EP2134105B1 - Dispositif de traitement audio et procédé pour effectuer un traitement de correction de caractéristiques de fréquence pour un signal d'entrée audio - Google Patents

Dispositif de traitement audio et procédé pour effectuer un traitement de correction de caractéristiques de fréquence pour un signal d'entrée audio Download PDF

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Publication number
EP2134105B1
EP2134105B1 EP08010723.8A EP08010723A EP2134105B1 EP 2134105 B1 EP2134105 B1 EP 2134105B1 EP 08010723 A EP08010723 A EP 08010723A EP 2134105 B1 EP2134105 B1 EP 2134105B1
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EP
European Patent Office
Prior art keywords
signal
audio
frequencies
room
circuit
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EP08010723.8A
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German (de)
English (en)
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EP2134105A1 (fr
Inventor
Tadeo Spraggon-Hernandez
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Alpine Electronics Inc
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Alpine Electronics Inc
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/307—Frequency adjustment, e.g. tone control
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/03—Synergistic effects of band splitting and sub-band processing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/13—Acoustic transducers and sound field adaptation in vehicles
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/07—Synergistic effects of band splitting and sub-band processing

Definitions

  • the present invention is directed to an audio processing device configured to receive an audio input signal from an audio source, and comprising an equalizing circuit configured to execute frequency characteristic correction processing for a received audio signal and an output circuit coupled to an output of the equalizing circuit and configured to supply an audio output signal to a speaker for audio output from the speaker. Further, the invention is directed to a method of performing frequency characteristic correction processing for an audio input signal.
  • an electrical signal from an audio source having audio information content (hereinafter referred to as an "audio signal") is provided to one or multiple speakers which transform the received audio signal into sound.
  • an audio system installed in a room environment, the sound is reflected by reflecting objects such as a wall of the room, wherein sound is reflected back into the room and interferes with sound directly emitted from the speakers at the position of the listener, which may cause coloration of the sound heard by the listener.
  • an audio system may use an equalizing circuit configured to execute frequency characteristic correction processing for a received audio signal.
  • Such equalizing circuit is typically used for filtering and/or correcting an overall frequency response of the audio signal having specific signal frequencies in order to exert influence on the sound spectrum at the position of the listener, for example, to compensate for reflection effects as mentioned above.
  • room parameters of a room environment which influence sound reflection are not known prior to installation of the audio system since such room parameters vary depending on a particular environment in which the audio system may be installed. Varying room parameters may be, for example, a result of varying distances of opposed walls of different room environments in which the audio system may be installed, or different absorbing/reflecting objects present in the room. In this respect, it is known in the art to measure such frequency response between speaker and listener position in the particular audio system installation and to set a respective filter parameter of the equalizing circuit accordingly.
  • An output circuit is coupled to an output of the equalizing circuit and supplies an audio output signal having a correspondingly corrected frequency characteristic to the speakers for audio output from the speakers.
  • Room modes can be identified by peaks and dips in the frequency response of the acoustic transfer function between speaker and listening position, typically only at low frequencies (e.g., below 150 Hz) where their density is not too high over space.
  • Room modes are caused by the formation of standing waves inside a room or cavity, like in a vehicle or car cabin. Standing waves will appear in cases where the wavelength of the sound wave is on the order of the room dimension (half wavelength relationship) and also reflections, e.g. at room walls, are present.
  • FIG. 1 An example of a particular room mode for a given cubic room or cavity is shown in Fig. 1 .
  • the room R comprises a distance D between the opposed walls W which is a multiple of ⁇ /2 of the wavelength ⁇ of the room mode RM.
  • a room mode may significantly cause sound coloration at the listener's position since room modes may cause amplifying or attenuation effects for signal components at the particular room mode frequency.
  • Fig. 2 shows an exemplary room mode measurement over time, taken from "Comparison of Modal Equalizer Design Methods", Poju Antsalo1, et. al., AES Convention 2003 March 22-25, page 5 .
  • Fig. 2 shows that all room modes decay over time within a particular so-called decay time, which is in the present example approximately 500ms. It is also possible to see that several modes are present and they decay also differently in the room, thus each having a different decay time. The highest peaks determine the dominating modes in the room.
