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Srinath Srinivasan
India
Registrace 25. 05. 2016
Welcome to my Channel! I am Srinath Srinivasan. The goal of this channel is to educate everyone about various physical phenomenon that we experience in everyday life. Predominantly this channel has a strong focus on explaining various phenomenon and concepts of Sound and its applications as audio and acoustics in a clear, simple and intuitive way! I use physical visual aids for better understanding and make learning simple. Currently I'm using presentation style which make learning and grasping concepts more interesting and immersive.
Other areas that I have explored in the past include DIY (Do It Yourself) Arduino projects, creative sound effects (SFX), slow motion effects and Augmented Reality (AR).
Other areas that I have explored in the past include DIY (Do It Yourself) Arduino projects, creative sound effects (SFX), slow motion effects and Augmented Reality (AR).
Overlap in digital signal processing
Overlap is the amount of time data used within a time signal by two neighboring Fast Fourier Transforms (FFT). Overlap is expressed in percentage. While windowing a signal ensures that there is no spectral leakage, some amount of information is lost due to the windowing process. Overlapping is one way to overcome the limitations of windowing. Learn about the process of overlapping a time signal in this video.
Other relevant videos:
Windowing explained:
czcams.com/video/tgH-a-vaiq8/video.html
Types of windowing explained:
czcams.com/video/JZRTJRTnYNU/video.html
Other relevant videos:
Windowing explained:
czcams.com/video/tgH-a-vaiq8/video.html
Types of windowing explained:
czcams.com/video/JZRTJRTnYNU/video.html
zhlédnutí: 48
Video
Sound Diffraction explained
zhlédnutí 31Před měsícem
Sound Diffraction is the bending of sound waves around small objects or obstacles. Sound diffraction is the reason why we are able to hear something although we cannot see it. Sound diffraction depends on the wavelength of the wave as well as the size of obstacle or opening. Learn about sound diffraction in this video and cases where one does experience it in everyday life.
Triggering acoustic measurements
zhlédnutí 14Před 2 měsíci
What is Triggering in Acoustic measurements? A trigger is often required to start recording at a given time or once a threshold has been crossed. A trigger would thus initiate in starting a recording once the defined condition has been satisfied. Learn more about triggering and its applications in this video.
Wavelet Transform examples
zhlédnutí 126Před 3 měsíci
The Wavelet Transform is a type of Time-frequency analysis. In Wavelet Transform, a multi resolution windowing is used to capture signals with good time and frequency resolution. In this video three short signals are analyzed using the Wavelet Transform. To learn about the basics of The Wavelet Transform: czcams.com/video/DWYzroBudTw/video.html
Tolerance curves in acoustics
zhlédnutí 31Před 4 měsíci
In order to evaluate the acoustic performance of a product, tolerance curves are used. Tolerance curves help to assess whether a product is within acceptable limit or is violating the limit. Learn more about Tolerance curves and their types used in acoustics in this video.
Real Time Analyzer (RTA) explained
zhlédnutí 227Před 5 měsíci
A Real time analyzer (RTA) is an audio device that measures and displays the frequency spectrum of an audio signal in “real time”. There are two types, Analog RTA and Digital RTA. Analog RTA uses hardware whereas Digital RTA uses Digital Signal Processing (DSP) technology. Learn more about RTA in this video.
Playback speed
zhlédnutí 27Před 7 měsíci
Playback speed offers the ability to either speed up or slow down a given time data recording. Playback speed however does not preserve the pitch of the recording. The pitch changes based on the speed. Playing back a rapid acoustic event at low speeds is sometimes insightful. Learn more about Playback speed in this video.
Low background noise
zhlédnutí 50Před 7 měsíci
Background Noise or Ambient Noise is any sound present in the space other than the one that is being monitored. Background Noise levels are measured to provide a reference point for analyzing a sound source in the environment. Can low background noise affect acoustic measurements? Learn more about Low background noise in this video. Background Noise: czcams.com/video/WxQXu-SuMv8/video.html Loga...
Frequency Response Function (FRF) explained
zhlédnutí 3,1KPřed 8 měsíci
A Frequency Response Function (FRF) is a function used to quantify the response of a system to an excitation, normalized by the magnitude of this excitation, in the frequency domain. FRF is a frequency based measurement function. It is used to identify the resonant frequencies, damping and mode shapes of a physical structure. Learn more about FRF in this video. Mode Shapes Playlist: czcams.com/...
Time-Frequency resolution explained
zhlédnutí 1,2KPřed 9 měsíci
Microphones and Accelerometers are the sensors which capture Sound and Vibration as analog signals. These analog signals are discretized by the computer i.e. they are broken into discrete samples and stored as Digital signals. Time Domain represents how a signal changes with respect to time. Frequency Domain represents how much of the signal lies within a particular band of frequencies. There i...
