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Moving picture experts group international standards
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A browser plug-in transmits audio files in the form of streams through the Internet.
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Ogwobis
Comparative characteristics
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Moving picture experts group international standards
MP3 file
MPEG-4 MIDIWMAREALAUDIOV Qogvorbis has expanded the editing function of this paragraph.
The common feature of audio file formats is: playing or processing audio files in a computer, that is, converting audio files from digital to analog. This process is also composed of sampling and quantization. The lowest frequency of sound that the human ear can hear is 20Hz to the highest frequency of 20 Hz, which is inaudible to the human ear. So the maximum bandwidth of audio file format is 20 Hz, so the sampling rate needs to be between 40-50hz, and each sample needs more. The standard of audio digitization is that the signal-to-noise ratio is 16 bits-96dB per sample. PCM adopts linear pulse code modulation, and the length of each quantization step is the same. This standard is used to make audio files.
Edit this paragraph to develop.
Audio formats change with each passing day. By 2008, audio formats include: CD format, WAVE(*. WAV), AIFF, Au, MP3, MIDI, WMA, RealAudio, VQF, OggVorbis, AAC and APE.
Edit this CD
Cd discs are used to store files in cd format.
CD format sound quality comparison audio format. So when it comes to audio format, CD is naturally the forerunner. In the "open file type" of most playback software, you can see *. CDA format, that is, CD audio track. The standard CD format is 44. 1K sampling frequency, 88K/ s rate and 16 quantization bits. Because the CD track can be said to be almost lossless, its sound is basically loyal to the original sound, so if you are an audio enthusiast, CD is your first choice. It will make you feel the sound of nature. A CD can be played in a CD player or in a computer through various playing software. The CD audio file is a *. CDA file, which is only an index information, does not really contain sound information, so no matter what the length of CD music is, you should put *. The CDA file seen on the computer is 44 bytes long. Note: * in CD format cannot be copied directly. CDA file to hard disk for playback. You need to use audio capture software like EAC to convert files in CD format into WAV. If the quality of optical drive is up to standard and EAC parameters are set properly, it can be said that audio capture is basically lossless. I recommend you to use this method.
Edit this wave
Waveform file audio image
Wave (*. WAV) is a sound file format developed by Microsoft. It conforms to the specification of PipfResource exchange file format and is used to save the audio information resources of WINDOWS platform, which is supported by WINDOWS platform and its applications. “*。 WAV "format supports many compression algorithms such as MSADPCM and CCITT A LAW, and supports many audio bits, sampling frequencies and channels. Like CD format, the sampling frequency of standard WAV file is 44. 1K, the rate is 88K/ s, and the quantization bit is 16. Look, the quality of audio files in WAV format is almost the same as that in CD format.
AIFF edits this paragraph.
Aiff (audio interchange file format) format and Au format, which are very similar to WAV, are also supported in most audio editing software. Data graph stored in AIFF format
AIFF is short for audio interchange file format. It is an audio file format developed by Apple, supported by MACINTOSH platform and its applications, and LIVEAUDIO in Netscape browser also supports AIFF format. So it's not common. AIFF is the standard audio format on Apple computer, which is a part of QuickTime technology. The characteristic of this format is that the format itself has nothing to do with the meaning of data, so it is favored by Microsoft, and the WAV format is also developed accordingly. Although AIFF is an excellent file format, it is not very popular on PC platform because it is the format on Apple computer. However, because Apple computers are mostly used in multimedia production and publishing industry, almost all audio editing software and playback software support AIFF format to some extent. As long as Apple computer is still there, AIFF will always have a place. Because of its inclusiveness, AIFF supports many compression technologies.
Edit this paragraph AU
Audio file is a digital audio format introduced by SUN Company. AU file was originally a digital sound file under UNIX operating system. Since the WEB server on the Internet in the early days was mainly based on UNIX, the file in AU format is also a common sound file format on the Internet today.
Edit this MPEG
Embedded MPEG4 decoding system
MPEG is the abbreviation of Moving Picture Experts Group. This expert group was established in 1988, and is responsible for establishing the video and audio compression standard for optical discs. MPEG audio file refers to the sound part of MPEG standard, namely MPEG audio layer. At present, MP3 is the most common music format on the Internet. Although it is a kind of lossy compression, its greatest advantage is that it can achieve a higher compression ratio with minimal sound distortion. The formats included in MPEG include MP 1, MP2, MP3 and MP4.
Edit this MP3.
