Category: AAC normalizer

WebM: everything you need to know about the Google format


Free Download Mp4Gain
picture

 

What is the WebM?

WebM is a container format (with extension * .webm) for multimedia files, that is, for videos and audio files. In the same container the video codecs VP8 and VP9 are used, as well as the Vorbis and Opus audio codecs. At the Google I / 0 2010 conference, the company announced its plan for WebM to be an alternative to the existing MP4 format with its H.264 codec from the beginning. The consumer can use the latter at no cost when watching a video, but developers who want to work with the codec must pay the license fees. On the contrary, WebM is an open source project with which anyone can work without paying rights for it.

WebM is designed for use with HTML5. The VP8 and VP9 codecs are designed so that in those cases where considerable compression must be carried out, the extraction can still occur with little computing power. The objective of this design is to allow the reproduction of Internet videos on virtually any device (regardless of whether it is a desktop computer, a tablet, a smartphone or a multimedia device such as a Smart TV). It is not surprising that YouTube, being a subsidiary of Google, converts all its videos to the WebM format, regardless of the format of the original file. Despite everything, YouTube still supports H.264 for those who cannot play WebM.

WebM has become a political issue within the Internet community. While Google tries hard to consolidate this audio and video format, other important market players such as Apple or Microsoft cling to formats like MP4. The main reason is, above all, the patent system: both software companies use a group of MPEG-LA patents, since it is responsible for maintaining the patents of the used codecs and charging royalties for them. Google is trying to circumvent these patents with WebM.

This situation has already led to legal problems in the past, the VP8 codec being the point of contention. Several companies have criticized that their codec patent has been ignored. Google would have reached an agreement with MPEG LA, however, Nokia is not part of this patent pool and believes its rights have been ignored. A first lawsuit, in which the company faced its competitor HTC before the courts, whose devices support V8, was dismissed by the Mannheim regional court.

WebM vs. MP4: advantages and disadvantages

While WebM is relatively young, MP4 (MPEG-4 Part 14) and H.264 have been used for many years. Due to its age, this format and the codec have become a standard: you will find few applications that do not support MP4. In addition to Internet services and PC and MAC software, many other devices (such as camcorders) can also use MP4. The high degree of acceptance makes the format interesting for both manufacturers and users.

But Google has been marked somewhat with the open source character of WebM: using the format is no cost to manufacturers, developers or end users. In addition, the software is distributed under an open BSD license.

The fabric behind the MP4 or H.264 license is opaque: most users, even those who create videos in a professional way, do not know if they have a valid license with the purchase of hardware or software or if any video violates The license right. WebM eliminates this confusion. The MPEG LA already announced in 2010 that the use of the H.264 codec would also be free in the future, provided that the videos created were already free for users.

For many users, the performance of both formats is more important than the controversies surrounding their patents: it is for some reason that H.264 has positioned itself as the leader of the codecs in recent years. The quality of MP4 videos of this encoding is generally considered very good. H.265 exceeds it in some aspects. WebM also convinces with the image and audio quality, but VP8 does not reach the level of H.264. To what extent the image quality of VP9 approaches H.265 (also known as HEVC) is a controversial issue; some believe that both are equal, while others say that the quality of VP9 does not reach that of H.264.

Two other determining characteristics when comparing codecs are the file size and the speed of encoding and decoding. Both directly influence the utility: for fast data transmission over the Internet, the size should be kept as small as possible. This is especially relevant in the mobile Internet field. H.264 has a bad reputation for creating, in comparison, large files. At the same time, decoding on the user’s site


Free Download Mp4Gain
picture


Mp4Gain Main Window
picture


Mp4Gain Features
picture


Free Download Mp4Gain
picture

What is a codec? Audio and video compression

 

Check our codecs and containers guide to not confuse you anymore. Learn what formats suit you.

Has it happened to you that you download a video file and then you can’t use it on your player? Or that you finally finish editing your video clip and it takes years to upload to the Internet? You might think it’s a problem with your file. You are not in error, only that the question is more specific: it is the codec and container you are using.

Perhaps they are somewhat strange terms, but they are gaining more and more publicity due to the growing online video and audiovisual production community. So if you plan to start your career as a youtuber, take into account the information, because if you end up with a final video with a weight of 1 GB it will not be fun to wait for it to upload…

In this guide we will explain what each of these elements consists of and how they work. We will talk about both: video and audio.

What is a codec?

Those who are dedicated to video editing know very well that storage space can be a problem. It is better to have the material you record in its original format, but most of the time this implies a considerable amount of GB of space. For example, if you record an hour of content with a high-definition camera you may need … up to 410 GB! This is complicated to keep it, much more if you want to transmit to other media. It is here that the subject gets interesting.

