What is Audio Compression Threshold and How it Affects Sound Quality


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What is Audio Compression Threshold and How it Affects Sound Quality

Audio Compression
Audio Compression
Audio Compression
Audio Compression

Introduction

Audio compression is a technique used to reduce the dynamic range of an audio signal. It is commonly used in music production to make a recording sound louder and more impactful. However, compressing audio too much can lead to a loss of detail and a reduction in sound quality. In this article, we will explore the concept of audio compression threshold and how it affects sound quality.

Understanding Audio Compression

Audio compression is the process of reducing the dynamic range of an audio signal by attenuating the louder parts of the signal while leaving the quieter parts untouched. The main purpose of audio compression is to make the overall level of the audio signal more consistent, which can make it easier to listen to and mix with other tracks.

However, compression can also introduce artifacts such as pumping, breathing, and distortion, which can affect the quality of the sound. Therefore, it’s important to understand the parameters of audio compression, such as threshold, ratio, attack, and release, to achieve the desired sound.

“Compression is like a lens in photography. Just as a lens can bring certain parts of an image into focus while blurring others, compression can bring certain parts of an audio signal into focus while reducing the dynamic range.” – Bobby Owsinski, The Mixing Engineer’s Handbook

What is Audio Compression Threshold?

The compression threshold is the level at which the compressor starts to attenuate the audio signal. In other words, it’s the point at which the compressor kicks in and starts reducing the level of the audio signal. The threshold is usually set in decibels (dB), and it can range from -60 dB to 0 dB or higher.

Setting the compression threshold too low can result in over-compression, where the compressor is constantly active and the audio signal loses its natural dynamic range. On the other hand, setting the threshold too high can result in under-compression, where the compressor doesn’t kick in enough and the audio signal remains too dynamic. Therefore, finding the right compression threshold is crucial for achieving the desired sound.

“The compression threshold is the gatekeeper of the compressor. If you set it too low, the compressor will work too hard and the sound will lose its natural dynamics. If you set it too high, the compressor won’t work enough and the sound will be too dynamic.” – Bob Katz, Mastering Audio: The Art and the Science

How Compression Threshold Affects Sound Quality

The compression threshold can have a significant impact on the sound quality of an audio signal. Setting the threshold too low can result in a squashed and lifeless sound, while setting it too high can result in a dynamic and uncontrolled sound. Therefore, it’s important to find the right balance between dynamic range and consistency.

Additionally, different instruments and sounds require different compression thresholds. For example, a snare drum may require a higher threshold than a vocal track, as the snare drum has a shorter decay time and more transient peaks. Therefore, it’s important to adjust the compression threshold for each individual track to achieve the desired sound.

“The compression threshold is like a knife. Use it wisely,
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“The compression threshold is like a knife. Use it wisely,
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How to determine the right compression threshold

Determining the right compression threshold can be tricky, and it can take some trial and error to find the sweet spot. Here are a few tips to help you get started:

  • Start with a low threshold: A good starting point is a threshold of around -30dB. This will ensure that you are compressing the quieter parts of your audio, without squashing the louder parts.
  • Listen carefully: When you apply compression, listen carefully to the changes in the audio. You want to make sure that the compressor is not introducing any unwanted artifacts or distortion.
  • Experiment with different settings: Try adjusting the threshold up and down to see how it affects the audio. You may also want to experiment with the attack and release times, as well as the ratio setting.

Remember, there is no one-size-fits-all solution when it comes to compression. You will need to experiment to find the settings that work best for your particular audio.

“Compression is a great tool, but it’s easy to overdo it. Always err on the side of subtlety, and remember that sometimes a little goes a long way.”

– Brian Eno

The importance of a balanced mix

One of the most important aspects of audio compression is ensuring that your mix is balanced. If one element of the mix is too loud, you may be tempted to apply heavy compression to bring it down to the same level as the other elements. However, this can result in a dull and lifeless mix.

The key is to start with a well-balanced mix. This means that each element of the mix should be at a similar volume level, without any one element dominating the others. Once you have a balanced mix, you can then use compression to add subtle enhancements and make the mix sound even better.

“A good mix is all about balance. Each element of the mix should have its own space, and nothing should be too dominant.”

