Digital Room Correction Project

[boconnor]

I have commenced a project to do digital room correction (DRC) of my sound system. I thought it might be useful to report on the techniques used and the progress of the project.

I begin by describing my existing use of DRC techniques and then describe how I intend to enhance the system.

Existing System

The existing system has been in place now for about 6 weeks. It has convolution filters running under the Squeezebox system. The equipment used to create and play the filters is:

• Test signal generation: sine wave log sweep, played via the SqueezeBox receiver running the Inguz Audio test signal plug-in
• Mic and mixer: Behringer omni-directional microphone ECM8000 (BEHRINGER: ECM8000) and Behringer UB1204 mixer with phantom power.
• Recording software and hardware: Audacity under Windows, running on a laptop. Mixer output was fed to a Creative Audigy 2ZS PCMCIA sound card in the laptop.
• Filter generation: DRC program by Denis Sbragion, running under Windows. (DRC: Digital Room Correction, freeware)
• Filter convolver: Inguz Audio plug-in for Logitech Squeezebox (http://inguzaudio.com/RoomCorrection/, freeware)

The process used was:

• a 20 to 20k Hz sine wave log sweep test signal was played through the existing sound system. The signal lasted for about 30 seconds. Separately, each speaker played the signal and the room response was excited. The SPL of the signal was manually adjusted to get a good recording level based on a visual check of the Audacity recording screen during the recording.
• the room response of the system was measured with the microphone positioned in the best listening position. The mic was on a mic stand, with the mic capsule oriented vertically at listener ear height. The listening setup is a standard triangle with a stereo pair of speakers, and with the prime listening position at the apex of the triangle. Speakers are at ear height, about 1.58 metres apart, and measured from their midpoint some 2.1 metres from the listening position. There is no acoustic room treatment in the room. Approximate room dimensions are 4 metres (w), 4 metres (d), 4.5 metres (h). The room is furnished with rugs, window drapes and fabric couches.
• the room measurements were processed through the DRC program. This has a range of target curves that can be selected. I opted for the ‘erb’ curve which uses a psychoacoustic target that is fundamentally a flat target curve but with a slight roll-off of the bass response. The program created a correction filter for each speaker.
• the left and right correction filters were converted into a stereo .wav file. The stereo filter was loaded into the Squeezebox system using the Inguz Audio plug-in. In my Squeezebox system iTunes is the media source for music files. All files are in Apple lossless format. When playing music the data stream is extracted in real time from iTunes then convolved at the PC server with the correction filter, then transmitted by WiFi to the Squeezebox receiver in the listening space. From there it goes into the analogue input chain of amplifiers and speakers.

The sound of the existing system is excellent. In particular the bass is very tight and controlled. It is much better than the 1/3 octave equaliser solution I had been using before the shift to digital room correction filters.

Proposed System

The goal of the proposed system is to recreate, as far as possible, the sound environment of a recording space that is physically larger than the listening space. That is, to increase the chance that the listener suspends their disbelief that the room is larger than it actually is.

The three main changes to the existing setup to achieve this are:

• running the filters on a dedicated PC rather than through the Squeezebox system.
• moving from stereo to multi-speaker reproduction. Specifically from a 2.1 system to a 6.1 system.
• running separate filters on each of the four surround speakers using impulse responses recorded from large acoustic spaces.

There are three advantages to running the filters by PC compared with by Squeezebox. First, with an appropriate sound card and other hardware more than 2 channels of audio can be processed. This allows for experimentation with surround sound formats. Second, at present only the audio output from the Squeezebox is processed against the filters. I have a DVD/CD player and home theatre system and those outputs can’t at present be put through the filters. Third, although the Inguz Audio plug-ins are really excellent software (and free!) they require the Squeezebox software to remain at version 7.2 or below. I have had to re-install old versions of the Squeezebox server software to remain compatible with the Inguz Audio software.

Equipment list for proposed setup:

• Test signal generation: sine wave log sweep, played from the filter software running on the laptop.
• Mic and mixer: Behringer omni-directional microphone ECM8000 and Behringer UB1204 mixer with phantom power.
• Recording software and hardware: Not yet decided. I am leaning towards the Audiolense software by Juice HiFi (Juice HiFi) running under Windows on a laptop – but there are other options such as Acourate by Ulrich Bruggemann (http://www.acourate.com/). The Audiolense product seems better at driving the Audigy card during recording. Both products are payware. The sound card will remain the Creative Audigy 2ZS PCMCIA sound card.
• Filter generation: Probably the same software as the recording software, Audiolense running under Windows.
• Filter convolver: BruteFIR running under Linux (BruteFIR, freeware).
• PC for running the filters: a fan-less, small PC with an appropriate sound card, running Linux.
• Analogue input and output: Behringer ADA8000 DAC/ADC unit (BEHRINGER: ADA8000). This provides 8 channels of simultaneous input and 8 channels of simultaneous output via two ADAT connectors.

