Digital Audio Bit Depth: Understanding the Basics


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Digital Audio Bit Depth: Understanding the Basics

Audio Bit Depth
Audio Bit Depth
Audio Bit Depth
Audio Bit Depth

What is Digital Audio Bit Depth?

Digital audio bit depth refers to the number of bits used to represent each sample in a digital audio signal. Bit depth is a crucial aspect of digital audio because it affects the accuracy and dynamic range of the signal.

In digital audio, sound is captured and processed as a series of discrete samples, with each sample representing the amplitude of the sound wave at a specific point in time. The bit depth determines the number of possible amplitude values that can be represented in each sample.

How Does Bit Depth Affect Audio Quality?

The higher the bit depth, the more accurately the digital audio signal can represent the original analog waveform. A higher bit depth allows for a greater dynamic range, which means that the quietest sounds can be represented with more accuracy, and the loudest sounds can be represented without distortion.

For example, a 16-bit audio signal can represent 65,536 possible amplitude values, while a 24-bit audio signal can represent 16,777,216 possible amplitude values. This means that a 24-bit audio signal can capture a wider range of dynamic levels and is capable of greater accuracy and detail than a 16-bit audio signal.

What is the Relationship Between Bit Depth and Signal-to-Noise Ratio?

As the bit depth increases, the signal-to-noise ratio (SNR) also increases. SNR is the ratio between the desired signal (the audio) and the background noise.

A higher bit depth means that there are more possible amplitude values for each sample, which reduces the amount of quantization noise in the signal. Quantization noise is a type of distortion that occurs when the analog signal is converted to digital.

How is Bit Depth Measured?

Bit depth is measured in bits per sample. Common bit depths in digital audio include 16-bit, 24-bit, and 32-bit.

What is Dithering?

Dithering is a process used to reduce the distortion caused by quantization error in digital audio. When an analog signal is digitized, the conversion process rounds the amplitude of each sample to the nearest possible value.

Dithering adds a small amount of random noise to the signal before it is quantized, which allows for a smoother transition between amplitude values and reduces the audible effects of quantization error.

What is the Difference Between Bit Depth and Sample Rate?

While bit depth determines the number of possible amplitude values in each sample, sample rate determines the number of samples taken per second. A higher sample rate allows for greater accuracy in capturing the original analog waveform, but it does not affect the dynamic range or accuracy of each individual sample.

What is the Ideal Bit Depth for Recording and Mixing?

The ideal bit depth for recording and mixing depends on the intended use of the final product. For most applications, a bit depth of 24 bits is considered to be sufficient, as it provides a wide dynamic range and high accuracy.

However, for applications that require extreme accuracy and detail, such as classical music recording, a higher bit depth may be necessary.

What is the Relationship Between Bit Depth and File Size?

As the bit depth increases, the file size of the digital audio also increases. This is because a higher bit depth requires more storage space to represent the additional amplitude values.

What is the Relationship Between Bit Depth and Processing Power?

Higher bit depths require more processing power to manipulate and process. This is because the additional amplitude values must be calculated and stored in memory.

What Happens When a High Bit-Depth Audio File is Converted to a Lower Bit-Depth Format?

When a high bit-depth audio file is converted to a lower bit-depth format, the result is a loss of some of the original audio data. This is because the lower bit-depth format has fewer bits to represent the audio data, which means that some of the information is lost in the conversion process.

For example, if a 24-bit audio file is converted to a 16-bit format, the conversion process will discard the least significant 8 bits of each sample. This can result in a loss of some of the subtle nuances and details in the audio, which can be particularly noticeable in quiet passages or when the audio is heavily processed.

It’s worth noting that some audio formats, such as MP3 and AAC, use lossy compression to reduce the file size. This means that even if the original file was at a high bit-depth, converting it to a lower bit-depth format such as MP3 will result in a further loss of data due to the compression algorithm.

What is Dithering and How Does it Help with Bit Depth Reduction?

Dithering is a technique used to reduce the impact of bit-depth reduction when converting high-resolution audio to a lower resolution format. It works by adding a small amount of random noise to the audio signal before it is truncated to the lower bit depth.

This noise effectively masks the truncation distortion, allowing the audio to retain some of its original detail and clarity. Dithering is particularly useful when converting from a higher bit-depth format to a lower bit-depth format, as it can help to mitigate the loss of information that would otherwise occur.

How Does Bit Depth Affect Audio Quality?

