Category: Mp3 Normalizer

Sound formats and audio normalization


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WAV: It is the “pure” sound format, without any compression. Its weight is huge, as is its quality. Only recommended for professional works or to edit the audio before transferring it to a format with compression.
MP3: We’ve talked about him in the previous pages. Without a doubt, it is the most popular and widespread format. His appearance changed the way we listen to music.
OGG: It is the audio format of GNU / Linux, the free software MP3 version. It has all the virtues of MP3 (and more), but not all portable players can use it, but it is getting more and more.
WMA: Microsoft format, your own version of the MP3. It compresses quite well, but it is not as widespread as the MP3. Nor can all portable players use it.
MID: It is the audio format also known as MIDI (Musical Instrument Digital Interface). It is the only format that can not play more than music simply because what it contains inside are not sounds. Simplifying, it contains a series of instructions for special software included in all systems, a kind of digital synthesizer that can generate sounds like those of many musical instruments. The MID has inside what notes they have to sound and with what instruments: a score.

It is important to clarify the distinction between audio format and audio codec. The codec encodes and decodes the audio data while this data is archived in a file that has a specific audio format.

Most of the formats listed below are container formats, formats that group different types of data. Most of these container formats have only one codec associated, next to which metadata is stored. However, there are formats that group audio and video data produced by different codecs. Some of these container formats that group different types of data are: MP4, Ogg, WAV, QuickTime Format, AVI.

In this article we talk about audio formats, but we are really discussing the properties of the codec associated with the format.

When classifying audio formats we can distinguish three large groups.

No data compression: These are real sound waves that have been captured and converted to digital format without further processing. As a result, uncompressed audio files tend to be the most accurate.
With compression, without loss of data: Compression algorithms are used to reduce file sizes; It basically works by eliminating redundancy.
With compression and data loss: It is a form of compression that loses data during the compression process. In the context of audio, that means sacrificing quality and fidelity to decrease file size. The good news is that, in most cases, we will not notice the difference when listening.

volume booster

Compression

Compression is a process that involves reducing the dynamic range of an audio signal.

An apparatus, called a compressor, analyzes the gain of the input signal and, according to certain parameters set, those parts that exceed a level or threshold determined according to the desired configuration are attenuated.

In principle, compression is perceived a decrease in overall volume; In fact, this is because the compressor reduces the gain of the “peaks”, that is, of the parts that accumulate greater sound energy.

However, several very interesting objectives are achieved:

The resulting sound sounds more balanced and compensated, there is not much difference between the soft and strong parts of the signal
We gain headroom space (the difference between the nominal level and the saturation point) and we can increase the overall volume of the signal a little more without “touching the ceiling” (the peaks were attenuated). As a consequence, the parts that previously sounded with little force will now be heard better.
It will allow to integrate the signal with greater ease and clarity in the general mix.

Standardization

Normalization is an atypical dynamic process, very different from compression, limitation, expansion or noise reduction:

It does not reduce the relative dynamic range of the audio signal.
It is not applied in “real time”, or at the moment, but it is a process that is carried out “a posteriori”, on the previously recorded material.
The process to normalize audio is summarized as follows:

Normalization analyzes the material and detects its highest volume peak. It then increases its gain to the maximum possible without exceeding the reference level (from which distortion would occur).
Taking as reference the same proportion of increase applied in the previous step increases the level of the rest.
The signal, in general, will sound with a greater volume. The maximum volume level that we can reach depends on the limit marked by the highest peak.


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CBR and VBR What are they and what is the difference?

 

Both acronyms correspond to two coding modes used for audio and video and their meaning is as follows:

CBR (Constant Bit Rate): Constant bit rate.
VBR (Variable Bit Rate): Variable bit rate.
Constant bit rate
In CBR mode, the bit rate per second that will be used in the coding process is set numerically and this will be maintained constantly for the entire duration of the audio or video clip.

Variable bit rate

When we use VBR, an average of the bit rate per second that will be used in the coding process is established numerically and this, according to analysis of the characteristics of each image frame, varies decreasing and increasing according to the information needs that occur during the audio or video clip.

