Digital audio source, that is, digital audio format, originally referred to CD. After CD is compressed, it has derived a variety of formats suitable for playback on Walkmans. These compressed formats can be divided into two categories: Yes Loss compressed and lossless compressed. The compression mentioned here refers to the conversion of PCM-encoded or WAV-format audio streams into other formats through special compression processing, thereby achieving the effect of reducing file size. Lossy/lossless refers to whether the sound signal retained by the new file is reduced compared to the original PCM/WAV format signal after compression.
Audio Related Parameters
What is the bit rate? Of course I can't directly explain to you that "rate is bit rate". Everyone should have noticed a small message when playing sound files with some software. For example, "128Kbps", "1411Kbps"...Some friends know that, under normal circumstances, the larger the number in front of "Kbps", the better the sound effect. For example, CD is "1411Kbps". So, what exactly do these numbers represent? Simply put, it is how much data is converted into sound every second. The reason why the sound quality of CD is better than MP3 is that CD has more information than MP3 in every second. For example, compared with a CD file of 1411Kbps for a 128Kbps MP3 file, the amount of data converted per second is nearly 12 times less than that of a CD. The same song, CD sounds much more delicate, MP3 uses less data to express the same content, and its level of detail is of course not as good as CD.
Sampling rate is also a very common term. The specific expression is "XXHz", where "XX" is a specific number. For example, "44100Hz (44.1KHz)", "32000Hz (32KHz)" and so on. As mentioned before, digital audio files are composed of many “points”, so the sampling rate is actually a “number” standard for collecting these “points”. Obviously the sampling rate of "44100Hz" is higher than that of "32000Hz", so more points are collected per unit time (1 second). The more points per unit time, the more complete the sound information, and of course the closer it is to reality. Therefore, if the rate is the same, the file of "44100Hz" is better than "32000Hz".
The compression algorithm is divided into lossless compression and lossy compression. If the data accuracy requirements are relatively high, then choose lossless compression, but the compression ratio is relatively low, and the compression is actually similar to the uncompressed. The compression ratio of lossless compression is very high. Take MP3 as an example. It can achieve a compression ratio of 1:10 at a bit rate of 128Kbit/s. Of course, the price to pay for this is that the compressed data cannot be restored to the original data. This kind of compression algorithm is usually used in the audio and video fields. The compression algorithm designed for human physiological characteristics can have a high compression ratio under the premise of ensuring the effect.
For the "lossless audio" we most often say, it generally refers to the file format with a sampling rate of 16bit/44.1kHz in the traditional CD format. It is known as lossless compression because it contains 20Hz-22.05kHz. It is named after the frequency response frequency that completely covers the audible range of the human ear.
Lossless audio compression may be unfamiliar to everyone, but this does not mean that the lossless compression technology is not well developed. On the contrary, in the field of lossless compression, there have long been many excellent works, such as APE, FLAC, WavPack, LPAC, WMALossless, AppleLossless, La, OptimFOG, Shoten, etc.
The full name of WMA is WindowsMedia Audio, which is an audio format promoted by Microsoft. The WMA format achieves a higher compression ratio by reducing data traffic but maintaining sound quality. The compression ratio can generally reach 1:18, and the generated file size is only half of the corresponding MP3 file.
The sound quality and volume characteristics of WMA and MP3 can be summarized as follows: At low bit rate (less than 128Kbps), WMA has a smaller volume than MP3, and the sound quality is better than MP3; while at high bit rate (greater than 128K), MP3’s sound quality is Better than WMA.
Compared with MP3, the biggest feature of WMA is that it has strong protectability. It can be said that the introduction of WMA is aimed at the shortcomings of MP3 without copyright protection. WMA can be added to prevent copying through the DRM (Digital Rights Management) solution, or to limit the playback time and number of playbacks, or even the restrictions of the playback machine, which can effectively prevent piracy.
The full name of MP3 is Moving Picture Experts Group Audio Layer Ⅲ. To put it simply, MP3 is an audio compression technology. Since the full name of this compression method is MPEGAudio Layer 3, people call it MP3 for short. It was born in 1993, and its "parents" are Faunhofe IIS (Faunhofe IIS) and Thomson (Thomson) in France.
MP3 uses MPEGAudio Layer 3 technology to compress music into files with a smaller capacity at a compression ratio of 1:10 or even 1:12. In other words, it can compress files to a smaller size with little loss of sound quality. Degree. It also maintains the original sound quality very well. It is precisely because of the small size and high sound quality of MP3 that the MP3 format has almost become synonymous with online music. The MP3 format of music per minute is only about 1MB in size, so the size of each song is only 3-4 megabytes. Use an MP3 player to decompress (decode) MP3 files in real time, so that high-quality MP3 music can be played.
