Post by jazz1 March 16, 2013 (1 of 9)
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Just received the new Joachim Raff Symphony No2 SACD, it is an excellent recording nearly in the CC class.
I am just a little confused about the DSD and 24/96 mention, I always thought
that DSD was 1 bit/?? Am I wrong??
BTW it is a wonderful SACD musically the only music of Raff I knew was from the fantastic string quartet SACD
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Post by Gussy March 16, 2013 (2 of 9)
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This means it is an original 24/96 PCM recording and then encoded to DSD for the SACD - don´t worry, the vast majority of (new) SACDs are recorded this way...
If it sounds good, it is good!!!
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Post by Ear March 16, 2013 (3 of 9)
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Hey Jazz
The 96/24 refer to the recording mode. It means that the SACD was made from 96/24 PCM recording which was converted to DSD to produce an SACD. The SACD still is DSD ( 1 Bit)
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Post by DSD March 16, 2013 (4 of 9)
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I'm still trying to understand how one converts from PCM to DSD without converting to analog first.
In this case we have a 24/96 PCM recording meaning there are 24 ones and zeros that describe 1/96,000th of a second of music.
DSD on the other had doesn't describe the waveform each time but measures if the waveform goes up, down or stays the same. Getting three states with Base-2 computer code (1s and 0s) is a bit tricky, if the waveform goes up it outputs a 1, if it goes down it outputs a 0, if it stays the same it outputs alternating 1's and 0's. These numbers are fed into the memory buffer, and each new sample is compared to the previous sample, I understand the reason for the memory is the DAC has to see two numbers the same for the waveform to raise or fall.
So with PCM we have a very small slice of the complete waveform which is why it is easier to edit, DSD we have movement of the waveform up and down. So how does one change from one to the other keeping things in the digital domain? I've been trying to wrap my head around how this is done for over a decade, maybe someone here knows.
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Post by AmonRa March 16, 2013 (5 of 9)
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DSD:
without converting to analog first.
It is just mathematics. Two different methods to describe the waveform, computer can convert between these two with less added distortions than converting the signal to analog and back. DA and AD conversions are ever perfect, neither is the analog circuitry between them. Math is pure with the right algorithms.
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Post by AmonRa March 16, 2013 (6 of 9)
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ever = Never
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Post by Scytales March 16, 2013 (7 of 9)
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DSD said:
So how does one change from one to the other keeping things in the digital domain? I've been trying to wrap my head around how this is done for over a decade, maybe someone here knows.
It is done for more than a decade in almost every digital to analogue chipset, where the 16bits/44,1 kHz or 24 bits/96 or 192 kHz PCM (with eventually digital filtering in-between) read on CD or DVD respectively is passed through a delta-sigma modulator to convert the bitstream in high frequency modulation (over MHz) with fewer bits (typically 3 to 8, though 1 bit delta-sigma converters has existed).
The operation is entirely digital.
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Post by hiredfox March 16, 2013 (8 of 9)
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DSD said:
I'm still trying to understand how one converts from PCM to DSD without converting to analog first.
It is called approximation.
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Post by tailspn March 16, 2013 (9 of 9)
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hiredfox said:
It is called approximation.
+1
:)
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