The sounds in nature are very complex and the waveforms are extremely complex. Usually we use pulse coded modulation coding, ie PCM coding. PCM encoding converts continuously changing analog signals into digital signals through three steps of sampling, quantization, and encoding.
Sampling
The digital audio system reproduces the original sound by converting the acoustic waveform into a series of binary data (the original sound is an analog signal). The device used to implement this step is an analog-to-digital converter (A/D converter, or ADC, or Analog to digital convert). It samples the sound waves at a rate of tens of thousands of times per second. Each sample records the state of the original simulated sound wave at a certain moment, called the sample. By connecting a series of samples, you can describe a sound wave. The number of samples sampled per second is called the sampling frequency or recovery rate, and the unit is HZ (Hertz). The higher the sampling frequency, the higher the frequency of the sound wave that can be described. The sampling rate determines the range of the sound frequency (equivalent to the pitch) and can be represented by a digital waveform. The range of frequencies represented by waveforms is often referred to as bandwidth. To correctly understand the audio samples can be divided into the number of bits sampled and the frequency of the sample.
Number of samples (sampling accuracy)
The sound files in the computer are represented by the numbers 0 and 1. So the essence of recording on a computer is to convert an analog sound signal into a digital signal. Conversely, during playback, the digital signal is restored to an analog sound signal output. The number of samples can be understood as the resolution of the sound processed by the capture card. The larger the value, the higher the resolution and the more realistic the sound of recording and playback. The bit of the capture card refers to the binary digit of the digital sound signal used by the capture card when acquiring and playing the sound file. The bits of the capture card objectively reflect the accuracy of the digital sound signal's description of the input sound signal. The 8 bits represent the 8th power of 2 - 256, and the 16 bits represent the 16th power of 2 - 64K.
Sample rate
The number of times the media object is sampled per unit time, in Hz. The sampling frequency refers to the number of times the recording device samples the sound signal in one second. The higher the sampling frequency, the more realistic the sound is restored. In today's mainstream acquisition cards, the sampling frequency is generally divided into 22.05KHz, 44.1KHz (44100Hz), 48KHz three levels, 22.05 KHz can only achieve the sound quality of FM broadcast, 44.1KHz is the theoretical CD sound quality limit, 48KHz is more precise. For the sampling frequency higher than 48KHz, the human ear can not be distinguished, so there is not much use value on the computer.
The sampling rate of 5 kHz can only achieve the sound quality of people's speech.
The 11 kHz sampling rate is the lowest standard for playing small sounds and is a quarter of the CD sound quality.
The 22 kHz sampling rate can achieve half the sound quality of the CD, and most websites currently use this sampling rate.
The 44kHz sampling rate is a standard CD sound quality that achieves good hearing.
Number of channels
Divided into mono mono; stereo stereo. Of course there are more channels. Give a list, the channel is more, the effect is good, two channels, indicating that only the left and right sides have sounds coming over, four channels, indicating that the front and rear sounds are transmitted to the bit rate (bitrate)
Also called the code rate. For the encoding format, the amount of audio data per second after compression encoding is indicated. Calculation formula: bit rate = sampling rate x sampling accuracy x number of channels. Unit kbps, where k is 1000
VBR, ABR, CBR
VBR (Variable Bitrate) dynamic bit rate. That is, there is no fixed bit rate, and the compression software instantly determines what bit rate to use based on the audio data during compression. This is the algorithm developed by Xing. They encode the complex part of a song with high Bitrate and the simple part with low Bitrate. Although the idea is good, unfortunately, the VBR algorithm of the Xing encoder is very poor, and the sound quality is far from CBR. Fortunately, Lame perfectly optimized the VBR algorithm to make it the best encoding mode for MP3. This is the recommended encoding mode when the quality is the premise of the file size.