  • onset time of a room mode which indicates the time for forming the corresponding room mode upon emitting a particular sound signal which forms the corresponding room mode.
  • Such onset time of a room mode is typically in the order of the above mentioned decay time for that room mode.
  • the above mentioned equalizing circuit may be used, for example, to filter or correct signal components of the audio signal having frequencies where room modes occur in order to compensate for the amplifying effect of the corresponding room mode at the listener's position.
  • signal components of the audio signal may be filtered and/or corrected so as to gain a flattened or "equalized" signal response over the frequency band without peaks or dips.
  • frequency characteristic correction processing or equalizing processing herein referred to as equalizing compensation or EQ compensation in the following
  • equalizing compensation or EQ compensation may negatively affect the sound at low frequencies as a matter of steady state EQ compensation of room modes. Sound like short drum or short bass sounds are affected by the EQ compensation in the same manner as longer lasting low frequency sound, such as long (e.g. lasting 800 ms or longer) and deep sound produced by a pipe organ, which excites and creates a corresponding room mode. Therefore, conventional EQ compensation is a compromise between the optimization for steady state (long lasting sound) and the dynamic case (short lasting sound).
  • a compensation system may include an aggregate of parametrically tuned filters wherein, generally, some of these filters are resonant and tuned to suppress counter-productive sound reproduction that would occur from motion breakup, standing waves and other resonant behaviour.
  • an audio processing device is provided which comprises a dynamic parameter selector which is connected to a detector through which, for example, dynamic changes can be identified.
  • the detector provides dynamic information about input data wherein the applicable content of the data may be determined and applicable dynamic parameter changes may be made by the dynamic parameter selector.
  • JP 2000-261900 A an acoustic device using a steady state sound field correction method is provided.
  • microphones are installed at optional positions in a room, multiple speakers measure acoustic characteristics at the positions, analyze the measured values and are installed at prescribed positions in the room.
  • a system control circuit controls audio signals supplied to the speakers on the basis of the analyzed value, and controls a standing wave at an optional listening position so as to correct the frequency characteristic thereby making the frequency characteristic flat at the listening position through changing the frequency characteristic in the room.
  • the sound pressure level at the measured position can closely be made constant thereby reducing a sound pressure difference within a listening plane.
  • the invention is directed to an audio processing device according to the features of claim 1. Further, the invention is directed to a method of performing frequency characteristic correction processing for an audio input signal according to the features of claim 10. Embodiments and advantageous features of the invention are mentioned in the dependent claims.
  • examples of an audio processing device comprise an input terminal configured to receive an audio input signal from an audio source, an equalizing circuit having an input and an output, wherein the input is coupled to the input terminal and the equalizing circuit is configured to execute frequency characteristic correction processing for a signal received at the input, and an output circuit coupled to the output of the equalizing circuit and configured to supply an audio output signal to a speaker for audio output from the speaker.
  • a signal analyzing circuit is coupled to receive the audio input signal and is configured to analyze a frequency characteristic of the audio input signal for detecting the presence of a signal power of the audio input signal at at least one frequency of a room mode.
  • the signal analyzing circuit is further coupled to a control terminal of the equalizing circuit for dynamically adapting the frequency characteristic correction processing at the at least one frequency of a room mode over time when the presence of the signal power of the audio input signal at the at least one frequency of a room mode is detected.
  • examples of the invention include a method of performing frequency characteristic correction processing for an audio input signal, comprising the steps of analyzing a frequency characteristic of an audio input signal for detecting a presence of a signal power of the audio input signal at at least one frequency of a room mode, performing, in an equalizing process, frequency characteristic correction processing for the audio input signal, dynamically activating and deactivating the frequency characteristic correction processing at the at least one frequency of a room mode over time if the presence of the signal power of the audio input signal at the at least one frequency of a room mode has been detected, and supplying from the frequency characteristic correction processing an audio output signal to a speaker for audio output from the speaker.
