volume increaser for mp3 Archives

Mp3 volume increaser, Mp3 Volume Booster

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Mp3 volume increaser, Mp3 Volume Booster

The manipulation by users when encoding music into formats such as mp3, with the intention and the need to save space on the hard drive… but that caused the domestic user to choose which bitrate, samplerate, etc. I was going to use, it meant a disparity not only in the volume level, a dilemma between the volumes of the different mp3s, but in general problems of large differences in quality.

Mp4Gain is obviously the most appropriate answer, it is the quintessential software that is used to achieve, not only achieve an mp3 volume increaser or mp3 volume booster, but in general a standardization of the sound level in terms of sound in general. You can even equalize them all so that they have a very similar “color”.

Mp4Gain has the most efficient algorithm to optimize and normalize the volume on the market, but also its approach to the problem is so sophisticated and with the latest technology, according to the technology of current playback devices, that it achieves a quality that is unrivaled.

Mp4Gain is the product of many years of advanced research that has matured with each new version.

You can also acquire it with a lifetime license or with a payment plan every four months (extremely cheap), whichever is more comfortable for you and your needs.

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Mp3 volume increaser, volume equalizer

It’s been a while since some people say or post that mp3 is dead.

It is an impulsive and hardly credible assessment.
The mp3 emerged as an urgent need to save space, because the WAV file is very large and there were those who paid attention to the functioning of the human ear and soon discovered some things they could do to save space, in addition to zip compression.

He first noticed that there was a lot of information about frequencies that we humans can’t hear and that this information took up a lot of space, so removing it would not affect the sound quality and would save space.

Then I noticed that there is a phenomenon called audio masking which meant that if we hear a sound in a range of frequencies and seconds or milliseconds later we get another sound in nearby frequencies, we won’t hear the second ones… so we can remove them. .. and so, there is a whole series of phenomena typical of human hearing that are used by mp3 in an intelligent way.

But there are also the compression settings that each one uses, such as the bit rate and that many times DOES affect the sound quality.

Experiments with professional musicians have shown that even a trained ear cannot distinguish a GOOD mp3 from a lossless file.

Mp4Gain is capable of generating high quality mp3, mmp4, flac, wav, etc.

Mp3 volume increaser

Why is everyone looking to increase the volume of their mp3s?

Mp3 volume increaser

In reality, they also seek to increase the content of their mp4 videos and other audio formats such as FLAC, etc.

And it’s not that people have gone crazy and just want to increase the volume of their mp3s for a fad. No way. Actually it is a necessity, because it often happens that one or more of the mp3s sounds with a noticeably different volume, lower or higher.

This is the same thing that happens with videos, where one or several mp4 clearly sound at a different volume and it is necessary to normalize the volume gain to avoid this problem and make them all sound at the same level of loudness.

Mp4Gain is the solution. It is definitely the program that best achieves this. It definitely has the best standardization algorithm in the whole world, for a reason radio and TV broadcasting stations, music producers, musicians, etc. buy Mp4Gain !
It is very simple to use but if you are an advanced user it offers many options, but if you are a beginner user, you will only have to click a button and you will have managed to normalize the volume of all your video and music files.

What is the MP3 compression principle?

Mp3 Volume Booster

In fact, there are many audio compression technologies and MP3 compression technology is not the best.

Volume Booster

But now it seems that it is still mainstream.
Musical signals have many redundant components, including spacing and information that the human ear cannot distinguish (such as weak signals mixed with a strong background). The CD sound is not compressed and uses a fixed sampling frequency of 44.1 kHz, which can ensure good playback of maximum dynamic music. Of course, the amount of data is the same where the amount of information is less, so there is a possibility of compression. The audio bandwidth of 20 ~ 20 kHz (upper CD player can be extended up to 2 Hz) has become the current music standard. To reduce sound distortion, MP3 adopts an encoding algorithm called “sensory encoding technology”: the audio file first undergoes spectral analysis during encoding, and then the noise level is reduced by filter, and then the remaining components are quantized by quantization The bits are scattered and arranged, and finally an MP3 file with a higher compression ratio is formed, and the compressed file can achieve a sound effect closer to the original sound source during playback . Although it is a lossy compression, its biggest advantage is very little sound distortion in exchange for a higher compression ratio. And now MP3 adopts a variable compression ratio (VBR) compression technology similar to Dolby AC-3. The sampling compression ratio depends on the amount of information in the music, and the masking effect of the human ear is used to reduce redundant data.

Mp3 Increase Volume Part 3

In the two previous parts about Mp3 Volume Increaser, we have seen how and why it is necessary to normalize the volume of audio and video files (something that only Mp4Gain can do) and for them we have begun to delve into how the compression.

Increase MP3 Volume

Audio compression algorithms

coding

Audio compression technology refers to the application of suitable digital signal processing technology to the original digital audio signal stream (PCM encoding), without losing the amount of useful information, or under the condition that the loss introduced be insignificant, reduce (compress) its code rate, and also called compression encoding. It must have a corresponding inverse transform, called decompression or decoding. Audio signals can introduce a great deal of noise and some distortion after passing through a codec system.

1. Redundant audio signal information
Digital audio signals, if transmitted directly without compression, would consume a large amount of bandwidth. For example, if the sample rate of a two-channel digital audio set is 44.1 KHz and each sample value is quantized to 16 bits, its code rate is:

2*44.1kHz*16bit = 1.411Mbit/s

Such a large bandwidth will bring a lot of difficulties for signal transmission and processing, so the audio data must be processed with audio compression technology to transmit audio data effectively.