MP3 format was born in Germany in 1980s. The so-called MP3 also refers to the audio part of the MPEG standard, that is, the MPEG audio layer. According to the difference of compression quality and coding processing, it is divided into three layers, which correspond to three kinds of sound files: * .mp1* .mp2/* .mp3. It should be noted that the compression of MPEG audio files is lossy compression. MPEG3-3 audio coding has a high compression ratio of10:12:1,while basically keeping the low audio part undistorted, but sacrificing the high audio part from 12KHz to 16KHz in the sound file. Music files with the same length are stored in * * .mp3 format, generally only * *110. WAV file, and the sound quality is not as good as CD format or wav format Because it has small files and good sound quality; So at the beginning of its appearance, no other audio format can match it, thus providing good conditions for the development of * * .mp3 format. Until now, this format is still all the rage, and its position as the mainstream audio format is hard to shake. But the copyright problem of MP3 music has not been solved, because MP3 has no copyright protection technology, which means anyone can use it clearly. There are many sampling frequencies for MP3 compressed music. The sampling frequency of 64Kbps or lower can save space, and the standard of 320Kbps can also achieve extremely high sound quality. With the MP3 encoder equipped with MusicMatch Jukebox 6.0, Fraunhofer IIS Mpeg Lyaer3, a 3-minute song is encoded at the frequency of 128Kbps, and a 2.82MB MP3 file is obtained. The default CBR (fixed sampling frequency) technology can sample a song at a fixed frequency, while VBR (variable sampling frequency) can increase the sampling frequency to obtain higher sound quality when the music is "busy", but the generated MP3 file may not be played on some players. Set the level of VBR to be basically the same as that of the previous CBR file, and the generated VBR MP3 file is 2.9MB. MP3 is the lossy compressed digital audio format with the largest number of users up to 2008. Its full name is MPEG (Moving Picture Experts Group) Audio Layer -3. When it first appeared, its coding technology was not perfect. It is more like a coding standard framework, which is left for people to improve. Early MP3 coding used fixed bit rate (CBR), and 128KBPS means coding at a fixed data rate of 128KBPS-you can increase the bit rate to 320KBPS at the highest, and the sound quality will be better. Naturally, the file size will increase accordingly. Because the coding method of MP3 is open, we can choose different acoustic principles for compression based on this standard framework, so Xing Company soon introduced the variable rate compression method (). Its principle is to encode the complex part of a song with high bit rate and the simple part with low bit rate. Only in this way can we further achieve the unity of quality and quantity. Of course, the VBR algorithm of the early Xing encoder is very poor, and the sound quality is far from CBR (fixed bit rate). However, this algorithm points out a direction, and other developers have also introduced their own VBR algorithm, which has been improving the effect. LAME is recognized as the best one at present, which perfectly realizes the VBR algorithm and is completely free software. The development team composed of enthusiasts has been developing and perfecting it. LAME developed ABR algorithm based on VBR. ABR(AverageBitrate (ABR is an interpolation parameter of VBR. LAME created this coding mode in view of the poor file volume ratio of CBR and the uncertain file size generated by VBR. Within the specified file size, ABR takes every 50 frames (30 frames is about 1 sec) as a segment, with relatively low traffic for low frequency and insensitive frequency and high traffic for high frequency and large dynamic performance, which can be used as a compromise between VBR and CBR. Shortly after MP3 came out, it opened up a brand-new music field with its high compression ratio of 12: 1 and good sound quality. However, the openness of MP3 inevitably leads to copyright disputes. In this context, MP4 with smaller files, better sound quality and more effective copyright protection came into being. There is no necessary connection between MP3 and MP4. First of all, MP3 is an international technical standard for audio compression, and MP4 is indeed a brand name.
Edit this MPEG-4
MPEG-4 standard is a video compression standard for multimedia applications, which was released by the International Moving Picture Expert Group in June 2000. It adopts object-based compression coding technology. Before encoding, the video sequence is analyzed and each video object is separated from the original image. Then, the shape information, motion information and texture information of each video object are encoded separately, and the temporal redundancy between consecutive frames is removed through motion prediction and motion compensation superior to MPEG-2. Its core is based on Content-basedscalability, which can assign priority to every object in the image, and use high spatio-temporal resolution to represent more important objects, and use low resolution to represent less important objects (such as the background of the monitoring system), or even not display them. Therefore, it has the ability to allocate resources adaptively, and can realize high-quality and low-rate image communication and video transmission. MPEG-4 is widely used in network multimedia, video conference, multimedia monitoring and other image transmission systems because of its high quality and low transmission rate. Most mature MPEG-4 applications at home and abroad are based on PC-level client and server mode, but few are applied to embedded systems. Most embedded MPEG-4 decoding systems use commercial embedded operating systems, such as WindowsCE and VxWorks, which have high cost and poor flexibility. If embedded Linux is used as the operating system, it is not only convenient to develop, but also can save costs, and can be reduced according to the actual situation, with less resources, strong flexibility, good network performance and wider application scope.
