What Is Audio Sampling Rate: A Comprehensive Explanation

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What Is Audio Sampling Rate: A Comprehensive Explanation

Introduction

Audio sampling rate is a fundamental concept in digital audio that refers to the number of samples per second used to represent an analog audio signal in digital form. In this article, we’ll explore the technical details of audio sampling rate, its importance in digital audio, and its impact on audio quality and file size.

Sampling Rate Fundamentals

The concept of audio sampling rate is based on the Nyquist-Shannon sampling theorem, which states that in order to accurately represent an analog signal in digital form, the sampling rate must be at least twice the highest frequency present in the signal. This means that a signal with a highest frequency of 20kHz (the upper limit of human hearing) must be sampled at a rate of at least 40kHz in order to be accurately represented.

Sampling rate is measured in Hertz (Hz), which refers to the number of samples per second. Common sampling rates in digital audio range from 44.1kHz (used in CDs) to 192kHz (used in some high-resolution audio formats).

Sample Rate Conversion

In some cases, it may be necessary to convert audio from one sampling rate to another. Sample rate conversion involves resampling the audio data to a different rate, which can be done using digital signal processing techniques. However, sample rate conversion can introduce artifacts and reduce audio quality, especially when downsampling from a higher rate to a lower rate.

There are various reasons why sample rate conversion may be necessary, such as when mixing audio tracks with different sampling rates, or when preparing audio for distribution on different platforms with varying requirements.

Audio Quality and Sampling Rate

The sampling rate has a significant impact on audio quality, with higher sampling rates generally resulting in better fidelity and more accurate representation of the original signal. However, the benefits of higher sampling rates are limited by the limitations of human hearing and the practical limitations of digital audio technology.

While there is debate about the benefits of “high-resolution audio” formats with sampling rates above 44.1kHz, it is generally accepted that sampling rates above 96kHz provide little additional benefit in terms of audio quality.

Bit Depth and Sampling Rate

The bit depth of an audio sample refers to the number of bits used to represent the amplitude of the signal at each sample point. Higher bit depths allow for more precise representation of the signal, but also result in larger file sizes. The bit depth and sampling rate are related, as increasing the bit depth requires more data to be stored for each sample.

There is a trade-off between sampling rate and bit depth, as higher sampling rates require more data to be stored per second, which can limit the maximum bit depth that can be used without exceeding practical file size limits. However, this trade-off can be mitigated by using efficient audio compression techniques.

Sample Rate in Practice

Common sampling rates in digital audio include 44.1kHz (used in CDs), 48kHz (used in digital video), 88.2kHz, 96kHz, 176.4kHz, and 192kHz. Streaming services such as Spotify and Apple Music typically use lower sampling rates for their audio streams, with 44.1kHz being a common choice.

The Nyquist Theorem, named after the Swedish-American physicist Harry Nyquist, states that the sampling rate should be at least twice the highest frequency component in the signal being sampled. This is why the standard CD quality sampling rate is 44.1 kHz, which is just above the upper limit of human hearing.

However, it is important to note that there are higher sampling rates available, such as 48 kHz, 96 kHz, and even 192 kHz. These higher sampling rates can provide more detail and accuracy in the digital representation of the analog signal. However, they also require more storage space and processing power.

Another important factor to consider is the bit depth, which is the number of bits used to represent each sample. The more bits used, the more accurate and detailed the representation of the analog signal. CD quality uses a bit depth of 16 bits, but higher bit depths such as 24 bits are also available.

It is worth noting that some argue that higher sampling rates and bit depths may not necessarily result in audible improvements in sound quality, especially when considering the limitations of human hearing. Additionally, some argue that the increased storage and processing requirements may not be worth the potential improvements.

In conclusion, the sampling rate is a crucial component in the digital representation of analog audio signals. A higher sampling rate can provide more detail and accuracy in the digital representation, but also requires more storage and processing power. The Nyquist Theorem provides a guideline for choosing the appropriate sampling rate based on the highest frequency component in the signal. Additionally, the bit depth is another factor to consider in the accuracy and detail of the digital representation. While higher sampling rates and bit depths are available, the potential improvements in sound quality must be balanced against the increased storage and processing requirements.

