Tag: sample rate and bitrate for mp3

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“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 2

Thus, there is no strict unified standard for determining “whether or not the sound source is high resolution”, and whether or not it is high resolution is not determined only by the older formats such as FLAC and WAV.

Since the amount of data is required to be greater than or equal to that of an audio CD and, in particular, that the number of quantization bits must be 24 bits or more, a sound source with a “sample rate of 44.1 kHz or more and a number of quantization bits of 24 bits or more “is high resolution. It seems good to think. Of course, it should be noted that the sound quality is not always high quality sound because it meets the high resolution conditions because it is not an audible standard.

Relationship between audio format and codec (for PCM)
Uncompressed or lossless compressed
Linear PCM is the sampled data itself, and theoretically does not deteriorate unless conversion processing is performed. However, since efficient data storage is not considered, the file size increases as the sample rate and the number of quantization bits increase. This is why multi-MB songs with compressed sound sources like MP3 are converted to tens of MB and hundreds of MB in high resolution.

That’s where the “lossless” codec is used. The purpose is to make the data compact by signing the linear PCM (processing to organize the data arrangement / storage pattern according to a certain rule). During playback, it is converted to the original linear PCM in real time, and in theory there is no deterioration in sound quality. “FLAC” and “ALAC” are typical examples, and linear PCM can be reduced to a data size of approximately 60%. The characteristic is that the sound quality does not deteriorate theoretically because the original information is left completely when encoded.

On the other hand, the “lossy compression” codec can achieve a high compression rate so that the data size is 10% of the original, but it is said to sound out of the audible band (human ears cannot perceive) during encoding. High frequency band) is removed. Sounds processed by lossy compression codecs are not classified as high resolution because the presence of sounds outside the audible band is believed to have a great influence on the expression of the realistic sound field and depth, which is You can tell that it is the advantage of high resolution. The way of thinking is dominant. In fact, even the JEITA and Japan Audio Association definition mentioned above does not include high resolution lossy compressed sound sources.

Compressibility and sound quality trends of the main digital audio formats
method Typical format Sound quality Compression rate
(assuming PCM is 100%)
Uncompressed WAV ◎ 100%
AIFF
Lossless compression FLAC ◎ Approximately 60-70%
A THE C
Lossy compression MQA About 15-25%
MP3 △ ～ ○ About 10 to 20%
CAA
Ogg Vorbis
WMA

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“High resolution” basics. 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 indicating the number of steps in which the amplitude of a signal is expressed when an analog signal is converted to a digital signal (AD conversion) for linear PCM generation. The higher the value, the finer the amplitude of the sound can be captured, and the closer the waveform is to the original sound (analog signal), the more accurately the sound can be reproduced with high resolution.

The large number of quantization bits is directly related to the resolution of the data. For example, when the number of quantization bits is 1 (1 bit), the width of the expression is 2 steps of “0 or 1”, but in 2 bits, there are 4 steps of “00” and “01”, ” 10 “and” 11 “. They can be expressed. Similarly, 4-bit has 16 steps, 8-bit has 256 steps, 16-bit has 65,536 steps, and 24-bit has 16,777,216 steps, allowing for detailed expression.

The “dynamic range” of the maximum / minimum sound ratio that can be handled with the sound waveform data is determined by the number of quantization bits. The dynamic range of human hearing is about 120 dB, but when the number of quantization bits is 16 bits, it reaches 96 dB, but when it is 24 bits, it reaches 144 dB, and when it is 32 bits, reaches 192 dB. (increases by 6 dB for each additional bit) If it is a high resolution sound source, it can handle tiny sounds to powerful sounds with a margin.

Original waveform data. The vertical axis is the sample rate and the horizontal axis is the number of quantization bits.

The higher the sampling frequency (finer the horizontal axis) and the greater the number of quantization bits (finer the vertical axis), the richer in information the “high resolution” sound becomes.
Definition of “high resolution”
High-resolution quality audio sources (hereinafter referred to as high-resolution sound sources) excluding DSD are distinguished by the sample rate and number of quantization bits described above. The term “more than CD specifications” is often used, which means that it is based on the sampling frequency (44.1 kHz) and the number of quantization bits (16 bits) of the CD.

