echo types, how to cancel echo in each type, which is more complex, echo cancellation implementation in matlab
prepared by : OLA MASHAQI ,, SUHAD MALAYSHE
echo types, how to cancel echo in each type, which is more complex, echo cancellation implementation in matlab
prepared by : OLA MASHAQI ,, SUHAD MALAYSHE
- Obtained the Fast Fourier Transform of signals.
- Designed and Validated Low Pass, High Pass, and Band Pass filters in compliance with the specifications.
- Produced and compared graphs of the results upon processing.
Digital signal processing through speech, hearing, and PythonMel Chua
Slides from PyCon 2013 tutorial reformatted for self-study. Code at https://github.com/mchua/pycon-sigproc, original description follows: Why do pianos sound different from guitars? How can we visualize how deafness affects a child's speech? These are signal processing questions, traditionally tackled only by upper-level engineering students with MATLAB and differential equations; we're going to do it with algebra and basic Python skills. Based on a signal processing class for audiology graduate students, taught by a deaf musician.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Impr...a3labdsp
In the recent years, hybrid reverberation algorithms have been widely explored aiming to reproduce the acoustic behavior of real environment at low computational load. On this basis, exploiting the advantages introduced from hybrid reverberation structures, a novel approach for the reproduction of moving listener position through impulse responses (IR) interpolation has been presented in this paper. In particular, the presented methodology allows to remove
redundant information in large IR database also decreasing the memory usage and the computational complexity required to perform the auralization operation. The effectiveness of the proposed approach has been proved taking into account a real IR database and also providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
Distance Coding And Performance Of The Mark 5 And St350 Soundfield Microphone...Bruce Wiggins
A paper presented at the Institute of Acoustics Reproduced Sound 25 conference in 2009 looking at the response of two SoundField Ambisonic microphones to sound sources at different distances from the microphone.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
The Southeast Regional Acoustics Consortium (SEAC) met in March 2012 at Florida International University bringing together academic institutions, federal and regional fisheries and environmental management agencies, and private industry that conduct active acoustics research in the coastal environments of the US from North Carolina to Texas and the US Caribbean. Informal presentations and discussions highlighted the latest tools for fisheries research, organized around high-priority research objectives and management drivers (e.g., stock assessment improvements, integrated ecosystem assessments) and HTI’s Pat Nealson conducted a presentation to help demystify FM Slide/Chirp signals in hydroacoustics for fisheries assessments.
Images may contain different types of noises. Removing noise from image is often the first step in image processing, and remains a challenging problem in spite of sophistication of recent research. This ppt presents an efficient image denoising scheme and their reconstruction based on Discrete Wavelet Transform (DWT) and Inverse Discrete Wavelet Transform (IDWT).
Research: Applying Various DSP-Related Techniques for Robust Recognition of A...Roman Atachiants
This paper approaches speaker recognition in a new way. A speaker recognition system has been realized that works on adult and child speakers, both male and female. Furthermore, the system employs text-dependent and text-independent algorithms, which makes robust speaker recognition possible in many applications. Single-speaker classication is achieved by age/sex pre-classication and is implemented using classic text-dependent techniques, as well as a novel technology for text-independent recognition. This new research uses Evolutionary Stable Strategies to model human speech and allows speaker recognition by analyzing just one vowel.
Noise reduction is the process of removing noise from a signal. In this project, two audio files are given: (1) speech.au and (2) noisy_speech.au. The first file contains the original speech signal and the second one contains the noisy version of the first signal. The objective of this project is to reduce the noise from the noisy file
Echo and reverberation effects are used extensively in the music industry. Here we will design a digital filter that will create the echo and reverb effect on audio signals.
- Obtained the Fast Fourier Transform of signals.
- Designed and Validated Low Pass, High Pass, and Band Pass filters in compliance with the specifications.
- Produced and compared graphs of the results upon processing.