ABR (Average Bitrate) is an interpolation parameter of VBR. Lame created this encoding mode for the poor file size ratio of CBR and the variable size of VBR file generation. ABR is also known as "Safe VBR", which is a segment of every 50 frames (30 frames, about 1 second) in a specified average Bitrate. Low-frequency and insensitive frequencies use relatively low traffic, high frequency and large dynamic performance. Use high traffic. For example, when specifying a wav file with a 192 kbps ABR, Lame will fix 85% of the file with 192 kbps and then dynamically optimize the remaining 15%: the complex part is encoded with more than 192 kbps, the simple part Coded with less than 192 kbps. Compared with the 192kbps CBR, the 192kbps ABR is similar in file size, and the sound quality is improved a lot. The ABR code is 2 to 3 times faster than the VBR code in the speed, and the quality is better than the CBR in the range of 128-256 kbps. Can be used as a compromise between VBR and CBR.
CBR (Constant Bitrate), a constant bit rate, means that the file is a bit rate from beginning to end. Compared to VBR and ABR, the file it compresses is very large, but the sound quality is not significantly improved.
Lossless and lossless According to the process of sampling and quantification, audio coding can only achieve signals that are infinitely close to the natural world, at least the current technology is not possible to make it exactly the same. This is because the signals in nature are continuous, and the audio encoded values ​​are discrete. Therefore, any digital audio coding scheme is lossy, which means that no audio can completely restore the sound of nature.
In computer applications, PCM coding can achieve the highest level of fidelity. It has been widely used for material preservation and music appreciation, including CD, DVD and WAV files. Therefore, the PCM convention has become lossless coding, but this does not mean that the PCM can ensure that the signal is absolutely fidelity, and the PCM can only achieve maximum infinite proximity.
We habitually include MP3 in the category of lossy audio coding, which is relatively PCM encoded.
Emphasis on the relative loss of coding is lossy and non-destructive, and it is very difficult or even impossible to achieve true loss. Just like, we use decimals to express the pi, no matter how high the precision of the decimal, we can only approach infinitely, rather than the value that is really equal to the pi.
Why use audio compression technology to calculate the bitrate of a PCM audio stream is a very easy thing, sample rate value &TImes; sample size value &TImes; channel number bps. A sampling rate of 44.1 kHz, a sampling size of 16 bits, a two-channel PCM encoded WAV file, its data rate is 44.1K & TImes; 16 & TImes; 2 = 1411.2 Kbps. We often say that 128K MP3, the corresponding WAV parameter, is this 1411.2Kbps. This parameter is also called data bandwidth. It is a concept with bandwidth in ADSL. By dividing the code rate by 8, you can get the data rate of this WAV, which is 176.4KB/s. This means that the one second sampling rate is 44.1KHz, the sampling size is 16bit, and the two-channel PCM encoded audio signal requires 176.4KB of space, which is about 10.34M in one minute, which is unacceptable to most users. Especially for those who like to listen to music on the computer, to reduce the disk usage, there are only 2 ways to reduce the sampling index or compression. It is not advisable to reduce the indicators, so experts have developed various compression schemes. Because the purpose and target market are different, the audio quality and compression ratio achieved by various audio compression codes are different, which we will mention in the following articles. One thing is for sure, they are all compressed.
Frequency vs. Sample Rate The sample rate represents the number of times the original signal is sampled per second. The common sample rate of audio files we have is 44.1KHz. What does this mean? Suppose we have 2 sine wave signals, 20Hz and 20KHz, each of which is one second in length, corresponding to the lowest frequency and the highest frequency we can hear. We respectively sample 40KHz of these two signals, we can get What kind of result? The result is that the 20 Hz 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, recording low frequency information is much more detailed than high frequency. This is also why some audiophiles accuse CDs of having digital sounds that are not real enough. The 44.1KHz sampling of CDs does not guarantee that high-frequency signals are better recorded. To record high-frequency signals better, it seems that a higher sampling rate is required, so some friends use a sampling rate of 48KHz when capturing CD tracks, which is not desirable! This actually has no benefit to the sound quality. For the tracking software, keeping the same sampling rate as the 44.1KHz provided by the CD is one of the guarantees of the best sound quality, instead of improving it. Higher sample rates are only useful when compared to analog signals. If the sampled signal is digital, do not try to increase the sample rate.
