Re: [music-dsp] proper scaling of an FFT

On Wed, Nov 4, 2015, at 04:56 PM, Phil Burk wrote: > Is there a "right" way or a "wrong" way or is there just "my" way? I think in the context of a library there are two questions here -- what scaling should an FFT implementation use (behind the scenes in the functions it provides), and what

Re: [music-dsp] warts in JUCE

(I have never used JUCE, don't know its interfaces, and have no side to take) On Fri, Sep 4, 2015, at 08:38 PM, robert bristow-johnson wrote: > i find it odd that it seems to all inherent to a parcel of sound > represented in a computer are the number of samples and the number of > channels

Re: [music-dsp] warts in JUCE

On Sat, Sep 5, 2015, at 01:23 AM, robert bristow-johnson wrote: > same problem (if that *is* a problem) if you pass sampleRate alongside > the AudioSampleBuffer rather than inside of it. I agree, that would be the worst of both worlds. (At least from this perspective -- there may be other

Re: [music-dsp] warts in JUCE

On Sun, Sep 6, 2015, at 01:50 PM, robert bristow-johnson wrote: > otherwise, you're always going to be passing sampleRate > along with every AudioSampleBuffer. This bit here -- this is the part that doesn't match real practice. You would typically pass the sample rate when you construct a

Re: [music-dsp] warts in JUCE

On Sat, Sep 5, 2015, at 11:37 PM, Daniel Varela wrote: > Robert is right, sample rate is part of the audio information Robert is certainly right -- an encapsulation of an audio buffer in isolation must include the sample rate. I don't expect anyone in this thread disagrees with that. All the

Re: [music-dsp] Real-time pitch shifting?

On Sat, 19 May 2018, at 21:34, RJ Skerry-Ryan wrote: > It may not be the state of the art, but RubberBand > is, I believe, the best open source > pitch shift / time stretch library out there at the moment, and can run in > realtime on modern CPUs. See here

Re: [music-dsp] Real-time pitch shifting?

On Mon, 21 May 2018, at 19:54, Matt Ingalls wrote: > "Rubber Band implements pitch-shifting using the standard technique of > resampling… > This method has the advantage of making the amount of pitch adjustment > extremely accurate and stable, even for small changes.” It also means you only

Re: [music-dsp] Real-time pitch shifting?

On Tue, 22 May 2018, at 18:31, Matt Ingalls wrote: > As far as I can understand it, it seems to be along the same lines as > that Bernsee blog, but with peak detection. I think that should work a lot better than what you have -- although Eric Brombaugh's experience earlier in the thread may be

Re: [music-dsp] Real-time pitch shifting?

On Tue, 22 May 2018, at 14:09, Sound of L.A. Music and Audio wrote: > With the voice it is even more tricky, since the formant shaping is > different for other frequencies. One reason is, that there are more than > one "equalizer" involved. Just putting the whole track to a hight frequ > will

Re: [music-dsp] Real-time pitch shifting?

On Mon, 21 May 2018, at 22:08, robert bristow-johnson wrote: > From: "Chris Cannam" <can...@all-day-breakfast.com> > > I assume this wouldn't work with your use-case as you want to keep the > > frames synchronised so you can resynthesise everything in one go.

Re: [music-dsp] looking for a flexible synthesis system technically and legally appropriate for iOS development

On 8 February 2011 15:50, Stefan Kersten s...@k-hornz.de wrote: the known precedents make it a risky undertaking trying to distribute _any_ GPL'd application through the app store, because apple might decide to take it out in any moment; not a sound foundation to build any business model on ...

Re: [music-dsp] Window presum synthesis

On 20 April 2012 14:15, Domagoj Šarić dsar...@gmail.com wrote: ...but unfortunately all of these however seem to stop at the analysis stage (i.e. we've gotten a more precise spectrum and that's all we need). Mark Dolson's CARL phase vocoder (http://www.crca.ucsd.edu/cmusic/cmusic.html) from

Re: [music-dsp] stuck with filter design

On 18 November 2012 22:24, Bjorn Roche bj...@xowave.com wrote: Great. I guess that means LADSPA does not use the usual [-1,1] range. LADSPA doesn't enforce anything -- it's really up to the host. But the spec in header does say For audio it is generally assumed that 1.0f is the `0dB' reference

[music-dsp] Last call: SoundSoftware 2013: Workshop on Software and Data for Audio and Music Research

*** LAST CALL FOR PARTICIPATION *** *** New Registration Deadline: 24 June 2013 *** SoundSoftware 2013: Workshop on Software and Data for Audio and Music Research Queen Mary University of London 26 June 2013 http://soundsoftware.ac.uk/soundsoftware2013 If you're interested in attending our

[music-dsp] SoundSoftware 2014: Third Workshop on Software and Data for Audio and Music Research

(please forward as appropriate!) *** CALL FOR PARTICIPATION *** SoundSoftware 2014: Third Workshop on Software and Data for Audio and Music Research Queen Mary University of London, 8 July 2014 About the workshop: http://soundsoftware.ac.uk/soundsoftware2014 Registration link:

[music-dsp] Last call to register for SoundSoftware 2014!

suggests it will be just as rewarding. We look forward to seeing you there! Chris Cannam SoundSoftware.ac.uk -- 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