Tag: convert sample rate to bandwidth

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What is a bit rate?

Detailed explanation of bitrates for video and music files
What is a bit rate? image image
Bitrate is an unavoidable experience in video editing and music production.

The word bitrate sounds a bit complicated, but of course the video and audio files. It is also often found in various situations, such as mobile phones and Internet lines.

What is the bit rate? It’s surprisingly helpful to remember it sensually, so please remember it.

I think the person who visited this page is a beginner asking “What is bitrate?” , so I hope I can explain it in the easiest and simplest way possible.

Bitrate Basics
To briefly explain bitrate, it refers to the amount of data per second for communication such as video files, music files, and the Internet.

The so-called playback and communication are data streams. This flow uses a unit called bps (Bit Per Second) to indicate how many bits flow in one second (Per Second).

I think it’s easy to understand if you imagine the road that a car is driving on. The car is the data and the data flow is the road.

Considering that a highway with many cars carries a lot of people and luggage, it can be said that a large amount of data (information) flows. This is called high bit rate.

On the contrary, if there are few cars running, it can be said that little data (information) flows. This is called low bit rate.

Image of many cars driving on the road.
A state where the amount of data is large and the stream bit rate is high

Image that there are not many cars circulating on the road.
A state where the amount of data is small and the bit rate is low

Should the bitrate be high?
A high bit rate means a large amount of information. If there is a lot of information, it sounds great, like high image quality for videos and high sound quality for music, but it has a big drawback.

The downside of a high bitrate is that it increases the data capacity of a single file. If the data capacity is too large, it can take time to send it to someone or move data between hard drives, making it difficult to manage.

Therefore, the codec was developed to reduce data capacity while maintaining high image and sound quality. I’ll talk about codecs another time, but you need to set the bitrate considering the balance between picture quality, sound quality and file capacity so that the data capacity is easy to manage.

How to check the bitrate
You can check the bitrate by looking at the detailed information of video and music files on both Windows and Mac.

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Audio encoding compression rate

The compression ratio of audio encoding is determined by the bitrate at the time of encoding.

In the previous article (Sample Rate and Bitrate, Part 1 –Smile Engineering Blog), I mainly wrote about the original sound (PCM) bitrate, but this time I would like to write about the bitrate and the compression rate. coding…

Specifically, if you set a low bitrate, the compression ratio will be high, for example, if you save the file, its size will be small. As I wrote last time, the sound source bitrate (PCM) before compression is as follows.

Bit rate = sampling frequency (Hz) x number of quantization bits x number of channels
For example, a music CD is 44.1 kHz stereo and the bit rate is

Music CD bit rate: 44100 Hz x 16 bits x 2 channels (stereo) = 1411.2 kbps
If it is encoded with MP3, AAC , etc., for example 256 kbps, the compression rate (assuming the original sound is 100%) is about 18% and the file size is 1/5 or less .

Encode a music CD at 256kbps: 256kbps / 1411.2kbps = about 18%
If it is 4 minutes of music, the file size is as follows.

44.1 kHz sample rate recording 4 minute song file size
as the original sound 1411.2 kbps x 240 seconds = about 40.4 MB
Encode at 256 kbps 256 kbps x 240 seconds = about 7.3 MB (+ header)
These days, it’s not the time to get songs on CD, so it’s an old story…
If a song is 4 minutes long, you can save 16 songs on CD650MB with the original sound, but at 256 kbps as MP3 and AAC . If encoded, 89 songs can be recorded.

