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10 Common Audio Formats: Which Format Should I Choose?

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10 Common Audio Formats: Which Format Should I Choose?

Audio files come in various formats and sizes, in addition to our common MP3, there are AAC, FLAC, OGG, WMA, etc.

Why are there so many different audio standards? What format is the best? What should you choose when saving audio files?

In fact, all audio formats can be divided into 3 categories, and once you understand the characteristics of each of these 3 types, you can choose one of the three, and then choose a further subdivided category according to your specific needs.

Comparison of audio formats – operation, equipment – Introduction to 10 common audio formats: which format should I choose?
uncompressed audio format
The uncompressed format preserves the original audio waveform. The so-called original audio waveform refers to the result after the sound is captured and converted to a digital signal for storage. Raw audio waveforms are stored without any post processing. From the results, such an audio file will consume a lot of space. For 24-bit (24-bit) 96 KHz audio, you need about 34 MB of storage space per minute.

Uncompressed audio format: PCM
PCM is short for Pulse-Code Modulation, which is a technical method of simulating sound using digital signals. The sounds that we can hear in nature exist in the form of physical sound waves, and in order to simulate this sound wave with a digital signal, scientists decided to sample it and record it with another wave, which is what we call a pulse.

Therefore, digital audio has two basic concepts: sample rate and bit depth. The so-called sampling rate refers to the speed of how many times the sound is sampled. The so-called bit depth refers to the space occupied by a single sample. In uncompressed formats, the product of the sample rate, bit depth, and audio duration is the space occupied by the digital signal that represents the audio content.

PCM is a format commonly used on CDs and DVDs, but its derivative LPCM is often used in practice. LPCM is short for Linear PCM, which is Linear Pulse Code Modulation. Most PCM files in use today are actually LPCM files, so there is no need to differentiate between them.

Uncompressed audio format: WAV
WAV is short for Waveform Audio File Format, which is a standard audio format developed by Microsoft and IBM in 1991.

Many people think that WAV is an uncompressed audio format, but this is not entirely true. In fact, WAV is a “wrapper” type format, a common format developed by Microsoft to use audio for various platforms. So it is possible for WAV to contain compressed audio, but this is rarely the format of choice when using compression these days.

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What are the common audio formats?

Most frequent and popular sound formats

05 AAC format

AAC is actually short for Advanced Audio Coding. The algorithm adopted by AAC is different from that of MP3. AAC improves encoding efficiency by combining other features. The AAC audio algorithm far surpasses some previous compression algorithms (such as MP3, etc.) in terms of compression capabilities. In conclusion, AAC can provide better sound quality with 30% smaller file size than MP3 files.

It is one of the best lossy formats out there. There are multiple encodings, with bit rates up to 448 kbps. In terms of hardware support, advanced mobile phones and mp3 players are now generally supported.

06 VQF format

The VQF format is an audio compression technology jointly developed by YAMAHA and NTT. Its compression ratio can reach 1:18. Therefore, under the same circumstances, the compressed VQF file size is 30%-50% smaller than MP3, which is more suitable for online streaming, while the sound quality is excellent, near to CD quality (16-bit, 44.1 kHz stereo). But VQF has not revealed the technical standards, so far it has not become popular.

07 FLAC format

FLAC is short for Free Lossless Audio Codec, which can be interpreted as Chinese lossless audio compression coding, which is characterized by lossless compression. Unlike other lossy compression codes like MP3 and AAC, it can restore the sound quality of music discs. It is now compatible with many software and hardware audio products.
It can be said that FLAC is similar to MP3, but it is losslessly compressed, which means that the audio is compressed in FLAC without losing any information. And you can use the player to play compressed FLAC files, just like playing MP3 files.

08 APE format

APE is one of the most popular digital music file formats. Unlike lossy compression methods like MP3, APE is a lossless audio compression technology, which means that after compressing the audio data file read from the audio CD in APE format, you can also compress the file in APE format. The audio file is exactly the same as before compression, without any loss.

The file size of APE is about half that of CD, which is loved by many music lovers, especially for friends who want to stream audio CDs over the network, APE can help them save a lot of resources.

What are the common audio formats?

Most frequent and popular sound formats

01 MP3 format

MP3 is one of the most common audio formats. Its full name is Moving Picture Experts Group Audio Layer III. Simply put, MP3 is an audio compression technology that uses MPEG Audio Layer 3 technology to compress music into a smaller capacity file with a compression ratio of 1:10 or even 1:12. Compress files to a lesser extent in small cases. And it keeps the original sound quality very well.