PCM coded PCM pulse code modulation is an abbreviation of Pulse CodeModulation. In the previous text, we mentioned the general workflow of PCM. We don't need to care about the calculation method used by PCM final encoding. We only need to know the advantages and disadvantages of PCM encoded audio stream. The biggest advantage of PCM coding is that the sound quality is good, and the biggest disadvantage is the large size. Our common AudioCD uses PCM encoding, and the capacity of a single disc can only hold 72 minutes of music information.
WAVE
This is an old audio file format developed by Microsoft. WAV is a file format that conforms to the PIFF Resource Interchange FileFormat specification. All WAVs have a file header, the encoding parameter of the header audio stream. WAV does not have a hard code for encoding audio streams. In addition to PCM, almost all encodings that support the ACM specification can encode WAV audio streams. Many friends don't have this concept. Let's take AVI as a demonstration, because AVI and WAV are very similar in file structure, but AVI has more video streams. There are many kinds of AVIs that we come into contact with, so we often need to install some Decode to watch some AVIs. We have more DivXs as a kind of video encoding. AVI can use DivX encoding to compress video streams. Of course, other ones can be used. Encoding compression. Similarly, WAV can also use a variety of audio encoding to compress its audio stream, but we all commonly use WAVs whose audio streams are PCM encoded, but this does not mean that WAV can only use PCM encoding, and MP3 encoding can also be used in WAV. In the same way as AVI, you can enjoy these WAVs as long as the corresponding Decode is installed.
Under the Windows platform, WAV based on PCM encoding is the best supported audio format. All audio software can be perfectly supported. Because it can achieve high sound quality requirements, WAV is also the preferred format for music editing creation. Suitable for saving music material. Therefore, PCM-based WAV is used as an intermediary format, often used in the conversion of other encodings, such as MP3 to WMA.
MP3 encoding MP3 is currently the most popular audio compression format, which is widely accepted by everyone. Various MP3 related software products are emerging one after another, and more hardware products are also beginning to support MP3. We can buy VCD/DVD players. Many can support MP3, there are more portable MP3 players, etc., although several major music vendors are extremely disgusted with this open format, but can not prevent the survival and circulation of this audio compression format. MP3 has been in development for 10 years. He is the abbreviation of MPEG (MPEG: Moving Picture Experts Group) AudioLayer-3, which is a derivative code scheme of MPEG1. In 1993, it was successfully developed by the German Fraunhofer IIS Research Institute and Thomson Corporation. MP3 can achieve an amazing compression ratio of 12:1 and maintain a basic audible sound quality. In the days when the hard disk price was high, MP3 was quickly accepted by users. With the popularity of the Internet, MP3 was accepted by hundreds of millions of users. The release of MP3 encoding technology was actually very imperfect at first. Due to the lack of research on sound and human hearing, the early MP3 encoders were almost all coded in a rough way, and the sound quality was seriously damaged. With the continuous introduction of new technologies, MP3 encoding technology has been improved once and for all, including two major technical improvements.
VBR: MP3 format files have an interesting feature, which can be read while reading, which is also in line with the most basic characteristics of streaming media. In other words, the player can play without the pre-reading of the entire contents of the file, and read where to play, 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, it is determined that each segment of an MP3 file can have a separate average data rate for each frame without a special decoding scheme. So there is a technology called VBR (Variablebitrate, dynamic data rate), which can make a single bitrate for every segment or even every frame of an MP3 file. The advantage of this is that the maximum limit is guaranteed under the premise of ensuring the sound quality. The size of the file. The superiority of this technique is obvious, but it is really difficult to use it, because it requires the encoder to know how to assign a bitrate to each segment. This is like a dummy for an encoder without waveform analysis. This is the case, VBR technology does not appear as dazzling.
Through long-term acoustic research, experts have discovered that there is a shadowing effect on the human ear. The sound signal is actually an energy wave that propagates in air or other medium. The most direct response of the human ear to the sound energy, that is, the loudness or sound pressure is to hear the size of the sound. We call it loudness, which means loudness. The unit of energy is decibel (dB). Even with the same loudness, people will feel different sounds because of their different frequencies. The most easily heard by the human ear is the frequency of 4000 Hz. Regardless of whether the frequency is increased or decreased, even if the loudness is the same, everyone will feel that the sound is getting smaller. But when the loudness drops to a certain level, the human ear can't hear it, and each frequency has a different value.
