Questions tagged [sound]

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Measuring Amplitude of sound wave FFT

Please, pardon my ignorance. This may be a simple question, but I am still unable to explain what is happening. I am using a turtlebot to listen to sound emitted from a stationary sound source. My ...
elcymon's user avatar
  • 103
0 votes
1 answer
335 views

Uncertainty Relation between time and frequency

Typically, humans can hear sound waves in the frequency range 20 Hz to 20 kHz. If one wants to make digital record of sound such that no audible information is lost, what is the longest time interval ...
Quarkonia's user avatar
  • 103
2 votes
2 answers
299 views

Pitch perception - Why does our ear not directly detect the missing fundamental frequency?

My motivation for this question is related to biological phenomenon of our inner ear, so I initially posted in the biology community, but I suspect the answer has a DSP explanation, so I figured this ...
Kevin Wang's user avatar
-1 votes
2 answers
3k views

How can I calculate Reverberation Time (RT60) from the Room Impulse Response using Python?

I have the following Room Impulse Response(RIR) in wave format here. I wanted to calculate Reverberation Time (RT60) of it by plotting its Energy Decay Curve using this Python code. The RIR is ...
Sajil C K's user avatar
  • 149
0 votes
1 answer
206 views

is there a way to apply a median filter diagonally?

For example, some studies show how to smooth a spectrogram with a 2D median filter vertically and horizontally (along time or frequency). My question is how would I apply 2D median filter along the ...
Dan's user avatar
  • 70
0 votes
1 answer
639 views

How does it go about implementing pitch shifting without affecting the duration or speed on matlab? [closed]

I am creating a project that will record a sound sample from the world, for example, a bird's chirping. And then through pitch scaling, I could create different notes. I first tried to change the ...
Yihan Hu's user avatar
4 votes
1 answer
763 views

Algorithm for identifying beat grid in music with non-constant tempo

Are there any algorithms for automatically fitting a beat grid to a song (tempo matching) when the song has a slowly changing tempo or BPM? The output would be a vector of onsets. Here's a picture ...
R Tyler McLaughlin's user avatar
0 votes
0 answers
33 views

When we play several audio streams what is responsible for mixing them?

When we play several audio streams what is responsible for mixing them in single audio playing via speakers? Clearly it should be either OS combining the sounds or hardware getting multiple data ...
Eugene D. Gubenkov's user avatar
0 votes
1 answer
51 views

How to model airborne sound channel

I'm making a soft demodulation/decoding to communicate a microphone and a speaker in short range. I have to calculate the LLR using my constellation and the airborne sound channel probability density ...
Eligio Mariño's user avatar
4 votes
2 answers
14k views

Mathematical equation for the sound wave that a piano makes

Is there a mathematical function that can output the sound that a piano makes at different frequencies? So if the simplest equation for a sound wave (at a given frequency and for a given sample rate) ...
Murey Tasroc's user avatar
2 votes
1 answer
97 views

How exactly does sound get encoded by a cochlear implant processor?

I've been wanting to know more about how sound is processed by a cochlear implant processor, how they are programmed, and how they stimulate the magnets and electrodes inside. How do you take pure ...
HeavenlyHarmony's user avatar
2 votes
0 answers
816 views

PortAudio: sound fade in/out by scaling sample data

I've done already some fade in or fade out effect in portaudio, but that is not working properly. My callback function: ...
Lizard_Paszczyk's user avatar
0 votes
2 answers
140 views

recognize unwanted sounds, noises, and deleting them algorithm

I have a function that I need to use denoise algorithm on. What I thought is to divide it to smaller parts and then use Fourier Transform on each one of the parts, but I am not sure what to do next. ...
Holo's user avatar
  • 65
4 votes
1 answer
1k views

Polyphase Filter decomposition. It is not working

So, I want to do a polyphase implementation of a filter bank (I had problems with it since the beginning and I already asked for help to the community in the past, indeed: Norm MPEG-1 Layer III (Mp3) ...
f.gallardo's user avatar
12 votes
3 answers
5k views

Transmit data through sound between 2 computers (very close distance)

I'm writing an example on transmitting data through sound betwwen 2 computers. Some requirements: The distance is very close , i.e the 2 computers are basically adjacent to each other Very little ...
Dang Manh Truong's user avatar
1 vote
2 answers
155 views

Custom ringtone sounds good on computer, can't be heard on phone [closed]

I did a ringtone using Audacity and then used this website to convert to mp3. I sent this mp3 file to a friend who said he couldn't hear it well. I have an iPhone so I converted it to m4r. Both the ...
user1790813's user avatar
0 votes
2 answers
233 views

How to label sound events in noisy .wav files

I'm trying to find a way of automatically identifying birdsong in phone-recorded sound files (1min long). Currently the algorithm I'm using doesn't label all of the events I want it to. (it is ...
Finn Maunsell's user avatar
0 votes
2 answers
4k views

How to transform SoundWave Picture to sound?

