Audio, English is AUDIO, maybe you have seen the AUDIO output or input port on the back panel of the video recorder or VCD. In this way, we can explain audio in a very popular way, as long as it is a sound that we can hear, it can be transmitted as an audio signal. The physical properties of audio are too professional, so please refer to other materials. The sound in nature is very complicated, and the waveform is extremely complicated. Usually we use pulse code modulation coding, that is, PCM coding. PCM converts continuously changing analog signals into digital codes through three steps of sampling, quantization, and coding.
1. Basic audio concepts
(1) What is the sampling rate and sampling size (bit/bit).
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, there must be two sampling points in one vibration. The highest frequency that can be felt is 20kHz. Therefore, to meet the hearing requirements of the human ear, it is necessary to sample at least 40k times per second, expressed in 40kHz, and this 40kHz is the sampling rate. Our common CD has a sampling rate of 44.1kHz. It is not enough to have frequency information. We must also obtain the energy value of this frequency and quantify it to express the signal strength. The number of quantization levels is an integer power of 2, our common CD bit 16bit sampling size, that is, 2 to the 16th power. Sampling size is more difficult to understand relative to sampling rate, because it is an abstract point, as a simple example: Suppose that a wave is sampled 8 times, and the energy values corresponding to the sampling points are A1-A8, but we only use 2bit sampling size , As a result, we can only keep the values of 4 points in A1-A8 and discard the other 4 points. If we take a sample size of 3bit, then all the information of just 8 points will be recorded. The larger the value of sampling rate and sampling size, the closer the recorded waveform is to the original signal.
2. Loss and lossless
According to the sampling rate and sample size, it can be known that relative to natural signals, audio coding can only be infinitely close at best. At least the current technology can only do this. Relative to natural signals, any digital audio coding scheme is lossy. Because it cannot be completely restored. In computer applications, the highest level of fidelity is PCM encoding, which is widely used for material preservation and music appreciation. CDs, DVDs and our common WAV files are all used. 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. We have habitually included MP3 in the category of lossy audio coding, which is relative to PCM coding. Emphasis on the relative lossy and losslessness of coding is to tell everyone that it is difficult to achieve true losslessness. It is like using numbers to express pi. No matter how high the accuracy is, it is only infinitely close, not really equal to pi. value.
3. Why use audio compression technology
To calculate 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 = 1411.2 Kbps. We often say that 128K MP3, the corresponding WAV parameter, is this 1411.2 Kbps, this parameter is also called data bandwidth, it is a concept with the bandwidth in ADSL. Divide the code rate by 8, and you can get the data rate of this WAV, which is 176.4KB/s. This means that the sampling rate for storing one second is 44.1KHz, the sampling size is 16bit, and the two-channel PCM-encoded audio signal requires 176.4KB of space, and 1 minute is about 10.34M, which is unacceptable for most users. , Especially those who like to listen to music on the computer, to reduce disk usage, there are only two ways to reduce the sampling index or compression. It is not advisable to reduce the index, so experts have developed various compression schemes. Due to different uses and target markets, the sound quality and compression ratio achieved by various audio compression encodings are different, and we will mention them one by one in the following articles. One thing is certain, they have been compressed.
4. The relationship between frequency and sampling rate
The sampling rate indicates the number of times the original signal is sampled per second. The sampling rate of audio files we commonly see is 44.1KHz. What does this mean? Suppose we have 2 segments of sine wave signals, 20Hz and 20KHz, each with a length of one second, to correspond to the lowest frequency and highest frequency we can hear, sample these two signals at 40KHz, we can get What kind of result? The result is that the 20Hz signal is sampled 40K/20=2000 times per vibration, while the 20K signal is only sampled twice per vibration. Obviously, at the same sampling rate, the low-frequency information is much more detailed than the high-frequency information. This is why some audio enthusiasts accuse the CD that the digital sound is not real enough, and the 44.1KHz sampling of the CD cannot guarantee that the high-frequency signal is well recorded. To better record high frequency signals, it seems that a higher sampling rate is required, so some friends use 48KHz sampling rate when capturing CD audio tracks, which is not advisable! This is actually not good for the sound quality. For the ripping software, maintaining the same sampling rate as the 44.1KHz provided by the CD is one of the guarantees for the best sound quality, rather than improving it. Higher sampling rates are only useful when compared to analog signals. If the signal being sampled is digital, please do not try to increase the sampling rate.