Which of the two is recommended to use?
The use of one method or another depends fundamentally on two factors that cannot be analyzed separately since they are co-dependent:

The intended quality
available capacity

Let’s say we are going to make a video compilation on a double layer DVD with the capacity to store 8.5 GB. The video clips are in HD (720p) and although the figures that will be used for the example cannot be precise because they depend on the type of compression used, we will assume that in total, putting together all the clips we add 10 minutes.

The result of the compilation made in VBR to the standard commonly used for this quality (6-8 Mbit / s), would only be occupying 0.7GB of the total capacity of the disk, then then, according to our capacity budget, we can still increase the bit rate to increase the amount of information and consequently the image quality.

In this specific case, we could use the CBR mode to the maximum quality that the software / hardware that we are using allows us to increase and increase the bit rate for example to 9 Mbit / s, thus maintaining a constant good quality at all times of the film without any risk that the disc is not enough to record the total 10 minutes.

Returning to the example, suppose now that instead of 10 minutes, our clips total 90 minutes. Beforehand, we know that the 8.5GB disk will not be enough to hold that amount of information at constant maximum quality and that is when we use the VBR mode to compile.

Modality of one and two passes

The VBR mode can be configured in one or two pass mode and this refers to the fact that if we choose 1 pass, each image frame will be analyzed in fractions of a second (on the fly) and according to the information obtained, the rate of bits to apply during a certain number of frames in the sequence. This method encodes more quickly but sometimes, you get to notice the variations in image quality because in some way, the program tries to “guess” the behavior of the pixels during the following frames and when it varies unexpectedly in a cut of scene, sudden color variations or an increase in the action of the image, the bit rate applied is lower than required.

In the 2-pass mode, the first one dedicated exclusively to image analysis, then the software makes a budget and applies during the second pass the bit rate variation with much better result and virtually imperceptible quality transitions. When the scenes are relatively stable and static, the bit rate decreases and when variations in the intensity of brightness, colors or the action on the screen intensify, the bit rate increases. In this way, the coding program makes an optimal distribution by subtracting information where it is not necessary and adding it where the image requires it to finally be able to make the highest quality compilation in less capacity.

Explanation of advanced mp3 conversion settings

 

In this article we are going to address the audio coding settings that affect the sound quality. Understanding how conversion settings work can help you select the optimal sound coding properties in terms of file size relative to sound quality.

What is the bit rate?

The bit rate is the amount of data consumed to transmit the audio sequence per unit of time. For example, a bit rate of 128 kbps (kilobits per second) means that a second sound is encoded with 128,000 bits (1 byte = 8 bits). If you convert this into kilobytes, a second of sound occupies about 16 KB.

Therefore, the higher the bit rate of a track, the more space it will occupy on the computer. However, with the same format, a higher bit rate allows you to record the best quality sound. For example, if you convert an audio CD to MP3, the 256 kbps bit rate will provide much better sound quality than the 64 kbps bit rate.

Because today’s hard disk space is relatively cheap, it is recommended to convert to MP3 with a bit rate of at least 192 kbps or higher.

The bit rate can also be classified as constant or variable.

The difference between constant bit rate (CBR) and variable bit rate (VBR)

The constant bit rate means that the encoding of each audio segment consumes a constant amount of bits. However, the structure of the sound may be different, and the coding of a segment of silence requires much less bits than the coding of a segment of intense sound. Unlike the constant bit rate, the variable bit rate adjusts the quality of the coding at various intervals. Thus, intervals that are simple in terms of coding will use a lower bit rate, while more complex intervals will be coded with a higher bit rate. The use of a variable bit rate allows for better sound quality without increasing the file size.

What is the sampling frequency?

This term is used in the conversion of analog signal to digital form and defines the number of samples (signal level sample measurements) per second needed to convert a signal.

CBR vs VBR – which one to choose?

When you are going to pass a music CD to MP3 or AAC format you will have seen two different encoding options, the CBR and the VBR. Do you know the diference?

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CBR (Constant bitrate) encoding

CBR is a type of encoding in which a fixed bit rate is always used, so if we encode a song at 192 Kbps, the resulting file will have a bitrate of 192 Kbps for the entire duration of the song.