– Rick Rubin

The dangers of overcompression

While compression can be a powerful tool for enhancing the sound of your audio, it can also be easy to overdo it. Overcompression can result in a number of unwanted artifacts, including distortion, pumping, and breathing.

One of the main dangers of overcompression is the loss of dynamic range. Dynamic range refers to the difference between the loudest and quietest parts of your audio. When you apply too much compression, you reduce the dynamic range, resulting in a flat and lifeless sound.

Another danger of overcompression is the loss of transients. Transients are the short, sharp peaks in the audio that give it its punch and energy. When you apply too much compression, you can squash these transients, resulting in a dull and uninspired sound.

“Compression is a great tool, but it’s important to remember that it’s just one tool in the toolbox. Don’t rely on it too heavily, and always remember to use it in moderation.”

– Tony Maserati

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Audio and Video Compression Basics

Audio and Video Compression Basics

Audio and Video Compression Basics
Audio and Video Compression Basics
Audio and Video Compression Basics
Audio and Video Compression Basics

 

As we rely more and more on digital media, understanding the basics of audio and video compression becomes increasingly important. Compression is the process of reducing the size of digital files without sacrificing too much quality. Without compression, media files would take up a lot more space on our hard drives, making it difficult to store and share them. In this article, we’ll explore the fundamentals of audio and video compression and how it works.

Understanding Audio Compression

Audio compression is the process of reducing the dynamic range of an audio signal. Dynamic range is the difference between the quietest and loudest parts of a sound recording. Compression reduces this difference, making the quieter parts louder and the louder parts quieter. This is useful for improving the overall balance of a mix, and also for preventing distortion when the loudest parts of a recording exceed the maximum level of the recording medium.

Compression can be applied during recording or in post-production, using software tools like mp4gain. When done properly, compression can improve the clarity and punch of a recording, making it sound more polished and professional. However, overuse of compression can lead to a loss of detail and a “squashed” sound that lacks dynamics.

As musician David Byrne said in his book “How Music Works”:

“A good mix is one where the listener can hear and feel everything that the musicians and the engineer intended to be there.”

Understanding Video Compression

Video compression is the process of reducing the size of a video file by removing redundant or unnecessary data. This is done by encoding the video using a codec, which stands for “coder-decoder”. Codecs use complex algorithms to analyze each frame of a video and compress it in a way that minimizes the loss of quality.

There are two types of video compression: lossless and lossy. Lossless compression reduces the size of a video file without any loss of quality, but it’s not as effective as lossy compression in terms of file size reduction. Lossy compression, on the other hand, sacrifices some quality to achieve a smaller file size. The level of quality loss depends on the amount of compression applied.

When it comes to video compression, there are many factors to consider, including the resolution, bit rate, and frame rate. By adjusting these parameters, you can find the right balance between file size and quality for your particular needs.

As filmmaker and author Robert Rodriguez once said:

“Filmmaking is a chance to live many lifetimes.”

Compression Techniques for Audio and Video

There are many compression techniques used in audio and video, each with its own strengths and weaknesses. In audio, the most common type of compression is called “peak compression”, which reduces the volume of loud sounds that exceed a certain threshold. Another type of compression, called “multi-band compression”, divides the audio signal into multiple frequency bands and applies compression to each band separately.

For video compression, the most popular codecs are H.264 and HEVC (High-Efficiency Video Coding). H.264 is widely used for streaming video on the internet, while HEVC is more efficient but requires more processing

Audio Compression Techniques: Understanding the Basics

Audio Compression Techniques: Understanding the Basics

Audio Compression
Audio Compression
Audio Compression
Audio Compression

What is Audio Compression?

Audio compression is the process of reducing the size of digital audio files by removing redundant or unnecessary information, while maintaining the perceived quality of the original sound. This is done by using various algorithms that analyze and modify the audio data in a way that reduces its file size.

Types of Audio Compression Techniques

There are two main types of audio compression techniques: lossy and lossless.

Lossy Compression

Lossy compression algorithms are used to achieve high compression rates, but at the cost of some loss in quality. In lossy compression, some of the original audio data is discarded or modified in a way that reduces its size. The amount of data that is removed or modified depends on the compression algorithm used.

Some popular lossy compression algorithms include MP3, AAC, and WMA. These algorithms are commonly used for music streaming, online radio, and other applications where high compression rates are necessary.