The term codec refers to the process of compression and decompression of video or audio. It is a tool that encodes the video through algorithms and converts it into information. This way you can decrease the file size.

The choice of codec depends on different factors. You should take into account mainly the means of reproduction for the final product. However, coding is not enough for reproduction, it is also necessary to “package” the information to be able to present it. We are talking about containers.

What are those containers?

Suppose you just finished editing a video. The final file contains both images and audio, so you need a way to display it just as you prepared it. This “package” is basically what many refer to when they talk about the format of a file. Then, a container can accept different codecs, while players can use certain containers. For example, the VLC player accepts almost all containers.

Lossless and lossless codecs (lossy and lossless)

There are different types of compression, as we will see later. However, all of them can be divided into two categories: with or without loss. Loss of what? Quality. For example, in the case of audio files, it is not the same to listen to a song in FLAC (Free Lossless Audio Codec) format to one in MP3 (MPEG Audio Layer III). The first is coded in such a way that almost no information is lost at the time of compression, that is, fidelity is maintained.

The same goes for the video. When you want to save storage space, files with loss are compressed, that is, lossy. This makes them much easier to manage. However, it is inevitable to deal with the loss of data and, therefore, fidelity of the image or audio. On the other hand, when you want to maintain the highest possible quality and you have no problem of space, compressors are used without loss or lossless. Again, it all depends on the purpose of your file.

Digital video formats: how to differentiate them

As with text documents, photographs or audios, digital video is available in different formats or extensions.

In this sense, today we find DVD and Blu-Ray, although some of us still keep in an old VHS closet and maybe some Betacam.

But a second meaning or meaning of video formats refers to their encoding, since in digital video, as with a computer program, any file is written in a certain code.

In videos, the code influences image quality, sound quality, whether or not it includes subtitles and, especially, the relationship between quality and file size.

Thus, today we consume digital audiovisual content through physical discs (DVD, Blu-Ray), through streaming and through IPTV (Internet television), but we also handle digital video files, especially for content that we generate ourselves.

Next we will review the most common digital video formats that we can find, what is their origin and what benefits they offer. I apologize in advance for the gibberish of acronyms.

AVI

We start with the most popular format that we will find. Video files with an .AVI extension have their origin in a format that was launched in 1992 and is so popular that most smart TVs, DVD / Blu-Ray players, video game consoles and operating systems play it.

AVI is an acronym for Audio Video Interleave and not many know that it was created by Microsoft as a digital alternative without dependence on a physical format such as the then popular DVD.

Among its advantages, it allows you to include several audio channels and host content generated with different codecs (AC3 or MP3 for audio, DivX or Xvid for video), which can be an advantage but also an inconvenience with which players.

MP4

MP4 or MPEG-4 is one of the most modern formats, launched in 1998 as a standard for playing video and audio in a single digital file.

MPEG stands for Moving Picture Experts Group, the expert group that has established digital audio and video standards and was formed by two international organizations, the ISO (International Organization for Standardization) and the IEC (International Electrotechnical Commission).

In summary, the MPEG and MPEG-2 format were launched in 1993 and 1995 respectively as standards for encoding digital audio and video. To understand each other, any DVD offers its audiovisual content in MPEG-2.

MP4 also supports several audio channels, but has the advantage of allowing more image and sound quality in a less heavy file, as it compresses data better. Apple, for example, opts for this format and derivatives for its iTunes content.

Related to MP4 we can find M4V (video) or M4A (audio).

MKV

The MKV video format is an open format, free to pay rights, and whose full name is Matroska, like traditional Russian dolls.

MKV saw the light at the end of 2002 and has become popular thanks to the fact that within a single MKV file we can store, together with the audio channel, several channels or audio tracks and several subtitle tracks.

Like MP4, it offers very good audio and video quality in a small space. And as a curiosity, the WebM format that allows you to integrate online video via HTML, is inspired by Matroska.

FLV

The FLV or Flash Video format was created by Macromedia, and subsequently acquired by Adobe. This format is usually found as an FLV or SWF extension.

Like the other Flash content, FLV videos are designed for online playback from the browser through Adobe Flash Player.

As we saw in a previous article, Flash will stop developing in 2020, although we still find pages that use it.

MOV

I said before that Apple is currently betting on MP4 (and AAC) to facilitate multimedia content. But its star format for many years was MOV.

MOV, from QuickTime Movie, is also called QuickTime File Format, and today it is still the default format of QuickTime, the macOS video player.