It is the speed at which data is processed or transferred.
It is usually measured in seconds and the most common units are:
Kb / s or Kbps (remember that the lower case “b” is bits, not bytes).
Mb / s or Mbps.
Also called: bitrate, bit-rate and BR.
The main advantage of using CBR is that the coding is a bit faster (compared to VBR). However, the resulting files are not as well optimized in size and quality.

 

CBR coding also has another advantage and we know in advance the transfer rate we need. For example, if we set a bitrate of 300 Kbps, we already know that with a 320 Kbps connection we will be able to transmit the data without suffering cuts, so it is usually used in real-time transmissions or streaming.

VBR encoding (bitrate variable)

VBR is an encoding method that allows a variable bit rate, this means that the bitrate of an audio file can increase or decrease dynamically depending on the complexity of the sound.

If the music is very simple or there is silence for a few seconds the bitrate can go down and then go back up in the more complex areas of a song.

What is bitrate?

What is bitrate?

You will surely have heard the word bitrate on many occasions when an expert talks about digital format videos. But, if you don’t know what bitrate is or what it is, we tell you in this article so that from now on you will be clear about what is being talked about when bitrate is mentioned.

Like the resolution and the final format of the digital video, another of the determining factors to obtain an excellent image quality is bitrate. Specifically, bitrate is the flow or data rate, that is, the amount of information when playing a video that reads our computer per second.

By resolution we mean the size, that is, the amount of information you have for each centimeter or for every inch (although they are not really technically measured in centimeters or inches, but we explain it this way to be clear). That is why a low quality video looks pixelated if we enlarge them, it is because it lacks information, it lacks greater quality of details.

For that reason, and in the same way that happens with the size of the image, the greater the data flow, the greater the quality of the material.

The bitrate can be even more decisive than the size of the image to define its quality. The reason? Although we have a large video, if the data flow is poor, the material will be of poor quality.

For example, a VCD of 352×288 resolution and 1150 kbits / s will be of higher quality than one of 720×576 and 300 kbits / s.

In this same example, if one of them has a larger screen size, its bandwidth is low, since this data stores the information related to the luminaire and the color of the video.

For that reason, when the data flow is poor, the computer must group a large number of pixels that contain the same information, generating a redundancy that affects the quality of the video.

Because, although the video is large, the computer only gets few data with which to “fill in” that large space. It has little detail again, because it has little information for the space it should detail.

Bitrate implies the amount of details or data that flows through each period of time. The more data we have, the more detail can be obtained and in greater detail, we will obtain a higher quality.

The equation is that simple: quality depends on detail. Just as in a camera the number of megapixels matters, in the same way a higher bitrate will give us more details per second and that will directly impact the quality of the video.

What is bitrate? Bitrate video, audio, internet and more

HomeAudio Y VideoWhat is Bitrate? Bitrate video, audio, internet and more …
What is bitrate? Bitrate video, audio, internet and more …

Surely we have heard the word bitrate countless times when an expert user refers to a video or audio in digital format, and we have come to know that it is the element that defines the flow of data. But what exactly is bitrate? The question arises because not only in these fields is this parameter used.

Like the resolution and the final format of the digital video or audio, another determining factor to obtain excellent quality in an image or sound is, without a doubt, bitrate, a parameter that perhaps is not always taken into account and that not only applies to the field of audio or video. That is why in this article we will find a lot of information to perfectly understand what bitrate is.

Bitrate: Why it is so important in our digital life

Electronic devices have reached unthinkable operating speeds just a few years ago, and that is why today we hope that our device, be it a smartphone or a tablet, a computer or a hard disk, will respond to us at the moment and without hesitation. In this they have to see many and varied factors, but one of the most important is the bit rate at which it can exchange or process information.

The term bit rate, used in computing and telecommunications systems, basically refers to the amount of bits that can be transmitted in a given unit of time through a transmission system or between two digital devices. Depending on the context in which the term is used, the bit rate, or bitrate in English, is measured in Kbit / s or Mbps, kilobits per second or megabits per second, respectively.

Regardless of the unit of measurement for defining bitrate, higher numbers always mean better and higher quality values, although we must not forget that low bit rate values ​​can also mean less signal processing by the hardware, very convenient in equipment such as smartphones, tablets or netbooks.

