Audio compression


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Audio compression

Audio compression
Audio compression

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

Audio compression
Audio compression

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.

The storage unit of the computer is ” byte (Byte)”. In the computer, 1 byte consists of 8 bits, that is, 8b(bit)=1B(Byte). In computer parlance, data storage is expressed in decimal and data transmission is expressed in binary, so 1KB=1024B=1024×8b. This is also part of the reason why the hard drive capacity we see does not match the actual capacity.

Go back and talk about audio compression, the bitrate of the audio is actually the compression ratio. So the bitrate really just defines the size of the file, but because under normal conditions the larger the file, the less data you lose, so the sound quality is relatively higher. However, the bit rate itself does not directly affect the quality of the file. For example, if we take a 128kb file as the source file, even if it is converted to a 320kb file, the sound quality will not be better than 128kb. .

So what exactly do the numbers and letters in the bitrate mean? First look at the full name of 128k “128kbps”, let’s try to break it down: 128 is a number, k is a thousand symbol, b is a unit, s is a second, and ps is actually “/s”. Thus, 128kbps is 128kb/s. That’s 128kb per second.

Note that the b here is a lowercase b, or bit. Knowing this, we can calculate the approximate storage space that a 128kb file occupies: 128*1000=128000b/s÷8=16000B/s÷1024=15.625KB/s*60=937.5KB/min÷1024=0.9155 MB/ min. So 128Kb audio file size is about 0.92M or 916Kb per minute, so 128Kb mp3 is about 1M in size. You can test and check it locally.

Before talking about lossy and lossless, there are two words to explain to you, that is, we will see CBR and VBR when compressing MP3. And CBR is constant bit rate, constant bit rate; VBR is variable bit rate, dynamic bit rate. In theory, VBR’s way is to automatically correct some bitrates according to the specific frequency of the sound in the source audio file, to achieve a smaller file with the same bitrate effect.

Let’s talk about lossy and lossless. In a nutshell, lossy compression is about achieving compression by removing some less important data from existing data; lossless compression is about achieving compression by optimizing the layout. Since these compression methods involve deeper technical knowledge, we won’t say more, and we can probably look at it this way: lossy compression is like removing some unimportant particles in an article to achieve the purpose, after decompression, it is you deleted the content cannot be recovered; Lossless is achieved through typesetting. After decompression, complete WAV data can be obtained, just like our commonly used winzip and WinRAR.


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

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

Audio Bitrate
Audio Bitrate

【Preface】

Audio Bitrate
Audio Bitrate

Some time ago, a colleague came across a very troubled client. The mess was said to have been caused by the client asking him to provide song files larger than 100MB-200MB in size. And my colleagues don’t know much about audio formats, so they started endlessly fumbling about FLAC, WAV and audio size. In the end, the colleague did not clearly explain to the customer what was going on.

After that, other things happened that made me feel that in the music industry there are too many practitioners around me who have an extremely poor understanding of music and even lack some basic knowledge related to music. I don’t even have the idea to understand, which makes me very sad. It seems that music has only one product attribute, and our practitioners only need to organize the shelves, encode various products, and recommend products to users with the big data of user purchase records, and they don’t need to worry about why the users users like this. Brands, what features these products have, and provide users with various services with cold data.

Therefore, I think it is necessary to write something. I don’t expect practitioners to become people who really love music. I just hope that even if you still think of “her” as a commodity, you can first figure out what you’re selling. and what is..

PS: The contents of the first lesson are multimedia files. Since the relevant content involves a lot of technical issues, it seems a bit boring, but if you read it carefully, you will see that it is actually very easy to understand, but these basic knowledge can be very helpful.Improve your skill well. Also expect more interesting content about records, musical styles, etc. which I will post soon.

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.

What is the proper audio bitrate? Part 2

What is the proper audio bitrate? Part 2

audio bitrate
audio bitrate

What does audio bitrate mean?

audio bitrate
audio bitrate

 

Are higher audio bitrates better?

 

Encoding rate (Kbps) * total length of the song (seconds) / 8 = file size (KB), if divided by 1024, it is the size in MB.
mp3 is lossy compression, the smaller the file, the higher the loss. The relationship between bitrate and audio and video compression is simply that the higher the bitrate, the better the audio quality and video, but larger is the encoded file; yes The opposite is true for lower bitrates.
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.
The bitrate value is compared to the actual audio:
16 Kbps = phone sound quality
24 Kbps = increase phone sound quality, short wave transmission, long wave transmission, European standard medium wave transmission
40 Kbps = American standard medium wave transmission 56 Kbps
= voice
64 Kbps = voice boost (best bit for mobile phone ringtones) = tape (best setting for mobile phone stereo MP3 player, best setting for low-end MP3 player
112 Kbps = FM stereo radio FM 128 Kbps ) 160 Kbps = Hi-fi HIFI (best setting for mid-range MP3 player to high-end MP3 players) 192 Kbps=CD (best setting for high-end MP3 player) high-end ) 256 Kbps= Studio Music Studio (for music enthusiasts) In fact, with the advancement of technology, bit rates are also getting higher and higher, MP3 has a maximum bit rate of 320 Kbps, but some formats can achieve higher bit rates and superior sound quality. For example, the emerging APE audio format can provide true audiophile-level lossless sound quality and smaller volume than WAV format, and its bit rate is typically 550kbps.

