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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 ).

Why are MP3 bitrates often multiples of 32? (power of 2) part 2

Depending on the resource, VBR can be encoded by changing the bitrate between a fixed rate above each frame, or by sharing the available bits in adjacent frames (effectively producing a non-standard bitrate for the two frames combined).

the fixed frame depends on the sampling rate, 1152 samples per frame. There is no limit to the size of the frame itself, nor to the base 2 size of the frame (ie 417 bytes for a 128 kbit/s MP3 sampled at 44.1 kHz).

In the end, a file encoded at 126kbps will sound worse than a file encoded at 128kbps, and similarly a file encoded at 131kbps will sound better. However, MP3s are encoded according to the compression psychoacoustic model of a specific encoder. The amount by which a file sounds “better” or “worse” at a given bitrate largely depends on the algorithm used to implement the model, but in general higher bitrates allow for more data, presumably for rebuild a more accurate original transmission. audio signal

Why are MP3 bitrates often multiples of 32? (power of 2)

Some people say:

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

Since MP3 is just a sequential encoding of sound waves, why is it important to represent each second in kilobits divisible by 2?
Does a music player like iTunes continue to read the file and play the encoded sound regardless of the second limit, or does it read the file every second?
In the latter case, reading a 256kbps file requires reading slightly fewer memory pages than a 257kbps file, but the player can always read 256KB chunks, regardless of their bitrate, and just process them automatically. incremental, right, Bar?
Are 128kbps MP3 songs popular simply because it’s a generally accepted bitrate, or do they really have any advantages over 126kbps and 131kbps files, apart from a 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/second. There are a few others defined in the MPEG-2 standard, but they are also multiples of 2 (actually all multiples of 8 in the range 8 to 160 – see the table called “Bitrate Index” in the link above) .

Technically, there is nothing that limits the MP3 bitrate to a multiple of 2, since variable bitrate encoding can be used, or a custom bitrate can be achieved using some flags not used in the MPEG specification ( although this must be implemented manually). . In order 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 have a bitrate of two.

Audio bit rate, bit rate

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

Bitrate in sound refers to the amount of binary data per unit of time after converting an analog sound signal to a digital sound signal, which is an indirect measure of audio quality.

Bitrate refers to the sampling rate at which digital sound is converted from analog to digital format. The higher the sampling rate, the better the quality of the restored sound. As a benchmark for the efficiency of digital music compression, bit rate indicates the rate of the number of bits bps (bit per second, bits per second) transmitted per unit of time (1 second). Kbps (in layman’s terms is 1000 bits per second) is usually used as the unit. The bit rate of digital music on CD is 1411.2 kbps (that is, to burn 1 second of CD music, 1411.2 × 1024 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 take up a lot of memory capacity. The most commonly used bitrate for music files is 128 kbps, and MP3 files can generally use 8 to 320 kbps. In the same way, most of them are 32-256 Kbps. Of course, the wider the rate, the better, but 320 Kbps is the highest level at the moment.

Bitrate calculation formula
The basic algorithm is: [Bit rate] (kbps)=[file size] (bytes) X8/[time] (seconds)/1000

Special algorithm for audio files: [bit rate] (kbps) = [quantization sample point] (kHz) × [bit depth] (bit/sampling point) × [number of channels] (typically 2)

For example, the D5 drive has a capacity of 4.3G, which takes into account different audio formats, so it is calculated as 600M (so the remaining capacity is 4.3*1000-600=3700M), so the video file should not be larger than 3.7G, in this example, take The capacity of the video file is 3.446G, and the length of the video is 100 minutes (6000 seconds). The calculation result: the bit rate is approximately equal to 4933kbps.

What is a good bitrate guide for mp3 files? Part 2

To produce high-quality MP3 files of classical and jazz music, the optimal bitrate depends on the characteristics of the song.

Smooth jazz can usually be copied at 192kbps to create a good balance between file size and diminishing returns, although 256kbps may sound better in a home entertainment center. A classical orchestra should be 256kbps for a portable player, but if you want to burn a CD at home or in your car, a 320kbps file might be a better option.

For saturated music such as hard rock, metal, arena, pop, electronic and house music, 320 kbps will provide the best results. The higher the number of bits per second, the more complex acoustic envelope will be preserved.

If possible, it’s best to create MP3 files with variable bit rates. This allows the encoding program to determine if a particular frame of music requires the full bit rate. Otherwise, the program will reduce data retention for that frame, resulting in a smaller file without sacrificing quality. Forcing the program to “oversample” frames can produce artifacts.

While this article is intended as a general guide, he or she may be equally satisfied with a lower bitrate for a particular song or songs in general. Many factors affect our ability to judge the quality of music, not only the devices we use but also our activities while listening to it. For example, for those who listen to MP3 files while exercising or taking a walk, external noise can make it more difficult to tell the difference in quality. Conversely, audiophiles may prefer to sample at 320kbps, regardless of their equipment, type of music, or listening habits.

If you create your own MP3 files, there are other settings that affect quality. LAME is an excellent MP3 encoder that is free and has many graphical interfaces as the interface for this popular command line program. LAME allows users to adjust many settings to generate high-quality MP3 files in seconds. You can also experiment with various bitrates in your source file to find the best subjective balance between quality and file size.

What is a good bitrate guide for mp3 files?

