Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Ethan Duni
On Sun, Mar 8, 2020 at 8:02 PM Spencer Russell wrote: > In fact, the the standard STFT analysis/synthesis pipeline is the same > thing as overlap-add "fast convolution" if you: > > 1. Use a rectangular window with a length equal to your hop size > 2. zero-pad each input frame by the length of you

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread zhiguang zhang
Nowhere was it mentioned that there was an across the frame multiplication with a scalar as far as manipulating the transform coefficients. That might make it time variant. My concept was in the domain of audio engineering which reads a side-chain signal to obtain attenuation factors in the conte

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Spencer Russell
On Sun, Mar 8, 2020, at 7:41 PM, Ethan Duni wrote: > FFT filterbanks are time variant due to framing effects and the circular > convolution property. They exhibit “perfect reconstruction” if you design the > windows correctly, but this only applies if the FFT coefficients are not > altered betwe

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread zhiguang zhang
so Ethan, what is your definition of time invariance? because you say it's not time invariant because of time domain aliasing but then you say there is delay due to compute time. delay due to window and compute time is unavoidable and not to be factored into time invariance / variance. coding no

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Ethan Duni
> > If the system is suitably designed (e.g. correct window and overlap), > you can filter using an FFT and get identical results to a time domain > FIR filter (up-to rounding/precision limits, of course). The > appropriate window and overlap process will cause all circular > convolution artefact

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Greg Maxwell
On Sun, Mar 8, 2020 at 11:41 PM Ethan Duni wrote: > FFT filterbanks are time variant due to framing effects and the circular > convolution property. They exhibit “perfect reconstruction” if you design the > windows correctly, but this only applies if the FFT coefficients are not > altered betwe

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Ethan Duni
No, MDCT TDAC is the same. Perfect reconstruction only obtains if the coefficients are not changed at all. Coding noise causes (uncancelled) time domain aliasing that is shaped according to the window design. Limiting this effect is a primary aspect of MDCT codec design. Ethan > On Mar 8, 202

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread zhiguang zhang
Audio compression by definition 'alters' the transform coefficients and they get perfect reconstruction with no aliasing due to the transform alone. In fact 'TDAC' or time domain aliasing cancellation is a hallmark of the MDCT or DCT type IV which is ubiquitous in audio codecs. On Sun, Mar 8, 202

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Ethan Duni
FFT filterbanks are time variant due to framing effects and the circular convolution property. They exhibit “perfect reconstruction” if you design the windows correctly, but this only applies if the FFT coefficients are not altered between analysis and synthesis. If you alter the FFT coefficient

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread zhiguang zhang
The system is memoryless just because it is based on the DFT and nothing else, which is also why it's time-invariant. unless you alter certain parameters in real-time like the window size or hop size or windowing function, etc, any input gives you the same output at any given time, which is the de

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread zhiguang zhang
Well I believe the system is LTI just because the DFT is LTI by definition. The impulse response of a rectangular window I believe is that of a sinc function, which has ripple artifacts. Actually, the overlap-add method (sorry I don't have time to dig into the differences between overlap-add and

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread robert bristow-johnson
> On March 8, 2020 2:00 PM Zhiguang Eric Zhang wrote: > > it is not causal because the zero-phase system does not depend on past samples > > > On Sun, Mar 8, 2020 at 1:58 PM Zhiguang Eric Zhang wrote: > > the frequency response is a function of the windowing function > > > > what frequenc

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Zhiguang Eric Zhang
it is not causal because the zero-phase system does not depend on past samples On Sun, Mar 8, 2020 at 1:58 PM Zhiguang Eric Zhang wrote: > the frequency response is a function of the windowing function > > On Sun, Mar 8, 2020 at 10:34 AM robert bristow-johnson < > r...@audioimagination.com> wrot

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Zhiguang Eric Zhang
the frequency response is a function of the windowing function On Sun, Mar 8, 2020 at 10:34 AM robert bristow-johnson < r...@audioimagination.com> wrote: > > > > On March 8, 2020 10:05 AM Ethan Duni wrote: > > > > > > It is physically impossible to build a causal, zero-phase system with > non-tr

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread robert bristow-johnson
> On March 8, 2020 10:05 AM Ethan Duni wrote: > > > It is physically impossible to build a causal, zero-phase system with > non-trivial frequency response. a system that operates in real time. when processing sound files you can pretend that you're looking at some "future" samples. i gues

Re: [music-dsp] FIR blog post & interactive demo

2020-03-08 Thread Ethan Duni
It is physically impossible to build a causal, zero-phase system with non-trivial frequency response. Ethan > On Mar 7, 2020, at 7:42 PM, Zhiguang Eric Zhang wrote: > >  > Not to threadjack from Alan Wolfe, but the FFT EQ was responsive written in C > and running on a previous gen MacBook P