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The higher the bitrate, the higher the sound quality and the larger the file size.

but the quality of the source file determines the final quality.

From highest to lowest, the sound quality will be worse, but from lowest to highest, the sound quality will remain unchanged at most, but the file will be larger.Many

General mp3 are good with bit rate around 128, and also 3-4 BM in size.

The bitrate, choosing it, directly affects the size of your mp3 file and the listening experience. High compression ratio has high distortion, and low compression ratio has low distortion, but how do we find a balance point that we can accept on both counts? This requires careful exploration in the experiment. Considering that the sound quality of low bitrate files is not suitable for playing music, the minimum set is 128kbps, and four fixed bitrate files of 128, 192, 256 and 320 are used for comparison. and try.

The compression ratio of 128 kbps is still relatively rough, and the high-frequency part is highly distorted after compression. It sounds hollow, wrinkled, rough, and there are often flickering sounds. Misunderstanding, the compressed volume of a 3 minute 39 piece of music is 3414 Kb. Although the volume is not large, the sound is not satisfactory, and there are relatively large defects.

192kbps bit rate compression effect is much better than 128. First of all, the sound is solid, at least there is no empty feeling, the high-frequency distortion is also much less, the sound is compact, the noise is small and clean, and achieve relatively ideal listening The sound effect, just because the compression is still relatively strong, the detail performance is still not very good, the texture of musical instruments, especially instruments of wind, it is still very hard, unreal and lacks musicality. The compressed size is 5123kb, and I think the compression ratio is 128~ It is better to use it in a mp3 player with a capacity of ~256m, which can not only satisfy the basic sense of hearing, but also is suitable in size.128m can store about 95 minutes of music, and 256m can double to 190 minutes of music.

The 256 kbps compression rate is naturally a step higher than 192 in terms of sound quality. Take the first 10 seconds of the track, the low frequency of the cello is obviously less grainy, and the sound is more smooth and natural, with texture and texture. It is also clearer, with much more detail, the rendering of the atmosphere is more prominent, the rotation of parts in the following songs is also more expressive, the clarity of large and small signals is also improved, and the sound is more detailed and lasting. But at the same time, the file size has also increased to 6831kb, which is still affordable for a 256m mp3 player. It is not difficult to know by calculation. According to the bit rate of 256, about 135 minutes of music can be stored. Generally speaking, it is enough, 128m is a bit less and can only support a little over an hour, so it is recommended to use 192 bitrate for 128m.

320 kbps is the maximum bitrate that lame can provide. The final file generated is 8592kb which is about 8.4M. Compared to the 37M of the wav file the compression ratio is basically 4.5:1 but the generated mp3 file sounds very distorted Now on Compared with other 320 bit rate, the natural advantage is obvious, the tone, details, etc. are very delicate, basically achieve the sound quality of the original CD copy, especially in the CD player with playback function from mp3, the basic No difference, but I use relatively high-end earplugs with high resolution, plus my experience and skill with music and equipment, I can still hear a lot of differences compared to wav files, first Instead, the compressed mp3 sounds a bit The crunch feeling is relatively dry on the whole. Without the wav file, it sounds fresh and dynamic. In terms of final details, nuances and sense of space, the separation is not as high as the quality of the wav file, but it is quite close in terms of timbre, but the performance is poor and the digital flavor is relatively strong. So if you are using a miniature hard drive player like an iPod, I recommend you use 320kbps compression ratio, which can get the best listening experience. Of course listening to wav directly is the best~

The bit rate directly affects the sound quality.

High bitrate is good and low bitrate is bad.

The code rate is the number of data bits transmitted per unit of time during data transmission. Generally, the unit we use is kbps, that is, kilobits per second.