Recording with a sampling frequency of 44.1 kHz The number of songs that fit on a CD (650 MB) of a 4-minute song
The original sound (music CD) CD650MB / 40.4MB = about 16 songs
Encode at 256 kbps CD650MB / 7.3MB = about 89 songs
If you check the web, you will be able to hear and compare the sound quality due to the difference in bit rate. I think the condition is that everything is the same except the bitrate, but first of all there is a difference in sound quality depending on the original sound source’s sampling rate (PCM) and the number of quantization bits (the bit rate changes from the original sound ) . At the time of analog-to-digital conversion (ADC), the sound quality is determined by the conditions. No matter how high the bit rate is to encode a sound source in poor conditions, the sound quality will still be poor. Even with the same bitrate, the compression rate changes depending on the number of channels (stereo or monaural). Therefore, strictly speaking, the evaluation of sound quality cannot be judged by the difference in bitrate alone. For example, when 48 kHz and 44.1 kHz 16-bit PCM is encoded at 32 kbps to 320 kbps, the compression ratio is as follows.

16-bit PCM compression ratio (when the original sound is 100%)
48 kHz stereo encoded bit rate
(1536kbps) 48kHz monaural
(768kbps) 44.1kHz stereo
(1411.2kbps) 44.1kHz monaural
(705.6kbps)
320kbps 320 / 1536 = about 21% About 42% 320 / 1411.2 = about 23% About 45%
256kbps 256 / 1536 = about 17% About 33% 256 / 1411.2 = about 18% About 36%
192kbps 192 / 1536 = about 13% About 25% 192 / 1411.2 = about 14% About 27%
160kbps 160 / 1536 = about 10% About 21% 160 / 1411.2 = about 11% About 23%
128kbps 128 / 1536 = about 8% About 17% 128 / 1411.2 = about 9% About 18%
64kbps 64 / 1536 = about 4% About 8% 64 / 1411.2 = about 5% About 9%
32kbps 32 / 1536 = about 2% about 4% 32 / 1411.2 = about 2% about 5%
Comparison with the original sound
It’s a slightly twisted idea, but for example, which is closer to the original sound, stereo or monaural under the above conditions? Considering the compression rate, it’s the latter. Of course, stereo is superior to monaural in terms of expression, such as expressing depth of sound, so it makes sense to compare this to assess sound quality, but in encoding, compression is efficiently done using stereo. Since there are algorithms (M/S stereo and intensity stereo), the quality is not even half as monaural, and the compression is done efficiently by the amount of stereo.

How to improve video sound quality? Part 2

Bit rate
It is a numeric value of how much data is voice data in one second.
The unit is bps.

If the bit rate is high, the sound quality can be improved because one second’s sound can be played with a large amount of data.

For AAC and MP3 used in Internet video, the upper limit is 320 kbps. (MP3 is the upper limit of the specification. AAC may be higher depending on software.)

resume
To briefly summarize,

・ High sampling rate
・ High bit rate

If so, the sound quality can be improved in terms of data.

However, the higher the bitrate, the greater the amount of audio data.
If it is 320 kbps, it will be 300 MB in an hour.

I wrote that there is a trade-off between image quality and capability when deciding the image quality of a video, but the same idea applies to sound quality.

If you want to stream music clearly, you may want to consider improving sound quality at the expense of picture quality.
If it’s a movie, you can look at both picture quality and sound quality, even if it reduces viewer comfort.
If you just need to broadcast how the instructor speaks alone, like in a seminar, you probably don’t need as much sound quality.

However, if the original audio data is noisy or the sound is broken, there is little point in improving the sound quality of the data.

First of all, it may be the most important thing to try to record in the best possible environment.
I’d like to write an article about the recording method at another time.

In addition to that, it should be set to high sound quality as video data.

How to improve video sound quality?

Sample size, sample rate, bit rate

Sound is an integral part of video.
As was the case with the author, initially I will handle the video with the stance that it is only necessary to make a sound.

However, as I went over and over again, I gradually began to wonder “how can I make a good sound” and “what kind of knowledge do I need to have to optimize the sound parameters?” I will come.

In this article, we will explain the sound of videos, learn how to make it “sound good” and what kind of knowledge should be used to control the sound quality and amount of audio data.

What is “video sound”?
There are several types of voice data.
Parameters related to the sound quality of audio data
Sample size (bit depth)
Sampling rate
Bit rate
resume
Points to build a video distribution service Free distribution
Video Recording And Live Distribution Guide You Can Do In-House For Free

What is “video sound”?
Video is a combination of video and audio data.
Video and audio are separate, and a video file is a combination of different data methods (codecs).