Digital audio compression coding compresses the audio data signal as much as possible on the premise of ensuring that the signal is not audibly distorted. Digital audio compression coding is implemented by removing redundant components in sound signals. So-called redundant components refer to signals in the audio that cannot be perceived by the human ear and do not help determine the timbre, pitch, and other information of the sound.

Redundant signals include audio signals outside the range of human hearing and masked audio signals. For example, the frequency range of the sound signal that can be perceived by the human ear is 20 Hz to 20 KHz, and frequencies other than this frequency cannot be detected by the human ear and may be considered as redundant signals. In addition, according to the physiological and psychoacoustic phenomena of the human ear, when a strong signal and a weak signal exist at the same time, the weak signal will be masked by the strong signal and cannot be heard, so the weak signal can be regarded as a redundant signal. Do not send. This is the masking effect of human hearing, which is mainly manifested in the spectral masking effect and the time-domain masking effect, which are presented as follows:

1.1 Spectral masking effect
After the sound energy of a frequency is lower than a certain threshold, the human ear will not hear it, and this threshold is called the minimum audible threshold. When there is another sound with higher energy, the threshold value close to the frequency of the sound will increase considerably.

Mp3 Increase Volume Part 2

mp3 increase volume

We talked in the previous chapter about the need to achieve an mp3 volume increaser, although the volume increaser is not limited to mp3s in Mp4Gain. You can actually achieve a volume increaser in all major audio and video formats.

mp3 increase volume

To better understand, we were talking about and analyzing how the compression of an audio file works.

Quantification encoding. Quantization encoding uses a three-layer iterative loop model for bit allocation and quantization. These three layers include: frame loop, outer loop, and inner loop. The frame loop resets all iterative variables, calculates the maximum number of bits that can be provided to each data slice, and then calls the external iterative model; the outer iterative model first uses the inner iterative model, which quantizes the input vector by incrementing The size of the quantization step allows the quantized output to be encoded within a certain bit limit. Huffman coding has a limit on the maximum quantization value, so it is necessary to judge whether all quantization values exceed the limit. If it exceeds the limit, the inner iteration loop must increase the size of the quantization step and quantize again. Then determine the number of Huffman encoding bits such that the number of occupied bits is less than the maximum number of bits that can be provided by each encoding section computed by the frame cycle; otherwise, the size of the quantization step must be increased for requantization. When the quantization meets the requirements, store the final scale factor value, exit the outer loop, and compute the number of bits used to store each data section in the frame loop.

Mp3 Increase Volume

mp3 increase volume

First we need to understand how an mp3 works, how it compresses the original wav to a tenth of its size, to understand why we always need to normalize all the volumes of the different mp3s.

increase mp3 volume

Mp3 Increase Volume

A wav saves a lot of information, including information that humans cannot hear, also redundant information and ends up taking up a lot of space on the hard drive or on flash drives.

This has generated from the beginning the need to ensure that the different mp3s have a constant or similar volume when comparing some mp3s with others.

This currently with Mp4Gain can be applied to the most used audio and video formats.

Let’s understand a little how the compression of an mp3 works:

MP3 encoding is mainly composed of 3 main functional modules, including hybrid filter bank (subband filter and MDCT), psychoacoustic model, quantization encoding (bit mapping and bit factoring, and Huffman encoding).
1. Hybrid filter bank. This part includes two parts of the subband filter bank and MDCT. Subband filterbank encoding completes the mapping of the sampled signal from time domain to frequency domain and decomposes the specified audio signal into 32 subbands through the bandpass filterbank for output. All 32 subbands output by the subband filter bank have the same bandwidth, while the critical bandwidth derived from the psychoacoustic model is not. Therefore, to match each scaling factor band for encoding to the critical band, it is necessary to transform subband signals into MDCTs. Once the output of the subband filter bank is sent to the MDCT filter bank, each bank is subdivided into 18 frequency lines, resulting in a total of 576 frequency lines. Next, the number of bits allocated to the 576 spectral lines is determined using the signal-to-mask ratio of the computed subband signals in the psychoacoustic model.

2. Psychoacoustic models. The psychoacoustic model takes advantage of the masking effect of the human auditory system to remove a large number of irrelevant signals, in order to achieve the effect of compressing the audio data. To accurately calculate the masking threshold, the signal is required to have better resolution in the frequency domain, so the signal is Fourier transformed before using the psychoacoustic model. MPEG-I provides two psychoacoustic models. The first model is easy to compute and provides adequate accuracy when encoding at high bit rates. The second model is more complex and is generally used when encoding at lower bit rates. Psychoacoustic model II is generally used in MP3 encoding. The purpose of the psychoacoustic model is to find the masking threshold value of each subband and use it to control the quantization process. The implementation process of the psychoacoustic model generally consists of first using FFT to obtain the spectral characteristics of the signal and find the tonal components (some called musical components) and the non-tonal components (or noise components) at each frequency point according to the spectral features; The curve determines the masking domain value of each tonal component and non-pitch component at other frequency points; finally, the general masking domain of each frequency point is obtained and converted to the coding subband. For noise generated after quantization of the spectral value emitted by the subband filter bank, if the noise can be controlled below the masking threshold, the final compressed data decoded result may be indistinguishable from the original signal. The masking ability of a given signal depends on its frequency and loudness, so the end result of the psychoacoustic model is the signal-to-mask ratio (signal to mask ratio), which is the ratio of the intensity of the signal and masking. limit.