Comparison and implementation of MP3, WAV


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Comparison and implementation of MP3, WAV

WAV vs MP3
WAV vs MP3

Sound has three elements: pitch, volume, and timbre:

WAV vs MP3
WAV vs MP3

Pitch is determined by the frequency of the sound wave, the higher the frequency, the higher the pitch.
The volume is determined by the amplitude of the sound wave, the larger the amplitude, the louder the sound.
The timbre is determined by the “shape” of the waveform (sounds like square, triangle, and sawtooth are called impulse waves and sound individual).
An audio file is a file obtained by converting an analog signal to a digital signal. In general, there are 5 important parameters: encoding method, number of channels, sampling rate, bit depth, and bit rate.

Encoding: how this format organizes binary data and how it is compressed.
Number of channels: mono, dual or 5.1 channels, etc.
Sampling rate: The number of samples per second.
Bit Depth: The number of binary bits used to store the y value of the sample point.
Bitrate – The desired number of bits per second for the file.
We know that there is no compression in the WAV format, so its encoding method is to directly write all the sampled points to the file in order.

WAV file size (B) = number of channels * sample rate (Hz) * bit depth (bit) / 8 + the file header size (B, it’s 44B)

Implementation
When you open an mp3 or wav file with a text editor, you see numbers like this:
4944 3303 0000 0000 3d48 5459 4552 0000
0006 0000 0032 3031 3800 5444 4154 0000
0006 0000 0032 3230 3300 5449 4d45 0000
0006 0000 0031 3430 3600 5052 4956 0000
168e 0000 584d 5000 3c3f 7870 6163 6b65
7420 6265 6769 6e3d 22ef bbbf 2220 6964
3d22 5735 4D30 4D70 4365 6869 487A 7265
537A 4E54 637A 6B63 3964 223F 3E0A 3A78
6D70 6D65 7461 2078 6D6C 6E78 3D22
6F62 653A 6574 612F
5249 4646 2e3d 0e05 5741 5645 666d 7420
1200 0000 0300 0200 44ac 0000 2062 0500
0800 2000 0000 6461 7461 a026 0e05 8089
00bc 00e8 f0bb c09e 8dbc 00c2 87bc 80f1
d3bc 8063 ccbc c030 fcbc 8012 f4bc 20bb
13bd e051 0fbd c0b0 2dbd 6079 28bd 4012
46bd 6032 40bd c0e3 5dbd 6040 57bd c015
7cbd e035 74bd b058 8dbd 50e2 88bd f0a7 9dbd e0dd 98bd 70d3 acbd e0a9 a7bd
d043 b8bd b0da b2bd
00e3 c4bd 605c bfbd
This one above is the mp3/wav format of the same song. What is the difference between them?

WAV
structure
file header
The WAV format follows the RIFF Resource Interchange File Format, so the WAV format is actually a three-layer relationship, which is simplified here. Its file header format is as follows:

Address Carving type content
00H-03H 4 character * 4 RIFF resource file exchange flag
04H-07H 4 unsigned int The number of bytes from the next address to the end of the file.
08H-0BH 4 character * 4 WAV file WAVE logo
0CH-0FH 4 character * 4 fmt wave file flag, the last digit is 0x20 space
10H-13H 4 unsigned int The size of the subchunk file header. For the WAV subfragment, the value is 0x10.
14H-15H 2 short unsigned Format type, when the value is 1, it means the data is linear PCM encoding
16H-17H 2 short unsigned number of channels
18H-1BH 4 int unsigned Sampling rate
1CH-1FH 4 int unsigned Wave file bytes per second = sample rate Bit depth PCM / 8 channels
20H-21H 2 short unsigned DATA data block unit length = number of channels * PCM bit depth / 8
22H-23H 2 short unsigned Bit depth PCM
24H-27H 4 character * 4 data stamp data
28H-2BH 4 unsigned int Total length of data part (bytes)
struct WAVHeader
{ char RIFF[ 4 ]; ///Resource file exchange flag RIFF unsigned LEN; ///Number of bytes from the next address to the end of the file char WAV[ 4 ]; ///WAV file flag WAVE char FMT [ 4 ]; ///Wave fmt file pointer, last digit is 0x20 space unsigned SubchunkSize; ///The size of the sub-chunk file header, for WAV this sub-chunk, the value is 0x10 DATATYPE short unsigned; / //Format type, when the value is 1, it means the data is unsigned linear PCM encoding short CH ; ///Number of unsigned channels F; ///Unsigned sample rate BYTERATE; ///Number of bytes per second of wave file = sample rate*PCM bit depth/8*Number of unsigned channels

