MP3 Bitrate Calculation Methods

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MP3 Bitrate Calculation Methods

What is MP3 Bitrate?

MP3 bitrate is the amount of data that is used to encode an MP3 audio file. The higher the bitrate, the higher the quality of the audio file. However, higher bitrates also result in larger file sizes.

How is MP3 Bitrate Calculated?

MP3 bitrate is calculated using a variety of factors, including the complexity of the audio signal, the desired quality of the audio file, and the target file size.

What are the Different Types of MP3 Bitrate Calculation Methods?

There are two main types of MP3 bitrate calculation methods:

Constant bitrate (CBR): This method uses a constant bitrate for the entire audio file. This results in a consistent audio quality throughout the file, but it can also result in larger file sizes for complex audio signals.
Variable bitrate (VBR): This method uses a variable bitrate, which means that the bitrate changes depending on the complexity of the audio signal. This results in smaller file sizes for complex audio signals, but it can also result in a slight variation in audio quality.

Which MP3 Bitrate Calculation Method is Best?

The best MP3 bitrate calculation method depends on your needs. If you need a consistent audio quality throughout the file, then CBR is the best option. If you are more concerned about file size, then VBR is the best option.

How to Choose the Right MP3 Bitrate for Your Needs

When choosing the right MP3 bitrate for your needs, you should consider the following factors:

The type of audio signal: Complex audio signals, such as those that contain a lot of high-frequency content, require a higher bitrate than simple audio signals.
The desired quality of the audio file: If you want the highest possible quality, then you should use a higher bitrate. If you are more concerned about file size, then you can use a lower bitrate.
The target file size: If you have a specific file size in mind, then you can use a bitrate calculator to determine the bitrate that you need.

Conclusion

MP3 bitrate is an important factor to consider when encoding audio files. By understanding the different types of MP3 bitrate calculation methods and how to choose the right bitrate for your needs, you can ensure that your audio files sound great and have a reasonable file size.

Here are some additional details about the different types of MP3 bitrate calculation methods:

Constant bitrate (CBR): This method is the simplest to understand and implement. It is also the most reliable, as it ensures that the audio quality is consistent throughout the file. However, it can result in larger file sizes for complex audio signals.
Variable bitrate (VBR): This method is more complex than CBR, but it can result in smaller file sizes for complex audio signals. This is because VBR allows the encoder to use a lower bitrate for less complex parts of the audio signal, and a higher bitrate for more complex parts of the audio signal. However, VBR can result in a slight variation in audio quality, depending on the complexity of the audio signal.

Ultimately, the best MP3 bitrate calculation method for you depends on your specific needs. If you need a consistent audio quality throughout the file, then CBR is the best option. If you are more concerned about file size, then VBR is the best option.

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What is the difference between 128k and 320k music? Part 2

DJs: Understanding Bitrate & Audio Quality - On The Rise DJ Academy

Bit Rate, Sample Rate, Lossless, MP3, FLAC, APE, 320kb, 192kb, 128kb, 44.1khz, CBR, VBR. Does this bunch of various names make you both familiar and unknown?

The higher the bitrate, the better the sound quality. Lossless music is the highest sound quality, right? So, let’s start with the sound collection.

【Audio composition】

Nowadays, when we talk about audio, everything is digital audio. Digital audio consists of three parts: sample rate, sample precision, and number of sound channels.

Sample Rate: Both the sample rate, which refers to the number of samples per second when recording the sound, expressed in Hertz (Hz).

Sampling Precision: Refers to the dynamic range of the recorded sound, measured in bits (Bit).

Sound channel: the number of channels (1-8).

In simple terms, we can think of a sound wave as a curve. We know that the curve is made up of points, and the sampling rate is the number of points in the middle of the length per second (the horizontal axis in the figure above). Sampling precision is the number of points in the dynamic range (upper vertical axis). The finer the positioning of these two dimensions, the greater the true sound restoration and the better the sound quality. Of course, the larger the audio file will be. The customer mentioned by the above colleague said that the latest Hi-Res Audio format released by SONY is a 6-channel 192kHz/24-bit recorded audio file. The size of the lossless format, of course, will be more than 200 megabytes.

