Re: [music-dsp] Simulating Valve Amps
I'm aquainted to DSP and analogue electronics and have played a lot of guitar over the last 18 years.. I still think there is nothing like the sound of a good old valve amp.. On Tue, Jun 17, 2014 at 10:13 AM, Thomaz Oliveira thomazcha...@gmail.com wrote: I have made some simulations of valve amps.. I have writen some articles and a PHD thesis on why these amps are so hard to model... I'm sending you the link of one article on this topic: if you have any questions about simulation of valve amps please contact me. here is the link: http://www.dcc.ufla.br/infocomp/images/artigos/v12.1/art02.pdf On Tue, Jun 17, 2014 at 6:30 AM, Nigel Redmon earle...@earlevel.com wrote: Well…yes, aliasing is the main issue that separates the digital world from analog when it comes to amp modeling, but no, I don’t think it’s the main issue in simulating a good amp :-) There are a lot of details in simulating classic amps—the controls of the passive filters interact, etc. And you have to recreate the filtering before and after the “tube”—guitar amps aren’t about flat frequency response. And the cabinets are a huge part of the sound. Then there’s the approach that’s more like sampling (Kemper Profiling Amp), a totally different direction—does it sound better? I don’t know, it’s very cool, but I haven’t listened to it enough to know (only at NAMM Shows). Anyway, just keep in mind that the particular classic amps don’t sound “better” simply because they are analog. They sound better because over the decades they’ve been around, they survived—because they do sound good. There are plenty of awful sounding analog guitar amps (and compressors, and preamps, and…) that didn’t last because they didn’t sound particularly good. Then, the modeling amp has the disadvantage that they are usually employed to recreate a classic amp exactly. So the best they can do is break even in sound, then win in versatility. And an AC-30 or Matchless preset on a modeler that doesn’t sound exactly like the amp it models loses automatically—even if it sounds better— because it failed to hit the target. (And it doesn’t helped that amps of the same model don’t necessarily sound the same. At Line 6, we would borrow a coveted amp—one that belonged to a major artist and was highly regarded, for instance, or one that was rented out for sessions because it was known to sound awesome.) On Jun 17, 2014, at 2:09 AM, Aengus Martin aen...@am-process.org wrote: Hi, I've only the vaguest idea of this area but I do find it interesting. From what you said, Nigel, aliasing is the main issue. Is it the case then that amp modeling would be more or less a solved problem if you could sample at arbitrarily high rates? Cheers, Aengus. On Tue, Jun 17, 2014 at 6:58 PM, Nigel Redmon earle...@earlevel.com wrote: On Jun 16, 2014, at 7:51 PM, robert bristow-johnson r...@audioimagination.com wrote: one thing that is hard to replicate is a sample rate that is infinity (which is how i understand continuous-time signals to be). but i don't think you should need to have such a high sample rate. one thing we know is that for *polynomial curves* (which are mathematical abstractions and maybe have nothing to do with tube curves), that for a bandwidth of B in the input and a polynomial curve of order N, the highest generated frequency is N*B so the sample rate should be at least (N+1)*B to prevent any of these generated images from aliasing down to below the original B. if you can prevent that, you can filter out any of the aliased components and downsample to a sample rate sufficient for B (which is at least 2*B). This really goes out the window when you’re modeling amps, though. The order of the polynomial is too high to implement practically (that is, you won’t end up utilizing the oversampling rate necessary to follow it), so you still be dealing with aliasing. Modern high gain amps have huge gain *after* saturation. In practical terms, you round into it (with a polynomial, for instance), then just hard clip from there on out, and there goes your polynomial (it can be replaced by an approximation that's very high order, but what’s the point). Anyway, you pay your money, you make your choices. Obviously some really good musicians making really interesting music use modeling amps. They don’t have to be better than tubes, in order to be a win, just good enough to be worth all the benefits. If you’re a session music, you can bring in the truck with all of the kinds of amps that might be called on, or you can bring a modeling amp, for instance. And going direct into the PA or your recoding equipment…etc. I’m not going to make judgments on what people should like, so I’ll leave it at that. One happy thing about the aliasing is that, given a decent level of oversampling, it won’t be bad at lower overdrive levels. At the higher
Re: [music-dsp] Simulating Valve Amps
