Sample rate and bit rate of MP3 Part 3


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Sample rate and bit rate of MP3 Part 3

Sample rate and bit rate of MP3
Sample rate and bit rate of MP3

Sound is actually a type of energy wave, so it also has the characteristics of frequency and amplitude, with frequency corresponding to the time axis and amplitude corresponding to the level axis.

Sample rate and bit rate of MP3
Sample rate and bit rate of MP3

The wave is infinitely smooth, and the string can be considered to be made up of innumerable points. Since the storage space is relatively limited, in the process of digital encoding, the points of the string must be sampled. The sampling process consists of extracting the frequency value of a certain point. Obviously, the more points that are extracted in one second, the richer the frequency information that can be obtained. To restore the waveform, there must be two sampling points in one vibration. The highest frequency that can be felt is 20kHz, so to meet the auditory requirements of the human ear, at least 40k samples per second, expressed at 40kHz, and this 40kHz is the sample rate. Our common CD has a sample rate of 44.1 kHz. It is not enough to have only frequency information, we must also obtain and quantify the energy value of this frequency to represent the strength of the signal. The number of quantization levels is an integer power of 2, and the sample size of our common CD bit is 16 bits, that is, 2 to the power of 16. Sample size is harder to understand than bit rate. sampling, because it makes it seem abstract. For a simple example: suppose a wave is sampled 8 times, and the energy values ​​corresponding to the sampling points are A1-A8, but we only use 2-bit sampling size, as a result we can only keep the 4 point values ​​in A1-A8 and discard the other 4. If we use the 3bit sample size, all 8 point information is recorded. The higher the sample rate and sample size values, the closer the recorded waveform is to the original signal.

It is very easy to calculate the bit rate of a PCM audio stream, the value of the sample rate × the value of the sample size × the number of bps of the channel. A WAV file with a sample rate of 44.1 KHz, a sample size of 16 bits, and two-channel PCM encoding has a data rate of 44.1 K×16×2 = 1411.2 Kb/s. We usually say that 128K MP3, the corresponding WAV parameter, is this 1411.2Kb/s, this parameter is also called data bandwidth, it is a concept with the bandwidth in ADSL. Divide the code rate by 8 to get the data rate for this WAV, which is 176.4 KByte/s. This means storing a 1-second sample rate of 44.1 KHz, a 16-bit sample size, and a two-channel PCM-encoded audio signal, which requires 176.4 KB of space, which is approximately 10.34 M in 1 minute, which is unacceptable. For most users, especially friends who like to listen to music on the computer, to reduce disk usage, there are only 2 ways to downsample or compress. Lowering the index is not advisable, so experts have developed various compression schemes.

The minimum value of the 16-bit binary number is 0000000000000000, the maximum value is 1111111111111111, and the corresponding decimal numbers are 0 and 65535, that is, the difference between the maximum and minimum values ​​is 65535, that is, the dynamic range of the analog quantity that quantizes The difference can be 65535, which is 96.32 decibels, so quantization accuracy is only related to dynamic range and has nothing to do with frequency response. It also makes sense to set the dynamic range to 96 decibels. The painless limit sound pressure of the human ear is 90 decibels. The dynamic range of 96 decibels is sufficient for ordinary applications. Therefore, after quantization, analog waves within the 96 decibel dynamic range will not be quantized. Clipping distortion will occur.

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the highest digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone.


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Sample rate and bit rate of MP3 Part 2

Sample rate and bit rate of MP3 Part 2

Sample rate and bit rate of MP3
Sample rate and bit rate of MP3

The sample rate is similar to the number of frames of moving images.

