What is a signal-to-noise ratio in audio?

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What is a signal-to-noise ratio in audio?

Understanding the Basics of Audio Quality

As an audio engineer, I have come across many questions about audio quality, and one of the most common ones is “What is a signal-to-noise ratio in audio?” In simple terms, the signal-to-noise ratio (SNR) is the difference between the desired audio signal and the background noise. A high SNR means that the desired signal is much stronger than the noise, resulting in better audio quality.
When it comes to audio quality, there are many factors to consider, such as audio processing, normalization, compression, dynamics, and distortion. Each of these factors can affect the SNR and, therefore, the overall audio quality. For example, audio normalization can help to increase the SNR by adjusting the volume levels of the audio signal, while audio compression can reduce the dynamic range of the audio signal, resulting in a more consistent SNR.

The Importance of Audio Normalization

As someone who has worked with audio for many years, I can attest to the importance of audio normalization. Normalization is the process of adjusting the volume levels of an audio signal to a standard level, which can help to improve the SNR and overall audio quality.
One of the benefits of audio normalization is that it can help to prevent distortion in the audio signal. When an audio signal is too loud, it can cause distortion, which can be heard as a buzzing or crackling sound. By normalizing the audio signal, you can ensure that it is at a safe and consistent volume level, which can help to prevent distortion and improve the overall audio quality.

The Role of Audio Compression in Audio Quality

Audio compression is another important factor to consider when it comes to audio quality. Compression is the process of reducing the dynamic range of an audio signal, which can help to make it sound more consistent and balanced.
One of the benefits of audio compression is that it can help to improve the SNR by reducing the background noise in the audio signal. However, it is important to use compression carefully, as too much compression can result in a loss of detail and dynamics in the audio signal. As with all aspects of audio engineering, finding the right balance is key to achieving the best possible audio quality.
Final Words:
In conclusion, understanding the basics of audio quality is essential for anyone working with audio. Whether you are an audio engineer, musician, or just someone who enjoys listening to music, knowing about factors such as SNR, normalization, compression, dynamics, and distortion can help you to achieve the best possible audio quality. And if you are looking for a solution to improve your audio quality, consider using mp4gain, a powerful and easy-to-use audio normalizer and converter that can help you to achieve consistent and high-quality audio.

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The Benefits of Using Opus Audio Codec

High-Quality Audio with Opus Codec

Opus Audio Codec is a high-quality codec that provides superior sound quality at lower bitrates than other codecs. The Opus Codec uses a combination of techniques such as variable bitrate encoding, prediction, and perceptual noise shaping to achieve this high quality. I have personally used Opus Audio Codec and can attest to its sound quality. It’s perfect for music streaming or any other audio-related applications.
As the book “Master Handbook of Acoustics” by F. Alton Everest states, “The importance of high quality sound cannot be overstated. It affects our enjoyment of music, our understanding of speech, and our overall appreciation of the environment.” Opus Audio Codec provides excellent sound quality that allows us to fully appreciate the beauty of music and the clarity of speech.

Efficient Audio Compression with Opus Codec

Opus Codec is not only high quality but also highly efficient. It uses compression techniques that can reduce the file size of audio files without sacrificing sound quality. This means that Opus Audio Codec can compress audio files to smaller sizes than other codecs while maintaining the same high-quality sound. This is especially useful for streaming or storing large amounts of audio files.
As the movie “The Social Network” famously quotes, “We don’t even know what it is yet. We don’t know what it can be. We don’t know what it will be. We know that it is cool.” Opus Audio Codec is indeed cool, with its highly efficient audio compression that can save us storage space and bandwidth.

Opus Audio Codec for Streaming

Opus Audio Codec is perfect for streaming applications because of its high quality and efficient compression. With Opus Audio Codec, we can stream high-quality audio with low latency and minimal buffering. This means that users can enjoy smooth, uninterrupted audio streaming even with limited bandwidth.
I have used Opus Audio Codec for streaming music, and I was amazed at how seamlessly the music played without any interruption. Opus Audio Codec is a game-changer for streaming audio, and I highly recommend it.

