Thanks Mark, but I need to be able to do ASIO loopback recording for testing purposes and also loopback remastering so sound quality is important but a plain DAC is not useful to me. My Tascam has Inserts using TRS 6.5's bypassing the input gain/level stages, so I will make suitable adaptors and see if that works out to be a bit quieter and cleaner.
Today I just got hold of the pub recording I did a year ago with my filters all over backline and the recording gear. The result is super musical, super clear and with serious uber bottom end bass and it sure don't sound like any other pub recording I have heard including the Aussie bogan cheering after each song lol. Loopback remastering has glued the mix together with narrowed central mono sources and each instrument has gained it's own space in sound and depth
A bunch of classic/vintage outboard mastering processing boxes just got make redundant !.
Dan.
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In that case, the only things I know of at low cost are the usual home recording devices such as made by Focusrite, Tascam, etc. Personally, I don't prefer the sound of them, but the better new ones I find okay for casual/demo use. Even have an old Lynx II card which was once considered quite good. Compared to DAC-3, it seems obvious here that sound quality has improved substantially over time. Be nice if it didn't take so long.
Quite different. Makita is higher RPM's resulting in faster results and less tool bit life.
George already posted a loop-back with a multi-tone file I sent him, it was pretty good.
Mercury!!!
"The U.S. Postal Service has suspended retail services at the Pittsburgh Network Distribution Center in Marshall after a chemical spill.
Spokesman Tad Kelley said the chemical was identified as mercury.
“All employees are safe and accounted for and none were exposed to the contaminant,” Kelley said Tuesday. “Professional environmental contractors are conducting tests to evaluate, clean and certify the safety of the plant.”
"The U.S. Postal Service has suspended retail services at the Pittsburgh Network Distribution Center in Marshall after a chemical spill.
Spokesman Tad Kelley said the chemical was identified as mercury.
“All employees are safe and accounted for and none were exposed to the contaminant,” Kelley said Tuesday. “Professional environmental contractors are conducting tests to evaluate, clean and certify the safety of the plant.”
A few years ago I worked for a while on the Ffestiniog Railway, there was a young lad whose wife was expecting a child and he was so enamoured by his power tool armory that if she had a girl he was thinking of calling her Makita
You're right, it's installing right now...
Will report back if they still work.
Done updating.
Happy to report ARTA/Focusrite Scarlett still work under Win10 V1809/Build 17763.379.
Done updating.
Happy to report ARTA/Focusrite Scarlett still work under Win10 V1809/Build 17763.379.
They seem to have fixed the driver, I put it aside when the problem showed up the first time.
My issue is that I need to compare loopback recordings and if possible be able to compare the original file with the recordings. My old Tascam US-122 running ASIO and Reaper DAW does a pretty good job of maintaining consistent latency but not perfectly perhaps. Does anybody have advice on what settings I can tweak to improve this loopback time accuracy if possible, and is there a means of automatically effectively time aligning the loopback recordings with the original source file ?. The reason I require common clock is that I need to eliminate clock drift out of the variables when comparing recordings.....master clock drift in itself is not particularly a problem , but it is a critical problem when comparing recordings as in this application. It might be good to use my Ebay DAC as a better source feeding the Tascam input and I expect I can do this however I will lose ability to digitally compare recordings because of clock drifts. This is not a party stopper for remastering of existing recordings but it is useless for testing and comparison purposes. Any advice is welcome, thanks in advance.
Dan.
But , while you are Listening ....
I have both a real CD output and FLAC from my PC . NOperceivable differences either way.
Old technics CD or my new ASUS soundcard , my bitperfect sounds
'perfect". so all these 1M posts ... why ???
Edit - because of some imagined subjective superiority of some digital voodoo yours sound better than others ??
Edit 2 - Maybe , if some might smoke the right dope , and have excess $$$.
OS
would this get you to curse your gear (or your mate).
I have both a real CD output and FLAC from my PC . NOperceivable differences either way.
Old technics CD or my new ASUS soundcard , my bitperfect sounds
'perfect". so all these 1M posts ... why ???
Edit - because of some imagined subjective superiority of some digital voodoo yours sound better than others ??
Edit 2 - Maybe , if some might smoke the right dope , and have excess $$$.
OS
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Dan,
Regarding your question about loopback vs original recordings, I don't understand the exact process you plan to use, but if you rip a recording to one track, then playback that track you intend to modify in some way out to your sound device, then record the loopback the return digital audio to a new track, it should be easy enough to verify if the original and the recorded tracks are time aligned (if that's what you are talking about). You can expand out the time scale to that you can examine transients down to the sample level. Easy enough to do that at a few points over the course of a piece of music and know if there is any problem with time alignment. If there is, and it is a simple linear displacement, you can disable the snap to to grid function and slide the two tracks into perfect alignment down to the individual sample level. Again, just not sure what you are trying to do.
Regarding your question about loopback vs original recordings, I don't understand the exact process you plan to use, but if you rip a recording to one track, then playback that track you intend to modify in some way out to your sound device, then record the loopback the return digital audio to a new track, it should be easy enough to verify if the original and the recorded tracks are time aligned (if that's what you are talking about). You can expand out the time scale to that you can examine transients down to the sample level. Easy enough to do that at a few points over the course of a piece of music and know if there is any problem with time alignment. If there is, and it is a simple linear displacement, you can disable the snap to to grid function and slide the two tracks into perfect alignment down to the individual sample level. Again, just not sure what you are trying to do.