Lossless Compression

Lossless compression algorithms are used to compress digital audio files without losing any information. These algorithms are designed to reduce the size of the file by removing redundancies in the data, but without modifying any of the original information.

Some popular lossless compression algorithms include FLAC, ALAC, and WAV. These algorithms are commonly used for high-quality music streaming and for archiving music collections.

How Audio Compression Works

Audio compression works by analyzing the original audio data and then modifying it in a way that reduces its size while maintaining its quality. This is done using various mathematical algorithms that compress the data.

The most common way to compress audio data is to use perceptual coding. This method takes advantage of the human ear’s limitations in hearing certain frequencies and sounds. By removing these sounds, the audio data can be compressed without the listener noticing any loss in quality.

Another method of audio compression is predictive coding. This method uses mathematical algorithms to predict the next sample in a waveform based on previous samples. The difference between the predicted sample and the actual sample is then compressed and stored.

Why Audio Compression is Important

Audio compression is important because it allows us to store and transmit audio data more efficiently. This means that we can store more audio files on our devices and transmit audio data faster over the internet. Without audio compression, it would be impossible to stream music or podcasts over the internet.

12 Common Questions About Audio Compression Techniques

1. What is the difference between lossy and lossless audio compression?

Lossy compression algorithms are designed to achieve high compression rates at the cost of some loss in quality, while lossless compression algorithms are designed to compress audio files without losing any information.

2. Which audio compression algorithm should I use?

The choice of audio compression algorithm depends on the intended use of the audio file. Lossy compression algorithms like MP3 and AAC are commonly used for music streaming and online radio, while lossless compression algorithms like FLAC and ALAC are commonly used for high-quality music streaming and archiving.

3. How much does audio compression affect the quality of the original sound?

The amount of quality loss in audio compression depends on the compression algorithm used and the degree of compression applied. Lossy compression algorithms generally result in some loss in quality, while lossless compression algorithms do not.

4. How can I tell if an audio file has been compressed?

You can usually tell if an audio file has been compressed by looking at its file extension. Lossy compressed files usually have extensions like MP3, AAC

Audio (audio) compression comparison [mp3, wma, ogg, atrac] Part 2

Audio (audio) compression comparison [mp3, wma, ogg, atrac] Part 2

AUDIO COMPRESSION

[Sound source used and points of interest]
・ 1kHz sine wave
Check for noise or correction. Investigate if abnormal sounds are mixed by emphasis or noise different from the originally generated range.

Audio Compression

· White noise
Check the frequency characteristics. Use sounds that are emitted at the same level for all sounds from 0 to 20 kHz and see if they are reproduced correctly.

·music
Use real music and investigate the differences with the original.

[Bitrate Settings]
Fixed bit rate: 96kHz, 128kHz, 256kHz, 320kHz Variable bit rate: 96-160kHz, 192-320kHz.
However, depending on the software, 320kHz cannot be set fixed and 350 can be set, or the upper and lower limit bits cannot be specified in the variable, and the sound quality standard can be specified in several steps ( medium sound quality, high sound quality). quality).be. Also, there are some that are configured with average bitrate instead of variable bitrate, so understand that it’s not a completely fair comparison.

[Software used [encoder]]
・ MP3 system
Afternoon Koda Ver.3.11a [gogo.exe ver.3.11]
Lame Ivy Frontend Encoder Ver.2.91 [Lame.exe Ver.3.93]
B’s GOLD Ver.7.12 [Unknown]
RipAudiCO Ver.3.70 [leme_enc.dll Ver.3.93]

・OGG system
oggdropXPd Ver.1.7.11 [Unknown]
B’s GOLD Ver.7.12 [Unknown]

・ WMA
B’s GOLD Ver.7.12 [Unknown]
(For WMA, I tried 3 types of software in my environment, but the result was exactly the same (maybe the encoder itself uses the same thing?) And it corresponds by software Since the bitrate range was narrow, only used a type).

・ ATRAC
nothing special. For ATRAC, we recorded analog from a CD player to an MD deck, optically connected an MD deck to a PC, and measured what was captured by WAV.

