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What does the quality of an mp3 depend on? high resolution mp3

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What does the quality of an mp3 depend on? high resolution mp3

Factors influencing hearing quality

High quality

Lately, very high quality audios have been promoted… are they really convenient?

We could say that if we strictly base ourselves on technical aspects, they could be considered of higher quality.

For example, they get to use sample rates of more than double the highest currently used.

The same happens with the bit rate, they use numbers that until now were not used at all.

Pewro first we must ask ourselves if the equipment we use to read them (the computer, a cell phone, an mp4 player) are capable of handling these qualities and if the speakers or headphones are also enabled and built to do the same.

Otherwise we will end up paying a lot for this super audio and effectively get the same.

It is worth additionally thinking about whether our ears could differentiate between one and the other.

To what extent our ear perceives the difference between 4800 and 96000 as a sample rate.

What we must avoid is falling victim to the “numbers”, which will show us that in theory they will sound better, but avoid touching reality – for example the human ear or the quality of our speakers – and therefore the theory ends up being misleading.

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What does the quality of an mp3 depend on?

Factors influencing hearing quality

digital sound

We must consider that we are talking about digital sound.
The audio as we hear it from Daria is an analog audio. This means that it is a continuum, there are no partitions, cuts, chunks, etc.

On the other hand, digital audio is made up of thousands of points that make up a curve, but the curve is not continuous but is made up of a series of points.

Of course, the more points that curve has, the smoother the curve is and the more similar it is to the initial analog audio.

When the CD was developed, the conclusion was reached to make 44100 shots per second, so that the curve was smooth enough and could contain the sounds in the range that the human ear can perceive them.

Because there are sounds that are too high-pitched that we cannot hear them and also others that are so serious that we cannot perceive them.

It is even known that as the years go by, a person can perceive very high-pitched sounds less, unlike adolescents who perceive such high-pitched sounds better.

So the first factor to take into account will be to have 44,100 or 48,000 samples per second, in order to have a smooth curve, with high quality.

Recordings with less than that sample rate are not of high quality. Sample rate is called the number of samples taken per second to delineate the sound curve well.

So you take a naudio file and make sure it has a sample rate of at least 44100 or 48000 frames per second to know it’s CD quality.

There are higher samplerates, for example, 96000 but we will talk about it later.

Mp4Gain is a software that manages these parameters perfectly. If you really want high quality sound, Mp4Gain is the right tool for you.

Samplerate, what is sample rate

The sampling frequency is the time that results from the time between two samples. It is given in samples per second (S / s).

The level of the sampling frequency is a criterion for the reproducibility of the frequency of the sampled signal. The closer the sampling times are, the better the signal can be reproduced.

Sampling rate

Relationship between frequency and sampling frequency

For example, if an analog signal is sampled once per millisecond (ms), the sample rate is 1 kHz and the sample rate is 1000 samples per second. If the sampled signal has a frequency of 1 kHz, the signal is sampled once per period. It cannot be played. If, on the other hand, the frequency of the signal is 100 Hz, the signal is sampled ten times with the same sample rate. Therefore, the signal is easily reproducible. Therefore, the sampling frequency must be in a certain relation to the frequency of the signal. This relationship is through the given sampling theorem. Accordingly, the reproduction of the signal requires a sampling frequency that is at least twice the frequency of the signal. This applies to sine-type signals for their 1st harmonic, but not to square wave or pulse signals.

Audio sampling frequencies

In the case of voice transmission over ISDN with a maximum frequency range of 4 kHz, the sampling frequency is 8 kHz, which corresponds to a sampling interval of 125 µs. For audio with a maximum frequency range of 20 kHz, the sampling frequency is 44.1 kHz (22.67 µs) and 48 kHz (20.83 µs). For high-quality multi-channel audio, the sample rate can be up to 192 kHz. Much higher values are found for video and HDTV. For digital video, this results in a 6.5 MHz bandwidth for the luminance signal, a sampling frequency of more than 13 MHz and a sampling interval of 74 ns. The sample rate for HDTV is even higher with 74 MHz and a sample rate of 13.5 ns.
In the case of pulse-shaped signals, the sampling frequency must be many times greater than its fundamental oscillation, since otherwise important pulse parameters cannot be determined. If the sample rate is many times higher than the theoretically required sample rate, we are talking about oversampling.

Everything you need to know about samples and bits

I started delving into depth and sample rate in my last mixing / mastering tutorial, and while we’re not necessarily digital audio engineers, some background on what bit depth and sample rate is good information for anyone. participate in digital music. It’s something you always work with, whether you know it or not, and it’s great background information for understanding whether understanding the building blocks of digital audio is critical for personal gain or just to be able to sound smart just in case. where the conversation never comes up.

Samplerate

So the first thing to understand is that bit depth and sample rate only exist in digital audio. In digital audio, bit depth describes amplitude (vertical axis) and sample rate describes frequency (horizontal axis). So when we increase the number of bits we are using we are increasing the amplitude resolution of our sound and by increasing the number of samples per second we are using we are increasing the frequency resolution of our sound.

In an analog system (and in nature), the audio is continuous and fluid. In a digital system, the smooth analog waveform is only approximated by the samples and must be set to a limited number of amplitude values. When you sample a sound, the audio is divided into small sections (samples) and these samples are fixed at one of the available amplitude levels. The process of fixing the signal to an amplitude level is called quantization, and the process of creating the sample slices is, of course, called sampling.

In the diagram below you can see a visualization of this where there is an organic sine wave playing for one second. It starts in 0 seconds and ends in 1 second. The blue bars represent the digital approximation of the sine wave where each bar is a sample and has been set to one of the available amplitude levels. (This diagram is obviously much grosser than in real life).

samplerate

This second of audio would have 44.1K, 48K, etc. samples. From left to right depending on the selected sample rate when recording and it will cover -144dB at 0dB at 24bit (or -96dB at 0dB at 16bit bit). The dynamic range resolution (the number of possible amplitude levels for the sample to rest) would be 65,536 at 16 bits, and get this, 16,777,216 when logged at 24 bits.

Therefore, increasing the bit depth greatly increases our amplitude resolution and dynamic range. What is not so obvious is where the increase in dynamic range occurs. The added dB is added to the softest part of the sound since the amplitude can never exceed 0 dB. What this does is allow you to hear more delicate sounds (like a reverb tail running at -130 dB) to be heard, which might otherwise be cut off to a 16-bit, -96 dB sample.