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Old 20th August 2014, 06:22 AM   #1
fotios is offline fotios  Greece
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Default PCM1794 oversampling rate question

How does PCM1794 is handling the oversampling rate of its Delta-Sigma modulator?
In PCM1792 (the counterpart of PCM1794 managed through microcontroller) the OSR must be selected by the user through the micro software. The same applies to WM8741 of Wolfson, the user must select the suitable OSR of Δ-Σ modulator according to the sampling rate of the incoming digital audio signal.
PCM1794 does not offer this possibility. Why? It has the ability to detect the sampling rate and to automatically select the proper OSR?
Thanks for any information.
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Old 20th August 2014, 06:57 PM   #2
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You should know that i already have in use the pair of Wolfson WM8805 + WM8741 managed by a microcontroller.
As you probably know the DAC WM8741 has 3 oversampling rate settings, Low, Mid and High and according to each one a different set of filters is selected. From actual tests, if e.g. the Low OSR is selected and a higher rate digital signal (96KHz to 192KHz) is applied to WM8805 receiver, you could not hear nothing except noise. OTOH, if the High OSR is selected, any signal of any sampling rate can be converted without problem. But Wolfson suggests that the selected OSR should match with the sampling rate of incoming signal. For that reason, the microcontroller is informed from WM8805 for the sampling rate of it and accordingly selects the suitable OSR to WM8741 Δ-Σ modulator.
A similar process applies to PCM1792A. But not for the PCM1794. How on the earth this DAC could select the correct OSR? Or it is internally locked to a High OSR so as to be compatible with any signal of any sampling rate?
Do you think that PCM1794 has been developed for... NOS DACs?

P.S. My Asus Xonar Essence STX sound card is equipped with the PCM1794 and offers a great audio performance for any sampling rate. From actual audition tests it is directly comparable with WM8741.
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Old 21st August 2014, 12:52 AM   #3
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Unlike the Antikythera device in your avatar, it seems like these modern devices do not include instructions with them.

It perhaps lowers the sampling rate with 96 kHz and then lowers it further with 192 kHz.

The PCM1792 and WM8741 can operate in the highest oversampling with 192 kHz?

The PCM1792 was released later than the PCM1794, perhaps it's an improvement?

I'm not sure if they can operate in zero oversampling, they are both a hybrid deaign and I'm pretty sure the 6-bit section can operate in non-oversampling, not the Sigma-Delta section since S-D in Nos is just a lot of noise I think.
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Old 21st August 2014, 06:37 AM   #4
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Quote:
Originally Posted by Kastor L View Post
Unlike the Antikythera device in your avatar, it seems like these modern devices do not include instructions with them.

It perhaps lowers the sampling rate with 96 kHz and then lowers it further with 192 kHz.

The PCM1792 and WM8741 can operate in the highest oversampling with 192 kHz?

The PCM1792 was released later than the PCM1794, perhaps it's an improvement?

I'm not sure if they can operate in zero oversampling, they are both a hybrid deaign and I'm pretty sure the 6-bit section can operate in non-oversampling, not the Sigma-Delta section since S-D in Nos is just a lot of noise I think.
Hi Kastor L, thanks for the reply.
I don't know what exactly takes place inside PCM1794, i did an assumption based on WM8741 working in hardware mode. Still in this case it allows the user to select the proper OSR through the use of a 3 position switch. WM8741 includes a set of extra "PCM Digital Filters" before the Σ - Δ modulator (the oversampling is applied to modulator). PCM1792a and PCM1794 don't include a simillar filter section. Additionally, the OSR of PCM1792a is accessible only trough microcontroller. WM8741 looks like more flexible.
For the High OSR, i can confirm you that WM8741 is working flawlessly with signals of 192KHz, i have checked it on actual hardware. I think the same applies to PCM1792a and PCM1794 although i still haven't tried those DACs.
In both WM8741 and PCM1792a you could bypass all internal filters. In WM8741 datasheet is refered as 8FS mode which also is accessible only through microcontroller (software operation mode). I quote from WM8741 datasheet:
When MODE8X is set, the PCM data input to the WM8741 is applied only to the digital volume control and then the analogue section of the DAC system, bypassing the digital filters.
A simillar process also applies to PCM1792a, it is refered as DFTH: Digital Filter Bypass (or Through Mode) Control and also states that: The word (WDCK) signal must be operated at 8× or 4× the desired sampling frequency
PCM1794 can also be configured in bypass mode through 4 pins (hardware mode).
This functionality is offered by the 3 DACs (Wolfson and TI) when a DSP is desirable as an external digital filter to perform the interpolation function.
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Old 21st August 2014, 07:26 AM   #5
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I see, all noted.

Here is a service which changes 2x WM8740 to 2x WM8741, plus changes the digital filter setting from linear- to minimum-phase.

It's expensive and popular, have a look

Red Wine Audio Components
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Old 21st August 2014, 07:41 AM   #6
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Quote:
Originally Posted by Kastor L View Post
I see, all noted.

Here is a service which changes 2x WM8740 to 2x WM8741, plus changes the digital filter setting from linear- to minimum-phase.