Sample rate and bit rate of MP3
Sample rate and bit rate of MP3

For example, the sampling rate of movies is 24 Hz, the sampling rate of PAL format is 25 Hz, and the sampling rate of NTSC format is 30 Hz. When we play back the still images sampled at the same rate as the sampling frequency, we see a continuous image. In the same way, when a CD recorded at a sampling rate of 44.1 kHz is played back at the same rate, a continuous sound can be heard. Obviously, the higher the sample rate, the more coherent the sound will be heard and the picture will be seen. [Of course, the sampling rate that human auditory and visual organs can distinguish is limited, which is basically higher than sound sampled at 44.1kHZ, and most people haven’t noticed the difference. ]

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. [However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the higher digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone. ]

When you understand the above two concepts, bitrate is easy to understand. Take the phone as an example, 3000 samples per second, each sample is 7 bits, then the phone’s bit rate is 21000. And the CD is 44100 samples per second, two channels, each sample is 13 bit PCM encoded, so the CD bit rate is 44100*2*13=1146600, which means the data volume of the CD per second is about 144KB, and the capacity of a CD is 74 minutes equal to 4440 seconds , which is 639360KB=640MB.

MP3 sample rate and bit rate

MP3 sample rate and bit rate

MP3 sample rate and bit rate
MP3 sample rate and bit rate

When we listen to mp3 and watch movies, we will notice two parameters.

MP3 sample rate and bit rate
MP3 sample rate and bit rate

The most common ones are 44.1 KHz sample rate and 192 Kbps bit rate. So what is the sample rate and what is the bit rate? What is the relationship between them? Explain:

The process of converting an analog audio signal to a digital audio signal is called sampling. In a nutshell, how many data points does it take to record a 1 second long sound via waveform sampling. For example: the sound sample rate of 44.1 KHz is equivalent to spending 44,000 data points to describe the sound waveform for 1 second. In principle, the higher the sample rate, the better the sound quality; sampling frequency is generally divided into three levels: 22.05KHz, 44.1KHz and 48KHz; 22.05KHz can only achieve FM radio sound quality, and 44.1KHz is the theoretical limit of CD sound quality, 48KHz has reached DVD quality.

Sampling rate refers to the sampling frequency when converting sound (analog signal) to mp3 (digital signal), i.e. how many data points are sampled per unit of time. (The data for a sample point is 8 (or even more) bits long.)

Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (bit per second). The higher the bitrate, the more data transmitted and the better the sound quality.

It can be said that the sample rate and bit rate are like the horizontal and vertical coordinates on the coordinate axis. The sampling frequency on the abscissa represents the data points sampled per second. The bit rate on the ordinate represents the precision when quantizing analog quantities with digital quantities.

Difference between digital and analog

Difference between digital and analog

Difference Between Analog Signal and Digital Signal

The sound is analog. And sound is the vibration of the air. How is this sound vibration transmitted?

Analog VS Digital
For example, when a stone is thrown into a calm water surface, the ripples spread around it, but if
Cut in the direction of the waves and look at the cut end, the waveform is as shown in Fig.1.

Air waves spread from the point where sound is emitted even in air. Although it is invisible to the eye, it has a
similar waveform. This is the analog waveform of sound.

Therefore, although it is digital, when such a sound waveform is recorded or communicated by phone or wireless, as
shown in Fig. 2, the change in the analog waveform is electrically replaced with a series of numerical values ​​according to a certain promise. ..

When recording or communicating, if you handle it as analog, it is easy for noise to enter and the sound quality to deteriorate, but when trying
the waveform of the sound as digital = numerical data, you can eliminate that worry and
maintain a certain quality. You can do various processing while maintaining it.

(2) What is convenient when it is digital

Digital audio signals are convenient because they can be recorded and edited using a personal computer, for example.

In addition, 74 minutes of music can be recorded on a CD with a diameter of only 12 cm, and through digital compression processing
, music of the same length can be recorded on an MD with a smaller diameter.

Since digital signals can be compressed in this way, it is also convenient for storing large amounts of information.
Not only sound, but also video signals with a higher amount of information can be recorded and communicated at high speed by using compression technology.