It is precisely because of MP3’s small size and high sound quality that the MP3 format has become almost synonymous with online music. Use an MP3 player to uncompress (decode) MP3 files in real time so that high-quality MP3 music can be played. The highest bitrate is 320K, and the high-frequency part is one size fits all, which is its default.

02 WMA format

The full name of WMA is Windows Media Audio, which is an audio format promoted by Microsoft. The WMA format achieves a higher compression ratio by reducing the data stream while maintaining sound quality. The compression ratio can usually reach 1:18, and the generated file size is only half of the corresponding MP3 file. In addition, WMA can also be added to prevent copying through the DRM scheme, or add restrictions on playback time and playback times, and even restrictions on playback machines, which can effectively prevent piracy.

03 WAV format

WAV format is a lossless sound format developed by Microsoft Corporation, also known as waveform sound file. It is the oldest digital audio format and is widely supported by the Windows platform and its applications. The WAV format supports many compression algorithms, supports a variety of audio bits, sample rates, and channels. It adopts 44.1kHz 16-bit sampling rate. Therefore, the sound quality of WAV is almost the same as that of CD, and it is suitable for uploading to all major music platforms. But the WAV format is bulky and difficult to spread.

04 ASF format

The full name of ASF is Advanced Streaming Format, which is a media playback format developed by Microsoft and is suitable for playback on the Internet.

All the Audio Format Differences: Which One Should You Use?

Three classifications of audio formats

Characteristics and differences of different audio formats

Which audio format is right for you?

You already know MP3, but what about AAC, FLAC, OGG or WMA? Why are there so many audio file formats and is there any best audio format?

Comparison of audio formats

Audio files come in various types and sizes. While we’re all probably familiar with MP3, what about AAC, FLAC, OGG, or WMA? Why are there so many audio standards? Is there a better audio format? Which ones are important and which ones can be ignored?

All audio formats are divided into three main categories, and once you know what the categories mean, you can choose the format within the categories that best suits your needs.

uncompressed audio format

Uncompressed audio consists of actual sound waves that have been captured and converted to digital format without any additional processing. So uncompressed audio files tend to be the most accurate, but take up a lot of disk space: about 34MB per minute for 24-bit 96KHz stereo.

Audio file format: PCM

PCM stands for Pulse Code Modulation, a digital representation of the original analog audio signal. Analog sounds exist as waveforms. To convert a waveform into digital bits, the sound must be sampled and recorded at specific intervals (or pulses).

This digital audio format has a “sample rate” (how often the samples are made) and a “bit depth” (how many bits are used to represent each sample). There is no compression involved. Digital recordings are almost accurate representations of analog sounds.

Digital Audio Quality

Data rate refers to the data flow used by a video file in a unit of time, also called bit rate or bit stream rate.

The popular interpretation is the sampling rate, which is the most important part of image quality control in video encoding. Generally, the units we use are kb/s or Mb/s. Generally speaking, at the same resolution, the higher the code stream of the video file, the lower the compression ratio and the higher the image quality. The higher the code stream, the higher the sampling rate per unit time, the higher the data stream, the higher the accuracy, the closer the processed file is to the original file, the better the image quality, the clearer the image quality and the higher the decoding capability of the playback device is required.

Of course, the larger the code stream, the larger the file size. The calculation formula is file size = time X code rate/8. For example, a 720P RMVB file with a 1 Mbps stream of 90 minutes is common on the Internet and its volume is = 5400 seconds × 1 Mb/8 = 675 MB.

Generally speaking, a video file includes images and sounds, just like an RMVB video file, which contains video information and audio information. Audio and video have their own sampling methods and different bit rates, that is, the same video Audio and video file bit rate is not the same. And what we’re talking about is the bitrate of a video file, which generally refers to the sum of the bitrate of the audio and video information in the video file.

Taking the most popular and familiar RMVB video file in China as an example, VB in RMVB refers to VBR, which is short for Variable Bit Rate. The Chinese meaning is variable bit rate, which means that RMVB adopts dynamic encoding. In this way, a higher sample rate is used for complex dynamic images (singing and dancing, flying cars, wars, actions, etc.), while a lower sample rate is used for static images, and the resources are use rationally to achieve image quality and volume .Effect.

The most fundamental difference between code rate and sample rate is that the code rate is for the source file.