short DATAUNITLEN; ///DATA block unit length=channel number*PCM bit depth/unsigned 8 short BITDEPTH; ///PCM bit depth character DATA[ 4 ]; ///Unsigned data mark data DATALEN ; ///Total data section length (bytes) };


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Comparison and implementation of MP3, WAV

Comparison and implementation of MP3, WAV

WAV vs. MP3
WAV vs. MP3

An mp3 is 320kbps, 44100hz, what does this mean?

WAV vs. MP3
WAV vs. MP3

44100Hz represents the sample rate of the signal. The so-called sampling consists of obtaining the value y of the sound wave at the current moment every unit of time. Sampling is the process of discretizing continuous data (converting an analog signal to a digital signal).
image source

The sampling method mentioned above is called PCM (Pulse Code Modulation). According to the Nyquist-Shannon sampling law, the sampling rate must be at least twice the highest target frequency. The hearing range of the human ear is about 20Hz-20,000Hz (if you’re curious how loud you can hear, you can click here to test your ears), although recording software often has a 48,000 option Hz, but we can safely conclude: 44100Hz can meet almost all our needs, higher is just a waste of your memory and CPU. More than 48,000 samples are meaningless to the human ear, which is similar to 24 frames per second on a movie. 44100Hz happens to be the standard sample rate for almost all music released. In fact, for vocals and many instruments, high-frequency sounds are noise, so high sample rates can sometimes worsen sound quality (which is why we need to adjust the equalizer).

320kbps represents your bit rate/bit rate, which is short for kilobits per second, which represents the size of the data used to describe sound. In CD (uncompressed audio file), the bit rate is 1411.2 kbps, and the mp3 sound quality to achieve CD quality should be higher than 128 kbps / 44100 Hz (128 kbps can be said to be the most common bit rate). Generally, a higher number means better quality. The quality depends on many factors (such as the encoding algorithm). Many times we don’t need too high bitrate: our device can play mp3 and CD without difference (sound/sound card is normal).

A wav is 44100 Hz 16-bit stereo or 22050 Hz 8-bit mono, what does this mean? stereo/mono refers to dual/mono. For monophonic sound files, the sample data is an eight-bit short integer (short int 00H-FFH); for two-channel stereo sound files, each sample data is a 16-bit integer (int) and the upper eight bits (left channel) and lower eight bits (right channel) represent the two channels, respectively.

Sound is a mechanical wave, produced by the vibration of an object, and requires a medium to propagate. So, in essence, a sound is a waveform on an axis over time.

Mp3 (an audio encoding method) Part 3

Mp3 (an audio encoding method) Part 3

MP3 ENCODING

To generate bit-compliant (Layer 1.Layer 2.Layer 3) MPEGAudio files, ISO MPEG Audio committee members developed reference simulation software in C called ISO 11172-5.