Bit rate on the Internet

In the case of the bit rate applicable to the Internet, the higher bit rate is better, since the content we receive from the network arrives faster. In other words, the higher the bitrate we get from our ISP, the better the connection and we can work much more comfortably.

A higher bitrate in an Internet connection means streaming movies and video in high definition, playing online with no delay and downloading really large files without problems and in a few seconds.

In the event that we want to know exactly what the bitrate of our connection is, we can do so easily and comfortably by accessing with our browser a site that is responsible for performing this test. One of the best in the market is speedtest.net.

Bit rate in audio and video

If we talk about audio and video, the meaning of the term bit rate differs a bit from what we use for the Internet. In this context, the bit rate refers to the amount of data stored for every second of data that they reproduce. To take an example, an MP3 file of a 320 kbps song offers a much higher quality than the same 128 kbps encoded file, obviously as long as both files have been created from the same source.

But we must always remember that if the source from which we obtained the files was of poor quality, then the copy will also be of poor quality, it has been encoded at 128 kbps or 320 kbps.

This also happens with videos, a much higher bit rate will offer a much better viewing quality than a video with the same resolution but at a lower bit rate.

The bit rate could be expected to increase each time the resolution grows as a larger amount of data is being processed. This means that while high bitrate rates can offer excellent display quality, they also require much more effort to process part of the hardware, forcing it, especially in modest and older hardware, to produce pauses and cuts.

Another aspect that we must also take into account since it is very important, is that video file formats use different sets of compression algorithms, which could also offer high quality with a more discrete bit rate. However, the extra process load for these types of videos can also complicate the processor and the systems involved in decoding.

The quality of YouTube videos leaves much to be desired: they need an update

 

When we watch a video on the platform, we can usually appreciate that, despite finding videos in 1080p resolution, the compression applied by the platform is too aggressive. This causes the final quality of the video we are watching to differ greatly from that of the original file. The codec that YouTube uses is H.264 / MPEG-4 AVC, using various profiles or “levels” that specify the maximum resolution, frames per second and maximum bitrate of each quality.

We have analyzed a few videos, and we have taken a fairly representative one that is available on both Vimeo and YouTube to see how both platforms compress the videos. In addition, we have seen the maximum and minimum bitrate that each video can have according to the YouTube Help page for each resolution. The audio, as we discussed in summer, reaches 128 Kbps, leaving 320 Kbps only for YouTube Red users.

What sound quality (bitrate) do YouTube videos have?

The bitrate for 1080p videos is too low: 4K is the way to go
The bitrates that YouTube says it assigns to each video are the following, with the profile level in parentheses:

4K / 2160p
60 fps: Between 20,000 and 51,000 Kbps (L5.2)
30 fps: Between 13,000 and 34,000 Kbps (L5.1)
1440p
60 fps: Between 9,000 and 18,000 Kbps (L5.1)
30 fps: Between 6,000 and 13,000 Kbps (L5.0)
1080p
60 fps: Between 4,500 and 9,000 Kbps (L4.2)
30 fps: Between 3,000 and 6,000 Kbps (L4.1)
720p
60 fps: Between 2,250 and 6,000 Kbps.
30 fps: Between 1,500 and 4,000 Kbps.
480p: Between 500 and 2,000 Kbps.
360p: Between 400 and 1,000 Kbps.
240p: Between 300 and 700 Kbps.

In our tests, the bitrates we obtained for the previous video were the following:

4K at 30 fps
Vimeo: 19.4 Mbps (file size: 943 MB) (capture)
YouTube: 17 Mbps (file size: 821 MB) (capture)
1080p at 30 fps
Vimeo: 4.31 Mbps (file size: 219 MB) (capture)
YouTube: 3.2 Mbps (file size: 160 MB) (capture)
vimeo vs youtube compression

As we see, Vimeo files occupy more not only because of the lower compression of the videos, whose quality is superior to the naked eye, but that Vimeo’s sound quality doubles that of YouTube, since it reaches 256 Kbps by 128 Kbps from YouTube. So that you can see the difference in image quality, you can open the same New Zealand Ascending video on YouTube and Vimeo, and we have also left four captures at the same moment of each video so you can save them and see comfortably the video difference.