2. Sampling rate

Sample rate (also called sample rate or sample rate) defines the number of samples per second taken from a continuous signal to form a discrete signal, and is expressed in hertz (Hz). Sampling rate refers to the sampling frequency when converting an analog signal to a digital signal, i.e. how many points are sampled per unit of time. How many bits are in the data for a sample point? Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (bit per second). The higher the bitrate, the more data transmitted and the better the sound quality. Bit rate = sample rate x number of bits used x number of channels.

The sample rate is similar to the number of frames of moving images. For example, the sampling rate of movies is 24 Hz, the sampling rate of PAL format is 25 Hz, and the sampling rate of NTSC format is 30 Hz. When we play back the still images sampled at the same rate as the sampling frequency, we see a continuous image. In the same way, when a CD recorded at a sampling rate of 44.1 kHz is played back at the same rate, a continuous sound can be heard. Obviously, the higher the sample rate, the more coherent the sound will be heard and the picture will be seen. Of course, the sampling rate that human auditory and visual organs can distinguish is limited, which is basically higher than sound sampled at 44.1 kHz, and most people haven’t noticed the difference.

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the highest digits can only be distinguished by instruments.

Detailed Music Format Part 2

Music CD

That is, CD records. A CD can play sound files of approximately 74 minutes. The Windows system comes with a CD player. Also, the software that comes with most sound cards provides CD playback functionality, and even some CD-ROM drives are offline. from computer Can be used as a stand-alone CD player when powered on.

WMA with unlimited potential

In developing its own network media service platform, Microsoft primarily promotes ASF (Audio Streaming Format), which is an open standard that supports data transmission over various networks and protocols. It supports audio, video, and a variety of other types of multimedia. And WMA is short for Windows Media Audio, which is equivalent to an ASF file that contains only audio.
The compression ratio of WMA files can be as high as 1:18 in 80kbps 44kHz mode, which is basically the same as VQF. And the compression speed is doubled compared to MP3. So it should be more competitive than VQF.

Vorbis free music format

To avoid rising royalties charged by MP3 music companies, programmers at GMGI’s iCast company developed a new free music format, Vorbis, that rivals or even exceeds MP3 in sound quality. And it will be released over the internet and can be downloaded for free without worrying about infringement issues. But MP3 has become very popular on the Internet, and Microsoft’s Windows Media technology has also started to spread, and Vorbis’s outlook is still not optimistic.

Other audio formats

AIF/AIFF: A sound file format developed by Apple, supported by the MAC platform, and supports 16-bit stereo at 44.1 kHz.
AU: SUN’s AU Compressed Sound File Format, which only supports 8-bit sound, is a commonly used sound file format on the Internet, mainly created by SUN workstations.
CDA: CD audio track file.
CMF: A MIDI-like sound file developed by CREATIVE.
DSP: Abbreviation for digital signal processing. By improving the signal processing method, sound quality will be greatly improved and songs will be more pleasing to the ear.
S3U: MP3 playback file list
RMI: MIDI Instrument Sequence

Lossy compression:

AAC – Sound quality is second only to MPC at high bit rates and looks good at both high and low bit rates. The encoding speed is too slow!
MPC: Performance is average at low bitrate, not as good as MP3 and OGG encoded by Mp3Pro, sound quality is best at high bitrate, and encoding speed is
fast.OGG: The sound quality is better at a low bitrate, and the same is true at a high bitrate. Encoding is slightly slower.
MP3 (MP3Pro): Sound quality is lower than OGG at low bit rate and other aspects are the same as MP3
WMA: High and low bit rates are average, VBR is not supported and the highest is 192Kbit/s

lossless compression:

FLAC – Worst compression ratio of the four, decent encoding speed, good platform support.
PAC: Slightly slower encoding speed, third in compression ratio, good platform support.
APE: The fastest encoding speed, the best compression rate, and the platform is generally supported.
WV: The encoding speed is very fast, the compression rate is second among the four types, and it is only supported by the Windows platform.