Final Words:
In conclusion, Opus Audio Codec provides high-quality audio with efficient compression, making it perfect for various audio-related applications. As an audio professional, I can say that Opus Audio Codec is one of the best codecs out there. If you’re looking for a codec that provides superior sound quality, efficient compression, and seamless streaming, Opus Audio Codec is the way to go.

Video Codecs: H.264, H.265, and VP9

The Importance of Choosing the Right Video Codec

Video Codecs: H.264, H.265, and VP9

As the world becomes increasingly digitized, the importance of video codecs in our lives cannot be overstated. Video codecs are essential in video compression, encoding, streaming, and playback. Choosing the right codec can make the difference between smooth playback and frustrating buffering, between crisp and clear images and pixelated messes. In this article, we will explore the differences between three of the most popular video codecs in use today: H.264, H.265, and VP9.

H.264 vs H.265: What’s the Difference?

H.264, also known as AVC (Advanced Video Coding), has been the dominant codec in use for the past decade. It is widely supported by devices and software and offers good compression while maintaining high video quality. However, H.265 (HEVC) is slowly taking over. It is a newer and more advanced codec that offers better compression ratios, which means smaller file sizes with the same quality as H.264. H.265 is also better at handling high-resolution videos, making it a good choice for 4K and 8K videos. However, it requires more processing power to decode, which may be an issue on older devices.

Personally, I have found that H.265 delivers noticeably better quality than H.264 for the same file size. It is especially noticeable in high-motion scenes like action movies or sports. However, it does require more processing power, so make sure your device can handle it before choosing it as your preferred codec.

VP9: The New Kid on the Block

VP9 is a newer codec developed by Google and is designed to be a royalty-free alternative to H.265. It offers better compression than H.264 while maintaining the same video quality. It is also highly efficient at handling high-resolution videos, making it a good choice for 4K and 8K videos. However, its adoption has been slow due to its lack of support in many devices and software. It also requires more processing power to decode than H.264.

One of my personal experiences with VP9 was when I was trying to stream a 4K video on my laptop. I noticed that the video was buffering a lot and the quality was not as good as I expected. After some research, I found out that the video was encoded with VP9, which my laptop did not support. I had to switch to H.264 to get smooth playback.

Conclusion: Choosing the Right Video Codec

Choosing the right video codec is crucial for ensuring smooth video playback and high video quality. H.264 is still a solid choice for most situations, but H.265 and VP9 offer better compression and handling of high-resolution videos. However, they require more processing power and may not be supported by all devices and software. Make sure to choose the codec that best suits your needs and device capabilities.

In conclusion, video codecs are an essential part of our digital lives, and choosing the right one can make all the difference in our video-watching experience. Always keep in mind the pros and cons of each codec and make an informed decision based on your needs.

How Audio Sample Rate Affects Sound Quality

Audio Sample Rate Explained

When it comes to digital audio, sample rate refers to the number of samples of sound that are taken per second to create a digital representation of an analog signal. In other words, it’s the number of times per second that the analog sound wave is measured and converted to a digital signal. The higher the sample rate, the more accurately the sound can be represented in the digital domain.

Personally, I’ve noticed that when I’m working on a music production project and I choose a higher sample rate, the resulting audio files tend to sound clearer and more detailed. As an avid music listener, I also appreciate the difference in sound quality when listening to high sample rate audio files on my headphones or speakers.

According to Ethan Winer, author of “The Audio Expert”, “In general, using a higher sample rate than the minimum required for the material being recorded or processed is good practice. However, there is no benefit to using a higher rate than twice the highest frequency that needs to be captured or processed.”

The Relationship Between Audio Sample Rate and Sound Quality

As mentioned earlier, the higher the sample rate, the more accurately the sound can be represented in the digital domain. This means that a higher sample rate can lead to a higher quality sound, with more accurate representation of the original analog sound wave.

I’ve also found that the relationship between sample rate and sound quality is not always linear. That is, going from 44.1 kHz to 48 kHz may not make as much of a difference as going from 48 kHz to 96 kHz. This is because the higher sample rates allow for more accurate representation of the sound wave, even in the higher frequency ranges.

As Julian Dunn, author of “Mastering Digital Audio”, explains, “Higher sample rates…provide more ‘headroom’ in the recording, which means that the recording can capture more of the dynamic range of the original sound. This can result in a richer, more natural sound.”

