It depends what method is used.
If it's simply played back 'faster' without actually changing the data, then of course there is no lasting damage. This would be how a simple sampler would ordinarily do it, by changing the speed, ie data rate, of the playback.
If you resample so the same amount of audio plays back in half the time at the same sample rate - ...
You see that tree over there that you're barking up? It's the wrong one. There's nothing up there. Seriously.
You're contradicting yourself in your question.
You know there's no direct relationship between dBFS and dBSPL, but then you go on to hypothesize that there is.
You were right the first time.
The easiest way to look at this is that you are trying ...
In the special case that we are travelling with the sound source (e.g. in a plane), when we reach the speed of sound, the sound waves in the same direction will not be able to move away from the source. At this point, the sound waves will overlap which will cause accumulation of a lot of kinetic energy and resonance at every frequency in the sound. It will ...
The highly simplified answer is that sound in the real world is not sine waves with fixed frequencies. We learn about sound initially as frequencies or notes and simplify the concept down to thinking about various sine waves to make it easier to understand the fundamentals of what is going on.
The reality, however, is that all sound propagates through the ...
I believe Colin Hart's very excellent explanation is false in the second-to-last paragraph, beginning "NOW, ". If I am moving at the speed of sound and make a sound, the sound does NOT move at 686 mph relative to anything. The speed of sound is a property of the medium through which it propagates. It will always propagate at 343 (in this example). If the ...
Let's use some simpler numbers to illustrate the problem. Let's say the smallest string on a piano is 1m, and has a frequency of 200hz (it's a piano for giants). And let's say the lowest note on a piano is 100hz (only one octave because our giants have a narrow hearing range).
Given all other factors are consistent (tension, thickness, etc...), we know ...
This was a question asked in a Digital Sound and Music exam which has been copy-pasted here (even the "ex-pressed" has been left intact). The question only gives two marks. I think this is the answer:
f(t) = Amplitude * sin(2 * π * Frequency * t)
So in this case the Frequency would be replaced with 500.
The instantaneous sound pressure of a pure tone equals the ambient pressure (p0) with a superimposed pressure that varies in time as a sine function, i.e.: p(t) = p0 + A sin ωt, where A is the peak amplitude of the pressure variation and ω the angular frequency ( ω = 2πf ).
The amplitude A is related to sound pressure level L in dB by ...
Ok, let's start from the basics. Audio is usually recorded (sampled - converted from a continuous analogue signal to a digital record) and stored as a series of numbers representing the momentary voltage of the signal at regular intervals.
Sampling rate is the rate at which we check and record the momentary value of the analogue signal. Common sampling ...
Yes. This is all to do with wave interaction. Wave sources and sinks operate independently of each other, even when waves interact with each other and with the medium through which they travel.
This is more a physics question, however it is highly appropriate to sound.
A very good explanation can be found at:
Sound Design is a creative pursuit. Consequently the way to progress in Sound Design is to learn the common techniques involved, such as:
Pitch and Tone manipulation
Once you are familiar with these techniques, begin to experiment with particular sounds and see what you come up with.
There are ...
There are two factors that might muddy your results.
First, sound is absorbed by the air, so however you are measuring the sound level after it has hit the barrier, you won't know how much of the measured absorption was due to the air or the barrier, unless you measure twice - once right in front and once behind.
Second, the intensity of a continuous sound ...
A way to understand timbre and differences between instruments is to interpret harmonics by casting an eye on their spectrograms:
The flute has some tremolo visible on the high-end sustained harmonics that look consistent throughout the sample.
Piano, on the other hand, has gradient like harmonics fading towards the higher ones. Also the wavefile has a ...
Many stereos will generically say their range is from 20hz - 20kHz or somewhere close. This is the range of average human hearing. Each stereo will list somewhere it's frequency response. This is often done in a chart of some kind. In general the increase in volume by adjusting the volume knob would increase all frequencies in the "frequency range" by the ...
Sound pressure level is directly related to the amplitude of the waveform.
A pure tone is a sine-wave and sine-waves are defined by ω (omega) and t (time)
amplitude = sin(ωt) --- "sin" is the mathematical operator you did in trigonometry at school and t is time.
ω = 2 * π * f --- π is 3.141592654 (approx) and f is 500
So for 500Hz, ω = 3141.5927 (approx)...