As defined by the Japan Electronics and Information Technology Industries Association (JEITA), high-resolution audio must “either the sampling rate or the number of quantization bits exceed CD specifications”, and sources high resolution sound will follow this.

On the other hand, the Japan Audio Association also defines high resolution, and the recommended high resolution logo is given to audio equipment that guarantees its playability. This standard is divided into analog and digital systems, and the file format is also mentioned, such as setting the standard for high-resolution sound sources to be “96 kHz / 24-bit FLAC and WAV compliant.”

High resolution sound source

It can be said that “USB-DAC” is a secret weapon for playing music files on a computer with high sound quality.

Just add “USB-DAC” to the audio you use all the time, and you can enjoy very high quality sound! Therefore, this time, Sarah visited Onkyo Co., Ltd., which developed the most advanced “USB-DAC” that supports high-quality sound sources called “high resolution”, which has been released more and more in the last years. We also visited the audition room and asked Mr. Kurosawa, director of the high-quality music distribution site “e-onkyo music”, to teach us how to enjoy high-quality sound!
“Sample rate” and “bit rate”

Sara-chan: Hello! Wow, it’s a nice listening room! I’m excited!

Kurosawa: Hi Sarah! Today, I’m thinking about getting you to experience high-quality sound in a number of ways.

Sara-chan: Thank you! To enjoy music on your PC with high sound quality, “USB-DAC” is definitely important! But there are many “USB-DACs” and I don’t know what to choose.

Mr. Kurosawa: When choosing “USB-DAC”, it is a good idea to check the “sample rate” and the “bit rate”.

Sara-chan: Sa, Samp … Call frequency ?? What the hell is that ??

Kurosawa-san: “USB-DAC” is a device that converts sound from digital signals to analog signals, right? The “sample rate” indicates the number of digital samples of the audio signal acquired per second during the conversion. The “sample rate” determines the frequency range of the audio file. The higher this number, the closer the digital waveform will be to the original analog waveform and the softer the sound will be. On the other hand, “bit rate” indicates the amount of information per second. They are expressed in units of “Hz” and “bit”, respectively. By the way, do you know what the CD standard is?

Sara-chan: Well I’m sure it’s “44.1 kHz / 16 bit”!

Mr. Kurosawa: That’s right! It is said that the sound in the ultra high range above 20 kHz cannot be heard by the human ear, so the CD cuts off the inaudible sound. But even if you think you can’t hear it, you actually feel the vibrations in the air and it affects the sound of the frequencies you hear.
Since the high resolution sound source also records that part, it can be said that it is closer to a more realistic sound. First, at the music-making stage, work is often done at 96 kHz / 24-bit, which is why high-resolution sound sources are sometimes referred to as “studio master quality.” Recently, even more informational sound sources such as “192kHz / 24bit” have appeared. There are “44.1 kHz / 16 bit” and “96 kHz / 24 bit” sound sources here, so let’s compare them using ONKYO’s “DAC-1000 (S)”!

Sara-chan: Wow! You can feel the difference more than you imagined! It feels like a live performance is taking place right in front of you! I feel like the sound is expansive and I feel like I’m surrounded by sound! Anyway, it seems like I’ve never heard it on audio before! Impressed!

Mr. Kurosawa: Fufufu. Can you see the difference! However, even if you have a high resolution sound source such as “96 kHz / 24 bit”, there is no point in using a “USB-DAC” that does not support “96 kHz / 24 bit”. Therefore, when choosing “USB-DAC”, it is important to check the “sample rate” and “bit rate” to see if it is compatible with the sound quality you want to hear. By the way, recently there is even a “USB-DAC” that has a function to change the frequency called “upsampling”.

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry) part 2

2) When recording a 48 kHz music video to match the music played at 44.1 kHz on the site. In this case, it can be very difficult to match performance lips to post-production due to the different sample rates of the sound being played. It’s called sink drift.