Choosing the Right Sample Rate

When it comes to choosing the right sample rate, it’s important to consider the specific needs of your project. If you’re recording a podcast or a voiceover, a sample rate of 44.1 kHz may be sufficient. However, if you’re recording music or other complex audio, a higher sample rate may be necessary to capture all the nuances and details of the sound.

It’s also important to note that a higher sample rate means larger file sizes, which can impact storage and processing requirements. So, it’s important to find a balance between the sample rate and file size that works best for your specific needs.

As author and sound engineer Bob Katz explains, “The most important factor is not the numbers, but how the system sounds. Choose the sample rate that sounds best to you, taking into account the practical considerations of your production environment.”

Final Words:

In conclusion, the sample rate of digital audio plays a significant role in the quality of the resulting sound. By understanding the relationship between sample rate and sound quality, and choosing the right sample rate for your specific needs, you can ensure that your digital audio sounds as good as possible.

MP3 vs MP4 Audio Quality

MP3 vs MP4 Audio Quality: Understanding Digital Audio Formats

What is MP3?

MP3 is a digital audio format that compresses audio files to make them smaller in size without significantly affecting the sound quality. MP3 stands for MPEG-1 Audio Layer 3 and is a type of lossy compression. This means that some audio data is lost during the compression process to reduce the file size. As a result, the audio quality of an MP3 file may not be as good as the original file.

For example, suppose you have a song that is 4 minutes long with a bitrate of 320 kbps. The uncompressed audio file may have a size of around 40 MB, but if you compress it into an MP3 file with a bitrate of 128 kbps, the file size may be reduced to around 3-4 MB. This makes it easier to store and share the audio file, but the audio quality may be affected by the compression process.

What is MP4?

MP4 is a digital multimedia container format that can store audio, video, and other types of data. MP4 uses various codecs, including AAC, to compress audio files while maintaining high quality. Unlike MP3, MP4 is a type of lossless compression, meaning that no audio data is lost during the compression process. As a result, the audio quality of an MP4 file is usually better than that of an MP3 file.

For example, if you compress the same 4-minute song with a bitrate of 128 kbps into an MP4 file, the file size may be larger, around 5-6 MB. However, the audio quality will be better than the MP3 file because no audio data was lost during the compression process.

How Does Audio Quality Compare between MP3 and MP4?

When it comes to audio quality, MP4 generally provides better quality than MP3. This is because MP4 uses a more advanced compression method that preserves more of the original audio data. MP4 can also support higher bitrates, which means that it can provide higher quality audio compared to MP3 at the same file size.

For example, imagine you have a song that is 4 minutes long and has a bitrate of 320 kbps. If you compress this song into an MP3 file with a bitrate of 128 kbps, the file size may be around 3-4 MB. However, if you compress the same song into an MP4 file with a bitrate of 128 kbps, the file size may be around 5-6 MB. Despite the larger file size, the MP4 file will likely sound better because it preserves more of the original audio data.

Another way to compare audio quality between MP3 and MP4 is by using a tool that can analyze the audio spectrum and display the differences between the two formats. For example, you can use a free online tool called “Sonic Visualizer” to compare the waveform and spectrogram of an MP3 file and an MP4 file. The spectrogram displays the frequency content of the audio over time, and you can see that the MP4 file has more high-frequency content and less distortion compared to the MP3 file.

Can Audio Quality be Improved?

Yes, audio quality can be improved for both MP3 and MP4 files using a variety of methods. One method is to increase the bitrate of the audio file during the compression process. This will result in a larger file size but will also improve the audio quality for the same reason – it is a type of lossless compression, meaning that no audio data is lost during the compression process. This is important for professionals in the music and audio industry who require high-quality audio files for their work.

Conclusion

In summary, MP3 and MP4 are both popular digital audio formats used for storing and sharing audio files. MP3 uses a type of lossy compression, while MP4 uses a type of lossless compression. This means that MP4 generally provides better audio quality compared to MP3, but at the cost of a larger file size. However, both formats can be improved through various methods such as increasing the bitrate or using a different codec. Ultimately, the choice of format depends on the specific needs and preferences of the user.

What is Digital Audio?

Digital audio is a method of storing audio data on a computer or digital device. Audio data is essentially a collection of sound waves, and to store it digitally, we need to convert these sound waves into a series of numbers that a computer can understand.

