rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool

I use a pro interface although it does not have a mic pre. I have a separate analog pre and notice tenths of a ms shift between measurements when using a loopback timing reference because of the different clocks. This is workable but not entirely accurate.
you need one device to act as the master clock and the other to operate as a slave to that master clock
 
After installing the OpenDRC and checking the latency from our Antelope Orion 32 Spdif -> OpenDRC we measured 15ms latency after factory reset. With a FIR filter loaded it was 40ms. For reference we measured a DCX2496 with 100% processor load and it had a latency of 1,5ms.
15ms is too much for our application. With 40ms it's impossible to directly monitor and play piano or guitar. I'm returning the unit.

15ms seems pretty high as a base delay, even at 48kHz
Are you sure this was correctly measured, with an identical configuration as the DCX2496?
You should probably contact miniDSP support on their dedicated forum and see what they have to say about this.
 
Thanks info, does a PCI interface do that by itself on analog I/O or should/can we use its SPDIF/TOSLINK I/O and use external DAC/ADC units with build in pro terminals for bonding word clock.
I don't believe PCI interface can do this, sdpif (or adat/toslink) is one commonly available way to sync clocks. Some devices also have BNC connectors for this job.
 
At one point I tried to keep the first post up to date, but I lost track of it :(
I also try to keep the sourceforge project page up to date with links and tutorial.

Please feel free to gather links and ressources here and I will update these two pages!

Cool :) Not sure If Im up to the task gathering all ressources - But Im quite sure it would be helpful for many.
 
Just a thought. Phase-linearization could possibly be very beneficial for HT multi-channel setups, I think, where different model of speakers are used.

The multi-channel mix kind of relies on same phase all-around, I think.
Anyone tried it?
Maybe in an HT-scenario it makes sense to EQ all speaker-phases completely flat?

Maybe this is what automatic home-theater room-corrections like Dirac, Trinnov, etc, tries to do?
 
Last edited:
Dirac does exactly that. And for some other automotive customers they have an even more advanced algorithm called "Dirac Unison" that really take the concept to the edge.
19 drivers and 12 active channels with it´s own DSP and power amplifier. The 12 channels is measured up individually in frequency and in time domain. But it doesnt end there.. the reverberation for each channel is measured. By processing you can control the reverberation by using all 12 channels to cancel out some sound in the later reverberation situation... i guess you understand what i try to say.. Maybe that is a good idea to develop in rephase as well? or maybe that option already exist?

Anyway... the result is quite amazing. Since you have total control of the reverberation field you can "record" reverberation pattern from other environments so the DSP mimics different concert halls as well.
 
Something like the CABS / Double bass array concept but in a car, and the resulting reverberation is variable. Cool :)

This dsp-steered reverberation, and dsp-steered directivit like B&O Beolab, is very cool concepts to me :)

Esl 63, I hope you can let me in to that car lab with dirac unison someday. It’s on my way home from work :) It’s in an XC90 I guess. Is it on the market yet?

Btw, does the electrification of the cars affect the sound?! Better SNR while driving,
I guess.

Active noise-reduction must be possible as well in car? Like your old ”active bass trap”?
 
Last edited:
By processing you can control the reverberation by using all 12 channels to cancel out some sound in the later reverberation situation... i guess you understand what i try to say.. Maybe that is a good idea to develop in rephase as well? or maybe that option already exist?

You mean addressing the reverberation field with an additional delayed filter?
I don't think one can (should) really try to cancel out reflection, but the global reverberation field and power response tonal balance could probably be somewhat corrected, to an extend...

rePhase is more focused toward correcting the source/speaker itself and not the room, as I feel these kind of things are better addressed using passive/acoustical room correction and good speaker directivity behavior, but this would sure be an interesting feature to add...
 
Last edited:
The reverberation control is mainly done below 300-400 Hz i guess.
So by measuring the exact size of the room and the exact placement of the multi speaker arrangement. And of course some discrete measurment of all channels, one at a time it is possible to calculate the correction "anti phase" signal for cancellation of the modes that is long lived...

Rajapruk, the XC90 has had this feature for almost 3 years...
The active bass trap works well but its quite elegant to DSP correct it!
 
The reverberation control is mainly done below 300-400 Hz i guess.
So by measuring the exact size of the room and the exact placement of the multi speaker arrangement. And of course some discrete measurment of all channels, one at a time it is possible to calculate the correction "anti phase" signal for cancellation of the modes that is long lived...
I would take the opposite approach: the lower range can be addressed to an extend using "normal" EQ applied to the direct response, but the diffuse field EQing could be of some use to address the upper part of the range where the one axis and power response can differ due to crossover "anomalies" (LR power response dips, etc.) or directivity behavior.
I don't think one can really cancel out reflections in any predictable/reliable way for more than one point in space.
 
I am not sure that is the best way to go making the phase flat when the amplitude response is not. I agree with wesayso that having the phase more closely follow generated minimum phase sounds better. What does generated minimum phase look like on that measurement?
.

This is not a responce I would use. The main point with the post is to show in an short form that REW, rephase, and an STM32F4 prosessor can be used to correct sound as close to linear or wanted responce, as one needs.

There are many postings like this showing how good the RePhase EQ is, but not sure if many shows
* How close REW room sim is to actual measurements in bass.
* Shows how Rephase can be used with downsampling and FIR filtering. So that 2048 taps is enough for 1 Hz frequency resolution. (Under 70Hz in this case). Rephase Fs of 750Hz.

Phase linearization in bass can be useful if a multisub solution is made. And the user has time to do the linearization + don't mind latency. Nothing for me, but it is possible. The rationale is to avoid cancelling because of phasedifference vs frequency with different make subs.
 
15ms seems pretty high as a base delay, even at 48kHz
Are you sure this was correctly measured, with an identical configuration as the DCX2496?
You should probably contact miniDSP support on their dedicated forum and see what they have to say about this.

After contacting MiniDSP with my claim they concurred, I quote: "In normal operation, the DA8 has some latency (at least 20 ms) because there are various DSP and FIR blocks which could incurred some delay even if you are not using it."
 

ra7

Member
Joined 2009
Paid Member
I cannot get REW output to be imported in Rephase 1.2. What gives? I've tried three different separators, checked line endings (CR LF, on windows machine), tried removing the header. I keep getting the same 'wrong format' error.

Here's how it looks:

* Measurement data measured by REW V5.19 Beta 4
* Source: C:\RoomCorrection\drc-3.2.1\sample\left_uncorr.wav
* Format: Imported Impulse Response, 44100 Hz sampling
* Dated: Jul 3, 2017 5:55:44 PM
* REW Settings:
* C-weighting compensation: Off
* Target level: 65.0 dB
* Note: Impulse Response imported from C:\RoomCorrection\drc-3.2.1\sample\left_uncorr.wav 44100 Hz sampling, 32-bit data
* Measurement: left_uncorr
* Smoothing: ERB
* Frequency Step: 0.33646142 Hz
* Start Frequency: 2.0187378 Hz
*
* Freq(Hz) SPL(dB) Phase(degrees)
2.019 6.942 -31.517
2.355 7.187 -63.269
2.692 7.437 -93.285
3.028 7.692 -121.903
3.365 7.954 -149.356
3.701 8.221 -175.815
4.038 8.495 158.596
4.374 8.775 133.781
4.710 9.062 109.667
5.047 9.355 86.193
 
Last edited:
Looks good enough to me ra7, how does REW say when you try import its own exported txt or frd product, also you could try what happens if you edit away those first 6 lines where phase goes radical in advance direction or remove their minus symbol, beside that you could share txt or frd file here or try update REW to latest v5.19 beta 7.