Abstract: The drone comprises a camera (14), an inertial unit (46) measuring the drone angles, and an extractor module (52) delivering image data of a mobile capture area of reduced size dynamically displaced in a direction opposite to that of the variations of angle measured by the inertial unit. The module analyses the image data elements of the useful area to assign to each one a weighting coefficient representative of a probability of belonging to the sky, and defines dynamically a boundary of segmentation (F) of the useful area between sky and ground as a function of these weighting coefficients. Two distinct groups of regions of interest ROIs are defined, for the sky area and for the ground area, respectively, and the dynamic exposure control means are controlled as a function of the image data of the ROIs of one of these groups, in particular by excluding the ROIs of the sky area.

Abstract: The drone comprises a camera (14), an inertial unit (46) measuring the drone angles, and an extractor module (52) delivering image data of a mobile capture area of reduced size dynamically displaced in a direction opposite to that of the angle variations measured by the inertial unit. Compensator means (52) receive as an input the current drone attitude data and acting dynamically on the current value (54) of an imaging parameter such as auto-exposure, white balance or autofocus, calculated as a function of the image data contained in the capture area.

Abstract: The invention relates to a method of dynamically determining the duration of exposure for the capture of an image implemented in a drone comprising a substantially vertical-view camera. The method comprises a step (21) of measuring of the horizontal speed of displacement of the drone, a step (22) of measuring the distance between said drone and the ground, and a step (23) of determining the duration of exposure based on the measured speed of displacement of the drone, the distance measured between said drone and the ground, a predetermined quantity of blurring and the focal length of said camera.

Abstract: The drone comprises a camera (14), an inertial unit (46) measuring the drone angles, and an extractor module (52) delivering image data of a mobile capture area of reduced size dynamically displaced in a direction opposite to that of the variations of angle measured by the inertial unit. The module analyses the image data elements of the useful area to assign to each one a weighting coefficient representative of a probability of belonging to the sky, and defines dynamically a boundary of segmentation (F) of the useful area between sky and ground as a function of these weighting coefficients. Two distinct groups of regions of interest ROIs are defined, for the sky area and for the ground area, respectively, and the dynamic exposure control means are controlled as a function of the image data of the ROIs of one of these groups, in particular by excluding the ROIs of the sky area.

Abstract: The headset includes an active noise control, with an internal ANC microphone (28) placed inside the acoustic cavity (22) and delivering a signal including an acoustic noise component. A digital signal processor DSP (50) comprises a feedback ANC branch (54) applying a filtering transfer function (54, HFB2) to the signal delivered by the ANC microphone, and a mixer (60) for mixing the signal of the feedback branch with an audio signal to be reproduced (M). The headset comprises a movement sensor (64) mounted on one of the earphones. The DSP comprises an anti-saturation module (68) for analyzing concurrently i) the signal delivered by the internal microphone (28) and ii) the signal delivered by the movement sensor (64), and verifying whether current characteristics of these signals fulfill or not a set of predetermined criteria. Upstream from the feedback ANC filter (54), an anti-saturation filter (70, HFB1) is selectively switched as a function of the result of this verification.

Abstract: The headset includes an active noise control, with an internal ANC microphone (28) placed inside the acoustic cavity (22) and delivering a signal including an acoustic noise component. A digital signal processor DSP (50) comprises a feedback ANC branch (54) applying a filtering transfer function (54, HFB2) to the signal delivered by the ANC microphone, and means (46) for mixing the signal of the feedback branch with an audio signal to be reproduced (M). The headset comprises a movement sensor (64) mounted on one of the earphones. The DSP comprises means (68) for analysing concurrently i) the signal delivered by the internal microphone (28) and ii) the signal delivered by the movement sensor (64), and verifying whether current characteristics of these signals fulfil or not a set of predetermined criteria. Upstream from the feedback ANC filter (54), an anti-saturation filter (70, HFB1) is selectively switched as a function of the result of this verification.