Aliasing in Digital Signal Processing
zhlédnutí 668Před 9 měsíci
Aliasing is an irreversible signal distortion in the sampling process resulting from failure to meet the assumption of the sampling theorem. The detected erroneous signal is thus an alias of the original signal. By processing the acquired signal at the right sampling frequency, it is possible to prevent aliasing. Learn about this in detail in this video. Low Pass Filter explained: czcams.com/vi...
Acoustic A-weighting explained
zhlédnutí 874Před 10 měsíci
Acoustic A-weighting is a commonly used family of curves relating to the measurement of Sound Pressure Level. A-weighting is applied to measured sound levels in an effort to account for the relative loudness perceived by the human ear. The unit of A-weighted Sound Pressure Level is expressed in dB(A). Learn more about A-weighting in this video. A-weighting conversion: srinath96.blogspot.com/202...
Playback filters
zhlédnutí 44Před 11 měsíci
Audio Filters amplify or attenuate a particular frequency or band of frequencies. Playback Audio Filters provide the ability to listen to audio with filters applied. Playback is useful as it gives immediate feedback whether a particular filter is helpful or not. Learn more about playback filters in detail in this video. Audio filters playlist: czcams.com/play/PLTkzvdFCLGkilsqVOXtlmYJOlLxtj1Aib....
Time data corruption
zhlédnutí 54Před rokem
Time data or time recording consists of Sound Pressure with respect to time. Pressure and time form the basis of all sound waves we hear. A proper time data is the one which accurately represents the sound field. Presence of any additional unwanted information in the recording would result in “corruption”. Learn more about different ways in which time data can get corrupted in this video. Backg...
the image was helpful to understand
This is a fantastic introductory video to windowing!
Thanks that was the best and the simplest
number code?
Thank you for the lesson Sir. I am actually working on the dynamic analysis of a reinforced concrete bridge structure. But when I carried out numerical modal analysis considering 30 modes I realized the sum of my mass participation ratio for each mode is not even up to 80% of total mass, highest was at 77%. My main problem is: about the first 20 mode shapes had really small mass participation and instead of the first modes contributing a lot to the mass, I had the last modes contributing instead (to have up to 77% mass participation). From my knowledge of modal analysis, the first modes are usually the most important and the ones contributing a lot, but that's not the case with my analysis results. What could possibly be the problem Sir? (I assigned boundary conditions to both ends of the bridge)
You're welcome. As I am not from Structural engineering background but rather Mechanical, I cannot guide you accurately but will try to answer generally. If higher are modes are dominant in your structure, it could be either that higher modes are really contributing to the overall deformation or the lower modes have been suppressed by choice of boundary conditions (ex if the structure became too stiff). For example, consider a plate with too many accelerometers mounted on it. This certainly increases the system stiffness to such an extent that the lower modes are unable to make an impact. So the important thing to note here is that accelerometers need to be mounted in such a way that it does not alter the physical properties of plate (do not add mass or increase stiffness). Higher modes usually have complex shapes and are more localized. In contrast, lower modes have simple shapes and are more globalized. Also the excitation frequency range is important. Usually broad band excitation is given so as to check how the system responds overall. Certain frequencies will get excited which indicates resonance. But if the excitation frequency range is limited to high frequencies then the lower frequencies were not excited in the first place. So ensure all possible frequencies of interest are excited. These are the possible scenarios I could think of.
@@SrinathSrinivasan Thank you very much for your response Sir. Will try to follow your advice and see if there are improvements
Well explained. Thank you Sir
كم تردد ذهب
Sounds like ambience to horror game
STUPID
Thank you!!!!
this is so helpful, accurate and easy to understand
Great video!
Trespassers be warned $ purge commences amen
Thank you for that explanation. No doubt it's a a difficult subject to understand and one even more difficult to explain.
Have no Idea if you Will read this coments but if yes that was an amading video. Please what is the bast configuration of the piramids and which material it is more apropriate for such porpouse thanks.
There are many parameters to consider while designing an Anechoic/Semi-anechoic chamber. The most important is the cut-off frequency of the chamber which depends on the chamber dimensions and the wedge height. The lower the cut-off frequency, the larger should be the room size and the wedge height. Wedges help achieve near free field condition not only by absorbing high frequencies but also by trapping the lower frequencies due to their tapered shape. The wedge material must be a good absorber (ex. fiberglass, foam) with 0.99 absorption coefficient over a defined frequency range.
