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In Reply to: RE: Saracon posted by Tony Lauck on September 11, 2011 at 09:19:35
fmak uses a SACD transport to play the physical disc into his dac.
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I use both methods (3 transports) and software on PC. Hardware is better as I can control quality of components, power supply, and interface. Software doesn't sound as good.
With dsd material, the bandwidth of your replay system is very imortant. I use between 50k and 70k analog filter.
I am not surprised that Saracon converted files sound 'clean', because of the 22k filter. In general, I prefer a much more wide open system where the effects of the filters are moved as far from 20k as possible.
With the foobar2000 converter the filter rolls off around 70 Khz. If one looks at the 192/24 PCM samples (zoom in to individual samples) one can clearly see the residual high frequency noise. Or one can see it in the "voice print" that I posted, showing a wide area of haze starting around 40 Khz. I don't like the Weiss strategy. I'd rather have some of this noise than have musical signal filtered out by the converter. But the real problem is with DSD itself and/or the modulators used to create DSD bitstreams. They create this damn noise, and the noise isn't pure like properly dithered PCM, it's all tied up with the musical signal.
From implementing and then evaluating the output of some sample modulators published by Philips engineers I found that 1-bit noise is still present below 20 kHz, albeit at low levels. I don't believe these published modulators can even encode 44/16 PCM and get back the original 16 bit samples unchanged, proving in one sense that DSD has lower resolution than RBCD. But perhaps this was because I wasn't decoding these modulators correctly. If someone has links to papers on this subject I would be interested. My guess is that it will never be possible to get decent encoding out of a D-S modulator, but I believe it may be possible to design a different type of encoder that will do better. (If you read the DSD encoding section of the SARACON manual you will see that Weiss's encoder has a lot of klugery to prevent it from "lockup" caused by signal overload. A dubious system design, IMO.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
is not a 1 bit system; 5 bit apparently.
The Ultrasopnic noise needs to be dealt with by an analog filter, and thru a replay chain that is not upset by the out of band signal.
The ideal transient response is one factor that makes dsd better than pcm.
"is not a 1 bit system; 5 bit apparently."
Huh? A DSD bit stream is a string of single bits. How these bits are generated is part of an encoder and how they are used is part of a decoder. Encoders and decoders can be implemented in analog or digital form, etc... It is not possible to conduct an intelligent technical discussion of a communication system (i.e. any system involving audio formats or media) without an understanding of the fundamental difference between the system architecture and an implementation of that architecture. From your comments, it appears that you do not understand this fundamental distinction.
FWIW, when I implemented the sample delta-sigma modulator described in the Philips paper it was implemented using 64 bit arithmetic. There was a low pass filter that rejected essentially all of the signal above 22 kHz. Despite this, there was residual noise in the audio band, and it was well above the dither noise in 16 bit Redbook. I put various test signals through the modulator, e.g. DC, sine waves at various frequencies, etc., and looked at the levels and spectral plots of the (filtered) output. One of the filters that I used was a Windowed SINC filter with 512K terms. None of this ran in anything close to real-time, it was just to test how well the modulator worked. It sucked.
"The ideal transient response is one factor that makes dsd better than pcm."
It's about the only one. And it has nothing to do with DSD vs. PCM, it comes just from the higher sampling rate which permits a larger frequency range between the pass band and the stop band of the output filters needed to decode any sampled data system. One can get equivalent transient response by increasing the PCM sampling rate, albeit at the cost of more bits, but bits are cheap today, compared to when DSD was first proposed.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
I even believe Sony engineers have even acknowledged this description.
It's all Greek to me, though.
Chris
In the course of an SACD record-playback system the original analog audio signal exists in various formats, certainly analog at the microphones and speakers and in 1 bit format as encoded on the shiny spinning disks. However, various conversions take place "between the sheets" and other formats may be used inside of hardware boxes and computer systems. The Sony and Philips engineers have done their best to obscure the situation in their published papers, as this keeps the technical limitations of their "invention" Greek to most people.
One of the problems with the 1-bit format is that it can't be edited, not even a simple gain change or fade-out. One of the ways that Sony tried to overcome this limitation is to convert to 8 bits at the same 64x sample rate. In this form the recording can be edited and then converted back to 1 bit without too much loss of sound quality and is used in some SACD workstation software. Another approach is to convert to 24 bit (or 32 bit floating point) at 8x (352.8 kHz). This is the format favored by other SACD workstation vendors and is supported by various converters. The advantage to this format is that multiple editing processes can be performed in the DXD format, thereby reducing the inevitable generation loss converting into/out of DSD. (There are two costs to this approach: a slight dulling in potential transient response due to the 8x lower sampling rate and a significant increase in bit sizes.)
Perhaps this Digital Audio Denmark marketing white paper will be Greek, perhaps not. :-)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
If you have no first hand experience of dsd hardware and playback, what are you harping on about? Chris and I, plus others have been using it for years.
Stop blowing hot air and post some sensible comments based on experience and understanding, not postulation and wonky theorising.
who do not, and have not tried to understand sacd and it's implementation.
The Sony DSD chip is not a 1 bit implementation but much more complex. This is well known if you are into audio and not pcs.
DSD's impulse response has been what pcm filter gurus have been trying to emulate with different algorithms and is a direct consequence of the format. This is well known from the BEGINNING.
If you want to seriously evaluate the format instead of just wanting to rip and play on a pc, you need to have the primary interest in audio and its playback, not software.
Please help me to understand fmak. Thanks
Edits: 09/12/11
The data format is one bit but a 1 bit system has such poor performancve that it is not worth using. In reality, if you go deeper. the processing is much more than that and there is a similarity between dCS dac technology (ring dac) and the decoding/processing system. This was whyu dCSD was amongst the first companies that included dsd playback. one has to go beyond the label in many cases, as is so often the case in computer software.
It is interesting that pundits who totally rejected dsd only a few months ago are now trying to make 'definitive' pronouncements on it. These guys are just format 'gurus' who have little interest in how they operate and sound.
Once again, you have demonstrated that you do not understand the difference between architecture and implementation. Furthermore, you have demonstrated an inability to recognize what you do not know. You are therefore either unwilling or unable to overcome your ignorance. I will not be having any further discussion with you on this matter, due to the unfortunate combination of your bad attitude and your willful ignorance.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
your hot air on a format that you totally rejected a few months ago.
I have followed your nposts on dsd and they are all virtually without substance.
You won't see me shelling out any $$ for hardware that comes with DRM built-in. That includes SACD players and iLoks.
Does Saracon still require an iLok?
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
But you will spend $thousands on faulty software or one which carries NO clarity on what it is doing. Hence all these tests which you have to carry out.
Tony,
I have a friend that is an amateur musician that has a recording studio at home. He lives quite close to me and has the Saracon software. He lets me use it on his computer. Yes, his has an iLock.
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