Generally, the files we read will know the following parameters:
MP3 bit rate
Sampling frequency: the number of sampling points that pass one second; by
For example, the most common sample rate of 44.1k means that there are 44,100 sampled data points in one second;
Bit depth: Bit depth is also called sampling bit depth, and the bit depth of the audio determines the dynamic range.
Our common 16-bit (16-bit) can record a dynamic range of about 96 decibels. Well, roughly you can know that each bit can record about 6 decibels of sound. Similarly, the 20-bit recordable dynamic range is about 120 dB; 24 bits is about 144 dB.
If we define 0dB as the maximum value, then the amplitude of the sound is calculated by extending down, then the dynamic range of CD audio is “-96dB ~ 0dB”. And so on, the dynamic range of 24-bit HD-Audio is “-144dB~0dB”. It can be seen that at higher bit depths, a greater dynamic range is available and lower levels of detail can be recorded.
Number of channels: usually single channel or two channels
Through the above parameters, the bit rate of the audio can be calculated
\color{blue}{code rate = sample rate (44.1k) * bit depth (16) * number of channels (2) = 1411.2kbps}
Audio bitrate, also known as bitrate: Refers to the amount of data that can pass through an audio stream per second. For example, 128 kbps, where ps (per second) is per second and kb is thousand bits, then 128 kbps means that the amount of data that can be transmitted in one second is 128 thousand bits. For files of the same format, the higher the bitrate, the better the sound quality. But for audio files of different formats, the same bitrate does not mean the same sound quality.
CD quality sound, usually two channels, compressed at 128kbps MP3, one second of data is converted to 128kbits. Some details are lost, which saves a lot of disk space or bandwidth.
After knowing the audio code rate, you can get the size of the whole audio file = duration (300s) * code rate (1411.2)/1024/8=51.67M
The second thing you want to read is that you should never convert to a lower bitrate. bitrate to a higher bitrate stream and I hope it sounds better. You cannot gain quality by increasing the bit rate. This is absolutely correct. If you try to convert the bitrate, you will actually lower the quality of the MP3 file.
mp3 bit rate
If you want an MP3 with a higher bitrate than it currently has, you’ll need to go back to the source (CD, etc.) and extract the audio at full quality. You can then convert that file to a higher bitrate MP3 file.
The last thing you need to know is that converting between lossy formats is not recommended as you will still lose quality. However, it is possible to convert between formats losslessly while preserving quality.
On the other hand, many tests done even with recording engineers or professional musicians have shown that less than 1% of the population can distinguish between an mp3 with good bitrate (196 for example) and a samplerate of 44100 or more.
There is a sector of puritanism that defends tooth and nail the use of losless formats, because in theory they seem to be better, but the limitations and peculiarities of human hatred mean that a loosy file with a high bitrate and a high samplerate is enough for the human ear.
Mp4Gain is the most professional and polished program to help you in all these matters and you can achieve the highest sound quality with any format.
This can be useful if, for example, you need to reduce the size of an MP3 file. A 320kbps MP3 file, the highest bit rate allowed for MP3 files, can be reduced to 192kbps to significantly reduce the size of the MP3 file.
There is a drop in quality, but the difference is negligible for most listeners using standard speakers or headphones. If you’re an audiophile, you’ll probably never use the MP3 format, except for expensive audio equipment.
You are most likely using a lossless format such as PCM, WAV, AIFF, FLAC, ALAC, or APE audio, compressed or uncompressed. Uncompressed PCM audio files are about 10 times larger than CD-quality MP3 files.
The MP3 format is a lossy format, which means sacrificing audio quality to keep the file size relatively small. Almost every site will tell you not to convert lossless audio files to MP3 unless you might lose some audio quality.
<Almost all the time. The only time it might make sense is if you have a high bitrate audio file in a low quality format like WAV. For example, it might make sense to convert a 96kbps WAV file to MP3, but only if you choose a bitrate of 192kbps or higher. A higher bitrate in an MP3 file will allow it to maintain the same quality as a WAV file, even if it has a lower bitrate.
Audio Bitrate: What is the Bitrate of Music Part 3
Audio Bitrate
How should bitrate be interpreted in an audio file?
Audio Bitrate
That is, the higher the bitrate, the better the audio and video quality, but the larger the encoded file, the lower the bitrate, the situation is reversed. For example: encode audio and video at 500 Kbps. Where bps is bit 1K=1010=1024 b is bit (bit) s is second (second) p is per (per) So encoding with 500kbps means that audio data and video encoded need to use 500K bits per second to Indicates that the bit rate is used to represent the code rate of information transmitted in the baseband transmission system. The bit rate Rb refers to the number of binary bits transmitted per unit of time, and the unit is b/s. For example, the transmission code rate of a computer serial port is up to 115200b/s. The symbol rate or baud rate Rs refers to the number of modulation symbols transmitted per unit of time, that is, the information transmission rate of ternary and more than ternary digital code streams, and the unit is baud/s In M-ary modulation, bit The relationship between the rate Rb and the baud rate Rs is: Rb=Rslog2M In a word, the bit rate indicates the amount of data transmitted per second.