The bit depth of an audio file can have a significant impact on its perceived quality. Generally speaking, higher bit-depth files can capture more detail and nuance in the audio, resulting in a more accurate and realistic reproduction of the original recording.

For example, a 16-bit audio file has a maximum dynamic range of 96 dB, while a 24-bit file has a maximum dynamic range of 144 dB. This means that a 24-bit file can capture much quieter sounds and much louder sounds than a 16-bit file, resulting in a more accurate representation of the original recording.

That being said, the impact of bit depth on perceived audio quality can vary depending on a number of factors, including the quality of the recording equipment, the mastering process, and the listening environment.

What is the Difference Between Bit Depth and Sample Rate?

While bit depth and sample rate are both important aspects of digital audio, they refer to different things. Bit depth refers to the number of bits used to represent each sample in an audio file, while sample rate refers to the number of samples per second that are taken to create the audio file.

In other words, bit depth determines the level of detail captured in each sample, while sample rate determines the temporal resolution of the audio. Both bit depth and sample rate can have an impact on the perceived quality of an audio file, and both are important considerations when working with digital audio.

What is the Best Bit Depth for Audio Production?

The best bit depth for audio production depends on a number of factors, including the specific needs of the project and the available hardware and software. In general, however, a bit depth of 24 bits is considered to be a good choice for most recording and production purposes.

This is because a 24-bit depth provides a high level of detail and dynamic range, while also being widely supported by modern recording equipment and software. That being said, there may be situations where a lower bit depth may be sufficient. For example, if the final audio product will only be distributed online or through streaming services, a 16-bit depth may be acceptable as it will still provide decent quality while reducing file size and download times. Additionally, if the recording environment is not optimal and contains a high level of background noise, a lower bit depth may actually be preferable as it can help mask the noise.

How does bit depth affect audio quality?

Bit depth plays a critical role in determining the quality of digital audio recordings. The higher the bit depth, the greater the dynamic range and level of detail that can be captured in a recording. This results in a more accurate and faithful reproduction of the original sound source. In contrast, a lower bit depth may result in a loss of detail and accuracy, leading to a less faithful reproduction of the original sound.

Can bit depth be converted after recording?

While it is possible to convert the bit depth of a digital audio file after recording, it is generally not recommended. This is because bit depth conversion can result in a loss of information and a decrease in overall audio quality. If possible, it is best to record at the desired bit depth from the start to ensure the highest possible quality.

What are some common bit depths used in digital audio?

The most common bit depths used in digital audio are 16-bit, 24-bit, and 32-bit. 16-bit is the standard for CDs and is widely used in digital audio recording for distribution on streaming platforms. 24-bit is increasingly becoming the standard for professional recording due to its high level of detail and dynamic range. 32-bit is relatively new and provides an even greater level of detail and dynamic range, but is not yet widely supported by all recording equipment and software.

Does bit depth affect the final file size of an audio recording?

Yes, bit depth does affect the final file size of an audio recording. A higher bit depth requires more data to represent each sample, resulting in larger file sizes. For example, a 24-bit audio file will be larger than a 16-bit audio file of the same duration and sample rate.

What is dithering in relation to bit depth?

Dithering is a technique used to reduce the audible effects of quantization distortion when converting from a higher bit depth to a lower bit depth. When reducing the bit depth, some of the information from the original recording must be discarded. This can result in audible distortion and noise. Dithering adds a small amount of random noise to the audio signal to mask this distortion and make it less audible.

Can different bit depths be mixed in the same audio project?

Yes, different bit depths can be mixed in the same audio project. However, it is important to note that mixing different bit depths can result in a loss of quality for the higher bit depth audio. When mixing different bit depths, it is best to convert all audio to the same bit depth before mixing to ensure the highest possible quality.

What is the relationship between bit depth and sample rate?

Bit depth and sample rate are both important factors in determining the quality of digital audio recordings. Bit depth refers to the number of bits used to represent each sample, while sample rate refers to the number of samples taken per second. Higher bit depths and sample rates result in higher quality recordings with greater detail and accuracy.

Can bit depth affect the sound of analog audio recordings?

No, bit depth does not affect the sound of analog audio recordings. Bit depth only applies to digital audio recordings.


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The bit rate directly affects the sound quality.

The bit rate directly affects the sound quality.

audio bit rate
audio bit rate

High bitrate is good and low bitrate is bad.

audio bit rate
audio bit rate

The code rate is the number of data bits transmitted per unit of time during data transmission. Generally, the unit we use is kbps, that is, kilobits per second.