5. Flow characteristics
With the development of the Internet, people have put forward requirements for listening to music online. Therefore, it is also required that audio files can be read and played at the same time, instead of reading all the files and then replaying them, so that you can listen to them without downloading. Up. It is also possible to encode and broadcast at the same time. It is this feature that enables online live broadcast, and it becomes a reality to set up your own digital radio station.
Several supplementary concepts:
What is a divider?
The frequency divider is to distinguish the sound signals of different frequency bands, amplify them separately, and then send them to the speakers of the corresponding frequency bands for replay. When high-quality sound is reproduced, electronic frequency division processing is required. It can be divided into two types: (1) Power divider: located after the power amplifier, set in the speaker, through the LC filter network, the power audio signal output by the power amplifier is divided into bass, midrange and treble, and sent to Individual speakers. The connection is simple and easy to use, but it consumes power, audio valleys appear, and cross* distortion occurs. Its parameters are directly related to the speaker impedance, and the speaker impedance is a function of frequency, which deviates greatly from the nominal value. The error is also large, which is not conducive to adjustment. (2) Electronic frequency divider: A device that divides weak audio signals into frequency. It is located in front of the power amplifier. After the frequency is divided, a separate power amplifier is used to amplify each audio frequency band signal, and then send them to the corresponding speakers. unit. Because the current is small, it can be realized with a smaller power electronic active filter, which is easier to adjust, reducing power loss and interference between speaker units. The signal loss is small and the sound quality is good. However, this method requires an independent power amplifier for each channel, which has high cost and complex circuit structure, and is used in professional sound reinforcement systems. (From av_world)
What is an exciter?
The exciter is a harmonic generator, a sound processing device that uses the psychoacoustic characteristics of people to modify and beautify the sound signal. By adding high-frequency harmonic components to the sound and other methods, you can improve the sound quality, tone color, increase the penetration of the sound, and increase the sense of space of the sound. Modern exciters can not only create high-frequency harmonics, but also have low-frequency expansion and music style functions, making the bass effect more perfect and the music more expressive. Use exciters to improve sound clarity, intelligibility and expressiveness. Make the sound more pleasing to the ears, reduce listening fatigue, and increase loudness. Although the exciter only adds about 0.5dB of harmonic components to the sound, it actually sounds like the volume has increased by about 10dB. The auditory loudness of the sound is obviously increased, the three-dimensional feeling of the sound image, and the increase of the separation of the sound; the positioning and layering of the sound are improved, and the sound quality of the reproduced sound and the reproduction rate of the tape can be improved. Because the acoustic signal loses high-frequency harmonic components during transmission and recording, high-frequency noise appears. At this time, the former uses an exciter to compensate the signal first, and the latter uses a filter to filter out high-frequency noise, and then creates a high-pitched component to ensure the quality of the playback sound. The adjustment of the exciter requires the sound engineer to judge the sound quality and tone of the system, and then make adjustments based on subjective listening evaluation.
What is an equalizer?
Equalizer is an electronic device that can adjust the amplification of electrical signals of various frequency components separately. It compensates for the defects of speakers and sound field by adjusting electrical signals of different frequencies, compensates and modifies various sound sources and other special effects. , The equalizer on the general mixer can only adjust the high frequency, intermediate frequency, and low frequency electrical signals separately. There are three types of equalizers: graphic equalizer, parametric equalizer and room equalizer. 1. Graphic equalizer: also known as chart equalizer, through the distribution of push-pull keys on the panel, it can intuitively reflect the equalization compensation curve that is called up, and the increase and attenuation of each frequency are clear at a glance. It uses constant Q technology, each frequency The point is equipped with a push-pull potentiometer, no matter whether a certain frequency is increased or attenuated, the frequency bandwidth of the filter is always the same. The commonly used professional graphic equalizer divides the 20Hz~20kHz signal into 10 segments, 15 segments, 27 segments, and 31 segments for adjustment. In this way, people choose frequency equalizers with different numbers of segments according to different requirements. Generally speaking, the frequency points of the 10-band equalizer are distributed in octave intervals. In general, the 15-band equalizer is a 2/3-octave equalizer, and when used in professional sound reinforcement, the 31-band equalizer is 1 The /3-octave equalizer is mostly used in more important occasions where fine compensation is required. The graphic equalizer has a simple structure and is intuitive and clear, so it is widely used in professional audio. 2. Parametric equalizer: also known as a parametric equalizer, an equalizer that can finely adjust various parameters of the equalization adjustment. It is mostly attached to the mixer, but there is also an independent parametric equalizer. The adjusted parameters include frequency bands and frequency points. , Gain and quality factor Q value, etc., can beautify (including ugly) and modify the sound, make the sound (or music) style more distinctive and colorful, and achieve the desired artistic effect. 3. Room equalizer is an equalizer used to adjust the frequency response characteristic curve in the room. Due to the different absorption (or reflection) of different frequencies by decorative materials and the influence of normal resonance, it is necessary to use a room equalizer to The frequency defects in sound construction should be objectively compensated and adjusted. The finer the frequency band, the sharper the adjusted peak, that is, the higher the Q value (quality factor), the finer the compensation during adjustment. The thicker the frequency band, the wider the adjusted peak. When the sound field transmission frequency characteristic curve is more complicated It is difficult to compensate at times.