  • the equalizing processing may be dynamically adapted depending on the type of sound. Sound like short drum or short bass sound may not be affected or may be less affected by the EQ compensation, whereas longer lasting low frequency sound is subjected to EQ compensation to reduce the negative effects of room modes.
  • the invention accounts for the effects that short lasting low frequency sound, in contrast to long lasting low frequency sound, will not excite or create room modes so that, in this case, the corresponding signal components of the audio signal having these frequencies will not be subjected to EQ compensation which, if applied, may distort the sound in the absence of a corresponding room mode.
  • the signal analyzing circuit stores a predetermined onset time for at least one frequency of a room mode. If the signal analyzing circuit detects the presence of the signal power of the audio input signal at the at least one frequency of a room mode, the signal analyzing circuit determines a duration of the detection of the presence of the signal power and compares the duration with the stored onset time of the corresponding room mode. If the duration is equal to or greater than a predetermined time threshold, the signal analyzing circuit controls the equalizing circuit for activating the frequency characteristic correction processing for the at least one frequency of a room mode.
  • the signal analyzing circuit further determines a signal power of the audio input signal at the at least one frequency of a room mode compares the determined signal power with a signal power threshold value, wherein if the determined signal power is equal to or greater than the signal power threshold value, the signal analyzing circuit controls the equalizing circuit for activating the frequency characteristic correction processing at the at least one frequency of a room mode. In this way, it is accounted for the effect that sound which exceeds a particular signal power will more likely excite a room mode than sound which is lower than the signal power threshold.
  • the signal analyzing circuit stores a predetermined decay time for at least one frequency of a room mode.
  • the signal analyzing circuit further determines a signal power of the audio input signal at the at least one frequency of a room mode and compares the determined signal power with a signal power threshold value. If the signal analyzing circuit has previously controlled the equalizing circuit for activating the frequency characteristic correction processing at the at least one frequency of a room mode, the signal analyzing circuit determines whether the determined signal power is lower than the signal power threshold value.
  • the signal analyzing circuit controls the equalizing circuit for deactivating the frequency characteristic correction processing for the at least one frequency of a room mode after the decay time has elapsed counted from the point of time of when the signal analyzing circuit has determined that the determined signal power is lower than the signal power threshold value.
  • the signal analyzing circuit stores a table containing multiple frequencies of respective room modes, wherein the table associates a respective predetermined onset time and a respective predetermined decay time of the room modes with a corresponding one of the multiple frequencies.
  • the audio processing device may further include a buffer circuit coupled between the input terminal and the equalizing circuit for successively storing a number of samples of the audio input signal, wherein the signal analyzing circuit is coupled to the buffer circuit for analyzing a frequency characteristic of the audio input signal.
  • the buffer circuit has a length which is capable to store samples of the audio input signal during a period of time which is in the order of a room mode onset or decay time.
  • the audio processing device of the invention also includes a low pass filter coupled between the input terminal and the equalizing circuit and having an output providing an output signal to the equalizing circuit containing only lower frequencies of the audio input signal.
  • a high pass filter may be coupled between the input terminal and the output circuit and in parallel to the series connection of the low pass filter and equalizing circuit, wherein the output circuit comprises a summation circuit for adding a signal provided from the high pass filter and a signal provided from the equalizing circuit for supplying the audio output signal to the speaker.
  • the audio processing device may be installed in an interior vehicle environment, such as inside a car cabin.
  • the signal analyzing circuit stores a table containing multiple frequencies of room modes determined by measurement in the vehicle environment. In this way, an audio system may be adapted to the particular circumstances of a car environment.
  • At least one calibration signal may be transmitted into a room environment, such as the car cabin, the calibration signal containing a predetermined frequency spectrum.
  • a signal response particularly a frequency response of the acoustic transfer function between speaker and listening position, is measured for determining frequencies of room modes.
  • a table is stored containing the determined frequencies of room modes which are used, in an operating stage, for the activating and deactivating of the frequency characteristic correction processing.
  • Fig. 3 shows a basic block diagram of an audio processing device according to the invention.
  • the audio processing device 1 comprises an input terminal 2 configured to receive an audio input signal 20 from an audio source which is not shown in Fig. 1 .
