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An analog signal from microphone is convrted to digital form. Analog signal is like water. Digital signal is like multiple boxes with water. These boxes has infinite thin walls. The format should have no quality losses lossless. Such PCM bitstream has no size compression.

And audio data in some high resolutions may not be sent via a connection interface SPDIF, in instance. Channel number is limited by allowable throughput too. Compression may be lossless or cause sound quality losses. Read details Sometimes, size compressed audio is called as bitstream.

Read more DAC is digital to analog converter kind of audio interface. Audio output is connector, audio data transmition protocol and hardware, included to an audio interface, to transmit audio data to other device. Audio output may transmit signal in analog form. It's analog audio output. Audio signal in musical system amplifier, AV-receirer, etc. Special device - analog-to-digital converter - rapidly measure momentary values of the audio signal its voltage.

Let's imagine a machine, that can form water level by the written value sequence. And we get the same water wave. Analog-digital converter ADC is a device, that periodically measure analog signal voltage and send the measured values as numbers in digital form to PCM digital audio output. PCM encoding is the conversion of an analog signal to digital form.

Quantization is the measurement step of the voltage level of an analog signal. Samples may be stored and transmitted without altering of information.

It is the main advantage of digital signals, comparing analog ones. Sample rate sampling rate is a number of samples per second measured in Hz, Hertz. As rule, an analog signal is coded as real numbers math definition , that are usual numbers we use permanently. Let's pay attention to "theoretical" word. Real implementations require to account other factors too.

Read below about myths, where we'll discuss, why higher sample rates are used. In simple words it is not exact math definition the Nyquist—Shannon sampling theorem may sound as:. Below we will consider the theorem details, when More exact the theorem wording in sound terms: Endless analog sine signal may be coded to digital form and restored with sampling rate 2 times more the signal 's frequency.

M ore samples per finite signal duration keep more information about source signal to restore it from digital to analog form. More samples per duration, it is closer to infinity.

Alternatively, the input samples may be processed via Hilbert transform. It converts real numbers to complex ones. Analog-digital converter capture full frequency band at the input. It adds noise to the coded digital signal. But the analog filter isn't steep enough. Also in DAC sampling rate may be increased oversampling to better work with the analog filter.

Oversampling works with the digital filter in pair. There is a myth that non-multiple resampling causes more distortions, than multiple one. But in case and Hz, resampling is applied the same way.

Maximum value of the word is the maximal positive value of an analog signal at ADC input. Its code is:. Minimal value of the word is maximal negative value of the analog signal at ADC input. Rounding is bit depth reducing via removing of one or more bits with altering of reduced number according to removed bit s. Codes of analog values, stored into the words have precision limitation. The limitation is defined by total number of measured levels L.

So stored codes samples are not equal exactly to real analog voltage. Quantization error is difference between sample digital value and real voltage of analog signal. The energy of quantization noise is constant in total band. Thus, increasing of the total band of an analog signal after DAC sampling rate increasing decrease the noise level in the audible range [ It happens because audible range has a fixed width.

In the digital domain quantization noise level is decreased about 6 dB for Fourier transform length 2 times more. In the digital domain, N Q is the same independently sample rate. But the Fourier transform divide digital band to parts small sub-bands. Fourier transform is converting oscillogram time domain to spectrum frequency domain.

In digital audio, we mean discrete Fourier transform in most cases. The discrete mean, that spectrum is divided to taps. FFT fast Fourier transform is case of Fourier transform. It's length is 2 K , where K is integer number. If there are tips 2 times more, noise energy is redistributed.

And each tap have energy 2 times lesser. If we make tap width as before the redistributing tap width at the part A of the picture , noise level will 2 times lesser. Because square of noise is constant. It happens on computer display, when tap width have same pixel width on a screen. Read below more about bit depth, quantization noise and dynamic range for 16 bit implementations.

But it is not so. Because "the stairs" are smoothed by analog filter at the digital-analog converter output. But that's not exactly true. Because the analog filter isn't ideally "brick wall".

Half of the aliases are flipped horizontally. In ideal audio system without non-linear distortions these aliases will inaudible. In the table noted only file abilities, that author know. If you have additional information to correct description or other, contact us.

Sometimes files with same extension may contains different extensions. A reading software player, converter, editor, other parse file. As rule, file consists of data blocks. These blocks have identifiers. And the reading software recognize the block types. Sometimes the software check data integrity. If there are non-correct data, the software may to reject file opening depend on implementation. Size compressed file types are used for saving hard disk space.

Especially, it is actually for portable devices: digital audio players DAP , mobile phones, etc. Portable devices are able to playback multichannel files. But it is listened at stereo headphones, as rule. So multichannel records consume disk space to extra channels. The space extra size issue may be solving via downmixing audio files to stereo.

It is impossibly to get rid of jitter in real music systems. Because there are electromagnetic interference, non-stability of clock generators, power line interference issues. Quantization error cause non-linear distortions. It correlate with musical signal. Correlated distortions are considered as especially unwanted to perceived sound quality. Dither is extremely low level noise, that added to musical signal before ADC or before bit depth truncation prior to DAC.

To reduce noise in audible band, noise shaping may be applied. It looks like "pushing" of noise energy to upper part of frequency range. But the shaping demands of band reserve to the "pushing".