Digital signal processing through speech, hearing, and PythonMel Chua
Slides from PyCon 2013 tutorial reformatted for self-study. Code at https://github.com/mchua/pycon-sigproc, original description follows: Why do pianos sound different from guitars? How can we visualize how deafness affects a child's speech? These are signal processing questions, traditionally tackled only by upper-level engineering students with MATLAB and differential equations; we're going to do it with algebra and basic Python skills. Based on a signal processing class for audiology graduate students, taught by a deaf musician.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Impr...a3labdsp
In the recent years, hybrid reverberation algorithms have been widely explored aiming to reproduce the acoustic behavior of real environment at low computational load. On this basis, exploiting the advantages introduced from hybrid reverberation structures, a novel approach for the reproduction of moving listener position through impulse responses (IR) interpolation has been presented in this paper. In particular, the presented methodology allows to remove
redundant information in large IR database also decreasing the memory usage and the computational complexity required to perform the auralization operation. The effectiveness of the proposed approach has been proved taking into account a real IR database and also providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
Distance Coding And Performance Of The Mark 5 And St350 Soundfield Microphone...Bruce Wiggins
A paper presented at the Institute of Acoustics Reproduced Sound 25 conference in 2009 looking at the response of two SoundField Ambisonic microphones to sound sources at different distances from the microphone.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
The Southeast Regional Acoustics Consortium (SEAC) met in March 2012 at Florida International University bringing together academic institutions, federal and regional fisheries and environmental management agencies, and private industry that conduct active acoustics research in the coastal environments of the US from North Carolina to Texas and the US Caribbean. Informal presentations and discussions highlighted the latest tools for fisheries research, organized around high-priority research objectives and management drivers (e.g., stock assessment improvements, integrated ecosystem assessments) and HTI’s Pat Nealson conducted a presentation to help demystify FM Slide/Chirp signals in hydroacoustics for fisheries assessments.
Images may contain different types of noises. Removing noise from image is often the first step in image processing, and remains a challenging problem in spite of sophistication of recent research. This ppt presents an efficient image denoising scheme and their reconstruction based on Discrete Wavelet Transform (DWT) and Inverse Discrete Wavelet Transform (IDWT).
Research: Applying Various DSP-Related Techniques for Robust Recognition of A...Roman Atachiants
This paper approaches speaker recognition in a new way. A speaker recognition system has been realized that works on adult and child speakers, both male and female. Furthermore, the system employs text-dependent and text-independent algorithms, which makes robust speaker recognition possible in many applications. Single-speaker classication is achieved by age/sex pre-classication and is implemented using classic text-dependent techniques, as well as a novel technology for text-independent recognition. This new research uses Evolutionary Stable Strategies to model human speech and allows speaker recognition by analyzing just one vowel.
Noise reduction is the process of removing noise from a signal. In this project, two audio files are given: (1) speech.au and (2) noisy_speech.au. The first file contains the original speech signal and the second one contains the noisy version of the first signal. The objective of this project is to reduce the noise from the noisy file
Echo and reverberation effects are used extensively in the music industry. Here we will design a digital filter that will create the echo and reverb effect on audio signals.
Howling occurs when there is an acoustic coupling between a microphone and a speaker, whereby already amplified sound finds its way back into the amplifier through the microphone repeatedly, by making a positive feedback loop that in a way amplifies itself.
This includes discussion of DSP applications such as two band digital crossover system,woofers, sqawkers, tweeters, interference cancellation in ECG, speech noise reduction, speech coding and compression, CD recording system
Embedded systems increasingly employ digital, analog and RF signals all of which are tightly synchronized in time. Debugging these systems is challenging in that one needs to measure a number of different signals in one or more domains (time, digital, frequency) and with tight time synchronization. This session will discuss how a digital oscilloscope can be used to effectively debug these systems, and some of the instrumentation considerations that go along with this.
This presentation includes the discussion of Digital Signal Processing applications such as two band digital corssover system, woofers, sqawkers, tweeters, interference cancellation in ECG, speech noise reduction using FIR/ IIR filters, speech coding and compression, CD recording system
4. Vent configuration + Hearing
aids On
Vent configuration + Hearing aids Off
BTE 0mm 1 mm 2mm 3mm 0mm 1mm 2 mm 3mm
ITE 0mm 1 mm 2mm 3mm 0mm 1mm 2 mm 3mm
OPEN No hearing aids, using KEMAR’s in ear microphones.