Digital audio

 

Digital audio is the representation of sound signals through a set of binary data. A complete digital audio system usually begins with a transceiver (microphone) that converts the pressure wave that represents the sound to an analog electrical signal.

This analog signal goes through an analog signal processing system, in which limitations in frequency, equalization, amplification and other processes such as compaction can be performed. The equalization aims to counteract the particular frequency response of the transceiver used so that the analog signal closely resembles the original audio signal.

After analog processing the signal is sampled, quantified and encoded. Sampling takes a discrete number of analog signal values ​​per second (sampling rate) and quantification assigns discrete analog values ​​to those samples, which means a loss of information (the signal is no longer the same as the original). The coding assigns a sequence of bits to each discrete analog value. The length of the bit sequence is a function of the number of analog levels used in the quantization. The sampling rate and the number of bits per sample are two of the fundamental parameters to choose when you want to digitally process a certain audio signal.

The digital audio formats try to represent that set of digital samples (or a modification) of them efficiently, so that it is optimized depending on the application, either the volume of the data to be stored or the processing capacity necessary to obtain the starting samples. In this sense there is a very widespread audio format that is not considered digital audio: the MIDI format. MIDI does not start from digital samples of sound, but stores the musical description of the sound, being a representation of the score of the same.

The digital audio system usually ends the reverse process to that described. The set of samples they represent are obtained from the stored digital representation. These samples go through a digital-analog conversion process providing an analog signal that after a processing (filtering, amplification, equalization, etc.) affects the output transceiver (speaker) that converts the electrical signal to a pressure wave that represents Sound.

Digital audio quality

The quality of the digital audio depends strongly on the parameters with which that sound signal has been acquired, but they are not the only important parameters for determining the quality.

One way to estimate the quality of digital sound is to analyze the signal difference between the original sound and the sound reproduced from its digital representation. According to this strategy we can talk about a specific signal to noise ratio. For audio systems that perform lossless digital compressions, this measure will be determined by the number of bits per sample and the sampling rate.

The number of bits per sample determines a number of quantification levels and these a signal-to-noise ratio of carrier peak that depends quadratically on the number of bits per sample in the case of uniform quantification. The sampling rate establishes a higher level for the spectral components that can be represented, and linear distortion may appear in the output signal and aliasing (or spectral overlap) if the signal filtering is not adequate.

For digital systems with another type of compression, the signal to noise ratio can indicate very small values ​​even if the signals are identical to the human ear.

The reason is that the signal to noise ratio is not a good parameter of sound quality measurement because the quality perceived by the listener is determined by the response of the human ear to the sound waves, which does not perceive many of the possible differences Logically, if the signals are very similar, the ear cannot differentiate them, but they can also be very different and can be perceived as the original signal. Therefore, the evaluation of the quality of a digital system through sensitivity parameters of the human ear and specific tests with specialized listeners seems more appropriate.

It is in this sense that the quality of digital audio systems is evaluated today. Both MPEG and Dolby Digital (AC-3), which establish perceptual compressions, perform test benches to estimate the quality of the encodings.

MP3 vs FLAC vs AAC vs OGG: what differences does each audio format have?

 

Although streaming platforms such as Spotify are more fashionable than ever thanks to their great musical variety, reduced price and convenience of not having to manually download the files, many users still prefer to have music stored locally, for which there are formats like MP3, FLAC, AAC and OGG.

These formats are currently the most widespread for music on our devices, being able to pass files between PC and mobile without relying on the Internet and without being afraid of depleting our data rate. Most of the formats that we are going to deal with are formats that compress information, and therefore have quality losses. About the compression of images and files we talked a while ago.

 Why does a compressed file occupy less?

Audio formats with losses: MP3, AAC and OGG

The first of the formats that we are going to try is MP3. This format, whose acronym stands for MPEG Audio Layer III, is the most commonly used format with loss of quality. It is not the one that offers the best quality or best compression, but its great compatibility has made the standard format for music for decades.

