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Windows 11 - Bass redirection to a Subwoofer
- Thread starter -Jim-
- Start date
Next question: global volume control. It looks like VB-Audio Matrix disables Windows Mixer and taskbar volume widget's ability to control volume. It shows the device in Windows Mixer with levels, but changing those levels does nothing.
Is the only way to control global playback volume, for all VAIO channels, in the app that is playing -- e.g., the Tidal player's volume control?
Is the only way to control global playback volume, for all VAIO channels, in the app that is playing -- e.g., the Tidal player's volume control?
It shouldn't. The flow goes app > windows mixer > eapo > matrix > hardware out.
If you adjust volume in windows mixer, it affects all subsequent stages.
If something weird happens there, make sure eapo is installed post-mix and manually set post-mix stage.
But for better overall control and more flexibility, you could try using the banana/potato mixer too. In that case you'd be creating a virtual asio device instead of wdm vaio in matrix and assigning it to potato's A1 out. Then use one of potato's virtual wdm devices as the default windows output and install eapo on potato instead. eapo can also hook directly into the potato outputs. Plenty of options either way.
If you adjust volume in windows mixer, it affects all subsequent stages.
If something weird happens there, make sure eapo is installed post-mix and manually set post-mix stage.
But for better overall control and more flexibility, you could try using the banana/potato mixer too. In that case you'd be creating a virtual asio device instead of wdm vaio in matrix and assigning it to potato's A1 out. Then use one of potato's virtual wdm devices as the default windows output and install eapo on potato instead. eapo can also hook directly into the potato outputs. Plenty of options either way.
I wonder if a reboot would solve it. Perhaps Windows needs to get comfortable with this new VAIO device.
In any event, it's already working better than I imagined possible without buying fancy DSP hardware. Thank you for all your patient help!
In any event, it's already working better than I imagined possible without buying fancy DSP hardware. Thank you for all your patient help!
Nope, no change after reboot. See attached -- volume mixer with master volume muted, but output is unchanged/full volume. You can see the little darker gray bars showing Tidal is playing. Those darker gray bars move with the music level. They become green when unmuted. But levels cannot be changed.
Attachments
Even though I too have had expected VB-MATRIX would behave as you pointed, my careful evaluation/measurement proved it is not so, because VB-MATRIX (as well as ASIO4ALL) cannot synchronize the "trigger" of multiple DAC unit as I shared in my post #783, #804 (using ASIO4ALL) and #1,021 (using VB-MATRIX).
I highly recommend you checking/measuring the "trigger lags" by using a music track having sharp timing marker on the top (start) and bottom (end) of the track just like I have done.
As I wrote in my post #783 and #804, I can compensate the "trigger lags" by using upstream DSP software "EKIO"'s group delay settings, but the method is valid/applicable only if the "trigger lags" would be strictly kept unchanged/the-same throughout your listening sessions, but there is no guarantee at all for this "hope". Consequently, this approach should not be recommended for routine applications.
As I shared in my post here #389, in case if two of the same brand DAC units (like two or three of RME DAC units) have special synchronization mechanism through the aggregated ASIO driver, of course we can eliminate "trigger lags" and "drift of synchronization". Also please refer to #842 on my project thread.
assume you are doing this, right?
Even though I can well understand your routing setup, I believe you need careful validation measurement of "trigger sync" and "sync after starting music, i.e. lack of sync drift" by using a music track having sharp timing marker on the top (start) and bottom (end) of the track just like I have done in my post #783, #804 (using ASIO4ALL) and #1,021 (using VB-MATRIX).
In any way, if the possible "trigger lags" and/or "drift of synchronization" wound be within acceptable time span, say within 10 msec or 5 msec, and if such minimum sync issues would give you no subjective audible issue at all, then you may proceed/apply your routing configuration as I wrote in my post here #393.
It is important and worthwhile, however, to know the possible nature and amount of "trigger lags" and "sync drift" of your multichannel DAC system including their consistency.
I'm not disputing your observations. It's just that I've been using this setup for multichannel playback for quite some time now and I haven't noticed any audible symptoms.
Now granted, I'm only using it for content consumption. That's the benefit of the rme interface. The asio driver remains available for daw use. Wouldn't trust this amalgamation of 3-4 devices for anything serious obviously.
But for the purpose I'm using it, it's proven remarkably stable and resilient. I've also measured it with rew quite a few times, with my original delays applied and haven't seen any deviations. The delays I've set for the mlp whenever I first set the system up, still keep the signals in sync, in multiple measurements spanning several months.
Out of curiosity, how would the problem you're describing manifest in an observable way without hunting for it? Has it ever troubled you in regular use and how?
This post space is reserved for my response to @Adahn. 
Hopefully I will respond to you this evening after completing my rather busy end-of-the-year business and private duties (my wife is serious!) today...
OK, now I am back!
In order to properly and adequately respond to your kind inquiry, I believe I need talking briefly about my “historic” sequences in time-alignment (and phase-continuity) measurements and tunings in my multichannel multi-SP-driver multi-amplifier full-active stereo audio system.
