qute's pulse array noise shaper, which will have a small price in transparency. But since this noise shaper runs at 2048 FS not 64 FS, at 20kHz, it is 10,000 times more capable of resolving signals than DSD. At 1kHz it is billions of times more capable than DSD. So the transparency loss is very very small compared with the enormous problems of using DSD natively. In Qute, I used a moving average filter, and the signal was always up-sampled, not decimated. At 100kHz the filter gave 50 dB worth of rejection.
Now in Hugo, we have a potentially much more serious problem with DSD, as Hugo has to do volume control and cross-feed EQ. This means it has to be converted to PCM, and at a rate the cross feed and volume control works at - which is 16FS. So the filtering was much more challenging now, as I had to decimate the signal too (make it a smaller sample rate). This meant a new design, as the Qute filter would have aliasing problems due to not enough stop band rejection - it needed much more than 50dB filtering that Qute had. So I decided on a sledge hammer approach to aliasing problems, by having 140 dB of rejection. This actually is much better than pro standard ADC aliasing filters, but that is another story. The other benefit of this filter was that it removed the DSD noise at 100 kHz, as it had 110 dB worth of rejection at that frequency. Now I could have the benefits of PCM with DSD in that out of band noise is non-existent.
So this filter was designed, verified, measured and listened too. Now I was worried about the listening tests, as I had not decimated DSD before. I listened to the Qute filter directly, and compared it to the Hugo decimating filter, with no volume or cross feed, with identical gain on the two filters. The Hugo DSD filter sounded very much better - DSD was a lot smoother and warmer and more natural, with no detectable loss in transparency or timing. So the benefits of the much better filtering at 100 kHz far outweighed the potential decimation errors by a very big factor.
So apologies if you could not keep up with the technicalities, it is very complex issue - but people kept asking. I hope that you have gotten a flavour of the difficulties involved. Simple and false statements like "native DSD is best" hide a very much more complex reality.
So I need to explain why native DSD is a bad idea inside the DAC. Firstly, Hugo needs to see the original file that you can get hold off, so if it was originally PCM, use that, if it was DSD, feed that to Hugo via DoP. So my comments about native DSD apply about inside the DAC, something people do not see.
Firstly, some history. I first started getting involved in designing DAC's in 1989, when I heard Phillips Bitstream DAC the SAA7320. Compared to multi-bit DAC's at the time, it was a revelation - digital was starting to sound smooth and refined. Now these DACs were PDM types - that is 1 bit with 256 times oversampling - technically exactly the same as DSD but running at 256 times not 64. Now I started with these DAC's, made improvements, and I realised that the noise shapers were limiting resolution, so I started using multiple chips each with their own dither, to improve resolution. Noise shapers convert PCM to lower resolution data like 1 bit DSD. Also I found that the out of band noise from these noise shapers were overloading the analogue sections, giving noise floor modulation, making it sound harder. Also the DAC's were innately very sensitive to clock jitter. To try to solve these problems I designed the PDM1024, which had multiple noise shapers (improve resolution) and digital filtering (delay and add) to help with the jitter sensitivity and the out of band noise problems. Now the PDM1024 (early 90's now) gave a big step forward, but I could not resolve all of these problems. So I started developing Pulse Array, which was a multi-bit noise shape technology. To solve the noise shaper resolution problems, it runs at 2048 FS and is 5th order or better. This theoretically approaches 90dB more noise shaper resolution than PDM at 256 FS, and 150 dB more resolution than DSD 64. The Pulse Array modulation scheme also has the benefit in that it has much lower master clock jitter sensitivity than native DSD/PDM and, more importantly, has no jitter induced noise that is signal dependent as it is a constant clocking scheme - so it has no innate noise floor modulation. Also, by running at 2048 FS, the noise shaper noise at 1MHz is much lower - about 1000 times lower noise than usual DAC's. This means a simple analogue single stage with minimal filtering, so you get much more transparency. Also, the analogue active section has a much easier time, as RF induced noise floor modulation is fundamentally easier.
