Definition of sampling bits, sampling rate and bit rate in audio (transfer) Part 3

1. Why do many professional standards reach 24bit/192KHz?

It is now common to use the 48kHz or 96kHz recording rate in engineering, and only convert to the 44.1kHz CD format during the final mastering process, which reduces distortion caused by multiple sample rate conversions.

In the field of computing, the AC97 specification, which is an audio hardware codec standard, only specifies 48 kHz. This causes nearly all input and output signals to be resampled (the professional term is called sample rate conversion, or SRC). SRC generally causes loss of sound quality, and the simpler (ie poorer) SRC algorithms can cause significant deterioration of sound quality. But this is already a fait accompli.

2. Since 44K is enough, why use 192KHZ to record?

First of all, 20kHz is just the hearing threshold for most people, i.e. the human ear is very insensitive to sounds above 20kHz. Insensitivity to attention does not mean a total inability to perceive. The tones of most musical instruments (especially pianos and strings) are rich in higher harmonics, known in musical terms as higher harmonics. CD audio with a cutoff frequency of 22.05 kHz gives people who are used to listening to real instruments an unnatural feel, especially in the high frequencies, because the Nyquist cutoff frequency distorts the signal from harmonics. of higher frequencies.

Second, digital recordings often require post-processing. Audio processing can introduce more distortion into the signal, including signal distortion, spectral aliasing, and more. If the original signal is only sampled at 44.1 kHz during recording, it must be upsampled before post-processing to expand the sample rate. Since this expansion is “fake”, there is really no more useful original signal, and the quality of the upsampling algorithm will also affect the distortion of the original recording signal, so this approach is undesirable. Therefore, it is common practice to sample at a higher frequency.

In today’s fully professional digital recording studios, recording, mixing and mastering are no longer compliant with the CD standard, instead the HD audio standard is preferred. which:

Use 24Bit 48KHz, 24Bit 96KHz, 24Bit 192KHz three specifications to record, of course, 24Bit 48KHz is used by some small recording studios, because their processor resources are limited. And all the big recording studios use 24bit 96KHz and 24bit 192KHz for recording.

So what are the benefits of such a recording specification?

1. Comply with HD audio standard, which is also the main standard in the future. The finished product can be directly applied to HDCD, DVD-Audio, Blu-ray disc, digital music download business and digital player business to media.

2. Fully take care of the digital video and video business, and the multi-channel film and video will adopt the HD audio specification. Including the use of portable mobile digital video equipment.

3. Fully take care of the consumer audio playback business, such as: Intel HD-Audio audio standard, AC97 audio codec, MP3 / mp4 / phone / game console portable audio highest quality audio playback.

Currently, the highest quality standard in the professional recording industry is: 24 bits deeper than a specific point, 192000 Hz sampling rate, referred to as “24 bits/192 KHz”. Of course, this standard will continue to improve in the future, and it is also possible to move towards 32Bit 384KHz.

In fact, the (genuine) products sold in the current CD market are usually HDCD discs at the lowest level, when you buy discs, you find that they are basically HDCD logos, that is, a CD contains two audio tracks: Normal CD track and HDCD track. The CD track records a 16-bit signal at 44.1 KHz (this is the compatible content on this disc, considering early CD players), and the HDCD track records a 24-bit signal at 96 KHz ( this is the main content of the disc). Ordinary CD players can only play CD audio track signals, and HDCD audio tracks require an HDCD player to play (in fact, most DVD players today can play HDCDs, and modern computers work even better).

Definition of sampling bits, sampling rate and bit rate in audio (transfer) Part 2

Bitrate values ​​compared to real audio:

16 Kbps = phone quality
24 Kbps = increase phone quality, shortwave transmission, longwave transmission, European standard medium wave transmission
40 Kbps = American standard medium wave transmission
56Kbps=Voice
64 Kbps = boost voice (best bitrate setting for cell phone ringtones, best setting for cell phone mono MP3 players)
112 Kbps = FM stereo FM transmission
128 Kbps = tape (best setting for a mobile phone stereo MP3 player, best setting for a low-end MP3 player)
160 Kbps = HIFI high fidelity (best setting for mid to high end MP3 players)
192Kbps=CD (best setting for high-end MP3 players)
256Kbps=Studio Music Studio (for music enthusiasts)
In fact, with the advancement of technology, the bitrate is also getting higher and higher, the maximum bitrate of MP3 is 320Kbps, but some formats can reach higher bitrates and superior sound quality.
For example, the emerging APE audio format can provide true audiophile lossless sound quality and smaller volume than WAV format, and its bit rate is usually 550kbps—–950kbps.
Common coding patterns:

Dynamic bit rate VBR (Variable Bitrate), ie there is no fixed bit rate. The compression software immediately determines which bitrate to use based on the audio data during compression. This is a method that takes into account the quality of the file. and file size The recommended encoding mode;
ABR (Average Bit Rate) Average Bit Rate is an interpolation parameter of VBR. LAME created this encoding mode in response to the low file volume ratio of CBR and the variable size of files generated by VBR. Within the specified file size, ABR takes every 50 frames (about 1 second for 30 frames) as a segment. A relatively low flow rate is used for low frequency and insensitive frequencies, and a high flow rate is used for high frequencies and high dynamic performance. It can be used as VBR and CBR, a compromise option.
CBR (constant bit rate), constant bit rate, means that the file has a bit rate from start to finish. Compared to VBR and ABR, the compressed file size is very large and the sound quality will not improve significantly compared to VBR and ABR.
In simple terms:

In a nutshell, sample rate and bit rate are like horizontal and vertical coordinates on the coordinate axis.

The sampling rate on the abscissa represents the number of samples per second.

The bit rate on the ordinate represents the precision when quantizing analog quantities with digital quantities.

The sample rate is similar to the number of frames of moving images. For example, the sampling rate of movies is 24 Hz, the sampling rate of PAL format is 25 Hz, and the sampling rate of NTSC format is 30 Hz. When we play back the still images sampled at the same rate as the sampling frequency, we see a continuous image. In the same way, when a CD recorded at a sampling rate of 44.1 kHz is played back at the same rate, a continuous sound can be heard. Obviously, the higher the sample rate, the more coherent the sound will be heard and the picture will be seen. Of course, the sampling rate that human auditory and visual organs can distinguish is limited, which is basically higher than sound sampled at 44.1 kHz, and most people haven’t noticed the difference.

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the highest digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone.

Definition of sampling bits, sampling rate and bit rate in audio (transfer)

Number of samples (sample size):

The number of sampling bits can be understood as the resolution of the sound processed by the capture card. The higher the value, the higher the resolution and the more realistic the sound recorded and played back. The first thing we need to know: sound files on the computer are represented by the numbers 0 and 1. So the essence of recording on the computer is to convert the analog sound signal into a digital signal. On the contrary, during playback, the digital signal is restored to an analog sound signal output. The capture card bit refers to the binary digits of the digital sound signal used by the capture card when capturing and playing sound files. The bits on the capture card objectively reflect the accuracy of the digital sound signal’s description of the input sound signal. 8 bits represent the eighth power of 2–256 and 16 bits represent the sixteenth power of 2–64K. For comparison, for the same musical data, a 16-bit sound card can divide it into 64,000 precision units for processing, while an 8-bit sound card can only process 256 precision units, resulting in a large loss of signal. sampling effect is naturally incomparable.

It is usually said in the market, 16bit/24bit/32bit. The higher the value, the better the sound.

Sampling rate:

Sample rate (also called sample rate or sample rate) defines the number of samples per second taken from a continuous signal to form a discrete signal, and is expressed in hertz (Hz). The inverse of the sample rate is called the sample period or sample time, which is the time interval between samples. The sampling theorem states that the sampling frequency must be greater than twice the bandwidth of the sampled signal. Another equivalent statement is that the Nyquist frequency must be greater than the bandwidth of the sampled signal.

If the signal bandwidth is 100 Hz, the sample rate must be greater than 200 Hz to avoid aliasing.

In other words, the sampling frequency must be at least twice the frequency of the largest frequency component of the signal; otherwise the original signal cannot be recovered from the signal samples. Oversampling refers to the sampling rate that exceeds twice the bandwidth of the signal, so that the poorly performing analog anti-aliasing filter can be replaced with a digital filter.

Bit rate:

Bitrate refers to the sampling rate at which digital sound is converted from analog to digital format. The higher the sampling rate, the better the quality of the restored sound. As a benchmark for the efficiency of digital music compression, bit rate indicates the rate of the number of bits bps (bit per second, bits per second) transmitted per unit of time (1 second). Kbps (in layman’s terms is 1000 bits per second) is usually used as the unit. The digital music bitrate on CD is 1411.2 kbps (ie, to record 1 second of CD music, 1411.2 × 1024 bits of data are required). time unit (1 second) The amount of data (BIT) is large, which means the sound quality of the music file is good. However, when the BITRATE is high, the file size increases, which will occupy a large amount of memory capacity. they are 32-256 Kbps. Of course, the wider the rate, the better, but 320 Kbps is the highest level at the moment.