Edit this MIDI
MIDI format output schematic diagram
MIDI (Musical Instrument Digital Interface) format is used by people who often play music. MIDI allows digital synthesizers to exchange data with other devices. The MID file format is inherited from MIDI. MID file is not recorded sound, but a set of instructions that record sound information and then tell the sound card how to reproduce music. This MIDI file only needs about 5 ~ 1 0kb of music per minute. MID files are mainly used for original musical instruments, amateur performances of pop songs, game tracks and electronic greeting cards. *. * effect. Mid file playback depends entirely on the grade of sound card. * the biggest use. Mid format belongs to the field of computer typesetting. *.MID files can be written in composition software, or the music played by an external sequencer can be input into the computer through the MIDI port of the sound card to make *. Intermediate file.
Edit this WMA
WMA (Windows Media Audio) format is a heavyweight player from Microsoft. Its background is tough, and its sound quality is better than MP3 format and far better than RA format. Like the VQF format developed by Yamaha Company in Japan, it aims to achieve a higher compression ratio than MP3 by reducing data traffic but maintaining sound quality. Generally, the compression ratio of WMA can reach about 1: 18. Another advantage of WMA is that content providers can add copyright protection through DRM (Digital Rights Management) schemes such as Windows Media Rights Manager 7. This built-in copyright protection technology can limit the playing time and times, even the playing machine. This is good news for music companies that have been overwhelmed by piracy. In addition, WMA also supports audio streaming technology, which is suitable for play online on the Internet. As the pioneer of Microsoft's online music, it can be said that it is advanced in technology and powerful, and it is not more convenient to install a player like MP3. The seamless combination of windows operating system and Windows Media Player allows you to play WMA music directly as long as you install Windows operating system. The new version of Windows Media Player7.0 adds the function of directly converting CD into WMA sound format. In the newly produced operating system Windows XP, WMA is the default encoding format. Everyone knows what happened to Netscape, and now the "wolf" is coming again. WMA is a format that can adjust the sound quality during recording. The same format, good sound quality can be comparable to CD, high compression rate can be used for network playback. Although it is not very popular on the Internet now, with the large-scale promotion of Microsoft, it has been recognized and strongly supported by more and more websites. In the field of online music, it is almost equal to * * .mp3, and it is also carving up the world laid by Real in terms of online playing. Therefore, almost all audio formats feel the pressure of WMA format. WMA format file structure diagram
According to the official information released by Microsoft, WMA format is extremely protective, and it can even limit the playing machine, playing time and playing times, and has considerable copyright protection ability. It should be said that WMA is aimed at the shortcomings of MP3 without copyright restrictions-ordinary users may welcome this format, but as copyright owners, record companies prefer music compression technology that is difficult to copy, and Microsoft's WMA takes care of the needs of these record companies. In addition to copyright protection, WMA also deepens the compression ratio, with the goal of making the file size smaller under the same sound quality (of course, it is only effective when the MP3 bit rate is lower than 192KBPS. In fact, when using LAME algorithm to compress MP3 format, it is generally reflected that the sound quality is better when MP3 is higher than 192KBPS than WMA).
Edit this paragraph, RealAudio
RealAudio is mainly suitable for online voice RealAudio format and website server connection diagram on the network.
I'm glad to hear that most users are still using Modem with a speed of 56Kbps or lower, so typical playback is not the best sound quality. Some download websites will prompt you to choose the best real file according to your modem speed. Real has several file formats: RA(RealAudio), RM(RealMedia, RealAudio G2), RMX(RealAudio Secured) and so on. The characteristics of these formats are that they can change the sound quality with the different network bandwidth, and on the premise of ensuring that most people can hear smooth sound, the audience with richer bandwidth can get better sound quality. Recently, with the general improvement of network bandwidth, Real Company is launching a format for network broadcasting, which can reach the sound quality of CD. If your RealPlayer software can't handle this format, it will remind you to download a free upgrade package. Many music websites provide audition versions of songs in real format. Now the latest version is RealPlayer 9.0, and the 39th issue of Computer News also introduced RealPlayer 9.0 in detail, so I won't go into details here.
VQF edits this paragraph.
