1. Audio
Refers to sound waves with a frequency between 20 Hz and 20 kHz that can be heard by human ears.
If you add a corresponding audio card to the computer, which is what we often call a sound card, we can record all the sounds, and the acoustic characteristics of the sound, such as the level of the sound, can be stored as files on the computer's hard disk. Conversely, we can also use a certain audio program to play the stored audio file to restore the previously recorded sound.
2. Sampling frequency
Refers to the number of sound samples obtained per second. Sound is actually a kind of energy wave, so it also has the characteristics of frequency and amplitude. The frequency corresponds to the time axis and the amplitude corresponds to the level axis. The wave is infinitely smooth, and the string can be regarded as composed of countless points. Because the storage space is relatively limited, the points of the string must be sampled during the digital encoding process.
The sampling process is to extract the frequency value of a certain point. Obviously, the more points are extracted in one second, the more frequency information is obtained. In order to restore the waveform, the higher the sampling frequency, the better the sound quality. The more real the restoration is, but at the same time it occupies more resources. Due to the limited resolution of the human ear, too high a frequency cannot be distinguished. The sampling frequency of 22050 is commonly used, 44100 is already CD sound quality, and sampling over 48,000 or 96,000 is no longer meaningful to the human ear. This is similar to the 24 frames per second in movies. If it is stereo, the sample is doubled and the file is almost doubled.
According to the Nyquist sampling theory, in order to ensure that the sound is not distorted, the sampling frequency should be around 40kHz. We don’t need to know how this theorem came about. We only need to know that this theorem tells us that if we want to record a signal accurately, our sampling frequency must be greater than or equal to twice the maximum frequency of the audio signal. Remember, it is the maximum frequency. .
In the field of digital audio, commonly used sampling rates are:
8000 Hz-the sampling rate used by the phone, which is sufficient for human speech
11025 Hz-Sampling rate used by the phone
22050 Hz-sampling rate used for radio broadcasting
32000 Hz-sampling rate used by miniDV digital video camcorder, DAT (LP mode)
44100 Hz-Audio CD, also commonly used in the sampling rate of MPEG-1 audio (VCD, SVCD, MP3)
47250 Hz-sampling rate used by commercial PCM recorders
48000 Hz-sampling rate for digital sound used in miniDV, digital TV, DVD, DAT, movies, and professional audio
50000 Hz-sampling rate used by commercial digital recorders
96000 Hz or 192000 Hz-the sampling rate used for DVD-Audio, some LPCM DVD audio tracks, BD-ROM (Blu-ray Disc) audio tracks, and HD-DVD (High Definition DVD) audio tracks
3. the number of sampling bits
The number of sampling bits is also called the sampling size or the number of quantization bits. It is a parameter used to measure the fluctuation of the sound, that is, the resolution of the sound card or can be understood as the resolution of the sound card processed by the sound card. The larger the value, the higher the resolution, and the more realistic the sound recorded and played back. The bit of the sound card refers to the binary digits of the digital sound signal used by the sound card when collecting and playing sound files. The bit of the sound card objectively reflects the accuracy of the digital sound signal's description of the input sound signal. Common sound cards are mainly 8-bit and 16-bit. Nowadays, all mainstream products on the market are 16-bit and above sound cards.
The amplitude of each sampled data is recorded, and the sampling accuracy depends on the number of sampling bits:
1 byte (that is, 8bit) can only record 256 numbers, that is, the amplitude can only be divided into 256 levels;
2 bytes (that is, 16 bits) can be as small as 65536, which is already a CD standard;
4 bytes (that is, 32bit) can subdivide the amplitude into 4294967296 levels, which is really unnecessary.
4. the number of channels
That is the number of sound channels. Common mono and stereo (dual-channel) have now developed to four-sound surround (four-channel) and 5.1 channels.
(1) Single Road
Mono is a relatively primitive form of sound reproduction, and early sound cards used it more commonly. Mono sound can only be sounded using one speaker, and some are also processed into two speakers to output the same sound channel. When monophonic information is played back through two speakers, we can clearly feel that the sound is from two speakers. It is impossible to determine the specific location of the sound source that is transmitted to our ears from the middle of the speaker.
(2) Stereo
Binaural channels have two sound channels. The principle is that when people hear a sound, they can judge the specific location of the sound source based on the phase difference between the left and right ears. The sound is allocated to two independent channels during the recording process, so as to achieve a good sound localization effect. This technique is particularly useful in music appreciation. The listener can clearly distinguish the direction from which various instruments come from, which makes the music more imaginative and closer to the on-site experience.