MP3 coding quality is divided into: fixed bit rate (CBR), average bit rate (ABR) and dynamic bit rate (VBR). The early MP3 encoding technology was not perfect. For a long time, most people used 128Kbps CBR (fixed encoding rate) format to encode MP3 files. Until recently, VBR (variable encoding rate) and ABR (average Encoding rate) compression method appeared, encoding bit rate up to 320Kbps, MP3 files began to improve in sound quality, and the appearance of LAME brought a qualitative leap to this progress.
The full name of Ogg should be OGGVobis (oggVorbis), 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. OGGVobis has a very outstanding feature, that is, it supports multi-channel. With its popularity, listening to DTS-encoded multi-channel works with a Walkman will not be a dream in the future.
Vorbis is the name of this audio compression mechanism, and Ogg is the name of a project that intends to design a completely open multimedia system. At present, the plan only implements OggVorbis.
The extension of Ogg Vorbis files is. OGG. The design format of this file is very advanced. OGG files created now can be used in future
AAC is actually an abbreviation for Advanced Audio Coding. AAC is an audio format jointly developed by Fraunhofer IIS-A, Dolby and AT&T. It is part of the MPEG-2 specification. The algorithm used by AAC is different from that of MP3. AAC combines other functions to improve coding efficiency. AAC's audio algorithm far exceeds some previous compression algorithms (such as MP3, etc.) in compression capabilities. It also supports up to 48 audio tracks, 15 low-frequency audio tracks, more sample rates and bit rates, multi-language compatibility, and higher decoding efficiency. In short, AAC can provide better sound quality while being 30% smaller than MP3 files.
M4A is the file extension of the MPEG4 audio standard. Mentioned in the MPEG4 standard, the common MPEG4 file extension is. mp4. Since Apple began to use it in its iTunes and iPod. Since m4a has distinguished between MPEG4 video and audio files. The extension m4a has become popular. Currently, almost all software that supports MPEG4 audio supports it. m4a. most commonly used. The m4a file is in AAC format (file), but other formats, such as AppleLossless or even mp3 can also be placed. In the m4a container (TC Note: The concept of this container is similar to the .mkv file). It is safe to put audio only. Change the extension of mp4 file to. m4a so that it can be played in your favorite player, and vice versa.
FLAC is the abbreviation of FreeLossless Audio Codec, the full name should be called OGGFLAC, Chinese can be interpreted as lossless audio compression code. It is part of the OGG project, which of course is open source and free. It is no wonder that it has been supported by many MP3 manufacturers so quickly.
FLAC is a well-known set of free audio compression coding, which is characterized by lossless compression. The FLAC compression ratio can reach 2:1, which is a very high ratio for lossless compression; and it has a fast decoding speed. It only needs to perform integer operations to complete the entire decoding process, which requires very low CPU computing power. , So ordinary Walkman can easily realize real-time decoding.
Unlike other lossy compression codes such as MP3 and AAC, it will not destroy any original audio information, so it can restore the sound quality of music CDs. It is now supported by many software and hardware audio products. In short, FLAC is similar to MP3, but it is losslessly compressed, which means that audio is compressed in FLAC mode without losing any information. This compression is similar to the Zip method, but FLAC will give you a larger compression ratio because
FLAC is a compression method designed specifically for the characteristics of audio, and you can use the player to play FLAC compressed files, just like you usually play your MP3 files.
APE is a currently popular digital music file format produced by Monkey’s Audio. It appeared earlier than FLAC, and it is also more famous than FLAC. Different from lossy compression methods such as MP3 and OGG, APE is currently the only recognized audio lossless compression format in the world, which means that when you compress the audio data file read from an audio CD into APE format, you still You can restore the APE format file again, and the restored music file is exactly the same as before compression, without any loss. And now more and more people spread it on the Internet, because the compressed APE file capacity is more than half smaller than the WAV source file, which can save the time for transmission and make it easier to spread! As the sampling rate of APE is as high as 800kbps～1400kbps, which is close to 1411.2kbps of music CD and much higher than 128kbps of MP3, its compressed sound quality is almost the same as the sound quality of the source file, and its sound quality has been strict The blind listening test has been recognized by enthusiasts all over the world. These features of APE are what other lossless compression formats are competing to emulate.
Before the advent of APE, music fans believed that it is the best way to save their favorite music materials in CD or WAV, but the advent of APE is enough to make them change this view, because APE can keep the music signal lossless , And can compress WAV files with a much higher compression ratio (close to 2:1) than WAV, and can play directly without decompression. Since the compressed APE file is only about half the size of the original file, the APE format is loved by many music lovers, especially for friends who want to transmit audio CDs over the network, APE can help them save a lot of resources. APE is so popular that it is easier to download APE format files online.