What’s been purchased so far:

• Filter convolver: BruteFIR running under Linux (well not purchased, more like downloaded and compiled under Linux).
• PC for running the filters: I purchased a small PC from eBay for $200. It has one PCI slot and a PCMCIA slot. It uses a Via 800Mhz Mini-ITX Motherboard with fan-less heat sink and has 512 MB of memory, more than enough to run the filters. It does not have a hard disk. Initially I intend to run the BruteFIR convolver using a USB stick loaded with a disk-less version of Linux “SPBlinux” (SPB-Linux 2.1 beta) and the BruteFIR software.
• Sound card: RME Digi9636 Hamerfall PCI sound card (RME Intelligent Audio Solutions - Hammerfall Lite). This is a superseded card, which I bought on eBay for about $200. It has a S/PDIF digital input and output channel, and four ADAT connectors that can be configured as 16 input channels and 16 output channels at 44.1k or 48k sample rates.
• Analogue input and output: Behringer Pro-8 ADA8000 DAC/ADC unit.

Next step: get the mini PC and BruteFIR software working for just 2 channels to confirm that the PC/sound card combination and the Behringer ADA8000 link is working.

I’ll provide updates as things progress.

I am happy to answer any questions or explain the project in more depth for those that are interested.

[phofman]

Hi,

Nice plan, hats off. How are you going to transfer audio between the playback NTB and the brutefir linux filter?

[boconnor]

Originally Posted by phofman

Nice plan, hats off.

Thanks.

How are you going to transfer audio between the playback NTB and the brutefir linux filter?

I'm going to answer this based on the NTB you mention being the same as the Squeezebox receiver I have in the lounge room. If I have misunderstood your question please let me know.

The S/PDIF coaxial output from the Squeezebox receiver will be input into the coax input of the RME sound card in the PC. The correction filters running in the PC (under BruteFIR) will convolve the input and send the digital output to the RME’s ADAT channels. Then via toslink optical cable the output will go to the Behringer ADA8000 where the data stream is converted to line level analogue. Then its just a normal analogue signal which gets input in the amps and speakers.

[lazycatken]

Some questions:
1. Why did you choose erb configuration file? Why not stronger correction?
2. Does the convolution process 32 bits floating point signal? And what's the data format that fed to DAC? 16bits?
3. How do you set volume (or magnitude of amplification after convolution)? I mean, in order to achieve higher resolution, supposedly, signal must be normalized...
4. Can you talk more about your amplifier and speaker?
5. Did you use pa-xx.x.txt as your target frequency response curve? Due to room dimension, max 4.5m, it's impossible to generate bass under 38Hz...
6. Any microphone frequency response correction is applied?
7. Looks like iTunes plays music, then Inguz Audio plug-in does convolution. Sorry, I have no any idea about how Squeeze works. Does it work just like a sound card, thus, iTunes output music to the sound card?

[boconnor]

Originally Posted by lazycatken

Some questions: 1. Why did you choose erb configuration file? Why not stronger correction?

I found the bass response for the room was smooth and tight using the erb target. The stronger targets produced a slight boom in the bass. I like to compare the room response to what I remember from live concert performances, and they have a seamlessness from mid to bass that really appeals to me. That’s what I’m trying to reproduce in the listening room.

2. Does the convolution process 32 bits floating point signal? And what's the data format that fed to DAC? 16bits?

The drc program produces two files, one each for each channel. I understand they are 32 bit floating point files. They are spliced together by an Inguz Audio program to produce a stereo .wav file that is loaded with the SqueezeBox server software to convolve the data stream before it is transmitted to the receiver.

3. How do you set volume (or magnitude of amplification after convolution)? I mean, in order to achieve higher resolution, supposedly, signal must be normalized...

It’s a trial and error process. The output after the filter is applied is often over the normal digital 0db limit so I set a variable in a Inguz configuration file that attenuates the original signal to make room for the affect of the filter. At present the attenuation is -19db. It’s a compromise, since its been set to prevent clipping of the loudest music I play, but for some recordings it means the amp gain has to be turned up quite a bit to compensate for the attenuation.

