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Old 4th April 2013, 12:09 AM   #1
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Default TI PCM1792 DAC: change firmware in PC/soundcard

May have missed this in the Archives ...
If one has a modern "high-end" PC sound card, or any computer sound section with a firmware-configurable DAC (like PCM 1792, etc.), how can one change this DACs parameters -- aka "User-Programmable Function Controls" -- (on-the-fly) in the computer? I haven't been able to find applicable drivers or plugins for my soundcard (Xonar) or software-audio-player (Foobar2000).
Ref: See Table 3: http://www.ti.com/lit/ds/symlink/pcm1792.pdf
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Old 4th April 2013, 12:15 AM   #2
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With the Xonar cards the 1792 is controlled via the PCI bus interface chip over the I2C protocol. What the interface chip does to the 1792 is, as far as I am aware, determined by the ASUS software. ASUS already control the internal volume control on the PCM1792 and alter the oversampling rate with the sampling frequency. What is it exactly you would want to get access to?
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Old 4th April 2013, 12:51 AM   #3
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Quote:
Originally Posted by 5th element View Post
What is it exactly you would want to get access to?
The digital filter (fast/slow), sampling rate, etc.

I did find the 1792 User/Dev. Guide:
http://www.ti.com/litv/pdf/sleu037
(see pg. 1-5,1-6)
...and Demo/control software for Win.-based PCs:
Audio Converter - Audio DAC - PCM1792A - TI.com

I installed the Demo prog. but haven't figured out how to get it to DIRECTLY interact with the 1792.
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Old 4th April 2013, 11:44 PM   #4
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You cannot access the 1792 on the ASUS card with the TI software, that is meant to work only with the TI demo board.

With regards to the sampling frequency, you can already change this with the ASUS control panel and the differences between the fast and slow roll off are largely accademic. Some hardware setups might benefit from the lower latency (slow) filters, but from a sound quality point of view, I doubt you'd able to hear anything. The slow roll off filter is still very steep in absolute terms, it's just less steep than the fast filter.
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Old 7th April 2013, 06:33 AM   #5
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Default Time domain vs. freq. domain perf.

Quote:
Originally Posted by 5th element View Post
You cannot access the 1792 on the ASUS card with the TI software, that is meant to work only with the TI demo board.

With regards to the sampling frequency, you can already change this with the ASUS control panel and the differences between the fast and slow roll off are largely accademic. Some hardware setups might benefit from the lower latency (slow) filters, but from a sound quality point of view, I doubt you'd able to hear anything. The slow roll off filter is still very steep in absolute terms, it's just less steep than the fast filter.
For some reason, the Xonar Control Center did not install on my PC. Now that I've done so, I do see the sample-rate control feature, but have no idea whether this is implemented in the 1792 or "in software only" via the main PC CPU ???
As far as fast/slow roll-off... it may not be JUST the freq-domain performance; i.e., sometimes the latter means better concomitant time-domain reconstruction. E.g., the "min. phase"or "apodizing" filters that have been getting a lot of attn lately. Wadia has used something similar since the late 1980's.
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Old 7th April 2013, 11:18 AM   #6
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Quote:
Originally Posted by hollowman View Post
For some reason, the Xonar Control Center did not install on my PC. Now that I've done so, I do see the sample-rate control feature, but have no idea whether this is implemented in the 1792 or "in software only" via the main PC CPU ???
As far as fast/slow roll-off... it may not be JUST the freq-domain performance; i.e., sometimes the latter means better concomitant time-domain reconstruction. E.g., the "min. phase"or "apodizing" filters that have been getting a lot of attn lately. Wadia has used something similar since the late 1980's.
DACs are slave devices in that they don't have any say in the data rate, it is provided for them, not by them. When you select 48/96/192k in the ASUS control panel you are directly changing the sample rate being delivered to the DAC by the PCI audio interface chip.

Now some DACs do require that you alter the internal oversampling rate when the sampling frequency changes, the PCM1792 in default mode is not one of them however.

With regards to the fast/slow filters, yes the latency of the slow filter is less than that of the fast, so in theory, I think, would show less pre ringing. Without details of the ASUS drivers though, I don't know which ASUS use as standard.
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Old 7th April 2013, 06:45 PM   #7
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Default Upsampling?

Quote:
Originally Posted by 5th element View Post
DACs are slave devices in that they don't have any say in the data rate, it is provided for them, not by them. When you select 48/96/192k in the ASUS control panel you are directly changing the sample rate being delivered to the DAC by the PCI audio interface chip.
I should've been clearer....what I meant was what is delivered to the Xonar's decoder DSP. Does the Xonar control center simply act as a multi-position switch (it detects native sample rate of certain file or disc). So, is some software-based processing done betw. Asus and Windows/Intel? I doubt it, but audio players like Foobar do feature their own plugins that work in pure software only.
I think it's more likely as you suggest: hardware acceleration only (i.e., in the sound card). On that note ... can one effectively upsample non-high-rez data: e.g., send typical 16/44.1 RedBook data to 1792 in 96k or 192k? I haven't tested this yet -- maybe others have (subjective) and even taken some measurements (objective; graphs; RightMark)???
I have a MusicalFidelity outboard D/A that offers 96 or 192 as switchable upsamples ... via some dedicated asycn chip (curiously, one can't turn upsampling off completely). It also uses a TI DAC, I think.
Oh ... FWIW ... on the MF, 96k sound better to my ears.
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Old 7th April 2013, 07:35 PM   #8
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Why not just use a software sample rate converter?
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Old 7th April 2013, 08:04 PM   #9
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Quote:
Originally Posted by counter culture View Post
Why not just use a software sample rate converter?
If one has a dedicated "audio computer", this makes sense. Not the avg. Joe who is an obscene and vulgar multi-tasker ...
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Old 7th April 2013, 08:50 PM   #10
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Why not, the upsampling does not have to be particularly low latency, and even a fairly long polyphase filter is not **that** expensive to compute.

The upsampler could run in the disk IO thread which can easily have a MB or so of buffer between it and the card.

I see no reason not to do this on the main CPU even of a very conventional multitasking box, a few percent of the CPU is a lot of mac operations.
Not that I think it will make much difference, my hearing is shot past about 15Khz or so, so 44.1K is **way** more then I really need.

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