Now this happened in 1995. At the same time, silicon DAC designers were on a similar path - moving performance DAC's to multi-level noise shaping, away from single bit. At this time DSD started, which was moving in the opposite direction - instead of 256 FS it had reduced to 64 FS, simply because of data rate limitations on optical disks. Now as I have talked about in earlier posts, DSD has a major benefit - it does not have the big timing problems of PCM - but it suffers from much poorer resolution, and creates more distortion and noise than PCM. Using the WTA filter addresses (I won't say eliminates the timing issue because I think we need more taps than today to do that) the timing problems of PCM, giving you the potential of better resolution from PCM and overall better sound.
Now when DSD was first presented, it was claimed that processing could be maintained natively, that is if you wanted to add volume control or freq EQ, you could modify the bit-stream directly and re-noise shape the OP. But people quickly found out that this was not possible. When you re noise shape DSD 64 it very quickly degrades in terms of noise performance - you simply cannot connect 3 or 4 noise shaped stages together, unless you want awful performance. With regular PCM this is not the case, you can add as many stages together and it won't significantly change the measured performance. This is why DSD is initially recorded with PCM at 352.4 kHz - the DXD standard. Then it can be mixed, EQ etc, with minimal losses. Then finally it gets converted from DXD to DSD. And if you can get hold of DXD master recordings and the DSD you can hear the transparency losses of DSD (try 2L website they have free samples).
Now with Qute I did have a choice - I could easily use the DSD data and delay it, then feed it into the 4E DAC. But this would be a very bad thing to do, as DSD is -20dB at 100 kHz, and this noise is distorted, signal correlated, and would cause noise floor modulation in the OP stage. Also, it would be very master clock jitter sensitive, so you would get jitter induced noise floor modulation. The result would be back to the 90's, sub 100 dB dynamic range, distortion, noise floor modulation, whistles, pops and gurgle noises... together with poorer sound stage, poorer detail resolution and a hard aggressive sound quality.
So, in Qute it is digitally filtered, upsampled, filtered again. At this point you still have identical performance, as these steps can be done transparently, that is the audio spectrum is identical to the original signal. It is next fed into Qute's pulse array noise shaper, which will have a small price in transparency. But since this noise shaper runs at 2048 FS not 64 FS, at 20kHz, it is 10,000 times more capable of resolving signals than DSD. At 1kHz it is billions of times more capable than DSD. So the transparency loss is very very small compared with the enormous problems of using DSD natively. In Qute, I used a moving average filter, and the signal was always up-sampled, not decimated. At 100kHz the filter gave 50 dB worth of rejection.
Now in Hugo, we have a potentially much more serious problem with DSD, as Hugo has to do volume control and cross-feed EQ. This means it has to be converted to PCM, and at a rate the cross feed and volume control works at - which is 16FS. So the filtering was much more challenging now, as I had to decimate the signal too (make it a smaller sample rate). This meant a new design, as the Qute filter would have aliasing problems due to not enough stop band rejection - it needed much more than 50dB filtering that Qute had. So I decided on a sledge hammer approach to aliasing problems, by having 140 dB of rejection. This actually is much better than pro standard ADC aliasing filters, but that is another story. The other benefit of this filter was that it removed the DSD noise at 100 kHz, as it had 110 dB worth of rejection at that frequency. Now I could have the benefits of PCM with DSD in that out of band noise is non-existent.
So this filter was designed, verified, measured and listened too. Now I was worried about the listening tests, as I had not decimated DSD before. I listened to the Qute filter directly, and compared it to the Hugo decimating filter, with no volume or cross feed, with identical gain on the two filters. The Hugo DSD filter sounded very much better - DSD was a lot smoother and warmer and more natural, with no detectable loss in transparency or timing. So the benefits of the much better filtering at 100 kHz far outweighed the potential decimation errors by a very big factor.
So apologies if you could not keep up with the technicalities, it is very complex issue - but people kept asking. I hope that you have gotten a flavour of the difficulties involved. Simple and false statements like "native DSD is best" hide a very much more complex reality.
audionewbi
Headphoneus Supremus
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I think we are over thinking all the technicalities. HUGO sounds great and it really doesn't matter to me what it does with DSD files.
Diavel
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Only issue I have is with HD USB connection to iMac (OSX Mavericks) need to change sample rate in Audio Midi setup in utilites sometimes as sound is high pitched , squawky & distorted sounding once doing this no issue not sure if some software OS quirk.
last day at the Axpona Hi-Fi Show, Chicago... don't forget to visit the Noble-Chord stand and grab a listen to some outstanding products namely the K10 & Hugo.