MP3 ENCODING

It can demonstrate the first real-time DSP-based hardware decoding of compressed audio on some non-real-time operating systems. Various other MPEG audio was developed in real time for digital broadcasting (DAB radio and DVB TV) for consumer receivers and set-top boxes.
Later on July 7, 1994, Fraunhofer-Gesellschaft released the first MP3 encoder called l3enc.
The Fraunhofer development team selected the .mp3 extension on July 14, 1995 (previously the extension was .bit). Using Winplay3 (released September 9, 1995), the first real-time software MP3 player, many people were able to encode and play MP3 files on their own personal computers. Since hard drives at the time were relatively small (such as 500MB), this technology was essential for storing entertainment music on computers.
MP2, MP3 and Internet
In October 1993, MP2 (MPEG-1 Audio Layer 2) files appeared on the Internet and were often played by Xing MPEG Audio Player and later MAPlay developed by Tobias Bading for Unix. MAPplay was first released on February 22, 1994 and ported to the Microsoft Windows platform.
The only MP2 encoder products at first were Xing Encoder and CDDA2WAV, a CD ripper that converts audio tracks from CDs to WAV format.
Often considered the father of the online music revolution, the Internet Underground Music Archive (IUMA) was the first hi-fi music site on the Internet, with thousands of licensed MP2 recordings before MP3 and the web became popular. .
From the first half of 1995 to the end of the 1990s, MP3 began to flourish on the Internet. MP3’s popularity is largely due to the success of companies and software packages such as Winamp released by Nullsoft in 1997 and Napster released by Napster in 1999, and they are mutually reinforcing. These programs make it easy for normal users to play, create, share and collect MP3 files.
The debate about sharing MP3 files between peers has spread rapidly in recent years, mainly because compression makes file sharing possible, uncompressed files are too large to share. Since MP3 files are widely spread over the Internet, Napster has been sued by some of the major record labels to protect their copyright (see Copyright).
Commercial online music distribution services, such as the iTunes Music Store, often choose other proprietary or DRM-enabled music file formats to control and limit the use of digital music. Formats that support DRM are used to protect copyrighted material from copyright infringement, but most protection mechanisms can be broken in some way. Computer experts can use these methods to generate unlocked files that can be freely copied. One notable exception is Microsoft’s Windows Media Audio 10 format, which has yet to be cracked. If a compressed audio file is desired, the recorded audio stream must be compressed and the sound quality will be degraded.
streaming audio quality
Because MP3 is a lossy compression format, it offers a variety of options for different “bit rates,” that is, the number of encoded data bits needed to represent the audio per second. Typical speeds are between 128 kbps and 320 kbps (kbit/s). In contrast, the uncompressed audio bitrate on a CD is 1411.2 kbps (16 bits/sample × 44100 samples/sec × 2 channels).
MP3 files encoded with lower bit rates generally play at a lower quality. If you use too low a bitrate, “compression artifact” (sounds not present in the original recording) will appear during playback. A good example of compression noise is the sound of compressed cheering; due to its randomness and sharp changes, encoder errors are more pronounced and sound like echoes.

Mp3 (an audio encoding method) Part 2

Mp3 (an audio encoding method) Part 2

mp3 3ncoding

MPEG-1 Audio Layer 2 encoding began as a digital audio broadcast (DAB) managed by Egon Meier-Engelen at the German Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later known as Deutsches Zentrum für Luft- und Raumfahrt, German Space Center). )draft.

mp3 encoding

This project is funded by the European Union as a EUREKA research project, and its name is commonly known as EU-147. The study period for EU-147 was from 1987 to 1994.
2. By 1991, two proposals had emerged: Musicam (called Layer 2) and ASPEC (Adaptive Spectrum Sensing Entropy Coding). The Musicam method proposed by Philips of the Netherlands, CCETT of France, and the Institut für Rundfunktechnik of Germany was chosen due to its simplicity, error robustness, and lower computational effort in high-quality compression. The Musicam format based on subband coding is a key factor in determining the MPEG audio compression format (sample rate, frame structure, header, sample points per frame). This technology and its design philosophy are fully integrated into the definition of ISO MPEG Audio Layer I, II and later Layer III (MP3) formats. The standard was developed by Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II) under the auspices of Prof. Mussmann (University of Hannover).
3. A working group consisting of Leon Van de Kerkhof from the Netherlands, Gerhard Stoll from Germany, Yves-François Dehery from France and Karlheinz Brandenburg from Germany absorbed design ideas from Musicam and ASPEC and added their own design ideas to develop an MP3. MP3 can achieve MP2 sound quality from 192 kbit/s to 128 kbit/s.
4. All of these algorithms eventually became part of the first group of MPEG standards, MPEG-1, in 1992, resulting in the international standard ISO/IEC 11172-3 published in 1993. Further work on MPEG audio was eventually became part of the MPEG-2 standard, a second group of MPEG standards developed in 1994, officially known as ISO/IEC 13818-3, first published in 1995.
5. The compression efficiency of the encoder is generally defined by the bit rate, because the compression rate depends on the number of bits (: in: bit depth) and the sampling rate of the input signal. However, there are often products that use CD parameters (44.1 kHz, two channels, 16 bits per channel, or 2×16 bits) as the compression ratio reference, and the compression ratio using this reference is usually higher, which which also shows that the compression ratio is very important for lossy compression problems.
6. Karlheinz Brandenburg used Suzanne Vega’s song Tom’s Diner on CD to test MP3 compression algorithms. This song is used because the song’s smooth and simple melody makes it easier to hear glitches in the compressed format during playback. Some jokingly refer to Suzanne Vega as “the mother of MP3”. Some more serious and critical audio extracts (glockenspiel, triangle, accordion…) from the EBU V3/SQAM reference CD are used by professional audio engineers to assess the subjective perceived quality of the MPEG audio format.