Detailed music format

classic wave

As the most classic Windows media audio format, the WAVE file is widely used, which uses three parameters to represent sound: the number of sampled bits, the sample rate, and the number of channels.
The channels are divided into mono and stereo, and the sample rates are generally 11025 Hz (11 kHz), 22050 Hz (22 kHz), and 44100 Hz (44 kHz). The capacity occupied by the WAVE file = (sampling frequency × sampling bits × channel) × time/8 (1 byte = 8 bits).

traditional mod

MOD is a wavetable-like music format, but its structure is similar to MIDI, it uses real samples, and the volume is small. In the earlier DOS era, MOD was often used as background music for games. Modern mods can contain many audio tracks in many formats, such as S3M, NST, 669, MTM, XM, IT, XT, and RT.

midi music computer

MIDI is short for Musical Instrument Data Interface. Records the sound played by the instrument digitally (each note is recorded as a number), and then synthesizes these records via FM or wavetable during playback: FM synthesis is the sound of the instrument is simulated by mixing the multi-frequency sounds; wavetable synthesis consists of storing the sound samples of the instrument in the wavetable of the sound card and extracting the sound from the wavetable as you play.

Boss Boss MP3

It can be said that MP3 is famous, it uses MPEG Audio Layer 3 technology to compress the sound with a compression ratio of 1:10 or even 1:12, with a sampling rate of 44kHz and a bit rate of 112kbit/s. .
MP3 music is music stored in digital form. If you want to play it, you must have a corresponding digital playback and decoding system. Generally, MP3 digital music is decoded by special software and then restored to a waveform sound signal for playback output. This type of software is called For MP3 players, such as Winamp, etc.

Overlord RA series online

RA, RAM, and RM are Real’s mature network audio formats, using “streaming audio” technology, making them well suited for network streaming. Information such as copyright, singer, producer, mail and song title can be added during production.
RA can be called the supreme lord of multimedia communication on the Internet. It is suitable for streaming on the Internet and is currently the best format for listening to online music online.

VQF with high compression ratio

VQF or TwinVQ is an audio compression technology developed by Nippon Telegraph and Telephone and Yamaha Corporation.
The audio compression rate of VQF is almost twice that of standard MPEG audio and can reach approximately 1:18 or even higher. And popular compression formats like MP3 and RA are usually only around 1:12. But it still won’t affect the sound quality, when VQF compress music at 44kHz-80kbit/s audio sampling rate, its sound quality will be better than 44kHz-128kbit/s MP3, when compress at 44kHz-96kbit/s , the music is close to 44kHz-256kbit/s MP3.

MD minidisc

MD (ie MiniDisc) is a comprehensive portable music format released by SONY in 1992. The compression algorithm it uses is ATRAC technology (the compression ratio is 1:5). MD is divided into Recordable MD (Recordable, with two heads of magnetic head and laser head) and Single Play MD (Prerecorded, only laser head).
The powerful editing function is the strong point of MD. You can quickly select tracks, move tracks, merge, split, delete and edit track titles. It is more personalized than CD and you can have your own MD album at any time. MD products include MD Walkman, MD bedside audio, MD car audio, MD recording deck, MD camera gun and MD driver, etc.

Principle of mp3 and file format analysis. Part4

The three bytes starting at 1397H are 54 41 47, which store the “TAG” character, indicating that this file has ID3 V1.0 information.

The 30 bytes starting at 139AH store the name of the song, the first 4 bytes that are not 00 are 54 45 53 54, which means “TEST”;
the 4 bytes starting at 13F4H are 04 19 14 03 and the year of storage is “04/25/2003” ”;
the last byte is 4E, which represents the music category, and the code name is 78, that is, “Rock&Roll”; the
other bytes are all 00, and no information is stored.

4 Conclusions
As an important multimedia data type, people are always looking for more efficient compression methods and new sound file formats. In the MP3 file, the MDCT transform is used, which is a quasi-optimal transform with a simple structure and easy programming, which avoids the problem that the optimal transform (KL) is difficult to solve for the eigenvalues and eigenvectors of the covariance. matrix.

Through the analysis of the MP3 file format, it is not difficult to find its shortcomings. Each frame of an MP3 file has the same 4-byte frame header, which requires some space overhead for an MP3 file with a large number of frames. ID3 stores the music description information. The proprietary, copyright, and other information in the frame header is also description information. The music description information is a bit messy.

In any case, the development of MP3 is unstoppable. MP3 has become a recognized sound data format. MP3 is becoming a hot spot in the field of multimedia information processing along with JPEG images and PDF documents.

Principle of mp3 and file format analysis. Part 3

The ID3 standard MP3 frame header does not consider storing complex information such as song title, author, album name, year, etc., except some simple music description information such as privacy, copyright and original, which are very necessary in MP3 applications.