· To measure
Wave Space Ver.1.31

【others】
Although it is different from the main theme, I converted it to WAV for the visual measurement of each standard (because WaveSpace only supports wav), but the position where the sound of the WAVized data ends and the total playback. We discovered that there was a difference in time. , so we also investigated it.

3.Hardware 3.
Originally, the equipment used should be described in detail here, but the hardware environment is different for each individual, and this survey is only a guide in the first place, and it will be different if other people do the same. is a possibility of results, I will omit the detailed description of the hardware. (The thing is that I don’t have enough equipment to publish)

【Results of the test】
See the following for a summary of the results of each survey.
・White noise measurement result
・1kHz sine wave measurement result
・Music measurement results
・Simple file size and comment list

[Discussion]
ah There seems to be no big difference in file size (between the same bitrate)
stomach. Sound quality appears to be MP3 < WMA < OGG at low bit rates
(MP3: 128 = WMA: 96 < OGG: 96).
Hare. There is little difference at high bit rates
(there is a slight difference in the treble range, but it seems you won’t notice the difference unless you’re in a very good environment).
Worker. The difference in the encoder software was more than I expected
(especially in MP3)

“My conclusion”
[Less than 128]
If you’re worried about popularization (compatibility), [WMA] is good, and if you basically use it alone, [OGG] is good.
(I am worried about the amount of noise or the correction, but I sacrificed a bit on the sound quality anyway, so I chose the one that covers up to the high range as it is. Also, due to the relationship between ① and ②, mp3 is another with the same sound quality.The file size will be larger than

[With 256 and more]
The variable bit rate (192-320) of [Afternoon Koda] is good.
The fixed 320 is good for sound quality, but there is little difference between the fixed 256 and variable high-quality sound, and it seems that you can barely understand it even if you listen to it. If the sound quality is about the same, the smaller the file size, the better.

[Other impressions]
About OGG
I had high expectations for OGG, but I was concerned about the measurement result at 1 kHz, whether it was noise or correction. However, I find the relationship between sound quality (wide playback band) at low bit rates and file size to be excellent. At high bit rates the sound quality and file size are about the same as MP3s so I think MP3s are advantageous considering the penetration rate but I think they are doing pretty well considering the fact that they have just been developed. expected in the future

Audio (audio) compression comparison [mp3, wma, ogg, atrac]

Audio (audio) compression comparison [mp3, wma, ogg, atrac]

Compressed Audio

MP3-typed audio, etc., for storing music that was recorded on cassette tapes, music borrowed from CD rental shops or purchased music CDs, or for easy enjoyment with a portable player or car.

compressed audio

More and more people are recording with compression technology. However, there are many standards such as WMA recommended by Microsoft as well as MP3 when it comes to audio compression. Also, since the sound quality and compression rate of each standard change depending on the bit rate setting and the like, there is a wide variety of compression methods depending on the combination of the standard and the setting.

So, I wanted to check what the sound quality and file size would be when recording with which standard and with which settings, and select the standard that suits my purpose, so I took this survey. However, due to the investigation of the ideas of fans, the software and equipment used were covered by those that are freely obtainable in hand or on the net, so the result may be different from the original performance. , but it is only one. Take it as an example.
Since this test focuses on sound quality, it does not test at a low bit rate, which deteriorates sound quality.

Finally, in conducting this survey, I referenced many documents on the Internet. We would like to express our gratitude to each person (individual/corporation) for facilitating us to review materials that have been researched and created with considerable effort from their respective points of view. The sites I mainly referred to will be featured at the bottom of this page, so I recommend that those who are viewing this also take a look.

[Survey outline]
1. 1. Destination standards
As mentioned above, there are many audio compression standards, but here we have limited them to MP3, WMA, OGG, and ATRAC. The standards and reasons for the survey are shown below.

・MP3 ( Moving Picture Experts Group 1 Audio Layer – 3 )
I chose it because it is probably the best known and most popular standard and there are many compatible players for the same reason.

・WMA ( Windows Media Audio ) _ _
It is widely known alongside MP3. Recently, it has become compatible with car audio and DVD players. Also, according to a theory, the same bitrate is rumored to have higher sound quality and compression than MP3, so I chose it.