4. Can you talk more about your amplifier and speaker?

I have a 4.1 setup. The front left and right speakers are modified Mirage Omnisat V2. I like their innovative speaker geometry. They use a specially constructed parabolic “wave guide” above the midrange and tweeter drivers that disperses the sound wave through a large angle into the listening space. This is designed to overcome the narrowing of the power response as frequency increases from the omni-directional low frequencies to the directional high frequencies.

Close up of Mirage Omnisat.jpg

Also,

• The speakers are a small acoustic radiator relative to the size of the room. This means they are closer to the ideal of a point source radiator.
• The enclosures are rigid and non rectangular so are less likely to emit out of phase secondary radiation into the listening area compared with rectangular wooden box designs.

They also have some practical advantages, being sufficiently small and light to be placed on adjustable stands, and easily moved around the room. Also they come with built in mounting screws that can be used with inexpensive home theatre stands.

I have mounted them so that the midrange driver is at the same height as the ears of the listener when sitting on the lounge at the optimal listening position.

The Omnisats use a very simple (3 component!) passive crossover. I removed the passive crossover board and replaced the banana plug speaker cable connectors with RCA connectors. That gave me one RCA cable per driver so I can run a simple stereo RCA cable to each enclosure from the amplifier.

I left the simple foam padding inside each enclosure that is part of a normal acoustic suspension design.

Due to the size and type of midrange driver, and the small enclosure size, the Omni’s are not going to have much bass output below 90Hz. So I decided to add the Siegfried Linkwitz designed “Pluto” subwoofers as front bass units (see here Linkwitz Lab - Loudspeaker Design for his website, this is the same Linkwitz of Linkwitz-Riley crossover design fame).

They seemed an ideal match for the satellites given they were designed to marry with omni-directional satellites with a low frequency roll-off.

Front Satellite Crossovers

I use a Behringer DCX2496 digital crossover for the front speakers. It provides 6 output channels which is perfect since I need 3 channels per side.

The Behringer has a number of different types of crossover filters (Butterworth, Bessel and Linkwitz-Riley), with a variety of slopes. I chose Linkwitz-Riley crossovers with a slope of 48db per octave.

The front bass units are crossed over at 100Hz. The crossover from midrange to tweeter is 3kHz. The passive crossover had used 2.7kHz. I adjusted the crossover value whilst listening to the speakers and found the 3kHz point sounded smoother than the 2.7kHz point.

Front Satellite Amplification

The front bass units are powered by a second hand Kenwood power amp. The satellites are powered by a second hand Yamaha RX-V520 AV receiver where I use the four 70W power amps in the receiver in external decoder mode.

Other speakers

OK, so that’s for the front speakers. For the rear speakers I use a pair of unmodified Spherex speakers (no longer manufactured) which are mounted on adjustable stands. These are the same design as the Mirage front speakers (they licensed the technology), but with smaller midrange and tweeter drivers.

For the subwoofer system I use a home built subwoofer done to a Linkwitz design, with a 12” Peerless driver in a 50L sealed box. It is driven by a Behringer A500 power amp in bridged mode. Because it is too close to the listening position compared with the satellites I digitally delay the signal being fed to the sub so that it is acoustically the same distance from the sub to the listener as from the front satellites to the listener.

I use a Behringer CX310 crossover to cross over the sub. The setting is 60Hz, with 24db per octave Linkwitz-Riley crossovers.

The rear surround speakers are powered with a Yamaha DSP-A1, which also does all digital stream decoding.

I don’t have a front centre speaker. I use an $80 Behringer mixer to take the three pre-amp signals (front left, centre, front right) from the Yamaha and mix them to the front stereo channels. This allows me to control the level of the dialogue and adjust the centre channel tone using the equaliser on the mixer.

5. Did you use pa-xx.x.txt as your target frequency response curve? Due to room dimension, max 4.5m, it's impossible to generate bass under 38Hz...

No, I used the erb-xx.x.txt curve. Not sure about the no bass under 38Hz. I think room modes and bass calculations can go a bit haywire in normal rooms (compared to the ideal rooms used in the equations). Also, the lounge room is open to another room of the same size so the combined volume is quite large. For instance the satellites are positioned in an archway joining the two rooms so they effectively don’t have any rear wall directly behind them.

6. Any microphone frequency response correction is applied?

No. I am aware that Denis Sbragion’s correction file for the Behringer ECM8000 was just based on testing one mic. I think if you are going to use an adjustment file it should be based on a reasonable number of samples.