What is the proper audio bitrate?

What is the proper audio bitrate?

audio bitrate
audio bitrate

What does audio bitrate mean?

audio bitrate
audio bitrate

What is the proper audio bitrate? When recording gameplay videos and putting them on the Internet, the bit rate used to suppress the video depends on the end use. If you are going to share it on an online video site like Youku, the requirements for videos will vary from site to site. Youku’s requirements for video formats are as follows: Generally speaking, when uploading videos to online sites, it is recommended that the video be encoded in H264/X264, and the video bitrate should be 1600Kbps;
What is the proper audio bitrate?

 

When recording gameplay videos and putting them on the Internet, the bit rate used to suppress the video depends on the end use.

 

If you are going to share it on an online video site like Youku, the requirements for videos will vary from site to site. Youku’s requirements for video formats are as follows:

 

Generally speaking, when uploading to online video website, it is recommended that the video be encoded by H264/X264, the video bit rate should be 1600Kbps; audio must be AAC encoded and audio bitrate must be 128 Kbps.

 

If you plan to put it on the network drive and share it with friends, it is recommended to use a higher bit rate. For example, the video is encoded by H264/X264 and the video bit rate is 2400Kbps; the audio is AAC encoded and the audio bitrate is 128 Kbps.

What does audio bitrate mean?

 

The code rate is the number of data bits transmitted per unit of time during data transmission. Generally, the unit we use is kbps, that is, kilobits per second.
A common understanding is the sample rate. The higher the sample rate per time unit, the higher the precision, and the processed file will be closer to the original file, but the file size is proportional to the sample rate, so almost all encoding formats pay attention to This is how to use the lowest code rate to achieve the least distortion. The cbr (fixed code rate) and vbr (variable code rate) derived from this core are all items in this regard, but things are not absolute, In terms of audio, the higher the bit rate, the lower the compressed ratio, the smaller the sound quality loss and the closer it is to the sound quality of the audio source.
Basically, the sound quality of the two data 44.1 and 128 in the MP3 song attribute is very good.

What do the bits, bit rate and sample rate of an audio file mean?

What do the bits, bit rate and sample rate of an audio file mean?

bits, bit rate and sample rate
bits, bit rate and sample rate

For example, the common mp3 format audio source

bits, bit rate and sample rate
bits, bit rate and sample rate

In order to store a continuous physical signal (well, tell me about Planck’s constant…) in a computer, it must be converted to a digital signal. In acoustics, a digital signal is a digital representation of the amplitude of the sound wave at any moment.

Sound waves are longitudinal waves, which are difficult to draw. The following figure is replaced by transverse waves (the concept of longitudinal waves is the phenomenon that the density of air or other media changes regularly due to energy. The peaks represent high density, the troughs represent low density, and the horizontal line is the average density, i.e. silent state)

 

Using high school physics, waves contain two dimensions, one is intensity and the other is time. “Number of digits” indicates how many levels sound waves are divided into from the strongest to the weakest; “Sampling Rate” determines the precision of the time axis or the sampling density, that is, the length of time represented by each red dot, and the code rate is one second The number of dots on the clock, multiplied by the space that each point occupies.
So the so-called 24 bits consist of dividing the intensity of the sound wave by 2 at power level 24, occupying 3 bytes of space. Obviously, the finer the grade, the more details are restored.

The sample rate is generally 44100 Hz for CD (Hertz = times/second), 48000 Hz for DVD, and 96000 Hz as standard. As with the number of digits, the more points you get in a single second, the more details you retrieve. Why does CD take this value? Because the hearing range of the human ear is generally believed to be between 20 and 20,000 Hz. A peak and a trough need to be represented, and at least two sampling points are required. Therefore, the CD can represent the sound of 22050 Hz at most, but this sound does not have any detail, because if there are only two peak and valley points, the average waveform is completely lost. Therefore, there will be a higher sampling rate.

If it’s in a lossless uncompressed format, the bit rate is strictly equal to the number of bits * sample rate * number of channels. And typically, the MP3 bitrate you can see just represents how much capacity the format needs to describe this one second of audio.

