Tag: mp3 volume booster

Loudness equalizer


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Loudness equalizer

Loudness equalizer

Equalizing the volume or loudness of a song or a video is something that all of us who usually save mp3s, mp4s, flac, etc. we usually and want to do.

Loudness equalizer

Why do we want to achieve Loudness equalizer?

In the first place, because normally we have obtained our mp3, mp4, etc. files from very different places. and that implies that they have been encoded with very different bitrates and configurations.

Encoding with different bit rates affects the volume level

Certainly, as this subtitle points out, we could say that using different parameters such as bit rate mainly, but also others, often affects the volume level.

With which we will understand that obtaining numerous audio or video files from different sources will force us later to perform a normalization of the volume level or Loudness equalizer to be able to adjust the volume level or loudness and keep our files the same.

Mp4Gain is the perfect solution with which we will be able to equalize the loudness of our music or videos, achieving that when playing them there are no sudden changes but that they all sound harmonically at the same volume level and with a very similar sound, pleasantly similar.


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Bitrate or Kilobytes per second (kbps)

 

What corresponds to talk now is about the quality of these paintings. Let’s take a frame and see its quality and weight in kb … This is called bitrate or kbps which means KiloBytes per Second or Kilobytes per second, and is the measure used to determine the image quality, the higher the bitrate, the image quality will be better, but the weight of the file will be greater too. There are standards and limits for bitrate, this for the MPEG format. For a DVD, which is an MPEG2 video format, the limitation is more than 1500 and less than 8000 kbps, for VCD it is exactly 1150 kbps, and for example a MiniDV tape of a handycam uses a bitrate of over 30000, this is a lot of quality and when you go to hard disk you see the large file size. For the other formats it depends on the codec and its configuration, being able to compare in quality an MPEG2 format with 3000 kbps with an AVI with Xvid codec at 1500 kbps, this difference is given by the way in which the video is compressed, the Xvid codec It does better than the MPEG2 format.

Now, bitrate can be constant throughout the video or it can be adjusted to changing needs, this is Constant Bitrate (CBR) and Variable Bitrate (VBR), when we use Constant Bitrate all video always uses the same amount of information to show Each frame, whether you need it or not, this can make the video heavier than it could be or that it does not have the right quality in certain parts or has plenty of information where it is not needed.

Let’s see the following example, where a small video clip is shown where it requires a lot of bitrate or kbps and another part where it does not need so much, encoded in MPEG CBR and VBR. We see the difference in the weight of the files and their comparison with an Avi with DivX codec with a lower bitrate. In minute 2 another video is shown where very little bitrate is needed and as when coding in VBR or CBR it does not greatly affect the final result.

analog recording

When we use Bitrate Variable we can define the maximum and minimum to use and let the coding program scan the video and adjust the amount of information as necessary, for this you can use one or two passes (usually no more than two are used), that is, the program goes through the video once, calculates and in the next pass applies the bitrate correctly. It can also be done in a single pass, but the calculation can be wrong, this because to make the necessary bitrate changes makes use of something called keyframes or keyframes. In addition there are programs that detect scene changes and there apply keyframes to change the amount of bitrate occupied, this may be in one pass, but it may also not be as accurate as two passes.

To prevent the file from being excessively complex and heavy to process, when encoding an amount of information is allocated for video blocks, these blocks are defined by the keyframes, for example, there may be 90 frames between each keyframe, and if the video run at 30 frames per second, then the bitrate can change only every 3 seconds of video (3 seconds x
30 frames = 90 frames), and if there is a change of scene, for example, from a dark and slow shot to a scene with light and moving, if we are in the middle of a block between keyframes we will notice that the scene in motion is seen bad, with large video blocks and little definition, this can happen when the coding is done in a single pass using Bitrate Variable, so it is preferred to use 2 passes, in this way the program is responsible for detecting the change of scenes where there is need to adjust the bitrate, and accommodate the keyframes of
correct way, so that in each change the quality of the video is adjusted and seen
all right. And what happens when we have a whole video with movement, well, the bitrate remains high for most of the time, this will make a video that is heavier than another of the same duration but where the content is more static while maintaining good quality . If we use Constant Bitrate for both videos, one with a lot of movement, and another with more static content, the result will be very similar, it will only change the quality of the content, unless we use a very high bitrate to give quality to the video with movement, which it would be excessive for a video of more static scenes, which would generate a heavy file without any need. Therefore, we have videos that can be heavier than others, and videos with different qualities. This is where we need to know what should be adjusted when encoding a video and how automatic adjustments do not always give the same results, this because each video may have different coding needs.

Does bitrate influence? A 320 kbps Mp3 sounds better than a 128 kbps one?

Much has been speculated about the bitrate. Most people do not understand clearly what it is. A few understand, but almost nobody knows if a file with 320 kbps really sounds different or better than the same file but with 128 kbps.

The easiest way is to test:

The first is at 128 kbps

Now let’s hear the 320 kbps option

Notice the difference? In case the note is because it was encoded using the Mp4Gain.
Normally it is almost impercentible, but using a good encoder you get to notice some subtle difference.

It should be taken into account that at higher kbps, if there is a higher quality – although it is not always noticeable – and will always use more disk space.

Therefore it is not the best option to say “all my mp3s will be 320 kbps”, unless the space does not mean any problem at all.

Normalize audio: Normalization and normalizer

Normalization – Normalize audio

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 normalizer is a device that falls into the category of dynamics processors.

Analyze the target signal, detect its highest volume peak and increase its gain to the maximum level possible without distorting.
With the same proportion the level of the rest of the signal increases.
The signal, in general, will sound with a greater volume.

