Audio quality: learn about bits, sample rate and formats

Audio setups for recorders or audio interfaces can be complex, but if you’re going to be working with , or audiobooks and you want to have a good one audio qualityyou must know the parameters when recording and exporting files, whether in Audacity (free), Reaper, Adobe Audition, as well as, in .

Therefore, here we are going to talk about the differences between the sampling rates (sample rate), resolution (bit depth), rates of file compression and even variations of formats. This way, you will know what the options are and you can guarantee good results.

We will explain why we recommend recording in uncompressed format (wav, for example in 24 bit Y 48kHz, and the reason why, in most cases, we don’t need more than a 192kbps MP3 to export good quality audio.

We will also talk about the possibility of greater compression in podcast files, which can be generated in MP3 of 64kbps, bowfacilitating their online consumption.

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Formats, extensions and codecs: what do they mean?

Simply put, the Format is the file type, identified by its extension (*.mp3, *.wav, *.ogg, *.wma, etc.), which often indicates how the file has been encoded or what its codec

As an example, a file in the MP3 format has the extension *.mp3 and codec MPEG-1 Audio Layer III.

Normally, these endings are mixed, but what matters is knowing that, just like in the videos, files with the same type of extension do not always have the same codec, and vice versa.

Therefore, it may happen that a software that plays your files refuses to play another of the same extension, which may indicate that different codecs were used. If this happens to you, the solution is to use other software to read the file or do a conversion (re-encoding), which can be done even in video editors.

These variations of formats and codecs depend on the options of the companies that develop the software to execute files.

The records They are usually divided into two types: without compression or compressed.

1. Files without compression

Some audio recording equipment allows you to record files without losing any of your data. These uncompressed files can be generated in various formats and extensions, such as WAV, AIFF, FLAC Y A THE C. For those who know a little about photography, they are equivalent to RAW or DNG.

Being very heavy, it is only recommended to use these lossless formats (lossless) in some cases, such as:

  • When the consumer can process the final product (files intended for sound banks, for example);
  • When the recording occurs on physical media (CD, DVD and Blue-Ray);
  • For the audiophile market (as a matter of perceived value and high quality assurance).
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However, even if you don’t want to end the process with a WAV (one of the most common), the formats lossless they can be very useful in the editing stage. Because they contain a lot of information, they withstand more extreme alterations without degrading the sound quality.

In this way, through plugins, conversions and processing, it is possible to manipulate them more freely, guaranteeing high quality, even if a compressed file is generated later.

2. Compressed files

Most of the equipment available on the market (cameras, smartphones and even audio recorders) usually deliver already compressed files, which they are more practical, easier to process and require less storage space, with very small sizes (in bytes).

Some examples of these formats are: 3GP, AAC, M4A, OGG, WMA Y MP3, which is without a doubt the best known. Those files are like the JPEG or GIF in the field of images.

They are generated by means of a complex algorithm, seeking to keep only the most relevant sound information. Depending on the compression mode, we can generate a MP3 from a WAV and have a file 10 times smaller, without perceptible alterations in sound.

Despite being very popular, the format MP3 It is already considered obsolete, since there are others like the ACC (extension .acc either .m4a) enable even smaller, higher-quality files.

Even so, MP3 is still widely used, since much of the software and equipment was developed with it in mind. That is why we will use it as an example when talking about compression rates.

Compression rate: how is it related to audio quality?

Now that we’ve said that a file can be compressed and still maintain audio quality, you should know that compression level can vary greatly.

By the value of the compression rate (or bitrate) we can control the size of the file and, therefore, the quality of the sound.

For example, a 320kbps MP3 (kilobits per second) can sound just as good as uncompressed audio from a CD or DVD. As the value of the bit ratethe size of the file decreases, but the sound losses become perceptible, depending on the audio in question.

Here are some references to get an idea of ​​how that rate affects sound quality:

  • 320kbps – audio that does not differ from CD quality;
  • 192kbps – no significant losses for most people;
  • 128kbps – slightly perceptible losses;
  • 96kbps – quality similar to FM radio;
  • 32kbps – similar to AM radio;
  • 16kbps – similar to short wave radio (“walkie-talkie”).