Even before I could get OKTO DAC8PRO (8-Ch sync DAC unit), I semi-objectively measured and noticed through REW’s wavelet analysis of room air sound that the sound of my L&R sub-woofers delays about 15 msec – 25 msec against the main SP system (ref. #17, #18, #20, #21, #22).
After the arrival of OKTO DAC8PRO at my listening room in May 2020 (ref. #92), I explored my rather long and intensive journey towards selection and determination of four amplifiers to dedicatedly and directly drive woofers, midranges, tweeters and super-tweeters (my L&R large and heavy sub-woofers, YAMAHA YST-SW1000 has built-in powerful amplifiers). You can find the summary of my amplifier selection journey based on my policy of “right-person-in-right-place” in my post #311 and #413; throughout the journey I have been using JRiver MC, ASIO4ALL, DSP “EKIO” and DAC8PRO, i.e. I have/had no trigger-lag nor sync-drift issue among the eight (8) DAC channels.
Then, after I almost completed/established my multichannel audio system in January 2022 (ref. #508), I decided to perform rather intensive measurements and tunings on time-alignments (and phase-continuities) among all of the five SP drivers (i.e. sub-woofer, woofer, midrange, tweeter and super-tweeter) by my somewhat unique but fully validated reliable reproducible measurement methods; I would like to avoid any loop-back analysis like REW does/did, but would like to establish my own primitive and simple reliable methods.
I started such approach firstly for the time-alignment between sub-woofer to main SP’s woofer in 1 msec precision applying “Precision Time-shifted Multiple Fq Rectangular Sine Tone Burst Method” (ref. #493), and also applying “Tone Burst Energy Peak Matching Method” (ref. #494). Both of the two methods unanimously gave the result that the subwoofer sound delays 16.0 msec against main SP system, and hence the 16.0 msec group delay settings in DSP “EKIO” for woofer, midrange, tweeter and super-tweeter have been justified for time-alignment with subwoofer.
Next, I moved on to 0.1 msec precision time-alignment between woofer and midrange by applying “Precision Single Sine Wave Matching Method” (ref. #504). I believe this method is a kind of ultimate method/goal for not only precision time-alignment but also for optimization of phase-continuity between the two SP drivers controlled by upstream DSP, and several people in ASR agreed it is so. As the results, I found that my woofer delays 0.3 msec against midrange, and there were/are no delay between midrange, tweeter and super-tweeter (in 0.1 msec precision which is more than enough precision).
Consequently, I could find/confirm that group delay DSP settings of 0.00 msec (none) for subwoofer, 16.0 msec for woofer, 16.3 msec for midrange, 16.3 msec for tweeter and 16.3 msec for super-tweeter were/are the optimal for time-alignment (as well as phase continuity) among all the SP drivers in my own multichannel audio setup.
I also found that the “Precision Single (or Three, Eight) Sine Wave Matching Method” and the 3D-color spectrum analysis (representing 3D sound energy distribution) thereof using ADOBE Audition 3.0.1 are also very much effective and powerful in assessment of transient characteristics (step response) of subwoofers and woofers, as well as for determination of optimal crossover Fq (ref. #504, #507, #495, #497).
I myself, as well as my wife and several audio enthusiast colleagues/musicians, can subjectively find/hear the great benefit of such complete time-alignment among the all SP drivers, for example as shared in my posts #520 and #687.
- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520
- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687
BTW, for subjective listening tests/evaluations, I have been using my own consistent “Audio Reference/Sampler Music Playlist” consists of 60 music tracks selected from various genres having excellent recording quality; I have dedicated thread (ref. here #1 on that independent thread) on such audio reference playlist.
Of course, the pros of my such time-alignment tunings remained unchanged when I implemented wide-3D reflective dispersion of super-tweeter sound using random-surface hard-heavy crystal glass material enabling improved/better stereo sound perspective/image (ref. #921, #926, #927, #929).
After briefly talking all the above, and after your kind understanding on the above, now I shall go into my response to your present inquiry.
In September 2023, while I was fully enjoying music listening with thus established my main multichannel audio setup, just for my personal curiosity (nothing more, nothing less) I thought again about my long-lasting interest on “possible (or not) synchronization of multiple independent DAC units (each has independent ASIO driver) in Windows-PC-based audio setup”.
I decided performing my rather strict “experiments” on this regard (completely in outside of my main audio system) as I shared in my posts #783 and #804 which you have been already reading through; at that time, I was still using ASIO4ALL as system-wide ASIO routing center.
Now you can understand why I was/am concerning about possible synchronization of multiple independent DAC units in 0.1 msec precision/accuracy; please be reminded that I have established 0.1 msec precision time-alignments in my main audio system, and therefore I need 0.1-msec-level precision of synchronization if it might be possible to synchronize multiple independent DAC units.
As you are already aware of, unfortunately, even using ASIO4ALL, I clearly measured/observed the “trigger lags” between the multiple independent DAC units; the “synchronization drift” after the trigger through the end of the music track, however, was negligible/undetectable; I mean the “trigger lags” were/are strictly kept throughout the playback of the music track. In any way, as far as we have “trigger lags”, we should not use such multiple independent DAC units in our PC ASIO driver based audio system.