Mp3 (an audio encoding method)

Mp3 (an audio encoding method)

Mp3 encxoding

MP3 is an audio compression technology, its full name is Moving Picture Experts Group Audio Layer III, called MP3.

mp3 encoding

It is designed to drastically reduce the amount of audio data. Using MPEG Audio Layer 3 technology, music is compressed into a smaller capacity file with a compression ratio of 1:10 or even 1:12, and for most users, playback quality is not as good as the original uncompressed. audio Significant decrease. It was invented and standardized in 1991 by a group of engineers at the Fraunhofer-Gesellschaft research organization in Erlangen, Germany. Music stored in the form of MP3 is called MP3 music, and a machine that can play MP3 music is called an MP3 player.

Motion Picture Expert Compression Standard Audio Layer 3 foreign name Moving Picture Expert Group Audio Layer III research organization Fraunhofer-Gesellschaft type audio coding advantage Drastically reduce the amount of audio data defect sound quality loss
content
1 Features
2 story
▪ origin
▪ go to the masses
3 audio quality
4 patent issues
transmission characteristics
MP3 converts the time-domain waveform signal to a frequency-domain signal by taking advantage of the human ear’s insensitivity to high-frequency sound signals and splits it into multiple frequency bands, using different compression rates. for different frequency bands and increasing the compression ratio for high frequencies (even ignoring the signal) Use a small compression ratio for low frequency signals to ensure that the signal is not distorted. In this way, it is equivalent to discarding the high-frequency sound that is basically inaudible to the human ear [1], keeping only the audible low-frequency part, thus compressing the sound with a compression ratio of 1:10 or even 1: 12. Because the full name of this compression method is called MPEG Audio Player3, people call it MP3 for short.
According to the MPEG specification, AAC (Advanced Audio Coding) in MPEG-4 will be the next generation of the MP3 format.
Compared to CD, FLAC and APE lossless compression formats, the sound quality of the highest parameter MP3 (320 Kbps) is not much different.
MP3 players are dying
When they first came out, MP3 players were at the forefront of the digital revolution. However, sales of iPods and other MP3 players in the UK fell sharply in 2012 as consumers turned to other digital products such as smartphones.
In 2012, sales of MP3 players in the UK market were £110m ($178m), just 29% of the £381m in 2011, according to market research firm Mintel. Mintel expects total MP3 player sales in the UK market to halve by 2017. In the worst case scenario, total MP3 player sales in the UK market will be just 25 million dollars five years later. [23]
1. MP3 is a data compression format;
2. Discards pulse code modulation (PCM) audio data that is not important to the human ear (similar to JPEG is a lossy image compression), resulting in a much smaller file size;
3. MP3 audio can be compressed according to different bit rates, providing a variety of trade-offs between data size and sound quality. The MP3 format uses a mixed conversion mechanism to convert audio domain signals. time in frequency domain signals;
4. 32 band polyphase integral filter (PQF);
Modified discrete cosine filter (MDCT) of 5, 36 or 12 taps; each subband size can be independently selected between 0…1 and 2…31;
6. MP3 not only has extensive client software support, but also has a lot of hardware support, such as portable media players (referring to MP3 players), DVD and CD players, outgoing calls