3) Another point It seems that this is a problem that occurs at the time of recording, but there is a problem that the sound changes gradually when the sound recorded separately using a cheap recorder is synchronized with the sound recorded in the reference of the video. it seems that there are moments. In this case, it seems that it is necessary to manually and quickly advance the video a bit and match it with the audio file, or extract some frames at the important points of the audio and synchronize it. It seems that this has nothing to do with the sample rate, so I will describe it so as not to cause misunderstandings.

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry)

In video production, record the frame rate for shooting and the sample rate for recording. Remember this is one of the basics for shooting and recording.

First, about the difference in sampling frequency. Generally speaking

44,100Hz (44.1kHz) is the standard in the music industry

48,000Hz (48kHz) is the sound standard in the video industry

The difference between the two sample rates is just that. I spoke of the sample rate as
the frame rate in video in another article, “Sound Principles Required for Video Production,” Sample Rate and Bit Depth. ”
In other words, the higher the sample rate in Hz, the softer the sound will be.

There are several theories about the historical background of 44,100Hz.
I would like to introduce you to one of the most logical.

First, when sampling sound, you need a sample rate that is at least twice the highest frequency you are recording. This is the sample rate required to obtain a minimum of the waveform. This is because it is not possible to record a sound that has the character of a wave if there is only one place to take a sample. Most people say that the audible range is 50 Hz to 16,000 Hz. Double is 32 kHz, but it seems that the harmonic components that make up the tone need to be recorded in order to record the voice correctly. Only when this is taken into account does it appear that up to 44,100Hz is required. Click here for more details.

Sound Processing “I want to hear my voice clearly” (link outside Vook’s site)

What happens when the sample rate is low?
When digitizing analog information, if the sample rate is not high, the high-frequency information will be hidden in the low-frequency information.
Then the high-frequency sound will be recorded as low-frequency sound. Specifically, see the following illustration.
This is called aliasing.

See also: Wikipedia https://en.wikipedia.org/wiki/Aliasing#/media/File:AliasingSines.svg

In any case, by definition, 48,000 Hz has better sound quality than 44,100 Hz. The video industry has introduced 48,000 Hz.

One problem that sometimes occurs is that “I was recording a video at 48 kHz and the separately recorded microphone was set to 44.1 kHz.” At first I thought that different sample rates would be a big deal, but it doesn’t really seem to be the case.

Audio recorded at a small sample rate just has a small number of samples per second, but since there is almost no difference between 44.1 kHz and 48 kHz, I think the difference is barely noticeable when listening to the sound regularly. At 96 kHz, the sound quality is even higher, but the number of samples is so large that ordinary people cannot hear it at all.

In some cases, the sample rate is really important.

1) By writing the audio actually recorded with a different sample number as a video file. This is because the sample rate must be converted to a video sample rate that is different from the conventional 44.1 kHz and 96 kHz sample rates, that is, 48 ​​kHz. Software that specializes in video editing seems to have sound distortion at this point.

About the frequencies used for audio (sample rate, PCM, DSD, etc.)

On this occasion, I would like to explain the frequencies used in digital audio and their meanings.

Recently, the high-resolution sound source has increased, such as 192KHz Toka, 11.2MHz, as the frequency has been written or will, what frequency?

I would like to explain the frequency used for said audio taking as an example the Combo384 installed in the USB-DAC used in LV2.0.

1. 1. What is the sampling frequency?

Music distribution is becoming mainstream these days, but audio was first digitized on CDs, which are still on the market.

You often hear that the sample rate of a CD is 44.1 KHz. Since digital signals are basically 0 or 1, to reproduce up to the 20 KHz limit that can be heard by the human ear, a resolution of twice that frequency is required. Furthermore, the frequency was decided to be 44.1 KHz taking into account the digital signal processing margin. Since the music signal is a set of sine waves, it is 44.1 KHz that can be shaken at the maximum frequency of 20 KHz.

2. What are 16 bits and 24 bits?

As you often hear, CDs are sometimes described as 44.1KHz / 16bit. This 16 bit is the volume of the sound. Since 16 bits can express the size of 2 raised to 16, there are 65536 different sizes.