・OGG (Ogg Vorbis)
It may not be familiar to you yet, but although MP3 requires a license, the number of compatible players is gradually increasing due to the fact that it is unlicensed but offers high sound quality and high compression. Since it is (apparently) high-performance and license-free, it is easy to develop encoders and playback software, so we chose it with the expectation that it will spread in the future.

・ATRAC ( Advanced TR Transform Acoustic Coding ) _ _
This name may not be familiar to you, but you can understand the standard adopted by MD. Many people think that MD has the same high sound quality as CD, and since it is widely used as a storage medium for music, it was used as a reference for comparison.

・ Reason for not targeting other standards
There are many compression standards in addition to the above, but there are few compatible software and players, and considering the interaction with others (although I cannot say publicly), I judged that the comparison with the three types above is adequate. In addition, there is a standard called OpenMG (ATRAC3) recommended by SONY, etc., and there is no need to adopt other than SONY in mobile players, etc., but there are still few (limited) supported players, and recording is done. except for VAIO users, since it is difficult to do so, it was excluded from the target.

2. 2. Survey method
The three types of sounds selected for the survey were converted to various bit rates of each standard, visually compared to the original sounds, and listened to and evaluated. Also, I heard rumors that although the standard is the same, there are differences depending on the conversion software, so I used various types of software (encoder). the detail is just below.

What do the audio sample rates and sample sizes mean?

What do the audio sample rates and sample sizes mean?

The human hearing range

You can see that MP3 audio files have audio in the number of bits (in seconds) that the player uses, that is, the bit rate that indicates the quality of the audio.

human hearing range

But I am confused with the terms sample rate and sample size. Are they dependent on bit rate and sound quality? Or can it be explained in understandable terms?

This is a great article on the three terms you are asking. In summary, here are three definitions.

Bit rate: the amount of data per second. This can vary within the file (variable bit rate) and can have static values.
Sample Rate – The rate at which audio is measured per second. It is usually measured in kilohertz (kHz). The usual number you can see is 44.1 kHz. This is directly related to the bit depth or the number of bits measured in each cycle.
So at this point you need to do some math and you can see that the bitrate is in bits per second (usually measured in megabits per second). Therefore, bit rate = sample rate x bit depth. As far as I know, your sample size is just one of these 1-second chunks of data.

If you run pure math, you will find that these files are very large, but there are some compression algorithms that have been adopted to keep the files low without a significant loss of quality.

The sample size or bit depth is included, which is a measure of the number of bits in the sample, which is a direct quality measure. However, this only applies to PCM sampling. For irreversible formats like mp3, the sample size doesn’t really define the quality.

See Audio Bit Depth for more information.

1
2012/02/10Florist
Sample rate = There is no sample rate. Of audio samples transported per second

Sample size = The sample size determines the maximum dynamic range of a digitized sound. Dynamic range is the ratio of the maximum amplitude to the minimum non-zero amplitude of a signal, generally expressed in decibels (dB).

The sampling frequency affects the quality of the recorded sound. Therefore, a higher sample rate will improve the quality as the number of bits increases, but will require more data and result in larger files. The bit rate used to store the samples used to store the sampled data also affects the quality of the recording. Bit rate is the amount of space that can be used to store sampled data per second. The higher the bit rate, the better the sound, but more space is required to store the file.

Relationship between human audible range and sample rate

Relationship between human audible range and sample rate

Audio Sample Rate

The two main factors that indicate the performance of an audio interface are the number of sample bits and the sample rate.

sample rate

Of these, the number of sample bits is expressed as a numeric value, such as 16 bits or 24 bits, and last time I introduced that the dynamic range differs based on the difference in the number of sample bits. In other words, we have also used graphs to show that the difference in the number of bits is the precision with which very quiet sound can be expressed.
So what about the other sample rate? The sampling frequency is also called the sampling frequency, but the unit is usually kHz. The most commonly used are 32 kHz, 44.1 kHz, 48 kHz, and 96 kHz.
The Roland audio interfaces introduced last time, such as the UA-1X and UA-3FX, as well as the UA-1D and UA-20, are models that support 44.1 kHz and 48 kHz.