Since the video file has the above structure, the image quality and sound quality can be adjusted separately.
Therefore, the sound quality of the video depends on the audio data part.

There are several types of voice data.
There are several data methods for voice data.
There are two main types, each of which has multiple codecs (data methods).

lossless codec
A data method that has a low compression rate but can restore the original sound quality.
FLAC, ALAC, etc

lossy codec
A data method that has a high compression rate but cannot restore the original sound quality when played back.
MP3, AAC, WMA, etc

What is used in Internet video is a lossy codec with a high compression ratio.
There is a lot of AAC and MP3, and it seems that most of the videos that are generally seen these days are AAC.
Therefore, to aim for high sound quality in Internet video distribution, it is important to understand how the sound quality of AAC or MP3 (lossy codec) goes up or down.

Parameters related to the sound quality of audio data
The factors that affect the sound quality of audio data are listed below.
By understanding and understanding the following and setting them properly when creating audio data, you will be able to control sound quality properly.

Sample size (bit depth)
The larger the sample size, the finer the volume of the sound, so it sounds softer. If the sample size is small, it sounds harsh to the ear.
If it is 16-bit, it is played by dividing the loudness of 65536 by 2 to the 16th power.
If it is 24 bits, it will be 2 to the power of 24, which is 16777216 division.

Actually, when you create a 32-bit MP3 file and an 8-bit MP3 file from the same WAV file and compare them, the 8-bit has a “sir” noise and the 32-bit has the same level of fluency. sounds like the original rice field file. (The bit rate and sample rate of the two files are the same.)
In the case of MP3 and AAC, the bit rate also greatly affects the sound quality.
Sampling rate
This is the frame rate on the video.

It is a numerical value that indicates how many times a second is divided into data, and if it is 44.1 kHz, it means that the sound is divided into 44100 times and converted into data.
The finer the cut, the higher the sound reproducibility and the better the sound quality.

CD is 44.1 kHz, and it seems that the range said to be audible to the human ear can be almost covered by this 44.1 kHz. So even if the sampling rate is higher than that, the human ear can’t feel it.

However, many people feel that the sound is different at high sample rates, so there are audio files with higher sample rates.

44.1kHz or 48kHz is often used for the audio of Internet videos.
Since the music industry uses 44.1kHz and the video industry uses 48kHz, it is recommended to use 48kHz for audio used for video unless there is a particular problem.

When collecting audio materials with different sampling rates into one video, problems such as sound deviation will occur.

SAMPLING FREQUENCY

When the distribution begins, you may encounter such a phenomenon that
“the video and audio change gradually, although at first it is not so much.”

One of the causes is: -There is also a
error when setting the sample rate of the audio signal
.

If you are experiencing sound drift, you may want to review your settings.

Note:
What I’m talking about in this article is something I don’t understand, and although I mentioned “countermeasures” at the bottom for now, I really don’t understand the cause.
Still, I think the “countermeasure” described here will lead to an improvement for those who have problems with sound deviation, so I’ll post it.
(The author has made a clear improvement on the sound gap.)
Maybe there is an error in the article about the way of thinking, understanding, etc. note that.

What is sample rate and bit depth?

Sample rate is like a unit used when recording and playing back audio digitally,
y represents how much audio is sampled (sampled) per second.
It seems that Hz (hertz) is often used as a unit.

Bit depth is the number of bits by which a sample is represented and is expressed as either 16 bits or 24 bits.
It is used in the form of how many stages the difference in the sound that can be recorded is divided from the silent state to the maximum volume state, and the higher the bit depth, the more delicately the sound can be recorded and reproduced. . .

The minimum unit
of the vertical axis of the sound waveform (the one seen in voice editing software) is
the smallest unit of
horizontal axis of bit depth, which is determined by the sample rate.

