What do the audio sample rates and sample sizes mean?


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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.


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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!

Understand what audio compression is

Understand what audio compression is

Audio Compresion

A container format is a data format that “encapsulates” other encoded data. It often contains “meta information” about the encoded data, or has a way of storing several separate streams of encoded data, or something like that.

Adio Compression

The encoding produced by the codec is the real essence of the data stream.

The most common example I can think of is the Ogg / Vorbis format. Ogg is the container format and Vorbis is the encoding. So you have an Ogg file and inside there are these little segments that contain encoded data. Each block contains a stream of Vorbis-encoded data and nothing else. For example, a cube might have the name of an artist and the title of a song stamped on it.

So, back to technology:

If you already have lossy music like mp3 or ogg / vorbis, converting it to lossless format will only take up (a lot) of disk space and will NOT, at all, NOT improve the audio quality at all. You can’t create loyalty when it’s already lost. Unless you’re writing a Visual Basic GUI on some popular TV show called CSI, but that’s fantasy, not reality.

If you have music in other lossless formats and want to convert it to FLAC, you can.

Be careful when using the term “WAV”. Wav doesn’t have to be lossless; in fact, WAV is just a container for the various possible formats. In this sense, it is similar to AVI. You can have lossless WAV if it is just raw PCM data, but you can also embed MPEG-1 Layer III (lossy) data in a WAV file.

It is possible to lose data when converting from one lossless format to another if you reduce the precision of the data. For example, if you convert an unsigned 16-bit PCM data stream at 48000 Hz to 8-bit PCM data at 44100 Hz, you lose precision in two ways: samples are merged from 48000 to just 44100 at a time. second (leading to data loss), and the data needs to be scrambled to fit the information into just 8 bits instead of 16 per sample, which will drastically degrade the quality.

Every digital audio stream, even encoded with a compressed encoder (lossy or lossless), has the following sample format properties, which are important elements that describe the properties of the stream:

An example of bit width and depth, i.e. 8 bit, 16 bit, etc. Bit widths and depths are slightly different and there is also big endian / endian byte order (which does not affect quality) and signed or unsigned sign (which does not matter either) affects quality but does affect encoder / decoder operation with data). The key point to remember is “the more bits the better”. So 32-bit is better than 16-bit, etc.

Frequency, also known as the sample rate. The more the better, because more “samples” of sound are played per second. Imagine sliding your finger across a deck of cards and seeing the cards blur; this is essentially how digital sound occurs. Each sample is a map, and if you have more maps flying per second, the sound is softer. For example, you would really notice if you were flipping only 5 cards per second, but everything would be blurry if you were flipping thousands of cards per second. So it’s even better, because it’s more natural and closer to reality, which is analogous and infinitely divisible (well, up to Planck units, but this is debatable and off-topic).

Lossless simply means that if you use the same or better sample format in the output that you used in the input, you won’t lose any data.

Therefore, if you change from 16-bit to 32-bit sample format, you will not lose data. But if you go from 32 bit to 16 bit, you will lose data.

So the answer to your question about whether it makes sense to use FLAC depends on the original data: if you have 64-bit WAV files that were originally recorded in this 192,000 Hz (or 192 kHz) sample format, and you convert them to “format Standard 16-bit 44.1kHz FLAC, you’ll lose a ton of data. But if your WAV file is 8-bit with 22100 samples per second and you convert it to 16-bit FLAC with 44100 samples in second, you won’t lose data. and you can even increase the file size depending on whether you gain lossless compression or a smaller sample format.

The sample format will affect the amount of space the file takes up, so “bigger” bits and a “faster” sample rate will take up more space.

When it comes to practical considerations and human hearing, you won’t notice if you convert very high-quality originals to 16-bit FLAC at 44.1 kHz. But you won’t notice any improvement if you convert MP3 to FLAC either. As such, you need to evaluate what format your raw data is in before deciding what to do.