![]() ![]() SPL will usually be quoted in dB relative to a very quiet sound, where 0dB is the threshold of hearing (virtually silent). The general term for sound magnitude is "Sound Pressure Level" (SPL). These take into account the power, distance and frequency content of the sound. When dealing with "loudness" of sound, there are a number of different standards that are used for measuring the magnitude. If you reduce the level of this signal so that it has a peak amplitude of 0.5, then it has a peak level of -6dB (minus 6 decibels full scale) ![]() (more precisely it should be 0dBfs, but commonly just called 0dB). If you have a signal that has a peak amplitude of 1.0, then it has a peak level of 0 dB. In digital audio, 0dB is usually considered to be the maximum valid signal level (though in certain special circumstances it is possible to exceed 0 dB).Īll other signals are less than 0 dB and are measured as negative values - for example Similarly, if you import or record a sound and amplify it so that the peak level reaches the top/bottom of the track (+/- 1.0), then the peak level is 0dB. If you generate a tone in Audacity with "Amplitude = 1.0", the generated tone will have a peak level of 0dBfs. Signal levels are usually described with reference to the "full scale" magnitude (dBfs). You can see this numerical scale in Audacity if you look at the vertical track scale in the default track view. In the case of digital audio, the reference level is represented numerically by sample values of +/- 1.0 It is a measure of magnitude relative to a reference level. But I can't seem to find any clips which straight out define the starting decibel level.ĭecibels (dB) are a relative measure, not an absolute measure. You see we're working in a vacuum until you tell us a lot more about the "show."ĬAPLAB wrote:I simply need an audio file that I can manipulate to several decibel levels to be used for research purposes, ei. If you start with a -80 tone and reduce it 20dB, you will get flat-line in 16-bit sound. That's easy as long as you don't hit the limits at either end. If you start with an arbitrary correct sound file and Effect > Amplify it -10dB, then the result will be 10dB lower than whatever it was you started with. If you start out life with a "normal" audio file of someone speaking and reduce it 60dB, then there will be almost no sound left if you try to play it through a sound system whose noise floor is -60dB. 16-bit sound only has about a 90dB range between maximum peak loudness and the channel floor - and that's before you try to squeeze the test through analog electronics which tend to be far worse. Then there's the technical specification. Did the sound start out inside the computer? If it's an external sound, then you're going to run into all the variables of your sound card or other interface device. You wouldn't be the first researcher trying to nail down all the variables to perform a project. If it's just about anything else, it's fogged in complexity. If it's straight, pure tones, then Whomper's solution will work. Of what? The answers change a great deal depending on the sound file. Not the average loudness or any of the other ways to measure level. on the one highest peak in the performance. ![]()
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