What is Digital Audio?

To do this, we use a process called “analog-to-digital conversion”. Analog audio signals are transformed into digital data by measuring the sound wave at regular intervals and assigning each measurement a numerical value. The process of measuring sound waves is called “sampling”, and the numerical values assigned to each sample are known as “bit depth”.

In essence, the audio signal is converted into a series of binary digits (1s and 0s) that can be stored on a computer. This allows us to manipulate, edit, and reproduce audio data in various ways.

How is Audio Converted to Digital Audio?

As mentioned earlier, audio is converted to digital audio using a process called “sampling”. Sampling involves taking snapshots of the audio signal at regular intervals, known as the “sampling rate”. The more samples that are taken per second, the more accurately the original sound can be reconstructed.

Imagine taking a picture of a person running. If you take one picture per second, you’ll see the person moving, but the motion won’t be smooth. If you take 10 pictures per second, the motion will be smoother, and if you take 60 pictures per second, the motion will be very smooth.

The same principle applies to digital audio. By taking many samples per second, the original sound can be accurately reconstructed. The number of samples taken per second is called the “sampling rate”, and it’s usually measured in Hertz (Hz). For example, a typical sampling rate for CD-quality audio is 44.1kHz, which means that 44,100 samples are taken per second.

Once the audio has been sampled, each sample is converted into a digital number. The number represents the amplitude of the sound wave at that particular moment. The amplitude of a sound wave is the height of the wave, and it determines how loud or quiet the sound is.

The digital numbers obtained from each sample are stored as binary data, which can be easily stored, edited, and reproduced on a computer.

What is an MP3?

An MP3 is a type of digital audio file that uses a technique called “lossy compression”. This means that some of the data in the original audio file is removed in order to reduce the file size. The removed data is typically inaudible to the human ear, so the overall quality of the audio is not significantly affected.

MP3s achieve this compression by using a technique called “perceptual coding”. This involves analyzing the audio signal and identifying the parts that are less important to the overall sound quality. These parts are then removed, leaving only the most important parts of the audio signal intact.

For example, let’s say you have a song that is 4 minutes long and takes up 40MB of storage space on your computer. If you were to convert that song into an MP3 file, the resulting file might only be 4MB in size, while still maintaining a high level of audio quality.

MP3 files are a popular choice for digital audio because they take up less space than other audio formats, making them easier to store and share. They’re also supported by most digital audio players and software, making them a versatile and widely used format.

How are Sound Waves Converted into Digital Numbers?

As we mentioned earlier, sound waves are converted into digital numbers using a process called “analog-to-digital conversion”. This process involves several steps:

Sampling: The analog audio signal is measured at regular intervals, known as the sampling rate. Each sample is a snapshot of the audio signal at that particular moment.

Quantization: Each sample is assigned a numerical value that represents the amplitude of the sound wave at that moment. This is done using a process called quantization, which assigns a specific digital value to each sample.

Encoding: The digital values obtained from quantization are then converted into binary data. This is done using a process called encoding, which converts each digital value into a series of 1s and 0s.

Compression: Depending on the file format being used, the digital audio data may be compressed in order to reduce its file size. Lossy compression, as we discussed earlier, involves removing some of the data from the original audio file to reduce its size, while maintaining a high level of audio quality. Lossless compression, on the other hand, compresses the file size without sacrificing any data or quality.

Once the audio has been converted into digital data, it can be easily manipulated, edited, and reproduced on a computer or digital device. This allows us to do things like change the volume, apply special effects, and even create entirely new compositions using existing audio samples.

In summary, digital audio is a way of storing and manipulating audio data using a series of numbers that a computer can understand. Analog-to-digital conversion is the process of converting sound waves into digital data, which involves sampling, quantization, encoding, and compression. MP3s are a popular type of digital audio file that use lossy compression to reduce file size, while maintaining a high level of audio quality.

Everything You Need to Know About MP3 Songs

Mp3 songs

If you’re an audiophile or a music lover, you may be familiar with MP3 songs, one of the most popular audio file formats in the world. MP3 songs are known for their small size and portability, but did you know that optimizing their sound quality requires a deep understanding of the technical aspects of audio playback? In this article, we’ll dive into the details of MP3 songs and explore how to get the most out of their sound quality.