What is a compression limiter?
Compression limiter is a collective term for compressor and limiter. It is a processing device for audio signals, which can compress or restrict the dynamics of audio electrical signals. The compressor is a variable gain amplifier, and its amplification factor (gain) can automatically change with the strength of the input signal, which is inversely proportional. When the input signal reaches a certain level (the threshold is also called the critical value), the output signal increases with the increase of the input signal. This situation is called Compressor; if it does not increase, it is called Limiter. In the past, the compressor used Hard-knee technology, and the input signal reached the threshold as soon as the input signal reached the threshold. The gain is immediately reduced, so that there will be a dynamic sudden change of the signal at the inflection point (the turning point of the gain change), which makes the human ear clearly feel that the strong signal is suddenly compressed. In order to solve this shortcoming, the modern new compressor adopts soft-knee technology. The compression ratio change of this compressor before and after the threshold is balanced and gradual, making the compression change difficult to detect, and the sound quality is further improved. . The compressor can maintain a certain balance between the volume of the instrument and the singer during the recording process; ensure the balance of various signal strengths. Sometimes it is also used to eliminate the vocalists of singers, or to change the compression and release time to produce the special effect of "reversal sound" in which the sound changes from small to large. In the broadcasting system, it is used to compress the program signal with a larger dynamic range to increase the average emission level under the premise of preventing modulation distortion and preventing transmitter overload. In the sound reinforcement system of the dance hall, the compressor compresses the signal while maintaining the original program style, reducing the dynamics of the music to meet the requirements of the sound reinforcement system and artistic activities. Although the compressor has many uses, modern compressors generally adopt new technologies such as soft knees, which can further reduce the side effects of the compressor of the compressor, but it does not mean that the compressor does not destroy the sound quality. Re-existed. Therefore, in the sound reinforcement system, do not abuse the limiter, even if you want to use it, you should use the reducer to process the signal with caution. This is not only a need to protect power amplifiers and speakers, but also a need to improve sound quality.
What is the signal-to-noise ratio (S/N)?
The signal-to-noise ratio refers to the signal power at a reference point in the line and the inherent noise power when there is no signal
The ratio is expressed in decibels (dB). The higher the value, the better, which means less noise.
What is decibel
Decibel (dB) is a standard unit that expresses relative power or amplitude level. Expressed in dB. The larger the decibel number, the louder the sound emitted. In calculation, every 10 decibels increase in decibels, the sound level will be approximately ten times the original.
dB: deciBel decibel. It is used to express the relative level of two voltages, powers or sounds.
dBm: A variant of decibels, 0dB = 1mW into 600 Ohms
dBv: A variant of decibels, 0dB = 0.775 volts.
dBV: A variant of decibels, 0dB = 1 volt.
dB/Octave: decibel/octave. The expression of the slope of the filter, the larger the number of decibels per octave, the steeper the slope.
This concept is relatively complicated, we use physics calculations to illustrate:
In order to express the strength of the sound, people introduced the concept of "sound intensity", and measured its magnitude by the amount of sound energy passing through a unit area vertically in 1 second. The sound intensity is represented by the letter "I", and its unit is "Watts/m2". According to the regulations, if the sound energy perpendicular to the unit area is doubled within 1 second, the sound intensity will also double. Therefore, the sound intensity is an objective physical quantity that does not change with people's feelings.
