SLIDE 1
1
Introduction
§ Types of delay-based audio effect
– Delay – Chorus – Flanger – Reverberation
2
Delay-based Audio Effect Graduate School of Culture Technology - - PowerPoint PPT Presentation
Delay-based Audio Effect Graduate School of Culture Technology - - PowerPoint PPT Presentation
CTP 431 Music and Audio Computing Delay-based Audio Effect Graduate School of Culture Technology (GSCT) Juhan Nam 1 Introduction Types of delay-based audio effect Delay Chorus Flanger Reverberation 2 Perception of Time
SLIDE 2
SLIDE 3
Perception of Time Delay
§ The 30 Hz transition
– Given repeated click sound (e.g. impulse train):
- If the rate is less than 30Hz, they are perceived as discrete
events.
- As the rate is above 30 Hz, they are perceive as a tone
– Demo: http://auditoryneuroscience.com/?q=pitch/click_train
§ Feedback comb filter: y(n) = x(n) + r y(n-M)
– Models sound propagation and reflection with energy loss – If M < fs/30: generate a tone
- E.g. Karplus-strong model of tone generation
(https://en.wikipedia.org/wiki/Karplus%E2%80%93Strong_string_synthesis) – If M > fs/30: generate a looped delay
- E.g. Delay effect
3
SLIDE 4
Delay
§ Delay effect
– Generate repetitive loop delay – Feedback coefficient controls the amount of delayed input – Can be extended to stereo signals such that the delay output is “ping-ponged” between the left and right channels – The delay length is often synchronized with music tempo – The delayline is implemented as a “circular buffer”
4
+
x(n)
feedback
y(n)
Dry
+
Wet
Delay Line
SLIDE 5
Chorus
5
- Chorus effect
– Gives the illusion of mul?ple voices playing in unison – By summing detuned copies of the input – Low frequency oscillators are used to modulate the posi?on of output tops à This causes the pitch of the input (resampling!)
LFOs
x(n) y(n)
Dry
+ +
Wet
Delay Line
SLIDE 6
Flanger
6
- Flanger effect
– Originally generated by summing the output of two un-locked tape machines while varying their sync (used to be called “reel-flanging”) – Emulated by summing one sta?c tap and variable tap in the delay line
- Feed-forward combine filter where harmonic notches vary over
frequency. – LFO is oPen synchronized with music tempo
x(n)
+
LFOs Sta?c tap Variable tap
y(n)
+
Wet Dry
Delay Line
SLIDE 7
Reverberation
§ Natural acoustic phenomenon that occurs when sound sources are played in a room
– Thousands of echoes are generated as sound sources are reflected against wall, ceiling and floors – Reflected sounds are delayed, attenuated and low-pass filtered: high-frequency component decay faster – The patterns of myriads of echoes are determined by the volume and geometry of room and materials on the surfaces
7
Sound Source Listener Direct sound Reflected sound
SLIDE 8
Reverberation
§ Room reverberation is characterized by its impulse response (IR)
– E.g. when a balloon pop is used as a sound source
8
10 20 30 40 50 60 70 80 90 100
- 0.4
- 0.2
0.2 0.4 0.6 0.8 1 CCRMA Lobby Impulse Response time - milliseconds response amplitude direct path early reflections late-field reverberation
- The room IR is composed of
three parts
– Direct path – Early reflec?ons – Late-field reverbera?on: high echo density
- RT60
– The ?me that it takes the reverbera?on to decay by 60 dB from its peak amplitude
SLIDE 9
Artificial Reverberation
§ Mechanical reverb
– Use metal plate and spring – Plate reverb: https://www.youtube.com/watch?v=XJ5OFpvX5Vs
§ Delayline-based reverb
– Early reflections: feed-forward delayline – Late-field reverb: allpass/comb filter, feedback delay networks (FDN) – “Programmable” reverberation
§ Convolution reverb
– Measure the impulse response of a room – Do convolution input with the measured IR
9
SLIDE 10
Delay-based Reverb
10
Z-M
+
x(n)
_
+
y(n) AllPass filter / Comb filter (when one tap is absent)
- The lengths of delaylines are chosen
such that their greatest common factors is small (e.g. prime numbers)
- The mixing matrix is chosen to be
unitary (orthonormal)
+
x(n) Feedback Delay Networks Z-M1 Z-M2 Z-M3
+
a11 a12 a13 a11 a12 a13 a11 a12 a13 y(n)
- A reverb is constructed by cascading
mul?ple AP or FFCF units
SLIDE 11
Convolution Reverb
§ Measuring impulse responses
– If the input is a unit impulse, SNR is low – Instead, we use specially designed input signals
- Golay code, allpass chirp or sine sweep: their magnitude
responses are all flat but the signals are spread over time – The impulse response is obtained using its inverse signal or inverse discrete Fourier transform
11
s(t)
LTI system
r(t)
test sequence measured response
n(t) h(t)
measurement noise
r(t) = s(t) ∗ h(t) + n(t),
SLIDE 12
Convolution Reverb
12
500 1000 1500
- 0.5
0.5 sine sweep, s(t) amplitude frequency - kHz sine sweep spectrogram 200 400 600 800 1000 5 10 500 1000 1500 2000
- 1
- 0.5
0.5 1 sine sweep response, r(t) time - milliseconds amplitude time - milliseconds frequency - kHz sine sweep response spectrogram 500 1000 1500 2000 5 10 100 200 300 400 500 600 700 800 900 1000
- 0.04
- 0.02
0.02 0.04 0.06 0.08 measured impulse response time - milliseconds amplitude