7. Looks like iTunes plays music, then Inguz Audio plug-in does convolution. Sorry, I have no any idea about how Squeeze works. Does it work just like a sound card, thus, iTunes output music to the sound card?

No. The SqueezeBox server software runs on a PC where the music is also located. After taking data from iTunes and convolving with the filter, the SqueezeBox software sends the data stream to a wireless transmitter. From there it gets picked up by a SqueezeBox wireless receiver, which has the digital to analogue converters. Then its on to the amps and speakers.

[boconnor]

Originally Posted by lazycatken

Some questions:5. Did you use pa-xx.x.txt as your target frequency response curve? Due to room dimension, max 4.5m, it's impossible to generate bass under 38Hz...

Sorry, I made a mistake when I answered the first part of this question, concerning the .txt file used by DRC.

You were right lazycatken - I have looked at the DRC configuration files and the 'erb' control file actually uses the pa-xx.xx.txt file for the target frequency response curve calculations.

[lazycatken]

Originally Posted by boconnor

I found the bass response for the room was smooth and tight using the erb target. The stronger targets produced a slight boom in the bass. I like to compare the room response to what I remember from live concert performances, and they have a seamlessness from mid to bass that really appeals to me. That’s what I’m trying to reproduce in the listening room.

In my system, extrme or even stronger got better bass performance(less stationary waves)
So far, I tested four systems (different rooms & equipments), stronger correction produced better bass resolution in all the four systems.
The only issue prevents taking stonger correction is pre-echo...

It’s a trial and error process. The output after the filter is applied is often over the normal digital 0db limit so I set a variable in a Inguz configuration file that attenuates the original signal to make room for the affect of the filter. At present the attenuation is -19db. It’s a compromise, since its been set to prevent clipping of the loudest music I play, but for some recordings it means the amp gain has to be turned up quite a bit to compensate for the attenuation.

Volume setting issue makes me be interested in the internal signal format of convolver...
No doubt, the convoluton process is under 32 bits floating point format. And then? 16bits?
I'm wondering down converting from float32 to interger16 must result in some loss...
Thus, I use 24 bits, sounds better. Although, not sure its due to the issue that I concerned...

I have a 4.1 setup. The front left and right speakers are modified Mirage Omnisat V2. I like their innovative speaker geometry. They use a specially constructed parabolic “wave guide” above the midrange and tweeter drivers that disperses the sound wave through a large angle into the listening space. This is designed to overcome the narrowing of the power response as frequency increases from the omni-directional low frequencies to the directional high frequencies.

Attachment 144559

Yes, the design is very impressive...
Thus, I mentioned it in my blog couple months ago 沈浸在音樂之ä¸*...: 有æ„�æ€�的單體è¨*計--Mirage Uni-theater
But, it may not be an appropriate choice for DRC...
You know, the major issue that DRC tried to solve is stationary wave in common listening room. DRC does not do much on high frequency range. High frequency must be taken care by traditional passive room treatment. That's why Denis Sbragion holds a heavily damped listening room.
Thus, I guess, the best speaker choice for DRC is one that pretty directional. Just like studio near field monitor speakers...

I have mounted them so that the midrange driver is at the same height as the ears of the listener when sitting on the lounge at the optimal listening position.

Looks like, your speakers do not be placed on a position to get better correction by DRC?
DRC documents suggets put speakers close to wall. I did it, got better result.

The Omnisats use a very simple (3 component!) passive crossover. I removed the passive crossover board and replaced the banana plug speaker cable connectors with RCA connectors. That gave me one RCA cable per driver so I can run a simple stereo RCA cable to each enclosure from the amplifier.

That's interesting... Why did you choose RCA instead of Y spade or banana?

The front bass units are crossed over at 100Hz. The crossover from midrange to tweeter is 3kHz. The passive crossover had used 2.7kHz. I adjusted the crossover value whilst listening to the speakers and found the 3kHz point sounded smoother than the 2.7kHz point.

Did you do measurement? Just curious about what it can be seen on measurement result...

For the subwoofer system I use a home built subwoofer done to a Linkwitz design, with a 12” Peerless driver in a 50L sealed box.
...
No, I used the erb-xx.x.txt curve. Not sure about the no bass under 38Hz. I think room modes and bass calculations can go a bit haywire in normal rooms (compared to the ideal rooms used in the equations). Also, the lounge room is open to another room of the same size so the combined volume is quite large. For instance the satellites are positioned in an archway joining the two rooms so they effectively don’t have any rear wall directly behind them.