When it comes to audio digitally, the two are often written as a set.
For example:
CD: 16bit 44.1kHz
DVD: 24bit 48kHz

A notation like 44100 Hz 16 bits appears even in the audio interface settings.
The values ​​that can be set and the sound quality that can be handled differ for each audio interface.

Why does sound change due to sample rate setting error?
I can’t explain this area well.
When I was afraid of the sound gap, I searched for various information and looked at it,
but I couldn’t find anything like a good explanation.

Even if you look at the site that has information on measures against sound deviation, it says that if there is a hint of one or two lines
and continuous deviation of the sound, you should check if the sample rate is misaligned
. my environment, but…

in my environment
By the way, in my environment, when I rewatch the recording when the distribution sound was out of sync, there was a symptom that the
audio played a bit earlier (recording time axis was a bit shorter)
. .. It was a subtle speed boost that I’d miss until someone told me, but it was certainly fast.

The range of sample rate settings that VirtualAudioCable itself can handle was quite wide (checked in the dedicated settings app), so if I wasn’t careful, before I knew it
(default?), the “default format” was set to 48kHz in windows settings. What I was doing was a blind spot.
The sample rate was fixed on the Windows side.
After fixing this, continuous sound space during distribution was improved.

How to set the sample rate
Sampling Rate Adjustment Concept
The sample rate should be the same as much as possible.
Since distribution software and mixing software handle multiple inputs, they (should) convert such differences, but still the sound quality deteriorates each time the sample rate is converted.
This time, it appears that there is a sound gap involved.

We will try to unify the sample rate settings here as much as possible for those used during distribution.
I think it is better to unify at 44.1 kHz or 48 kHz.

What are Encoding, Codec, Bitrate, Sample Rate Part 2

Bit rate / Sampling rate

Bit rate :
the amount of data processed per unit of time, which is used to indicate how much information can be processed and sent/received per second, such as
kbps.
Sampling rate (sampling rate/sampling frequency) :
represents the amount of sample data (frequency/frequency of occurrence?) per unit of time when converting a digital signal to an analog signal.
・ 44.1kHz or 48kHz
That’s how it is …

Fountain

Bit per second (bit per second) is a unit of data transfer rate (bit rate in JIS information processing terminology). It is defined as the number of bits that have passed (ie transferred) a virtual or physical point in the data transfer path per second.

▶ ︎ https://ja.wikipedia.org/wiki/ Bits per second
Sampling rate (sample rate) is sampling, which is a process necessary to convert analog waveforms, such as voice, into digital data, and is the sampling rate per unit of time. The commonly used unit is Hz.

Also called sample rate or sample rate.

▶ ︎ https://ja.wikipedia.org/wiki/Sampling rate
❤︎3, what is the standard file format?
This will come in handy when creating your own data, so it’s worth knowing.

*
If it’s a CD, even if you create a larger file with a 44.1 kHz

* iTunes is a
AAC file format with a
256 kbps bit rate
sample rate
5~10MB

It’s standard…

So the sample size (number of bits) is not shown in the compressed file, so there is no comparison by the number of bits, but it seems that Wav files and CD sound sources have a quality sound superior to compressed files.

It is easy to understand when compared to the bitrate on the WAV file side.

* iTunes sound source purchase file example

hello Adele
Hello Adele
Bit rate: 256 kbps Sampling frequency: 44.1
kHz
CCA
9.8MB

Sia-Viva
Sia Alive
Bit rate: 256 kbps Sampling frequency: 44.1
kHz
CCA
8.9MB

Spicy Red Chili Getaway
Red Hot Chili Getaway Rate
bits: 256kbps
Sampling frequency: 44.1 kHz
CCA
8.4MB

Original WAV-min file
By the way, the one I created uncompressed with Protools from the original sound source file is the one shown in
next figure.

adela hello mp3
Bit rate: 2304kbps
Sample size: 24 bits Sampling rate: 48
kHz
WAV
63.7MB

Also, when I encoded Adele Hello to MP3 with iTunes,
bit rate: 256 kbps → 160 kbps Sampling rate: 44.1
kHz
AAC → MP3
9.8MB → 5.9MB
.