Measuring and analyzing 17 different scenarios,
5. A relationship between the input and the output
of a system.
input transfer function = output
6. input tf = output
A ) If input is then output is also =>
OUTPUT/INPUT = TF
Not stable
B) delta function stuff = stuff =>
What if stuff = input
Then output = input;
Therefore;
tf = input output
7. “A Maximum-Length Sequence (MLS) is a
periodic two-level signal of length P = 2^N – 1,
where N is an integer and P is the periodicity,
which yields the impulse response of a linear
system under circular convolution. The impulse
response is extracted by the deconvolution of
the system’s output when excited with an MLS
signal.”
http://www.commsp.ee.ic.ac.uk/~mrt102/project
s/mls/MLS%20Theory.pdf
9. We used a 17 period long MLS signal.
The signal was re-sampled by ADC/DAC to
24414 Hz.
Time Domain Frequency Domain
10. Transfer function of one’s sounds localization
system from a point in space.
Involves shape of the pinna, shoulders effect, hair
and more. Left
Right
HRTF_L
HRTF_R
Conv(Input (Mono) & Impulse Response(L,R)) = Output(L,R)
Input : Desired Sound
Impulse Response : HRTFalpha (L,R) => Desired Direction
Output : Sound interpreted from angle alpha
0 degrees
alpha
alpha
MLS + Chrip
signal
Time domain
alpha
….
Recorded Signal HRTF for angle alpha
3D Audio
Reconstruction
11. Interaural time difference - ITD
Interaural level difference - ILD
Spectral information
20. What impulse response do we want?
Loudspeaker?
Room?
Pre-Amplifier?
Coupler?
Microphone?
RM1?
NO
NO
NO
No
NO
NO
21. We want KEMAR’s ear response to source’s
location in a room regardless of,
Loudspeaker
Room
Pre-Amplifier
Coupler
Microphone
RM1
responses.
22. What we have,
Impulse Response
(Loudspeaker+room+coupler+microphone+pre-amp+RM1+hrir)
What we need,
hrir
Solution,
(Loudspeaker+room+coupler+microphone+pre-amp+RM1+hrir)
minus
(Loudspeaker+room+coupler+microphone+pre-amp+RM1)
= hrir
23. Procedure:
1st : Remove KEMAR and replace with a
measurement microphone similar to one’s in
KEMAR’s ear at the same exact location to get,
(Loudspeaker+room+coupler+microphone+pre-amp+RM1)
let’s call the combination of all these responses, ‘room’
response for now.
2nd: Compensate for “room” response if necessary.
24. How to compensate?
A) conv(room^-1, (room+hrir)) = hrir (1)
Same thing as
abs(FFT(room+hrir))/abs(FFT(room))
(Note: Once in frequency domain use linear convolution length for the number of FFT
points to avoid time aliasing).
Problem:
Ill-condition frequency bins that introduces sever spectral coloration to results.
(1) Project MaRIE. "CRTools Compatible HRIR." GN Resound
26. How to compensate?
B) Constant Regularization (1)
abs(FFT(room+hrir))/abs(FFT(room)+β)
Avoid spectral coloration at the cost of losing some room
compensation.
(1) Choueiri . Optimal Crosstalk Cancellation for Binaural Audio with Two
Loudspeakers. BACCH Audio. Princeton University.
27. Room Response
Inverse Room Response
The magnitude at the ill-conditioned frequency
dropped by 10 times (20 dB).
Ill-condition
frequency bin
shifted
28. How to compensate?
C) Frequency Dependent Regularization (1)
abs(FFT(room+hrir))/abs(FFT(room)+β(frequency))
Avoid spectral coloration at the cost of losing a smaller amount
of room compensation.
0 if Room(i) > threshold, for i= 1 : #fftpoints /2
β(frequency) =
β if Room(i) < threshold, for i= 1 : #fftpoints /2
(1) Choueiri . Optimal Crosstalk Cancellation for Binaural Audio with Two
Loudspeakers. BACCH Audio. Princeton University.
29. Room Response
Inverse Room Response
Only the bins below the threshold are boosted.
Getting some of the room compensation back
from constant regularization.
30. How to compensate?
D) Filter Inversion
Example: Looking at room response as a high pass filter.
=> Flat out the response and deal with phase separately.
31. 1- Fit a curve to the room response to avoid compensating for
FFT artifacts.
We only want to partially compensate for the shape of the filter without
introducing new artifacts to the system.