This “trigger lag” measurements led me to the compromise of quasi-synchronization (or I should say sham-synchronization) of multiple independent DAC units in 0.1 msec precision by applying suitable “group delay” in upstream DSP configuration, but such compromise is valid only if the “trigger lags” would be strictly maintained/kept unchanged throughout our audio listening sessions on the day, and/or weeks, months, in the specific audio setup; we need to strictly “measure” the “trigger lags” to be compensated by DSP’s group-delay settings every time before our listening session, and we should confirm it has been maintained/kept at the end of the listening session.
Consequently, such DSP-based quasi-compensation of “trigger lags” would not be useful for our daily music listening sessions, and definitely it should not be recommended for our daily utilization; I mean that I could found the way of DSP-based tentative compensation of trigger-lags only for my curiosity.
After all the above, I replaced ASIO4ALL by VB-AUDIO MATRIX because of much better GUI operation in MATRIX as well as its stability/robustness as system-wide ASIO/VASIO/VAIO routing center (ref. #851, #854 and #858).
Recently, on November 27, when I received the update notice from VB-AUDIO on their MATRIX 1.0.2.5, I was rather impressed by their description of better/improved “synchronization” among the ASIO, VASIO, VAIO physical/virtual audio devices (ref. #1,016) which led me to the similar measurements (ref. #1,021) just like I have done in September 2023 using ASIO4ALL (ref. #783, #804); I had faint expectation of possible “trigger synchronization” by VB-MATRIX 1.0.2.5 among the multiple independent DAC units each has its own ASIO (and/or WDM/WASAPI) drivers.
Unfortunately, as shared in detail in my post #1,021, even using “Internal Clock” as sync-Master in VB-MATRIX 1.0.2.5, the “trigger lags” still do exist among the multiple DAC units which was also confirmed by kind response given by one of the VB-AUDIO’s engineering staffs as follows;
The ”Strict” synchronization feature of VB-MATRIX 1.0.2.5 is not intending your point of trigger synchronization of multiple USB-DAC units; VB-MATRIX strictly simultaneously sends the music (sound) track into connected multiple DAC units, but the exact trigger timing of each DAC would be dependent on its own ASIO or WDM (WASAPI) drivers as well as Windows audio playback priorities which are out of control by VB-MATRIX, as you have suggested.
Consequently, just as you find in your posts #783 and #804, there would be always some trigger timing lags between the USB-DAC units unless otherwise they have some “sync trigger” mechanism with each other through their ASIO driver just like you have already described in your post #842 for RME Fireface UFX III.
In VB-MATRIX side, however, you may optimize/minimize the “trigger timing lags” by settings VAIO sync option to “STRICT” and change Matrix Latency Performance Mode to “Optimal”.
Consequently, as I wrote in #1,021, my present “conclusions” are;
Even with VB-MATRIX 1.0.2.5, we cannot exactly/strictly synchronize the “triggers” of multiple independent USB-DAC units; the trigger timing lags depend on each ASIO and/or WDM (WASAPI) driver as well as Windows audio playback priority orders.
We can optimize/minimize the trigger timing lags by VB-MATRIX’s sync option “STRICT” and latency performance “Optimal”.
And as I wrote here #389 on the remote thread, I believe we should clearly separate the two points/issues in this regard of multiple-channel DAC processing;
1. trigger (kick-off/start-up) synchronization (or not) among the DAC channels, even if the strict synchronization (drift correction) could be achieved afterwards,
2. clock synchronization = drift correction among the multiple DAC channels after starting/triggering playback of a music track.
I also wrote there as follows;
We can completely "solve" both of 1. and 2. by using sync multichannel DAC unit like OKTO DAC8PRO (I use it), TOPPING DM7, all-new MOTU 16A, etc. all of which have dedicated multichannel USB ASIO driver.
We can also "solve" the issues if we can achieve all synchronized AES/EBU digital inputs into the multichannel DAC (or such multiple DAC units) if they have AES/EBU digital inputs (like OKTO DAC8PRO), since AES/EBU digital signal contains clock sync pulses in it. This has been widely achieved in pro audio market in multichannel recording and sound editing.
Furthermore, even in the case of USB ASIO configuration, we can fully synchronize certain "same brand" multiple DAC units (like RME Fireface UFX III) if they have some "sync trigger and drift correction" mechanism with each other through their dedicated USB ASIO driver (ref. #842 on my project thread). In the case, each of the multiple DAC units needs to be connected to USB ports of PC (or Mac) so that the ASIO driver can recognize all the sync-aggregated DAC channels.
Furthermore, I also wrote in my post #393 on that thread as follows;
You would please do not be too much "strict" on your audio system.
You may use multiple DAC units (with preamplifier gain controls) if the trigger timing difference(s)/lag(s) and the possible synchronization drift(s) are within acceptable time scale, say within 10 msec, and if you have no audible "issues" on it. You would please trust your own "subjective listening preference" with your ears and brain at your listening position. Our ears and brain are not identical to measurement microphone and sound analysis software.