It is very unpleasant to have a playlist with mp3 files playing at different volumes. However, it is possible to normalize (equalize) the volume of our mp3s.

Loudness normalizer

You can use the Loudness Normalizer to obtain a specific loudness.

Increasing the loudness to a specific value may cause clipping. To solve it, a peak limiter can be part of the process. The Loudness Normalizer increases the loudness and limits the peaks of the signal at the same time if necessary, to obtain the desired loudness.

This process takes place in several stages, first an analysis and then the final rendering.

What is an audio compressor?

When we talk about compression we also talk about dynamic range. Recall that the dynamic range is the difference in amplitude between the lowest and highest part of a signal. In the compression process, basically and technically it consists in decreasing the dynamic range of a signal.

These are the most common controls or points to control in an audio compressor:

Threshold: This is the point from which the compressor will start working. Any signal that exceeds this point will be compressed.

Ratio: This is the reason for the compression that will be applied to the signal that exceeds the threshold. We see it and find it as 2: 1, 4: 1, etc. The first number means the number of input decibels that exceed the threshold, and the second number the output decibels. In other words, for “X” number of decibels that exceed the threshold, “Y” number of decibels will be output.

Attack: It is the time it takes for the signal to compress after having exceeded the threshold. We see it and find it commonly expressed in milliseconds (ms).

Release: How soon the compressor stops working after the signal falls below the threshold.

Knee: This is a parameter that subtly modifies the compressor threshold. A low setting (Hard knee or 0) means that the compressor will act only from the set threshold. A high setting (Soft knee) will allow the compressor to act gradually from before the signal reaches the threshold. In this way, the threshold can be treated as a range and not as a specific point. As the signal approaches the threshold compression increases. When it exceeds it, it continues to increase until the entire compression ratio is applied.

What it is and how to perform a volume normalization on your MP3

 

What it is and how to perform a volume normalization on your MP3

Have you ever heard the term audio normalization, without being sure of what it meant? As a lover of music and technology, I also encountered such a doubt many years ago. Basically, giving a short definition, it is about the standardization of the volume, or rather, of the audio spectrum with respect to other subjects, usually of the same disc.

And that, to put it more simply, is the equalization of the volume of the different tracks on a disc. The reasons are many, and usually if the tracks are extracted from the same job they already have the same volume and gain, but what happens if we want to make a mixtape? For example, we decided to make a compilation called The Best 100 Rock Songs in History. Surely have songs from The Beatles or The Rolling Stones, and therefore from different albums. Depending on the year, type of mastering, etc. etc., we can end up with a CD that contains many different volumes, something that can be annoying when listening. That is just one of the reasons to normalize our MP3 collection.

There are add-ons for players that allow us to normalize on the fly. In fact we can say that programs like Spotify already do this by means of the option to equalize volume of all the songs, however the application that I present below allows us to permanently normalize modifying MP3 files and many other formats, both audio and Of video..

This is Mp4Gain, which stands out for its simplicity of use and is presented under an interface that is ideal to understand exactly what a normalization is and see the before and after. When we open the application we find a window in which we have a grid, which will be populated when we add files or folders, and a keypad with various options.

How do we normalize? Simply change the gain through the specific menu for this.

By pressing OK the application will start working and save our files with the same gain, so it is ideal that before doing the first tests we make a backup. It must also be taken into account that it is an operation that can take time, something that depends on the speed of our processor, the number of issues to normalize and also the size and quality of them.

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.

What is an audio compressor.

In the field of professional sound, a compressor is an electronic sound processor designed to reduce the dynamic range of the signal without noticing its presence too much. This task is done by reducing the system gain, when the signal exceeds a certain threshold.

Traditionally, compressors have been electronic equipment with one or two rack units, but software versions of them have appeared for some years.

A compressor acts in such a way that it attenuates the electrical signal by a certain amount (normally measured in decibels) and from a certain input level. The objective is to ensure that the resulting dynamic excursion is lower than the original, to protect certain equipment against possible signal peaks or, if it is a saturated sound, to try to hide the error.

Reasons to compress a signal

-Control the energy of the signal: The human ear is very sensitive, so the compression must be smooth and subtle so as not to capture it. This type of compression is used when there is a signal in which the intensity varies, so it is compressed to achieve a more constant signal within the values ​​assigned to it.

-Control the peak level of the signal: Often the equipment is limited, so the amplifiers can saturate and therefore be damaged. In this case the compression is used to control the signal and thus protect the equipment.

-Reduce the dynamic range of the signal: By attenuating the peaks of a signal, we reduce its dynamic range. Many devices are limited by the peaks, and this allows the RMS level of the signal to be raised.

Compressor Uses

In the field of music, its use ranges from applications for musical recordings to live sound. For example, it is often used to add more glued to the sound, an effect that is achieved by compressing the signal to subsequently apply a gain to the output of the device, which usually conceals possible interpretation failures by the artist, at least as Dynamic control refers. A compressor is highly recommended (and with certain musical styles, indispensable) for when using an electric bass. The slapping effect (hitting the strings with the finger) produces extremely high output peaks (20 dB or 10 times more than normal), which at low output levels generate distortion, and at high volumes (as in recitals) they can cause serious damage to the amplifier, and even the speaker (an excess of “excursion” can cause the speaker to tear from its suspension). Even in the (theoretical) case of a musical system with an infinite dynamic range, the difference, auditory speaking, using or not the compressor is imperceptible. Its use is also very frequent in voices, since not all singers use the appropriate technique so the signal level varies constantly.

-It is widely used in broadcasting, to improve the speaker’s diction.
-Compress during mastering improves the sound definition of the final mix.
-To protect the equipment (speakers).

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.