  • a buffer circuit 4 is coupled to the input terminal 2 for successively storing a predetermined number of samples of the audio input signal 20.
  • the buffer circuit 4 has a length which is capable to store samples of the audio input signal 20 during a period of time which is in the order of a room mode onset or decay time.
  • a low pass filter 5 is coupled between the buffer circuit 4 and an equalizing circuit 7, wherein the output of the low pass filter 5 is providing an output signal to the equalizing circuit 7 containing only lower frequencies of the audio input signal, for example below a cut frequency Fc of 200 Hz.
  • a high pass filter 6 is coupled in parallel to the series connection of the low pass filter 5 and the equalizing circuit 7.
  • the audio processing device further comprises an output circuit coupled to the output of the equalizing circuit 7 which is configured to supply an audio output signal 30 to a speaker (not shown) for audio output from the speaker.
  • the output circuit comprises a summation circuit 8 for adding a signal provided from the high pass filter 6 and a signal provided from the equalizing circuit 7 for supplying the audio output signal 30 to the speaker.
  • a static equalizing circuit 9 may be connected downstream to the summation circuit 8 for static EQ compensation of the audio signal. The static equalizing circuit 9 does not compensate the audio signal at the frequencies of room modes, which EQ compensation is handled by the equalizing circuit 7.
  • the input of the equalizing circuit 7 is coupled to the input terminal 2 via the buffer circuit 4 and the low pass filter 5.
  • the equalizing circuit 7 is configured to execute, for specific frequencies of room modes, frequency characteristic correction processing (referred to in this example as EQ compensation) for a signal received at the input.
  • a signal analyzing circuit which includes a dynamic analysis portion 11 and a control portion 12, is coupled to the input terminal 2 through the buffer circuit 4 and the low pass filter 5 to receive the audio input signal 20 via the buffer circuit 4 and the low pass filter 5.
  • the signal analyzing circuit 11, 12 is configured to analyze a frequency characteristic, at least at signal components of lower frequencies, of the audio input signal 20 for detecting the presence of a signal power of the audio input signal 20 at or near one or more frequencies of a room mode.
  • detecting the presence of a signal power of the audio input signal at or near one or more frequencies of a room mode means detecting of a particular signal power within a narrow frequency band around one or more room mode frequencies, such as F1, F2, as explained below.
  • the signal analyzing circuit, particularly its control portion 12 is coupled to a control terminal of the equalizing circuit 7 for dynamically adapting the EQ compensation for specific frequencies of room modes over time, as described in more detail in the following.
  • a calibration signal may be transmitted into the particular room environment, such as the car cabin.
  • the calibration signal contains a predetermined frequency spectrum and a signal response or frequency response of the acoustic transfer function between speaker and listening position is measured for determining frequencies of room modes.
  • the car cabin or any other room
  • This kind of information will be used as a priori knowledge for the functioning of the EQ compensation according to the principles of the invention.
  • a table of frequencies and associated decay times for each room mode will be determined. This is done by a separate measurement as outlined above. Similar measurements (so-called “Waterfall” measurements as discussed above and shown with reference to Fig. 2 ) should be done for determining the respective onset time of the room modes. Again, a table with frequencies and associated onset times will be determined. An example of such a table is shown in Fig. 4A depicting a table 21 in which the frequencies F1 to F3 and the corresponding onset times of exemplary principal room modes are stored. Another example of a table is shown in Fig. 4B depicting a table 22 in which the respective decay times of the exemplary principal room modes at the frequencies F1 to F3 are stored.
  • the onset and decay time can vary for each room mode frequency. Decay time and onset time can also be the same, i.e. the room "behaves" in a symmetric way.
  • two tables may be used, one for decay time and one for onset time.
  • Fig. 3 the function of the audio processing device as shown in Fig. 3 will be explained with particular reference to Fig. 5 , showing an exemplary time diagram for two particular audio input signal frequencies and the associated activating and deactivating of the frequency characteristic correction processing over time for these audio input signals.
  • the buffer circuit 4 functioning as input buffer, implements a signal analysis buffer of a certain length.