Especially in communication, a two-way digital multiplex communication can be realized communicating multiple pieces of information with a single wire.
In addition to electrical signals, laser light can also be used for optical communication, making communication possible at extremely high speeds.

(3) What is the sampling frequency?

Digital signals are processed at predetermined fixed time intervals.
The sample rate (sample rate) indicates how many times a second is processed and is expressed as Fs or fs.

The sampling frequency unit is Hz (Hertz), and the
44.1 kHz (kilohertz) sample rate means 44,100 pieces of data are processed per second.
(K represents 1000 times)

AD conversion converts a continuous analog signal into a digital signal,
measures the size of the signal at each moment determined by the sampling frequency (sampling) and converts
the result in a binary number (quantization).

On the other hand, DA conversion converts a digital signal into an analog signal,
It reads the digital signal in the sample rate time interval and connects it smoothly.

Since digital signals can be reproduced up to half the sampling frequency, how much
The higher the sample rate, the higher the playable frequency and the better the sound quality.
In familiar areas, 44.1 kHz is used for CD, and 48 kHz is used for DAT and B modes of satellite transmission.

In addition, recent professional equipment uses high sampling frequencies (high sampling), such as 88.2 kHz and 96 kHz,
and are designed to faithfully reproduce even higher frequency sounds to improve sound quality.

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality PART 2

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality PART 2

Audio bit rate

What is the bit depth that determines the sound quality?
What is the bit depth that determines the quality of the sound?
The bit rate is calculated by multiplying the two factors of bit depth and sample rate. Bit depth represents the amount of data per divided sample and is an element that affects the quality and expressiveness of the sound.

The sample rate is the finely divided horizontal axis and the bit depth is the one that overlaps the vertical axis. By providing a large amount of data (bit depth) to each sample, it is possible to create finer and more accurate audio data.

Bit depth is the precision of each image in animation production. The higher the bit depth, the more expressive the sound and effects will be, and the higher the sound quality will feel.

What is the sample rate that determines the smoothness?
Of the bit rates that determine audio quality, the sample rate is an element that represents the number of data divisions per second on the horizontal axis. It shows how many tens of thousands of data are divided per second and the higher the number of divisions, the higher the sound reproducibility and sound quality.

It is even easier to understand if you consider it a mechanism similar to animation production. The more images you use per second, the smoother your character will move, and the higher the sample rate (the number of data divisions), the smoother your sound will sound. Also, the amount of data increases according to the size of the sample rate.

In conclusion
To improve sound quality, it is important to increase the audio bit rate. The audio bit rate is determined by two factors: the bit depth, which determines the expressiveness of the sound, and the sampling rate, which determines the smoothness of the sound.

However, keep in mind that the higher the audio bit rate, the more beautiful the audio will be and will also be affected by the original sound source. If you want to create high-quality data from the moment of recording, why not ask a production company that has high-quality recording equipment?

I think there are many people who are concerned about the balance between capacity and sound quality when it comes to the audio system mounted on a computer. Professional engineers and mixers will come up with the best balance, so don’t hesitate to contact us on these points.

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality

Digital Audio Basics

Bit rate is one of the factors that determine the quality of a video job.

BitRate

Among them, the one that determines the audio finish is the audio bit rate. Understanding the bit rate will allow you to control the sound quality and create better quality audio.

So on this occasion, I will explain bit depth and sample rate which are indispensable when talking about sound quality.

What is a bit rate?
What is a bit rate?
■ Bit rate represents high quality

Bit rate is a numerical value that represents the amount of information in the data, and the height of the bit rate is proportional to the quality of the data.

By looking at the bit rate, you can see how much data is packed in one second. Generally, the higher the bit rate, the higher the sound quality, and the sound and video are more realistic.

■ Audio bit rate and video bit rate

In the case of video, the bit rate is divided into two types, video and audio, and the overall quality of the video is determined by the height of each bit rate.

