question: Why does the pitch of a sound become harder to distinguish when it is at a Lower volume?
comment: The pitch you hear has to do with how fast the note makes your eardrum vibrate. Volume has to do with how much force is behind the vibration. At high volumes there is a lot of vibration so it's easier for your brain to pick up on the pattern your eardrum is registering. At much quieter volumes there is much less vibration so it's harder for your brain to pick out this pattern; also note, at VERY high volumes the vibrations are so forceful that they can start to distort and your brain again has difficulty picking up the pattern.
passages: When sensorineural hearing loss (damage to the cochlea or in the brain) is present, the perception of loudness is altered. Sounds at low levels (often perceived by those without hearing loss as relatively quiet) are no longer audible to the hearing impaired, but sounds at high levels often are perceived as having the same loudness as they would for an unimpaired listener. This phenomenon can be explained by two theories, called "loudness recruitment" and "softness imperception".
   Some pipe organs makes use of this phenomenon as a resultant tone, which allows relatively smaller bass pipes to produce very low-pitched sounds.
  
  Section::::Audio processing applications.
   Varying intensity levels can also have an effect on masking. The lower end of the filter becomes flatter with increasing decibel level, whereas the higher end becomes slightly steeper. Changes in slope of the high frequency side of the filter with intensity are less consistent than they are at low frequencies. At the medium frequencies (1–4 kHz) the slope increases as intensity increases, but at the low frequencies there is no clear inclination with level and the filters at high center frequencies show a small decrease in slope with increasing level. The sharpness of the filter depends on the input level and not the output level to the filter. The lower side of the auditory filter also broadens with increasing level. These observations are illustrated in Figure H.
   Since CDs were not the primary medium for popular music until the late 1980s, there was little motivation for competitive loudness practices then. The common practice of mastering music for CD involved matching the highest peak of a recording at, or close to, digital full scale, and referring to digital levels along the lines of more familiar analog VU meters. When using VU meters, a certain point (usually −14 dB below the disc's maximum amplitude) was used in the same way as the saturation point (signified as 0 dB) of analog recording, with several dB of the CD's recording level reserved for amplitude exceeding the saturation point (often referred to as the "red zone", signified by a red bar in the meter display), because digital media cannot exceed 0 decibels relative to full scale (dBFS). The average level of the average rock song during most of the decade was around −16.8 dBFS.
   As this effect is more pronounced with higher input signals, the harder "attack" of a note will be compressed more heavily than the lower-voltage "decay", making the latter seem louder and thereby improving sustain. Additionally, because the level of compression is affected by input volume, the player can control it via their playing intensity: playing harder results in more compression or "sag". In contrast, modern amplifiers often use high-quality, well-regulated power supplies.
  
  Section::::Approaches.:Speaker distortion.
   These frequency shifts in response to environmental stimuli have been shown to improve performance in perceptual behavior tasks in adult mice that were tone-reared during auditory critical period. Adult learning and critical period sensory manipulations induce comparable shifts in cortical topographies, and by definition adult learning results in increased perceptual abilities. The tonotopic development of A1 in mouse pups is therefore an important factor in understanding the neurological basis of auditory learning.
   Similarly the response of a "low shelf" (or "low shelving or "bass control") can be represented as
  
  In this case the inclusion of the leading factor simply indicates that the response at frequencies much higher than "f" or "f" is unity and that only bass frequencies are affected.
   The energy density of sound waves decreases as they spread out, so that increasing the distance between the receiver and source results in a progressively lesser intensity of sound at the receiver. In a normal three-dimensional setting, with a point source and point receptor, the intensity of sound waves will be attenuated according to the inverse square of the distance from the source.
  
  Section::::Damping.
   Wallach et al. did not systematically vary the intensities of the two sounds, although they cited research by Langmuir et al. which suggested that if the second-arriving sound is at least 15 dB louder than the first, the precedence effect breaks down.
  
  The "Haas effect" derives from a 1951 paper by Helmut Haas.
   Often, the acoustic summing of a room and system are much inferior to the ideal, so the specific relative level will change from −3 dB to 0 dB as the mono signal is panned from center to hard left or right, and this sounds natural. The idea is that when one directs signals left and right with the pan pot, the perceived loudness will stay the same, regardless of latitude.
   In September 2011, Emmanuel Deruty wrote in "Sound on Sound", a recording industry magazine, that the loudness war has not led to a decrease in dynamic variability in modern music, possibly because the original digitally-recorded source material of modern recordings is more dynamic than analogue material. Deruty and Tardieu analyzed the "loudness range" (LRA) over a 45-year span of recordings, and observed that the crest factor of recorded music diminished significantly between 1985 and 2010, but the LRA remained relatively constant. Deruty and Damien Tardieu criticized Sreedhar's methods in an AES paper, saying that Sreedhar had confused crest factor (peak to RMS) with dynamics in the musical sense (pianissimo to fortissimo).
   In case (a), above, the usual situation is that the derived low pass response attenuates at a much slower rate than the fixed response. This requires the speaker to which it is directed to continue to respond to signals deep into the stopband where its physical characteristics may not be ideal. In the case of (b), above, both speakers are required to operate at higher volume levels as the signal nears the crossover points. This uses more amplifier power and may drive the speaker cones into non-linearity.
  