  • the length of this buffer should be in the order of the decay or onset time of the room mode having the lowest frequency.
  • the low pass filter 5 is designed as quadrature mirror filter (QMF) with a high pass filter (HPF).
  • the cut frequency may be at 200 Hz, as room modes as a result of reflection at room walls or the like are typically present at lower frequencies.
  • the high pass filter 6 is designed as QMF with a low pass filter (LPF).
  • the cut frequency may also be at 200 Hz.
  • the summation circuit 8 adds the signal coming from the output of the HPF 6 and the signal coming from the output of the equalizing circuit 7.
  • the static equalizing circuit 9 implements the car audio equalization derived from static measurements (tuning) without considering room mode compensation. In the tuning process every frequency where a room mode is present is analyzed. Any negative cut necessary to achieve a desired target curve will be shifted to the equalizing circuit 7.
  • the equalizing circuit 7 (“EQ Cuts per Frequency Room Mode") comprises a series of cut filters at the room modes present in the car cabin (F1, F2, F3, etc., see Fig. 4 ) implemented as second order section IIR filters.
  • the EQ control portion 12 of the signal analyzing circuit controls the equalizing circuit 7. The controlling will be activated by the output of the dynamic analysis portion 11 of the signal analyzing circuit. If the dynamic analysis portion is requesting action, the equalizing circuit 7 will filter the low frequency signal from the LPF 5 applying frequency cuts, i.e. compensating room modes.
  • peaks of the signal components having a specific frequency of or near a room mode are cut in order to equalize, in connection with the static equalizing circuit 9, the frequency characteristic of the audio signal to a substantially flat curve over the frequency band (flat signal response or frequency response of the acoustic transfer function between speaker and listening position). If action is not required the equalizing circuit 7 will be gradually switched into a bypass filter (room modes are not compensated).
  • the dynamic analysis portion 11 controls the time for activation and deactivation of the equalizing circuit 7. It performs frequency analysis over time to determine activation and deactivation of the equalizing circuit 7 via the EQ control portion 12.
  • the analysis of the frequency characteristic of the audio input signal 20 is functioning as follows:
  • the initial state of the equalizing circuit 7 will be an "all pass" on all frequencies (0 db for all frequencies).
  • This filter will contain gain variable cutting filters (notch profile) at each room mode frequency, e.g. as listed in previously mentioned tables 21 and 22 of Fig. 4 .
  • power or energy band detection will be performed by means of a narrow band filter at the frequencies where room modes appear (around F1, F2, F3, etc.). This means that a gain threshold will be compared to the current spectral energy or power on that band.
  • signal power or energy is estimated from several blocks in time from buffer 4 (previously FFT transformed) by means of averaging within a narrow frequency band around F1, F2.
  • the detection is positive (a room mode frequency was detected in the audio signal).
  • the energy band detection is repeated periodically (in small blocks of samples) wherein the duration of the detection of the presence of the signal power to be determined is the time in samples or milliseconds that the input signal provides a positive power detection for or near a room mode frequency.
  • a different timer (e.g. counter) will be assigned for each individual room mode frequency, for example a timer1 for F1, timer2 for F2, timer3 for F3 and so on.
  • Each timer will be started at a respective point of time (e.g. Ton (F1) for compensating a signal at frequency F1, Ton (F2) for compensating a signal at frequency F2) depending on the positive detection of its corresponding room mode frequency.
  • the respective timer will be deactivated at a respective point of time (e.g. Toff (F1) for deactivating compensating at frequency F1, Toff (F2) for deactivating compensating at frequency F2) depending on the non-detection (low energy or not present) of its corresponding room mode frequency.
  • Each timer is activated after energy band detection is positive, i.e. the energy or power of the audio input signal for or at the respective room mode frequency (i.e., within the narrow frequency band around the respective room mode frequency) is greater than the threshold (i.e., a corresponding room mode frequency was detected in the audio signal).