I have made some simulations of valve amps.. I have writen some articles and a PHD thesis on why these amps are so hard to model... I'm sending you the link of one article on this topic: if you have any questions about simulation of valve amps please contact me. here is the link: http://www.dcc.ufla.br/infocomp/images/artigos/v12.1/art02.pdf On Tue, Jun 17, 2014 at 6:30 AM, Nigel Redmon earle...@earlevel.com wrote: Well…yes, aliasing is the main issue that separates the digital world from analog when it comes to amp modeling, but no, I don’t think it’s the main issue in simulating a good amp :-) There are a lot of details in simulating classic amps—the controls of the passive filters interact, etc. And you have to recreate the filtering before and after the “tube”—guitar amps aren’t about flat frequency response. And the cabinets are a huge part of the sound. Then there’s the approach that’s more like sampling (Kemper Profiling Amp), a totally different direction—does it sound better? I don’t know, it’s very cool, but I haven’t listened to it enough to know (only at NAMM Shows). Anyway, just keep in mind that the particular classic amps don’t sound “better” simply because they are analog. They sound better because over the decades they’ve been around, they survived—because they do sound good. There are plenty of awful sounding analog guitar amps (and compressors, and preamps, and…) that didn’t last because they didn’t sound particularly good. Then, the modeling amp has the disadvantage that they are usually employed to recreate a classic amp exactly. So the best they can do is break even in sound, then win in versatility. And an AC-30 or Matchless preset on a modeler that doesn’t sound exactly like the amp it models loses automatically—even if it sounds better— because it failed to hit the target. (And it doesn’t helped that amps of the same model don’t necessarily sound the same. At Line 6, we would borrow a coveted amp—one that belonged to a major artist and was highly regarded, for instance, or one that was rented out for sessions because it was known to sound awesome.) On Jun 17, 2014, at 2:09 AM, Aengus Martin aen...@am-process.org wrote: Hi, I've only the vaguest idea of this area but I do find it interesting. From what you said, Nigel, aliasing is the main issue. Is it the case then that amp modeling would be more or less a solved problem if you could sample at arbitrarily high rates? Cheers, Aengus. On Tue, Jun 17, 2014 at 6:58 PM, Nigel Redmon earle...@earlevel.com wrote: On Jun 16, 2014, at 7:51 PM, robert bristow-johnson r...@audioimagination.com wrote: one thing that is hard to replicate is a sample rate that is infinity (which is how i understand continuous-time signals to be). but i don't think you should need to have such a high sample rate. one thing we know is that for *polynomial curves* (which are mathematical abstractions and maybe have nothing to do with tube curves), that for a bandwidth of B in the input and a polynomial curve of order N, the highest generated frequency is N*B so the sample rate should be at least (N+1)*B to prevent any of these generated images from aliasing down to below the original B. if you can prevent that, you can filter out any of the aliased components and downsample to a sample rate sufficient for B (which is at least 2*B). This really goes out the window when you’re modeling amps, though. The order of the polynomial is too high to implement practically (that is, you won’t end up utilizing the oversampling rate necessary to follow it), so you still be dealing with aliasing. Modern high gain amps have huge gain *after* saturation. In practical terms, you round into it (with a polynomial, for instance), then just hard clip from there on out, and there goes your polynomial (it can be replaced by an approximation that's very high order, but what’s the point). Anyway, you pay your money, you make your choices. Obviously some really good musicians making really interesting music use modeling amps. They don’t have to be better than tubes, in order to be a win, just good enough to be worth all the benefits. If you’re a session music, you can bring in the truck with all of the kinds of amps that might be called on, or you can bring a modeling amp, for instance. And going direct into the PA or your recoding equipment…etc. I’m not going to make judgments on what people should like, so I’ll leave it at that. One happy thing about the aliasing is that, given a decent level of oversampling, it won’t be bad at lower overdrive levels. At the higher the overdrive levels, the harder it is to hear aliasing through all that harmonic distortion you’re generating. So it could be worse... Stephen, I’m a former Line 6 DSP engineer, in the interests of full disclosure. I have no idea how you Fender modeler rates.
Re: [music-dsp] WDF Gyrator in BlockCompiler anyone ?
Hi, I tried to use the block compiler, though researchers from finland (from the same research lab) do not recomend to use it anynmore, since it became unstable and does not have support anymore since Dr Karjalainen passed away (the creator of block compiler. Newer WDF tools will soon be available. Cheers Thomaz -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] WDF Gyrator in BlockCompiler anyone ?
I did a research on that seminar you just presented I know that the block compiler was good enough for simple models. They worked up to a model of a tube audio transformer. After that no further work was produced, when it became unstable for complex models. Rafael paiva is working on a C++ environment for future WDF models (rcdpa...@gmail.com ), he could give you further feedback when it will be ready. Block compiler reads Clisp code which makes it somehow difficult to use. Cheers Thomaz -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
