Does ReaInsert work yet?

[SonicAxiom]

Originally Posted by msmucr

Just for my curiosity, after occurrence of it, have you tried to save project, close reaper and open it again? If it remedy issue and delay skew will be fixed, i would suspect some issue with internal PDC recalculation (Reainsert shouldn't be any different than other compensated plugins). If it stays, i would probably address it to driver/hardware (I don't know, some dynamic buffering for instance, but it will be very uncommon).

Yes, I saved, closed and reopened the projects. Sometimes, everything stays correct, sometimes I have to correct the sample value slightly.

[EpicSounds]

now I can't get ReaInsert to not work right in V4.77.
But I do have 5 also installed using the same resources folder. This is weird because I could only get it close enough to cause comb filtering just a few weeks ago.

In preferences Audio>Recording I have "Use driver reported latency" checked, and 0 for both recording and playback offsets.

In Reainsert I disable "automatic device latency adjustment", bypass my hardware compressor, then click ping detect.

At 512 buffer size, 44.1kHz the reported latency is 1664 samples. The ping detected additional latency is at 1237.

It also works if you disable the device reported latency, and the automatic latency adjustment, then manual ping.

On my system ReaInsert breaks IF:

If driver reported latency and automatic device latency adjustment both active = very phasey with 50% mix

If driver reported latency is OFF and automatic device latency adjustment is ON.

If I have my hardware compressor active = off by 1 sample

If I disable delay compensation in the plugin


Tested this on master bus with ART PRO VLA II.

[LDstudios]

Originally Posted by EpicSounds

Tested this on master bus with ART PRO VLA II.

I think the peak ADC experience is when an analogue insert is added with parallel routing. For example, bussing your drums to two groups - one with maybe some slight eqing, and the other with hardware eqing and compression. If they can exist side by side without comb filtering we are onto a winner.

It will also give the plugin delay compensation a workout as well. It should be possible to add & remove plugins to the group without hardware inserts on it while retaining phase coherency.

[EpicSounds]

I did set mix to 50%. Same also works in parallel on separate tracks. I wanted it on the master so I could better hear the phase cancellation on a more complex sound rather than one element of the mix.

Even 1 sample off results in strong phase cancellation, Comb Filtering, Sea shell effect whatever you want to call it.

[LDstudios]

Originally Posted by EpicSounds

I did set mix to 50%. Same also works in parallel on separate tracks. I wanted it on the master so I could better hear the phase cancellation on a more complex sound rather than one element of the mix.

Even 1 sample off results in strong phase cancellation, Comb Filtering, Sea shell effect whatever you want to call it.

Yeah absolutely! Even with absolute sample for sample accuracy there is a chance that latency in the DA -> AD loop is a fraction of sample. Too small for any program dealing with 1 sample adjustments to actually compensate for.

My reasoning behind the parallel bussing is just to see how functional it is when aligning against other plugins & channels in the mix.

[Electric Ring]

I m on 4.77, on a Macbook Pro with OS X 10.10.3 and have been measuring latency with a pass through cable (unbalanced output directly to an input channel). I use a pulse wave form (I design audio hardware for a living, and have done lots of latency validation before, pulses or a 15 kHz single cycle sin burst are the most accurate methods). A single pulse is better if the latency is long. So if latency is fully compensated, I should be able to send out my pulse, record it on another track, and the two tracks should both line up in time, to the sample. Then latency to the connector would be compensated. Would not compensate for the extra latency in digital effects this way, the ones where the DSP is bypassed in analog or with true bypass would be the hardest to compensate, but I digress.

ReaInsert latency compensation just plain doesn't work. Measures 64 ms, not even close. There also appears to be about 37ms before anything is actually recorded. Nothing changes when I check to automatically compensate for driver latency (Mac means no driver settings to muck around with), or when I manually ping regardless of the automatic compensation check box. Latency does not move. Have not tried manually adjusting values yet, though I suspect they do nothing. I also tried recording in the mono latency compensated mode, which does pull the latency in to 21 ms.

Something is definitely screwy with the 4.77 code. I tried increasing the additional delay compensation field, and while you can shift the recorded pulse in time, it does not behave like one would expect, i.e., as you keep increasing, the latency will get smaller then increase.

For the other case, where I just route the the pulse to an output, connect input to output, and record the input track, I get a 4 ms latency. The project is 48 kHz and that is suspiciously 192 samples.

I can see how this would make Reaper unusable. Not sure when I will have time to try the new beta that supposedly fixes this issue.

[Bristol Posse]

I gave up on REinsert a couple of years ago.

now I just route each channel to an output and from there to a channel on the desk and just do everything in hardware and record the final two track back in to the DAW.

Constantly having to ping and re-ping, comb filtering, impossible to parallel process etc etc was just sucking all the time and life out of mixing.

I don't think the devs use a lot of hardware and so it's just not important to them. There have been many threads and feature requests all of which have been more or less ignored.

But hey, I paid $40 for a license toward the end of version 3 so what are you gonna do. They should just ditch it all together rather than trying to sell it as a feature but just having it half assed and not really usable

[lxm]

Originally Posted by Bristol Posse

I gave up on REinsert a couple of years ago.

now I just route each channel to an output and from there to a channel on the desk and just do everything in hardware and record the final two track back in to the DAW.

Constantly having to ping and re-ping, comb filtering, impossible to parallel process etc etc was just sucking all the time and life out of mixing.

I don't think the devs use a lot of hardware and so it's just not important to them. There have been many threads and feature requests all of which have been more or less ignored

I cant seem to get it to ever ping correctly. Always second guessing how things line up everytime I use hardware. It sucks. REALLY sucks. Kills mood and workflow. Maybe Studio One V3 is worth a go. They advertise perfect hardware integration. I need that. More and more hardware based these days. UGHHHHHHHHHHHHH

[Nixon]

Electric Ring have you tried the latest v5 pres? There has been something done regarding reainsiert. Wonder what your measurements will show here.

Edit: Oh I just read that you've noticed the betas.