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.
Discretization (discretization frequency – ing.) – transcoding an analog signal into digital by reading the characteristics of the signal at a given moment and converting it into a digital data matrix (approx. 100010110).
The sampling rate is a parameter that allows you to know the number of calls to an analog (or digital) signal in a given period of time (usually one second), to record frequencies in digital form or to convert to an analog signal.
If we rely on Kotelnikov’s theorem, then to record a lossless signal, a sample rate is required that is two or more times greater than the maximum sound frequency of the played track. That is, in theory 44,100 Hz will be sufficient for most recordings, which is more than 2 times higher than the threshold for human audible frequencies, but this is not entirely true.
The higher the sampling frequency, the more accurately the sound will be reproduced in an analog or digital signal. However, the more conversions made from analog to digital and vice versa, the more the precision and quality of the original signal recording will be lost.
The maximum sample rate for 2010 was 2,822,400 Hz and was compliant with the Super Audio CD (SACD) standard. Most multimedia centers, home theater systems have DACs (digital-to-analog converters) and ADCs (analog-to-digital converters) with a sample rate of 192,000 Hz.
To convert a signal into analog, special chips are used: DACs (digital to analog converters). To convert the signal to digital, ADCs (analog to digital converters) are used.
These microchips and chipsets have a variety of characteristics other than sample rate, such as THD, the amount of interference introduced by the transformation, the number of possible false errors, no saving a digital signal, and so on.
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.
To convert a so-called analog audio to digital, we use a process called: sampling. Sampling is done on a converter (or sound card). The principle is to take regular snapshots, which are the measurements of the analog signal voltage, and transform them into digital data whose language is numbers (numbers).
Here is a diagram representing the samples included in the amplitude of a sound wave. The number of samples in this wave defines the sampling frequency or sampling frequency.
Sampling frequency
The sampling rate is expressed in hertz (Hz) or (kHz). The following values are commonly found: 44,100 Hz, 48,000 Hz, 96,000 Hz, 192,000 Hz. The CD and the digital world standard are 44,100 Hz. This means that for every second, there are 44,100 samples. (samples) reproduced. The higher the sample rate (number of “snapshots” of the audio taken per second), the more accurate the analysis and coding of the music in digital data. The sampling rate affects the audio frequency range from the lowest to the highest pitch that can be stored.
Sampling frequency
16-bit / 44.1 kHz coding was the best quality available when the CD was released in the early 1980s, but things have changed, and we can now record and distribute music at higher bit-depth levels and sample rates. These formats have been used in studio recordings and for mastering for many years.
High-resolution audio (HRA) matches any recording format above the 16-bit / 44.1 kHz standard for CDs, and HRA recordings usually use 24-bit encoding, providing a greater dynamic range than CD and sampling rates up to 192 kHz . This is the pinnacle of HRA business records. First and foremost, it’s about getting as close as possible to the sound heard in the studio.
Which sampling rates should you choose?
In order to capture the smallest details at high frequencies, we need to try more frequently. The way it works is that a given sampling rate can accurately detect audio frequencies down to just under half its value. For example, a sample rate of 48 kHz can accurately detect audio frequencies as low as just below 24 kHz. This limit for half the sampling frequency is called the Nyquist frequency and is named after one of the engineers who developed the calculation behind the sampling principle.
The human ear can generally hear in the following spectrum: 20 Hz – 20,000 Hz. As we have just seen, for no obvious loss, the sampling rate must be at least twice as high as the maximum frequency contained in the audio when digitizing. The sampling rate must be at least 40,000 Hz for a correct result for our ears.
This is why 44 100 Hz resolution is the most widely used because it allows us to cover the spectrum up to 22 050 Hz. We even benefit from a small margin because we could have rounded up to 40,000 Hz, but it also means that if you export your music at a sampling rate higher than 44,100 Hz, your ear can’t hear the difference.
Anti alias filters
The first thing an analogue to digital converter does to analogue audio before sampling is to filter all frequencies above the Nyquist limit of the desired sampling frequency. If not filtered, all frequencies above Nyquist are injected again into the sample. This is called an alias effect.
Fortunately, almost all converters on the market today have implemented high-quality anti-aliasing filters. As a result, it seems undesirable aliasing effects are not, and all frequencies below the Nyquist recorded accurately. In most cases, as long as you use a good quality converter and a sampling rate of at least 44.1 Khz, you can record all frequencies in the area of human hearing in an orderly manner. Since the analog to digital converter measures each sample, you have to assign a number to that sample, as that is what makes it digital instead of analog.
How about sound cards up to 192,000 Hz?
There are two benefits to working at a very high frequency:
The first is that the drivers for your sound card (especially professional converters) will be optimized for a given sampling rate. In general, the ASIO drivers for your drives are optimized to the maximum sample rate it offers: 96,000 Hz and 192,000 Hz in most cases. This may be surprising, but it will have less delay and more relief for the microprocessor with a higher 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 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.