The sampling frequency is approximately the following depending on the type of use (k is the thousand-bit symbol, 1khz=1000hz):

8khz – used for phones etc, is enough to record human voices.

22.05khz: transmission use frequency.

44.1kb: Audio CD.

48khz: used in DVD and digital TV.

96khz-192khz: used for DVD-Audio, Blu-ray HD, etc.

The common range of sample precision is 8 bits to 32 bits, with 16 bits generally used on CD.

Having said that, my friends are starting to get confused. It’s not the bitrate that determines the sound quality, so why is everyone saying that 320kb sound quality is better than 128kb?

What is the difference between 128k and 320k music?

192k is a turning point. Below 192K, the sound quality is relatively damaged, especially the high-frequency part above 16Khz will be cut off.

In short, mp3 above 192k, ordinary home equipment can no longer hear the difference in CD sound quality, except for golden ears and hi-fi equipment. Of course, these data are not 100% reliable. There are always people on the internet sharing fake mp3 above 192K. In fact, they are converting low bitrate music to high bitrate through software, but the sound quality will not improve. Windows Media Player compresses the resulting mp3 is absolutely wonderful. No matter how high the compressed bitrate is, it will cut perfectly at around 16K, so if you want to compress MP3 yourself, don’t use Windows Media Player.

Well, in fact, the bit rate should be said to be another dimension, it is a compression of audio files.

Nowadays, most of the audio formats that we use regularly are based on the original “WAV” file of the audio CD (44.1khz sampling rate, 16bit sampling precision, 2ch). The original recorded sound data is stored in an array, which is in PCM format, while WAV format is an encoding format developed by Microsoft, and its function is to play the PCM format data through encoding.

Since the data in WAV basically completely restores the PCM data, MP3, AAC and other lossless encoding formats are basically recompressed based on the WAV files. Therefore, we can simply think that WAV is the original audio format and other audio formats are compressed formats.

When it comes to compression, storage and transmission are inseparable. The purpose of compression is to improve storage and transmission. Therefore, before we talk about compression, we need to understand the basic units of computers.

We all know that the computer is a binary number system, and the files stored by the computer are made up of two numbers, 0 and 1. Therefore, the computer’s transmission is based on each number, and each number is called 1 ” bit”. For example, for an audio piece, its basic data is “0,1,1,1,0,1, 1 ,0”, and when transmitting, these numbers are transmitted one by one. The sampling precision mentioned above is this unit.

Why are MP3 bitrates often multiples of 32? Part 2

Technically, there is nothing to limit the MP3 bitrate to a multiple of 2, as variable bitrate encoding can be used, or a custom bitrate can be achieved using some flags not used in the MPEG specification (although it must be implemented manually).

For MP3 to be MPEG-compliant, and therefore compatible with most MP3 decoders, it must have a bitrate defined by the specification, so all CBR-encoded MP3 files must have a bitrate that is a multiple of two.

Depending on the resource, VBR can be encoded by changing the bitrate between a fixed rate above each frame, or it can be encoded by sharing the available bits in adjacent frames (effectively generating a non-standard bitrate for the two frames combined). The length of a given frame depends on the sampling rate, there are 1152 samples per frame. There is nothing to limit the size of the frame itself, nor is there any limit to making the frame size base 2 (i.e. a 128 kbit/s MP3 with a 44.1 kHz sample rate would have a frame size of 417 bytes).