Another format of Yamaha Company is *. VQF, its core is to reduce data traffic while maintaining sound quality to achieve higher compression ratio. The audio compression ratio of Vqf is nearly twice that of standard MPEG audio compression ratio, which can reach 18: 1 or even higher. That is to say, pressing a 4-minute song (WAV file) into MP3 requires about 4MB of hard disk space, while the same song, if using VQF audio compression technology, only needs about 2MB of hard disk space. Therefore, MP3 and RA are not rivals of VQF in terms of audio compression ratio. Under the same conditions, the compressed VQF file is 30% ~ 50% smaller than MP3, which is more convenient for online communication and has excellent sound quality, close to CD (16-bit 44. 1kHz stereo). It can be said that the technology is also advanced, but due to poor publicity, this format is difficult to use. *.vqf can be played with Yamaha's player. At the same time, Yamaha also provides software to convert from *. Wav file to *. Vqf file. This document lacks features and publicity. When VQF compresses music at the audio sampling rates of 44KHz and 80kbit/s, its sound quality is better than that of MP3 at 44KHz and 128kbit/s, while when VQF compresses music at the frequencies of 44KHz and 96kbit/s, its sound quality is almost equivalent to that of MP3 at 44KHz and 256 kbit/s. Few people can hear SoundVQ compression when listening to playback effects. VQF audio file format
You only need Pentium 75 or higher computer configuration to play VQF. Of course, if you use a Pentium 100 or higher machine, VQF can run better. In fact, playing VQF only needs about 5 ~ 10% more CPU than playing Mp3. VQF or TwinVQ technology was developed by NTT and Yamaha, but their application software is free. Only NTT and Yamaha have not published the source code of VQF.
Edit this paragraph OggVorbis
OggVorbis is a new audio compression format, similar to existing music formats such as MP3. But one difference is that it is completely free, open and without patent restrictions. Vorbis is the name of this audio compression mechanism, while Ogg is the name of a project to design a completely open multimedia system. At present, the plan has only realized OggVorbis. The extension of OggVorbis file is *.OGG, and the design format of this file is very advanced. This file format can continuously improve the size and sound quality without affecting the old encoder or player. VORBIS uses lossy compression, but reduces the loss by using a more advanced acoustic model. Therefore, OGG encoded at the same bit rate sounds better than MP3. In addition, there is another reason that MP3 format is protected by patent. If you want to publish your own works in MP3 format, you need to pay royalties to Flawn Hof (the company that invented MP3). VORBIS has no such problem at all. Ogwobis scheme
For music fans, the obvious advantage of using OGG files is that they can get better sound quality with smaller files. Moreover, because OGG is completely open and free, making OGG files will not be restricted by any patents, and it is expected to obtain a large number of encoders and players. This is why there are so few MP3 encoders, and most of them are commercial software, because Flawn Hof collects royalties. Vorbis uses a completely different mathematical principle from MP3, so it faces different challenges when compressing music. Vorbis and MP3 files encoded at the same bit rate have the same sound quality. Vorbis has a well-designed and flexible annotation, which avoids such complicated operations as ID3 tagging of MP3 files. Vorbis also has a bit rate scaling function: the bit rate of a file can be adjusted without recoding. Vorbis file can be divided into small pieces and edited with sample granularity; Vorbis supports multiple channels; Vorbis files can be logically connected and so on.
Edit this paragraph comparison
As the standard of digital music file format, WAV format is too large and inconvenient to use. Therefore, we usually compress it into MP3 or WMA format. Compression methods include lossless compression, lossy compression and mixed compression. MPEG and JPEG belong to mixed compression. If compressed data is restored, the data is actually different. Of course, the human ear is indistinguishable. Therefore, if MP3 and OGG formats are restored from compressed state, there will be loss. But even if the APE format is restored, the original sound quality can be preserved without loss. Therefore, APE can be compressed and restored without losing treble texture. On the premise of completely maintaining the sound quality, the compression capacity of APE is appropriately reduced. Take one of the most common 38MBWAV files as an example. About 25MB after compressed into APE format, 13MB less than at the beginning. Today, with the increase of MP3 capacity, 25 million songs are no longer a monster. 1GB mp3, you can put 4 CDs, that is, more than 40 songs, that's enough! The formats supported by MP3 are MP3 and WMA. Because MP3 is lossy compression, the emphasis is on the sampling rate, which is generally 44. 1KHZ. There is also a bit rate, that is, data stream, which is generally 8-320KBPS. When encoding MP3, let's see if it supports variable bit rate (VBR). Most MP3 players now support it, which can reduce the effective file size. WMA is an audio format promoted by Microsoft, which is relatively smaller than MP3. [ 1]?