The more important - meaningful - divisions are direct field and reverberant field. The point at which they change is called the critical distance. The difference between the fields are spatial. The direct sound comes straight from the instruments, the reverberant field comes from all around you. In the reproduction the direct field is reproduced by the direct field of the speakers. The reverberant field is reproduced by the reflected sound from behind and beside the speakers. The secret of good speaker design is the radiation pattern and speaker positioning to project a good image model of these fields, including a flat power response.
Thanks for uploading this. Can you please do an example of extracting mode shapes using an impact hammer on a structure?
I had done some examples of mode shapes in general. You can watch them here : czcams.com/play/PLTkzvdFCLGkjrGqSw_C5J7iduYqwkVcZj.html
@@SrinathSrinivasan Thanks!!
If the room you are in echoes, then wouldn't this alter the true frequency of your voice?
The problem here is not with echoes but rather reverberation. Reverberation occurs when multiple reflections of the same sound arrives at slightly different times at the microphone creating one prolonging sound. As the room had flat walls and tiled floors, there was more opportunity for sound waves to undergo reflections. A reverberant environment does cause sound to be unintelligible and can increase the perceived amplitude of sound. However, it does not alter the frequency content of the sound since the environment (room) is not producing any sound but only reflecting already created sound.
Did you use a logarithmic scale for the PSD axis?
Both PSD and frequency axis are in logarithmic scale. The graph therefore is a log-log graph.
Thank you so much for this video. Sincerely, your videos make signal processing so simple and interesting. Thank you Brother.
Great video thank you!
What do you mean by 100 elements in each plate?
For FEM (Finite Element Method) Simulation, one of the prerequisite is, the object of interest has to be discretized into finite particles called elements. In this video, the plate was divided into 100 elements. Larger the number of elements, better the results but also greater is the computational cost.
Thank you sir, perfect explanation👍🏻
Is reverberant field and diffuse field the same thing?
Yes.
@@SrinathSrinivasan thanks
Very helpful thank you❤❤
FFT is an algorithm for DFT.
Can you help me??
Good job 👍💯. I have generated wavelets with the theory of trigonometric partition equations we can create wavelets. czcams.com/video/p0Zc9onKQ0Q/video.htmlsi=dX1K0xLtJ2iWSgf5 czcams.com/video/3Ebvypj577E/video.htmlsi=zeLyrZc54430eV5b czcams.com/video/DV1iJV0oa7Y/video.htmlsi=798Te0vetj90Q7j1
thank you
thank you so much, beautifully explained!!
Thanks ! What would be the difference if we replace our wavelet with bandpass filter ? Would the latency be the same ?
Octave analysis is performed by passing a signal through many bandpass filters. While this is good for longer sounds, it may not be the best approach when it comes to analyzing short sounds where both time and frequency information are important. If we use FFT/Octave approach there are some things to consider. First the frequency resolution is fixed because the duration of signal is very low (ex. Door close sound). Higher frequency resolution can be obtained only when duration of signal is longer. To learn more about time-frequency relation check the link in video description. Second, this approach uses fixed frequency resolution over the entire spectrum, which also means that time resolution is fixed. The wavelet analysis overcomes this by using multi-resolution windowing so there is no further dependency on time and frequency. Hence one can achieve good time and frequency resolution at the same time.
@@SrinathSrinivasan Very thanks for your answer ! But in fact my question was not what i intended in my mind... I was thinking about iir bandpass, (Generally i use original signal minus 2th allpass.) From what i understand, wavelet use fir ? But what is the difference if we replace them with those bandpass iir filter ? would the latency be worse or the same ?
Yes, a wavelet can be considered as an FIR filter as it has a finite impulse response in the time domain without recursive components. Now merely comparing FIR and IIR with respect to latency, IIR has less latency as you probably know that IIR filter being recursive in nature uses past outputs as input resulting in fewer calculations compared to FIR which is non-recursive. The less latency of an IIR comes at an expense of phase issues. These can be rectified by using All pass filters but if higher order All pass are required then again it means more computation which can increase latency.
@@SrinathSrinivasan Thanks !! I'm just testing my first FIR wavelet at 110 hz with 4096 tap. For the phase this is exact but note that at the exact frequency of interest, the phase could be exact, yet it degrade around that frequency. (depending of the type of bandpass, as a combination of lp and hp will maybe not do this ) I just test to see the difference, with only one bandpass (that is original signal minus a 2th allpass) the phase doesn't degrade so much around the frequency, but we have another problem, the envelope of the signal is extremely smoothed as 5X longer than the original, where the Fir have approximately a 1.7X longer envelope. Now i could use more bandpass filter in series with lower coefficient to get a less degraded envelope, But now, the phase is degraded even more around the frequency.. To get almost the same envelope elongation with IIR, i need to use 8 bandpass in series. But now the latency is even greater than with the FIR !... Well, this is a little approximate because i have some tool to visually see if the filter have the same response but it's not ultra precise, also, the IIR have a less sharp attenuation... But it give me an idea, using a IIR filter that degrade the envelope but have less latency and try to stop the detection with a FIR or more filter in series. But maybe i did'nt use the right windows for the wavelet, here a Hamming windows ?