In 1996, in the “Studio 3” project, FricKemp proposed to add description information for storing songs at the end of the MP3 file and formed the ID3 standard. Until now, ID3 V1.0, V1.1, V2 .0, V2, .3 and V2.4 standards have been formulated. The higher the version, the richer and more detailed the relevant information is recorded.
The ID3 V1.0 standard is not complete and the information stored is too small to store lyrics, album covers, images, etc. V2.0 is a fairly complete standard, but it brings difficulties in writing software, although there are many people in favor of this format, very few are actually implemented in software. The vast majority of MP3s still use the ID3 V1.0 standard. This standard uses the last 128 bytes at the end of the MP3 file to store ID3 information. See Table 3 for instructions on using these 128 bytes.
Table 3 Final ID3 V1.0 File Description
length in
byte (byte) Description
1-3 3 Stores the “TAG” character, which indicates the ID3 V1.0 standard, followed by the song information.
4-33 30 Song name
34-63 30 Author
64-93 30 Album name
94-97 4 Year
98-127 30 Notes
128 1 MP3 music category, a total of 147 types.

3.3 File example
Open a file called test.mp3 in VC++ with the following content:
000000 FF FB 52 8C 00 00 01 49 09 C5 05 24 60 00 2A C1
000010 19 40 A6 00 00 05 96 41 34 18 20 80 08 26 48 29
000020 83 04 00 01 61 41 40 50 04 00 C1 2 41 50 64
…
0000d0 Fe FF FB 52 80 01 EE 90 65 6E 02 30
0000E0 32 0C CD CD CD CD 46 16 41 89 B8 408 89 300 408
0000F0 33 B7 00 00 01 02 FF FF FF F4 E1 2F FF FF FF FF
……
0001A0 DF FF FF FF FB 52 8C 12 00 E 01 FE 90 58 6E 09 A0 02
000150 8513 B0 AC 45 F6 19 61 26 26
0001C0 05 AC B4 20 28 94 FF FF FF FF FF FF FF FF FF FF
…
001390 7F FF FF FF FD 4E 00 54 41 47 54 45 53 54 00 00
0013A0 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
001400
00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
00 00 00 00 00 00 00 00 00 00 00
001410 00 00 00 00 00 00 4E
File length is 1416H (5.142K), frame header is: FF FB 52 8C, converted to binary:
11111111 11111011
01010010
10001100T

Principle of mp3 and file format analysis. Part 2

MP3 uses perceptual audio coding (Perceptual Audio Coding) this distortion algorithm.

The frequency range of sound perceived by the human ear is 20 Hz to 20 kHz. MP3 cuts out a lot of redundant signals and irrelevant signals. The encoder transforms the original sound into the frequency domain through a mixed filter bank and uses a psychoacoustic model. to estimate that it may be only The perceived noise level is quantized and converted to Huffman coding to form an MP3 bit stream. The decoder is much simpler, its task is to extract the sound signal from the encoded spectral line components through inverse quantization and inverse transformation. The MP3 encoding and decoding process is shown in Figure 1.
2.4 Modified Discrete Cosine Transform The cosine transform
Modified Discrete CT (MDCT) refers to converting a time-domain data set to frequency-domain data in order to know the changes in the time domain. MDCT is an enhancement of the DCT algorithm. The first fast algorithm is fast Fourier transform (FFT), but FFT has complex operations, MDCT are real operations, easy to program.
When compressing audio data, first divide the original sound data into fixed blocks, and then perform direct MDCT (direct MDCT) to convert the value of each block into MDCT 512 coefficients. The 512 coefficients are restored to the original sound data, and The original before and after sound data is inconsistent because redundant and irrelevant data is removed during the compression process. The FMDCT transformation formula is:
k=0, 1,
.
n0=(N/2+1)/2, X(n) is the time domain value, X(k) is the frequency domain value. If N takes 1024 points, it becomes 512 frequency domain values.
The IMDCT transformation formula is:

n=0, 1, …, N-1
MDCT itself does not compress data, it simply maps the signal to another domain, and quantization compresses the data. When bit allocation is done on the quantized transformed samples, the entire quantized block must be considered the smallest, which is called lossy compression.
3 File Format Analysis
MP3 MP3 file data is made up of multiple frames, and the frame is the smallest unit of the MP3 file. Each frame, in turn, consists of a frame header, additional information, and sound data. The playback time of each frame is 0.026 seconds and its duration varies with the bit rate. Some MP3 files have extra bytes at the end that contain description information for non-audio data.