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Comparison and implementation of MP3, WAV

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Comparison and implementation of MP3, WAV

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

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

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

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).
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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 To wav converter

Mp3 to Wav Converter

Mp3 to wav

MP3 to WAV converter

Converting an mp3 to a wav is really easy. Most likely, the mp3 file itself was encoded from a wav.

Convert from Wav to Mp3

In fact, this whole matter of compression to mp3 format began with the search to save space, without losing quality. Why would you want to save space? Firstly to be able to stream the music file over the internet, in days when bandwidth was low, and secondly to save space at a time when available hard drives were quite limited in capacity. The technology did not exist at all neither on the internet nor on hard drives that we have today.

What’s the point of converting an mp3 back to a wav?

The mp3 must have discarded information from the original wav and that information will not be recovered if we do the reverse conversion, in this case from mp3 to wav.
Let us remember what was said above: it was necessary to reduce the size and to achieve this, psychoacoustics were used to find out what information was redundant or could be discarded without it meaning an “audible” loss for human beings.

The mp3 audio format is considered lossy for this reason, because it discards information in order to be compressed. And that information cannot be retrieved if we do the reverse conversion.

Sure, many experiments have been done that have shown that 99% of people can’t tell a 192kb compressed mp3 apart from the original wav, so converting the mp3 to wav again would produce a wav that is indistinguishable from the original wav. most people compare it to the original wav.

mp3 to wav converter using Mp4Gain

Mp4Gain is, in addition to a very modern and efficient normalizer (the most efficient in fact), it is also a very efficient converter. Therefore, you could use Mp4Gain to convert mp3 to wav and get excellent results.

What differentiates MP3 from AAC? Part 3

AAC or MP3

WAV audio file

M4A vs MP3

WAV is a waveform audio format. This is a high-quality audio file that is often used like a CD. WAV files are not compressed and therefore take up more disk space than MP3 or AAC.

Because WAV files are not compressed (called a “lossless” format), they contain more data, resulting in a better, more subtle, and more detailed sound. A WAV file typically requires 10MB of audio per minute. By comparison, MP3 takes up about 1 MB per minute.

WAV files are supported by Apple devices, but are not commonly used except by audiophiles.

WMA audio file
WMA stands for Windows Media Audio. This is a file type popularized by Microsoft Corporation who invented it. It is the default format used by Windows Media Player on Mac and PC. It competes with MP3 and AAC formats and offers compression and file sizes similar to those formats. Not compatible with iPhone and iPad.

AIFF audio file
AIFF stands for Audio Interchange File Format. Another uncompressed audio format, AIFF, was invented by Apple in the late 1980s. Like WAV, it takes up about 10MB of storage space per minute of music. Because it does not compress audio, AIFF is a higher quality format preferred by audiophiles and musicians. Because it was invented by Apple, it is compatible with Apple devices.

Apple Lossless Audio File
Another Apple invention, the Apple Lossless Audio Codec (ALAC), is the successor to AIFF. Released in 2004, it was originally a proprietary format. Apple made it open source in 2011. Apple Lossless balances smaller file sizes with better sound quality. Its files are typically about 50% smaller than uncompressed files, but with less sound quality loss than MP3 or AAC.

FLAC audio file
Free Lossless Audio Codec) is an open source audio format popular with audiophiles. You can reduce the file size by 50-60% without degrading the audio quality too much. FLAC is not supported on iTunes or iOS devices, but will work with other software installed on your device.

What differentiates MP3 from AAC? Part 2

AAC vs MP3

How MP3 works with Apple Music and iTunes

AAC Vs. MP3

MP3 is probably the most popular digital audio format on the web, but it’s not available on Apple Music’s iTunes store or in this format (more on that in the next section). Still, mp3 is compatible with Apple Music, iTunes, and all iOS devices like iPhone and iPad. You can get MP3 files from:

Digital download store.
Rip songs from CDs, depending on the music conversion settings.
Many music file sharing services.
All about AAC audio files
AAC stands for Advanced Audio Coding. It is a type of digital audio file that has been promoted as the successor to MP3. AAC generally provides higher quality sound than MP3 while using the same amount of disk space (or less).

