digital crossover on the cheap cheap

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I was wondering if anyone has attempted using software to improve the sound for their exiting systems?

I'm planning on designing and building my speakers (dsps and multiple way amps) later... But first fix up my existing system:

I listen to mostly my own recordings (flacs) have a two way JMR speakers and a Red Wine Audio Sig 16. I was wondering if I could brighten it up a bit, adjust for room characteristics etc. by:

1) Use a mike and digital room correction software to find the crossover frequency, ideal delay of the tweeter, and generally flatten the frequency response.

2) Simply remix my flacs using sox or ffmpeg filters to compensate for my own system and room weaknesses? Instead of active DSPs to provide crossover, just create a script to alter all my flacs at once once I get the settings right, and just keep my old equipment until my evil plans for a new system come to fruition.

Someone must have done that already right? Has anyone had success?

Cheers,
b
 
Tweet, a graphic EQ introduces noise, I guess...
which is the opposite of what Digital People want to achieve : clean, crystal sound...

About the delay in a tweeter: right ! The woofer is big and slow, the tweeter is
little and light
Then there's the passive crossover which introduces anticipation or delay to phase which is reflected in the time domain as in each cycle...:spin:
 
I was wondering if anyone has attempted using software to improve the sound for their exiting systems?

I'm planning on designing and building my speakers (dsps and multiple way amps) later... But first fix up my existing system:

I listen to mostly my own recordings (flacs) have a two way JMR speakers and a Red Wine Audio Sig 16. I was wondering if I could brighten it up a bit, adjust for room characteristics etc. by:

1) Use a mike and digital room correction software to find the crossover frequency, ideal delay of the tweeter, and generally flatten the frequency response.

2) Simply remix my flacs using sox or ffmpeg filters to compensate for my own system and room weaknesses? Instead of active DSPs to provide crossover, just create a script to alter all my flacs at once once I get the settings right, and just keep my old equipment until my evil plans for a new system come to fruition.

Someone must have done that already right? Has anyone had success?

Cheers,
b

Pico...., I was responding to that which I have highlighted in his post as an interim measure, it may well be all that he needs to get the sound he wants from his speakers. Sure it's a bit limited but 'until my evil plans for a new system come into fruition'.

C.M
 
JRiver is very powerful 64 bit DSP engine, it sounds great. You could use it to EQ your passive speakers. But to use it for a digital crossover you will need multichannel DACs and amps.

Most inexpensive digital active crossover boxes like Behringer, miniDSP, Rane, etc, have mediocre sound quality, but they are easy to adjust the filter parameters.

A DIY analog crossover, like Rod Elliot's Project 125, sounds excellent and is easy and cheap to build, but the filters and delay are not adjustable.

Excellent sounding digital crossovers are expensive, but they allow for easy adjustment, and room correction. DEQX is a stand alone box. Or PC software like AudioLense, Acourate or JRiver, outputting to a good quality multichannel DAC like exasound e28 or Buffalo 3.
 
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Thanks for your replies and ideas!

I should clarify that my source is a raspi pi b+ running runeaudio.com which probably wouldn't have the gumption to do any dsp work on the fly. That is why I was thinking of pre-batching it.

Again, this is until I design and build a new system. I'm a complete newbie so my only quals are lurking here. :D

Thank you again for the pointers.
b
 
I have a beringher 31 band graphic EQ. Its bottom of the range, almost. Includes two way Butterworth 2nd order fully adjustable crossover points, and sub output.

Compared with a DCX , miniDSP it is fairly cheap and allows for easy adjustment.

That being said, I bought all the parts for my analogue active filters AND amplifiers for less than either of the options.

A cheap multichannel soundcard and software is probably cheapest way, IF you have amplifiers already. Just be sure to check the low end roll.off on the soundcard... (mine is 40hz, even in stereo mode,meaning that it still expects you to use a sub...very annoying)
 
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Loafmeat. I was not able to get Google to translate your link. I'm interested because it's something I've wondered about, but not tried.

Were you able to get the different sections you needed mapped to the various channels of the 7.1 receiver OK? I can easily see how it should work, but have never seen it actually work.
 
@Pano
I was trying out the mapping of my HDMI 7.1 AVR.
In Jack connect one by one
Take a cheap loudspeaker and connect it to your AVR then you can hear the high or the mid or the low pass.
Take not the chassis of the loudspeaker you want to activate for this test.
That I have done for the Front, Surround and Surround Back in my case
Quite simple.
Don't take my example. Try it out

For my AVR the HDMI mapping is as follows:
X-Over-Out 7+8 =>Playback 7+8 = High = Frontspeaker
X-Over-Out 5+6 =>Playback 3+4 = Mid = Suround Speaker
X-Over-Out 3+4 =>Playback 1+2 = Low = Suround Back Speaker and additional to LFE for a SUB on Playback 6
X-Over-Out 1+2 = without mapping = below 20Hz
Maybe for other AVR it is different I don't know.

My actual additional setup is as follows.

Capture Stream => 12 band EQ => Xover => each stereo Channel a 5 band EQ => each stereo channel a Limiter => HDMI Out => ARV HDMI in => AVR DA Converter => AVR Amplifier => Loudspeaker Chassis

Its raealy good in this combination. You have a lot of possibilities to adjust (or destroy ) the sound.

Regards
 
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Thanks. It all makes sense, I just have not had a chance to try the channel mapping thru an HDMI output to a 7.1 receiver. JRiver would be my choice for the player, EQ, mapping and crossovers. But there are plenty of other ways to do it.

At least your Onkyo seems to have identical amps for all the channels, which is nice.
You put it in LCPM mode, right? Or rather it detects that and routes the signals where you want them? A 1-to-1 mapping.
 
@Pano

Yes, my AVR has identical power for each channel

All is digital from PC to AVR. DA Conversion is done by the AVR.
Each channel give a PCM Stream up to 192 kHz.
I adjust it to 44,1 kHz because my original music streams are 44,1 kHz (CD or internet streams or MP3)

With 44,1 kHz and the setup I descibed my PC has a DSP Load of 13 % and 0 Xruns
With higher sample rate the DSP Load will grow up.
With 192 kHz I have a DSP load of 50%.
It works but I cannot hear any difference to 44,1 kHz.
I think because the original streams are sampled in 44,1 kHz.
Internally the PC and the software is working with 32 bit.

The AVR automatically is recognizing the right sample rate an will switch correctly.

Regards
Guenter
 
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