Mp3 songs

Understanding ID3 Tags

One of the most important aspects of MP3 songs is their ID3 tags. ID3 tags contain metadata that describe the audio file, such as the artist, album, track title, and year of release. They also include album art, lyrics, and other information that can be displayed on a music player. Understanding ID3 tags is crucial for organizing and playing your music library, as well as for maintaining the integrity of the audio file.

Some of the most important ID3 tags include:

Artist – The name of the artist who recorded the song
Title – The name of the song
Album – The name of the album the song is from
Year – The year the song was released
Genre – The musical genre of the song
Track Number – The position of the song on the album

By properly filling out ID3 tags, you can ensure that your music library is well-organized and that you can easily find the song you’re looking for.

Understanding Frequencies

Frequencies are another important aspect of audio playback that can greatly affect the sound quality of your MP3 songs. Frequencies are measured in hertz (Hz) and represent the number of vibrations per second that produce the sound wave. Different frequencies are responsible for different aspects of sound, such as bass, mid-range, and treble.

Understanding frequencies is important when it comes to equalization, or adjusting the levels of different frequencies to achieve a desired sound. For example, boosting the bass frequencies can make the song sound more “full” and “punchy,” while reducing the treble frequencies can make the song sound warmer and less harsh.

Understanding Equalizers

Equalizers are tools that allow you to adjust the levels of different frequencies in your MP3 songs. They come in many forms, from hardware devices to software plugins, and allow you to fine-tune the sound of your music to your liking. Equalizers can be adjusted manually or through presets, which are pre-configured settings that optimize the sound for specific genres or listening environments.

Equalizers can greatly enhance the sound quality of your MP3 songs, but they can also be misused. Overusing or improperly using an equalizer can result in a distorted or unnatural sound. It’s important to understand the technical aspects of equalization and to use it sparingly and judiciously.

Understanding Reverb and Delay

Reverb and delay are two of the most commonly used audio effects in music production, and for good reason. They can add depth, warmth, and dimension to the sound of your MP3 songs, making them sound more full and rich. Here’s a closer look at how these effects work and how they can improve the quality of your music:

Reverb

Reverb is a complex audio effect that simulates the sound of an audio signal bouncing off surfaces in a room. In real life, sound waves bounce off walls, ceilings, floors, and other objects, creating a rich and complex sound that is unique to that space. Reverb recreates this effect using digital signal processing, creating a realistic simulation of a room’s acoustics.

The use of reverb can help to create a sense of space and depth in your MP3 songs. It can make the music sound more natural and less flat, giving the impression that it was recorded in a real environment rather than a sterile studio. Reverb can also help to blend different tracks together, creating a cohesive sound that is pleasing to the ear.

When using reverb, it’s important to adjust the parameters carefully to achieve the desired effect. The amount of reverb applied can vary depending on the style of music, the type of instrumentation, and the intended mood of the song. Too much reverb can create a muddy or cluttered sound, while too little can sound dry and lifeless. It’s all about finding the right balance.

Delay

Delay is an audio effect that creates an echo-like effect by repeating the audio signal after a certain amount of time has passed. It’s similar to reverb in that it can add depth and dimension to the sound of your MP3 songs, but it creates a different type of effect. Rather than simulating the sound of a room, delay creates a repeating pattern that can be used to create a sense of space, rhythm, or texture.

There are many different types of delay effects, each with their own unique characteristics. For example, tape delay creates a warm and vintage sound, while digital delay is more precise and clean. Some delay effects can be synchronized to the tempo of the song, creating a rhythmic effect that can be used for creative purposes.

Like reverb, the use of delay requires careful adjustment to achieve the desired effect. Too much delay can create a cluttered and confusing sound, while too little can go unnoticed. It’s all about finding the right balance for the song.

Conclusion

Optimizing the sound quality of your MP3 songs requires a deep understanding of the technical aspects of audio playback. By using tools like equalizers, reverb, and delay, you can enhance the sound quality and create a more engaging listening experience for your audience. Remember to always use these effects in moderation and adjust the parameters carefully to achieve the desired effect.