Although sound intensity is an objective physical quantity, there is a very big difference between the magnitude of sound intensity and the sound intensity that people subjectively feel.In order to conform to people's subjective perception of sound intensity, the concept of "sound intensity level" has been introduced in physics. The decibel is a unit of sound intensity level, which is one-tenth of the bell.
How is the sound intensity level regulated? What does it have to do with sound intensity?
The measurement proves that the human ear has different sensitivity to sound waves of different frequencies. It is most sensitive to 3000 Hz sound waves. As long as the sound intensity of this frequency reaches I0 = 10-12 watts/m2, it can cause hearing in the human ear. The sound intensity level is specified based on the minimum sound intensity I0 that can be heard by the human ear, and the sound intensity of I0 = 10-12 watts/m2 is specified as the zero-level sound intensity, that is to say the sound intensity at this time The level is zero bels (also zero decibels). When the sound intensity doubles from I0 to 2I0, the sound intensity felt by the human ear does not double. Only when the sound intensity reaches 10I0, the human ears feel the sound intensity doubled. The sound intensity level corresponding to this sound intensity is 1 beel = 10 decibels; when the sound intensity becomes 100I0, the human ears feel the sound strong Weak increases by 2 times, the corresponding sound intensity level is 2 Bel=20 decibels; when the sound intensity becomes 1000I0, the sound intensity felt by the human ear increases by 3 times, and the corresponding sound intensity level is 3 Bel=30 decibels. So on and so forth. The maximum sound intensity that the human ear can withstand is 1 watt/m2 = 1012I0, and its corresponding sound intensity level is 12 bels = 120 decibels.
Formula: Sound pressure level (dB) = 20Lg (measured sound pressure/reference sound pressure value)
Old fish's note: When the measured sound pressure is the same as the reference sound pressure, the calculated result after taking the logarithm is 0dB. On analog audio equipment, it can be greater than 0dB, but digital equipment does not. Digital calculation requires a measurement, and there is no infinite value. Therefore, in the digital equipment and software we use, 0dB has become a reference standard value.
2. Introduction to common audio formats and players
The characteristics and adaptability of mainstream audio formats
All kinds of audio coding have their technical characteristics and applicability in different occasions. Let's roughly explain how to apply these audio coding flexibly.
4-1 PCM encoded WAV
As mentioned earlier, the PCM encoded WAV file is the format with the best sound quality. Under the Windows platform, all audio software can provide support for her. There are many functions in WinAPI provided by Windows that can directly play wav. Therefore, when developing multimedia software, wav is often used in large quantities for event sound effects and background music. PCM encoded wav can achieve the best sound quality under the same sampling rate and sample size, so it is also widely used in audio editing, non-linear editing and other fields.
Features: The sound quality is very good, supported by a large number of software.
Applicable to: multimedia development, preservation of music and sound effect materials.
4-2 MP3
MP3 has a good compression ratio. The mid-to-high bit rate mp3 encoded by LAME is very close to the original WAV file in terms of sound. Using appropriate parameters, LAME encoded MP3 is very suitable for music appreciation. Since MP3 has been introduced for a long time, coupled with fairly good sound quality and compression ratio, many games also use mp3 for event sound effects and background music. Almost all well-known audio editing software also provide support for MP3, you can use mp3 like wav, but because mp3 encoding is lossy, the sound quality will drop sharply after multiple editing, and mp3 is not suitable for saving material. But the demo as a work is really excellent. The long history and good sound quality of mp3 make it one of the most widely used lossy encodings. A large number of mp3 resources can be found on the Internet, and mp3player is becoming a fashion day by day. Many VCDPlayer, DVDPlayer and even mobile phones can play mp3, and mp3 is one of the best supported encodings. MP3 is also not perfect, and it does not perform well at lower bit rates. MP3 also has the basic characteristics of streaming media and can be played online.
Features: Good sound quality, relatively high compression ratio, supported by a large amount of software and hardware, and widely used.
Suitable for: Suitable for music appreciation with higher requirements.
4-3 OGG
Ogg is a very promising code, which has amazing performance at various bit rates, especially at low and medium bit rates. In addition to its good sound quality, Ogg is also a completely free codec, which lays the foundation for more support for Ogg. Ogg has a very good algorithm that can achieve better sound quality with a smaller bit rate. The 128kbps Ogg is even better than the 192kbps or even higher bitrate mp3. Ogg's treble has a certain metallic taste, so this defect of Ogg will be exposed when coding some solo instruments with high requirements for high frequencies. OGG has the basic characteristics of streaming media, but there is no media service software support, so digital broadcasting based on ogg is not yet possible. Ogg's current state of being supported is not good enough, no matter it is software or hardware, it can't be compared with mp3.