Mr Tripper (indeed), You are not understanding the problem. Sure your cd/cdp and flac/pc files sound same enough on subjective testing despite clock drifts, because drift is cyclical slow variation in clock rate and not readily discernible as pitch shift.....the frequency offset value is too low plus the rate of change is slow ie 10's of seconds. However when comparing multiple recordings (digital subtraction in DAW) slow clock litter is laid bare and results in continual variation in cancellation over the length of the recordings. With loopback recording using common clock oscillator any relative clock drift between recordings is eliminated and this is mission critical when testing for subtle differences between subsequent recordings/files.
Edit 1 - This is not about claiming 'superiority' but it is about discriminating differences in cables, gear and tweaks and objective affirmation of subjective findings. But since you mentioned it, some remastering I have done lately does quite remarkably subjectively improve existing recordings....that's what remastering is about !. Edit 2 - I don't understand your second edit.....are you on this smoke you mention ?. If all systems were affordable and sounded essentially the same we could all just pack up and go do something useful. The fact is that every system is different and for varying reasons, another fact is that systems can very easily be changed subjectively (and objectively) and are not nearly as immutable as we are led to believe. What if every affordable system did sound a standard same clean, clear and musical and immune to external influence ?.....I would call that a leap forward.
Dan.
Any pro grade digital audio should have clocks stable to .5 ppm the AES minimum standard. However in that environment you would normally use a master clock to lock everything to a common reference. Looping through external devices will add some delay, especially if they are digital, not to mention the latency in the I/o links.
Consumer and audiophile grade stuff may not meet these tolerances or features since its not necessary for simple playback.
Consumer and audiophile grade stuff may not meet these tolerances or features since its not necessary for simple playback.
I am doing two things, the first is to digitally compare at least two recordings of an 'original' which can be music or defined test signal. The Tascam does a pretty good job of recording multiple loopbacks that are the 'same'...there is some amplitude noise and I suspect DAC/ADC timing noise that prevents perfect and consistent cancellation, perhaps this is a master clock jitter artifact ?. I am also not certain that the delay between commencement of PB and delivery of ADC converted signal is perfectly consistent .....isn't that what ASIO is about ?. The need for timing precision is for discriminating 'subtle' changes in system behaviour caused by changes in the loopback path, be it cable, my filtering or ferrite filtering, even Myrtle blocks lol. By adding say 10 before recordings together and adding 10 after recordings together, and then comparing these two sets of recordings, random ie white amplitude noise is averaged out and the apparent noise floor is lowered and variation in peak amplitudes is averaged out too, iow system uncertainties are averaged out and valid difference signal is somewhat better discriminated.
Each of these 20 individual recordings will have individual noisy pitch shift due to jitter and drift during the loop recording process.....who knows the maths, how well does this timing uncertainty average out when comparing the said 20 recordings, anybody ?.
DAW comparison of original file and loopback recordings is useful but not essential and relies on extremely tight relative timing of the two files to gain valid minimum of difference signal.......devilishly difficult to do in practice with mouse and 'eyeing it' despite how far the waveform is zoomed in. Is there a way to effectively compensate for the loopback latency and ensure automatic and precise effective time alignment of source file and recordings ?.
The second thing I am doing is analog processing remastering of existing music recordings but without outboard gear apart from usb soundcard and loopback cable. The particular loopback cable used makes for difference and subjective improvement of the recording when realtime subjectively comparing to the original recording. This is part of what conventional remastering does anyway, but this method is taking things to a new, different, reproducible and standard-iseable level....also customisable for any particular 'house sound'.
Dan.
Thanks.
The latency of the usb soundcard and associated I/O causes the new recording to be time shifted wrt the original source file, precluding cancellation/inversion comparison directly and precisely.
So three questions, is ASIO usb soundcard device internal latency constant ?.
Is DAW to USB device I/O latency constant ?.
If the above two questions are affirmative, is there a Reaper DAW setting that compensates for known (and constant) loop latency and causes precise apparent time alignment of source file and recording file ?.
Dan.
Thanks, excellent suggestion. I may eventually do just that.
Your link doesn't bring me to a post, and unlike in the past, manually removing the page number from the URL doesn't work. I will try incognito mode.
Put a marker pulse in you file and you will know. ASIO latency is fixed and stable, and most any DAW compensates automatically for it.
Direct subtraction of original and recording doesn't work, we've explained that before. An awful lot of sophisticated pre-processing is needed to compensate for the simple linear frequency response effects that will otherwise dominate the residual. You might want to check out Paul Kane's DeltaWave software for this: Beta-test: DeltaWave Null Comparison software | Audio Science Review (ASR) Forum
Direct subtraction only works for recording A minus recording B with only a subtle change between A and B. But even then trivial frequency and level response changes must be factored out before subtraction.
Always conduct a null baseline test with two recordings of the same state, eg. "A", being subtracted. Note that minor unavoidable clock drifts (of the common ADC/DAC clock) and reference voltage shifts can quickly spoil the null depth, again linear differences becoming dominant, especially if you use block averaging to lower the uncorelated noise. Only way to circumvent is interleaved recording of A and B.
I've spend some 10 years to refine the process of direct subtraction... if you really want to find something relevant that isn't simple linear deviations (which we consider trivial and irrelevant) it is not that easy to get reliable and meaningful results.
And IME, don't even bother with cheap ADC/DACs. Look for something at least the quality level of an RME Adi-2 Pro FS.
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