I understand there's a few Hugo's left with 10% off.
the next Chord-Noble Event will be at Hi-End Show, Munich, Germany in May (might be one in UK before that).
I understand there's a few Hugo's left with 10% off.
the next Chord-Noble Event will be at Hi-End Show, Munich, Germany in May (might be one in UK before that).
woodcans
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It already is, they aren't making them anymore, afaik.
Could this be a limitation of your software player (or possibly Mavericks)? Using Amarra in ML, the sample rate is switched automatically. I would assume the same would be true for Mavericks (when using other audiophile software players).
lovethatsound
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Wow the dac in this hugo is amazing, people r saying it's better than some £7000 dacs,I myself am using with the sennheiser hdvd 800 amp and the hd800 with their balance cable and ive never heard the hd800s sing like this, it's turelly breathtaking,I'm using it in my main system as well, the hugo dac has upset alot of people who have spent thousands of pounds on a dac.a turelly amazing product.
audionewbi
Headphoneus Supremus
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I am in market for a good amp for HUGO. I know Mr Watt is against adding an amp in the section as anything added after the analog stage of HUGO will decrease transparency but surely there must be an amp out their that remains true to the philosophy of maintaing the same audio quality of the signal that is it feed right?
I have my eyes on wagnus epslion S.
I have my eyes on wagnus epslion S.
Diavel
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Don't think so as effects output from computer not software specific as includes video audio I think relates to sleep restart issue, but for info I use Fidelia,
verber
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I haven't done a head to head comparison and my time with each was a couple of days with a borrowed unit... but comparing my memories the Hugo was significantly better in just about every way I can think of
I would say a good amp will depend on the headphones. I think for all but the most power hungry (like HE-6) the Headamp GS-Xmk2 is either the best in the top tier. There are several tube amps which equal the GS-X for the HD800. For other headphones I would suggest looking in Headphone XYZ appreciation threads.
I appreciate Mr Watts philosophy about keeping things simple. Something I didn't do was compare my IEM between the Hugo direct, and the Hugo into the GS-X. My guess is that I wouldn't have heard a difference. I did heard a difference between the Hugo and Hugo into the GS-X with the HD800 (and other full size headphones). I think that is because the GS-X is a more capable amplifier while still being very transparent.
--Mark
Or... you can use an amp that ON PURPOSE add to the sound a coloration that you like. You will not be banned for this.
So, a nice TUBE AMP that you like it's signature could also be a solution that add on purpose some warm to the sound.
And.. if you like it better this way, then it is what count, even if you loose the famous ....purity of the HUGO. After all it is all about pleasure...
Myself, I have my HUGO attached to my beloved TU-05 portable tube Amp (on which I experienced a lot of tube rolling), and then, depending on my mood (or the Headphone or music used), I either plug the headphone direct in the HUGO or.... in the TU-05 (that is feed by HUGO)
podeschi
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I am using Onkyo HF Player app via cck to Hugo. Onkyo app doesn't allow for streaming. I have other apps that play music from dlna stream (8player, PlugPlayer, etc) but none of these output from iphone cck setup at higher res. is there a magic app out there that wirelessly grabs the high res files and outputs via cck to hugo in native high res fashion?
Sent from my iPad using Tapatalk HD
Sent from my iPad using Tapatalk HD
SoundFreaq
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Well, that's a tough one. I don't have any empirically sound evidence that the x5 would output a more bit-perfect file than the Onkyo player. It very well may. But carrying around an iPod touch is a much more convenient form-factor, not to mention UI. iPod is my go-to device for sure. The portability factor, combined with the UI experience, negate a theoretical increase in sound quality by using the X5. The Onkyo's HD EQ, upsampling options, combined with the fact that the Hugo nails the stated output bitrate of the player (with indicator color) every time, inspires confidence.
SleepyOne
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podeschi - Just to confirm you are able to play high-rez files (24/196) and DSD (2.8 & 5.6) using Onkyo app on your iPod touch without any problem, many thanks!
SoundFreaq - Thank you! Trouble free , good UI, plus other useful apps are always handy!
SoundFreaq - Thank you! Trouble free , good UI, plus other useful apps are always handy!
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