Sampling rate:
Digital audio system reproduces the original sound by converting sound waves into a series of binary data. The equipment to realize this step is analog-to-digital converter (A/D), which samples the sound waves at the rate of tens of thousands of times per second, and each sampling records the state of the original analog sound waves at a certain moment, which is called a sample.
A string of samples can be connected to describe a sound wave, and the number of samples per second is called sampling frequency or output, and the unit is HZ (hertz). The higher the sampling frequency, the higher the audio frequency that can be described. For each sampling system, a certain number of storage bits are allocated to represent the amplitude state of sound waves, which is called sampling resolution or sampling accuracy. Every time one bit is added, the number of states representing the amplitude of sound wave will be doubled, and the dynamic range state of 6db will be increased, that is, the dynamic range of 6db. A 2-bit digital audio system can represent thousands of states, that is, the dynamic range of 12db, and so on. If the number of bits continues to increase, the sampling accuracy will increase at a very fast speed. It can be calculated that 16bit can represent 65536 states, corresponding to 96db, while 20 bits can represent 1048576 states, corresponding to 120db. 24bit can represent as many as 167772 16 states. Corresponding to the dynamic range of 144db, the higher the sampling accuracy, the finer the sound wave reduction. (Note: Dynamic range refers to the range from the weakest sound to the strongest sound) The hearing range of human ears is usually 20HZ~20KHZ.
According to Nyquist sampling theorem, the waveform can be completely and truly restored by sampling at twice the frequency of sine wave, so the sampling frequency of digital recording wave is directly related to its highest restoration frequency index. For example, sampling at the sampling frequency of 44. 1KHZ can restore the highest frequency of 22.05KHZ, which is slightly higher than the hearing limit of human ears. (Note: MD can be recorded. For example, the sampling frequency of R900 is 44. 1KHZ, and there is a sampling frequency converter, which can convert the input 32KHz/44. 1KHZ/48KHZ into the standard sampling frequency of this machine, which is enough to record and truly reproduce the sound that everyone in the world can distinguish, so the sampling specification of CD audio is defined as 6544. 44KHZ, even if the recording of 16bit is realized in the ideal environment of high-precision electronic components that are almost impossible to manufacture in real life, it will still be troubled by problems such as filtering and sound specific bits, and people can still perceive some slight distortion, so many professional digital audio systems use 18bit or even 24bit for recording and playback.
Existing sampling methods:
MP3: The full name of MP3 should be MPEG 1 Layer-3 audio file. The Chinese translation of MPEG(Moving Picture Experts Group) is moving picture expert group, especially the moving video compression standard. MPEG audio file is the sound part of MPEG 1 standard, also called MPEG audio layer. According to the compression quality and coding complexity, it is divided into three layers, namely, Layer- 1, Layer2 and Layer3, which respectively correspond to three kinds of sound files, namely, MP 1, MP2 and MP3, and different levels of coding are adopted according to different purposes. The higher the MPEG audio coding level, the more complicated the encoder and the higher the compression ratio. The compression ratios of MP 1 and MP2 are 4: 1 and 6: 1-8: 1 respectively, while the compression ratio of MP3 is as high as10:1-2. However, MP3 uses lossy compression for audio signals. In order to reduce the distortion of sound, MP3 adopts "sensory coding technology", that is, when encoding, the audio file is analyzed in frequency spectrum, and then the noise level is filtered by a filter, and then the remaining bits are quantized and scattered, and finally an MP3 file with high compression ratio is formed. The compressed file can achieve a sound effect closer to the original sound source when playing back. (Another mp3 PRO:MP3 PRO encoder divides the recording into two parts: MP3 part and PRO part. The mp3 part analyzes the information in the low frequency band and encodes it into a normal mp3 file data stream. This enables the encoder to concentrate on coding less useful information and obtain better quality coding effect. At the same time, it also ensures the compatibility between mp3PRO files and old mp3 players. PRO part analyzes the high-frequency band information and encodes it as a part of mp3 data stream, which is usually ignored in old mp3 decoders. The new mp3PRO decoder will effectively use this part of the data stream and combine two sections (high frequency band and low frequency band) to produce a complete audio band, thus enhancing the sound quality. )
WMA:WMA is a file format encoded by Windows Media Audio developed by Microsoft. WMA is not aimed at the stand-alone market, but the network! The competitor is the famous Real Networks in the online media market. Microsoft claims that WMA can achieve sound quality close to CD at a bit rate of only 64kbps. Unlike the previous coding, WMA supports copy protection. It supports adding protection through Windows Media Rights Manager, and can limit the playing time and times, even playing the machine and so on. WMA supports streaming media technology, that is, reading and playing at the same time, so WMA can easily realize play online. Because it is a masterpiece of Microsoft, Microsoft has added support for WMA in Windows. WMA has excellent technical characteristics. With the strong promotion of Microsoft, this format has been accepted by more and more people.
