Windows relevant stuff. I don't think it's necessarily sound card dependent. A sound card for most uses is important when you record a sound into the 'puter and when you play it back out. Flanging, phasing, choruses, delays and their ilk(s) can all be considered mathematic operations and are quite splendidly handled by computers. Computer DSP-heads (much smarter smarter than you or me) have been writing audio applications for years now. Once you've got a sampled piece of music into your computer you can dick around with it as much as you want. Here's a start: http://www.hitsquad.com/smm/programs/Audacity/ A freeware audio editor a number of people I respect recommend (I've never used it). Supports VST plug-ins, which means HUNDREDS of FREE effects plug-ins are available. Audio editors allow you to pull up an audio (.wav) file from your computer and do various things to it... cut, paste, volume operations, transforms, effects, etc. Browse around http://www.hitsquad.com/smm/. Or go completely ape-shit, stomp-boxes-from-hell-in-a-chain-nine-miles-long, where-the-hell-did-THAT-sound-come-from; and try out Ross Bencina's excellent Audiomulch. His native plug-ins are scary weird.
How to start-off??
Started by ●November 2, 2005
Reply by ●November 4, 20052005-11-04
Reply by ●November 4, 20052005-11-04
Richard Owlett wrote:> Bevan Weiss wrote: >> Richard Owlett wrote: >> >>> Follow up set to comp.dsp as that is group I follow and am thinking >>> in those terms >>> >>> Bevan Weiss wrote: >>> >>>> krg wrote: >>>> >>>>> Posted To: >>>>> comp.dsp, sci.electronics.misc, rec.music.makers.synth >>>>> >>>>> Hello people, >>>>> Getting straight on to the topic, I read a lot about cool >>>>> effects like flanging, reverb, and ... <you got the idea>, and also >>>>> that how you can create these with your own programs in C. I believe I >>>>> know enough C to carry things on my own, but I just need a start. >>>>> All I want to know is how to get those samples of that fav song >>>>> of mine that go out of my PC, so that I can really grab them and do >>>>> algebra that sounds good to me and finally play it. >>>>> Thanks for reading my post and further more if you actually >>>>> help. >>>>> >>>> >>>> Not quite the right newsgroup for that question, however... >>>> You'll either need a decoder for the particular file format that >>>> you're wanting to play around with, or you'll have to look into >>>> directly modifying the Direct-Sound (or equivalent) buffers to allow >>>> for the effects you desire. >>>> >>>> Once you've got the buffer containing your samples then it's just a >>>> matter of applying these samples to a filter to produce the effects >>>> you desire. The filter structure would be entirely up to you, >>>> however given that it's audio that you're dealing with an IIR filter >>>> would probably suffice for pretty much everything that you're >>>> wishing to do. >>>> >>>> Quite a few modern sound cards already have the capability to >>>> provide these effects on-board (i.e. without additional CPU load) so >>>> you'd just need to look into utilizing those features if this is >>>> more what you're trying to do. >>> >>> >>> What cards would this be or can you suggest appropriate Google search >>> terms? >>> >>> What effects are possible on which cards? >>> >>> I'm mulling some wild ideas and need grist for the mill so to speak. >>> >>> Practicality is the *LEAST* of my concerns. >>> I am in "brain storming" mode ;} >> >> >> The creative audigy cards are the first that come to mind. They have >> a DSP onboard them which can be programmed to produce almost any >> effect you can imagine. You would have to do some coding if it was to >> be something really different than the norm however. >> > > I should have more carefully stated my question, especially considering > tread I tacked it to [thread had triggered musing] > > I'm interested in filtering a microphone input channel before its sent > on to some "off the shelf" [commercial or open source] speech > recognition software that expects to see input from a 'standard' sound > card. The intention is to offload my filtering experiments from main cpu. > > I repeat that this is "brain storming" mode as I won't have time to > really dive in until I retire in a few years.I'm pretty sure the audigy card can be made to do this. If you have a look into the kxproject drivers, then I'm sure they can apply dsp effects before passing the data into a directsound buffer or similar.