I would like to recreate a sound from a picture of a soundwave. But i see a lot of theorical question. And im not sure its possible.... I mean, it is possible to create a sound... But to make ...
Jebik's user avatar
  • 109
0 votes
1 answer
1k views

Normalizing audio waveforms code implementation (Peak, RMS)

I have some audio data (array of floats) which I use to plot a simple waveform. When plotted, the waveform doesn't max out at the edges. No problem - the data just needs to be normalized. I iterate ...
Andy Hin's user avatar
  • 103
1 vote
2 answers
2k views

Delay-and-sum beamforming: implementing steering in C

I am currently trying to understand how delay-and-sum beamforming works. I have found this website which explains the process of this technique very well in which the following C code is provided: <...
dll's user avatar
  • 137
1 vote
1 answer
40 views

Why does this video's audio change depending on the playing device?

Yesterday I noticed that if I played this video from my smartphone on speaker, one could barely hear anything. On the other hand, if I used earphones I could hear it perfectly. Then, I tried with ...
Tendero's user avatar
  • 4,980
1 vote
1 answer
1k views

Harmonic Product Spectrum algorithm. I dont understand one step

I was looking some Harmonic Product Spectrum algorithm examples and I came up with this: ...
Sebastian Araneda's user avatar
3 votes
2 answers
883 views

Generating inaudible sound waves using MATLAB

I want to generate inaudible sound waves (above 18kHz) with 48kHz sampling rate using MATLAB. Generating a sound wave with 18kHz frequency makes it inaudible. I've inserted the data in the 18kHz ...
danielle's user avatar
2 votes
1 answer
2k views

What does Fast and Slow Time Weighting signify for real-time audio signal processing?

I have a streaming audio signal that I am recording in real time. To make the explanation easy, say the signal that I receive is calibrated and has a sampling frequency, $F_s = 44100\textrm{ Hz}$. I ...
Arnav Mendiratta's user avatar
2 votes
1 answer
273 views

How to convert multi-band Energy Time Curve (ETC) to Impulse Response (IR)?

I am running my own sound simulator. Now I have obtained the Energy Time Curve (ETC, expressed by energy) by physical simulation, and they are stored in 8 discrete frequency bands (because decay of ...
Roy's user avatar
  • 21
1 vote
1 answer
4k views

How to find SPL of a WAV file given a calibration tone in WAV format

My ultimate aim is to find the Sound Pressure level of a digital WAV audio file and represent this (say, in MATLAB) as a dB SPL re 20 uPa on y-axis and frequency on x-axis. As this link mentions: ...
Arnav Mendiratta's user avatar
1 vote
2 answers
961 views

Generating sound of particular frequency matlab

I want to generate a high frequency (single) tone. I am using matlab (any other platform/tool for the same will also work). I am using the following snippet which I wrote by referring to various ...
Anup Agarwal's user avatar
0 votes
2 answers
468 views

Weird note structure on guitar audio waveform

Why do some times notes take these weird structures (circled below). That is a guitar monophonic audio waveform by the way (3 note melody). There are 3 notes played in this audio file. These are the ...
pavlos163's user avatar
  • 213
0 votes
3 answers
1k views

Guitar fundamental frequency vs pitch

When a note is played on the guitar, is the fundamental frequency always the strongest one or could a harmonic be the strongest one (in terms of amplitude). When could that happen? Also, when playing ...
pavlos163's user avatar
  • 213
1 vote
1 answer
77 views

Why doesn't a sudden loud noise sound high pitched?

The jumpiness or high change of a signal is due to the higher frequency components in the signal. So if I have a sound signal that increases suddenly in amplitude, why doesn't the signal sound very ...
JobHunter69's user avatar
6 votes
3 answers
7k views

Audio time stretching, without pitch shifting

This might be a Sound Design question, or a StackOverflow question since I am attempting to do this with Java. I would like to play back a sound at the same pitch, but stretched out in time. My ...
Phil Freihofner's user avatar
3 votes
2 answers
721 views

Sound absorption coefficients for frequencies higher than 8kHz and lower than 80Hz?

I am building a matlab simulation that applies room compensation to a 3D audio rendering system, and I'm searching for the frequency dependent sound absorption coefficients of various materials in ...
Luca's user avatar
  • 108
10 votes
3 answers
2k views

How does Siri recognize me saying "Hey Siri"?

