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.
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.