Reply by ●November 4, 20052005-11-04
Bevan Weiss wrote:> Richard Owlett wrote: > >> Bevan Weiss wrote: >> >>> Richard Owlett wrote: >>> >>>> Follow up set to comp.dsp as that is group I follow and am thinking >>>> in those terms >>>> >>>> Bevan Weiss wrote: >>>> >>>>> krg wrote: >>>>> >>>>>> Posted To: >>>>>> comp.dsp, sci.electronics.misc, rec.music.makers.synth >>>>>> >>>>>> Hello people, >>>>>> Getting straight on to the topic, I read a lot about cool >>>>>> effects like flanging, reverb, and ... <you got the idea>, and also >>>>>> that how you can create these with your own programs in C. I >>>>>> believe I >>>>>> know enough C to carry things on my own, but I just need a start. >>>>>> All I want to know is how to get those samples of that fav >>>>>> song >>>>>> of mine that go out of my PC, so that I can really grab them and do >>>>>> algebra that sounds good to me and finally play it. >>>>>> Thanks for reading my post and further more if you actually >>>>>> help. >>>>>> >>>>> >>>>> Not quite the right newsgroup for that question, however... >>>>> You'll either need a decoder for the particular file format that >>>>> you're wanting to play around with, or you'll have to look into >>>>> directly modifying the Direct-Sound (or equivalent) buffers to >>>>> allow for the effects you desire. >>>>> >>>>> Once you've got the buffer containing your samples then it's just a >>>>> matter of applying these samples to a filter to produce the effects >>>>> you desire. The filter structure would be entirely up to you, >>>>> however given that it's audio that you're dealing with an IIR >>>>> filter would probably suffice for pretty much everything that >>>>> you're wishing to do. >>>>> >>>>> Quite a few modern sound cards already have the capability to >>>>> provide these effects on-board (i.e. without additional CPU load) >>>>> so you'd just need to look into utilizing those features if this is >>>>> more what you're trying to do. >>>> >>>> >>>> >>>> What cards would this be or can you suggest appropriate Google >>>> search terms? >>>> >>>> What effects are possible on which cards? >>>> >>>> I'm mulling some wild ideas and need grist for the mill so to speak. >>>> >>>> Practicality is the *LEAST* of my concerns. >>>> I am in "brain storming" mode ;} >>> >>> >>> >>> The creative audigy cards are the first that come to mind. They have >>> a DSP onboard them which can be programmed to produce almost any >>> effect you can imagine. You would have to do some coding if it was >>> to be something really different than the norm however. >>> >> >> I should have more carefully stated my question, especially >> considering tread I tacked it to [thread had triggered musing] >> >> I'm interested in filtering a microphone input channel before its sent >> on to some "off the shelf" [commercial or open source] speech >> recognition software that expects to see input from a 'standard' sound >> card. The intention is to offload my filtering experiments from main cpu. >> >> I repeat that this is "brain storming" mode as I won't have time to >> really dive in until I retire in a few years. > > > I'm pretty sure the audigy card can be made to do this. If you have a > look into the kxproject drivers, then I'm sure they can apply dsp > effects before passing the data into a directsound buffer or similar.I think that http://kxproject.lugosoft.com/direct.php?language=en implies the answer is "No" when it says The 'Direct SPDIF Recording' is activated via kX Mixer (on the main page). Due to the nature of this recording method, the signal completely bypasses the DSP, which is why it is not affected by any uploaded effects or by the kX Mixer sliders and recording levels. I don't think I would gain anything over using an external DSP board (used as 'fancy' preamp).