WAV: This is an ancient audio file format developed by Microsoft. WAV is a file format, which conforms to the piff resource exchange file format specification. All wavs have a file header, which is the coding parameter of the audio stream. WAV has no hard and fast rules for the coding of audio streams. Except PCM, almost all codes supporting ACM specification can encode the audio stream of WAV. Many friends don't have this concept. Let's take AVI as an example, because AVI and WAV are very similar in file structure, but AVI has only one video stream. We are exposed to many kinds of AVI, so we often need to install some decoders to watch some AVI. DivX, which we are exposed to, is a kind of video coding. AVI can use DivX coding to compress the video stream, and of course, it can also use other coding to compress it. Similarly, WAV can also use a variety of audio codes to compress its audio stream, but we usually use WAV whose audio stream is pulse code modulated, but this does not mean that WAV can only use pulse code modulation, and MP3 coding can also be used in WAV. Just like AVI, you can enjoy these wavs as long as you install the corresponding decoder. Under the Windows platform, WAV based on pulse code modulation is the best audio format, and all audio software can support it perfectly. Because it can meet the requirements of high sound quality, WAV is also the preferred format for music editing and creation, which is suitable for saving music materials. Therefore, WAV based on pulse code modulation is used as an intermediate format, which is often used for mutual conversion of other codes, such as MP3 to WMA.
Ogg Vorbis: Known as MP3 killer! Where does Ogg Vorbis come from? OGG is the project name of a huge multimedia development plan, which will involve the coding development of video and audio. The purpose of the whole OGG project plan is to provide a completely free multimedia coding scheme for anyone! OGG's belief is: open! Free! Wobis is a playboy in Terry Pratt's fantasy novel Little God. This word became the official name of audio coding in OGG project. At present, Vorbis has been successfully developed and an encoder has been developed. Ogg Vorbis is a high quality audio coding scheme. Official data show that Ogg Vorbis can achieve better sound quality than MP3 at a relatively low data rate! Ogg Vorbis is also far more advanced than MP3 developed successfully in 1990s. It can support multiple channels. What does this mean? This means that Ogg Vorbis can encode all channels with the support of SACD, DTSCD and DVD audio (not yet), instead of MP3 encoding only two channels. The rise of multi-channel music has brought revolutionary changes to music appreciation, especially when enjoying symphony, which will bring more sense of presence. This revolutionary change is beyond the adaptability of MP3. Like MP3, Ogg Vorbis is a flexible and open audio coding. When the coding scheme has been fixed, it can obviously adjust the sound quality and improve the new algorithm. So its sound quality will get better and better. Similar to MP3, Ogg Vorbis is more like an audio coding framework, which can be continuously improved by introducing new technologies. Like MP3, OGG supports VBR.
Ra: Ra is a RealAudio format, which is a format that Internet users have more contact with. The online audition of music websites mostly adopts RealAudio, which is aimed at the media market on the network and supports very rich functions. The biggest trick is that this format can control its own bit rate according to the bandwidth of the audience, and improve the sound quality as much as possible on the premise of ensuring fluency. RA can support a variety of audio coding, including ATRAC3. Like WMA, RA not only supports reading while playing, but also supports using special protocols to hide the real network address of files, so as to realize the appreciation mode of only playing online without downloading. This is very important for record companies and record sales companies. With the vigorous promotion of all parties, RA and WMA are the most widely used audio media formats for online audition on the Internet.
APE:APE is a lossless compression format provided by Monkey Audio. Monkey's audio provides Winamp plug-in support, so this means that the compressed file is no longer a simple compressed format, but an audio file format that can be played like MP3. The compression ratio of this format is much lower than other formats, but it can be truly lossless, so it has won the favor of many enthusiastic users. Among many lossless compression schemes, APE is a format with outstanding performance, satisfactory compression ratio and fast compression speed, which has become the best choice for many friends to exchange feverish music privately.
ACC:AAC (Advanced Audio Coding) is a technology provided by Dolby Labs for the music community. AAC claims that "it can accommodate up to 48 audio tracks with a sampling rate of 96 KHz, and can provide music programs with 5. 1 channel, with the quality equivalent to ITU-R broadcasting and a data rate of 320Kbps". Compared with MP3, it has better sound quality and can save about 30% storage space and bandwidth. It is a technology developed according to the specification of MPEG-2.
ATRAC 3/ATRAC 3 Plus: Atrac 3 (adaptive transform acoustic coding 3) was developed by Sony Corporation of Japan, which is an upgraded version of Atrac adopted by MD. Its compression ratio (about twice that of Atrac) and sound quality are equivalent to MP3. The compression principle includes simultaneous masking, aging masking and equal loudness curve, which is similar to MP3. The copyright protection function of ATRAC3 uses OpenMG. At present, the portable player corresponding to ATRAC3 is mainly Sony's own products. However, in February 2000, the company signed patent licensing agreements with semiconductor manufacturers such as Fujitsu, Hitachi, NEC, Rohm, Sanyo and TI to manufacture and sell LSI for ATRAC3.