The “video data rate” shown in the video properties is the video bit rate and the bit rate assigned to “audio” is the audio bit rate. The “total bit rate” is the total of the two types of bit rates, video and audio, and is the bit rate of all video.

However, high bit rate audio is not always good. This is because if the quality of the original audio data or the equipment used for recording is poor, poor quality audio will be played. For example, if the original voice contains noise, the noise part will be reproduced realistically and the voice will be difficult to hear. Similarly, if the bit rate of the video is low, the movement will be choppy and unnatural, or the video will be uneven.

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 7

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 7

DSD vs. PCM

MQA encoding processing can be performed on 44.1 kHz to 768 kHz linear PCM sound sources and can be stored in existing file formats (container formats) such as FLAC, ALAC, and WAV. If you use a compatible device equipped with a dedicated decoder, you can “open origami” and demonstrate the original quality. It can be played with an unsupported device, but in that case, “Origami cannot be opened”, so normal PCM playback will be performed that does not include the original MQA information.

MQA is available for download at e-onkyo music. It has the advantage that it can be used with a low communication volume even with streaming type distribution, and abroad, the “TIDAL” music distribution is developing a high quality 96 kHz / 24 bit MQA streaming distribution ” TIDAL Masters “.

High Resolution Portable Player “DP-X1” Quickly Achieved MQA Compatibility
Current Status of Supported High-Resolution Formats and Music Distribution Services (as of July 2017)
WAV FLAC ALAC DSD MQA others
e-onkyo music 〇 〇 — 〇 〇 Dolby TrueHD
blackberry — 〇 — 〇 — —
OTOTOY 〇 〇 〇 〇 — —
Recochoku — 〇 — — — —
My sound — 〇 — — — —
slots — 〇 — 〇 — —
Is FLAC a powerful option right now?
Generally speaking, the high resolution format has been “PCM or DSD”, and for PCM, the choice has been “lossless compression” or “uncompressed”. There may be circumstances on the side of the playback device and sound preferences, but in terms of the balance between data compression effect (file capacity) and sound quality, “lossless” is a reasonable existence.

Among them, FLAC, which is easy to use on smartphones, will remain high resolution as source code is published and royalties are not incurred, it is already compatible with many compatible high resolution audio devices, and will be supported in the system level in the next iOS (iOS 11), will be the centerpiece of the format.

Looking around, there are signs of change, such as the emergence of a new face, MQA, and the spread of streaming services, but the audio format must have attractive content. Above all, I would like to hope that the record company / distribution service company is actively working in high resolution.

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 6

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 6

PCM DSD

Review the basics of “high resolution”. What is the difference between DSD, FLAC, MQA, etc.?

There are several formats, even if it says “high resolution”. If the format changes, the amount of sound information and thus the sound quality will change, the file size will also change, and whether the playback device / software will support it or not, it will also change, so choose a format is important. We will explain the main formats incorporating common technologies and unique pieces.

There are several high-resolution formats, but …
What is the sampling frequency?
Most digital sound sources are “linear PCM”. This is data obtained by digitizing (sampling) the sound waveform (analog signal) in a canned cycle, and that cycle is called the “sample rate.” If sampling is done every 1/44100 of a second it will be “44.1 kHz”, if it is 1/96000 of a second it will be “96 kHz”, if it is 1/192000 of a second it will be “192kHz”. This means that the implementation cycle of is shorter and the amount of information is greater. In other words, if you look at this number, you can see “how finely the sound was measured with respect to time.”

What is the number of quantization bits?
Value that indicates the number of steps in which the amplitude of a signal is expressed when an analog signal is converted into a signal.

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 4

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 4

DSD vs PCM

The number of audio devices that support ALAC has increased in recent years, and some high-resolution distribution sites handle it as well, but FLAC is compatible with more software / hardware, and there is a significant advantage in sound quality over FLAC. . it will eventually restore to the same linear PCM data), so there is no reason to choose it aggressively except for users who are unified with Apple products.

