Method and system for optimizing the low-frequency sound rendition of an audio signal

Abstract

A system and method for optimizing the low-frequency sound rendition of an audio signal, implementing variations in a plurality of parameters of the audio signal according to the volume level of the signal chosen by a user, in particular filtering or compression parameters, or parameters relating to the harmonics of the audio signal, while seeking to optimize the dynamics and the bandwidth of the audio signal according to the volume, in order to provide an optimal rendition to the user.

Claims

1. A system for optimising the low-frequency sound rendition of an audio signal, characterised in that it comprises means for varying a plurality of compression parameters of said audio signal according to the volume level of said audio signal chosen by a user, while seeking to optimise the dynamic range and the bandwidth of said audio signal according to the volume, in order to provide an optimal rendition to the user, comprising a plurality of gain modules, a harmoniser comprising a pre-harmonic filter, a harmonic generator, a post-harmonic filter, and further comprising a crossover module composed of two filters of the type IIR (Infinite Impulse Response) and a compression block comprising a speaker cut sub-module constituted of a biquad filter.

2. The system for optimising the low-frequency sound rendition of an audio signal according to claim 1, characterised in that said parameters are included in the group consisting of: the compression rate, the release time, the attack time, the recovery time, the threshold and the gain compensation.

3. The system for optimising the low-frequency sound rendition of an audio signal according to claim 1, characterised in that it comprises means for adjusting the characteristics of a high-pass filter according to the input level of said audio signal.

Description

BRIEF DESCRIPTION OF THE DRAWINGS

(1) The present disclosure will be better understood after reading the description, provided for illustration purposes only, of one embodiment of the present disclosure, with reference to the FIGURES, in which:

(2) FIG. 1 shows the system according to the present disclosure, in one embodiment thereof.

DETAILED DESCRIPTION

(3) The present disclosure relates to a method for optimising the low-frequency sound rendition of an audio signal, implementing variations in a plurality of parameters P1, P2, P3, . . . , PN of said audio signal according to the volume level V1, V2, . . . , VP of said audio signal chosen by a user, in particular filtering or compression parameters, or parameters relating to the harmonics of said audio signal, while seeking to optimise the dynamic range and the bandwidth of said audio signal according to the volume, in order to provide an optimal rendition to the user.

(4) As opposed to the methods and systems known in the prior art, the method according to the present disclosure varies numerous parameters of the audio signal in order to provide an optimal rendition to the user.

(5) Within the scope of the method according to the present disclosure, the aim is to optimise the dynamic range and the bandwidth of said audio signal according to the volume, in order to provide an optimal rendition to the user, at both a low and high volume, without being detrimental to the power handling of the system for reproducing said audio signal.

(6) This plurality of parameters is adjusted, then stored by a person skilled in the art, then retrieved by the reproduction system as a function of the volume level chosen by the user.

(7) Thus, the dynamic range is increased at a low volume level, while protecting the reproduction system at a high volume level.

(8) For example, for a volume variation ranging from 0 (minimum volume=−100 dB of attenuation) to 30 (maximum volume=0 dB of attenuation), seven steps are considered here for adjusting the parameters of the elements in FIG. 1. The intermediate parameters are interpolated here.

(9) With reference to FIG. 1, the following table is obtained:

(10) TABLE-US-00001 TABLE 1 Pa- ram- Volume level eters 0 5 10 15 20 25 30 14 14 × 0 14 × 5 14 × 10 14 × 15 14 × 20 14 × 25 14 × 30 40 40 × 0 40 × 5 40 × 10 40 × 15 40 × 20 40 × 25 40 × 30 41 41 × 0 41 × 5 41 × 10 41 × 15 41 × 20 41 × 25 41 × 30 42 42 × 0 42 × 5 42 × 10 42 × 15 42 × 20 42 × 25 42 × 30 16 16 × 0 16 × 5 16 × 10 16 × 15 16 × 20 16 × 25 16 × 30 17 17 × 0 17 × 5 17 × 10 17 × 15 17 × 20 17 × 25 17 × 30 51 51 × 0 51 × 5 51 × 10 51 × 15 51 × 20 51 × 25 51 × 30 52 52 × 0 52 × 5 52 × 10 52 × 15 52 × 20 52 × 25 52 × 30 19 19 × 0 19 × 5 19 × 10 19 × 15 19 × 20 19 × 25 19 × 30 61 61 × 0 61 × 5 61 × 10 61 × 15 61 × 20 61 × 25 61 × 30 62 62 × 0 62 × 5 62 × 10 62 × 15 62 × 20 62 × 25 62 × 30 21 21 × 0 21 × 5 21 × 10 21 × 15 21 × 20 21 × 25 21 × 30 22 22 × 0 22 × 5 22 × 10 22 × 15 22 × 20 22 × 25 22 × 30

