Julia Merino

Hi, I am implementing a patch to calculate the loudness of an incoming signal for an interactive live performance system. This system will have a parameter to change the duration of the sliding window between 1 and 30 seconds. My first approach was to use average~ however this doesn't work when suddenly changing the time interval from 1s to 10s for instance as I guess the buffer doesn't have access to the samples of the last 10s. For this reason, I've thought of implementing the average calculation by always storing in a delay buffer the last 30 seconds and from there get the last T time samples to calculate the accumulation (integral) and then divide that between T-time samples to get the average.

My question is how to get the last X samples from a delay~ buffer?

Hi, I am implementing a patch to calculate the loudness of an incoming signal for an interactive live performance system. This system will have a parameter to change the duration of the sliding window between 1 and 30 seconds. My first approach was to use average~ however this doesn't work when suddenly changing the time interval from 1s to 10s for instance as I guess the buffer doesn't have access to the samples of the last 10s. For this reason, I've thought of implementing the average calculation by always storing in a delay buffer the last 30 seconds and from there get the last T time samples to calculate the accumulation (integral) and then divide that between T-time samples to get the average.
My question is how to get the last X samples from a delay~ buffer?

Thanks a lot for you help!

Hi, I am implementing a patch to calculate the loudness of an incoming signal for an interactive live performance system. 

integral-of-a-signal-using-delay-buffer-how-to-get-the-last-x-samples-from-a-delay~-buffer

record 30 seconds of audio into buffer and read whatever portion of it using peek .

As alternative, you could store average per period, like 10 - 100 ms instead of each sample which would make storage and calculations much more effective

record 30 seconds of audio into buffer and read whatever portion of it using peek .
As alternative,  you could store average per period, like 10 - 100 ms instead of each sample which would make storage and calculations much more effective

That's a good shout. However, how could I constantly update the buffer with the new incoming samples?

get average level of 1 second which is your minimum report length

by taking 100 readings and averaging them.

meter~ set to report 10ms interval -> zl.group 100 = 1000 ms

30 average values get inserted into coll , you set how many

last seconds get taken into calculation of average level. 

here is what I woud do :
get average level of 1 second which is your minimum report length
by taking 100 readings and averaging them.
meter~ set to report 10ms interval -> zl.group 100 = 1000 ms

30 average values get inserted into coll , you set how many
last seconds get taken into calculation of average level. 



Thank you very much! I ended up using a slightly different approach using zl.stream instead but all seems to be working fine!

if you need to access samples other than the last one, why not use a tapping buffer instead of delay~ ?

i build loudness metering also around average~ because it is actually cheaper than using individual math objects. 

no idea why you would like to modulate the window size though. for loudness the mean-sqare is normally measured in chunks of 400 ms with 75% overlap. which is roughly what you get when using average~ with 400, followed by rampsmooth~ with 400, and then snapshot~ the values to be displayed (or accumulated).

alternatively to the rampsmooth+ you could take the analysed 100ms values and interpolate those values over 5 values. (2 in both directions. nowadays done using zl.stream, exactly.)

i build loudness metering also around average~ because it is actually cheaper than using individual math objects. 
no idea why you would like to modulate the window size though. for loudness the mean-sqare is normally measured in chunks of 400 ms with 75% overlap. which is roughly what you get when using average~ with 400, followed by rampsmooth~ with 400, and then snapshot~ the values to be displayed (or accumulated).

Thanks Roman! I actually checked your loudness meter :)

My client wants to be able to adjust the sliding window. I went for taking the avg~ in chunks of 100ms and making a 300-long list (30 seconds) with zl.stream so I could always have access to the last N average samples if the sliding window was increased.

Thanks Roman! I actually checked your loudness meter :)
My client wants to be able to adjust the sliding window. I went for taking the avg~ in chunks of 100ms and making a 300-long list (30 seconds) with zl.stream so I could always have access to the last N average samples if the sliding window was increased.

https://cycling74.com/forums/loudness-not-amplitude-control

Integral of a signal using delay buffer: how to get the last X samples from a delay~ buffer?