Abstract: The method comprises the steps of: a) converting the digital audio signal (PCM) into a voltage signal (VE); b) first lowshelf filtering of fixed gain (G2); c) calculating a value of excursion (x) of the loudspeaker; d) comparing the excursion with a maximum value and calculating a first gain of possible attenuation (G3); e) second lowshelf filtering of gain (G4+G3) taking into account the first gain of possible attenuation; g) comparing with the maximum saturation or clipping voltage (vMAX) and calculating a second gain of possible attenuation (G5); h) third lowshelf filtering gain (G6+G3+G5) taking into account the first and/or second gains of possible attenuation; i) comparing with the maximum saturation or clipping voltage (vMAX) and applying a gain of possible overall attenuation (G7); j) optionally compensating for the nonlinearities of the loudspeaker response; and k) reversely converting the signal (Vs) into a digital audio signal (S) without dimension, for later amplification.

Abstract: The loudspeaker enclosure (10) comprises a central channel (16) turned toward the listener, and left and right side channels (18L, 18R) oriented perpendicular to each other. The signal to be reproduced is separated into: i) a mono component correlated between the left and right signals; ii) left and right surround components decorrelated between the left and right signals; iii) a left component decorrelated between the mono component ant the left signal; and iv) a right component decorrelated between the mono component ant the right signal. The central channel and the side channels are piloted by combinations of these components according to a distribution that is a function of the mode of use of the loudspeaker enclosure, alone in front of a listener and as a pair in association with another similar loudspeaker enclosure, the two loudspeaker enclosures being arranged on the left and on the right of the listener.

Abstract: The method comprises: a) the emission of an ultrasound burst repeated at a predetermined recurrence frequency; and b) after each emission and for the duration of a time frame (n?1, n, n+1, . . . ) separating two consecutive emissions, the reception of a plurality of successive signal spikes appearing in the course of the same frame. These spikes include spurious spikes (E?n?1, E?n, E?n+1, . . . ) originating from the emitter of another drone, and a useful spike (En?1, En, En+1, . . . ) corresponding to the distance to be estimated. To discriminate these spikes, the following steps are executed: c) for two consecutive frames, comparison of the instants of arrival of the p spikes of the current frame with the instants of arrival of the q spikes of the previous frame and determination, for each of the p.q pairs of spikes, of a corresponding relative time gap; d) application to the p.

Abstract: The method comprises the steps of: a) converting the digital audio signal (PCM) into a voltage signal (VE); b) first lowshelf filtering of fixed gain (G2); c) calculating a value of excursion (x) of the loudspeaker; d) comparing the excursion with a maximum value and calculating a first gain of possible attenuation (G3); e) second lowshelf filtering of gain (G4+G3) taking into account the first gain of possible attenuation; g) comparing with the maximum saturation or clipping voltage (vMAX) and calculating a second gain of possible attenuation (G5); h) third lowshelf filtering gain (G6+G3+G5) taking into account the first and/or second gains of possible attenuation; i) comparing with the maximum saturation or clipping voltage (vMAX) and applying a gain of possible overall attenuation (G7); j) optionally compensating for the nonlinearities of the loudspeaker response; and k) reversely converting the signal (Vs) into a digital audio signal (S) without dimension, for later amplification.

Abstract: The method comprises the steps of: filtering the audio signal by means of a lowpass filter (101) with a cutoff frequency substantially equal to said cutoff frequency (F0) of the sound playback device; determining a fundamental frequency for reconstituting from the lowpass filtered audio signal; and generating a harmonic signal (Sharm) associated with said fundamental frequency to be reconstituted. It also comprises the steps of: detecting a time envelope (env(t)) of the lowpass filtered audio signal; adapting the dynamic range of said time envelope (env(t)) as a function of the frequency band under consideration; and reinjecting said harmonic signal in phase into said audio signal by addition after multiplying said harmonic signal (Sharm) with the adapted time envelope (envadapt(t)).