Reply by ●November 4, 20052005-11-04
Richard Owlett wrote:> Bevan Weiss wrote: >> Richard Owlett wrote: >> >>> Bevan Weiss wrote: >>> >>>> Richard Owlett wrote: >>>> >>>>> Follow up set to comp.dsp as that is group I follow and am thinking >>>>> in those terms >>>>> >>>>> Bevan Weiss wrote: >>>>> >>>>>> krg wrote: >>>>>> >>>>>>> Posted To: >>>>>>> comp.dsp, sci.electronics.misc, rec.music.makers.synth >>>>>>> >>>>>>> Hello people, >>>>>>> Getting straight on to the topic, I read a lot about cool >>>>>>> effects like flanging, reverb, and ... <you got the idea>, and also >>>>>>> that how you can create these with your own programs in C. I >>>>>>> believe I >>>>>>> know enough C to carry things on my own, but I just need a start. >>>>>>> All I want to know is how to get those samples of that fav >>>>>>> song >>>>>>> of mine that go out of my PC, so that I can really grab them and do >>>>>>> algebra that sounds good to me and finally play it. >>>>>>> Thanks for reading my post and further more if you actually >>>>>>> help. >>>>>>> >>>>>> >>>>>> Not quite the right newsgroup for that question, however... >>>>>> You'll either need a decoder for the particular file format that >>>>>> you're wanting to play around with, or you'll have to look into >>>>>> directly modifying the Direct-Sound (or equivalent) buffers to >>>>>> allow for the effects you desire. >>>>>> >>>>>> Once you've got the buffer containing your samples then it's just >>>>>> a matter of applying these samples to a filter to produce the >>>>>> effects you desire. The filter structure would be entirely up to >>>>>> you, however given that it's audio that you're dealing with an IIR >>>>>> filter would probably suffice for pretty much everything that >>>>>> you're wishing to do. >>>>>> >>>>>> Quite a few modern sound cards already have the capability to >>>>>> provide these effects on-board (i.e. without additional CPU load) >>>>>> so you'd just need to look into utilizing those features if this >>>>>> is more what you're trying to do. >>>>> >>>>> >>>>> >>>>> What cards would this be or can you suggest appropriate Google >>>>> search terms? >>>>> >>>>> What effects are possible on which cards? >>>>> >>>>> I'm mulling some wild ideas and need grist for the mill so to speak. >>>>> >>>>> Practicality is the *LEAST* of my concerns. >>>>> I am in "brain storming" mode ;} >>>> >>>> >>>> >>>> The creative audigy cards are the first that come to mind. They >>>> have a DSP onboard them which can be programmed to produce almost >>>> any effect you can imagine. You would have to do some coding if it >>>> was to be something really different than the norm however. >>>> >>> >>> I should have more carefully stated my question, especially >>> considering tread I tacked it to [thread had triggered musing] >>> >>> I'm interested in filtering a microphone input channel before its >>> sent on to some "off the shelf" [commercial or open source] speech >>> recognition software that expects to see input from a 'standard' >>> sound card. The intention is to offload my filtering experiments from >>> main cpu. >>> >>> I repeat that this is "brain storming" mode as I won't have time to >>> really dive in until I retire in a few years. >> >> >> I'm pretty sure the audigy card can be made to do this. If you have a >> look into the kxproject drivers, then I'm sure they can apply dsp >> effects before passing the data into a directsound buffer or similar. > > I think that http://kxproject.lugosoft.com/direct.php?language=en > implies the answer is "No" when it says > > The 'Direct SPDIF Recording' is activated via kX Mixer (on the main > page). Due to the nature of this recording method, the signal > completely bypasses the DSP, which is why it is not affected by any > uploaded effects or by the kX Mixer sliders and recording levels. > > I don't think I would gain anything over using an external DSP board > (used as 'fancy' preamp).That would imply that you already have the signal digitized. If you're using the sound card to perform the sampling process, then it shouldn't be a problem. The issue is primarily in the re-sampling process that occurs with SPDIF signals from things such as DVD players etc. If you're not worried about the sample rate etc then this shouldn't be an issue. This is the situation I see you being in. You're really just after some 'filtered' version of the analog input being fed into a directsound buffer for reading by the voice recognition software. This should be quite do-able, though you'd want to see what kind of DSP capabilities are available with that driver package in this regard. I recall last time I used it the DSP side of things was really nicely done. It represented everything using blocks with various inputs/outputs as needed, then you could just connect them together with virtual wires. This would then implement a particular signal flow.
Reply by ●November 4, 20052005-11-04
Bevan Weiss wrote:> Richard Owlett wrote:...>> I don't think I would gain anything over using an external DSP board >> (used as 'fancy' preamp). > > > That would imply that you already have the signal digitized. ...Not necessarily. A DSP board can accept continuous audio, process it, and put continuous audio out. From the outside, it can behave like an all-analog filter (albeit with some unusual properties). Sampling the signal more than once may not be a theoretically elegant solution, but providing a desired result with resources already on hand is elegance of a different sort. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●November 4, 20052005-11-04
Jerry Avins wrote:> Bevan Weiss wrote: > >> Richard Owlett wrote: > > > ... > >>> I don't think I would gain anything over using an external DSP board >>> (used as 'fancy' preamp). >> >> >> >> That would imply that you already have the signal digitized. ... > > > Not necessarily. A DSP board can accept continuous audio, process it, > and put continuous audio out. From the outside, it can behave like an > all-analog filter (albeit with some unusual properties). > > Sampling the signal more than once may not be a theoretically elegant > solution, but providing a desired result with resources already on hand > is elegance of a different sort. > > JerryI was interested in using the onboard DSP to manipulate the signal coming via one of the cards _analog_ inputs. I had gotten the, apparently mistaken impression,from http://kxproject.lugosoft.com/direct.php?language=en that one could only pass digitized signals to the DSP. Bevan was saying that the digital version of an analog channel input can also be passed to the DSP. That also makes other portions of the site I only skimmed make more sense. Part of the problem is that I've had no previous interests in the guts of sound cards.
