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What is the difference between 128k and 320k music? Part 3

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?

【Audio Compression】

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 we use regularly are based on the original “WAV” file of the audio CD (44.1khz sample rate, 16bit sample precision, 2ch). The original recorded sound data is stored in a matrix, which is in PCM format, while WAV format is an encoding format developed by Microsoft. Its function is to reproduce the data in PCM format 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. .

 

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

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 frequency is the number of points in the middle of the length per second (the horizontal axis of 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 previous 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.

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

I can’t fully understand music in words.show all

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【Preface】

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, some 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 merchandise attribute, and our practitioners only need to organize the shelves, encode various merchandise, and use the big data of users’ purchase records to recommend merchandise to users, no matter why to users. they like this. features that these products have, and use cold data to provide users with various services.

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 content of the first lesson is about media files. Since the relevant content involves a lot of technical issues, it seems a bit boring, but if you read it carefully, you will find that it is actually very easy to understand, but this 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.

Related Audio Attribute Part 3

How samples are combined

This is mainly for two-channel or multi-channel audio. For a two-channel audio, it can be combined in the following two ways:

interleaved Taking stereo as an example, a stereo audio sample is obtained by interleaving the storage of two mono samples.
flat. The samples of each channel are stored separately.

The data after FFmpeg audio decoding is stored in the AVFrame structure.

In packed format, frame.data[0] or frame.extended_data[0] contains all the audio data.
In Planar format, frame.data[i] or frame.extended_data[i] represents the data of the i-th channel (assuming channel 0 is the first), the size of the AVFrame.data array is set to 8, if If the number of channels exceeds 8, you should get the channel data from frame.extended_data.

sample format
The sample formats in FFmpeg are mainly:

copy code
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = – 1 ,
AV_SAMPLE_FMT_U8, /// < 8 bits unsigned
AV_SAMPLE_FMT_S16, /// < 16 bits
signed AV_SAMPLE_FMT_S32, /// < 32 bits
signed AV_SAMPLE_FMT_FLT, /// < float
AV_SAMPLE_FMT_DBL, /// < double

AV_SAMPLE_FMT_U8P, /// < 8 bits unsigned, flat
AV_SAMPLE_FMT_S16P, /// < 16 bits signed, flat
AV_SAMPLE_FMT_S32P, /// < 32 bits signed, flat
AV_SAMPLE_FMT_FLTP, /// < float, flat
AV_SAMPLE_FMT_DBLP, /// < double, flat
AV_SAMPLE_FMT_S64, /// < 64 bits
signed AV_SAMPLE_FMT_S64P, /// < 64 bits signed, plain

AV_SAMPLE_FMT_NB /// < Number of sample formats DO NOT USE if dynamically linked
};
copy code
to illustrate:

1. U8 (8-bit unsigned integer), S16 (16-bit integer), S32 (32-bit integer), FLT (single-precision floating-point type), DBL (double-precision floating-point type), S64 (64-bit integer), those not ending with P are interleaved structures, and those ending with P are flat structures.
2. Flat mode is FFmpeg’s internal storage mode, and the audio files we use are in packed mode.
3. The FFmpeg audio sample format that decodes different output audio formats is not the same. The test found that the data output by AAC decoding is in floating point AV_SAMPLE_FMT_FLTP format, and the data output by MP3 decoding is in AV_SAMPLE_FMT_S16P format (the mp3 file used is 16-bit deep). For the specific sample format, you can see the format member in the decoded AVFrame or the sample_fmt member in the AVCodecContext of the decoder.

Bit rate
The transfer rate per second (bit rate, also called bitrate). Like 705.6kbps or 705600bps, where b is a bit, ps is per second (per second), which means a capacity of 705600bit per second. Compressed audio files are often represented at double speed, for example CD quality MP3 is 128kbps/44100HZ. Note that the unit here is bit instead of byte. One byte is equal to 8 bits (bits). The bit is the smallest unit. It is generally used to describe network speed and various communication speeds. The byte is used to calculate the size. hard drive and memory.

Mbps is: Millionbit per second (millions of bits per second);
Kbps is: Kilobit per second (kilobit per second);
bps is: bit per second (bit per second), the corresponding conversion ratio is:

1Millionbit=1000Kilobit=1000000bit; 1Mbps = 1000,000bps; Again, this is the unit of speed, which refers to the number of bits transmitted per second. The unit of measure for data transmission speed K is the decimal meaning, but the K for data storage is the binary meaning. E.g:

The 1M bandwidth generally described is 1 Mbps = 1,000,000 bps = 1,000,000 / 8 / 1,000 = 125; therefore, the download speed of 1M bandwidth generally does not exceed 125KB/s
. 1000 = 12.5, so the maximum download rate of 100M bandwidth can reach 12.5MB/s
. Of course, the above is only the theoretical rate. In fact, the maximum download rate may not reach that much, and it is mainly affected by various losses, generally 100MB A broadband download rate of 10MB is not bad.

Related Audio Attribute Part 2

 

The higher the sampling, the more realistic and natural the sound will be.

 

The frequency recognition range for people is 20 HZ – 20,000 HZ. If 20,000 samples per second can be sampled, it will be enough to satisfy the needs of the human ear during playback. So 22050 The sample rate is commonly used, 44100 is already CD quality, and sampling more than 48000 is no longer meaningful to the human ear. This is similar to a 24 frames per second image from a movie.

 

Sampling bits
After sampling the audio for a sample, two steps must be performed for the sample:

1. Quantify. The quantization bits commonly used for audio quantization are:

8 bits (that is, 1 byte) can only register 256 numbers, that is, only the amplitude can be divided into 256 levels;

16 bits (ie 2 bytes) can be as small as 65536 numbers, which is already the CD standard;

32 bits (ie 4 bytes) can subdivide the amplitude into 4294967296 levels, which is really unnecessary.

The number of quantization bits is also called the number of sampling bits, bit depth, and resolution, and refers to how many levels the continuous intensity of the sound can be divided after being digitally represented. N-bit means that the intensity of the sound is divided equally into 2^N levels. 16 bits, it is level 65535. This is a very large number and people may not be able to tell the difference in sound intensity from 1/65,535. You can also say that it is the resolution of the sound card. The higher the value, the higher the resolution and the greater the ability to produce sound. The sampling multiple here is primarily addressing the strength characteristics of the signal, and the sampling rate is addressing the time (frequency) characteristics of the signal, which are two different concepts.

2. Binary encoding. That is, the result of the quantization, ie the single channel sample, is stored in a binary keyword. There are two storage methods:

Store the result of the quantization directly in the cast, that is, the two’s complement code;

The result of quantization is stored in floating point type, ie floating point encoding code.

Most PCM sample data formats use integers to store, and for some applications that require high precision, use floating point to represent PCM sample data.

frame
After the audio is quantized to a binary codeword, it must be transformed and the transformation (MDCT) is done in block units, and a block is made up of multiple (120 or 128) samples. A frame will contain one or more blocks. Common frame sizes are 960, 1024, 2048, 4096, etc. A frame records a sound unit whose duration is the product of the sample duration and the number of channels. The nb_samples in the AVFrame structure in FFmpeg represent the number of single channel audio samples in a frame.