Objective measurements of trigger mismatch and/or drift of synchronization (or not), however, would be worthwhile to objectively know the "actual room sound status" of your system with which you usually enjoy music listening excitements. Based on such objective data, sometimes further tuning would be possible for better subjective assessment/feeling of the system-sound in your acoustic listening environments.
I hope all the above would be properly and adequately responding to your present kind inquiry.
Edit:
You would please find in #931 and #1,009 the details of my latest system setup.
You can find here and here the Hyperlink Index for my project thread.
Hopefully I will respond to you this evening after completing my rather busy end-of-the-year business and private duties (my wife is serious!) today...
OK, now I am back!
In order to properly and adequately respond to your kind inquiry, I believe I need talking briefly about my “historic” sequences in time-alignment (and phase-continuity) measurements and tunings in my multichannel multi-SP-driver multi-amplifier full-active stereo audio system.
Even before I could get OKTO DAC8PRO (8-Ch sync DAC unit), I semi-objectively measured and noticed through REW’s wavelet analysis of room air sound that the sound of my L&R sub-woofers delays about 15 msec – 25 msec against the main SP system (ref. #17, #18, #20, #21, #22).
After the arrival of OKTO DAC8PRO at my listening room in May 2020 (ref. #92), I explored my rather long and intensive journey towards selection and determination of four amplifiers to dedicatedly and directly drive woofers, midranges, tweeters and super-tweeters (my L&R large and heavy sub-woofers, YAMAHA YST-SW1000 has built-in powerful amplifiers). You can find the summary of my amplifier selection journey based on my policy of “right-person-in-right-place” in my post #311 and #413; throughout the journey I have been using JRiver MC, ASIO4ALL, DSP “EKIO” and DAC8PRO, i.e. I have/had no trigger-lag nor sync-drift issue among the eight (8) DAC channels.
Then, after I almost completed/established my multichannel audio system in January 2022 (ref. #508), I decided to perform rather intensive measurements and tunings on time-alignments (and phase-continuities) among all of the five SP drivers (i.e. sub-woofer, woofer, midrange, tweeter and super-tweeter) by my somewhat unique but fully validated reliable reproducible measurement methods; I would like to avoid any loop-back analysis like REW does/did, but would like to establish my own primitive and simple reliable methods.
I started such approach firstly for the time-alignment between sub-woofer to main SP’s woofer in 1 msec precision applying “Precision Time-shifted Multiple Fq Rectangular Sine Tone Burst Method” (ref. #493), and also applying “Tone Burst Energy Peak Matching Method” (ref. #494). Both of the two methods unanimously gave the result that the subwoofer sound delays 16.0 msec against main SP system, and hence the 16.0 msec group delay settings in DSP “EKIO” for woofer, midrange, tweeter and super-tweeter have been justified for time-alignment with subwoofer.
Next, I moved on to 0.1 msec precision time-alignment between woofer and midrange by applying “Precision Single Sine Wave Matching Method” (ref. #504). I believe this method is a kind of ultimate method/goal for not only precision time-alignment but also for optimization of phase-continuity between the two SP drivers controlled by upstream DSP, and several people in ASR agreed it is so. As the results, I found that my woofer delays 0.3 msec against midrange, and there were/are no delay between midrange, tweeter and super-tweeter (in 0.1 msec precision which is more than enough precision).
Consequently, I could find/confirm that group delay DSP settings of 0.00 msec (none) for subwoofer, 16.0 msec for woofer, 16.3 msec for midrange, 16.3 msec for tweeter and 16.3 msec for super-tweeter were/are the optimal for time-alignment (as well as phase continuity) among all the SP drivers in my own multichannel audio setup.
I also found that the “Precision Single (or Three, Eight) Sine Wave Matching Method” and the 3D-color spectrum analysis (representing 3D sound energy distribution) thereof using ADOBE Audition 3.0.1 are also very much effective and powerful in assessment of transient characteristics (step response) of subwoofers and woofers, as well as for determination of optimal crossover Fq (ref. #504, #507, #495, #497).
I myself, as well as my wife and several audio enthusiast colleagues/musicians, can subjectively find/hear the great benefit of such complete time-alignment among the all SP drivers, for example as shared in my posts #520 and #687.
- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520
- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687
BTW, for subjective listening tests/evaluations, I have been using my own consistent “Audio Reference/Sampler Music Playlist” consists of 60 music tracks selected from various genres having excellent recording quality; I have dedicated thread (ref. here #1 on that independent thread) on such audio reference playlist.
Of course, the pros of my such time-alignment tunings remained unchanged when I implemented wide-3D reflective dispersion of super-tweeter sound using random-surface hard-heavy crystal glass material enabling improved/better stereo sound perspective/image (ref. #921, #926, #927, #929).
After briefly talking all the above, and after your kind understanding on the above, now I shall go into my response to your present inquiry.
In September 2023, while I was fully enjoying music listening with thus established my main multichannel audio setup, just for my personal curiosity (nothing more, nothing less) I thought again about my long-lasting interest on “possible (or not) synchronization of multiple independent DAC units (each has independent ASIO driver) in Windows-PC-based audio setup”.