In the end, a file encoded at 126 kbps sounds worse than a file encoded at 128 kbps, and likewise a file encoded at 131 kbps sounds better. However, MP3s are encoded for compression according to the psychoacoustic model of a specific encoder. The amount by which a file sounds “better” or “worse” at a given bitrate depends largely on the algorithm used to implement the model; however, in general, higher bit rates can hold more data, likely reproducing Build a more accurate raw stream audio signal

I strongly suspect that the reason the MPEG standard specifies multiples of 2 is because binary computers can often optimize math involving both themselves and programmers.
This is a begging question. Don’t you think there is a mathematical/arithmetic reason for the chosen bitrate value? Or doesn’t the mere presence of VBR justify any limits on possible bitrates?
@slhck I’ve just updated my answer to provide more relevant details, please let me know if this answers all questions.
MPEG 1 Layer-III (mp3) files are streams of frames.

This web page details the data structure of the framework.

As you can see, only 4 bits are allocated to determine the bitrate. When designing a format for live streaming, you don’t want to waste more space than describing the stream.

I’m not sure exactly why 4 bits was determined to be a good compromise between space footprint and “bitrate resolution” – for the particular bitrate chosen, they were probably chosen based on the lowest and highest quality range that the engineer considered acceptable. mp3 algorithm.

Probably most MP3 players read one frame at a time, probably trying to “early” buffer at least one frame when decoding/playing the current frame.

The size of the frame and possibly the RAM allocated to it is as follows:

FrameSize = 144 * BitRate / SampleRate when the padding bit is cleared.
FrameSize = (144 * BitRate / SampleRate) + 1When the padding bit is set.
Higher bit rate/sample rate = more RAM required.

128 Kbps is probably popular as it is the default setting for many encoders.

Also, a colleague gave me insight into the discussion: 128 Kbps also roughly translates to “minutes in a minute” (unverified though), probably has something to do with that as well.

When “raw” data is logged, that data is buffered in chunks. These blocks will obviously be powers of two. It’s conceptually easier if you have an integer number of blocks per second.

Why are MP3 bitrates often multiples of 32?

I understand why multiples of 2 are often found on computers due to their binary nature, but I can’t figure out that the most common mp3 bitrates (64kbps, 128kbps, 160kbps, 192kbps, 256kbps, 320 kbps, etc.) also tend to follow this rule.

Since MP3 is just a sequential encoding of sound waves, why does it matter that each second is represented by thousands of digits per second that are divisible by 2?
Do music players like iTunes continue to read the file and play the encoded sound regardless of where the second limit is, or will they read the file every second?
In the latter case, reading a 256kbps file requires slightly fewer memory pages than reading a 257kbps file, but the player can always read 256kbit chunks, regardless of their bitrate, and process them incrementally , it is right?
Is MP3 popular at 128kbps because it’s a generally accepted bitrate, or does it really have some advantages over 126kbps and 131kbps files? Very slight difference in quality/file size?

For constant bit rate (CBR) encoding, the MPEG-1 Audio Layer III standard specifies standard bit rates of 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s. There are other definitions in the MPEG-2 standard, but they’re also all multiples of 2 (actually all multiples of 8 are in the range 8 to 160 – see the table called “Bitrate Index” at the link above ).

MP3: the ideal sampling frequency according to each use.

With MP3 and other audio formats, it is important to use the same sample rate from recording to playback whenever possible. While you can convert the sample rate at any time, sample rate converters almost always produce artifacts. The following sample rates are ideal for various applications:
To convert music CDs to MP3, for example, using our media player instructions, it is better to use the original 44100 Hz sample rate.

Mp3 Bitrate

DVD and BluRay sound is generally stored and played at 48,000 samples per second. So here you should stick to the 48 kHz sample rate. When converting 96 kHz audio to MP3, 48 kHz often sounds better than 44.1 kHz.

For pure voice recordings using a sound recorder or other software, a sampling rate of 8 to 9 kHz is sufficient, since small microphones above 4 to 5 kHz contain little sound energy.