My speakers are Spica TC50, with 6.5" woofer. No subwoofer is companioned, though, I have one.
TC50 can't produce bass under 50Hz. I realized it's useless to compel TC50 to generate flat frequency response down to 20Hz. Thus, I modified the target frequency response curve.

No. I am aware that Denis Sbragion’s correction file for the Behringer ECM8000 was just based on testing one mic. I think if you are going to use an adjustment file it should be based on a reasonable number of samples.

I'm thinking about purchasing a mic with frequency response data...

No. The SqueezeBox server software runs on a PC where the music is also located. After taking data from iTunes and convolving with the filter, the SqueezeBox software sends the data stream to a wireless transmitter. From there it gets picked up by a SqueezeBox wireless receiver, which has the digital to analogue converters. Then its on to the amps and speakers.

Sorry, I didn't clearly present my question.
I'm thinking about playing music by iTune, then send music via LAN to a device, just like Squeezebox.
Right now, it's Apple Airport Express. Unfortunately, Airport Express can only handle 16bits music.
Thus, I'm thinking building my own. You know, the first issue will be "How to get music from iTune?"
A simple way is to install a virtual sound card. Not a good idea, right?
Another way is tried to hack the encryption key of RAOP. I'm not an expert of that...
Or license RAOP from Apple. I have to buy lottery first...
Or?

[boconnor]

Originally Posted by lazycatken

Volume setting issue makes me be interested in the internal signal format of convolver...No doubt, the convoluton process is under 32 bits floating point format. And then? 16bits? I'm wondering down converting from float32 to interger16 must result in some loss... Thus, I use 24 bits, sounds better. Although, not sure its due to the issue that I concerned...

The following is my understanding of how the process works. However, I am not an expert in digital signal processing so if anyone reading this wants to correct me please feel free.

The impulse response of the room is measured using a 16-bit resolution log sine sweep test tone. The actual measurement of the room response results in a 32-bit floating point format file. There are then a series of mathematical calculations done on this file which results in a 32 bit floating point output file, which is the correction filter. The correction filter is then put into a convolver. A convolver like BruteFIR also uses 32-bit internal floating point calculations. Within BruteFIR the word length of the final output file that actually gets converted into analogue will depend on the settings of the sound card.

The question then becomes is it better to use 24 bits or 16 bits in the final digital to analogue conversion? Remember that the original input data stream that goes into the convolver is likely to come from a CD source. That means that the original data stream is in 16-bit format. Even if it is up-converted to 24 bits, no additional information is created in that up sampling process. Provided the convolution algorithms are correctly designed, convolving a 16-bit input with a 32-bit floating point file should not result in any loss of information. Certainly not information that would be audible.

I think that the question can only ultimately be answered through psychoacoustic testing. A properly designed and executed test will indicate if there are audible improvements in going from 16 bits to 24 bits in the final DAC process. My view is that 16-bit resolution is good enough in normal listening environments, so I don't see a need to necessarily use 24-bit conversions.

But, it may not be an appropriate choice for DRC...You know, the major issue that DRC tried to solve is stationary wave in common listening room. DRC does not do much on high frequency range. High frequency must be taken care by traditional passive room treatment. That's why Denis Sbragion holds a heavily damped listening room. Thus, I guess, the best speaker choice for DRC is one that pretty directional. Just like studio near field monitor speakers... Looks like, your speakers do not be placed on a position to get better correction by DRC? DRC documents suggets put speakers close to wall. I did it, got better result.

I am very persuaded by the argument of Seigfreid Linkwitz concerning the best speaker geometry and placement of speakers in a room to generate a sound field that convinces the ear/brain combination that you are in the recording space rather than the listening room.

As I understand his ideas you need either dipole or monopole (omnidirectional) speakers which have a uniform polar response across mid to high frequencies. One of the reasons I quite liked the Omnisat design was that, with the use of the waveguide, they really seem to generate a polar response that is uniform across the mid and high frequencies. That, combined with a base unit which would be omnidirectional anyway, means that there is a very good chance that there is a nice consistency in the polar response across all useful frequencies.

The other issue is placement of the speakers in the room. I am also convinced by his arguments that you need at least a one metre gap between the speaker and any side walls. This ensures that there is not a wave of reflections that are so close in time compared with the direct wave from the speakers that the brain cannot distinguish the two waves and the illusion of phantom sources collapses.