I decided performing my rather strict “experiments” on this regard (completely in outside of my main audio system) as I shared in my posts #783 and #804 which you have been already reading through; at that time, I was still using ASIO4ALL as system-wide ASIO routing center.
Now you can understand why I was/am concerning about possible synchronization of multiple independent DAC units in 0.1 msec precision/accuracy; please be reminded that I have established 0.1 msec precision time-alignments in my main audio system, and therefore I need 0.1-msec-level precision of synchronization if it might be possible to synchronize multiple independent DAC units.
As you are already aware of, unfortunately, even using ASIO4ALL, I clearly measured/observed the “trigger lags” between the multiple independent DAC units; the “synchronization drift” after the trigger through the end of the music track, however, was negligible/undetectable; I mean the “trigger lags” were/are strictly kept throughout the playback of the music track. In any way, as far as we have “trigger lags”, we should not use such multiple independent DAC units in our PC ASIO driver based audio system.
This “trigger lag” measurements led me to the compromise of quasi-synchronization (or I should say sham-synchronization) of multiple independent DAC units in 0.1 msec precision by applying suitable “group delay” in upstream DSP configuration, but such compromise is valid only if the “trigger lags” would be strictly maintained/kept unchanged throughout our audio listening sessions on the day, and/or weeks, months, in the specific audio setup; we need to strictly “measure” the “trigger lags” to be compensated by DSP’s group-delay settings every time before our listening session, and we should confirm it has been maintained/kept at the end of the listening session.
Consequently, such DSP-based quasi-compensation of “trigger lags” would not be useful for our daily music listening sessions, and definitely it should not be recommended for our daily utilization; I mean that I could found the way of DSP-based tentative compensation of trigger-lags only for my curiosity.
After all the above, I replaced ASIO4ALL by VB-AUDIO MATRIX because of much better GUI operation in MATRIX as well as its stability/robustness as system-wide ASIO/VASIO/VAIO routing center (ref. #851, #854 and #858).
Recently, on November 27, when I received the update notice from VB-AUDIO on their MATRIX 1.0.2.5, I was rather impressed by their description of better/improved “synchronization” among the ASIO, VASIO, VAIO physical/virtual audio devices (ref. #1,016) which led me to the similar measurements (ref. #1,021) just like I have done in September 2023 using ASIO4ALL (ref. #783, #804); I had faint expectation of possible “trigger synchronization” by VB-MATRIX 1.0.2.5 among the multiple independent DAC units each has its own ASIO (and/or WDM/WASAPI) drivers.
Unfortunately, as shared in detail in my post #1,021, even using “Internal Clock” as sync-Master in VB-MATRIX 1.0.2.5, the “trigger lags” still do exist among the multiple DAC units which was also confirmed by kind response given by one of the VB-AUDIO’s engineering staffs as follows;
The ”Strict” synchronization feature of VB-MATRIX 1.0.2.5 is not intending your point of trigger synchronization of multiple USB-DAC units; VB-MATRIX strictly simultaneously sends the music (sound) track into connected multiple DAC units, but the exact trigger timing of each DAC would be dependent on its own ASIO or WDM (WASAPI) drivers as well as Windows audio playback priorities which are out of control by VB-MATRIX, as you have suggested.
Consequently, just as you find in your posts #783 and #804, there would be always some trigger timing lags between the USB-DAC units unless otherwise they have some “sync trigger” mechanism with each other through their ASIO driver just like you have already described in your post #842 for RME Fireface UFX III.
In VB-MATRIX side, however, you may optimize/minimize the “trigger timing lags” by settings VAIO sync option to “STRICT” and change Matrix Latency Performance Mode to “Optimal”.
Consequently, as I wrote in #1,021, my present “conclusions” are;
Even with VB-MATRIX 1.0.2.5, we cannot exactly/strictly synchronize the “triggers” of multiple independent USB-DAC units; the trigger timing lags depend on each ASIO and/or WDM (WASAPI) driver as well as Windows audio playback priority orders.
We can optimize/minimize the trigger timing lags by VB-MATRIX’s sync option “STRICT” and latency performance “Optimal”.
And as I wrote here #389 on the remote thread, I believe we should clearly separate the two points/issues in this regard of multiple-channel DAC processing;
1. trigger (kick-off/start-up) synchronization (or not) among the DAC channels, even if the strict synchronization (drift correction) could be achieved afterwards,
2. clock synchronization = drift correction among the multiple DAC channels after starting/triggering playback of a music track.
I also wrote there as follows;
We can completely "solve" both of 1. and 2. by using sync multichannel DAC unit like OKTO DAC8PRO (I use it), TOPPING DM7, all-new MOTU 16A, etc. all of which have dedicated multichannel USB ASIO driver.
We can also "solve" the issues if we can achieve all synchronized AES/EBU digital inputs into the multichannel DAC (or such multiple DAC units) if they have AES/EBU digital inputs (like OKTO DAC8PRO), since AES/EBU digital signal contains clock sync pulses in it. This has been widely achieved in pro audio market in multichannel recording and sound editing.