32. 2- Find the maximum value of the obtained curve and boost all
frequency bins to that value.
room response is boosted appropriately and resulted in a flat frequency
response.
33. As expected, low frequency bins are boosted the most to
compensate for the room response.
This method is basically, a variation of frequency dependent
regularization where the threshed is defined backward as the
max(FFT(room)).
34. 3- Compensating for phase.
Phase of the acquired signal before compensation,
Phase of the compensated signal,
Note: Phase is conjugate symmetric.
DC and middle frequency bin must be
ignored when reconstructing the new
hrir response.
36. Goal:
Does the impulse response corresponds to right
location?
How:
By comparing,
both perceptually and calculating the quantization error
between
the resulting signals.
37. KEMAR is 145 cm from loudspeaker, 5 cm
elevation.
HRTF angle ~ 60 degrees in Front Left
azimuth and ~ 5 degree elevation on top.
Sound pressure level at left ear ~ 75-80 dB
0 degree
60 degrees
Left
Right
38. Only the magnitude of the HRTFs
were compensated, since the phase
response of the room is linear.
Room Response
Original Binaural
Reconstructed with HRTF
+ Room Compensation
Reconstructed with HRTF
Perceptually:
39. Difference:
After synchronizing and normalizing the gain,
we
have,Difference Reconstructed L/R Compensated L/R
Binaural Left 20.38% 5.2%
Binaural Right 33.03% 15.55%
FYI:
Subtracting the two STFT at each frequency bin, taking the average over each
frequency bin and find the norm of the resulted vector => Difference.
42. MIT Media Lab - 1994
Sampling rate 44.1 kHz
Azimuth : 0 to 355 ~ 5 degrees step
Elevation : -45 to 90 degrees ~ 15 degrees
step
1024 samples
43. University of California Davis -2001
Sampling rate 44.1 kHz
Azimuth : -80 to 80 ~ 5 degrees step
Elevation : -45 to 270 degrees ~ 15 degrees
step
200 samples
44. GN Resound at Glenview, Illinois - 2014
Sampling rate 48828 kHz
Azimuth : 0 to 355 = 5 degrees step
Elevation : -30 to 90 degrees = 10 degrees
step
160 samples
45. Better Ear Strategy: Compare SNR of signal source
between ears and choose the signal with most positive
SNR. For polar plot, choose the signal between ears that
has the most attenuation in reference to the on-axis signal.
Audibility Strategy: Compare levels of signal source
between ears and choose signal with the most positive
level. For polar plot, choose the signal between ears that
has the least attenuation in reference to the on-axis signal.
(1) Andrew Dittberner, Chang Ma,, and Paul Sexton. "BASS Benchmark Project.“ Labyrinth
Program. GN Resound, 2014.
53. Why Calibration?
Is KEMAR facing the speaker at the (0az,0el)?
Is the robotic arm keep the same azimuth angle
as it goes to higher elevation angles?
There are different ways to operate calibration on
the system.
54. 1- Set the speaker and the KEMAR at (90az, 0 el) by
eyeballing it (or maybe use a level).
2- Define a reasonable azimuth and elevation threshold
for the eyeballing error, e.g. ±15 degree azimuth and
elevation.
3- Record the response at both ears for every point within
the threshold. The step size will define the resolution of
your calibration.
4- Take the RMS of the result from each ear and subtract
them in the log domain.
5- The maximum value from step 4 will correspond to the
actual (90az, 0 el).
6- Move the motor to the corresponding angle from step 5
and set it to (90az, 0 el).
55. th = threshold
Threshold on azimuth and elevation are arbitrary and could
be different values.
-
10*log10(rms
10*log10(rms
10*log10(rms
56. Verification: Left and Right RMS intersection
Azimuth Angles
Magnitude
dB
The intersection is @25 degrees.
57. KEMAR without its pinna on could form a
directional microphone. Auditorium calibration
might be more accurate if the KEMAR’s pinna are
removed( For higher frequency bands).
A similar procedure must also be done for
(0az,90el). It’s expected that the level difference
between left and right should be zero since they’re
symmetric (we’re looking for the minimum value
here).
A similar auditorium calibration procedure can be
done by focusing on the ITD instead of ILD. The
index at which Left and Right impulse response are
extracted must be the same at ( 0az, 0el).