The popular understanding is the sampling rate. The higher the sampling rate per unit time, the higher the precision, and the processed file is closer to the original file, but the file size is proportional to the sampling rate, so almost all encoding formats pay attention. It’s about how to use the lowest code rate to achieve the least distortion. The cbr (fixed code rate) and vbr (variable code rate) derived from this core are all articles in this regard, but things are not absolute, in terms of audio, the higher the bit rate, the lower the compressed ratio, the smaller the sound quality loss and the closer it is to the sound quality of the audio source.
The information in the computer is represented by binary 0 and 1, and each 0 or 1 is called a bit, which is represented by lowercase b, that is, bit (bit); uppercase B represents byte, ie byte, one byte = Eight bits, ie 1B=8b; the capital K in front stands for thousand, that is, thousand bits (Kb) or kilobytes (KB). Indicates the size of the file, usually using bytes (KB) to indicate the size of the file.

Kbps: The first thing to understand is that ps refers to /s, which is every second. Kbps refers to the speed of the network, that is, how many thousands of bits of information are transmitted per second (K means thousands of bits, Kb means how many thousands of bits), it is expressed in kb (kilobit), and in the case KBps means how many kilobytes are transferred per second. 1KBps = 8Kbps. The Internet speed of ADSL is 512 Kbps. If converted to bytes, it is 512/8 = 64 KBps (that is, 64 kilobytes per second).

A frame is a still image, and continuous frames form an animation, like a television image.
We normally say the number of frames. Simply put, it is the number of image frames transmitted in 1 second. It can also be understood that the graphics processor can update several times per second, usually expressed in fps (Frames Per Second). Each frame is a still image, and showing frames in rapid succession creates the illusion of movement. Higher frame rates result in smoother, more realistic animations. The more frames per second (fps), the smoother the motion is displayed.

What is the bitrate of the music?
It can also be called bit rate, which is nothing more than the amount of data reproduced per second by a type of music, the unit is expressed in bits, that is, binary bits. bps is the bit rate. b is bit, s is second, p is per, and one byte is equal to 8 binary bits. That is, the file size of a 4-minute song at 128bps is calculated as (128/8)*4*60=3840kB=3.8MB, which means that the same song with the same bit rate (bps) will not no matter what format (such as mp3 wma) The capacity is basically the same, which can only represent a transmission rate, not the sound quality. Due to different compression engines, the sound quality of different formats varies a lot. However, for the same format, the higher the bitrate, the larger the file and the better the sound quality.

What is the sample rate of the music?
Sampling rate refers to the number of samples per unit of time. The sampling rate is 44KHz, which means the number of samples per second is 44K, which means that 44,000 pieces of data are used to describe the sound waveform in 1 second. That is, the higher the sample rate, the better the sound quality. But he and bitrate are two completely different concepts.

What does the quality of an mp3 depend on? high resolution mp3

What does the quality of an mp3 depend on? high resolution mp3

high resolution mp3
high resolution mp3

Factors influencing hearing quality

high resolution mp3
high resolution mp3

High quality

Lately, very high quality audios have been promoted… are they really convenient?

We could say that if we strictly base ourselves on technical aspects, they could be considered of higher quality.

For example, they get to use sample rates of more than double the highest currently used.

The same happens with the bit rate, they use numbers that until now were not used at all.

Pewro first we must ask ourselves if the equipment we use to read them (the computer, a cell phone, an mp4 player) are capable of handling these qualities and if the speakers or headphones are also enabled and built to do the same.

Otherwise we will end up paying a lot for this super audio and effectively get the same.

It is worth additionally thinking about whether our ears could differentiate between one and the other.

To what extent our ear perceives the difference between 4800 and 96000 as a sample rate.

What we must avoid is falling victim to the “numbers”, which will show us that in theory they will sound better, but avoid touching reality – for example the human ear or the quality of our speakers – and therefore the theory ends up being misleading.

What is the difference between 128k and 320k music? Part 3

What is the difference between 128k and 320k music? Part 3

bit rate
bit rate

The sampling frequency is approximately the following depending on the type of use (k is the thousand bit symbol, 1khz=1000hz):

bit rate
bit rate

8khz – used for phones etc, is enough to record human voices.

22.05khz: transmission use frequency.

44.1kb: Audio CD.

48khz: used in DVD and digital TV.

96khz-192khz: used for DVD-Audio, Blu-ray HD, etc.

The common range of sample precision is 8 bits to 32 bits, with 16 bits generally used on CD.