Another widely used format for sound in recent years is AAC. It is very similar in MP3 performance, but has the advantage that it is able to offer the same quality in a smaller size. This is the reason why platforms like Apple’s iTunes use it, and the fact that Apple uses it has made its compatibility as great as MP3’s today. AAC is also used to compress stereo sound in movies of 1 or 2 GB in size that we find in various torrent portals, direct download or streaming.

The next most used format is OGG, or OGG Vorbis, it is a free alternative to AAC and MP3 (although the MP3 patent ended last May). Its size is similar to that of MP3, but its compression is smaller, keeping a higher audio quality than MP3, especially at high frequencies, which destroys the MP3 the lower the bitrate. In addition, while MP3 reaches 320 Kbps, OGG reaches up to 500 Kbps.

Lossless audio formats: FLAC, ALAC and WAV

On the other hand, we have FLAC. This lossless format is free, as indicated by its name (Free Lossless Audio Codec). The size of your files is between 5 and 10 times larger than MP3, but it has no losses, although the audio is “compressed.” Thus, it occupies much less than uncompressed formats such as WAV or AIFF, and maintaining the same sound quality.

The equivalent of FLAC in Apple is ALAC. Although it is not as efficient as FLAC (its files occupy more), ALAC owns Apple, and is the only alternative that can be used in iTunes, since the platform does not read FLAC.

In short, the best format to use is always FLAC if you can afford its large size, followed by AAC and OGG. If you have no choice, MP3, although it is the least desirable option, is the most widespread today, and what you will be forced to use for a lot of music on the network.

What audio formats exist? All you need to know

 

FLAC, WAV, AIFF, DSD … these are just some of the acronyms you can find when looking for a digital format. They are also accompanied by technical data such as sample rates and bit depth. So many terms can leave you more misplaced than a chicken in a dance. And unless you are an expert in digital sound, the process to choose the audio format that best suits your needs can be a mess. But if they explain it to you, the subject is relatively simple. That is why in Culturasonora we have prepared a complete guide on the different audio formats used. This will prevent any acronym from taking you on the dark side, dear Padawan.

Sample Rate and Bit Depth.
MP3s vs WAVs vs AIFF.
OGG vs FLAC vs ALAC.
What is the DSD format?
How to listen to the DSD?
MQA audio Hi-Res.
What is Bit Depth and Sample Rate?

These two concepts are basic. To understand how audio formats work, you need to know what Bit Depth and Sample Rate are. They are two measures that indicate the quality of a digital audio file. We will try to summarize it so that you stay with the general idea

When you read the specifications of the audio formats you find a couple of figures. For example: 32-bit / 192kHz or 24-bit / 96kHz. These numbers indicate the bit depth and the sample rate. These references tell us how much information the different formats transmit and the sound quality. For example, the audio we hear on a normal CD, or on a Spotify stream, is 16bit / 44.1kHz. Samples are always measured in Hertz (or hertz) and bit depth in Bits.
Softwares or hardwares do not usually work with a continuous flow of information but often use pieces, samples or samples to effectively manage the data that is transmitted. The sample rate is the number of samples per second that are obtained from a recording. The higher the number of times a device plays the samples, the higher the sound quality. Each of these extracts or samples has a certain amount of information, which is the bit depth, or bit depth.
To understand it better, we are going to make a slightly beast analogy, which is not entirely true, but which will help you to make sense of all this. What interests us. If you control a bit of photography and image you will get it right away: the sample rate would be something similar to the frames or frames per second of a video, and the bit rate would be similar to the pixels of a photograph. The higher the bit depth number, the more information each sample will have. The more pixels an image has, the more resolution each frame of a video will have. The more frames per second a movie has, the greater the definition. In short: the higher the number of the Bit Depth and the Sample Rate, the higher the quality of the audio file.