explain more on wavelets
Jeez this just made my day. It was so easy to easy to understand this way. You just want a periodic signal. That is the whole goal, with windowing that FTT and bam you have your awnser :0
Hello, I really like your video very much and the explanation helped a lot. However, when you showed the window at around 6:42, the window function seemed to be misplaced on the axis. I wonder if I am seeing it right :)
Thank you. The horizontal axis for window function should rather be lower, please ignore it.
Rajwant Sir OP 🔥🚩
Awesome video, unless I missed it, is there any specific reason why many manufacturers exclude well number 0? Esthetically they'll look symmetrical but wondering if there's a benefit in terms of sound. Maybe well number zero is not that influent? Thanks in advance!
The well number 0 is present in QRD of any order. When installing QRD assuming N7, it is recommended to use arrays of N7 as using only one will not give meaningful results. So either one can leave the well '0' on one side or split the well '0' evenly along the ends. The latter option ensures that one does not need to worry about symmetry during final assembly. I do not know which manufacturer excludes well '0' altogether but there is a well known QRD manufacturer - RPG Acoustical systems. Their QRD model QRD-734 as mentioned in their website offers well '0' split along both ends. Out of curiosity I used QRDude software to check if there is any difference in keeping well '0' as it is or splitting it along ends and there was no observable difference. You can check the findings in the link in the video description now. Finally my thoughts, even if a manufacturer were to neglect well'0' altogether, they still have to use end-fins on both sides to cover the diffuser sides. In that process, they have just added an approximate well '0' on both sides since the end-fin will have some finite thickness.
Oh thank you very much for your detailed explanation! Now it makes sense... It was my mistake, never noticed that most likely those looking "simmetric" in fact have larger sides to compensate... I prefer the simmetric design, unless is equally good!
fantastic video, better explanation provided than my prof. did. Bro is professional
Extremely informative! Thank you 😊
❤
Hello Srinath, awesome video, indeed!! Thanks. Just had a quick question. At 11.58min of your video, when you are describing the effect of increasing the sampling rate, 1) you are increasing the sampling rate from 2500Hz to 25000Hz. BUT 2) you are using the SAME signal (right?), that you previously sampled at a rate of 2500Hz. IF SO, then how come the bandwidth of the signal got changed (!). Let's say, I have a sinusoidal signal of frequency 5Hz. Does the bandwidth of the SIGNAL change When I sample it at 10 Hz and at 100Hz, respectively? Would be great if you could clarify this confusion!! Thanks in advance.
Sampling rate (in the time domain) is how many samples are being acquired in a given time. In this video, I considered two cases wherein the signal was being acquired with different sampling rates (2500Hz and 25000Hz). Now sampling rate has a consequence in the frequency domain which is bandwidth. One has to sample the signal at least 2.5 times than the highest frequency of interest. Hence in acoustic recordings, as our frequencies of interest lies within the human hearing range (20Hz - 20KHz), we usually sample at 48KHz. In other words a signal sampled at 48KHz would result in a bandwidth of up to 20KHz in the frequency domain. If the signal was only sampled at 24KHz then the bandwidth is now reduced to around 12KHz because the current sampling frequency is too slow to accurately sample the fast oscillating high frequency waves. To answer the other question: If your sine wave of 5Hz is sampled at 10Hz and 100Hz, the bandwidth in frequency domain is different in each case although you will only see 5Hz result in the spectrum. If you had an additional sine wave of say 20Hz then you cannot sample it with 10Hz as it is less than 20Hz, it will fail to capture it.
Hows this any different from a transfer function??
A transfer function is basically output/input. FRF is a measure of how the transfer function varies with respect to frequency.
Very nice
Rj sir in lakshya jee2024 batch PW
Great. Thanks for your efforts.
Hi, Are you going to produce videos in more advance wavelet analysis? I can not link the idea of the wavelet analysis with the filtering in discrete wavelet analysis. Thank you.
Thank you so much for this series
Hey Srinath. Thanks for the video. I'm a doctoral student at CalArts doing research on using megaphones as musical instruments. I would love to look at your charts and diagrams in higher definition. Would you be willing to share them? Thanks for considering!
Who Else came from Rajwant's Sir Communication Lecture ?
me
Me
Thanks lot a sir for explaining the noise level with very minute basic concept
legend