This format can also be found in many digital video cameras, since it offers very good quality

Audio normalization

Audio normalization

audio normalization

The normalization of the audio level is something that is achieved by applying a constant and maintained amount of gain, in volume, to an audio recording to bring the average peak amplitude to a desired level that has been previously defined. To which the same amount of gain is applied to the entire range, the signal-to-noise ratio generally does not change. Normalization differs from dynamic range compression, which applies different levels of gain to a recording so that the amplitude is within a minimum and maximum range. Standardization is one of the most common functions provided by a digital audio workstation.

Peak normalization

One type of normalization is peak normalization, in which the gain is changed to bring the highest PCM value or the highest peak of an analog signal to a given level.1

Since it only searches for the highest level, it does not take into account the apparent volume of the content. As such, peak normalization is generally used to change the volume in such a way as to ensure optimum use of the distribution medium in the mastering stage of a recording. loudness normalization.

Normalization of loudness

Another type of normalization is based on a loudness measure, in which the gain is changed to bring the average amplitude to an objective level. This average may be a simple measurement of average power, such as the RMS value, or it may be a measure of the loudness perceived by humans, such as that offered by ReplayGain.

Depending on the dynamic range of the content and the target level, the normalization of the loudness can lead to peaks that exceed the limits of the recording medium. Some software has the option of using dynamic range compression to avoid saturation when this happens. In this situation, the signal-to-noise ratio is altered.

volume booster

Modern Audio Normalization

Currently Mp4Gain uses an audio normalizationn that is more similar to that used in modern recording studios or live music group recitals.

It is a normalization of volume focused from a new perspective.

Under this new paradigm, not only does it achieve that all songs have the gain of loudness at the best possible level, but it also achieves that each instrument and / or voice obtains a level of gain that makes it audible. Achieve an optimized level of volume gain normalization.

There is no other normalizer in the market that obtains this level of result. People with training in hearing listening can easily notice the difference., very similar to that obtained with expensive hardware in radio stations or in recording studios or in recital consoles, combining limiters, modern compressors and other processors.
All these results that offer expensive hardware equipment, Mp4Gain does for a few dollars.

In fact, the opposite result is achieved than that achieved with masking, because with masking, which is a method used to compress music, you can no longer perceive some sounds that are behind a more audible sound, that is what is called masking, which leads to the loss of audio quality.

Mp4Gain manages to highlight hidden instruments and sounds, performing an audio normalization by frequency bands to achieve this.

That is why we say that Mp4Gain achieves the same results as those obtained through a series of hardware equipment (limiters, compressors, normalizers, etc.) that are very expensive, while Mp4Gain costs only a few dollars.

Digital Audio – Beginners guide

The Cost of a High Sampling Rate

Although it is true that high sampling rates produce better sound quality … that comes at a price.

That price translates into:

Higher processing load.
Less number of tracks.
Heavier audio files.
So you always give something in return. Professional studies can support higher sampling rates because they use better equipment.

But for most home studios, people often find that the standard 48 kHz configuration is the best.

Following…

4. Bit Depth

In order to understand what bit depth is, we first have to know what bits are.

A bit (or binary digit) is a single unit of binary code, with a value of 1 or 0.

The more bits, the more possible combinations. For example…

As you can see in the diagram below, 4 bits allow a total of 16 combinations.

4 bits

When used to encode information, each of these numbers is assigned a specific value.

As the number of bits increases, the possible values ​​grow exponentially.

4 Bits = 16 possible values
8 Bits = 256 possible values
16 Bits = 16,536 possible values
24 Bits = 16,777,215 possible values
With the bit depth in the digital audio, each value is assigned a specific amplitude of the waveform.

The greater the bit depth, the greater the volume increase between high and low … and a greater dynamic range in the recording.

A good rule of thumb is: for every extra bit, the dynamic range increases by 6dB.

For example:

4 Bits = 24 dB
8 Bits = 48 dB
16 Bits = 96 dB
24 Bits = 144 dB
In the end, what this means is that… the greater the bit depth, the less noise.

Because by adding more processing margin (or headroom), the useful signal (at the high end of the spectrum) can be recorded higher above the background noise (at the low end of the spectrum).

small vs large bit depths

Following…

5. Quantization Noise

Impressive that a 24-bit recording can result in almost 17 million possible values, right?

However, that remains much less than the infinite number of possible values ​​that exist in an analog signal.

Therefore, in almost all samples, the actual value is somewhere between two possible values. The solution of the converter is simply to round it or “quantify” it to the nearest value.

