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

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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?

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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 Bit Rate or Bitrate in the audio?

Do you want to change the bit rate in your MP3 files? This can be useful if you need to reduce the size of your MP3 files, for example. A 320 kbps MP3 file, the highest bit rate allowed for MP3 files, could be reduced to 192 kbps to significantly reduce the size of the MP3 file.

There would be a loss of quality, but the difference would be insignificant for most listeners who use standard speakers or headphones. If you are an audiophile, in addition to having expensive audio equipment, you probably never use the MP3 format anyway.

Most likely, you will use a compressed or uncompressed lossless format such as PCM Audio, WAV, AIFF, FLAC, ALAC or APE. An uncompressed PCM audio file is approximately 10 times larger than a CD-quality MP3 file.

The MP3 format is a lossy format, which means that the audio quality is sacrificed to maintain the relatively small size of the files. Almost all sites will tell you that you should never convert an audio file from lossless format to MP3 format unless you are well with the loss of audio quality.

What is the normal bit rate in an MP3?

A normal bit rate for an MP3 is between 128 kilobits per second, or kbps, and 320 kbps. Different bit rates in this range have different advantages; To extract music from a CD to MP3, choose the bit rate that best suits your purposes. In general, the higher the bit rate of an MP3, the better the audio quality, but the larger the file size. The most common bit rates in MP3s are 128 kbps, 192 kbps and 320 kbps.

Bit rate

The bit rate of an MP3 indicates the density of the audio information contained in the file. Therefore, every second in a 192 kbps MP3 contains 192 kilobits (24 bytes) of data. The higher the bit rate, the more information the MP3 contains; The more information it contains, the quality is closer to the original audio recording. The bit rate also determines the size of the MP3 file: a four-minute song encoded at 128 kbps occupies a little more than 3.5 MB, while the same song encoded at 320 kbps requires more than 9 MB of space.

Choice of a bit rate

When you convert a sound file or extract your CD collection to MP3, you must specify the bit rate you want the encoder to use. Choose a bit rate based on the type of audio you are encoding and on your desired purpose. If you are converting a conference or other piece of vocal audio, a bit rate of 128 kbps is more than enough. If you are encoding MP3 files for use with a portable media player, a bit rate of 192 kbps usually provides a good balance between audio quality and file size. If you are converting your CD collection to MP3 for archiving, encode the files with a bit rate of 320 kbps.

Variable bit rate VBR

Coding with a variable bit rate changes the bit rate of the MP3 depending on the type of data present at a certain point in the audio. For example, a variable bit rate encoder could encode a portion of dense information from an MP3 at 320 kbps; however, when the audio contains a silent section the encoder lowers the bit rate to 32 kbps. If the bit rate of an MP3 has an unusual number (for example, 204 kbps), this indicates that the creator of the MP3 encoded it with a variable bit rate.

Bit Rate Conversion

Although most MP3 conversion and encoding programs can modify an MP3 at a different bit rate, this is not a good idea, generally speaking. Converting an MP3 to a higher bit rate does not add audio information to the file, so the sound quality does not increase. Converting an MP3 to a lower bit rate reduces the size of the file, but it could also introduce audio defects, since the encoder is compressing a file that is already compressed. If you need an MP3 file to have a different bit rate, re-encode an MP3 from the original audio CD or WAV file.

Does the birate of an mp3 affect the quality or is it just an impression?

Since the mp3 appeared, I always understood that it is a lower format than the CD quality, no matter how much bit rate it contains. When it comes to mixing and producing my personal recordings, I have used wave support and when converting it to mp3 it is impossible not to distinguish the differences: at least, the reduction of bass and treble is very noticeable. The dishes of the drums or the bass are very opaque and hidden, sometimes they even disappear. But in these assessments I think that perhaps my subconscious has betrayed me in some cases.

First of all I will explain some basic things about mp3 and its bitrate (or bit rate). Compressing sound means loss of quality, so you have to pay attention to how much information is transmitted per unit of time: the most used encodings for mp3 are 64, 128, 192 and 320 kbps (kilobits per second). Thus, a song of 64 will occupy less space than one of 128 and so on. Nor should we forget the CBR, constant bitrate, and the VBR, variable bitrate. The latter is more advisable when compressing different parts of a recording with various bitrates.

Well, there has always been controversy and confrontation between those who prefer to save space and opt for the musical amount, claiming that the quality differences of the mp3 are almost imperceptible to our ears, and those who bet on the qualitative details of the music, preferring heavier files and with higher bitrates, emphasizing the multiple peculiarities that our auditory system loses to lower quality media. I have been a supporter of the second group, although current opinions about it make me doubt my choice.

I recently informed myself about a study conducted by programmer Jeff Atwood through his blog, in which he tried to discover if normal people, fond of music (but without becoming music experts) notice the differences between mp3 formats . More than 3,500 people participated in the study, who had to listen to 5 different audio files (with bitrates between 128 and 320 Kbps and one without compression) and vote from 1 to 5 depending on the quality they had received. Naturally, users were unaware of the characteristics of the media so as not to influence their objectivity. The results were as follows: the 128 Kbps CBR mp3 was undoubtedly considered the worst; the one of 160 Kbps VBR would be the one of better quality, surpassing even the one of 320 CBR (the variable bit rate would be higher than the fixed one). This is very curious since it is assumed that an original CD would house a quality between 192 and 256 Kbps. According to the study cited, the mp3 of 160 would have more quality than the compact, which seems absurd, so I tend to think that the Most people, after 160, do not distinguish some sounds from others, also taking into account that the subconscious can deceive us and make us imagine what it is not.

In short, the best way to compress, saving on storage and with optimal sound, would be thanks to the 192 Kbps VBR bitrate. From there, onwards, it is very difficult to appreciate nuances and alterations.

We must also consider something logical: if we recompress a file of 128 and convert it to 192 Kbps, improvements will not be achieved and we will lose space. If we do the opposite, go from 192 to 128 Kbps, we will reduce the quality somewhat but reduce the weight of the file.

I think this experiment obviates interesting data such as knowing the player or equipment from which the music was broadcast (computer, speakers, headphones, hi-fi equipment, etc.), listening time (our senses do not lend the same attention for 5 minutes than during 20) or musical style (electronic, rock, classical, etc.), since all these factors can greatly influence the final result.

Even, and this is proven: the volume (if the file is normalized) and the dynamics (a normalization like the one that only Mp4Gain does) manage to make one perceive the music as with higher quality.