Having said that, my friends are starting to get confused. It’s not the bitrate that determines the sound quality, so why is everyone saying that 320kb sound quality is better than 128kb?

【Audio Compression】

Well, in fact, the bit rate should be said to be another dimension, it is a compression of audio files.

Nowadays, most of the audio formats we use regularly are based on the original “WAV” file of the audio CD (44.1khz sample rate, 16bit sample precision, 2ch). The original recorded sound data is stored in a matrix, which is in PCM format, while WAV format is an encoding format developed by Microsoft. Its function is to reproduce the data in PCM format through encoding.

Since the data in WAV basically completely restores the PCM data, MP3, AAC and other lossless encoding formats are basically recompressed based on the WAV files. Therefore, we can simply think that WAV is the original audio format and other audio formats are compressed formats.

When it comes to compression, storage and transmission are inseparable. The purpose of compression is to improve storage and transmission. Therefore, before we talk about compression, we need to understand the basic units of computers.

We all know that the computer is a binary number system, and the files stored by the computer are made up of two numbers, 0 and 1. Therefore, the computer’s transmission is based on each number, and each number is called 1 ” bit”. For example, for an audio piece, its basic data is “0,1,1,1,0,1, 1 ,0”, and when transmitting, these numbers are transmitted one by one. The sampling precision mentioned above is this unit.

The storage unit of the computer is “byte (Byte)”. In the computer, 1 byte consists of 8 bits, that is, 8b(bit)=1B(Byte). In computer parlance, data storage is expressed in decimal and data transmission is expressed in binary, so 1KB=1024B=1024×8b. This is also part of the reason why the hard drive capacity we see does not match the actual capacity.

Go back and talk about audio compression, the bitrate of the audio is actually the compression ratio. So the bitrate really just defines the size of the file, but because under normal conditions the larger the file, the less data you lose, so the sound quality is relatively higher. However, the bit rate itself does not directly affect the quality of the file. For example, if we take a 128kb file as the source file, even if it is converted to a 320kb file, the sound quality will not be better than 128kb. .

 

What is the difference between 128k and 320k music? Part 2

What is the difference between 128k and 320k music? Part 2

bit rate
bit rate

Bit rate, sample rate, lossless, MP3, FLAC, APE, 320kb, 192kb, 128kb, 44.1khz, CBR, VBR. Does this bunch of various names make you both familiar and unknown?

bit rate
bit rate

The higher the bitrate, the better the sound quality. Lossless music is the highest sound quality, right? So, let’s start with the sound collection.

【Audio composition】

Nowadays, when we talk about audio, everything is digital audio. Digital audio consists of three parts: sample rate, sample precision, and number of sound channels.

Sample Rate: Both the sample rate, which refers to the number of samples per second when recording the sound, expressed in Hertz (Hz).

Sampling Precision: Refers to the dynamic range of the recorded sound, measured in bits (Bit).

Sound channel: the number of channels (1-8).

 

In simple terms, we can think of a sound wave as a curve. We know that the curve is made up of points, and the sampling frequency is the number of points in the middle of the length per second (the horizontal axis of the figure above). Sampling precision is the number of points in the dynamic range (upper vertical axis). The finer the positioning of these two dimensions, the greater the true sound restoration and the better the sound quality. Of course, the larger the audio file will be. The customer mentioned by the previous colleague said that the latest Hi-Res Audio format released by SONY is a 6-channel 192kHz/24-bit recorded audio file. The size of the lossless format, of course, will be more than 200 megabytes.

What is the difference between 128k and 320k music?

What is the difference between 128k and 320k music?

Bit Rate
Bit Rate

I can’t fully understand music in words.show all

Bit Rate
Bit Rate

 

【Preface】

Some time ago, a colleague came across a very troubled client. The mess was said to have been caused by the client asking him to provide song files larger than 100MB-200MB in size. And my colleagues don’t know much about audio formats, so they started endlessly fumbling about FLAC, WAV and audio size. In the end, the colleague did not clearly explain to the customer what was going on.

After that, some other things happened that made me feel that in the music industry there are too many practitioners around me who have an extremely poor understanding of music and even lack some basic knowledge related to music. I don’t even have the idea to understand, which makes me very sad. It seems that music has only one merchandise attribute, and our practitioners only need to organize the shelves, encode various merchandise, and use the big data of users’ purchase records to recommend merchandise to users, no matter why to users. they like this. features that these products have, and use cold data to provide users with various services.

Therefore, I think it is necessary to write something. I don’t expect practitioners to become people who really love music. I just hope that even if you still think of “her” as a commodity, you can first figure out what you’re selling. and what is..

