While these statements were believed to be true in the mid to late 70s and was revived again when digital came into its prime (mid 80s early 90s) this does not represent current thinking for the professional. What you do end up with is spending allot more money in equipment and in maintaining the system, with inferior sound as an end result. It's the American dream come true, bigger is better. Working more with the acoustic environment and speaker placement keeping it simple will yield a much better sound stage. Assuming of coarse GOOD EQUIPMENT!- A lot of audiophiles bi-amp whenever they can. A linux HTPC with two high-end audiocards (or.. dare I dream.. digital out + professional DACs) would make a killer bi-amp system. Id predict almost EVERY audiophile would consider this setup..
Thank you for agreeing with my entire point!Not to rain on your parade! But in recent years real professionals have been moving away from bi-amped or try-amped systems. In a good environment the associated filters, crossovers and the amplifiers themselves offer to much phase shift to create an accurate sound stage. Even thought time alignment tools are available, the frequency spectrum is still adversely effected by these methods.
I don't really see the problem. VLC already supports multichannel output (through directx or whatever) doesn't it? All you need to do is assign two soundcard channels per audio channel.For VLC to implement such a scheme would require more involvement with the various sound cards available for all hardware and operating system platforms and there isn't a really good way to do this without a major standards change. Also in case you haven't noticed this is being handled through DirectX and your sound card and has little to do with the software programs you run.
Correct again. Agreed again, while underlining the ordinarily. I.e. not in this case. Unless you are trying to tell me that splitting the digital FFT (or Discrete Cosine, if you wish to be nitpicky) into a 'high only' and a 'low only' mode introduces additional phase issues.This is the issue: You CAN'T effectively remove all phase issues ordinarily associated with crossover filters. By nature of the filter you will introduce phase error!
I was recently thinking the same thing and I posted a feature request. Then I found this thread. It's great to find that I am not the only one in need of this feature.Well, for the audiophiles amongst you, the request can completely inferred from the subject.
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Thoughts? Comments?
No, this won't work. When a signal is divided into frequency bands by a crossover, the shape of the filter curves are very important. It is not just matter of removing some frequencies and leaving other ones. It is also not as simple as equalization. The exact crossover frequency and the shape of the filter curve should be very precisely controlled. Otherwise it simply won't work as a crossover.- IN THEORY, a lossy audio signal lends itself extremely well to bi-amping since audio is stored in frequency segments. All you have to do (.. I think) is just NOT join the lower and higher frequency areas.
I think so too.- Digital crossfilters can, in theory, and with enough CPU power (i.e. realtime), be vastly superior to analog ones, especially if you don't first DAconvert the signal
- A lot of audiophiles bi-amp whenever they can. A linux HTPC with two high-end audiocards (or.. dare I dream.. digital out + professional DACs) would make a killer bi-amp system. Id predict almost EVERY audiophile would consider this setup..
Foobar2000 can do this with the help of a crossover plugin. At least two such plugins are freely available and functional. There is also a commercial VST plugin implementing crossover function but it's more difficult to use it in a media player.- No other media player has this feature! (frankly I think Im the first to think of this, maybe I should patent the idea!)
I think the best place is output filter for already decoded audio signal.Well the generic idea is that a purist would want to split the signal at the best place.
Best place is the earliest place.
Earliest place is inside VLC, preferrably even inside the audio decoding algorythms.
We are talking about the needs of ordinary audio enthusiasts (audiophiles), so we don't need to worry about where professionals are moving. Also professionals have different budget from us home users. Multiamplification is one of the relatively simple and affordable ways to get better sound quality, especially coupled with a bit of DIY. Recently it is very easy to get a multi-channel sound card and AV amplifier. Then all you need is to slightly modify the speakers. This is already possible with great resutls for music using foobar2000 and freeware plugins.Not to rain on your parade! But in recent years real professionals have been moving away from bi-amped or try-amped systems.
Yes, passive crossovers create a phase shifts and have other problems. That's why we want to use a software crossover in digital domain - it can counter all these effects.In a good environment the associated filters, crossovers and the amplifiers themselves offer to much phase shift to create an accurate sound stage. Even thought time alignment tools are available, the frequency spectrum is still adversely effected by these methods.