(11) It should be noted that the parameters mentioned in the left-hand column of Table 1 above correspond to the blocks shown in FIG. 1.

(12) If thirty adjustment steps are considered, a plurality of parameters (14×0, 14×1, 14×2, 14×3, etc.) thus corresponds to each volume level.

(13) In one embodiment, said parameters P1, P2, P3, . . . , PN are included in the group consisting of: the compression rate, the release time, the attack time, the recovery time, the threshold and the gain compensation.

(14) In one embodiment, said method comprises a step of adjusting the characteristics of said high-pass filter F according to the input level of said audio signal.

(15) The present disclosure further relates to a system for optimising the low-frequency sound rendition of an audio signal, comprising means for varying a plurality of parameters P1, P2, P3, . . . , PN of said audio signal according to the volume level V1, V2, . . . , VP of said audio signal chosen by a user, in particular filtering or compression parameters, or parameters relating to the harmonics of said audio signal, while seeking to optimise the dynamic range and the bandwidth of said audio signal according to the volume, in order to provide an optimal rendition to the user.

(16) In one embodiment, said parameters P1, P2, P3, . . . , PN are included in the group consisting of: the compression rate, the release time, the attack time, the recovery time, the threshold and the gain compensation.

(17) In one embodiment, said system comprises means for adjusting the characteristics of said high-pass filter F according to the input level of said audio signal.

(18) FIG. 1 shows the system according to the present disclosure, in one embodiment thereof.

(19) FIG. 1 shows that the system according to the present disclosure comprises an input signal 10 and an output signal 30.

(20) The input signal 10 is composed of a left channel 11 and a right channel 12.

(21) The output signal 30 is composed of a front left channel 31, a front right channel 32, a back left channel 33 and a back right channel 34.

(22) The system according to the present disclosure comprises a so-called “by-pass” element 13, a delay module 14 and a harmoniser 15.

(23) The system according to the present disclosure comprises two gain modules: a so-called “dry gain” module 16 and a so-called “wet gain” module 17.

(24) The system according to the present disclosure further comprises a so-called “crossover” module 18, a second delay module 19 and a compression block 20.

(25) Finally, said system according to the present disclosure comprises two further gain modules, a front low-frequency gain module 21, and a rear low-frequency module 22, situated before the output.

(26) The harmoniser 15 is composed of: a pre-harmonic filter 40 of the type IIR (Infinite Impulse Response) constituted in one example embodiment of two identical biquad low-pass filters; a harmonic generator 41 (if the input signal into this module is negative, the output signal of this module is zero); and a post-harmonic filter 42 of the type IIR (Infinite Impulse Response) constituted in one example embodiment of one biquad high-pass filter and of two identical biquad low-pass filters.

(27) The so-called “crossover” module 18 is composed of: a so-called “crossover high” sub-module 51: a filter of the type IIR (Infinite Impulse Response) constituted in one example embodiment of two identical biquad high-pass filters; and a so-called “crossover low” sub-module 52: 51: a filter of the type IIR (Infinite Impulse Response) constituted in one example embodiment of two identical biquad low-pass filters.

(28) Said compression block 20 is composed of: a so-called “speaker cut” sub-module 61 constituted of a biquad filter; and another sub-module 62, which is a stereo compressor, comprising adjustable parameters such as the threshold, the compression slope, the expansion slope, the attack time, the release time and the gain compensation.

(29) The above description of the present disclosure is provided for the purposes of illustration only. It is understood that a person skilled in the art can produce different variations of the present disclosure without leaving the scope of the patent.