(a good bitrate guideline for mp3 files?)

MP3 files are compressed audio files created from audio formats such as wave (.wav). Wave files replicate analog recordings and digital sound files at the expense of large file size, while MP3 files sacrifice some quality for a smaller footprint. There are several factors that mitigate the quality sacrifice during the conversion process. With the correct bitrate and settings, MP3 files can provide very high quality results, making them very close to the original wave files when played on portable audio players.

An mp3 player.

The balance between file size and quality is somewhat subjective. For audiophiles, any difference is noticeable. Others may simply not be able to tell the difference between a high quality MP3 file and a raw wave source. In many cases, the nuances of the sound environment will only become clearer when played through a high-quality stereo system.

MP3s are compressed digital music files that sacrifice quality for file size.
MP3 files are primarily targeted at portable audio players. In this field, high-quality MP3 files are played with incredible sound due to their small file size. With the limited memory of portable players, it makes sense that one would want MP3 files to be as small as possible while maintaining the highest possible quality.

For this, one of the most important factors when creating MP3 files is the bit rate. In general, the more bits per second that are preserved from the original file, the higher the quality of the MP3 and the larger the file size. Lower bit rates reduce size and quality. The idea is to use the bitrate for maximum realism without saving unnecessary data, which just creates larger files with no noticeable difference to the ear.

For voice recordings such as lectures or language lessons saved to waveforms, a bit rate of 32 kilobits per second (kbps) is acceptable, although 64 kbps may offer better quality, depending on the source. At 32 kbps, the sound may sound “flat”, but that’s understandable. A 64 kbps MP3 file created from a voice recording should sound nearly identical to the original.

Desaturated acoustic music with simple arrangements should work fine at 192kbps bitrate. You can choose 256 kbps if the music will be played on a high quality device. Music that falls into this category includes folk, boy band songs, easy listening, and folk music. There are also works by many classic artists such as James Taylor, Linda Longstadt, Jonny Mitchell, and Simon Garfunkel.

Audio Intro Part 3

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*Bit depth PCM/8 unsigned short BITDEPTH; ///Bit depth character PCM DATA[ 4 ]; ///Data flag data unsigned DATALEN ; ///Data partial total length (bytes) };

data organization
After the file header is the data part of the WAV file. Its data organization is: the left channel value of the first sample point, the right channel value of the first sample point, …, the left channel value of the last sample point, the right channel value of the last sample point value. Each value has a bit depth of bits.

Generate a simple wav
First complete the Wav header.

WAVHeader getHeader ( int number )
{
WAV Header res; memcpy (res.RIFF, “RIFF” , sizeof (res.RIFF)); memcpy (res.WAV, “WAVE” , sizeof (res.WAV)); memcpy (res.FMT, “fmt ” , size of ( res.FMT )); res.SubchunkSize= 0x10 ; res.DATATYPE= 1 ; res.CH= 2 ; res.F=F; res.BITDEPTH=DEPTH; res.BYTERATE=res.F*res.BITDEPTH/ 8 *res.CH; res.DATAUNITLEN=res.CH*res.BITDEPTH/ 8 ; memcpy(res.DATA, “data”

, size of ( res.DATA ));
res.DATALEN=num*res.DATAUNITLEN;
res.LEN=res.DATALEN+ 44 -8 ; returnres; }

First, define the key name – frequency comparison table.

const double keyf[]=
{ 27.5 , 29.1352 , 30.8677 , 32.7032 , 34.6478 , 36.7081 , 38.8909 , 41.2034 , 43.6535 , 46.2493 , 48.9994 , 51.9131 , 55 , 58.2705 , 61.7354 , 65.4064 , 69.2957 , 73.4162 , 77.7817 , 82.4069 , 87.3071 , 92.4986 , 97.9989 ,

103.826 , 110 , 116.541 , 123.471 , 130.813 , 138.591 , 146.832 , 155.563 , 164.814 , 174.614 , 184.997 , 195.998 , 207.652 , 220 , 233.082 , 246.942 , 261.626 , 277.183 , 293.665 , 311.127 , 329.628 , 349.228 , 369.994 , 391.995 , 415.305 , 440

, 466.164 , 493.883 , 523.251 , 554.365 , 587.33 , 622.254 , 659.255 , 698.456 , 739.989 , 783.991 , 830.609 , 880 , 932.328 , 987.767 , 1046.5 , 1108.73 , 1174.66 , 1244.51 , 1318.51 , 1396.91 , 1479.98 , 1567.98 , 1661.22 , 1760 , 1864.66 ,

1975.53 , 2093 , 2217.46 , 2349.32 , 2489.02 , 2637.02 , 2793.83 , 2959.96 , 3135.96 , 3322.44 , 3729.31 , 3951.07 , 4186.01 } ___ { “A-0” , “A#0” , “B- 0” , “C-1” , “C#1” , “D-1” , “D#1” , “E-1” , “F-1”, “F#1”

, “G-1” , “G#1” , “A-1” , “A#1” , “B-1” , “C-2” , “C#2” , “D-2″ , ” D#2″ , “E-2” , “F-2” , “F#2” , “G-2” , “G#2” , “A-2” , “A#2” , “B- 2” , “C-3” , “C#3” , “D-3” , “D#3” , “E-3” , “F-3” , “F#3” , “G-3” , “sun#3” ,