In 70s and 80s there was no way to build a software crossover because there were no affordable computers able to do this, and because computer audio interfaces were of very poor quality anyway. In 80s and 90s the multi-channel sound cards and AV amplifiers were still not as good and cheap as they are today. So it is today when multiamplification with computer as a source begins to make sense for large number of users.While these statements were believed to be true in the mid to late 70s and was revived again when digital came into its prime (mid 80s early 90s) this does not represent current thinking for the professional.
A lot of people came to difrerent conclusion by their experience, including myself.What you do end up with is spending allot more money in equipment and in maintaining the system, with inferior sound as an end result. It's the American dream come true, bigger is better. Working more with the acoustic environment and speaker placement keeping it simple will yield a much better sound stage. Assuming of coarse GOOD EQUIPMENT!
I don't think so, frankly speaking. VLC is already able to produce multi-channel sound. All that is needed is to implement a filter that will take stereo signal and produce a 4-channel or 6-channel output of the same stereo separated by frequency.For VLC to implement such a scheme would require more involvement with the various sound cards available for all hardware and operating system platforms and there isn't a really good way to do this without a major standards change.
There are already excellent free implementations of software crossover. (Links are in my feature request linked in my previous post). It works great for music in foobar2000. All we want now is to also watch movies using multi-amplified system.Also in case you haven't noticed this is being handled through DirectX and your sound card and has little to do with the software programs you run.
This problem is only relevant to analog crossovers. In digital domain it is possible to have crossover without any phase problems. Though a decent digital crossover should have a delay function configurable individually for each channel, so that precise optimization can be possible.This is the issue: You CAN'T effectively remove all phase issues ordinarily associated with crossover filters. By nature of the filter you will introduce phase error!
Let's forget this idea. It is a wrong track, because:- MP3's and many lossy encoders in fact split music up into signal bands as a means of frequency analysis and component removal.
- The MP3 decoder should therefore reconstitute its final signal by indeed recombining/mixing these signal bands anyway.
- If we prevent the decoder to do this, but instead keep highs and lows separated, we're quite on the right track already. Any introduced phase problems can not be more severe than they would be by just combining everything into a single channel.
Yay! 24 bit (or may be 24-bit 96 KHz) is the way as long as your sound interface hardware supports it and as long as the CPU is fast enough.- Obviously, switching up to 24bit, then performing the crossover would be neat-o.
Here is one really good and detailed article about multiamplification. And here is one practical guide to set up a multi-amplified system using a hardware active crossover. That Behringer crossover is good but it adds extra A/D and D/A conversion to the signal path, so it is better to implement crossover on a PC before the first D/A conversion. For music you can do this easily using foobar2000 player and foo_dsp_xover plugin. Audigy should work great, but you need a multi-channel amp and modified speakers. You need to disconnect their internal passive crossover network and route the terminals directly to the drivers (don't forget to add a capacitor to protect the tweeter). BTW for the best results you may want to use kernel streaming or ASIO output in foobar2000.This is quite an interesting thread. Being an audiophile myself, I would like to know where I can learn more about bi-amping, a link perhaps? I’ve been using a SB Audigy 2 Z/S soundcard with kx audio drvers (in place of Creative drivers) and the sound Is great!
If this "bi-amping" can make the sound even better, then I’m all for it.
Yeah, I better read the articles first before I read the rest of this post. Thanks.Here is one really good and detailed article about multiamplification. And here is one practical guide to set up a multi-amplified system using a hardware active crossover. That Behringer crossover is good but it adds extra A/D and D/A conversion to the signal path, so it is better to implement crossover on a PC before the first D/A conversion. For music you can do this easily using foobar2000 player and foo_dsp_xover plugin. Audigy should work great, but you need a multi-channel amp and modified speakers. You need to disconnect their internal passive crossover network and route the terminals directly to the drivers (don't forget to add a capacitor to protect the tweeter). BTW for the best results you may want to use kernel streaming or ASIO output in foobar2000.
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