So I feel like MP3, 160kbps and 44.1kHz are the breaking lines where the deterioration isn’t as noticeable.

WAV files are a backup instead of a listen

Typically, bit rate: 256 kbps, sample rate: 44.1 kHz, AAC is more than enough.

What are encoding, codec, bit rate, sample rate?

Basic DTM knowledge

Not limited to DTM people

Encode
encoder
codec
Bit rate
Sampling frequency
I think you hear it often

Do you only know by nuances?

So I hurried up and checked it out.

Table of contents that you can fly with a touch ↗︎
❤︎1, encode/decode/encoder/codec
❤︎2, Bit rate / Sampling rate
❤︎3, what is the standard file format?
Sponsorship

❤︎1, encode/decode/encoder/codec
As for the music files,

Coding :
File format conversion
・ WAV to MP3
Decode :
restore encrypted version
* Encoding can also refer to the compression of the file itself.
In that case, decoding also means returning the compressed data to an “audible” state.

Thank you for your comment <(_ _)>

Encoder/Decoder :
Software and equipment that encodes/decodes (encoder/decoder)
Codec :
how to convert file format
There seems to be a lossy compression method and a lossy compression method… In
other words, it seems that there are some formats that can be decoded and others that cannot be decoded
At the bottom of the codec page of the wiki it is written in detail
・MPEG audio codecs are irreversible compression methods, that is, they cannot be decoded…
・Apple Lossless (Apple Lossless Audio Codec, Apple Lossless): Decoding is possible with the lossless codec compression method installed in iTunes, QuickTime, etc.
・Windows Media Audio (WMA): It appears that you can choose which format the codec should have installed in Windows Media Player.

I checked the source on the wiki etc.

Encryption, also called encryption, is the conversion of digital data into a code according to the purpose according to certain rules. For details on the encoding method, please refer to the encoding method.

On a computer, it can also refer to file compression (also called “high-efficiency encryption”) or encryption. In this case, the function (software or hardware) that encodes is called an “encoder”.

Decode is also called decode and is an antonym for encode. To restore the encrypted information. The function to be decoded is called the “decoder”. Depending on the device that communicates and records information, it may be equipped with both an encoder and a decoder, and such bidirectional conversion function, conversion device, algorithm or the like is called a codec.

In a computer, the interpretation of a given machine language as an internal representation is called decoding, and its logic circuit is called a decoder. The entire mechanism that collects instructions and data and sends information to the arithmetic unit, centered on the decoder, is called the interface.

▶︎
A codec is a device or software that can encode (encode) and decode (decode) data bidirectionally using an encoding method. It is also used as a term to refer to the algorithm for that purpose.

▶︎
Hmmm…

What is the sampling rate (sample frequency)?

 

Differences in sound quality and how to check! It also explains the settings that need to be taken into account!
Sampling frequency setting Sound quality

plugin-boutique-2022-free
The sample rate is mentioned in DAW and audio interface settings. How should I set it?
I googled “sampling rate”, but I don’t understand all the difficult words…
The higher the sample rate, the better the sound quality?
I’m curious!

sample-rate-what-is
For those people, I would like to explain in a way that is easy to understand even for beginners, from the meaning of the sample rate that always appears when starting DTM, the relationship with sound quality, the confirmation method to the setting method. .

Please refer to that.

Table of Contents [ Close ]
1. What is the sampling rate?
2. Does the sound quality change depending on the sample rate?
3. How to check the sampling rate?
4. How should I actually set the sampling rate?
5. Points to consider in addition to the sampling rate
resume
1. What is the sampling rate?
Sampling frequency setting Sound quality
The sample rate is a frequency that indicates how accurately the sound is captured.

Sampling Rate, Sampling Frequency Also sometimes called Sampling Rate in English.

The higher the value, the more accurate the captured data will be.

Of course, if you want to import sound to your computer, you have to convert it to data. Capturing sound data is called sampling.