I am trying to understand how my iPhone can continually listening for me saying Hey Siri, Alexa, ...
nowox's user avatar
  • 201
0 votes
1 answer
995 views

Measuring the average frequency of signals

I have many signals/audio data of a hammer banging on a nail. As the nail gets deeper it seems obvious to me (just by hearing) that the pitch is higher at the end than at the beginning. I assume that ...
user3494047's user avatar
2 votes
1 answer
2k views

How to do a Fourier transform on input from a microphone and obtain the average amplitude of a certain frequency range [closed]

I'm building a sound visualizer with LED strips controlled by python, and I'm trying to learn how to use pyAudio to perform rapid Fourier transforms on live mic ...
Rushil Joshi's user avatar
5 votes
2 answers
3k views

Double the "speed", double the frequency?

Usually, when some sample is played-back at twice the rate it was recorded, the pitch of the recording sounds an octave higher. For example: $g(t) = \sin(2 \pi f t)$ and $g(2t) = \sin(2 \pi f 2 t) \...
A_A's user avatar
  • 10.6k
2 votes
3 answers
2k views

Bose Chaudhuri Hocquenghem (BCH) vs Reed Solomon (RS) Error correcting codes and sound transmission channel modeling

I've implemented an acoustic based information transmission system using a bandwidth of 2 kHz and the band between 18.5 and 20.5 kHz. I'm trying to find the best error correcting code for it, in the ...
VMMF's user avatar
  • 1,078
1 vote
1 answer
243 views

MFCC for very short signals

I have very short signals (usually 100 samples from a 44100 hz audio recording ). These are 'bang' noises that a hammer makes on a piece of metal. I want to use these short signals as input to some ...
user3494047's user avatar
0 votes
1 answer
1k views

Determine Sound Intensity in SPL

I am recording ultrasonic sound using "Earthworks M50 50kHz Measurement Mic" and the sound card is "Rolando Quad-Capture USB 2.0 Audio Interface". The recording was done using Matlab into a wav file. ...
Ahsan Raza's user avatar
1 vote
2 answers
3k views

Extracting sound pressure from wav File

I have recorded an ultrasonic wave from a microphone using MATLAB command. After plotting that recorded wav file, the graph shows the amplitude between $1$ and $-1$. Can anyone tell that what is ...
Ahsan Raza's user avatar
2 votes
0 answers
183 views

Peak detection for signals that contain high frequencies

I have an question about peak detection. My purpose is creating a plot with an impulse response and the decay line at / above it. What I see till now is that there are two options for finding peaks: ...
Jan-Bert's user avatar
  • 335
5 votes
3 answers
800 views

Signal processing for audio and speech

I have started learning about signals and I am interested in sound signals. There are some questions that I need to resolve. For example in music, difference of notes like 'sol' and 'la' are about ...
virtouso's user avatar
  • 277
3 votes
0 answers
7k views

What's spectral "tilt"?

I am looking at article Speech-in-noise intelligibility improvement based on spectral shaping and dynamic range compression. In paragraph 2.2 the article mentions "tilt" of the spectral envelope. The ...
Danijel's user avatar
  • 492
5 votes
3 answers
5k views

Best beginners guide to sound/audio

I'm trying to learn audio programming, but am struggling with the basics of audio. Audio programming is new to me (I'm a C/C++ developer with over 30 years of experience), but I don't have any ...
Robert Loggia's user avatar
3 votes
1 answer
666 views

Why are zero values added in the FFT of a concatenated noise signal?

After concatenating an 11-second *.WAV (sample rate 44100 Hz) file 27 times (just by gluing them head-to-tail), I obtained a ~300-sec noise signal. I used MATLAB(2015b) and the FFT function on both ...
AliceD's user avatar
  • 298
1 vote
1 answer
570 views

Channel weighting of Mono channel in EBU R128

I am developing a loudness meter conforming to the EBU R128 recommendation. Currently, it works with stereo files but not with mono files. In the ITU Recommendation, where the metering algorithms ...
Sebastian's user avatar
  • 185
3 votes
0 answers
118 views

Calculating Phase Response for 2 microphones given the input

I want to measure and compare the phase response of two identical microphones. To do this, I have a pressure chamber, which allows to apply the same pressure to both microphones. I apply a sweeped ...
user6522399's user avatar
4 votes
2 answers
887 views

Signals cross-correlation

I am trying to measure phase difference between two received waves which are recorded by two separate PC's (microphones). The two PC's starts recording at the same time, then I make a sine wave sound ...
Redan Hassoun's user avatar
0 votes
2 answers
61 views

Synthetic sounds to describe motion in 3d or 100d

Watch a butterfly flitting about, or an optimizer chugging along in 3d or 100d -- a sequence of points $X_1 \ X_2 \ X_3\ \ldots$ How could one generate synthetic sounds that convey moving fast or ...
denis's user avatar
  • 598
1 vote
0 answers
106 views

What can I do to improve quality of sound recording?

I have an issue with a sound recording I made with my Olympus LS-12. The recorder is recording a constant unknown sound at about 10kHz. The sound is visible in the sonagram below as a faint dark ...
luciano's user avatar
  • 111