  • the respective timer continues counting only if the energy is over the gain threshold over time. If the positive detection lasts longer than the time threshold for F1, F2, etc., the EQ control portion 12 will be activated. At this time the corresponding timer will be reset to zero. This indicates that currently the signal will excite a room mode, therefore steady state cuts or EQ compensation for F1, F2 (designated as "Cut EQ" in Fig. 5 ) should be applied.
  • the EQ control portion may implement a "fade in” and “fade out” for the activation and deactivation of the cut filters of the equalizing circuit 7 correspondingly. Time and profile for this fade in and fade out are parameters of the EQ control portion 12. Fading will be implemented for changing the gain cut of the notch filters of the equalizing circuit 7. This gain control of the equalizing circuit's 7 parametric EQs will be performed in order to ensure a soft transition between the states "cutting EQ mode" and "bypass setting" during transition from deactivation to activation and vice versa.
  • the EQ control portion 12 will keep the equalizing circuit 7 to be a bypass filter.
  • F1 40Hz
  • the parameters of the EQ control portion 12 it could be that during a fade in (activation) or fade out (deactivation) phase an opposite transition happens. For example, time for activation was achieved, fade in is in process and a transition to deactivation is detected. These special cases are shown in Fig. 5 .
  • FIGs 6 and 7 examples of audio signals are shown which represent a drum music signal.
  • a drum sound as shown is characteristical for having short lasting low frequency signals having relative high energy which may excite room modes if they last longer than the corresponding onset time.
  • such drum audio signals are not subjected to EQ compensation, since the typical drum "sound elements" as shown in Fig. 6 are shorter than typical onset and decay times of the corresponding room modes.
  • the duration time of the drum "sound element" (short hit on the drum) is in the order of 0.04 ms.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Stereophonic System (AREA)

Claims (12)

  1. Dispositif de traitement des audiofréquences comprenant :
    - un terminal d'entrée (2) configuré pour recevoir un signal d'entrée audio à partir d'une source audio ;
    - un circuit d'égalisation (7) possédant une entrée, une sortie et un terminal maître, l'entrée étant couplée au terminal maître (2) et le circuit d'égalisation (7) étant configuré pour exécuter un traitement de correction de caractéristique de fréquence pour un signal reçu à l'entrée ;
    - un circuit de sortie (8, 9, 3) couplé à la sortie du circuit d'égalisation (7) et configuré pour acheminer un signal de sortie audio à un haut-parleur pour une sortie audio à partir du haut-parleur ;
    - un circuit d'analyse de signaux (11, 12) couplé pour recevoir le signal d'entrée audio et configuré pour analyser une caractéristique de fréquence du signal d'entrée audio afin de détecter la présence d'une puissance du signal d'entrée audio à au moins une fréquence (F1, F2) d'un mode ambiant ;
    - dans lequel le circuit d'analyse de signaux (11, 12) mémorise une table (21, 22) contenant plusieurs fréquences (F1, F2, F3) de modes ambiants respectifs, la table associant un temps de réaction respectif prédéterminé et un temps de déclin respectif prédéterminé des modes ambiants à une fréquence correspondante parmi lesdites plusieurs fréquences ;
    - dans lequel le circuit d'analyse de signaux (11, 12) est couplé au terminal maître du circuit d'égalisation (7) pour adapter de manière dynamique le traitement de correction de caractéristique de fréquence du circuit d'égalisation (7) à une ou plusieurs fréquences (F1, F2) desdites plusieurs fréquences (F1, F2, F3) de modes ambiants respectifs au cours du temps lorsque la présence de la puissance du signal d'entrée audio auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant est détectée, le traitement de correction de caractéristique de fréquence étant activé pour lesdites une ou plusieurs fréquences d'un mode ambiant lorsque la durée de la détection de la présence de la puissance de signal est supérieure à un seuil de temps pour la fréquence respective d'un mode ambiant.
  2. Dispositif de traitement des audiofréquences selon la revendication 1, dans lequel :
    - lorsque le circuit d'analyse de signaux (11, 12) détecte la présence de la puissance du signal d'entrée audio auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant, le circuit d'analyse des signaux détermine une durée de la détection de la présence de la puissance de signal et compare la durée au temps de réaction enregistré de la fréquence respective ;
    - dans lequel, lorsque la durée est égale ou supérieure à un seuil de temps prédéterminé, le circuit d'analyse de signaux (11, 12) donne ordre au circuit d'égalisation (7) d'activer le traitement de correction de caractéristique de fréquence auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant.