Reply by ●November 4, 20052005-11-04
Jerry Avins wrote:> Bevan Weiss wrote: >> Richard Owlett wrote: > > ... > >>> I don't think I would gain anything over using an external DSP board >>> (used as 'fancy' preamp). >> >> >> That would imply that you already have the signal digitized. ... > > Not necessarily. A DSP board can accept continuous audio, process it, > and put continuous audio out. From the outside, it can behave like an > all-analog filter (albeit with some unusual properties). > > Sampling the signal more than once may not be a theoretically elegant > solution, but providing a desired result with resources already on hand > is elegance of a different sort. > > JerryAhh, sorry Jerry I was was referring to Richards mention of direct SPDIF recording. This would imply that he already has the signal digitized otherwise he'd just be looking at the standard AC97 ADC sampling process, which I'm sure can be routed through the DSP before being placed in a main memory buffer.
Reply by ●November 4, 20052005-11-04
Richard Owlett wrote: ...> I was interested in using the onboard DSP to manipulate the signal > coming via one of the cards _analog_ inputs. I had gotten the, > apparently mistaken impression,from > http://kxproject.lugosoft.com/direct.php?language=en > that one could only pass digitized signals to the DSP.Many DSP project boards include audio codecs. I can lend you one. ... Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●November 6, 20052005-11-06
Jerry Avins wrote:> Richard Owlett wrote: > > ... > >> I was interested in using the onboard DSP to manipulate the signal >> coming via one of the cards _analog_ inputs. I had gotten the, >> apparently mistaken impression,from >> http://kxproject.lugosoft.com/direct.php?language=en >> that one could only pass digitized signals to the DSP. > > > Many DSP project boards include audio codecs. I can lend you one. > >Thank you, but I'm not that far along yet. I know about as much about speech recognition as I do DSP [ not a thimble full ;] When reading posts to comp.speech.users about using SR in real world, I have "gut feel" that SR as a system is not properly factored. *EVERYTHING* is dumped on the main CPU. As there is so much discussion of appropriate acoustic environment and microphone used. I would apportion independent processors to: 1. filter that optimized speaker's formants 1.a. optimistically do some sort of "shifting" [pitch and ???] to a "standard" 2. filter to de-emphasize local interference 3. phoneme recognition 4. AI-like section that chooses between "to","too","two" etc. Have I displayed enough ignorance to show why I'm not quite ready to implement any hardware solution? I also have a problem with SR industry/marketplace. They are interested in "continuous recognition". I want "command and control"/"discrete recognition". Several VAR's have been patient enough to explain the problem. Having ~4 years till retirement, I should be able to become somewhat educated in DSP and SR.
Reply by ●November 7, 20052005-11-07
Richard Owlett wrote: ...> When reading posts to comp.speech.users about using SR in real world, I > have "gut feel" that SR as a system is not properly factored. > *EVERYTHING* is dumped on the main CPU.Why not? It's cheaper that way. One of my early radio designs passed the signal through the same tube three times: once at RF, again at IF, and a third time at audio. I resented needing a separate tube for the local oscillator, but at least it also contained the diode for the detector.> As there is so much discussion of appropriate acoustic environment and > microphone used. I would apportion independent processors to: > 1. filter that optimized speaker's formants > 1.a. optimistically do some sort of "shifting" [pitch and ???] to a > "standard" > 2. filter to de-emphasize local interference > 3. phoneme recognition > 4. AI-like section that chooses between "to","too","two" etc.Some people have one car to impress clients, another to take to the market, and yet another for vacations. (You need only one roof on a three-story house.)> Have I displayed enough ignorance to show why I'm not quite ready to > implement any hardware solution?Nah. Try harder. :-)> I also have a problem with SR industry/marketplace. They are interested > in "continuous recognition". I want "command and control"/"discrete > recognition". Several VAR's have been patient enough to explain the > problem.I don't even know what you're writing about.> Having ~4 years till retirement, I should be able to become somewhat > educated in DSP and SR.-- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