58. When these two peaks are
at the same index,
KEMAR is located 0az, 0
el)
60. The auditorium calibration assumes that the robotic arm
would move on a straight line (keeping the same azimuth
angle with respect to the KEMAR) toward higher elevation
angles. Turned out that’s not the case here.
61.
62. The laser pointer follows the center of the KEMAR
head at all elevation angles.
63. KEMAR receive reflection of the signal of the
walls and other items while receiving the
same signal from the speaker.
64. Where is it coming from?
It varies from 100~150
samples form the peak for
most azimuth angles. Given
the sampling rate this
translate to 70~100 cm
from the KEMAR.
Solution:
Use a measurement microphone as the second ear and place it in different position.
Warmer or colder?
66. The robotic arms were the main origins of
the early reflections in the systems.
A few acoustic sound absorption foams
on each arm decreased the reflection by
almost 40 dB.
67. Early reflection becomes more important
when using Hearing Aids, since they got
longer impulse responses. It’s harder to
detect the early reflection.
85. Collecting data for on elevation at 5 degree azimuth takes
about 923 seconds ~ 15 min, that’s about 4.5 hours to
collect data for all elevation at 10 degree resolution.
Not a good idea to save your database to Matlab memory
while measuring, no matter how more convenient it is!
Matlab WILL CRAHS!
86. Interpolation in time/Frequency domain
ITD/ILD
Reverberation Time
3D Audio
Beamforming for Source Localization
CIPIC Polar Pattern
CIPIC Delay and Sum Pattern
MIT Polar Pattern
MIT Delay and Sum Pattern
And more glitzy plots!
87. Why interpolation?
1. Higher Resolution and easier for analysis.
2. Creating a smoother transition for reconstructing 3D audio.
3. Easier to compare different HRTF databases of different
step sizes.
93. How? Subtracting the magnitude squred of
the HRTF at left ear from the right ear.
94.
95.
96. The time it takes for a signal to drop by 60 dB. In a
noisy environment, T60 is measured by
interpolating the linear region in the Energy Decay
Curve.
Where h(tau) is the room impulse response.
https://ccrma.stanford.edu/~jos/pasp/Energy_Decay_Curve.html
97.
98.
99. Goal : Make a sound moving smoothly
through all the angles in the database.
Help us identify the accuracy of the database and the possible
spectral coloration of the database on any desired signal.
3D Audio for 0 degree elevation:
CIPIC -80 to 80 MIT 0 to 355 CIPIC -80 to 80 MIT 0 to 355
102. How does Spatial Aliasing affect us in
localizing a sound source?
Assuming that the only cue we could use to localize a
source is the ITD cue (like wearing a hearing aid?).
103. We could use a set of techniques, called
Beamforming, to simulate localizing a sound
source.
The idea is…
Delay the reference signal until the
sum of the energy of the two signals
is at its maximum. That delay would
corresponds to the angle of arrival.
104. • Signals at 1 kHZ forming
a sound source at 20
degrees at sampling rate 5
kHz.
•Angle of arrival
detected correctly
•Angle of arrival
detected in-correctly
• Signals at 2.5 kHZ forming
a sound source at 20
degrees at sampling rate 5
kHz.
Beam pattern
105.
106.
107.
108.
109. CIPIC database
-80 to 80 azimuth
@ 0 elevation
MIT database
0 to 355 azimuth
@ 0 elevation
Both from DC to 22 k HZ, shown for Left ear.
-80 to 80 azimuth
110. ITD: stronger at low frequencies.
ILD: stronger at high frequencies.
Shorter wavelength, die faster, bigger ILD.
111. Distance between ear ~ 23 cm
23 cm = 343 m/s * t => t ~ 670µs
Frequency = 1/period =>
F ~ 1600 Hz
What does that mean?
Bigger topics, brief introduction, no need to introduce linear system, go from impulse MLS sequence, then HRTF,
Purpose . Spatial perception .
Plug hearing aids/ changing TF, simulate another polar plots distorted and undistorted. Goal
Directionality might be affected by the hearing aids.
appendix
So, how do we get a delta function? Maybe popping balloons? Won’t be accurate and won’t give you the response for every frequency.
Same energy at all frequency bins.