Furthermore, even in the case of USB ASIO configuration, we can fully synchronize certain "same brand" multiple DAC units (like RME Fireface UFX III) if they have some "sync trigger and drift correction" mechanism with each other through their dedicated USB ASIO driver (ref. #842 on my project thread). In the case, each of the multiple DAC units needs to be connected to USB ports of PC (or Mac) so that the ASIO driver can recognize all the sync-aggregated DAC channels.
Furthermore, I also wrote in my post #393 on that thread as follows;
You would please do not be too much "strict" on your audio system.
You may use multiple DAC units (with preamplifier gain controls) if the trigger timing difference(s)/lag(s) and the possible synchronization drift(s) are within acceptable time scale, say within 10 msec, and if you have no audible "issues" on it. You would please trust your own "subjective listening preference" with your ears and brain at your listening position. Our ears and brain are not identical to measurement microphone and sound analysis software.
Objective measurements of trigger mismatch and/or drift of synchronization (or not), however, would be worthwhile to objectively know the "actual room sound status" of your system with which you usually enjoy music listening excitements. Based on such objective data, sometimes further tuning would be possible for better subjective assessment/feeling of the system-sound in your acoustic listening environments.
I hope all the above would be properly and adequately responding to your present kind inquiry.
Edit:
You would please find in #931 and #1,009 the details of my latest system setup.
You can find here and here the Hyperlink Index for my project thread.
Last edited:
Appreciate the detailed writeup. If I did not misread anything, the problem is not drifting sync as I initially understood, but miniscule sample level delays between the various interfaces of the aggregate device.
Assuming a sample rate of 48k, 0.1ms is what? 5 samples or something? I doubt that kind of delay can be fixed and I'm certain it can't be perceived in a surround system. Moving your head a couple cm to the side will produce more of a delay. Even when connecting 2 rme interfaces with adat and synced clocks, you're gonna have a delay of 3-6 samples, that's unavoidable.
You might be overthinking this a tad, especially since the purpose of the system is purely listening and not simultaneous recording. I welcome your inquisitive exploration of these limitations nonetheless though.
Assuming a sample rate of 48k, 0.1ms is what? 5 samples or something? I doubt that kind of delay can be fixed and I'm certain it can't be perceived in a surround system. Moving your head a couple cm to the side will produce more of a delay. Even when connecting 2 rme interfaces with adat and synced clocks, you're gonna have a delay of 3-6 samples, that's unavoidable.
You might be overthinking this a tad, especially since the purpose of the system is purely listening and not simultaneous recording. I welcome your inquisitive exploration of these limitations nonetheless though.
Essentially, I agree with you; 1 msec precision in time-alignment tuning would be enough in ordinary multichannel home audio setups, I assume.
On the other hand, I believe the unacceptable "trigger lags" (up to ca. 20 msec in my experiments), however, are caused/dependent on Windows multiple ASIO trigger sequence and/or trigger priorities which are out-of-control by ASIO4ALL or VB-MATRIX; VB-AUDIO's staff kindly agreed and confirmed my experimental findings.
Just for your reference and interest, I usually convert on-the-fly (by JRiver MC) all the tracks into 88.2 kHz PCM (or sometimes into 96 kHz); for my personal rationales for such on-the-fly sample rate conversion, you would please refer to my post #532.
I essentially agree with you. This is why I also quoted/referred again my post #393 on that thread...
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Good news. ChatGPT helped me solve the Windows global volume control issue. Solution is right click on the VAIO aggregate device from within Matrix, open VAIO Control Panel, click Options menu at upper left, and click Enable Windows Volume Control. BUT Matrix must be run as Administrator in order to make the change persistent.
With that fix, now all is well. I am really happy with this setup, and I think it sounds excellent!
Good info for others reading the thread. In my setup I do only asio devices in matrix and passing them to voicemeeter for the wdm stuff, so haven't had to deal with this quirk. Glad to hear it's all working out for you.
10PSPMemSticksForMiku2007
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If you do want to go ahead with using a PC-based crossover, which is what I do, your software setup is quite over complicated just to use two DACs. If your motherboard built in audio is not good enough for some reason, I would recommend a sound blaster or xonar usb device.
I would suggest giving your motherboard audio a proper chance. Might not be good for headphones, probably fine for amps and speakers. Implementation varies by motherboard but I'm happy with the quality of my Realtek ALC1220 on ASRock Z370 Extreme4 so long as I have good gain structure. I got a decent loopback measurement from Rightmark. My speakers sound good and measure well. However if I have the amps turned up too much, or if I connect sensitive IEMs directly, there is too much noise coming from the PC. Other than the noise issue, there are no problems with the sound quality.
I can not think of desktop speakers and amps with enough distortion-free output to need a better DAC than this. Even my old Polk towers hit their limits long before the noise floor from this PC becomes audible, and I am adjusting volume in software. There are noise issues with heavy processing loads and gaming, but not so much with reading news and playing music. I imagine many people find USB DACs sound better only because the Windows limiter is less aggressive on digital outputs. You can disable it entirely with a regristry edit however.