It can be seen that this curve is basically a V-shape. When the frequency exceeds 15000 Hz, the human ear will feel the sound is very small. Many people who are not very good in hearing will not hear the frequency of 20000 Hz, no matter how loud the loudness is. . When the human ear hears two sounds of different frequencies and different loudness at the same time, the one with less loudness will be ignored. For example, during the day, it is difficult for us to hear the sound of the cooling fan in the computer, but at night it becomes a noise source, according to This principle, the encoder can filter out a lot of inaudible sound, to simplify the information complexity, increase the compression ratio, and not significantly reduce the sound quality. This shading is referred to as a simultaneous shading effect. However, the sound A is obscured by the sound B. If A is in the shielding range centered on B, the shadowing will be more obvious. This range is called the critical bandwidth. The critical bandwidth of each frequency is different, and the higher the frequency, the wider the critical bandwidth.
Frequency (Hz) Critical Bandwidth (Hz) Frequency (Hz) Critical Bandwidth (Hz)
50 80 1850 280
150 100 2150 320
350 100 2500 380
450 110 3400 550
570 120 4000 700
700 140 4800 900
840 150 5800 1100
1000 160 7000 1300
1170 190 8500 1800
1370 210 10500 2500
1600 240 13500 3500
According to this effect, the experts designed the human auditory psychology model. This model was introduced into the MP3 encoding, which led to a revolutionary sound quality revolution. MP3 encoding technology has been carrying the notoriety of poor sound quality, but this notoriety has now Gradually eluted. At this time, the VBR technology that has been buried has been radiant, and with the use of the psychological model, it has a strong temptation and lethality.
For a long time, many people have a bad impression on MP3. More people think that WMA's best sound quality is better than MP3. This is not true. At medium and high bit rates, well-coded MP3 is much better than WMA. Close to the CD sound quality, with not very good hardware support, not many people can distinguish the difference between the two, this is not a myth, although you can easily distinguish between MP3 and CD before blind listening, but now you can hardly guarantee that you can The resolution is correct. Because MP3 is an excellent code, it was buried before.
Sampling
The digital audio system reproduces the original sound by converting the acoustic waveform into a series of binary data (the original sound is an analog signal). The device used to implement this step is an analog-to-digital converter (A/D converter, or ADC, or Analog to digital convert). It samples the sound waves at a rate of tens of thousands of times per second. Each sample records the state of the original simulated sound wave at a certain moment, called the sample. By connecting a series of samples, you can describe a sound wave. The number of samples sampled per second is called the sampling frequency or recovery rate, and the unit is HZ (Hertz). The higher the sampling frequency, the higher the frequency of the sound wave that can be described. The sampling rate determines the range of the sound frequency (equivalent to the pitch) and can be represented by a digital waveform. The range of frequencies represented by waveforms is often referred to as bandwidth. To correctly understand the audio samples can be divided into the number of bits sampled and the frequency of the sample.
Number of samples (sampling accuracy)
The sound files in the computer are represented by the numbers 0 and 1. So the essence of recording on a computer is to convert an analog sound signal into a digital signal. Conversely, during playback, the digital signal is restored to an analog sound signal output. The number of samples can be understood as the resolution of the sound processed by the capture card. The larger the value, the higher the resolution and the more realistic the sound of recording and playback. The bit of the capture card refers to the binary digit of the digital sound signal used by the capture card when acquiring and playing the sound file. The bits of the capture card objectively reflect the accuracy of the digital sound signal's description of the input sound signal. The 8 bits represent the 8th power of 2 - 256, and the 16 bits represent the 16th power of 2 - 64K.
Sample rate
The number of times the media object is sampled per unit time, in Hz. The sampling frequency refers to the number of times the recording device samples the sound signal in one second. The higher the sampling frequency, the more realistic the sound is restored. In today's mainstream acquisition cards, the sampling frequency is generally divided into 22.05KHz, 44.1KHz (44100Hz), 48KHz three levels, 22.05 KHz can only achieve the sound quality of FM broadcast, 44.1KHz is the theoretical CD sound quality limit, 48KHz is more precise. For the sampling frequency higher than 48KHz, the human ear can not be distinguished, so there is not much use value on the computer.