UA-1X dal_4007_s.jpg dal_4002_s.jpg UA-20
UX-1X UA-1D UA-3FX UA-20
As many of you will know, CDs, which can be said to be representative of digital audio, are compatible with 44.1 kHz and with 44.1 kHz, that clear sound can be expressed. But why is it 44.1 kHz? Here is a clear medical basis. It is the relationship with the human audible range, that is, the audible frequency band.
Generally, the highest pitch that can be expressed is said to be half the sample rate. In other words, 44.1 kHz is up to 22.05 kHz and 48 kHz is up to 24 kHz. On the other hand, the range that humans can hear is said to be 20 Hz to 20 kHz for healthy people. Therefore, according to the theory, recording of 20 kHz or more does not make sense because humans cannot perceive it. However, considering a small margin, it is the CD standard that can be expressed up to 22.05kHz. However, the reason it became a medium number like 44.1kHz is that when CD was standardized, the VTR was used for digital recording, and the TV’s horizontal and vertical sync signal was 44.1kHz., It is said which was by using it.

■ Can humans really detect sounds above 20 kHz?

However, if you can’t really hear more than 20 kHz, there is no point in picking up frequencies above that. But is that true?
The answer is clear from the appearance of DVD-Audio, which has a sound quality superior to that of CDs. Yes, it is certainly difficult to recognize 20 kHz or more as a single signal, but when signals of various frequencies, such as music, are expressed in an overlapping way, the atmosphere of the sound that can be heard depends on whether 20 kHz or more is being output. o No. It makes a difference. When I listen to a CD and an analog record, sometimes I feel that the sound of the record is better, but it can also be said that this is the result of not setting an upper limit on the frequency in the case of analogs.
Here, let’s experiment a bit to see if it is true that “the highest pitch that can be expressed is half the sample rate.”

48 kHz 96 kHz 48 kHz 96 kHz
White noise expressed at a sampling frequency of 48 kHz (left) and a sampling frequency of 96 kHz (right). In the case of 48 kHz, the sound is output only up to about 24 kHz, but in the case of 96 kHz, all the sound is output flat. In the two graphs above, the horizontal axis was only up to 48kHz, so it looked completely flat at 96kHz, but when the horizontal axis is up to 96kHz and expressed in exponential notation, it is 48k, which is almost the same as the theoretical . value. You can see exactly what comes out.
The graph shown here shows the extent to which frequency is expressed by creating white noise that mixes evenly from low to loud sounds at 48 kHz and 96 kHz. If you look at this, you can see that the 48 kHz sample rate is up to about 24 kHz and the 96 kHz sample rate is up to 48 kHz. However, the two charts on the right side have an index on the horizontal axis, so it might not seem like much of a difference, but it does have a double number range.
You can say that this is the difference between 48kHz and 96kHz.

■ If you want to make a CD last, do you need 24-bit / 96 kHz specifications?

By the way, some people may have some doubts about the story so far? Yes, I would like to digitally record analog recordings and tapes and eventually convert them to a CD, but if the CD itself is 16-bit / 44.1 kHz, the specs, such as 24-bit / 96 kHz, are above spec. Is it unnecessary?
It certainly may not be necessary if you burn the recording as is to CD without any processing.

What is Sample Rate and Bit Rate Depth?

What is Sample Rate and Bit Rate Depth?

Audio Compression

Both image and video data have some numerical values ​​related to image quality, such as the number of pixels, the number of colors that can be expressed, and the number of frames per second in the case of video.

Audio Compression

Similarly, audio data also has two numerical values ​​related to sound quality, which are the sample rate and the bit rate. I do not understand the difficulty in either case, but I am sure I am not mistaken, so I will write about these two today.

Sampling rate
Let’s start with the sample rate.

Simply put, the sample rate is a numerical value that indicates “how loud the sound is recorded.” For some reason, when the sampling frequency is 44.1 kHz, it is not possible to record up to 44.1 kHz and it seems that it is possible to record up to about 22 kHz. Remember that you register up to half the frequency. If you’re wondering why that happens, google it (laughs).

It seems to have an effect on the sound of musical instruments that produce a crisp sound like cymbals, but I have never bothered to change the sample rate under the same conditions and compare them, so the amount of sound depends on the frequency of sampling. It is unknown if it will change. In professional environments, it is often recorded at 48 kHz. On rare occasions, the sample rate changes the sound quality, and some teachers boast that they can tell the difference. You seem to understand something. I would love to take a blind test, but I don’t have free time to go out with me.

