adrianfreed
commented
9 years ago According to some sources sonogram is the old world for spectrogram.
<research related aside that could be a distraction: Also note that old lore from Helmholtz suggests relative phase of partials is not audible. This was debunked in 1947 and regularly ever since. I am working with Meyer Sound on this because it is of great interest to John Meyer and David set them on an interesting path towards research on it. From the musical perspective this phase thing is interesting because one of the things that happens when you change the relative phase of partials is that you modulate the roughness of the perceived sound. This may have consequences for inharmonic sounds such as bells.
analyzer~ is buggy and I have no ETA for a fix so try to avoid it for the analysis side.
It sounds like the model being fitted here supposes that the original signal has fixed frequency partials.
My approach would be as follows:
Giant FFT on the whole file Identify the 20-30 largest peaks.
Feed the sound BACKWARDS into a bank of resonators tuned to the frequencies of those peaks with relatively short decay rates. Estimate the energy out of each filter using RMS or something similar. Reverse the resulting tracks back to the right temporal order.
You can fit the ADSR to these tracks, turn them into SDIF or whatever.
The reason for doing this in reverse is that it avoids ringing of the resonators messing up the attack.
Lots of us have had a hand at this sort of thing (Miller, Ali, John MacCallum, MichaelZ etc.) so there may be some value in collecting up the various approaches.
On Dec 14, 2014, at 1:33 PM, Edmund Campion notifications@github.com wrote:
This is assignment to build a Spectral Modeling patch/app for Music 29 SP2015. The due date is no later than February 20, 2015.
Here are basic details, but please sit down with Ed to go over everything before starting.
Description:
Load a soundfile of a single instrumental sample playing one note (e.g. a Sather Bell strike or a Saxophone playing G etc..). Playback of the sample should be on separate gain, have loop/no loop, have mute/unmute button.
Prefader gain from sample is sent to sonogram analysis onboard with patch.
Using sonogram analysis and probing sinusoid locate specific frequencies inside the sample. (This will entail going back to frequency domain analysis patch in MUSIC 29 and working out some better and more refined way to capture probed frequencies)
Once a frequency is located it gets added to an individual frequency control track that can post the frequency selected, play the frequency constantly, have full gain control, and have a mute/unmute button. Output should be mono.
Once many frequency are located, all the frequencies should appear as a mixing board, allowing for fine tuning of amplitudes and control of every frequency found. (maybe an upper limit of 20 frequencies?).
The mixed output of all frequencies and the original source sample should be sent to a dual master mono output, one for sinusoids, one for sample. The normal default of the master mono outpust gain is a slider output for each. The sinusoid master output will have a button that selects an ADSR feature. When the ADSR component is checked, then a window is enable to create a specific envelope that will auto match to the length of the currently loaded soundfile and will trigger each time the loaded soundfile begins.
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EdmundCampion
equilet
mattschumaker
This is assignment to build a Spectral Modeling patch/app for Music 29 SP2015. The due date is no later than February 20th, 2015.
Here are basic details, but please sit down with Ed to go over everything before starting.
Description:
Load a soundfile of a single instrumental sample playing one note (e.g. a Sather Bell strike or a Saxophone playing G etc..). Playback of the sample should be on separate gain, have loop/no loop, have mute/unmute button.
Prefader gain from sample is sent to sonogram analysis onboard with patch.
Using sonogram analysis and probing sinusoid locate specific frequencies inside the sample. (This will entail going back to frequency domain analysis patch in MUSIC 29 and working out some better and more refined way to capture probed frequencies)
Once a frequency is located it gets added to an individual frequency control track that can post the frequency selected, play the frequency constantly, have full gain control, and have a mute/unmute button. Output should be mono.
Once many frequency are located, all the frequencies should appear as a mixing board, allowing for fine tuning of amplitudes and control of every frequency found. (maybe an upper limit of 20 frequencies?).
The mixed output of all frequencies and the original source sample should be sent to a dual master mono output, one for sinusoids, one for sample. The normal default of the master mono outpust gain is a slider output for each. The sinusoid master output will have a button that selects an ADSR feature. When the ADSR component is checked, then a window is enable to create a specific envelope that will auto match to the length of the currently loaded soundfile and will trigger each time the loaded soundfile begins.