The sampling rate of 5 kHz can only achieve the sound quality of people's speech.
The 11 kHz sampling rate is the lowest standard for playing small sounds and is a quarter of the CD sound quality.
The 22 kHz sampling rate can achieve half the sound quality of the CD, and most websites currently use this sampling rate.
The 44kHz sampling rate is a standard CD sound quality that achieves good hearing.
Number of channels
Divided into mono mono; stereo stereo. Of course there are more channels. Give a list, the channel is more, the effect is good, two channels, indicating that only the left and right sides have sounds coming over, four channels, indicating that the front and rear sounds are transmitted to the bit rate (bitrate)
Also called the code rate. For the encoding format, the amount of audio data per second after compression encoding is indicated. Calculation formula: bit rate = sampling rate x sampling accuracy x number of channels. Unit kbps, where k is 1000
VBR, ABR, CBR
VBR (Variable Bitrate) dynamic bit rate. That is, there is no fixed bit rate, and the compression software instantly determines what bit rate to use based on the audio data during compression. This is the algorithm developed by Xing. They encode the complex part of a song with high Bitrate and the simple part with low Bitrate. Although the idea is good, unfortunately, the VBR algorithm of the Xing encoder is very poor, and the sound quality is far from CBR. Fortunately, Lame perfectly optimized the VBR algorithm to make it the best encoding mode for MP3. This is the recommended encoding mode when the quality is the premise of the file size.
ABR (Average Bitrate) is an interpolation parameter of VBR. Lame created this encoding mode for the poor file size ratio of CBR and the variable size of VBR file generation. ABR is also known as "Safe VBR", which is a segment of every 50 frames (30 frames, about 1 second) in a specified average Bitrate. Low-frequency and insensitive frequencies use relatively low traffic, high frequency and large dynamic performance. Use high traffic. For example, when specifying a wav file with a 192 kbps ABR, Lame will fix 85% of the file with 192 kbps and then dynamically optimize the remaining 15%: the complex part is encoded with more than 192 kbps, the simple part Coded with less than 192 kbps. Compared with the 192kbps CBR, the 192kbps ABR is similar in file size, and the sound quality is improved a lot. The ABR code is 2 to 3 times faster than the VBR code in the speed, and the quality is better than the CBR in the range of 128-256 kbps. Can be used as a compromise between VBR and CBR.
CBR (Constant Bitrate), a constant bit rate, means that the file is a bit rate from beginning to end. Compared to VBR and ABR, the file it compresses is very large, but the sound quality is not significantly improved.
Lossless and lossless According to the process of sampling and quantification, audio coding can only achieve signals that are infinitely close to the natural world, at least the current technology is not possible to make it exactly the same. This is because the signals in nature are continuous, and the audio encoded values ​​are discrete. Therefore, any digital audio coding scheme is lossy, which means that no audio can completely restore the sound of nature.
In computer applications, PCM coding can achieve the highest level of fidelity. It has been widely used for material preservation and music appreciation, including CD, DVD and WAV files. Therefore, the PCM convention has become lossless coding, but this does not mean that the PCM can ensure that the signal is absolutely fidelity, and the PCM can only achieve maximum infinite proximity.
We habitually include MP3 in the category of lossy audio coding, which is relatively PCM encoded.
Emphasis on the relative loss of coding is lossy and non-destructive, and it is very difficult or even impossible to achieve true loss. Just like, we use decimals to express the pi, no matter how high the precision of the decimal, we can only approach infinitely, rather than the value that is really equal to the pi.