DSD vs PCM

“ALAC”, which is compatible with Apple software / hardware, can also be used for high resolution playback.
Wav
Classification:
PCM compression: no
extension: .wav

It is widely used as a linear PCM container format (it can store various types of data, and the codec can be selected for musical use), and it has been widely adopted by high-resolution distribution sites. Since linear PCM is not compressed, the file size is large, but since no decoding (decoding) processing is required, there are quite a few people who dare to choose WAV from a sound quality point of view.

It is playable on most digital audio software / devices, but tags / metadata such as artist names and album images cannot be expected to display a lot. It is possible to store data, but the format is not officially defined, so it can be displayed in one environment but not another. FLAC and ALAC are more convenient than WAV when using an application like Roon that searches for information, such as related artists, by referencing metadata.

Linear PCM with a sampling frequency of 352.8 kHz (8 times 44.1 kHz) or 384 kHz (8 times 48 kHz) and a quantization bit rate of 24 bits or more is called “DXD” (Digital eXtreme Definition). Originally intended for SACD production (because DSD is not suitable for editing), it is now used as a distribution format as well.

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 3

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 3

DSD vs PCM

PCM or DSD

PCM to DSD

So far, we have looked at high resolution definition on the premise of PCM, but “DSD (Direct Stream Digial)” is also recognized as a type of high resolution sound source. Regarding DSD, which is not clarified in the JEITA definition, the Japan Audio Association definition recognizes products that support 2.8MHz / 5.6MHz DSD playback as compatible high-resolution devices.

Sound reproduction methods are almost completely different between PCM and DSD. DSD always records 1 bit width (1 bit is represented by “0” or “1”) subdividing only in the direction of the time axis.

In the case of PCM, the resolution can be increased and the dynamic range can be expanded by increasing the number of bits, but on the other hand, quantization noise occurs (distortion caused by an error in rounding up / down the amplitude value real). . DSD is set to 1 bit to create a state in which no quantization noise is produced and the amount of information is ensured by significantly increasing the sample rate.

DSD has a 1-bit amplitude of “0 or 1” and records sound information with little information in the direction of the time axis.
(The picture is taken from the product information page of Sony HDD audio player “HAP-Z1ES”).
Main high-resolution formats
Data as uncompressed or compressed, lossless or lossy as compressed, PCM or DSD is primarily a high resolution format, and depends on the playback device / software to be used in addition to the taste of the sound. Let’s explain the characteristics of each format for the formats handled by the so-called music distribution sites.

FLAC
Classification:
PCM compression: Yes (lossless)
Extension: .flac

“FLAC” is a representative of high resolution audio sources. It is an open source lossless codec (software whose source code is open to the public and can be freely improved / redistributed), and one of the reasons for its widespread use is that it does not generate royalties. It is thought that the sense of security that allows us to confirm that we are there also helped spread the word.

It is often created (encoded) using uncompressed linear PCM as the source, and can be compressed to a file size of around 60% compared to the source. There are many supported hardware / software. At the moment, iOS is late (some apps have already supported it), but it almost certainly is if it is a device that claims to support high resolution. If you are wondering which format to choose, choose FLAC and you are right.

Most of the hardware / software and distribution sites that claim to support high resolution support FLAC (image is blackberry screen).
ALAC (Apple Lossless Audio Codec)
Classification:
PCM compression: Yes (lossless)
Extensions: .m4a, .mov, .alac

Introduced when Apple realized lossless audio streaming using a compact router (AirTunes). Initially it was an original technology, but then it was made open source and is now popular as a lossless codec alongside FLAC.

It has been well supported by Apple products since its introduction, and the files can be generated (encoded) using iTunes. However, Apple products are designed not to consider high resolution playback, so even if you play high resolution ALAC files on iOS devices, the information will be cut (reduced) to CD quality.