Bit rate depth
This is a numerical representation of “how low a sound can be picked up (small change in volume)”. This can be a bit difficult to imagine.

The higher the bit rate, the smoother the waveform lines will be as the sound rises and falls, and the lower the depth of the bit rate, the rougher it becomes.

There are two options, 16-bit or 24-bit. There are also 32 bits at the moment.

Bitrate is likely to make a difference when recording percussion instruments such as drums (instruments with extremely loud volume). Some engineers record in 16-bit from scratch because the sound impression changes when 24-bit drum sound is converted to 16-bit for burning to CD. Unlike the sample rate, this is quite different.

Personal feeling about sample rate and bit rate.
First of all, the sound quality of commonly sold CDs is 16-bit at 44.1 kHz. And, in the professional field, it is often recorded at 24 bits and 48 kHz (which is called Neyonyonpachi). And the human audible range is said to be up to 20 kHz.

With that in mind, it is honestly ridiculous to see and hear something like “This audio interface supports up to XXkHz, so the sound is good …”. Just record at 2448. And there should hardly be any current audio interface model that doesn’t support 2448.

There are audio interfaces that support 192 kHz, but I honestly doubt the idea that the higher the sample rate, the better the sound quality. The basis of recording is to record the desired sound as loud as possible. To record sounds that are far from the human audible range, reducing the proportion of sounds that we really want (of course, sounds that can be heard by the human ear) is what we call high-quality sound. First of all, I think that high frequency sound is nothing more than noise like white noise. If you think that those high frequency sounds are generated by playing musical instruments, it means that the same or louder sounds are generated from fluorescent lamps and all machines, and those sounds are also recorded.

Data lost due to compression is irreversible Part 2

Data lost due to compression is irreversible Part 2

 

audio compression

[Quantization bit number (bit depth)]

Audio Compression

◉ Unit: bit
◉ Audio: Resolution related to volume. The higher the value, the more faithfully the quiet sound can be reproduced and the wider the theoretical dynamic range (ratio of the maximum and minimum volume values). 16-bit, 24-bit, and 32-bit floats are used primarily in production.
◉ If you compare it with the video …: Conceptually, it corresponds to the number of gradation bits. In terms of feel, it is almost the same as the dynamic range of the video. The wider the range, the greater the gradation possible without overexposure and underexposure.
◉ Remarks: There is no concept of the amount of quantization bits in compression formats such as MP3.
◉ Image of the number of quantization bits

When a square is cut on the vertical (volume) axis, the volume change less than one step cannot be reproduced, resulting in noise. In other words, the finer the squares, the more accurately the low volume can be reproduced. The actual number of steps in the number of bits in common use is as follows.

・ 16 bits → 65,536 steps

・ 24 bit → 16,777,216 steps

It can be seen that the 24-bit, which is said to be high-resolution, can reproduce the volume change much more accurately than the CD-quality 16-bit. In other words, 24-bit has a “wider dynamic range” than 16-bit.

[Sampling frequency]
◉ Unit: Hz
◉ Audio: Temporal resolution. Involved in the reproducible frequency range. If the frequency is low, the treble range will not be reproduced correctly. As the frequency increases, it is possible to reproduce frequencies above the audible range. Those used primarily in production are 44.1 kHz, 48 kHz, 96 kHz, and 192 kHz.
◉ If you compare it to video …: In terms of temporal resolution, it is equivalent to frame rate. The higher the speed, the smoother the video will be (in the case of sound, it is perceived as treble reproducibility rather than smoothness).
◉ Remarks: The upper limit of the frequency that can actually be reproduced is half the frequency. For example, if the speed is 96 kHz, it can be played up to
48 kHz ◉ Explanatory sampling frequency diagram

If you compare it to a video, you may understand it in some way. As of 2018, I think the lowest line quality that can be used regularly is the “16 bit / 44.1 kHz” used by CDs. If each value gets lower than this, it will collapse more and more so that it can be heard. If the number of bits is small, small sounds are converted to noise, and if the sampling frequency is small, the aliasing noise (noise that is inevitably generated by digitization. Moiré sound phenomenon) falls into the audible range and is comes back jarring. And note that half the value of the sample rate is the upper limit of the actual recorded / played rate. In other words, in the case of “44.1 kHz”, the actual recording / playback is up to about 22 kHz. The human audible range is said to be 20Hz to 20kHz, so that’s a sufficient value in terms of specs. By setting the sample rate to twice the upper limit of this audible range, overlapping noise is removed from the audible range, and by cutting it with a digital filter, jarring noise, which is CD quality, is removed. From this, you can see that “16 bit / 44.1 kHz” is the lowest line.