Why use audio compression technology to calculate the bitrate of a PCM audio stream is a very easy thing, sample rate value &TImes; sample size value &TImes; channel number bps. A sampling rate of 44.1 kHz, a sampling size of 16 bits, a two-channel PCM encoded WAV file, its data rate is 44.1K & TImes; 16 & TImes; 2 = 1411.2 Kbps. We often say that 128K MP3, the corresponding WAV parameter, is this 1411.2Kbps. This parameter is also called data bandwidth. It is a concept with bandwidth in ADSL. By dividing the code rate by 8, you can get the data rate of this WAV, which is 176.4KB/s. This means that the one second sampling rate is 44.1KHz, the sampling size is 16bit, and the two-channel PCM encoded audio signal requires 176.4KB of space, which is about 10.34M in one minute, which is unacceptable to most users. Especially for those who like to listen to music on the computer, to reduce the disk usage, there are only 2 ways to reduce the sampling index or compression. It is not advisable to reduce the indicators, so experts have developed various compression schemes. Because the purpose and target market are different, the audio quality and compression ratio achieved by various audio compression codes are different, which we will mention in the following articles. One thing is for sure, they are all compressed.
Frequency vs. Sample Rate The sample rate represents the number of times the original signal is sampled per second. The common sample rate of audio files we have is 44.1KHz. What does this mean? Suppose we have 2 sine wave signals, 20Hz and 20KHz, each of which is one second in length, corresponding to the lowest frequency and the highest frequency we can hear. We respectively sample 40KHz of these two signals, we can get What kind of result? The result is that the 20 Hz 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, recording low frequency information is much more detailed than high frequency. This is also why some audiophiles accuse CDs of having digital sounds that are not real enough. The 44.1KHz sampling of CDs does not guarantee that high-frequency signals are better recorded. To record high-frequency signals better, it seems that a higher sampling rate is required, so some friends use a sampling rate of 48KHz when capturing CD tracks, which is not desirable! This actually has no benefit to the sound quality. For the tracking software, keeping the same sampling rate as the 44.1KHz provided by the CD is one of the guarantees of the best sound quality, instead of improving it. Higher sample rates are only useful when compared to analog signals. If the sampled signal is digital, do not try to increase the sample rate.
PCM coded PCM pulse code modulation is an abbreviation of Pulse CodeModulation. In the previous text, we mentioned the general workflow of PCM. We don't need to care about the calculation method used by PCM final encoding. We only need to know the advantages and disadvantages of PCM encoded audio stream. The biggest advantage of PCM coding is that the sound quality is good, and the biggest disadvantage is the large size. Our common AudioCD uses PCM encoding, and the capacity of a single disc can only hold 72 minutes of music information.
WAVE
This is an old audio file format developed by Microsoft. WAV is a file format that conforms to the PIFF Resource Interchange FileFormat specification. All WAVs have a file header, the encoding parameter of the header audio stream. WAV does not have a hard code for encoding audio streams. In addition to PCM, almost all encodings that support the ACM specification can encode WAV audio streams. Many friends don't have this concept. Let's take AVI as a demonstration, because AVI and WAV are very similar in file structure, but AVI has more video streams. There are many kinds of AVIs that we come into contact with, so we often need to install some Decode to watch some AVIs. We have more DivXs as a kind of video encoding. AVI can use DivX encoding to compress video streams. Of course, other ones can be used. Encoding compression. Similarly, WAV can also use a variety of audio encoding to compress its audio stream, but we all commonly use WAVs whose audio streams are PCM encoded, but this does not mean that WAV can only use PCM encoding, and MP3 encoding can also be used in WAV. In the same way as AVI, you can enjoy these WAVs as long as the corresponding Decode is installed.
Under the Windows platform, WAV based on PCM encoding is the best supported audio format. All audio software can be perfectly supported. Because it can achieve high sound quality requirements, WAV is also the preferred format for music editing creation. Suitable for saving music material. Therefore, PCM-based WAV is used as an intermediary format, often used in the conversion of other encodings, such as MP3 to WMA.