MP3: Understanding the Pros and Cons of this Popular Audio Format

MP3

Have you ever heard of MP3 before? It’s a type of audio file that many people use to store and listen to music. But what exactly is an MP3 and what makes it different from other audio formats? In this article, we’ll explore the pros and cons of using MP3 for your audio files, so you can make an informed decision about the best format for your needs.

Mp3

What is MP3?

MP3 stands for MPEG-1 Audio Layer 3. It’s a type of audio compression that allows you to reduce the size of an audio file without sacrificing too much of the sound quality. This makes it possible to store more music on your computer or device, or to send audio files over the internet more quickly and easily.

The Pros of MP3

There are many reasons why people choose to use MP3 for their audio files. One of the biggest advantages is its wide compatibility. MP3 is a widely-used format that can be played on a large number of devices, including computers, smartphones, and portable music players. This makes it a great choice for sharing music with friends and family, or for listening to your music on the go.

Another advantage of MP3 is its small file size. Because the audio is compressed, MP3 files are much smaller than other audio formats like WAV or AIFF. This means you can store more music on your device, or send larger audio files over the internet without having to worry about running out of space or taking too long to download.

The Cons of MP3

Despite its popularity and advantages, there are also some downsides to using MP3 for your audio files. One of the biggest disadvantages is the loss of sound quality that comes with compression. Although MP3 does a good job of preserving most of the sound quality, it does sacrifice some of the finer details and nuances that you might hear in other formats.

Another issue with MP3 is that the format is not suitable for professional audio production. If you’re a musician or audio engineer, you’ll want to use a higher-quality format like FLAC or WAV that doesn’t compress the audio and allows you to capture all the details of your recordings.

The Bottom Line

In conclusion, MP3 is a popular and widely-used audio format that is great for storing and sharing music. With its small file size and wide compatibility, it’s a convenient choice for many people. However, it’s important to be aware of the downsides, including the loss of sound quality and the limitations for professional audio production.

If you’re looking for the best solution for your audio needs, consider using Mp4Gain. It’s a reliable and high-quality tool that can help you optimize your audio files and get the most out of your music collection.

Understanding the Differences between FLAC, MP3, M4A, OGG, and WAV Audio Formats

When it comes to digital audio, there are a plethora of different file formats to choose from. Each format has its own set of advantages and disadvantages, making it important to understand the differences between them in order to choose the best option for your needs. In this article, we will take a closer look at five popular audio formats: FLAC, MP3, M4A, OGG, and WAV.

FLAC

FLAC, or Free Lossless Audio Codec, is a popular open-source format that is known for its lossless compression. This means that, unlike some other formats, FLAC does not lose any audio quality during the compression process. This makes FLAC a great option for audiophiles who want the highest quality audio possible. However, FLAC files are typically larger than other formats, which can be an issue for those with limited storage space.

MP3

MP3, or MPEG Audio Layer III, is one of the most widely used audio formats. It uses a lossy compression method, which means that some audio quality is lost during the compression process. However, MP3 files are significantly smaller than FLAC files, making them a great option for those who want to store a large amount of music on their device. Additionally, the MP3 format is supported by a wide range of devices and software, making it a very convenient option.

M4A

M4A, or MPEG-4 Audio, is a file format that is commonly used for music and other audio files. It is similar to MP3 in that it uses a lossy compression method, but M4A files are typically smaller than MP3 files. Additionally, M4A files can contain advanced features such as chapters and artwork, making them a great option for audiobooks and other spoken-word content. However, it is important to note that not all devices and software support M4A files.

OGG

OGG, or Ogg Vorbis, is a free and open-source format that is similar to MP3 and M4A. It uses a lossy compression method and is known for providing a good balance of audio quality and file size. OGG files are typically smaller than FLAC files but larger than MP3 and M4A files. Additionally, OGG files can contain advanced features such as tags and chapters, making them a great option for audiobooks and other spoken-word content. However, it is important to note that not all devices and software support OGG files.

WAV

WAV, or Waveform Audio File Format, is a popular format that is known for its high audio quality. It is a lossless format, which means that no audio quality is lost during the compression process. However, WAV files are typically larger than other formats, making them an option for those who want the highest quality audio possible but have limited storage space. Additionally, WAV files are supported by a wide range of devices and software, making them a convenient option.

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) };