The master file
must be of high quality

That said, it’s hard to understand how sound quality changes at low bits and low sample rates without actually experiencing it.

Data lost due to compression is irreversible

Data lost due to compression is irreversible

Audio Compression

In this series, we will focus on the basic knowledge about “sound” that is necessary for video production, and we will make it easy to understand by omitting small and difficult things as much as possible, such as a little general knowledge and sound, including music. . I look forward to delivering it, so I look forward to working with you!

Audio Compression

Now, let’s talk about the first memorable event under the name [Digital Audio Basics]. There are several types of digital audio. Among them, I have summarized the main ones.

[Format types and functions]
◉ Uncompressed format: linear PCM (WAV, BWF, AIFF)
→ The most basic format for digital audio. BWF is a commercial WAV that can contain metadata.

◉ Lossy compression format: P3, AAC (MP4), MQA, etc.
→ Format used mainly for general purposes. In many cases, the information in the uncompressed data is shrunk and compressed. The data capacity is reduced, but the sound quality also deteriorates accordingly. MQA is a new format that is irreversible in terms of data, but reversible in terms of sound quality.

◉ Lossless compression format: FLAC, ALAC, etc.
→ Format mainly used for high-quality listening. It has the reversibility of being able to reproduce exactly the same sound quality as before compression, but the data capacity is not that small.

◉ Others: DSD (DSF, DSDIFF, etc.)
→ It is also called 1-bit audio, but since the concept is fundamentally different from multi-bit audio like linear PCM, it can be compared to “24bit” WAV, etc. in the same line I have not. Currently, it is one of the highest quality formats, but it has the weakness of not being editable.

How is it? I think there are several things, from the familiar ones to the ones you see for the first time, but among them, the one that is most suitable for today’s video production is “Linear PCM”! The reason is as follows.

1. Since it is an uncompressed format, it has excellent sound quality.

2. You can edit like cut and paste.

3. The digital voice tracker is the most popular Ma ‘around the world because the bet, any device, can be managed by software.

Since MP3 and AAC (MP4) are compressed formats, there is a considerable loss in sound quality. Depending on the compression ratio, it may not be obvious at first glance, but it is not suitable as processing-based material such as video production and music production. FLAC and ALAC are lossless compression formats that do not deteriorate sound quality, but do not significantly reduce capacity, and there is no software that can be edited natively (without conversion to other formats), so it is still unsuitable for the production. . DSD was adopted from SACD which appeared in 1999, and is said to be the most analog digital audio today, and it has a smooth texture that is different from linear PCM in terms of sound quality. This format has finally attracted attention in recent years, but due to its mechanism, it has the weakness that it cannot be edited as is, so on the production site, mainly one-shot music recording (recording without editing) and mixing (long-playing recording without editing) and mixing (often used as a master recorder when combining multiple sounds into one stereo or surround sound (also called track down). “Almost Ichi 択 linear PCM” video production, I think I could understand that you can refer to. Of course, if the compressed format does not make you uncomfortable, you can use it, but consider it as an emergency. If you still want quality, you must use linear PCM. The data lost by compression is irreversible. The file that will be the master of the work must be of the highest possible quality. By the way, whether you use WAV or AIFF, the sound quality is almost the same. However, co Considering compatibility, even Mac users can be relieved to use WAV for data transfer.

“16 bit / 44.1 kHz” is
the lowest line of CD quality

Now let’s dive a little deeper into linear PCM. There are “number of quantization bits” (bit depth) and “sample rate” (sample rate) that represent linear PCM specifications. Have you ever seen the notation “16 bit / 44.1 kHz”? This means that the original (analog) audio is sampled (digitized) 44,100 times per second at the 16-bit volume stage (2 raised to 16 = 65,536)! Still, I think it’s “what is this?”, So I tried to sum up the points by comparing it to the video!