Resampling Audio: Concepts and Methods


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Resampling Audio: Concepts and Methods

Resampling Audio
Resampling Audio
Resampling Audio
Resampling Audio

Introduction

Resampling is the process of changing the sample rate of an audio signal. This can be done to increase or decrease the playback speed of the signal, or to convert it from one sample rate to another.

In this article, I will discuss the concepts and methods of resampling audio. I will also provide some tips on how to resample audio effectively.

What is Resampling?

Resampling is the process of changing the number of samples per second in an audio signal. The sample rate is the number of times per second that an audio signal is sampled. For example, a CD-quality audio signal has a sample rate of 44,100 samples per second.

When you resample an audio signal, you are changing the number of samples per second. This can be done to increase or decrease the playback speed of the signal, or to convert it from one sample rate to another.

Why Resample Audio?

There are a number of reasons why you might want to resample audio. For example, you might want to:

Increase or decrease the playback speed of an audio file.
Convert an audio file from one sample rate to another.
Optimize an audio file for playback on a specific device.
Remove unwanted noise from an audio file.
How Does Resampling Work?

Resampling works by inserting or removing samples from the audio signal. When samples are inserted, the playback speed of the signal is increased. When samples are removed, the playback speed of the signal is decreased.

There are two main types of resampling:

Linear resampling is the most common type of resampling. It works by inserting or removing samples in a linear fashion. This means that the spacing between samples is constant throughout the audio signal.
Non-linear resampling is a more sophisticated type of resampling. It works by inserting or removing samples in a non-linear fashion. This means that the spacing between samples is not constant throughout the audio signal.
Which Type of Resampling Should I Use?

The type of resampling that you should use depends on the specific application. For most applications, linear resampling is sufficient. However, if you need to preserve the quality of the audio signal, then you should use non-linear resampling.

How to Resample Audio

There are a number of software applications that can be used to resample audio. Some popular examples include Audacity, Adobe Audition, and FL Studio.

Resampling Tips

Here are a few tips for resampling audio:

Use a high-quality resampling algorithm. This will help to preserve the quality of the audio signal.
Set the sample rate of the output file to the same sample rate as the input file. This will avoid any changes in the playback speed of the signal.
Use a high-quality audio converter. This will help to ensure that the resampled audio signal is of the highest quality.
Final Words About Resampling Audio

Resampling is a powerful tool that can be used to change the sample rate of an audio signal. It can be used to increase or decrease the playback speed of an audio file, to convert an audio file from one sample rate to another, or to optimize an audio file for playback on a specific device.

When resampling audio, it is important to use a high-quality resampling algorithm and to set the sample rate of the output file to the same sample rate as the input file. This will help to preserve the quality of the audio signal.

Resampling Audio for Beginners

Resampling audio can be a daunting task for beginners. However, it is not as difficult as it seems. Here are a few tips to help you get started:

Start by using a simple resampling algorithm. There are many free and open-source resampling algorithms available online.
Set the sample rate of the output file to the same sample rate as the input file. This will avoid any changes in the playback speed of the signal.
Use a high-quality audio converter. This will help to ensure that the resampled audio signal is of the highest quality.
With a little practice, you will be able to resample audio like a pro!

Conclusion

Resampling audio is a powerful tool that can be used to change the sample rate of an audio signal. It can be used to increase or decrease the playback speed of an audio file, to convert an audio file from one sample rate to another, or to optimize an audio file for playback on a specific device.


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Why upsampling? Part 1

Why upsampling? Part 1

Upsamplin

When it comes to improving digital sound quality, experts in this field agree on only one thing: As the sample rate increases, the sound quality improves dramatically.

Upsampling

Also, under the word “improve”, everyone already understands something of their own. All the variety of opinions on this topic boils down to the following: the sound becomes clearer, softer, more natural, the low frequencies are perceived more clearly.

However, these nuances are only noticed by listeners trained with a good ear for music on specially selected sound material and using technically advanced equipment.

There are many hypotheses that explain why sound quality is improved by sampling. Many technicians are inclined to believe that this relationship is due to distortions that arise from filtering and interpolation during reconstruction of the audio signal.

On a modern technical level, high-quality interpolators may be practically impossible to implement, therefore instead of improving them, manufacturers simply increase the sampling rate. Maybe it’s not about them at all.

Another version, which many music lovers adhere to, is that at a low sampling frequency, for example 44100 Hz, digital sound is completely devoid of nuances of high sounds, the main frequencies of which are above 7 kHz. , and at lower frequencies there are too few nuances for high quality. perception of music.

In fact, many musical instruments generate vibrations of up to 100 kHz. It is true that the proportion of energy that falls in the frequency band above 20 kHz is 0.01 to 2% for sounds of a harmonic nature and 0.02 to 68% for sounds created by a cymbal, triangle or hitting a drum rim. metal (hoop shot – Editor’s note).

Even the frequency range of speech in hissing-hissing sounds extends up to 40 kHz. Supporters of this version are not ashamed that a person cannot perceive sounds with a frequency higher than 20 kHz. Ultrasound is assumed to be perceived bypassing the auditory system, for example, through bone conduction.