Many people think that AAC is Apple’s proprietary format, but this is incorrect. AAC was developed by a group of companies that includes AT&T Bell Labs, Dolby, Nokia, and Sony. While Apple has embraced AAC music, AAC files can actually be played on many non-Apple devices, including phones running Google’s Android operating system, game consoles, and more.

How does the CAA work?
AAC is a lossy file format, just like MP3. To compress CD-quality audio into a file that takes up less storage space, data that will no longer affect the listening experience is typically removed at the high and low end. So AAC files don’t sound exactly the same as CD-quality files, but they generally sound good enough that most people won’t know the difference.

Like MP3, the quality of AAC files is measured by their bit rate. Common AAC bit rates include 128 kbps, 192 kbps, and 256 kbps.

How AAC works with Apple Music and iTunes
Apple has adopted AAC as its preferred audio file format. All songs streamed or downloaded from Apple Music, or sold on the iTunes store, are in AAC format. All AAC files provided by Apple are encoded at 256 kbps.

Other types of audio files supported by iPhone, iPad and Mac
While MP3 and AAC are the most popular audio files on iPhone, iPad, Mac, and other Apple products, they’re not the only ones that work. Let’s take a look at other widely used Apple supported audio formats.

What differentiates MP3 from AAC?

AAC Vs. MP3

People often call any music file “MP3”, but that’s not accurate.

AAC vs MP3 320

MP3 is a specific type of audio file and not all digital audio files are MP3s. If you use an iPhone or other Apple device, chances are most of your music isn’t MP3.

So what kind of files are your digital songs? This article details the MP3 file type, the more advanced AAC format used by Apple, and some other common audio file types that can be used with or without iPhone and iPod.

What is mp3 and how does mp3 work?
MP3 is an acronym for MPEG-2 Audio Layer 3. It is a digital media standard devised by the Moving Picture Experts Group (MPEG), an industry group that creates technical standards.

Songs saved in MP3 format take up less space than songs saved in CD-quality audio formats like WAV (more on that later). They do this by compressing the data in the song. Compressing a song to MP3 requires removing parts of the file that don’t affect the listening experience, usually the loudest and quietest end of the audio. Because some data has been removed, and because the sound of MP3 is not the same as the CD-quality version, MP3 is called a “lossy” compression format. has led some audiophiles to criticize mp3 for impairing the listening experience, even though many can’t tell the difference.

Because mp3s are compressed, more mp3 files can be stored in the same amount of space than files using a lossless compression format. In general, MP3s take up 10% of the space of a CD-quality audio file. So if the CD quality version of a song is 10MB, the MP3 version is about 1MB (this can vary depending on your taste) Audio Encoding Settings

).
Understanding bitrate and MP3
The audio quality of MP3s (and all digital music files) is measured by their bitrate. A higher bitrate means the file has more data and MP3s sound better. The most common bit rates are 128 kps, 192 kbps, and 256 kbps.

MP3 comes in two bit rates: constant bit rate (CBR) and variable bit rate (VBR). Many modern mp3s use VBR, which works by encoding parts of the song at a lower bit rate and at a higher bit rate. . smaller file. For example, a song with only one instrument is simpler and can be encoded at a lower bit rate. Parts of a song with more complex instruments require less compression to capture the full range. By changing the bitrate, the overall sound quality of the MP3 can be kept at a high level, while the file size can be further reduced.

Mp3: Audio Bitrate Calculator

Audio File Size Calculator Streaming Bitrate Calculator.

mp3 bit rate

Get the recommended high and low bitrate settings related to your network setup Audio Bitrate and File Size Calculator If the size of that audio file seems like a mystery, this is the tool you need to calculate the audio file size. The first part of the calculator calculates the bitrate of the uncompressed audio (for example, the size of the WAVE or BWF file). The second part calculates the file size for a given bit rate.
Audio Bitrate and File Size Calculator The Bitrate Calculator allows you to calculate the exact bitrate used to encode audio and video to achieve your desired file size. 3ivx MPEG-4 5.0 is the estimated audio size! Uncompressed audio bit rate. Per second: 48,000 24-bit samples; uncompressed bitrate for 1 channel:

Audio Bitrate and File Size Calculator, Audio Bitrate and File Size Calculator If the size of your audio files seems like a mystery, here are the tools you need to calculate your audio file size .