I have tried different placement of the speakers in the room and it certainly appears to me that speakers at least one foot (3 metres) from a side wall sound better, in the sense that the phantom images and the realism of the reproduction is better.

I also think that room correction is easier if the reflected waves are both lower in intensity and there is a longer time gap between the direct wave and the reflected one - on the basis that the algorithms will have less work to do in distinguishing the corrections needed for the direct wave compared with correcting the reflected waves. At higher frequencies the DRC program uses an increasingly short time window so that would imply to me that the corrections are attempting to deal more with the direct wave from the loudspeaker rather than the room reflections.

That's interesting... Why did you choose RCA instead of Y spade or banana?

Convenience in building really. The existing speakers had a pair of speaker connectors. When I removed the passive crossover board and the speaker connectors, I found that the holes were perfect for the insertion of a pair of RCA terminals. That way I can run just a stereo RCA cable to the speakers and drive both speakers in the cabinet with one cable.

Did you do measurement? Just curious about what it can be seen on measurement result...

When I originally equalised the satellites with a 1/3 octave equaliser I hired an anechoic chamber and used a 1/6 octave measurement process to look at the response of the speakers as I varied the crossover frequency.

To be truthful there wasn't much difference in the measured response as I varied the crossover frequency from 2.5 kHz through to 3.2 kHz. So my reasoning at that point was that it would be better to have a high crossover frequency for the tweeter given that it is so small and the less information out of bounds that it gets the less distortion there would be. On that basis I chose to 3 kHz crossover frequency.

I'm thinking about purchasing a mic with frequency response data...

Sounds like a good idea.

I'm thinking about playing music by iTune, then send music via LAN to a device, just like Squeezebox. Right now, it's Apple Airport Express. Unfortunately, Airport Express can only handle 16bits music. Thus, I'm thinking building my own. You know, the first issue will be "How to get music from iTune?" A simple way is to install a virtual sound card. Not a good idea, right? Another way is tried to hack the encryption key of RAOP. I'm not an expert of that...Or license RAOP from Apple. I have to buy lottery first...Or?

As I indicated earlier I am simply not convinced that trying to get 24 bit source material (either originally recorded that way or through up-sampling) is worthwhile. The vast majority of existing commercially available material is in 16-bit format. If you run with that then there are a number of solutions available including just using Apple airport express as it's currently configured.

By the way, I have enjoyed your comments and questions – if has forced me to think through some things that had not been clear in my own mind before responding to your posts.

[dviswa]

Hi Guys,

Nice to see this going on. I have been on the side lines, considering dipping my toes into DRC. I have read elsewhere on the net about the latency issue. Using it on Music is a no brainer, but for TV and DVD, the digital processing of the sound causes a delay loosing sync with the images on TV. This is the main reason, why I am still on the sidelines. Please let me know how you are addressing this. Or you are using it only for music?

Thanks,
Dinesh

[boconnor]

Originally Posted by dviswa

Hi Guys, Nice to see this going on. I have been on the side lines, considering dipping my toes into DRC. I have read elsewhere on the net about the latency issue. Using it on Music is a no brainer, but for TV and DVD, the digital processing of the sound causes a delay loosing sync with the images on TV. This is the main reason, why I am still on the sidelines. Please let me know how you are addressing this. Or you are using it only for music? Thanks, Dinesh

I'm happy to address the latency issue from the perspectives of both the existing and proposed systems.

Existing system

When using my 4.1 home theatre system the front left and the front right speakers are fed via a Behringer DCX2496 active crossover unit. This takes two channels and converts them to six channels. Each channel is processed using an eighth order (48 db per octave) Linkwitz-Riley crossover calculation. All six channels are processed through one DSP unit. I do not know the latency figure for this unit but there would obviously be some latency in doing those calculations.

But, I have not noticed any sync issues when watching images on the screen and listening to the dialogue from the front stereo speakers.

Proposed system

In the proposed system all audio channels will go through convolution filters plus the front left and right will then go through the Behringer processor. As I understand it there are three components to the magnitude of the latency delay (independent of what is happening in the Behringer): the latency within the sound card, the speed of the PC in doing the convolution calculations, and the programming efficiency of the convolution algorithms. The sound card I will be using, a RME card, has next to zero latency. The PC processor is quite fast. And the BruteFIR algorithms should be quite fast, since it has been designed to be very fast at convolving data files.

On that basis there should not be a problem with latency. But I guess I’ll find out when everything comes together.