Audio formats: MP3 vs WAV vs AIFF

What is the MP3 format?
If you are interested in getting some audio fidelity and decent sound from your files, you will want to avoid this format. Why? Because basically an MP3 is a file that sacrifices audio quality to minimize size. They weigh very little for any device to read. The negative? The compression of these files provides a poor, almost lifeless sound. Nowadays almost nobody uses that format seriously. Even its creators recently finished the license declaring her dead. But surely every now and then you find a zombie file with this format.
What is the WAV format?
WAV (Waveform Audio File Format) are equally common but better for anyone who wants a decent audio format. They are higher resolution files than MP3s. A WAV is an audio piece that is encoded with something known as Pulse Code Modulation (PCM), a medium that encodes analog audio parts and converts them into digital so that they can have the Sample rates and the Bit Depth of the that we have talked about before.
What is the AIFF format?
The audio format AIFF (Audio Interchange File Format) is very similar to WAV, since it also uses the PCM to encode analog audio pieces and present them in digital format. This format was born as an answer from Apple to the Microsoft WAV, and at the beginning it could only work on MAC computers. Currently, the AIFF and WAV are more or less interchangeable.
In summary…
To close this topic we will tell you that if you have a file in WAV or AIFF audio formats you will hear a piece of good quality sound. Normally these formats are used in files that we play through our services, such as the iTunes music library. We will not see them in online streaming services, which tend to use special types of files. Now we will review that point

Do you differentiate between an mp3 encoded at 128 and one at 320 kbp?

 

Surely more than once you starred in or attended a dispute between people who say that you notice a lot of difference between an MP3 encoded with one or another level of compression, or between a CD and an MP3. However, there are very few people able to distinguish these nuances. That’s why at mp3ornot.com we propose this challenge:

Are you able to differentiate between an mp3 encoded at 128 kbps from another at 320 kbps? If you think you have your ear developed enough to capture that difference, I challenge you to take the test … and then tell me.

Data:

The Mp3 (MPEG-1/2 Audio Layer 3) was one of the first types of audio compression with almost imperceptible losses to the human ear. Its compression rate is measured in kbps (kilobits per second), with 128 kbps being the standard quality, in which the file size reduction is about 90%, that is, a ratio of 10: 1. That compression rate can currently reach up to 320 kbps, the maximum quality, in which the file size reduction is about 25%, that is, a ratio of 4: 1, going before 192 kbps, 256 kbps, that is, the maximum quality that can be removed in Mp3.

The lossy compression method used in the compression of the Mp3 consists in removing from the audio everything that the human ear would normally not be able to perceive, due to phenomena of masking sounds and limitations of human hearing (although people with absolute hearing can perceive such losses).

How to compress an MP3 file

Knowing that the MP3 audio format has become the most standardized and used worldwide in recent years, we have thought it pertinent to talk about the different parameters that make an MP3 file respond to one quality or another.

The first thing we have to know is the meaning of MP3, and it is nothing more than a compressed digital audio format that although by nature suffers a loss of information in the conversion process, it is not audible by the human ear, which It implies an assumable loss since we will not be able to perceive it in broad strokes.

Generally, an MP3 file is capable of reducing the size of an original audio file without altering quality. What this means is that in the conversion process for example of an audio file with CD quality, the result of the MP3 file would be practically identical to the original, leaving as standard ratio 1 minute = 1 MB.

That said, we can begin to clarify some parameters that will determine the quality of an MP3 file, which in its vast majority, depends on the bitrate or Bitrate.

Impact of Bitrate in MP3 quality
The MP3 file format allows you to select the compression ratio of the source file. The margins at the domestic level are between 8 Kbps and 340 Kbps, with 128 Kbps being the transfer rate equivalent to CD quality.

Bitrate is the unit of measure for the rate of data transfer read from an MP3 file. The higher bitrate an MP3 file has, the greater the amount of data that a player can obtain in the unit of time (Second).

The more instrumental content or quality an MP3 audio file contains (sound effects, recorded audio tracks, high frequencies, low frequencies, etc.), the higher the transfer rate it will require to fully reproduce the information, and at this point, it is where it is defined The quality of the MP3 file, since if we compress that file, we reduce that bandwidth, we will be sacrificing some of that data, resulting in loss of information that will influence the final result of the MP3 conversion.

In summary:

If the file lasts 5 minutes and weighs 3 MB, we would be talking about a low quality MP3 file.

If the file lasts 5 minutes and weighs 9 MB, we would be talking about a high quality MP3 file.