Even though the signal is not entirely binary anymore, the spectrum is still flat, though noisy, over the desired frequency range, so we’re safe.
We’ll use method B to measure the impulse response( steering vectors) of one’s sound localization system. In order to get the delta effect, we can use MLS or chirp signals, which is a psudue random noise.
Localization cues. ITD better for < 1600 Hz, headshadow effect comes to importance > 1600 ILD and nervous system not very good at ITD at low frequencies, and spatial aliasing, forming beams for > 1600 Hz.
Too many, octave frequency 1 2 4 8 kHZ
Produce by : Delay and sum beamforming
Super low frequencies: omni response.
Narrow beam at zero degrees => good
But high energy at all these other angles, cannot distinguish the difference.
Of course we almost never hear one frequency, but a wide range of frequencies.
So , conclusion here is that there is more to sound localaization besides ITD.
Assumptions can be made here. Collect references, do they compensate? Papers associated with databases.
Let’s assume loudspeakers and all are all room response, that they have flat response. Room response if different at. Just talk about it, w ref
Reference. Need papers. Cited.
Based on the room and equipment, there might be few to many ill-conditioned frequency bins. These are the bins that would boost the signal to higher values and colorate the spectral of the
Original signal.
Too much detailed.
Source. Refer to the same database, different ways to compensate. Be consistent.
Fix the peak.
Room = fft(room);
Room = fft(room);
Room = fft(room);
Room = fft(room);
Room = fft(room);
Solution is to simply subtract the phases from each other, so the phase of the signal won’t be affected by the room. If room phase is linear, this process is unnecessary.
References. Citations.
My computation is right, then no need to room compensation? Verify a few point. Quantization noise, veify the process
My computation is right, then no need to room compensation? Verify a few point. Quantization noise, verify the process
Keep in mind, binaural sound always sound better than reconstructed and no one knows why yet!
Open ear HRTF
Pick one database.
Better ear strategy analogous to beamfroming pattern and audibility to omni pattern
Interpolation error based on the resolution at 355 to 5 degrees.
Do in frequency domain
At azimuth -80 degrees
level differences over elevation angles as we walk through different elevation angles. The time delay between each elevation is neglibgible.
At elevation -45 degrees. As expected, time delay is more important over azimuth angles, and level differences is more important over elevation angles.
Similar to microphone array and MIT results
Low frequency localization for about 300 Hz to 1500 Hz pretty good at zero degree elevation.
Similar, ITD doesn’t change over elevation angles with frequency. Huh! So, we can use ITD cues the same way at any elevation planes. It only depends on how big your head is! We can make the same beams toward the azimuth angles using ITD.
As expected, ILD is higher for F > 1500, but it’s divided to two ranges, 2500 to 6500 and 8500 to 11000 Hz.
As expected, ILD is higher at extreme angles, -80 and +80.
0 degrees is not very good, cone of confusion.
Front bottom : two ranges
Front 0 : the low range
Front up : low range wider
Top : low range wider and weaker
Smiley face!
Back Top: high range not very strong
Back 0: high and low both
Back bottom: high
Appenidx
They also look like happy, annoyed and angry faces!
Appendix.
Appendix.
You be the judge of their angles .
0 degrees front-
-80 degrees left same as 355
80 degrees right same as 0 degrees
Appendix
Appendix
appendix
appendix
appendix
-80 to 80 CIPIC for 0 elevation for selected frequency, azimuth to beam to spatial aliasing.
Polar pattern.
Put both polar plot.
-80 to 80 front side for selected elevation angle. Plot left and right. See headshadow effect.
What is the assumption? Beamforming + headshadow effect. Cut off 8 kHz.
Same data sphere plot.
All azimuth angles, limited elevation angles.
Goes from 0 to 22 kHZ
-80 left to 355
80 right to 0
MIT 355 t0 180 is like -80 to 80 in CIPIC
Fft(hrir)
fixed. Brain and detection. Remove
ITD good for F<1600, more than the wavelength is smaller than the distance between the ears, we have
Spatial aliasing.
High frequency : reflection problem
Interpolation error based on the resolution at 355 to 5 degrees.
Interpolation error based on the resolution at 355 to 5 degrees.
Better ear : min intersection
Audibility: max intersection