Remembering that these values ​​and descriptions are only an approximation, since the compression of the file acts differently in each type of audio. The more perceptible information (or the more complex the audio in question), the more room there is for compression to affect quality.

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That is why, for a podcast without a soundtrack, it may not be a problem to generate a file of just 64kbps, bow, with a single sign audio tapping simultaneously on the left channels (L) and right (R).

However, a music produced in a studio, with several different instruments, can suffer perceptible losses, even when the compressed file is of 128 kpbs, stereo, with a different signal for each speaker, right and left.

Here we are talking about fixed compression rates (CBRconstant bit rate), but there is also the possibility of generating files with variable rates, such as the calls VBR (variable bit rate) either APR (average bit rate).

In the VBR, the algorithm analyzes the audio and decides where it can compress more aggressively and where it should remove less information. The APR acts in a similar way, but remains at the average of the previously stipulated rate. Those two methods, despite being more intelligent, can cause incompatibility with some audio players.

when we talk about compression x qualityremember that there are no rules: each case is a case and it is necessary to evaluate them individually to know to what extent losses are acceptable, or when it is worth prioritizing ease of use (faster download or less storage impact, for example) before quality.

Remember that some websites and services re-encode the audio after uploading. As we have no control over this process, it may be a good idea to submit files with a slightly higher quality than necessary, so that there is a safety margin in case of further conversions.

Amplitude resolution: 16 bits or 24 bits?

If you’re going to be using a dedicated sound card/interface or recorder, you’re going to be presented with choices of bit depth. This is related to the digital audio standard PCM Y does not apply to compressed files.

The values ​​refer to the signal-to-noise ratio. In other words, it has to do with dynamics, or the volume levels that the file manages to record with quality.

It is as if it were an amplitude resolution of the sound. Thus, in theory, 16-bit audio manages to represent 65,536 volume levels between the lowest and highest value on the scale. Whereas in 24 bit, there are 16.7 million gradations.

Despite the great numerical disparity, in practice it is not a perceptible variation to our ears. But there is a technical difference that can, in some cases, give the 24-bit file an advantage at the time of capturing and editing.

We know that we must be careful with the input level when recording, so that the audio does not “explode” (generating clipping). That’s what happens when we let the graphic meter go too high, going past 0 dB (maximum value before digital saturation/distortion occurs). Therefore, a certain safety margin must be respected, called “headroom”.

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In 16 bitIn addition to being careful, it is also recommended to pay attention so that the input level is not too low.

The reason is that because there isn’t enough resolution to accurately record extremely faint signals, those sounds can sound digitally distorted or noisy, through a process called ditheringwhich seeks to mask those quantization flaws.

In this way, as the 16-bit file registers fewer volume gradations (48 dB less than the 24-bit file), theoretically there is a risk that, when increasing the volume in the software, there will be more of those beeps. In 24 bittechnically That risk no longer exists.

Regardless, surely there will be a number of noises (noise floor) coming from various sources, such as: cables, electrical network, preamplifiers, microphones, low-quality components, noise from the environment itself (“room noise”) and even derived from the natural operation of the equipment used (some manufacturers specify the value in the Handbook).

Thus, in practice, values ​​of bit depth they probably won’t influence your recording. So, if your computer barely supports 16 bits, don’t worry, because in the end it’s the same value of bit depth from an audio CD.

However, as a 24-bit file is no heavier than a 16-bit file, it is worth recording in this higher resolution whenever possible. In addition to ensuring a higher safety margin when processing the file digitally, 24-bit is the DVD and Blue-Ray standard. In this way, unnecessary conversions are avoided, in the event that the final audio is destined for one of those physical means of communication.

Currently, there are even computers that work in 32 bits, but, as we saw, you will hardly be able to benefit from something like this, since it is an option for specific cases.

Sampling rate: what does that value tell us?

Other values ​​that you will find are relative to the sampling rate (or sample rate). Those numbers depend on the number of times analog sound is recorded per second, to be digitally reconstructed (44.1 kHz equals 44,100 samples per second). It is as if it were the number of frames (frames) per second in a video, necessary to create the illusion of movement.

Those values ​​also refer to the maximum frequency (highest sound) possible to play in the file.

Let’s remember…

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