You can get decent quality consumer 6 or 8 channel dacs with ASIO support for about 100 bucks. For simplicity, I use media players that output to DirectSound so I don't have to worry about a second DSP chain for ASIO. Some people really need ASIO for production, but for most listening setups it is unimportant.
There is bass management built into Windows, but I recommend not using it, as it reduces volume on all channels too much. I'd rather use EqualizerAPO to have full control. I leave only a few db headroom to avoid Windows audio limiter.
Your easy solution probably is finding a cheap used AVR. If your setup is 7.1 or less, you don't need any modern HDMI features or Atmos. You may not need HDMI at all. An outdated AVR with decent amps is a great solution for PC audio.
I would suggest giving your motherboard audio a proper chance. Might not be good for headphones, probably fine for amps and speakers. Implementation varies by motherboard but I'm happy with the quality of my Realtek ALC1220 on ASRock Z370 Extreme4 so long as I have good gain structure. I got a decent loopback measurement from Rightmark. My speakers sound good and measure well. However if I have the amps turned up too much, or if I connect sensitive IEMs directly, there is too much noise coming from the PC. Other than the noise issue, there are no problems with the sound quality.
I can not think of desktop speakers and amps with enough distortion-free output to need a better DAC than this. Even my old Polk towers hit their limits long before the noise floor from this PC becomes audible, and I am adjusting volume in software. There are noise issues with heavy processing loads and gaming, but not so much with reading news and playing music. I imagine many people find USB DACs sound better only because the Windows limiter is less aggressive on digital outputs. You can disable it entirely with a regristry edit however.
You can get decent quality consumer 6 or 8 channel dacs with ASIO support for about 100 bucks. For simplicity, I use media players that output to DirectSound so I don't have to worry about a second DSP chain for ASIO. Some people really need ASIO for production, but for most listening setups it is unimportant.
There is bass management built into Windows, but I recommend not using it, as it reduces volume on all channels too much. I'd rather use EqualizerAPO to have full control. I leave only a few db headroom to avoid Windows audio limiter.
Your easy solution probably is finding a cheap used AVR. If your setup is 7.1 or less, you don't need any modern HDMI features or Atmos. You may not need HDMI at all. An outdated AVR with decent amps is a great solution for PC audio.
Thanks for your reply. My motherboard out is stereo only, so that wouldn't help me. I also find my amp and speakers sound better than my headphones. Given I am not satisfied with the motherboard's onboard audio out through headphones, compared to my dedicated DDC+DAC, I definitely don't want to degrade the source feeding my amplifiers and speakers.
OP
-Jim-
Member
- Thread Starter
- #39
Gents, you took my relatively "simple" thread over and drove it to regions way off topic => at least for me. I tried to follow where you were going but without the software (which looked interesting but complex) and walking through it's implementation, it all became too much to follow and relate it to my "simple" issue.
So back to my request for assistance please?
In the many weeks since my last post, I went down the Asus Product Support wormhole with numerous emails without any files, or possible solutions coming my way. We went on a Caribbean Cruise for a couple weeks (Jan 20 to Feb 4), and of course that's when they sent me the definitive email. Here's the just of it (edited my last name out for privacy):
Hello Jim,
Thank you for contacting ASUS Product Support. My name is Nino V. and it's my pleasure to help you with your problem.
My name is Nicko V.. First and foremost, thank you for your patience while we completed our review of this issue. We apologize for the delay and are grateful for the opportunity to continue to assist you.
Please process an RMA to send your unit for repair,
For the RMA process it will take about 7-10business days not including the shipping and the weekends. and also for the motherboard our facility will going to repair the unit but if the unit is unrepairable they are going to notify you regarding with the replacement recertified unit.
The recertified unit comes direct from the manufacturing site and has not been used by any customers but may have new and/or remanufactured parts inside the unit; however we cannot classify it as a brand new unit since these are warranty units that are not intended for resale. This is also standard or common practice for most electronic devices when utilizing the manufacturer’s warranty.
I did have a Chromebook with me and connected to the Ship's Wi-Fi was able to respond as follows.
Hi Nino V.,
Thank you for the email. I am amazed Asus has not done anything to rectify this issue over the many weeks (now months?) since I brought it to your attention.
Unfortunately I am out of the country for a couple weeks and only have a Chromebook with very limited email access with me. When I return home I will need my computer to catch up on many things that have occurred in my absence. My experience with RMAs is somewhat limited; but basically with transportation, weekends, etc., a 6 week cycle door to door is probably to be expected as a quick turn around, and something in the area of two months is not at all uncommon. I cannot function with my Main Computer out of service for either of those durations.
My understanding is very few motherboards get repaired with this process and typically a replacement is sent. So would it be possible for you to just send me one and I will re & re it withing 48 hours of receipt and courier the damaged one to you? Or failing that, as it's only the Audio section of the motherboard that's not functioning correctly, could you just ship me an Asus Xonar AE PCI sound card (or any decent PCI soundcard) and we can get this issue dealt with quickly?