PS: The content of the first lesson is about media files. Since the relevant content involves a lot of technical issues, it seems a bit boring, but if you read it carefully, you will find that it is actually very easy to understand, but this basic knowledge can be very helpful.Improve your skill well. Also expect more interesting content about records, musical styles, etc. which I will post soon.

Related Audio Attribute Part 3

Related Audio Attribute Part 3

Sample Rate
Sample Rate

How samples are combined

Sample Rate
Sample Rate

This is mainly for two-channel or multi-channel audio. For a two-channel audio, it can be combined in the following two ways:

interleaved Taking stereo as an example, a stereo audio sample is obtained by interleaving the storage of two mono samples.
flat. The samples of each channel are stored separately.

The data after FFmpeg audio decoding is stored in the AVFrame structure.

In packed format, frame.data[0] or frame.extended_data[0] contains all the audio data.
In Planar format, frame.data[i] or frame.extended_data[i] represents the data of the i-th channel (assuming channel 0 is the first), the size of the AVFrame.data array is set to 8, if If the number of channels exceeds 8, you should get the channel data from frame.extended_data.

sample format
The sample formats in FFmpeg are mainly:

copy code
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = – 1 ,
AV_SAMPLE_FMT_U8, /// < 8 bits unsigned
AV_SAMPLE_FMT_S16, /// < 16 bits
signed AV_SAMPLE_FMT_S32, /// < 32 bits
signed AV_SAMPLE_FMT_FLT, /// < float
AV_SAMPLE_FMT_DBL, /// < double

AV_SAMPLE_FMT_U8P, /// < 8 bits unsigned, flat
AV_SAMPLE_FMT_S16P, /// < 16 bits signed, flat
AV_SAMPLE_FMT_S32P, /// < 32 bits signed, flat
AV_SAMPLE_FMT_FLTP, /// < float, flat
AV_SAMPLE_FMT_DBLP, /// < double, flat
AV_SAMPLE_FMT_S64, /// < 64 bits
signed AV_SAMPLE_FMT_S64P, /// < 64 bits signed, plain

AV_SAMPLE_FMT_NB /// < Number of sample formats DO NOT USE if dynamically linked
};
copy code
to illustrate:

1. U8 (8-bit unsigned integer), S16 (16-bit integer), S32 (32-bit integer), FLT (single-precision floating-point type), DBL (double-precision floating-point type), S64 (64-bit integer), those not ending with P are interleaved structures, and those ending with P are flat structures.
2. Flat mode is FFmpeg’s internal storage mode, and the audio files we use are in packed mode.
3. The FFmpeg audio sample format that decodes different output audio formats is not the same. The test found that the data output by AAC decoding is in floating point AV_SAMPLE_FMT_FLTP format, and the data output by MP3 decoding is in AV_SAMPLE_FMT_S16P format (the mp3 file used is 16-bit deep). For the specific sample format, you can see the format member in the decoded AVFrame or the sample_fmt member in the AVCodecContext of the decoder.

Bit rate
The transfer rate per second (bit rate, also called bitrate). Like 705.6kbps or 705600bps, where b is a bit, ps is per second (per second), which means a capacity of 705600bit per second. Compressed audio files are often represented at double speed, for example CD quality MP3 is 128kbps/44100HZ. Note that the unit here is bit instead of byte. One byte is equal to 8 bits (bits). The bit is the smallest unit. It is generally used to describe network speed and various communication speeds. The byte is used to calculate the size. hard drive and memory.

Mbps is: Millionbit per second (millions of bits per second);
Kbps is: Kilobit per second (kilobit per second);
bps is: bit per second (bit per second), the corresponding conversion ratio is:

1Millionbit=1000Kilobit=1000000bit; 1Mbps = 1000,000bps; Again, this is the unit of speed, which refers to the number of bits transmitted per second. The unit of measure for data transmission speed K is the decimal meaning, but the K for data storage is the binary meaning. E.g:

The 1M bandwidth generally described is 1 Mbps = 1,000,000 bps = 1,000,000 / 8 / 1,000 = 125; therefore, the download speed of 1M bandwidth generally does not exceed 125KB/s
. 1000 = 12.5, so the maximum download rate of 100M bandwidth can reach 12.5MB/s
. Of course, the above is only the theoretical rate. In fact, the maximum download rate may not reach that much, and it is mainly affected by various losses, generally 100MB A broadband download rate of 10MB is not bad.

