What does the quality of an mp3 depend on? high resolution mp3
high resolution mp3
Factors influencing hearing quality
high resolution mp3
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.
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.
What is the fundamental difference between 44100 and 48000 Hz?
44100 vs 48000 hz
In fact, this is just a question of long-standing standards.
44100 vs 48000 hz
44100 vs 48000 hz
44100 is the CD standard.
48000 is the standard for DVD.
The difference in practice is so small that it will be impossible to notice it (I’ll tell you more: many people feel the difference between mp3 and wav, but they can’t tell which is better).
The stereotype has persisted that if you need to work with TV or movies / soundtracks, it is better to do it in 48000, suddenly some old equipment will not understand sampling.
But this is very, very unlikely these days, so there isn’t much of a difference.
It can record at 96000. There is a small chance that some plug-ins / sound effects can handle such recordings better, but it requires more CPU / RAM and much more hard disk space.
Between 16 and 24 bits, it will also be difficult to feel the difference, but at the request of the sound engineer, we wrote in 24 with the same thoughts (for plug-ins).
In general, write to 44100 if you don’t need to work with a specific television crew.
44100 vs 48000 hz
Choosing the Right Sample Rate: 44100 or 48000 hz
In the world of digital audio, the choice between 44,100 Hz and 48,000 Hz sample rates is a critical one. As an audio expert, I’ve spent years diving deep into this topic, examining the real-world scenarios where this choice can make or break a sound. In this article, I’ll guide you through this audio journey, shedding light on the differences and helping you make an informed choice.
44100 Hz – The Analog Heartbeat
When we talk about 44,100 Hz, it’s like stepping into a cozy vinyl record shop, where the warm crackles and pops surround you. This sample rate mirrors the heartbeats of analog audio, capturing the subtleties of your audio source much like a vintage vinyl record player.
Imagine: You’re in a dimly lit jazz club, and a saxophonist takes the stage. You close your eyes as the music begins. 44,100 Hz is akin to capturing every breath, every emotion, and every nuance of the saxophonist’s performance. It’s the sample rate that preserves the soul of analog sound.
48000 Hz – The Digital Precision
Contrastingly, 48,000 Hz feels like entering a state-of-the-art recording studio with a digital mixing console at the heart of it all. It’s the precision tool for audio in the digital age, where every sound wave is charted with utmost accuracy.
Visualize: You’re in a high-tech laboratory, and a scientist is conducting a finely tuned experiment. 48,000 Hz is like the precise instruments that measure every data point with accuracy. It’s the sample rate that excels in capturing the clarity and detail of digital audio.
The Real-World Decision
The choice between 44,100 Hz and 48,000 Hz ultimately depends on the nature of your audio project.
Subtitle: For Vintage Vibes
If you’re aiming for a warm, nostalgic sound reminiscent of classic records, 44,100 Hz is your choice. It’s like using a vintage camera to capture that old-world charm. This sample rate will maintain the character and imperfections of your audio source.
Subtitle: For Contemporary Clarity
When you require crystal-clear audio for modern projects, such as podcasts, video games, or high-quality music production, 48,000 Hz is your ally. Think of it as upgrading to a high-definition TV for the audio world. This sample rate ensures every detail is captured and reproduced faithfully.
Last words about right sample rate for your digital audio
As an audio expert, my journey has led me to understand that the choice between 44,100 Hz and 48,000 Hz is about preserving the essence of your sound in the most appropriate way. Each sample rate has its place in the vast world of audio, just as a painter chooses different brushes for different strokes on their canvas.
So, whether you’re embracing the warmth of the past or striving for the precision of the future, remember that the right choice of sample rate can be the difference between an audio masterpiece and a missed opportunity. Choose wisely, and let your sound shine in all its glory.
The fundamental difference between them in the coverage of the frequency range on the track (from 20Hz), the 44100 sample rate allows you to work in the range up to 22kHz, 48000 to ~ 25kHz, 96000 to ~ 35kHz, etc. 48 parameters o 96kHz are used in large studios where the reproduction of these frequencies and sound engineers strive for the slightest increase in sound quality, before and after conversion to the 44100 standard, the sound of the track objectively looks better, even though the human ear does not hear these frequencies, the psychoacoustic effect remains (the closest example: if you shoot video and plan to play back in fHD, you will prefer to shoot 4k with rear cropping for the sake of image quality, and no one will say there is no point in shooting 4k, the same is here).
It’s even more interesting in movies … because 44100Hz is the playback frequency at 24fps and 48000Hz is 25fps. If you record a video at 25 fps and the sound is separately on the recorder at 44100Hz, then the length of the tracks will not match and you will have to change the timbre of the original with a small time interval.
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.
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.
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.
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).
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.
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 limited bandwidth (B) signal (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 the human being listens 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 sample rate of 88.2kHz was generally applied, since at the time of mastering, with the symmetrical re-sampling at “half”, it was at 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.
