Tag: does sample rate matter

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The higher the frequency and bitrate at the time of recording, the better the sound?

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Is PC recording okay?

You know that the audio interface used for PC recording (hereinafter referred to as audio IF) has a notation like “24bit-96kHz”. It is a notation that corresponds to the production of the so-called high-resolution sound source.

* ↑ It seems that 24bit-192kHz is mainstream these days.

This number is significant and can have an effect during recording. Isn’t it used like “Isn’t it better to have a bigger number?”

I will tell you the idea of ​​frequency and bit rate, and what kind of settings you should do if you want to make a sound source with high sound quality. Let’s optimize the recording settings for home recording!

Index Index [ Close ]

What are bit rates and frequencies in recording?
What is the effect of frequency and bit rate?
Optimal frequency and bit rate for singing and recording
High bit rate/high frequency recording is recommended even when making a CD
latest

What are bit rates and frequencies in recording?

Bit rate and frequency
24bit-192kHz

24 bits is called the “bit rate” and 192 kHz is called the “frequency”. Once this is determined, the bitrate of the final sound source will be determined. The bit rate is the amount of information. It’s easy to assume that the higher the number, the better the sound, but goodness of sound = high bitrate doesn’t always add up.

Most audio IFs are initially set to “16 bit, 44.1 kHz”. So unless you change the setting, it doesn’t record at high speed like “24bit-192kHz”. I think there are many cases where I don’t really care about this area.

“16 bit, 44.1 kHz” is the speed of the CD. If you don’t generate a wave file at this speed, you won’t be able to burn it to a CD. It is the most hassle-free fare to drive normally. I also record at CD rate for a simple recording.

So if you increase the overall bitrate, will the sound quality improve?

What is the effect of frequency and bit rate?

Manipulating the sound quality during recording will increase the amount of information. How does increasing the amount of information affect the sound by adjusting the frequency and bit rate? Let’s look at the effects of frequency and bitrate.

Frequency (sampling rate)
Increasing the frequency will make the sound softer. This is because there are more points to pick up the sound.

The upper limit of the frequency that can be recorded differs approximately twice between 44.1 kHz and 96 kHz. You will be able to record high-frequency sounds well. The sound isn’t twice as good, but when recording acoustic instruments, 96kHz is said to be able to record more treble.

Bit rate
Increasing the number of bits will make the volume of the sound clearer.

Expressing the relationship between frequency and number of bits as above, “So the higher the better, the better!”. However, not many people can distinguish between a sound source made with “24bit-192kHz” and a sound source made with “16bit-96kHz”. This is because it reproduces even the parts that are inaudible to the human ear. That’s why some people say that they can’t tell the difference even if they listen to music on a device that has high resolution. I’m not bad at listening.

Recommended DTM Books
There is a book that carefully explains bitrate/frequency and the points that beginners can easily trip over with DTM. I also try to open this book as soon as I have trouble recording. It’s a book by Ken Fujimoto, famous in the DTM world, so you can rest easy ^^

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Frequency used for audio (sample rate, PCM, DSD, etc.)

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

Recently, the number of Hi-Res Audio sources has increased and the frequency is written as 192 KHz or 11.2 MHz. What is this frequency?

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

1. 1. What is the sample rate?

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

You often hear that the sample rate of a CD is 44.1 KHz. Since digital signals are basically 0 or 1, reproducing down to the 20 KHz limit that the human ear can hear requires a resolution of twice that frequency. In addition, the frequency was decided to be 44.1 KHz considering the digital signal processing margin. Since a music signal is a set of sine waves, they are 44.1 KHz which can be shaken at a maximum frequency of 20 KHz.

2. What are 16-bit and 24-bit?

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

Converting this to dB is 20LOG (65536), which is approximately 96 dB. The dynamic range of a CD (the difference between soft and loud sounds) is 96 dB.

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

3. 3. PCM format

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

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

In fact, let’s look at the output of COMBO384.

This is a signal called LRCLK (or FSCLK) where the yellow is changing 2CH (L/R), and a set of LRs are sent every 44.1 KHz.

The blue color below is divided into 32 bits by the DATA line, and the DATA L and R are output.

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.

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…