Features: It can achieve better sound quality than mp3 with a smaller bit rate than mp3, and it has good performance under high, medium and low bit rates.
Apply to: Use smaller storage space to get better sound quality (relative to MP3)
4-4 MPC
Like OGG, MPC’s competitor is also mp3. At medium and high bitrates, MPC can achieve better sound quality than competitors. At medium bitrates, MPC's performance is not inferior to Ogg. At high bitrates, MPC’s The performance is even more desperate. The sound quality advantage of MPC is mainly manifested in the high frequency part. The high frequency of MPC is much more delicate than MP3, and it does not have the metallic taste of Ogg. It is currently the most suitable lossy encoding for music appreciation. Because they are all new codes, they are similar to Ogg's experience, and they lack extensive software and hardware support. MPC has good coding efficiency, and the coding time is much shorter than OGG and LAME.
Features: Under medium and high bit rates, it has the best sound quality performance in lossy encoding, and under high bit rates, it has excellent high frequency performance.
Applicable to: music appreciation with the best sound quality under the premise of saving a lot of space.
4-6 WMA
The WMA developed by Microsoft is also loved by many friends. At low bit rates, it has a much better sound quality than mp3. The emergence of WMA immediately eliminated the once-popular VQF encoding. WMA with a Microsoft background has received good software and hardware support. Windows Media Player can play WMA and listen to digital radio stations based on WMA encoding technology. Because the player exists on almost every PC, more and more music websites are willing to use WMA as the first choice for online audition. In addition to the good support environment, WMA also has a very good performance at 64-128kbps bit rate. Although many friends with higher requirements are not satisfied, more friends with lower requirements have accepted this encoding. WMA is very The popularity is coming soon.
Features: Sound quality performance at low bitrates is hard to beat
Applicable to: digital radio setup, online audition, music appreciation under low requirements
4-7 mp3PRO
As an improved version of mp3, mp3PRO shows very good quality, full of treble, although mp3PRO is inserted in the playback process through SBR technology, but the actual listening experience is quite good, although it seems a bit thin, but it is already in the world of 64kbps There is no rival, even more than 128kbps mp3, but unfortunately, the low-frequency performance of mp3PRO is as broken as mp3. Fortunately, the high-frequency interpolation of SBR can more or less cover up this defect, so mp3PRO On the contrary, the low frequency weakness of WMA is not as obvious as that of WMA. You can feel deeply when you use the PRO switch of RCA mp3PRO Audio Player to switch between PRO mode and normal mode. Overall, the 64kbps mp3PRO has reached the sound quality level of the 128kbps mp3, with a slight win in the high frequency part.
Features: the king of sound quality at low bitrates
Suitable for: music appreciation under low requirements
4-8 APE
A new type of lossless audio coding that can provide a compression ratio of 50-70%. Although it is not worth mentioning compared to lossy coding, it is a great boon for friends who are pursuing perfect attention. APE can be truly lossless, rather than sound lossless, and the compression ratio is better than similar lossless formats.
Features: The sound quality is very good.
Suitable for: the highest quality music appreciation and collection.
3, audio signal encoding processing
(1) PCM encoding
PCM Pulse Code Modulation is the abbreviation of Pulse Code Modulation. In the previous text, we mentioned the general workflow of PCM. We don't need to care about the calculation method used in the final encoding of PCM. We only need to know the advantages and disadvantages of the PCM encoded audio stream. The biggest advantage of PCM encoding is good sound quality, and the biggest disadvantage is its large size. Our common Audio CD uses PCM encoding, and the capacity of a CD can only hold 72 minutes of music information.
As we all know, no matter how powerful the current multimedia computers are, they can only process digital information inside. The sounds we hear are all analog signals. How can the computer also process these sound data? Also, what is the difference between analog audio and digital audio? What are the advantages of digital audio? These are what we are going to introduce below.
Converting analog audio to digital audio is called sampling in computer music. The main hardware device used in the process is the Analog to Digital Converter (ADC). The sampling process actually converts the electrical signal of the usual analog audio signal into a number of binary codes called "Bit" 0 and 1, these 0 and 1 constitute a digital audio file. As shown in the figure below, the sine curve in the figure represents the original audio curve; the colored square represents the result obtained after sampling. The more consistent the two, the better the sampling result.