What is the proper bit rate for mp3 music files?
Bitrate is simply the number of bits per second transmitted by the media file and the unit is Kbp/s. The default bit rate of compressed MP3 files is 128 Kbp/s and the sound quality is similar to that of a CD. However, the bit rate of MP3 downloaded from the Internet is usually 192Kbp/s, the sound quality of 192Kbp/s is better than that of 128Kbp/s, and the space occupied by the file is not too large, so which is widely used. . However, if you want to get better sound quality, the bit rate should be higher than 320Kbp/s, and the sound quality can be really comparable to CD quality. However, the price you pay for doing this is that one song will take up about 10M of hard drive space. Generally speaking, the higher the bitrate, the better the sound quality, but it will take up more disk space.
What does “bitrate” mean in audio?
For example, 128kbps MP3 means that the amount of information per second is 128kb converted into bytes is 16KB 320kbps MP3, so the quality does not improve, on the contrary, it may decrease. There are also lossy compression formats such as MP3, wma, ogg .. These are lossy compression formats, which means that there will be loss in the compression process, but each compression algorithm is different, and the quality of the compressed sound is different. For example, when comparing the same bit rate, the sound quality of different formats is different. Hope it can help you.
Audio Bitrate: What is the Bitrate of Music Part 2
Audio Bitrate
What do sample rate and bit rate mean in a song?
Audio Bitrate
Bit rate Bit rate 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 bit rate value is compared with the actual audio: 16 KBPS = phone sound quality 24 KBPS = increase phone sound quality, shortwave transmission, longwave transmission, European standard medium wave transmission 40 KBPS = American standard medium wave transmission 56 KBPS = voice 64 KBPS = voice boost (best bit rate for mobile phone ringtones) Setting value, the best mobile phone mono MP3 player setting value) 112 KBPS = FM radio stereo FM 128 KBPS = tape (best setting value of mobile phone stereo MP3 player, best setting value of low-end MP3 player) 160KBPS= HIFI HIFI (best setting for MP3 players mid- to high-end) 192KBPS=CD (best setting for high-end MP3 players) 256KBPS=Studio Music Studio (for music enthusiasts) The sample rate is when the analog signal is ca is converted to a digital signal The sampling rate is related to the quality of the sound. The higher the sample rate, the better the high-frequency restoration of the sound file. The following is the different quality corresponding to different sampling rates 1.11,025Hz The sound of this sampling rate is similar to the sound quality of AM radio 2.22,050Hz The sound of this sampling rate is similar to the sound quality of FM radio, but less than 3.32 000 Hz This sample The sound of this frequency is higher than the sound quality of FM broadcast 4.44 100 Hz The sound of this sampling frequency reaches the sound quality of the CD audio 5.48000 Hz Sound at this sampling rate reaches DAT audio sound quality 6.96000 Hz Sound at this sampling rate reaches DAT audio sound quality The higher the sampling rate sound quality of the DVD audio, the more disk space the final sound file will take up. Usually we can choose 44.1KHZ sampling rate.
Bit rate refers to the amount of binary data per unit of time after converting an analog sound signal to a digital sound signal.
Audio Bitrate
The higher the bitrate, the better the sound quality (under the same encoding format, different formats cannot be compared). audio bitrate. Bit rate is a benchmark of digital music compression efficiency. Bit rate indicates the rate of the number of bits bps (bit per second, bits per second) transmitted in a 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 the CD is 1411.2 kbps (i.e. to burn 1 second of CD music, 1411.2 × 1000 data bits are required), the high bit rate of the music file means that the data must be processed in a unit of time (1 second) The amount (BIT) is large, that is, the sound quality of the music file is good. However, when the BITRATE is high, the file size increases, which will occupy a large amount of memory capacity. ranges in this sense, most of them are 32-256 Kbps. Of course, the wider the index, the better, but 320 Kbps is the highest level for the moment.
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.
A wav is 44100 Hz 16-bit stereo or 22050 Hz 8-bit mono, what does that mean? stereo/mono refers to dual/mono.
Audio Intro
For monophonic sound files, the sample data is an eight-bit short integer (short int 00H-FFH); for two-channel stereo sound files, each sample data is a 16-bit integer (int) and the upper eight bits (left channel) and lower eight bits (right channel) represent the two channels, respectively.
Sound is a mechanical wave, produced by the vibration of an object, and requires a medium to propagate. So, in essence, a sound is a waveform on an axis over time.
Sound has three elements: pitch, volume, and timbre:
Pitch is determined by the frequency of the sound wave, the higher the frequency, the higher the pitch.
The volume is determined by the amplitude of the sound wave, the larger the amplitude, the louder the sound.
