What is the sample rate and bit rate?


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What is the sample rate?

Sample Rate

Frequency is defined as the number of cycles of periodic motion per unit of time. The SI unit of frequency is called hertz (Hz, after its inventor Heinrich Hertz). One hertz corresponds to one cycle (or complete oscillation) per second.

Sample Rate

Example. Sound waves have a frequency in the range of approximately 20 to 20,000 Hz. This means that at any point along the path of the sound wave, the pressure will fluctuate from high to low, 20 to 20,000 times per second.

In digital audio, the maximum frequency that can be successfully recreated is half the sample rate. Therefore, with a sample rate of 44.1 kHz, frequencies up to 22.05 kHz can be recreated. Wave frequency refers to how many times per second a wave moves from its highest point to its lowest point and vice versa. It is usually measured in hertz (Hz) or cycles per second. The frequency of the wave determines its height. High-frequency waves have a high pitch, while lower frequencies have a lower pitch. The average person can hear frequencies from 15 or 20 Hz to about 20,000 Hz (20 kHz).

Analog wave The wave amplitude refers to half the distance between the highest point of the wave and the lowest point. The greater the amplitude of the wave, the greater its volume, which is generally measured in decibels (dB). The decibel range for human hearing is complex and depends on the frequency of the sound in question, the age of the person and the listening environment, but varies from approximately 0 to 120 dB, with each 10 dB change corresponding to a doubling of the perceived volume.

Absolute Threshold: ATH is the volume level at which a certain sound can be detected 50% of the time.

What is the bit rate?

Bit rate refers to the data transfer rate (that is, how many bits are transmitted in a given time), generally expressed in bits per second. Common units of bit rate are kilobits per second (Kbps) and megabits per second (Mbps). The term is also commonly used when talking about digital sampling and sample rates. For example, the MP3 audio compression algorithm is often configured to output files at a bit rate of 128 kbps. This means that the file contains an average of 128 kilobits for every second of audio (960 KB per minute). This is in contrast to CD audio, which is encoded as 44,100 16-bit stereo samples per second: 1411.2 kbps (16-bit x 44100 Hz x 2ch).

Often times, bytes are written in uppercase and are multipliers (for example, “KB” for kilobytes) and lowercase factors are bits (for example, “kb” for kilobytes). All modern computers use 8-bit bytes.

MP3 bit rate
The MP3 bit rate can be misleading. For example, an MP3 “constant bit rate” (CBR) of 128 kbps will use approximately 128 kilobits for every second of encoded audio (so the file size in bits divided by the length of the audio is approximately 128,000), and Your frame headers will appear at regular intervals, but internally, frame-by-frame, you can encode audio at bit rates higher or lower than 128 kbps by using a bit pool (the ability of a frame to use spare bits from a previous block). However, the size of this bucket, and thus the amount of variability, is limited, so 128 kbps will be very close to the effective bit rate throughout the file.

See also: 8D surround sound and how to do it
As another example, “128 kbps VBR MP3” is often incorrect, as the purpose of VBR is to allow each of the internal MP3 sectors to have its own bit rate. When people refer to the VBR MP3 bit rate, they are generally referring to the actual average bit rate of their frames. If the length of the encoded audio is known, then the “bit rate” can be the data size of the file divided by its duration, which will be fairly close to the same number. However, the length of an MP3 VBR cannot be accurately determined without scanning all the frames.


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Digital Sound and Sample Rate

Digital Sound and Sample Rate

Sample Rate

Given the wide availability of inexpensive digital audio equipment, we invite you to take a closer look at digital audio.

Sample Rate

Acoustic sound is a continuous process in time and in amplitude, that is, the air pressure changes smoothly with time and does not jump from one value to another. Acoustic sound can be converted into an electrical signal using a microphone that, depending on the change in air pressure, changes the electrical voltage it generates at the output. After the conversion of an acoustic sound into an electrical signal, continuity is maintained in time and in amplitude: the signal voltage changes in the same way that the air pressure changes, which is why this sound is called analog. We can record an electrical signal on magnetic tape and convert it back to sound using a loudspeaker that functions as a “reverse microphone”: it moves air in response to changes in voltage. Respectively,

Despite the fact that the analog electrical signal has regularly served humanity for decades, over time some of its representatives (of humanity) became clear that the analog signal and magnetic recording are not the best ways to transmit and store audio information, since both during transmission and during storage occur. unavoidable losses, i.e. sound degradation. At the same time, the transmission and storage of data on computers that operate exclusively on digital data can be done without any loss. The only question is how to convert analog audio to digital and vice versa.

To solve the first problem, there are special devices known as analog-to-digital converters (ADCs). These devices are capable of converting a continuous analog signal into a sequence of separate numbers, that is, making it discrete (English discrete – separate, consisting of separate parts). The conversion takes place as follows: the device measures the amplitude of the analog signal many times per second and outputs the measurement results in the form of numbers.

Analog signal
Sampling
Sampled signal
As seen in the figure, the measurement result is not an exact analog of a continuous electrical signal. How much does digital sound compare to analog? Obviously, this correspondence will be more complete the more often the measurements are made and the more accurate they are. The frequency at which measurements are taken is called the sample rate. And the precision of amplitude measurements is indicated by the number of bits used to represent the measurement result. This parameter is called the bit depth.

Sampling rate
So, the conversion of an analog signal to digital consists of two stages: sampling in time and quantization in amplitude. Time sampling means that the signal is represented by a number of its samples (samples) taken at regular intervals. For example, when we say that the sample rate is 44.1 kHz, it means that the signal is measured 44,100 times per second (in MO, the more intelligible term “sample rate” is usually used, however, “sample rate “is more correct.).

The main issue in the first stage of converting an analog to digital signal (digitizing) is to choose the sampling frequency of the analog signal. As already mentioned, the higher the frequency, the closer the digital signal is to the analog. However, in proportion to the increase in frequency, the following increases: a) the intensity of the digital data stream and the bandwidth capabilities of the interfaces are not unlimited, especially if several channels are recorded / played simultaneously; b) the computational load of digital effects processors and their computational capabilities are also limited; c) the amount of memory required to store the digital signal. Obviously a compromise is needed.

The choice of the sampling frequency affects the frequency range of the received digital sound or the maximum frequency of an analog signal, correctly represented in digital. The range of frequencies a person hears is believed to be 20 to 20,000 Hz. According to the well-known Nyquist theorem, in order for an analog (continuous in time) signal to be accurately reconstructed from its samples, the sampling frequency it must be at least twice the maximum audio frequency. An audio frequency equal to half the sampling frequency is called the Nyquist frequency and is the maximum frequency that a given digital system can store and reproduce correctly. Thus, if the real analog signal that we are going to digitize contains frequency components from 0 Hz to 20 kHz.