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.
What is bit rate? Knowledge of the MP3 audio format.
bit rate
Digital audio formats are audio signals that are recorded, processed, and reproduced in digital form.
bit rate
The emergence of digital audio formats is to meet the needs of high-fidelity playback, storage and transmission. Simply put, early analog audio formats had issues with playback distortion and glitches due to media wear. Since the advent of the CD, digital format audio files have become popular, but another problem has arisen: the limitation of the storage volume, and the CD still has the phenomenon of wear. Saving to hard drive (relatively longer storage time) is not a good solution when storage media (mainly hard drives) are still expensive at the time. The rise of the Internet has created a requirement for long-distance file transmission. Under the restriction of bandwidth, the demand to reduce file size has become more intense. All this has led to the generation of lossy compressed digital audio formats from external factors!
In terms of internal factors, with the improvement of computing and coding capabilities, the progress of various acoustic psychological models has promoted the emergence of various lossy compressed digital audio formats. Some of the most commonly used audio formats in MP3 players are briefly introduced below: MP3 (CBR, VBR, ABR), WMA, WAV, ADPCM, and the emerging audio formats AAC, ASF, and OGG.
Before introducing various digital audio formats, let’s clarify one concept: bitrate.
In the field of computing, all information is digitized. Bit is the smallest unit of data in a computer, it refers to a number of 0 or 1, which is a mathematical binary number, a “0” or “1” , is a bit. For example, when we say a 2-digit number, it means that it is a two-digit binary number, and there are 4 combinations of “00”, “01”, “10” and “11”, which represent 0, 1, 2 and 3 is four numbers.
The triumphant march of the MPEG-1 Layer 3 audio recording format (colloquially called MP3) is explained by the fact that a simple and effective method of compressing an audio file was proposed, allowing to store up to 12 hours of quality music acceptable on a standard CD-ROM disc.
Simply put, the MPEG-1 Layer 3 algorithm is based on the so-called “psychoacoustic” compression method, when frequencies and loudness levels that are not perceptible to the ear are excluded from the sounds in the spectrum. The spectrum “cleaned” in this way is divided into separate blocks (frames) of the same duration and compressed according to specified requirements. When played back, the signal is formed from a sequence of decoded frames.
The compression ratio depends on the parameters of the audio stream, which must be output after decoding the file.
The main parameter that determines the sound quality and the degree of compression is the so-called (what is) bit rate, the bandwidth measured in bits per second.
The higher the number, the better the sound quality and the lower the compression ratio. Since almost all MP3 files are recorded in stereo with a 44 kHz encoding rate and 16-bit depth, the determining factors for clear sound are: the source of the recording, the codec used, and the selected bit rate. .
The word codec is formed by combining the words encoder + decoder. This is a software module that allows you to encode or decode audio or video files according to your own algorithm.
The average value of the stream of 256 kbps provides a compression ratio of approximately 6: 1, for other values, the compression ratio changes proportionally. So, with 256Kbps stream, you can record music from six regular audio CDs to one CD, and with 128Kbps stream, you can record from twelve regular music discs.
There is a never-ending controversy between amateurs and professionals about the bit rate that provides good sound quality that matches the playback quality of an audio CD.
Some consider the level of 128 Kbps sufficient, others are satisfied only with the maximum value of the transmission: 320 Kbps. In all probability, both are right: the only difference is what is recorded and under what conditions it is played.
The bit rate at which digitized audio was encoded is usually indicated on the CD cover. For example, the entire Beatles collection can be purchased on three 128Kbps discs or six 256Kbps discs.
It is clear that in the second case the purchase price will be twice as expensive, but the quality is also better.
If the music is played in a home car, the 192Kbps transmission will provide sufficient sound quality, the best that you still won’t hear due to the strange noise. For listening on a computer or standalone player (MP3 player), a 256 Kbps stream is accepted.
But if the signal is sent unchanged to an external device and output to high-quality speakers, the maximum possible stream is desirable: 320 Kbps. Based on the above considerations, a stream of 256 Kbps can be considered universal: with a good recording quality, it will provide adequate playback in most cases.