Thanks for your cooperation and assistance. I look forward to your prompt reply.
Regards,
Jim
I sent that from the middle of the ocean on January 28th and Asus responded (below) on February 9th. (Ugh!)
Hello Jim ,
Thank you for contacting ASUS Product Support.
My name is Nicko V.. First and foremost, thank you for your patience while we completed our review of this issue. We apologize for the delay and are grateful for the opportunity to continue to assist you.
We unfortunately are not able to provide the customer with an ASUS sound card. We are only able to provide accessories that came with the motherboard, but it is subject to availability and eligibility
We would like to know if you have any further questions, comments and/or concerns, please do not hesitate to let us know, we will be glad to assist you directly.
1-812-282-2787 or 1-571-918-6030 (Monday - Friday, 6AM - 9PM PST - Saturday - Sunday, 6AM - 5PM PST )
Your case number for your future reference: CASE NO= N2511028163-0015
I must confess that once I got home and was recovering from the 4 hour Time shift and Jet Lag, I got the idea to check the warranty coverage for the motherboard. And using the serial number the Asus Tool said warranty expired in August last year based on the date of manufacture! I figured that that could be the end of the Asus Support (not that I've really had any from them) but I pressed on.
Last night, after receipt of their last email, I dutifully supplied all of the information in the RMA process, and I sent them a copy of my purchase receipt date February 7, 2023. Now I wonder if they'll say the warranty expired 2 days before I sent the form, which was after they confirmed I needed to go ahead with the RMA process (as they won't "provide the customer with an ASUS sound card")? Or will they revert to when I identified and documented the issue with them last November 25th?
Only time will tell. The RMA says they will respond within 48 hours.
So I've developed a plan B. I'm thinking of getting a decent PCI-e Soundcard without breaking the bank; and have narrowed it down to 2 but really worry about Windows 11 Drivers that work reliably as I've seen posts to the contrary.
Asus Xonar AE or the Creative Audigy FX V2
The plan is to only use the soundcard to send a 2.1 signal (to a Klipsch Sub directly and a small amp for a pair of JBL Loft 50s that I got for a song) and nothing more complex. Do you think these soundcards are suitable for the task, or is there something else that's as cost effective?
Thanks in advance for your suggestions.
OP
-Jim-
Member
- Thread Starter
- #40
Gents,
As I sort of expected, Asus denied the warranty claim yesterday afternoon:
Hello Jim,
You are receiving this email to hereby acknowledge receipt of your online RMA request. Thank you for allowing us to help you with your repair! We appreciate your loyalty to our products.
This is also a confirmation that based on your date of purchase (if provided) and/or serial number your ASUS product is no longer within the standard ASUS manufacturer warranty period. Please be advised that warranty terms vary depending on what type of ASUS product you have. Warranty applies only to products that are purchased new on the date of purchase from an authorized ASUS product reseller (we do not accept peer-to-peer sale invoices, such as, eBay, Amazon Market Place, etc.).
If you would like to continue with the RMA process, please continue with the provided link to pay the required, nonrefundable Diagnostic Fee. After payment is submitted, a confirmation email along with RMA instructions will be sent to your email address. Please be aware that this fee does not cover the actual repair costs, which will be provided once diagnosis has been completed by the technician at our repair center.
Service Case Number: N2511028163-0017
Serial Number: N4M0KK07A8209P7
Once again, thank you for choosing ASUS!
Best regards,
ASUS Customer Support
So I'm not going to pay for them to fix their obvious defect since day 1 of ownership. Now I'm looking at defeating the Realtek Sound in Bios and installing a better PCI-e Soundcard.
Which is better?
Asus Xonar AE or the Creative Audigy FX V2
Your opinion matters. Please advise.
As I sort of expected, Asus denied the warranty claim yesterday afternoon:
Hello Jim,
You are receiving this email to hereby acknowledge receipt of your online RMA request. Thank you for allowing us to help you with your repair! We appreciate your loyalty to our products.
This is also a confirmation that based on your date of purchase (if provided) and/or serial number your ASUS product is no longer within the standard ASUS manufacturer warranty period. Please be advised that warranty terms vary depending on what type of ASUS product you have. Warranty applies only to products that are purchased new on the date of purchase from an authorized ASUS product reseller (we do not accept peer-to-peer sale invoices, such as, eBay, Amazon Market Place, etc.).
If you would like to continue with the RMA process, please continue with the provided link to pay the required, nonrefundable Diagnostic Fee. After payment is submitted, a confirmation email along with RMA instructions will be sent to your email address. Please be aware that this fee does not cover the actual repair costs, which will be provided once diagnosis has been completed by the technician at our repair center.
Service Case Number: N2511028163-0017
Serial Number: N4M0KK07A8209P7
Once again, thank you for choosing ASUS!
Best regards,
ASUS Customer Support
So I'm not going to pay for them to fix their obvious defect since day 1 of ownership. Now I'm looking at defeating the Realtek Sound in Bios and installing a better PCI-e Soundcard.
Which is better?
Asus Xonar AE or the Creative Audigy FX V2
Your opinion matters. Please advise.
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