MP3 is lossy compression. In the compression process, some information is lost, but the lost information cannot be represented by the number of bits and the sampling rate. Generally, the higher the code rate, the less information is lost. Mathematically, bitrate and sound quality are proportional. As for whether you can hear it or not, it depends on many factors. The MP3 algorithm is not complicated, of course, to understand it you have to learn what the Fourier transform is.

There is also lossless compression (representing APE, FLAC, etc.), which also has a bitrate, and this bitrate has nothing to do with sound quality. It also describes how much capacity the file uses to describe one second of audio content, but the same audio content can be compressed to different sizes (compression ratios), similar to zip compression ratios. No matter how big you compress it, in the end it can be restored to the same file. So if you see someone looking for a lossless bitrate, you can basically conclude that the product is a bad pen.

Does MP3 quality depend on how much KBPS is the bitrate?

Does MP3 quality depend on how much KBPS is the bitrate?

MP3 quality
MP3 quality

KBPS = fast bitrate, the read speed must be to play this file smoothly,
because mp3, a common streaming format on the internet, can be downloaded while listening.

MP3 quality
MP3 quality

If the download speed is slower than the playback speed, it will stop. (LAG), and the bit rate
refers to the minimum required download speed, but since the lower the required download speed,
the higher the compression required, and MP3 is a destructive compression format, so the bitrate
also
will affect the quality of the file. Bitrate is not the biggest influencer on overall sound quality, but the main influencing factors are sample rate and bit depth. The
sample rate refers to the number of times your computer records the sound per unit of time. Usually,
the sample rate used for a CD is 44100MHz, so
you can get good quality by setting the file to this, but remember that the bitrate should be set to 96KBPS or higher.
Reduce distortion.

Normalize the volume and loudness of an mp3 or a video easily

Normalize the volume and loudness of an mp3 or a video easily

Normalize the volume and loudness of an mp3 or a video easily
Normalize the volume and loudness of an mp3 or a video easily

It’s absolutely easy if we use Mp4Gain, it only takes one click of a button and all audio and video files are volume normalized.

Normalize the volume and loudness of an mp3 or a video easily
Normalize the volume and loudness of an mp3 or a video easily

Today we find many problems with this volume issue because they are compressed by different compressors and above all using different bitrate and sample rate settings.

People don’t realize how important this whole issue is, but Mp4Gain solves it automatically. Not only through bitrate and samplerate, but also by making a deep analysis of each frame and optimizing each frequency band, so that the result is magnificent.

The largest number of inquiries we receive by email refer to that difference in volume levels in the mp3s and also between the mp4s.

And what we have been able to corroborate is that, to a large extent, many are due to having been encoded with wrong settings, for example a very low bitrate.

Because the bitrate implies the amount of information or detail that the audio or video can pass per second and this translates into the detail that a video has, for example. Which immediately affects the quality of the aforementioned video.

Mp4Gain is the solution to normalization problems.

Non-professional and easy to understand popular science on sample rate, bit depth, bit rate and lossless – Part 2

Non-professional and easy to understand popular science on sample rate, bit depth, bit rate and lossless – Part 2

sample rate, bit depth, bit rate and lossless
sample rate, bit depth, bit rate and lossless

Bit rate kbps (kp/s)

sample rate, bit depth, bit rate and lossless
sample rate, bit depth, bit rate and lossless

In lossless uncompressed formats (such as .wav), bit rate = sample rate x bit depth x number of channels. In lossy compression (for example, .mp3), the bitrate does not equal this formula, because the original information has been destroyed. The bitrate describes the amount of information about the audio in one second, so the total size of the sound file is the bitrate x the total duration. The bit rate is also called the bit rate and the unit is the bit rate (bps, bit per second). Usually 128kbps and 320kbps are bit rates when listening to songs, of which 320kbps is the highest bit rate of mp3 format. But compared to wav file with 44.1 kHz sample rate and 16 bit bit depth (calculate two channel bit rate is 44.1 x 16 x 2 = 1411.2 kbps), it is far from the same. After compression, the bit rate has changed. Bitrate in lossless compression has nothing to do with sound quality, and bitrate in lossy compression is positively correlated with sound quality.

 

lossless compression
Lossless compression refers to compression (conversion) between formats without loss. Regardless of the format that is compressed (converted), the sound quality is the same and can be restored to the same original file. Lossless generally refers to lossless compression, and there is no such thing as lossless code rate. The compression of various formats corresponds to an algorithm (or encoding), and a decoder is required to decode during playback, and different decoders can also affect the integrity of the decompressed file. Common lossless formats are:

wav – A Microsoft sound file format, which is the closest uncompressed format to real sound (followed by midi), supporting multiple sample rates and multiple quantization precisions. All lossless formats are essentially wav compression, which is converted back to wav when played.

flac: Free Lossless Audio Coded, which is an international general format, characterized by high compression ratio and mature encoding algorithm. When the flac file is damaged, it can still be played normally. Furthermore, this format is also the first lossless format widely supported by hardware.

monkey: The file format converted from CD ripping using Monkey’s audio software, but the advantage is not prominent and decoding is slow.

wma-lossless: It is also produced by Microsoft. It is characterized by a high compression ratio, but it has not become mainstream.

aiff: Produced by Apple, it is the standard audio format on Apple computers.