Abstract: The method comprises: a) the emission of an ultrasound burst repeated at a predetermined recurrence frequency; and b) after each emission and for the duration of a time frame (n?1, n, n+1, . . . ) separating two consecutive emissions, the reception of a plurality of successive signal spikes appearing in the course of the same frame. These spikes include spurious spikes (E?n?1, E?n, E?n+1, . . . ) originating from the emitter of another drone, and a useful spike (En?1, En, En+1, . . . ) corresponding to the distance to be estimated. To discriminate these spikes, the following steps are executed: c) for two consecutive frames, comparison of the instants of arrival of the p spikes of the current frame with the instants of arrival of the q spikes of the previous frame and determination, for each of the p.q pairs of spikes, of a corresponding relative time gap; d) application to the p.

Abstract: An automatic gain control system applied to a audio signal as a function of the ambient noise, the system comprising: an ambient noise estimator module suitable for establishing a current noise value estimated at least from a signal provided by a microphone; and an automatic gain control module suitable for applying to the audio signal gain of a value that is determined as a function of the current noise value received from the ambient noise estimator module. According to the invention, the ambient noise estimator module comprises an MCRA estimator suitable for establishing the current noise value from a signal provided by the microphone picking up the real noise, the echo of the music, and where appropriate speech. The system also includes a module for estimating the power of the audio signal and suitable for providing the automatic gain control module with a current power value for the audio signal.

Abstract: The device comprises a microphone for detecting a speech signal from a near speaker, and a loudspeaker for reproducing a speech signal from a remote speaker. The processing for canceling the interfering acoustic echo implements an adaptive linear filtering algorithm. Double talk situations are detected by: evaluating an index representative of the convergence or divergence of the algorithm; assessing a predetermined condition for detecting a double talk situation; and if the condition is satisfied, modifying at least one parameter of the algorithm in response to the detection. The representative index may be the norm of the gradient vector describing the adaptation of the filter from one iteration of the algorithm to the next, the conditions being a comparison between the gradient and a threshold. The parameter that is modified double talk situation may be the adaptation stepsize of the algorithm, and also the gain control of an echo suppression stage.

Abstract: The device comprises a microphone for picking up a speech signal from a near speaker, and a loudspeaker for reproducing a speech signal from a remote speaker. The processing for canceling the interfering acoustic echo implements an adaptive linear filtering algorithm. Double talk situations are detected by: evaluating an index representative of the degree of convergence or divergence of the algorithm; assessing a predetermined condition for detecting a double talk situation; and if said condition is satisfied, modifying at least one parameter of the algorithm in response to said detection. The representative index may be the norm of the gradient vector describing the adaptation of the filter from one iteration of the algorithm to the next, the conditions being a comparison between the gradient and a threshold. The parameter that is modified double talk situation may be the adaptation stepsize of the algorithm, and also the gain control of an echo suppression stage.

Abstract: The method comprises the steps of: filtering the audio signal by means of a lowpass filter (101) with a cutoff frequency substantially equal to said cutoff frequency (F0) of the sound playback device; determining a fundamental frequency for reconstituting from the lowpass filtered audio signal; and generating a harmonic signal (Sharm) associated with said fundamental frequency to be reconstituted. It also comprises the steps of: detecting a time envelope (env(t)) of the lowpass filtered audio signal; adapting the dynamic range of said time envelope (env(t)) as a function of the frequency band under consideration; and reinjecting said harmonic signal in phase into said audio signal by addition after multiplying said harmonic signal (Sharm) with the adapted time envelope (envadapt(t)).

Abstract: An automatic gain control system applied to a audio signal as a function of the ambient noise, the system comprising: an ambient noise estimator module suitable for establishing a current noise value estimated at least from a signal provided by a microphone; and an automatic gain control module suitable for applying to the audio signal gain of a value that is determined as a function of the current noise value received from the ambient noise estimator module. According to the invention, the ambient noise estimator module comprises an MCRA estimator suitable for establishing the current noise value from a signal provided by the microphone picking up the real noise, the echo of the music, and where appropriate speech. The system also includes a module for estimating the power of the audio signal and suitable for providing the automatic gain control module with a current power value for the audio signal.