The abscissa in the above figure is the sampling frequency; the ordinate is the sampling resolution. The grids in the picture are gradually encrypted from left to right, first increasing the density of the abscissa, and then increasing the density of the ordinate. Obviously, when the unit of the abscissa is smaller, that is, the interval between the two sampling moments is smaller, it is more conducive to maintaining the true condition of the original sound. In other words, the higher the sampling frequency, the more guaranteed the sound quality; similarly, when the vertical The smaller the coordinate unit is, the better the sound quality is, that is, the larger the number of sampling bits, the better.
Please pay attention to one point. 8-bit (8Bit) does not mean that the ordinate is divided into 8 parts, but 2^8=256 parts; the same way, 16-bit means that the ordinate is divided into 2^16=65536 parts; while 24 bits are divided into 2^16=65536 parts. Divide into 2^24=16777216 parts. Now let's perform a calculation to see how large the data volume of a digital audio file is. Suppose we use 44.1kHz, 16bit for stereo (that is, two channels)
(2) WAVE
This is an ancient audio file format developed by Microsoft. WAV is a file format that conforms to the PIFF Resource Interchange File Format specification. All WAVs have a file header, which is the encoding parameter of the audio stream. WAV has no hard and fast rules on the encoding of audio streams. In addition to PCM, almost all encodings that support the ACM specification can encode WAV audio streams. Many friends do not have this concept. Let's take AVI as a demonstration, because AVI and WAV are very similar in file structure, but AVI has one more video stream. There are many kinds of AVIs that we come into contact with, so we often need to install some Decode to watch some AVIs. DivX that we come into contact with is a kind of video encoding. AVI can use DivX encoding to compress video streams. Of course, other ones can also be used. Encoding compression. Similarly, WAV can also use a variety of audio encodings to compress its audio stream, but we are usually WAV whose audio stream is encoded by PCM, but this does not mean that WAV can only use PCM encoding. MP3 encoding can also be used in WAV. Like AVI, as long as the corresponding Decode is installed, you can enjoy these WAVs.
Under the Windows platform, WAV based on PCM encoding is the best supported audio format, and all audio software can perfectly support it. Because it can achieve higher sound quality requirements, WAV is also the preferred format for music editing and creation. Suitable for saving music material. Therefore, WAV based on PCM encoding is used as an intermediary format and is often used in the mutual conversion of other encodings, such as converting MP3 to WMA.
(3) MP3 encoding
As the most popular audio compression format, MP3 is widely accepted by everyone. Various software products related to MP3 are emerging in an endless stream, and more hardware products have begun to support MP3. There are many VCD/DVD players that we can buy. Can support MP3, there are more portable MP3 players, etc. Although several major music companies are extremely disgusted with this open format, they cannot prevent the survival and spread of this audio compression format. MP3 has been in development for 10 years. It is the abbreviation of MPEG (MPEG: Moving Picture Experts Group) Audio Layer-3, which is a derivative coding scheme of MPEG1. It was successfully developed in 1993 by the Fraunhofer IIS Research Institute in Germany and Thomson. MP3 can achieve an amazing compression ratio of 12:1 and maintain basic audible sound quality. In the days when hard disks were so expensive that year, MP3 was quickly accepted by users. With the popularity of the Internet, MP3 was accepted by hundreds of millions of users. The initial release of MP3 coding technology was actually very imperfect. Due to the lack of research on sound and human hearing, the early mp3 encoders were almost all coded in a crude way, and the sound quality was seriously damaged. With the continuous introduction of new technologies, mp3 encoding technology has been improved one after another, including two major technical improvements.
VBR: The MP3 format file has an interesting feature, that is, it can be read while playing, which is also in line with the most basic characteristics of streaming media. That is to say, the player can play without pre-reading the entire content of the file, where it is read, even if the file is partially damaged. Although mp3 can have a file header, it is not very important for mp3 format files. Because of this feature, each segment and frame of the MP3 file can have a separate average data rate without special decoding schemes. So there is a technology called VBR (Variable bitrate, dynamic data rate), which allows each segment or even each frame of the MP3 file to have a separate bitrate. The advantage of this is to ensure the sound quality.
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