DSD: I don’t know much about Sony Dafa and I can’t appreciate the spicy culture.

 

lossy compression
Lossy compression refers to the loss of sound information during the compression process, and the lost sound cannot be represented by the sample rate and number of bits. But the feature is that the compressed file becomes very small and is often used in streaming media. Common lossy formats are:

mp3: A complex algorithm developed to simulate human hearing, known as a “psychoacoustic model”. It improves the compression ratio, lowers the bit rate, and reduces the footprint by extracting some frequency bands in the audio, but at the same time, the details of the sound, such as the emotion of the human voice, the reverberation in the later stage, etc., have been deformed. It is also difficult to distinguish wav and mp3 quickly if you listen blindly and need to use equipment. MP3 is currently the most popular audio compression format, which can best preserve the sound quality before compression.

wma: Microsoft’s masterpiece, characterized by lower bitrate (such as 64kbps), wma can get smaller volume under the same sound quality conditions as mp3. And at ultra-low bit rates (like 16 kbps), wma sound quality is much better than mp3.

aac: The storage format for sound files on Apple computers.

ogg – Completely free, open, and patent-free, but less popular.

Non-professional and easy to understand popular science on sample rate, bit depth, bit rate and lossless

Non-professional and easy to understand popular science on sample rate, bit depth, bit rate and lossless

sample rate, bit depth, bit rate and lossless
sample rate, bit depth, bit rate and lossless

HZ sampling rate

sample rate, bit depth, bit rate and lossless
sample rate, bit depth, bit rate and lossless

The sound from the outside world is an analog signal, which is converted to a digital signal represented by 0 and 1 in the digital device and then stored. Digital signals are discrete, so sampling rate refers to the number of samples per second. The higher the sample rate, the more realistic the restored sound will be. Since the hearing range of the human ear is 20 Hz to 20 kHz, according to Shannon’s sampling theorem (also called Nyquist’s sampling theorem), in theory, audio formats with a sampling frequency greater than 40 kHz they can be called lossless formats. However, the sound obtained at the 40 kHz sampling rate does not have any detail and all frequencies are only sampled with a peak and a valley. The sample rate of general professional equipment is 44.1 kHz. 44.1 kHz is the lowest sample rate in professional audio, also known as “CD-quality sound” (22.05 kHz sample rate is broadcast-quality sound). There are 96kHz, 192kHz, etc., more detailed of course, hearing the details at these higher sample rates is ear and equipment dependent.

 

bit depth
To reproduce sound as accurately as possible, a high sample rate is not enough. Describes a sample point, the horizontal axis (time) represents the sample rate and the vertical axis (amplitude) represents the bit depth. 16bit means that 16 bits (2 bytes) are used to represent the level of the sample point (in general, it is proportional to volume) The degree of precision that can be achieved when encoding, i.e. the vertical axis is divided into 16 parts Describe the level, such as -3dB and -3.1415926dB accuracy difference. Similarly, there are 20 bits and 24 bits. 16-bit is considered to be the lowest bit depth standard in the field of professional audio and, like the 44.1 kHz sample rate, is the common standard for consumer and professional audio products. Bit depth is also directly related to the size of the signal-to-noise ratio, which directly affects the overall dynamic range of the recorded signal.

Bit rate and audio quality

Bit rate and audio quality

Bit rate and audio quality
Bit rate and audio quality

Audio and video quality

Bit rate and audio quality
Bit rate and audio quality

We know that there are two types of compression, called lossy, which discards information to make the file smaller, and lossless, which simply uses zip-type compression to reduce the size.

The quality of the audio depends on the number of bits that can be transmitted per second.

Let’s understand this concept well, it’s simple.

If the audio file has a certain amount of information and we can transmit ALL of it, then we will obtain a very high quality and there will be no loss of information.

On the other hand, if we must discard more data because we can only transmit a small amount, quality will necessarily be lost.

For this reason, the amount of information collected by the sample rate and the number of bits that can be transmitted per second go hand in hand. It will be useless to have a very good samplerate if the bitrate is low and forces us to transmit only a small portion of the abundant data available.

Many times when encoding a file, of any format, information is lost either by using a poor encoder or by mutilating the wrong settings.

Mp4Gain is highly efficient at all of this and uses the best settings for you to get the best quality, both when encoding and converting.