If the sound quality of radio plays and audiobooks is not that important to you, because you want to carry as many stories as possible on an MP3 player, for example, use a sampling rate of 22050 Hz, although it is quite low. With half the sample rate, you can also cut the MP3 bit rate in half without losing quality.
If you digitize your old cassettes, 32 kHz was sufficient as the sample rate, because the tapes barely register frequencies above 16 kHz anyway. In other words, it would be unnecessary to use a higher sample rate.

What is the sample rate?

It is the speed with which “photographs” are taken (actually samples, in this case sound) and the more they are taken per second, the higher quality will be obtained. Think that the sound is represented by curves, and a curve will draw better the more detail or more dots it contains. It is impossible to represent well a curve with 3 segments, even with 10. The more segments it has, the more faithful it will be and the more similar it is to the original.

Because the quality is exactly that: how similar is the encoding to the original version. And there are two factors that count a lot: Sample rate and bitrate. Of course, the higher the sample rate and the higher the bitrate we will find a greater utilization of the disk space, which at this point is not usually a priority.

The size it occupies on the hard disk

Recall that the mp3 emerged precisely as a solution to save space on the hard disk. It was unmanageable to pretend to have a large music collection in WAV format (original format, without compression) on one of those small hard drives from a few years ago.

On the other hand, trying to download a complete WAV of a song from the internet or transfer it from one computer to another was also unmanageable, since they took up too much disk space.

Then the mp3 and later all the other compression formats, sought to achieve a good audio quality occupying perhaps 10 times the space that a WAV occupied.

What is the constant and variable bit rate in an MP3?

The FLAC format is more modern and with less loss of quality than MP3 but with a much larger file size, so MP3 remains one of the most widespread and used digital audio formats. With MP3 encoding at 320 kbps, excellent quality is achieved for most domestic digital audio players and for the human ear itself. But what is the difference between MP3 with a constant bit rate and a variable bit rate? Let’s clear the doubt.

Constant Bit Rate vs. variable bit rate

The variable bit rate coding method, known by the acronym VBR (Variable Bit Rate), is intended to achieve the highest audio quality evenly throughout the entire music track by performing an intelligent bit allocation during The coding process. That is, the Bit Rate varies to keep the audio quality as constant as possible throughout the file. In general, the MP3 encoding method with a variable bit rate produces a higher quality level than the constant bit rate coding for a similar Bitrate encoding.

Constant bit rate coding (CBR – Constant Bit Rate), meanwhile, is an encoding method that, instead of varying the bit rate, varies the audio quality to adjust to a fixed and constant bit rate at length of the MP3 file.

When to use a Constant Bit Rate?

If your priority is to get a certain MP3 file size, or at least a more or less predictable size, you should use a constant Bit Rate encoding.

When using a constant Bit Rate, the final file size is predictable although the quality of the MP3 audio will not be constant along the audio track.

When to use a Bit Rate Variable?

Bit Rate Variable encoding was developed for use when the main priority of MP3 encoding is a consistent quality throughout the track and the final file size is not the most important.

In an audio track some sections will be more difficult to code than others. Coding with Constant Bit Rate will encode the audio at the same Bit Rate set in advance, both in these more complex sections and in the easier sections. On the contrary, the coding with Bit Rate Variable will assign less bits in the easiest sections and will assign more bits when the section requires it because of its complexity. In this way, VBR encoding achieves a more uniform audio quality throughout the entire MP3 while the CBR achieves a constant Bit Rate.

In other words, if you use Constant Bit Rate the quality will be varied to reach the specified Bit Rate while using a Variable Bit Rate the Bit Rate is varied to achieve the highest possible and uniform quality.

All this does not mean that you don’t have to specify a Bit Rate for VBR encoding. Normally a fork will be specified, for example, 192-320 kbps, which means that a minimum of 192 kbps will be allocated for the easiest sections and up to a maximum of 320 kbps in the most complex sections of the music track.

The negative part is that MP3 encoding with Bit Rate Variable is not supported by some players. For me this disadvantage is part of the past and only if you try to play an MP3 in a really old player will not be able to read the file.