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.
As of 2008, MP3 is the lossy compressed digital audio format with the largest number of users. 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, while MP4 is a brand name. 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 a boon for music companies that have been plagued 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 more convenient to install an MP3 player. 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 that Real has laid. Therefore, almost all audio formats feel the pressure of WMA format. 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). RealAudio is mainly suitable for online music appreciation on the Internet. At present, 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 introducing a format for network broadcasting to realize CD sound quality. 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. 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. In other words, 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 for converting 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. 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. 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, MP3 format is protected by patent for another reason. 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.
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. The full name of AMR is Adaptive Multi-Rate, which is mainly used for audio of mobile devices. The compression ratio is relatively high, but the quality is poor compared with other compression formats. Because it is mostly used for voice and phone calls, the effect is still very good.
I. Classification
1.AMR: Also known as AMR-NB, compared with WB below, the voice bandwidth range is 300-3400 Hz, and the sampling frequency is 8KHz.
2.AMR-WB:AMR broadband,
Voice bandwidth range: 50-50-7000Hz 16KHz sampling.
"AMR-WB" is called "adaptive multirate-wideband", that is, "adaptive multirate wideband coding", and its sampling frequency is 16kHz. It is a wideband speech coding standard adopted by ITU-T and 3GPP, also known as G722.2 standard. AMR-WB provides a voice bandwidth of 50 Hz to 7000 Hz, and users can subjectively feel that voice is more natural, comfortable and easy to distinguish than before.
In contrast, the EFR (Enhanced Full Rate) sampling frequency used by GSM is 8kHz, and the voice bandwidth is 200 ~ 3400 Hz.
The advantages of applying AMR-WB to narrowband GSM (full rate channel 16k, GMSK) are that it can adopt three kinds of coding, namely 6.6kb/s, 8.85kb/s and12.65kb/s. When the network is busy, the C/I deteriorates, and the encoder can automatically adjust the coding mode, thus enhancing QoS. In this application, AMR-WB has better immunity than AMR-NB.
The application of AMR-WB in EDGE and 3G can fully reflect its advantages. Adequate transmission bandwidth ensures that AMR-WB can adopt nine kinds of codes ranging from 6.6kb/s to 23.85 KB/s * *, and the voice quality exceeds that of PSTN fixed telephone.