Bi-amping support for VLC

[puntloos]

Well, for the audiophiles amongst you, the request can completely inferred from the subject.

The short version is:

- I want VLC to output the low-frequency components of the output sound to be sent to 'line out channel 1' and the mid- and high-frequency components of the output sound to 'line out channel 2'. (and do this for all sound channels in the sound stream)

The longer version:
Bi-amping is digitally splitting (cross-filtering) the sound signal into a high-frequency part (with all lows removed) and a low-frequency part (with all highs removed), as early as possible in the audio chain, and then eliminating(removing!) usually inferior analog crossover filters in your speakers, finally directly wiring the speakers onto your amplifiers. (yes, for playing a stereo signal, you will need 2 stereo amplifiers).

The benefits are a little technical, but roughly speaking they are:

- higher overall wattage without having to purchase larger amplifiers
- Better frequency response
- More detailed soundstage

The longest version:
http://sound.westhost.com/bi-amp.htm

Reasons I ask:

- 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.
- 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..
- No other media player has this feature! (frankly I think Im the first to think of this, maybe I should patent the idea!)


Thoughts? Comments?

[Lurker]

I have bi-amping active monitors in all 5 channels in my 5.1 setup. VLC is just one of the sound sources, so I'd avoid a special wiring especially for VLC audio output. I just wonder why do you need it, or is VLC is the only audio source for you?

[puntloos]

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.

Anyway for me personally Ive recently bought a 1-way speaker system instead of the 2-way thing I was considering so the bi-amping thing is a moot point for me

[DJ]

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.

- 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..

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!

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.

[puntloos]

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.

Thank you for agreeing with my entire point!

To recap:

- If VLC 'bi-amps' by separating highs from lows digitally (perhaps even at the fourier decoding functions that happen in most lossy formats) you would effectively remove all phase issues ordinarily associated with crossover filters.
- This of course is assuming a setup where the 'high-freq components' of a certain sound channel get directed to exactly the same amplifier setup (perhaps even the same amplifier) as the 'low-freq components'. If the amplifiers for both signalpaths are identical there would be no phase shift, which is exactly the point of the bi-amping in the first place.

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.

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.

(and of course from that point on, be sensible about what signals to feed at what volume settings to what speaker parts.. sending a bass signal to a tweeter is bad, m'kay)

Anyway for me personally the issue is moot - I have 1-way speakers. (Quad 989 electrostats) but I still think this would be a cool addon.

[DJ]

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!

[puntloos]

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!

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.

Short version: they don't. Or at least shouldn't, if they're properly built. And as you have picked up I hope, the only 'crossover filtering' we are doing is exactly in that spot, inside VLC.

In my proposed audio chain, the equipment that you have control over (we can't start to compensate for perceived possible errors in the recording gear, can we) introduces NO additional phase change between highs and lows, compared to a single-amped system.

Let me recap:

- Assume a mono source.
- Assume we use a stereo amplifier
- Assume its signal paths are identical
- Assume a two-way speaker with a decent crossover filter which splits the signal, and doesn't attenuate the high component. Normally it does this since tweeters are more sensitive, but that bit is irrelevant here.
- Assume a 44khz 16bit source

Then:

- We need to split the signal into highs and lows as early as possible
- 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.
- Even if this can't be prevented, a well-tuned digital crossover filter is MUCH better at preventing phase differences than analog ones (inside, say, a speaker).
- Obviously, switching up to 24bit, then performing the crossover would be neat-o.
- When we have 2 decoded ('WAV') digital signals, we send those to the left and right channel.
- The separated signal path through the identical amplifier should not cause any additional phase changes (again, compared to sending 1 complete signal)
- No crossover-ing in the speaker

[Kirr]

Well, for the audiophiles amongst you, the request can completely inferred from the subject.

...

Thoughts? Comments?

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.

I mostly agree with puntloos here. A couple of comments:

- 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.

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.

Separation into frequency bands used in lossy audio compression has nothing to do with crossover function. They are two totally unrelated matters. Depending on particular details of lossy compression scheme is also not a general solution.

- 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..

I think so too.

- No other media player has this feature! (frankly I think Im the first to think of this, maybe I should patent the idea!)

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.

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.

I think the best place is output filter for already decoded audio signal.

[Kirr]

Not to rain on your parade! But in recent years real professionals have been moving away from bi-amped or try-amped systems.

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.

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.

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.

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.

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.

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 people came to difrerent conclusion by their experience, including myself.

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.

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.

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.

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.

[Kirr]

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!

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.

[Kirr]

- 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.

Let's forget this idea. It is a wrong track, because:

1. Frequency filters used in lossy compression don't have anything in common with crossover high-pass and los-pass filters. Crossover filters are tuned very precisely for particular loudspeaker drivers. They have precisely defined curve shape and deviating from that shape will ruin the sound.

2. Digital crossover should have configurable filter type, order and crossover point.

3. Many people who use multiamplification don't use lossy compression at all. Even if we use lossy compression there are 1000 methods and algorithms. We simply can't expect that particular algorithm is used. It is much more general and sane to implement digital crossover totally independently from a decoding stage.

- Obviously, switching up to 24bit, then performing the crossover would be neat-o.

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.

[CloudStalker]

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.

[Kirr]

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.

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.

[CloudStalker]

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.

Yeah, I better read the articles first before I read the rest of this post. Thanks.

[axel1973]

i guess it would be a lot of coding and work for a thing that 60% of the poll say NO THANKS!

that speaks for itself.

[Jean-Baptiste Kempf]

Anyway, that may appear this summer.

[agnos]

It IS possible to get the fddshow audio decoder's crossover to work...... then you would have to set that as default audiodecoder in winamp...Vlc...WMP...Wmpc...etc My first Crossover was the foobar one and started looking for tons of plugins... but in the end I ended up with something I already had... K-lite codecpack -> fddshow

succes

[nstewart]

11 years but.. I actually NEED this for a project. It's not "audiophile" but it is a biamp system taking advantage of a couple of otherwise great for my purposes drivers that need some eq and efficiency matching.