Related Audio Attribute Part 2

Related Audio Attribute Part 2

Sampling
Sampling

 

The higher the sampling, the more realistic and natural the sound will be.

Sampling
Sampling

 

The frequency recognition range for people is 20 HZ – 20,000 HZ. If 20,000 samples per second can be sampled, it will be enough to satisfy the needs of the human ear during playback. So 22050 The sample rate is commonly used, 44100 is already CD quality, and sampling more than 48000 is no longer meaningful to the human ear. This is similar to a 24 frames per second image from a movie.

 

Sampling bits
After sampling the audio for a sample, two steps must be performed for the sample:

1. Quantify. The quantization bits commonly used for audio quantization are:

8 bits (that is, 1 byte) can only register 256 numbers, that is, only the amplitude can be divided into 256 levels;

16 bits (ie 2 bytes) can be as small as 65536 numbers, which is already the CD standard;

32 bits (ie 4 bytes) can subdivide the amplitude into 4294967296 levels, which is really unnecessary.

The number of quantization bits is also called the number of sampling bits, bit depth, and resolution, and refers to how many levels the continuous intensity of the sound can be divided after being digitally represented. N-bit means that the intensity of the sound is divided equally into 2^N levels. 16 bits, it is level 65535. This is a very large number and people may not be able to tell the difference in sound intensity from 1/65,535. You can also say that it is the resolution of the sound card. The higher the value, the higher the resolution and the greater the ability to produce sound. The sampling multiple here is primarily addressing the strength characteristics of the signal, and the sampling rate is addressing the time (frequency) characteristics of the signal, which are two different concepts.

2. Binary encoding. That is, the result of the quantization, ie the single channel sample, is stored in a binary keyword. There are two storage methods:

Store the result of the quantization directly in the cast, that is, the two’s complement code;

The result of quantization is stored in floating point type, ie floating point encoding code.

Most PCM sample data formats use integers to store, and for some applications that require high precision, use floating point to represent PCM sample data.

frame
After the audio is quantized to a binary codeword, it must be transformed and the transformation (MDCT) is done in block units, and a block is made up of multiple (120 or 128) samples. A frame will contain one or more blocks. Common frame sizes are 960, 1024, 2048, 4096, etc. A frame records a sound unit whose duration is the product of the sample duration and the number of channels. The nb_samples in the AVFrame structure in FFmpeg represent the number of single channel audio samples in a frame.

Related Audio Attribute

Related Audio Attribute

Sample Rate
Sample Rate

channel, sample rate, sample bits, sample format, bit rate

Sample Rate
Sample Rate

 

The PCM obtained from audio sampling contains three elements: channel, sample rate, and sample rate.

channel
When people hear the sound, they can locate the sound source. By setting the sound source to different positions, a better listening experience can be created. If the position of the audio is adjusted with the image, a better audio-visual experience will be obtained. Effect. Common channels are:

monkey monkey
Two channels, stereo, the most common type, including left and right channels
2.1 channels, adding a bass channel on the basis of two channels
5.1 channels, including one front channel, one front left channel, one front right channel, one surround left channel, one surround right channel, and one bass channel, first used in early theaters
7.1 channel, on the basis of 5.1 channel, the surround left and right channels are divided into surround left and right channels and rear left and right channels, mainly used in BD and modern theaters
Next is a two-channel audio system.

 

 

Sampling rate
Audio sampling is the conversion of sound from an analog signal to a digital signal. The sample rate is the number of times the sound is collected per second and is also the number of samples per second of the resulting digital signal. When sampling sound, common sample rates are:

8,000 Hz – telephone sampling rate, sufficient for human speech
11,025 Hz – sample rate for AM radio
22,050 Hz and 24,000 Hz – sample rate for FM radio
32,000 Hz – sampling for miniDV digital camcorder, DAT (LP mode)
44,100 Hz – Audio CD, also commonly used in MPEG-1 audio (VCD, SVCD, MP3) Sample rate 47 250
Hz – Sampling frequency
48,000 Hz for commercial PCM recorders – for miniDV, digital TV, DVD, DAT, movies, and pro audio Sampling rate 50,000 Hz for 2,000 – 96,000 or 192,000 Hz digital sound
for commercial digital sound recorders
– DVD-Audio, some LPCM DVD soundtracks, BD-ROM (Blu-ray Disc) and HD-DVD (High Definition DVD) soundtracks The sample rate used by the audio track
2.8224 MHz: The sample rate used by Direct Stream Digital’s 1-bit sigma-delta modulation process.