It's expensive and popular, have a look

Red Wine Audio Components
Thanks for the link, it gave me some ideas to try in my actual project.
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Old 21st August 2014, 10:21 AM   #7
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Quote:
Originally Posted by Kastor L View Post
I see, all noted.

Here is a service which changes 2x WM8740 to 2x WM8741, plus changes the digital filter setting from linear- to minimum-phase.

It's expensive and popular, have a look

Red Wine Audio Components
I looked again in redwineaudio modifications of their RWAK 120/S/B digital audio player. It says that they replace the two WM8740 with WM8741 and that configure them to work in hardware mode. They state that is better this, because WM8741 volume control is bypassed: We discovered that even with the volume set to MAX (setting 75), the output voltage of the WM8740 signal is still attenuated in the digital domain Hmm... that is curious, i haven't noticed something like this in my hardware.
Also that they use the “minimal phase digital filter”. But what from the 3 offered by WM8741? a) Minimum phase ‘soft-knee’ filter b) Minimum phase half-band filter c) Minimum phase apodising filter ???
Curious things... Why they don't make a firmware upgrade in microcontroller?
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Old 21st August 2014, 11:13 AM   #8
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Hmm that is curious.

Firmware upgrade to microcontroller? So the user can change the settings?

Yes it's a great idea.

I'm not sure which filter of the three you write they use.
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Old 21st August 2014, 05:10 PM   #9
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Quote:
Originally Posted by Kastor L View Post
Hmm that is curious.

Firmware upgrade to microcontroller? So the user can change the settings?

Yes it's a great idea.

I'm not sure which filter of the three you write they use.
Yes, indeed is curious.
Look what is refered in WM8741 datasheet regarding filter selection:
There is a "tristate" input pin named "FSEL" which can be tied to Vcc = logic state 1, or to GND = logic state 0, or it can be left open = High Z state.
In hardware mode and according to OSR selected, the following filters will be selected by default:

1. For FSEL = 0 and OSR = Low Rate (32 to 48KHz)
Linear phase half-band filter for backward compatibility
For FSEL = 0 and OSR = Mid or High Rate (88.2 to 192KHz)
Linear phase half-band filter for backward compatibility

2. For FSEL = 1 and OSR = Low Rate (32 to 48KHz)
Minimum phase apodising filter
For FSEL = 1 and OSR = Mid or High Rate (88.2 to 192KHz)
Linear phase ‘brickwall’ filter

3. For FSEL = Hi Z and OSR = Low Rate (32 to 48KHz)
Linear phase apodising filter
For FSEL = Hi Z and OSR = Mid or High Rate (88.2 to 192KHz)
Minimum phase ‘soft-knee’ filter

redwineaudio says that the WM8741 is configured in hardware mode. So the FSEL pin should constantly be in 0 or 1 or Hi Z logic state.
redwineaudio also says that the "minimal phase" digital filter is allways selected.

According to the above table THAT IS IMPOSSIBLE!
If FSEL is tied to GND = 0, apparently we can't talk about "Minimal phase" filter.
If FSEL is tied to Vcc = 1, for Low OSR indeed the "Minimum phase apodising" filter will be selected , but for Mid - High OSR the "Linear phase brickwall filter will be selected.
Finally, if FSEL pin is left open = Hi Z, for Mid - High OSR indeed the "Minimum phase soft-knee" filter will be selected, but for Low OSR the "Linear phase apodising" filter will be selected.

Here, we have a contradiction in the description of redwineaudio.
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Old 21st August 2014, 07:53 PM   #10
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I know very well what they do in redwineaudio for the advertised modification as i tried same things in my WM8741 DAC.
Actually, in commercial devices the use of WM8741 in software mode is a big messing for the end user as he pay some thousands just to hear music and not to make experiments. For this purpose you could buy the WM8741 evaluation board. For that reason, WM8741 is configured to work in hardware mode. But still in hardware mode, and if we talk for a consistent audio device, the OSR of Σ-Δ modulator of WM8741 must be changed within 3 levels according to the sampling rate of incoming digital signal for the selection of correct filters. There is a tristate input pin, named OSR, that is offered by WM8741 for this purpose. You have two choices for selection of correct OSR: a) to connect the OSR pin in a 3 position switch for manual selection and b) to use a microcontroller which gets the info of the sampling rate of incoming signal from the SPDIF receiver chip and accordingly changes the states of a pin connected to OSR input of WM8741 and: outputs a logic 1 for High OSR or 0 for Low OSR or is turned in input to present Hi-Z state in the OSR pin for Medium rate. At the same time, the FSEL tristate input of WM8741 must be changed through another pin of micro, if we like to allways use the Minimum or Linear Phase Apodising filter (which is the suitable filter for pre-ringing elimination) for any OSR. As you could understand, the use of a micro is NECESSARY for automated funcionality. It is not practical the use of manually operated switches for the end user.
But given that if the OSR pin is constantly tied to Vcc, to get a logic level 1, then the WM8741 Σ-Δ modulator is also constantly working in High OSR and can convert any signal of any sampling rate. But this is the dishonest solution as you couldn't never get the best possible audio performance.
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Last edited by fotios; 21st August 2014 at 07:56 PM.
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