The popular understanding is the sampling rate. The higher the sampling rate per unit time, the higher the precision, and the processed file is closer to the original file, but the file size is proportional to the sampling rate, so almost all encoding formats pay attention. It’s about how to use the lowest code rate to achieve the least distortion. The cbr (fixed code rate) and vbr (variable code rate) derived from this core are all articles in this regard, but things are not absolute, in terms of audio, the higher the bit rate, the lower the compressed ratio, the smaller the sound quality loss and the closer it is to the sound quality of the audio source.
The information in the computer is represented by binary 0 and 1, and each 0 or 1 is called a bit, which is represented by lowercase b, that is, bit (bit); uppercase B represents byte, ie byte, one byte = Eight bits, ie 1B=8b; the capital K in front stands for thousand, that is, thousand bits (Kb) or kilobytes (KB). Indicates the size of the file, usually using bytes (KB) to indicate the size of the file.

Kbps: The first thing to understand is that ps refers to /s, which is every second. Kbps refers to the speed of the network, that is, how many thousands of bits of information are transmitted per second (K means thousands of bits, Kb means how many thousands of bits), it is expressed in kb (kilobit), and in the case KBps means how many kilobytes are transferred per second. 1KBps = 8Kbps. The Internet speed of ADSL is 512 Kbps. If converted to bytes, it is 512/8 = 64 KBps (that is, 64 kilobytes per second).

A frame is a still image, and continuous frames form an animation, like a television image.
We normally say the number of frames. Simply put, it is the number of image frames transmitted in 1 second. It can also be understood that the graphics processor can update several times per second, usually expressed in fps (Frames Per Second). Each frame is a still image, and showing frames in rapid succession creates the illusion of movement. Higher frame rates result in smoother, more realistic animations. The more frames per second (fps), the smoother the motion is displayed.

What is the bitrate of the music?
It can also be called bit rate, which is nothing more than the amount of data reproduced per second by a type of music, the unit is expressed in bits, that is, binary bits. bps is the bit rate. b is bit, s is second, p is per, and one byte is equal to 8 binary bits. That is, the file size of a 4-minute song at 128bps is calculated as (128/8)*4*60=3840kB=3.8MB, which means that the same song with the same bit rate (bps) will not no matter what format (such as mp3 wma) The capacity is basically the same, which can only represent a transmission rate, not the sound quality. Due to different compression engines, the sound quality of different formats varies a lot. However, for the same format, the higher the bitrate, the larger the file and the better the sound quality.

What is the sample rate of the music?
Sampling rate refers to the number of samples per unit of time. The sampling rate is 44KHz, which means the number of samples per second is 44K, which means that 44,000 pieces of data are used to describe the sound waveform in 1 second. That is, the higher the sample rate, the better the sound quality. But he and bitrate are two completely different concepts.

Some details of the sample rate

For many years it was thought that the sample rate or sampling frequency did not decisively influence the final quality of the digital audio; There are currently several engineers who record in 44.1K or 48K without really knowing why they do it. With the advent of new and better computers, interfaces, ports and protocols, 88.2K, 96K and up to 192K entered the discussion table on the best sample rate to use. It has always been the subject of discussion between engineers and audiophiles; some argued that they did hear the difference between different sample rates and others that did not, and the topic has been subjected to millions of A / B tests with very high quality equipment, causing all kinds of opinions found and uncompromising, fights and friendships of years broken

While this is a basic issue of digital audio, it is always surrounded by a halo of mystery, mysticism and magic (like every sound theme), which is well worth clarifying.

What is the sample rate?

This topic, although it occurs in the first or second class of digital audio, is not always understood correctly. In scholastic thinking, sample rate is defined as the amount of audio samples transported and taken per second. Since this is a unit of measurement over a second and with events that occur cyclically, the Hertz (1 / Frequency) is used as a unit. Obviously we cannot talk about this subject without referring to the Nyquist sampling theorem, which was tested by Shannon almost twenty years after its publication and in which it is stated that for a signal of limited bandwidth (B) (for example, a vibraphone reaches 14.917Hz), the sampling frequency must be twice its bandwidth (2 * B). Then, taking the previous example, we can say that: 2 * B → 2 * 14.917Hz → The sampling frequency for 14.917Hz should be 29.834Hz. This would be equivalent to 29,834 samples per second (1/29, 834) to be able to regenerate the signal of a vibraphone without error. Hence, it is taken that the highest frequency that human beings listen to is 20kHz and if we apply Nyquist it should be 40kHz, but it takes 44.1kHz to meet the demanding ears and for a matter of multiples.

