DSD (SACD) FIR based software crossover - Page 2 - diyAudio
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Old 1st April 2014, 04:31 PM   #11
Julf is offline Julf  Europe
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Quote:
Originally Posted by emaster View Post
Yes, you partially right.
Pulse density on the filter's input transforms to the signal amplitude samples. But unlike the regular PCM, difference between the following two samples (after the filter) never exceeds the LSB value - due to the high frequency rate. So, the delta-sigma modulator restores DSD stream with a maximum available quality (someone can name it 'bit-perfect').
OK, so technically it is PCM that encodes the amplitude difference instead of the absolute value - I guess pretty much precisely like "DSD-wide", as used by some digital editing / workstation software.
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Old 1st April 2014, 05:09 PM   #12
emaster is offline emaster  Russian Federation
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OK, so technically it is PCM that encodes the amplitude difference instead of the absolute value - I guess pretty much precisely like "DSD-wide", as used by some digital editing / workstation software.
Exactly! And also, intermediate sample format in DSDXOVER is 64-bit double, instead of 8-bit in DSD-wide.
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Old 1st April 2014, 05:21 PM   #13
Julf is offline Julf  Europe
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Thanks!
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Old 2nd April 2014, 01:01 AM   #14
Jay is offline Jay  Indonesia
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IMO, the arduino ide with its c++ based syntactic sugar is easy enough to implement logic for home-brew devices.
Yes I have an arduino module here but I'm not familiar with C++. But it is part of DOTNET so I think I may switch to it, but for application user interface I think I still need other language.

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It seems, windows 7 is more preferable one for the advanced user.
And windows 8 is a step towards to the phone/pad-like UI style.
When I used Windows7 I had problems with networking, printing, etc. I thought it might have serious incompatibility issues (so I went back to XP).

I really don't like the user interface of Windows 8, but I think it shouldn't be difficult to "customize" the OS. The operation is smooth enough until now (only a few days). But I have used Windows Server 2008 R2 or 2012 which does not accept Windows 8 as a client. And if you think Windows7 is okay then I may look at it again.

Regarding software based DSP, what is your experience regarding the audibility of latency in the audio stream?
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Old 2nd April 2014, 08:53 AM   #15
emaster is offline emaster  Russian Federation
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Originally Posted by Jay View Post
but for application user interface I think I still need other language.
I think c# is a reasonable choice at this moment.

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Regarding software based DSP, what is your experience regarding the audibility of latency in the audio stream?
Of course, there are latency. Phase-linear filters requires as larger block of data for processing as the higher frequency resolution for cut frequencies required. Measurements for the first version of xDSP shows in->filter->out latency about 15-20 ms for ASIO version and FIR filter with 1024 taps.
For the directsound version latency is about 150-400 ms due to system input/output latency.
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Old 2nd April 2014, 08:58 AM   #16
phofman is offline phofman  Czech Republic
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Quote:
Originally Posted by Jay View Post
Regarding software based DSP, what is your experience regarding the audibility of latency in the audio stream?
Latency is noticable when there is an event to compare with - e.g. video (synclip), pushing a MIDI keyboard button or watching a DAW displaying the currenty played clip. Of course all DSP'd channels must preserve the same delay.
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Old 2nd April 2014, 09:13 AM   #17
emaster is offline emaster  Russian Federation
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Latency is noticable when there is an event to compare with - e.g. video (synclip), pushing a MIDI keyboard button or watching a DAW displaying the currenty played clip. Of course all DSP'd channels must preserve the same delay.
You absolutely right!
It seems, playback start latency is not the most important parameter for the music listening (if delay not exceeds 0.5-1 seconds).
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Old 3rd April 2014, 03:16 AM   #18
Jay is offline Jay  Indonesia
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Originally Posted by emaster View Post
It seems, playback start latency is not the most important parameter for the music listening (if delay not exceeds 0.5-1 seconds).
Not the most important parameter, for sure. But I think I asked the wrong question because I'm sure I'm more critical than most regarding the audibility of things. I have little experience with this DSP latency but with group delay in loudspeakers I tend to feel that group delay makes music not interesting. But it is hard to set a "standard" of maximum delay allowable, when there are too many compromises at stake.

But your xDSP's 15-20ms is just perfect I would say.
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Old 3rd April 2014, 06:34 AM   #19
emaster is offline emaster  Russian Federation
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I have little experience with this DSP latency but with group delay in loudspeakers I tend to feel that group delay makes music not interesting.
Group delay that varies with frequency it's a property of non-linear phase filters (IIR) without additional phase correction.
Since a FIR filter is a linear phase filter and has constant group delay, all frequency components have equal delay times.
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Old 14th April 2014, 12:41 PM   #20
andpa is offline andpa  Russian Federation
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The one of the best DRC system now is the Acourate by AudioVero. It allows to make the calculations fo FIR for specific audio system. But it cannot work with DSD stream. So is it possible to use calculated in Acourate FIR in your program?
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