  Section::::Classification.:Models and Simulation.
   One of the well-established methods of studying tonotopic patterning in the auditory cortex during development is tone-rearing. In mouse Primary Auditory Cortex (A1), different neurons respond to different ranges of frequencies with one particular frequency eliciting the largest response – this is known as the "best frequency" for a given neuron. Exposing mouse pups to one particular frequency during the auditory critical period (postnatal day 12 to 15) will shift the "best frequencies" of neurons in A1 towards the exposed frequency tone.
   Certain kinds of ADCs known as delta-sigma converters produce disproportionately more quantization noise at higher frequencies. By running these converters at some multiple of the target sampling rate, and low-pass filtering the oversampled signal down to half the target sampling rate, a final result with "less" noise (over the entire band of the converter) can be obtained. Delta-sigma converters use a technique called noise shaping to move the quantization noise to the higher frequencies.
  
  Section::::Example.
   Temporal Theory posits that the cause is from looking at the phase locking to tell what the pitch is. This theory has a hard time explaining diplacusis. There are some examples of pitch which don’t have an "edge" on the basilar membrane, which this would account for—i.e. white noise, clicks, etc. (need a reference)
   The ability to differentiate frequencies of two successive tones was also tested for this elephant using a similar conditioning paradigm. The elephant’s responses were somewhat erratic, which is typical for mammals in this test. Nevertheless, the ability to discriminate sounds was best at frequencies below 1 kHz particularly at measurements of 500 Hz and 250 Hz.
   When measuring hearing thresholds it is always easier for the subject to follow a tone that is audible and decreasing in amplitude than to detect a tone that was previously inaudible.
  
  This is because 'top-down' influences mean that the subject expects to hear the sound and is, therefore, more motivated with higher levels of concentration.
  
  The 'bottom-up' theory explains that unwanted external (from the environment) and internal (e.g., heartbeat) noise results in the subject only responding to the sound if the signal to noise ratio is above a certain point.
   Because a downward compressor only reduces the level of the signal, the ability to add a fixed amount of "make-up gain" at the output is usually provided so that an optimum output level is produced.
  
  Section::::Controls and features.:Look-ahead.
   In rock music from the late 1960s to the 2000s, the timbre of specific sounds is important to a song. For example, in heavy metal music, the sonic impact of the heavily amplified, heavily distorted power chord played on electric guitar through very loud guitar amplifiers and rows of speaker cabinets is an essential part of the style's musical identity. You could not take the heavily amplified electric guitar part and substitute it with the same notes played on a grand piano or pipe organ.
  
  Section::::Psychoacoustic evidence.
   In audio, a related measurement, signal-to-noise ratio, is usually defined as the difference between the nominal level and the noise floor, leaving the headroom as the difference between nominal and maximum output. It is important to realize that the measured level is a time average, meaning that the peaks of audio signals regularly exceed the measured average level. The headroom measurement defines how far the peak levels can stray from the nominal measured level before clipping. The difference between the peaks and the average for a given signal is the crest factor.
   The "loudness" control associated with a loudness compensation feature on some consumer stereos alters the frequency response curve to correspond roughly with the equal loudness characteristic of the ear. Loudness compensation is intended to make the recorded music sound more natural when played at a lower levels by boosting low frequencies, to which the ear is less sensitive at lower sound pressure levels.
  
  Section::::Normalization.
  
  Loudness normalization is a specific type of audio normalization that equalizes perceived level such that, for instance, commercials do not sound louder than television programs. Loudness normalization schemes exist for a number of audio applications.
  
  Section::::Normalization.:Broadcast.
   The ATH is the lowest of the equal-loudness contours. Equal-loudness contours indicate the sound pressure level (dB SPL), over the range of audible frequencies, that are perceived as being of equal loudness. Equal-loudness contours were first measured by Fletcher and Munson at Bell Labs in 1933 using pure tones reproduced via headphones, and the data they collected are called Fletcher–Munson curves. Because subjective loudness was difficult to measure, the Fletcher–Munson curves were averaged over many subjects.
   Some organ pipe ranks are designed to be overblown. For instance, a given pipe of the harmonic flute stop is twice the length of pipes of other stops designed to sound the same pitch. When such a pipe is overblown, it sounds a fundamental tone one octave higher than other pipes of its length. For instance, a harmonic flute pipe that is 16 feet long is designed to sound the same note as most 8-foot pipes.
  
  Section::::Further reading.
   A "high shelf" or "treble control" will have a frequency response "|H(f)|" whose square is given by:
   The "unmasked threshold" is the quietest level of the signal which can be perceived without a masking signal present. The "masked threshold" is the quietest level of the signal perceived when combined with a specific masking noise. The amount of masking is the difference between the masked and unmasked thresholds.
  