44.1K or 48K to 88.2K or 96K, the correct division

At the dawn of the digital audio era, Nyquist was used to use the sampling resolution of 44.1K, used at that time audio CD format that played at 16bit / 44.1kHz. With the advent of DVD and Blu Ray as video and audio formats, resolutions such as 24Bits / 48K or 24Bits / 96kHz began to be used. Although for many years there were recordings that were made in 24Bits / 88.2kHz or 24Bits / 96kHz, at a certain time of mastering, before sending it to the disk duplicator, the audio suffered a mutilation that reduced it to 16Bits / 44.1kHz as It was ordered by the CD format. This process should be carried out with equipment specially designed for this function and in stages so that the audio did not suffer a very noticeable cut and the bad conversion was evidenced. Although the old and dear Dither was applied since then to compensate for this process (something like “grain” in the cinema. Watch a film without “grain” and it will look like HD even though it was filmed in 1980 on tape and goes to notice until the makeup of the actor and the assembly of the special effects, something otherwise disagreeable).

Generally, to prevent the audio from mutilating or applying several conversions that degrade it, it was decided at what resolution to record before pressing the REC button (we will not mention those that come down directly with your DAW from 24Bits / 96kHz to 16Bits / 44.1kHz in one step to export the audio … there is a place reserved especially for them in hell). If the audio was going to end on CD, a 88.2kHz sample rate was generally applied, since at the time of mastering, with the symmetric re-sampling at “half”, it was 44.1kHz.

Sounds better?

The subjective point of this is that we expect recordings to “sound” better at a higher sample rate. The reality is that if we record in high sample rates, with very good sampling, our sound will not “sound better”, but will be more detailed. Obviously, if our sound source is bad, our microphones and preamps too and so on, no matter how much we record at 192K, the result will not be the best. Now, if we use a good sound source, good audio chain and a good converter, everything will be obviously good. But don’t confuse; We are talking about detail here, not if it will sound more “warm,” “fat,” or “full-bodied.” This translates into a more homogeneous capture of the entire frequency spectrum, both audible and non-audible.

CPU, disk and plug-ins

Obviously, having a higher sample rate means that our processor must do more calculations, since it has to process more samples (or audio samples). Depending on the amount of plug-ins that we use before a multitrack in high resolution, our use of both DSP and native processors (the computer equipment), will increase significantly, making it very difficult or impossible to work. There are several options to overcome this problem, from buying more processor or DSP, using fewer processes or external equipment (hybrid mixing), to borrowing a machine. The only option that should never go through our minds is to lower the resolution of the audio, process and upload it again. The serious problem that comes with this is a cut in the audio, which is not reversible and what is limited and trimmed, so it stays.

Another aspect to consider is that the storage speed must be in accordance with the audio resolution we use. Suppose we want to record at 24Bits / 96kHz; The transfer rate would be: 2304kbits / second. Now, calculating the amount of tracks, we should use a disc that really reaches us in speed for this transfer rate (topic to be developed in another article).

In these times, storage size is not a problem, but speed is. Having three terabyte disk drives are generally used for 5400 rpm dish disks; the least that should be used if they are not solid state disks, would be 7200 rpm plate disc drives. Obviously, with 5400 rpm discs, we would have a third reduction in the final transfer speed and reading and writing possibilities called “iops” (in out per second or in and out per second), which have a certain number, depending on the disk, capacity and arrangement of the same (RAID) which, depending on how much we demand in the resolution of the audio, amount of channels, processing (plug-ins) and expected latency (if we record with real-time monitoring), we will surely face some problems like “clicks” and / or “pops” in our audio.

Clock

The importance of using a good clock (or clock) and being in sync with all the elements that belong to our audio chain is vital. Recall that a few articles ago we have exposed this topic in detail, but it should be reinforced in this article. Several ADC and DAC converters of economic interfaces do not perform sampling and quantization in the correct or expected manner; External clocks or protocols such as Dante help the synchronization between several devices to be correct and improve the audio quality. Much of the final quality of our work in audio is in this part of the process and it is important that if we take our work and passion seriously, we begin to pay attention to these kinds of details that are generally overlooked.