MP3 encoding MP3 is currently the most popular audio compression format, which is widely accepted by everyone. Various MP3 related software products are emerging one after another, and more hardware products are also beginning to support MP3. We can buy VCD/DVD players. Many can support MP3, there are more portable MP3 players, etc., although several major music vendors are extremely disgusted with this open format, but can not prevent the survival and circulation of this audio compression format. MP3 has been in development for 10 years. He is the abbreviation of MPEG (MPEG: Moving Picture Experts Group) AudioLayer-3, which is a derivative code scheme of MPEG1. In 1993, it was successfully developed by the German Fraunhofer IIS Research Institute and Thomson Corporation. MP3 can achieve an amazing compression ratio of 12:1 and maintain a basic audible sound quality. In the days when the hard disk price was high, MP3 was quickly accepted by users. With the popularity of the Internet, MP3 was accepted by hundreds of millions of users. The release of MP3 encoding technology was actually very imperfect at first. Due to the lack of research on sound and human hearing, the early MP3 encoders were almost all coded in a rough way, and the sound quality was seriously damaged. With the continuous introduction of new technologies, MP3 encoding technology has been improved once and for all, including two major technical improvements.
VBR: MP3 format files have an interesting feature, which can be read while reading, which is also in line with the most basic characteristics of streaming media. In other words, the player can play without the pre-reading of the entire contents of the file, and read where to play, 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, it is determined that each segment of an MP3 file can have a separate average data rate for each frame without a special decoding scheme. So there is a technology called VBR (Variablebitrate, dynamic data rate), which can make a single bitrate for every segment or even every frame of an MP3 file. The advantage of this is that the maximum limit is guaranteed under the premise of ensuring the sound quality. The size of the file. The superiority of this technique is obvious, but it is really difficult to use it, because it requires the encoder to know how to assign a bitrate to each segment. This is like a dummy for an encoder without waveform analysis. This is the case, VBR technology does not appear as dazzling.
Through long-term acoustic research, experts have discovered that there is a shadowing effect on the human ear. The sound signal is actually an energy wave that propagates in air or other medium. The most direct response of the human ear to the sound energy, that is, the loudness or sound pressure is to hear the size of the sound. We call it loudness, which means loudness. The unit of energy is decibel (dB). Even with the same loudness, people will feel different sounds because of their different frequencies. The most easily heard by the human ear is the frequency of 4000 Hz. Regardless of whether the frequency is increased or decreased, even if the loudness is the same, everyone will feel that the sound is getting smaller. But when the loudness drops to a certain level, the human ear can't hear it, and each frequency has a different value.
It can be seen that this curve is basically a V-shape. When the frequency exceeds 15000 Hz, the human ear will feel the sound is very small. Many people who are not very good in hearing will not hear the frequency of 20000 Hz, no matter how loud the loudness is. . When the human ear hears two sounds of different frequencies and different loudness at the same time, the one with less loudness will be ignored. For example, during the day, it is difficult for us to hear the sound of the cooling fan in the computer, but at night it becomes a noise source, according to This principle, the encoder can filter out a lot of inaudible sound, to simplify the information complexity, increase the compression ratio, and not significantly reduce the sound quality. This shading is referred to as a simultaneous shading effect. However, the sound A is obscured by the sound B. If A is in the shielding range centered on B, the shadowing will be more obvious. This range is called the critical bandwidth. The critical bandwidth of each frequency is different, and the higher the frequency, the wider the critical bandwidth.
Frequency (Hz) Critical Bandwidth (Hz) Frequency (Hz) Critical Bandwidth (Hz)
50 80 1850 280
150 100 2150 320
350 100 2500 380
450 110 3400 550
570 120 4000 700
700 140 4800 900
840 150 5800 1100
1000 160 7000 1300
1170 190 8500 1800
1370 210 10500 2500
1600 240 13500 3500
According to this effect, the experts designed the human auditory psychology model. This model was introduced into the MP3 encoding, which led to a revolutionary sound quality revolution. MP3 encoding technology has been carrying the notoriety of poor sound quality, but this notoriety has now Gradually eluted. At this time, the VBR technology that has been buried has been radiant, and with the use of the psychological model, it has a strong temptation and lethality.
For a long time, many people have a bad impression on MP3. More people think that WMA's best sound quality is better than MP3. This is not true. At medium and high bit rates, well-coded MP3 is much better than WMA. Close to the CD sound quality, with not very good hardware support, not many people can distinguish the difference between the two, this is not a myth, although you can easily distinguish between MP3 and CD before blind listening, but now you can hardly guarantee that you can The resolution is correct. Because MP3 is an excellent code, it was buried before.
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