Two voices are currently the most common use. In karaoke, one is for playing music and the other is for singer's voice; in VCD, one is dubbing in Mandarin and the other is dubbing in Cantonese.
(3) Four-tone surround
The four-channel surround defines four sound points, front left, front right, rear left, and rear right, and the audience is surrounded by these four sound points. At the same time, it is also recommended to add a subwoofer to enhance the playback processing of low-frequency signals (this is the reason why 4.1-channel speaker systems are popular today). As far as the overall effect is concerned, the four-channel system can bring the listeners surround sound from multiple different directions, can obtain the auditory experience of being in a variety of different environments, and give users a brand-new experience. Nowadays, four-channel technology has been widely integrated into the design of various mid-to-high-end sound cards, becoming the mainstream trend of future development.
(4) channel
5.1 channels have been widely used in various traditional theaters and home theaters. Some of the more well-known sound recording compression formats, such as Dolby AC-3 (Dolby Digital), DTS, etc., are based on the 5.1 sound system. The ".1" channel is a specially designed subwoofer channel that can produce subwoofers with a frequency response range of 20 to 120 Hz. In fact, the 5.1 sound system comes from 4.1 surround, the difference is that it adds a center unit. This center unit is responsible for transmitting the sound signal below 80Hz, which is helpful to strengthen the human voice when watching the film, and concentrate the dialogue in the middle of the entire sound field to increase the overall effect.
At present, many online music players, such as QQ Music, have provided 5.1-channel music for trial listening and downloading.
5. frame
The concept of audio frames is not as clear as video frames. Almost all video encoding formats can simply think of a frame as an encoded image. However, the audio frame is related to the encoding format, which is implemented by each encoding standard. Because if it is PCM (unencoded audio data), it does not need the concept of frames at all, and it can be played according to the sampling rate and sampling accuracy. For example, for dual audio with a sampling rate of 44.1kHZ and a sampling accuracy of 16 bits, you can calculate that the bit rate is 44100*16*2bps, and the audio data per second is a fixed 44100*16*2/8 bytes.
The amr frame is relatively simple. It stipulates that every 20ms of audio is a frame, and each frame of audio is independent. It is possible to use different encoding algorithms and different encoding parameters.
The mp3 frame is a bit more complicated and contains more information, such as sampling rate, bit rate, and various parameters.
6. cycle
The number of frames required for one processing by an audio device is used as a unit for data access and audio data storage by the audio device.
7. interlaced mode
The storage method of digital audio signals. The data is stored in continuous frames, that is, the left channel samples and right channel samples of frame 1 are recorded first, and then the recording of frame 2 is started.
8. non-interlaced mode
First, record the left channel samples of all frames in a period, and then record all the right channel samples.
9. bit rate
The bit rate is also called the bit rate, which refers to the amount of data played by music per second, and the unit is expressed in bits, which are binary bits. bps is the bit rate. b is bit (bit), s is second (second), p is every (per), one byte is equivalent to 8 binary bits. That is to say, the file size of a 4-minute song of 128bps is calculated like this (128/8)*4*60=3840kB=3.8MB, 1B (Byte)=8b (bit), generally mp3 is beneficial at around 128 bit rate , It is also about 3-4 BM in size.
In computer applications, the highest fidelity level is PCM encoding, which is widely used for material storage and music appreciation. It is used in CDs, DVDs and our common WAV files. Therefore, PCM has become a lossless encoding by convention, because PCM represents the best fidelity level in digital audio. It does not mean that PCM can ensure the absolute fidelity of the signal. PCM can only achieve the greatest degree of infinite proximity.
Calculating the bit rate of a PCM audio stream is a very easy task, sampling rate value × sampling size value × channel number bps. A WAV file with a sampling rate of 44.1KHz, a sampling size of 16bit, and dual-channel PCM encoding, its data rate is 44.1K×16×2 = 1411.2Kbps. Our common Audio CD uses PCM encoding, and the capacity of a CD can only hold 72 minutes of music information.
A dual-channel PCM encoded audio signal requires 176.4KB of space in 1 second, and about 10.34M in 1 minute. This is unacceptable for most users, especially those who like to listen to music on the computer. Disk occupancy, there are only two methods, downsampling index or compression. It is not advisable to reduce the sampling index, so experts have developed various compression schemes. The most original are DPCM, ADPCM, and the most famous is MP3. Therefore, the code rate after data compression is much lower than the original code.