The first part of the calculator calculates the bitrate of the uncompressed audio (for example, the size of the WAVE or BWF file). The second part calculates the file size for a given bit rate. The Bitrate Calculator allows you to calculate the exact bitrate used to encode audio and video to achieve your desired file size. 3ivx MPEG-4 5.0 is a

Bitrate calculator estimates audio size! Uncompressed audio bit rate. Per second: 48,000 24-bit samples; 1-Channel Uncompressed Bitrate: In a simplified way, bitrate refers to the number of bits that can be transmitted or received per second. Bitrate is used to encode the number of bits into.
Bitrate Calculator The Bitrate Calculator allows you to calculate the exact bitrate used to encode audio and video to achieve the desired file size. 3ivx MPEG-4 5.0 is the estimated audio size! Uncompressed audio bit rate.

Per second: 48,000 24-bit samples; uncompressed bitrate for 1 channel:
Get the bitrate or bit depth of an audio wav file In simple terms, bitrate is the number of bits per second that can be transmitted or received. The bit rate is used to encode the number of bits. If the size of the audio file seems like a mystery, this is the tool you need to calculate the size of the audio file. The first part of the calculator counts bits.

Get the bitrate or bit depth of an audio wav file to estimate the size of the audio! Uncompressed audio bit rate. Per second: 48,000 24-bit samples; 1-Channel Uncompressed Bitrate: In a simplified way, bitrate refers to the number of bits that can be transmitted or received per second. Bitrate is used to encode the number of bits into.
Audio Bitrate Calculator – Inaudible Discussion If audio file size seems like a mystery, this is the tool you need to calculate audio file size.

The first part of the calculator calculates the bit rate for the DVB-S2, DVB-S2X and DVB-S standards, calculates the bit rate and bandwidth, the net bit rate, up to 32 APSK.

Audio Bitrate Calculator – Inaudible Discussion Put simply, bitrate refers to the number of bits per second that can be transmitted or received. The bit rate is used to encode the number of bits.

If the size of the audio file seems like a mystery, this is the tool you need to calculate the size of the audio file. The first part of the calculator counts bits.

44.1kHz PCM

In our experience, 16-bit and 44.1 kHz provide the best audio quality you can experience.

PCM

Anything beyond that format tends to waste disk capacity, since the 44.1 kHz HD sample rate originated with PCM adapters in the late 1970s that recorded digital audio onto video tape, especially the Sony PCM-1600 introduced in 1979 and introduced in this series It has flourished in later models. This became the basis for Compact Disc Digital Audio (CD-DA) as defined in the 1980 Red Book standard. In other words, the digital audio standard for CD audio is 44.1 kHz/16 bits. PCM Audio and Home Theater PCM is used for CD, DVD, Blu-ray and other digital audio applications. When used in surround sound applications, it is often called Linear Pulse Code Modulation (LPCM). The reason for this is that in the past, computer sound cards could only handle 48kHz PCM data, so the 44.1kHz PCM data had to be resampled, which would consume processing power. So the CD-ROM drive has an audio cable that feeds the analog audio to the sound card for playback, avoiding the need for resampling.