If this is converted to dB at 20LOG (65536), it will be approximately 96dB. The dynamic range of a CD (the difference between low and loud sounds) is 96 dB.

For DVD and Hi-Res, it can be 24-bit, but in this case, it’s 144 dB in 16.77 million steps.

3. 3. PCM format

So what is the actual signal? In the case of the PCM format, the standard called I2S, which can support up to 32 bits in sample rate, is common. In the case of a CD, being stereo, the data has a frequency of 44.1 KHz with 32 bits of 2 channels (L, R) alternately (although in reality 16 bits are used).

Therefore, to process this digitally, a processing capacity of 44.1KHz x 2CH x 32bit = 2.8224MHz is required.

Sample rate and bit rate Part 2

Sampling theorem

It is a very simple explanation, but it can express up to half the sample rate. When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when talking about signal processing or other timing) .

44.1 kHz

Why is there a poorly separated rate of 44.1? .. ..

Didn’t technicians deliberately make cumbersome clocks to prevent music CDs from being easily copied? I heard something like that. When I searched, it seems this happened (?) Due to the convenience of an old PCM recorder. In this age, it is difficult to know what 44.1 kHz is in development. The 44.1 kHz ↔ 48 kHz sampling conversion is a headache. For example, USB audio (USB audio device class) exchanges data at 1 ms intervals. In the case of 48kHz, the data is 48 samples, but when considering 44.1kHz, it will be 44 samples (x9) and 45 samples (x1) in 10ms. If you cheat on a sample when there are 45 samples (tentatively), it will be 44.0kHz. I think it’s more like that with voice and music, and the human ear usually cheats (it’s just my personal opinion).
However, the objective evaluation method will soon come to an end. For example, you can clearly see that you were fooled by a sine wave (sine wave) (maybe you are unexpectedly on the market).

Number of quantization bits

The sampling had to take a value in the direction of time (discretization), but the quantization had to take a value in the direction of amplitude. The range that it is possible to display the volume of the sound, which is often heard, “96 dB dynamic range” means that the number of quantization bits is 16 bits, and the musical signal is reproduced in the range of 0 to 65535. dog. The number of quantization bits is also called the bit depth or bit depth.

Bitrate

In communication, it indicates how many bits of data are transferred per hour and is generally expressed in bps (bit / s) of how many bits are transferred (processed) per second. If it is low, the size when saving as a file is small and there is space on the transmission line for communication. For example, when an audio (1 channel) is compressed to 1/3, the 3 channel audio can be sent at the same bit rate. Excuse the old story, but considering from the age of analog communication (analog mobile phone), digitization + compression will be able to support multiple calls with the same radio wave.

Finally

I often hear what is called Hi-Res Audio. The sampling frequency is said to be 96 kHz or 192 kHz, which is over 48 kHz, the number of quantization bits is 24 bits, and the limit (high range) of human hearing is about 20 kHz, but it expresses frequencies higher than that. It will be. It is the same bit rate as the image from a long time ago. .. ..
By the way, it seems that dogs can hear up to 60 kHz and cats up to about 64 kHz.

Hi-res audio example
Sampling frequency Number of quantization bits Number of channels bit rate Frequency that can be expressed
192 kHz 24 2 9,216 kbps 96 kHz
192 kHz 16 2 6,144 kbps 96 kHz
96 kHz twenty-four 2 4.608 kbps 48 kHz
96 kHz 16 2 3,072 kbps 48 kHz
48 kHz twenty-four 2 2,304 kbps 24 kHz
Considering the limit of human hearing (about 20 kHz), according to the sampling theorem, 48 kHz or 44.1 kHz is a sufficient frequency, but what about all of them? .. ..
In my case, I cannot distinguish the high resolution range, but it should be able to reproduce the discarded frequency at 48 kHz to 96 kHz, and when the number of quantization bits is in the 24-bit range, the sound pressure (dB) is a little. Feels like it’s going up (?) (It’s just a story from my ears).
I’d like to make a comparison if I get the chance, but I don’t think I can tell by ear without a proper regenerator (like an expensive analog amp).

Is it time for cats and dogs to get verified in the acoustic industry? .. ..