The timbre is determined by the “shape” of the waveform (sounds like square, triangle, and sawtooth are called impulse waves and sound individual).
An audio file is a file obtained by converting an analog signal to a digital signal. In general, there are five important parameters: encoding method, number of channels, sampling rate, bit depth, and bit rate.
Encoding: how this format organizes binary data and how it is compressed.
Number of channels: mono, dual or 5.1 channels, etc.
Sampling rate: The number of samples per second.
Bit Depth: The number of binary bits used to store the y value of the sample point.
Bitrate – The desired number of bits per second for the file.
We know that there is no compression in the WAV format, so its encoding method is to directly write all the sampled points to the file in order.
WAV file size (B) = number of channels * sample rate (Hz) * bit depth (bit) / 8 + file header size (B, it’s 44B)
Implementation
When you open an mp3 or wav file with a text editor, you see numbers like this:
44100Hz represents the sample rate of the signal. The so-called sampling consists of obtaining the value y of the sound wave at the current moment every unit of time. Sampling is the process of discretizing continuous data (converting an analog signal to a digital signal).
image source
The sampling method mentioned above is called PCM (Pulse Code Modulation). According to the Nyquist-Shannon sampling law, the sampling rate must be at least twice the highest target frequency. The hearing range of the human ear is about 20Hz-20,000Hz (if you’re curious how loud you can hear, you can click here to test your ears), although recording software often has a 48,000 option Hz, but we can safely conclude: 44100Hz can meet almost all our needs, higher is just a waste of your memory and CPU. More than 48,000 samples are meaningless to the human ear, which is similar to 24 frames per second on a movie. 44100Hz happens to be the standard sample rate for almost all music released. In fact, for vocals and many instruments, high-frequency sounds are noise, so high sample rates can sometimes worsen sound quality (which is why we need to adjust the equalizer).
320 kbps represents your bitrate/bitrate, which is shorthand for kilobits per second, which represents the size of the data used to describe sound. In CD (uncompressed audio file), the bit rate is 1411.2kbps, and the mp3 sound quality to achieve CD quality should be higher than 128kbps/44100Hz (128kbps can be said to be the most common bit rate). Generally, a higher number means better quality. The quality depends on many factors (such as the encoding algorithm). Many times we don’t need too high bitrate: our device can play mp3 and CD without difference (sound/sound card is normal).
What is bit rate? Knowledge of the MP3 audio format. Part 2
bit rate
Bitrate is a benchmark indicator of the efficiency of digital music compression.
bit rate
The bit rate represents the number of bits bps (bit per second, bits per second) transmitted per unit of time (1 second). We usually use kbps (in simple terms, it is per second) clock 1000 bits) as the unit. The bit rate of digital music on CD is 1411.2 kbps (ie recording 1 second of CD music requires 1411.2 × 1024 bits of data). The higher the bit rate of the music file, the more data (Bit) must be processed in a unit of time (1 second), and the better the sound quality of the music file. However, when the bit rate is high, the file size increases, which will occupy a large amount of storage capacity. 8 to 320 kbps.
1. WMA (Windows Media Audio, Windows Media Audio)
As a Microsoft media compression method, it is a part of the technology that only compresses audio data in Windows Media Technologies. The sound quality is similar to MP3 and can be compressed with half the technology of MP3. It has the copyrighted Windows Media Rights Manager and can be played by installing it in WMP (Windows Media Player, Windows Media Player). Due to the strong influence of Microsoft and Windows, as well as major copyright reasons, the major American record companies, EMI and BMG, officially confirmed that they use the WMA method developed and produced by Microsoft. It is believed that this advanced method will become even more popular in the future.
2. MP3 (CBR, VBR, ABR)
MP3 is currently the most widely used and widely used lossy compressed digital audio format. It has been explained above and will not be repeated here.
CBR (constant bit rate)
CBR is the oldest and simplest MP3 encoding (compression) method. When this method is used for encoding, the bit rate of the entire file is the same, in other words, the bit rate used by the MP3 file per second is the same. Although the music file has sections of varying complexity, the encoder always keeps the bitrate constant, unless you use the highest sound quality; otherwise the sound quality of the different sections of the MP3 file will vary. The more complex the passage, the worse the sound quality. Its biggest advantage is that the file size is fixed, which is convenient for calculating storage space.
VBR (Variable Bit Rate, Variable Bit Rate)
VBR is a variable encoding rate MP3 compression method. Its principle is to encode the complex part of a song with a high bit rate and the simple part with a low bit rate. Through this dynamic adjustment of the encoding rate, the sound quality can be improved. additionally obtained and the size of the file. Its main advantage is that the entire song can approximately meet our sound quality requirements, but the disadvantage is that the size of the compressed file cannot be estimated during encoding.
Most MP3 players released now support VBR, but although some machines can play songs in VBR format, they can’t display the playing time correctly. Nowadays, a lot of high-quality MP3 music is encoded in VBR.