The resulting distortion, known as quantization noise, takes place in 2 phases of the recording process:

at the beginning, during the A / D conversion, and
at the end, during mastering
With mastering, the sampling frequency / bit depth of the final track is usually reduced by converting to the final digital format (CD, mp3, etc.).

When that happens, some of the information is erased and “re-quantized”, generating more distortion in the sound.

The most frequent solution to deal with this problem is …

6. Dither

When reducing a 24-bit file to 16 bits, the screen is used to mask much of the resulting distortion …

Adding a low level of “random noise” to the audio signal.

As it can be difficult to visualize the concept in audio, to explain it, we usually turn to the popular analogy of the screen plot.

Is that how it works:

When a color photo is converted to black and white, a mathematical estimate is made to determine if each color pixel should be “quantized” in a black pixel, or a white one …

As is the case when digital audio samples are quantized.

As you can see in this picture, the “before” photo is pretty bad, right?

dither

But with the plot …

a small number of white pixels are randomly distributed in black parts, and …
a small number of black pixels are randomly distributed in white parts …
By adding that “random noise” to the image, the “after” photo looks much better. Well, the screen in the audio works very similarly.

Following…

7. Latency

The GREAT PROBLEM of current digital studies is the amount of latency that accumulates in the signal chain, especially with DAWs.

With all the calculations that are processed, the audio signal takes time to leave the system between a few milliseconds and a few DOCENAS of milliseconds.

Between 0-11 ms of latency – it is short enough, so a normal person does not notice it.
Between 11-22 ms – an annoying delay is heard which it is difficult to get used to.
More than 22 ms – there is so much delay that it is impossible to play or sing at tempo with the track.
In a normal digital signal chain there are usually 4 phases that contribute to the total latency:

A / D conversion
DAW Buffer
Delay of the Plugins
A / D conversion
The A / D and D / A conversion are the least harmful, contributing to total latency with less than 5 ms.

But nevertheless…

The DAW buffer and certain plugins (including compressors and virtual instruments) can add up to 20, 30 or 40 ms or

Beginner’s Guide to Digital Audio for Recording Music

62c-digital audio When recording at home began to become popular …

It happened for a simple reason:

The analog equipment of the past decades was being slowly but inexorably replaced …

For a new generation of audio interfaces and other digital equipment that was cheaper and easier to use.

And that trend has continued since then.

Today … digital audio is the standard in almost all studios, both professional and amateur.

However, surprisingly, there are few people who really understand what it is about.

So let’s see what it is about:

1. The Rise of the Digital Age

binary code Although digital audio is the standard in today’s music …

It has not always been that way.

Originally, music information only existed as sound waves in the air.

Then, as technology progressed, people discovered new ways to convert that information to other formats, including:

notes on a page
electrical signals inside a cable
radio waves in the atmosphere
relief on vinyl records
But in the end, with the rise of computers, digital audio ended up being the dominant format in the music production industry, since it allowed copying and transporting songs in a simple and free way.

And the device that made all that possible was … the digital converter.

Let’s see how they work …

2. Digital Converters

In recording studios there are 2 types of digital converters:

Those that are an independent device, which are normally seen in more advanced studies, or …
Those that are integrated into the audio interfaces, which are usually seen in home studios.
To convert the audio to binary code, they take tens of thousands of samples (samples) per second to make an “approximate” image of the analog waveform.

The image is not accurate because in the intervals between samples, the converter basically has to guess what is happening.

Digital waveform

As you can see in the diagram, in which:

the red line is the analog signal, and …
the black line is the conversion …
The results are not perfect, but they are good enough to generate excellent sound quality.

How excellent? That depends largely on …

3. Sample Rate

Check out this image:

sample rate diagram

As you can see…

When taking more samples per second, the highest sampling rate:

Collect more real information,
Go less to the estimate, and
It generates a much more accurate image of the analog signal.
Logically, the end result is … better sound quality.

Let’s talk about specific data:

Normal sampling frequencies in professional audio range around:

44.1 kHz (audio CD)
48 kHz
88.2 kHz
96 kHz
192 kHz
The minimum of 44.1kHz is due to a mathematical principle known as …

The Nyquist-Shannon Sampling Theorem

To record digital audio accurately, converters have to capture the entire human listening spectrum, which is between 20Hz – 20kHz.

According to the Nyquist-Shannon Sampling Theorem …

To capture a specific frequency, at least 2 samples are needed for each cycle … to measure both the upper and lower points of the sound wave.

That means that recording frequencies of up to 20kHz require a sampling rate of 40kHz or more, which explains why the audio CDs are just above that minimum, at 44.1kHz.

Sound formats and audio normalization

 

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.

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?

► The best turntables of 2019: purchase, configuration and more
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? 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.