* Currently, sampling refers primarily to capturing existing sounds. (Use of existing music, recording material, etc.)

Rate means fee or commission in Japanese.

In other words, sampling = capturing sound data. Rate = rate, commission

Literally translated, “sampling rate = sound data capture rate. It will be a commission.”

In other words, the sample rate indicates how accurately the sound is captured.

For example, if the sampling rate is 48kHz, the data will be taken by dividing it by 48000 times per second.

The higher this value is, the finer the sound will be sampled.

It may be easier to understand if you think of the version where each block of pixel art is a sound.

splicing accessories-2022
2. Does the sound quality change depending on the sample rate?
Sampling rate loads sound quality
A common question is whether sound quality changes with frequency.

Of course it will change.

However, beginners and those without expensive equipment won’t notice much even if it changes.

Then I’ll show you how to check it!

Sample rate and bit rate

Recently, sample rates and bit rates have become even more common on music distribution sites and music players. Are there many people who don’t know what it’s like, even though they hear it often?

So this time, I will explain sample rate and bit rate in an easy to understand way!

Please note: to explain it in an easy to understand way, there are some parts that are strictly different. note that.

1. Reasons sample rate and bit rate exist in the first place
Sample rate and bit rate are terms that come up when digitizing and digitizing audio. Audio devices such as headphones and speakers typically transmit analog audio signals. Sampling rate and bit rate are terms that appear only in the digital case, so they have almost no relation to analog audio signals.

The reason that sample rates and bit rates appear in the digital world is that there are problems converting analog signals to digital. Analog signals are constantly changing and it is not possible to accurately record the changes.

That’s where sample rate and bit rate come into play. Records the value of a constantly changing analog signal at regular intervals. An audio file is a collection of records as a piece of data. By making this digital audio file, it can be handled on a computer or smartphone and exchanged as data.

By the way, the log is the analog audio data log. However, in the case of vinyl, the original sound cannot be accurately recorded and there will be some differences. The slight difference shows up as the discs own sound quality.

2. What is the sampling rate?
I wrote that by digitizing in 1., I will record how much the value is at regular intervals. If you explain the sample rate in one word, it means “how often you are recording”.

Sampling rate is often expressed in the base unit of “kHz (kilohertz)”. As you know, “kilometer” is the same as “kilometre” and “km”, which means 1000 times. On the other hand, “Hz” indicates how many times it vibrates per second. In the case of sampling frequency, there is no problem in understanding how many times a second is recorded.

That is, in the case of 48kHz, it is recorded 48,000 times per second.

Basically, the more you record during 1 second, the closer it is to an analog signal, so the sound quality will be higher. However, the amount of data will increase proportionally, so it needs to be balanced.

The “Nyquist frequency” is an index that determines the balance. To briefly explain what it looks like, half the sample rate is the loudest pitch that can be produced at that sample rate.

In other words, in the case of a sound source with a sampling frequency of 48kHz, it is possible to record sounds up to 24kHz, which is half, and it is not possible to record sounds higher than that.

Since humans can basically only recognize sounds up to 20 kHz, it doesn’t make much sense to increase the sample rate too much. However, recent research says that “I think I actually feel it in the part that is not my ear.” However, it is a difficult part because there is a limit to the high-pitched sound that audio equipment can produce.

2. What is a bit rate?
When it comes to how often the sample rate is recorded, the bit rate is an indication of how much value there is at the time. Bit rate is basically expressed as “bit”. Since it is a drive that appears on a personal computer, it can be an unpleasant drive for those who are not good with personal computers.

However, the bitrate is easy. Imagine the horizontal line on the graph. The bit rate is the precision with which this horizontal line is prepared. It means that 1000 pieces are prepared in 1 bit. I think this makes it easier to understand.

The value of how much represents the bit rate indicates the volume. In other words, the higher the bitrate, the higher the volume, and subtle changes in volume can be recorded. It’s okay if you remember a lot about bitrate.