Rumors that harmonics above 20 kHz contribute significantly to sounding have led to the creation and widespread introduction of analog-to-digital converters using 96 kHz and 192 kHz sample rates; The sampling frequency is expected to increase to 384 kHz.

Based on modern knowledge of human perception of sound, it must be assumed that the relationship between digital sound quality and sampling frequency is due to the transformation of the quantization error spectrum in the audio frequency range.

In the technical literature, this topic is considered only for a particular mathematical model, when music is represented by a signal with a uniform distribution in level and frequency. In this case, the quantization errors are converted to noise with a uniform spectral density from 0 Hz to the Nyquist frequency.

For every doubling of the sampling frequency, the spectral noise density is reduced by half and the signal-to-noise ratio increases by 3 dB. Since the pressure resolution limit is approximately 1 dB, these decibels are unlikely to have a noticeable effect on sound perception in the high-frequency region. Based on these numbers, it is absolutely impossible to draw tentative conclusions about the change in sound quality.

In order to relate the spectrum of quantization errors, sampling frequency and sound quality, in this article it is proposed to use a tonal signal as a music model, as is usual to evaluate the quality of sound paths. This approach is largely based on materials published in the “Sound Engineer” magazine.

The results can be summarized as follows. Unlike analog audio, digital audio is the product of amplitude modulation. This is manifested in a rigid functional dependence of the frequency multiplicity factor quantization error spectrum of the audio signal frequency F and the sampling frequency fs, represented as the ratio of the prime numbers y and x (k = fs / F = y / x). The frequency spectrum of quantization errors is always discrete and is uniquely determined by the multiplicity factor; the components of this spectrum are also uniquely determined by the amplitude of the audio signal, expressed in quanta. This means that the mechanism for the formation of the quantization error spectrum does not depend on the number of bits used. With an increase in quantization bit depth.

Why upsampling?

Why upsampling?

Upsampling

When it comes to improving digital sound quality, experts in this field agree on only one thing: with an increase in sample rate, sound quality improves dramatically.

UPSAMPLING

Why upsampling?
When it comes to improving digital sound quality, experts in this field agree on only one thing: As the sample rate increases, the sound quality improves dramatically. Also, under the word “improvement”, everyone already understands something of their own. All the variety of opinions on this topic boils down to the following: the sound becomes clearer, softer, more natural, the low frequencies are perceived more clearly.

However, these nuances are only noticed by listeners trained with a good ear for music on specially selected sound material and using technically advanced equipment.

There are many hypotheses that explain why sound quality is improved by sampling. Many technicians are inclined to believe that this relationship is due to distortions that arise from filtering and interpolation during reconstruction of the audio signal.

On a modern technical level, high-quality interpolators may be practically impossible to implement, therefore instead of improving them, manufacturers simply increase the sampling rate. Maybe it’s not about them at all.

Another version, which many music lovers adhere to, is that at a low sampling frequency, for example 44100 Hz, digital sound is completely devoid of nuances of high sounds, the main frequencies of which are above 7 kHz. , and at lower frequencies there are too few nuances for high quality. perception of music.

In fact, many musical instruments generate vibrations of up to 100 kHz. It is true that the proportion of energy that falls in the frequency band above 20 kHz is 0.01 to 2% for sounds of a harmonic nature and 0.02 to 68% for sounds created by a cymbal, triangle or striking the metal edge of a drum (hoop shot – Editor’s note).

Even the frequency range of speech in hissing-hissing sounds extends up to 40 kHz. Supporters of this version are not ashamed that a person cannot perceive sounds with a frequency higher than 20 kHz. Ultrasound is assumed to be perceived bypassing the auditory system, for example, through bone conduction.

Rumors that harmonics above 20 kHz contribute significantly to sounding have led to the creation and widespread introduction of analog-to-digital converters using 96 kHz and 192 kHz sample rates; The sampling frequency is expected to increase to 384 kHz.

Based on modern knowledge of human perception of sound, it must be assumed that the relationship between digital sound quality and sampling frequency is due to the transformation of the quantization error spectrum in the audio frequency range.

In the technical literature, this topic is considered only for a particular mathematical model, when music is represented by a signal with a uniform distribution in level and frequency. In this case, the quantization errors are converted to noise with a uniform spectral density from 0 Hz to the Nyquist frequency.

With each doubling of the sampling frequency, the spectral density of the noise is reduced by half and the signal-to-noise ratio increases by 3 dB. Since the pressure resolution limit is approximately 1 dB, these decibels are unlikely to have a noticeable effect on sound perception in the high-frequency region. Based on these numbers, it is absolutely impossible to draw tentative conclusions about the change in sound quality.

In order to relate the spectrum of quantization errors, sampling frequency and sound quality, in this article it is proposed to use a tonal signal as a music model, as is usual to evaluate the quality of sound paths. This approach is largely based on materials published in the “Sound Engineer” magazine.

The results can be summarized as follows. Unlike analog audio, digital audio is the product of amplitude modulation. This is manifested in a rigid functional dependence of the quantization error spectrum of the frequency multiplicity factor of the audio signal F and the sampling frequency fs, represented as the ratio of the prime numbers y and x (k = fs / F = y / x). The frequency spectrum of quantization errors is always discrete and is uniquely determined by the multiplicity factor; the components of this spectrum are also uniquely determined by the amplitude of the audio signal.