The 44.1 kHz sample rate originated with PCM adapters in the late 1970s that recorded digital audio to videotape, notably the Sony PCM-1600 introduced in 1979, and carried over to later models of the Serie. This became the basis for Compact Disc Digital Audio (CD-DA) as defined in the 1980 Red Book standard. In other words, the digital audio standard for CD audio is 44.1 kHz/16 bits. PCM Audio and Home Theater PCM is used for CD, DVD, Blu-ray and other digital audio applications. When used in surround sound applications, it is often called Linear Pulse Code Modulation (LPCM). The reason for this is that, in the past, computer sound cards could only handle 48kHz PCM data, so the 44.1kHz PCM data had to be resampled, which would consume processing power. . So the CD-ROM drive has an audio cable that feeds the analog audio to the sound card for playback, avoiding the need for resampling. Pulse Code Modulation (PCM) or 44.1 kHz used on CD. Some devices may use a 96kHz or 192kHz sample rate, but the advantage is that

In other words, the digital audio standard for CD audio is 44.1 kHz/16 bits. PCM Audio and Home Theater PCM is used for CD, DVD, Blu-ray and other digital audio applications. When used in surround sound applications, it is often called Linear Pulse Code Modulation (LPCM). The reason for this is that, in the past, computer sound cards could only handle 48kHz PCM data, so the 44.1kHz PCM data had to be resampled, which would consume processing power. . So the CD-ROM drive has an audio cable that feeds the analog audio to the sound card for playback, avoiding the need for resampling.

What is bit rate? Knowledge of the MP3 audio format.

MP3 Bitrate

Digital audio formats are audio signals that are recorded, processed, and reproduced in digital form.

mp3 bit rate

The emergence of digital audio formats is to meet the needs of high-fidelity playback, storage and transmission. Simply put, early analog audio formats had issues with playback distortion and glitches due to media wear. Since the advent of the CD, digital format audio files have become popular, but another problem has arisen: the limitation of the storage volume, and the CD still has the phenomenon of wear. Saving to hard drive (relatively longer storage time) is not a good solution when storage media (mainly hard drives) are still expensive at the time. The rise of the Internet has created a requirement for long-distance file transmission. Under the restriction of bandwidth, the demand to reduce file size has become more intense. All this has led to the generation of lossy compressed digital audio formats from external factors!

In terms of internal factors, with the improvement of computer operation and coding capabilities, the progress of various acoustic psychological models has promoted the emergence of various lossy compressed digital audio formats. Some of the most commonly used audio formats in MP3 players are briefly introduced below: MP3 (CBR, VBR, ABR), WMA, WAV, ADPCM, and the emerging audio formats AAC, ASF, and OGG.

Before introducing various digital audio formats, let’s first clarify a concept: bitrate.

In the field of computing, all information is digitized. Bit is the smallest unit of data in a computer, it refers to a number of 0 or 1, which is a mathematical binary number, a “0” or “1” , is a bit. For example, when we say a 2-digit number, it means that it is a two-digit binary number, and there are 4 combinations of “00”, “01”, “10” and “11”, which represent 0, “11” in decimal respectively. 1, 2 and 3 are four numbers.

Bit rate, let’s see this, you don’t need radio quality to compare MP3 quality

MP3 Quality

Bit rate refers to the number of bits transmitted per second, and the unit is bps (Bit per second). The higher the bit rate, the higher the data transmission.

MP3 Quality

The bit rate in sound refers to the sampling rate of the conversion of digital sound from analog to digital format. The higher the sampling rate, the better the quality of the restored sound. The bit rate (bit rate) principle in video is the same as in sound, which refers to the sample rate converted from analog signal to digital signal.

Bitrate refers to the sampling precision (quantization precision) of converting digital sound from analog to digital format, that is, the number of bits per sample of sound. The higher the sampling precision (quantization precision), the better the quality of the restored sound.
Bit rate is a benchmark indicator of the compression efficiency of digital music. Bit rate indicates the rate of bps (bit per second, bits per second) transmitted per unit of time (1 second). The unit is usually kbps (1000 bits per second in colloquial terms). The bit rate of digital music on CD is 1411.2 kbps (that is, to burn 1 second of CD music, 1411.2 × 1000 data bits are required), the high BIT RATE of the digital music file music means that it should be processed in a unit of time (1 second) The amount of data (BIT) is large, which means that the sound quality of the music file is good. However, when the BITRATE is high, the file size increases, which will occupy a large amount of memory capacity. they are 32-256 Kbps. Of course, the wider the rate, the better, but 320 Kbps is the highest level at the moment.