  3. Dispositif de traitement des audiofréquences selon la revendication 1 ou 2, dans lequel :
    - le circuit d'analyse de signaux (11, 12) détermine en outre une puissance du signal d'entrée audio auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant et compare la puissance de signal déterminée à une valeur seuil de puissance de signal ;
    - lorsque la puissance de signal déterminée est égale ou supérieure à un seuil de temps prédéterminé, le circuit d'analyse de signaux (11, 12) donne ordre au circuit d'égalisation (7) d'activer le traitement de correction de caractéristique de fréquence auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant.
  4. Dispositif de traitement des audiofréquences selon la revendication 2 ou 3, dans lequel :
    - le circuit d'analyse de signaux (11, 12) détermine en outre une puissance du signal d'entrée audio auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant et compare la puissance de signal déterminée à une valeur seuil de puissance de signal ;
    - lorsque le circuit d'analyse de signaux (11, 12) a précédemment donné ordre au circuit d'égalisation (7) d'activer le traitement de correction de caractéristique de fréquence auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant, le circuit d'analyse de signaux détermine le fait de savoir si la puissance de signal déterminée est inférieure à la valeur seuil de puissance de signal ;
    - et lorsque la puissance de signal déterminée est inférieure à la valeur seuil de puissance de signal, le circuit d'analyse de signaux (11, 12) donne ordre au circuit d'égalisation (7) de désactiver le traitement de correction de caractéristique de fréquence auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant après que le temps de déclin respectif se soit écoulé à compter à partir du moment où le circuit d'analyse de signaux a déterminé que la puissance de signal déterminée est inférieure à la valeur seuil de puissance de signal.
  5. Dispositif de traitement des audiofréquences selon l'une quelconque des revendications 1 à 4, comprenant en outre un circuit tampon (4) couplé entre le terminal d'entrée (2) et le circuit d'égalisation (7) pour mémoriser de manière successive plusieurs échantillons du signal d'entrée audio, le circuit d'analyse de signaux (11, 12) étant couplé au circuit tampon (4) pour analyser une caractéristique de fréquence du signal d'entrée audio.
  6. Dispositif de traitement des audiofréquences selon la revendication 5, dans lequel :
    le circuit tampon (4) possède une longueur qui lui permet de stocker des échantillons du signal d'entrée audio au cours d'un laps de temps qui est de l'ordre d'un temps de déclin ou de réaction en mode ambiant.
  7. Dispositif de traitement des audiofréquences selon l'une quelconque des revendications 1 à 6, comprenant en outre un filtre passe-bas (5) couplé entre le terminal d'entrée (2) et le circuit d'égalisation (7) et possédant une sortie procurant un signal de sortie au circuit d'égalisation, contenant uniquement les fréquences inférieures du signal d'entrée audio.
  8. Dispositif de traitement des audiofréquences selon la revendication 7, comprenant en outre :
    - un filtre passe-haut (6) couplé entre le terminal d'entrée (2) et le circuit de sortie (8, 9, 3) et monté en parallèle à la liaison en série du filtre passe-bas (5) et du circuit d'égalisation (7) ;
    - dans lequel le circuit de sortie (8, 9, 3) comprend un circuit sommateur (8) pour additionner un signal provenant du filtre passe-haut (6) et un signal provenant du circuit d'égalisation (7) pour procurer le signal de sortie audio au haut-parleur.
  9. Dispositif de traitement des audiofréquences selon l'une quelconque des revendications 1 à 8, dans lequel le dispositif de traitement des audiofréquences (1) est implémenté dans un environnement intérieur de véhicule et lesdites plusieurs fréquences (F1, F2, F3) de modes ambiants contenus dans la table (21, 22) ont été déterminées via une mesure dans l'environnement du véhicule.
  10. Procédé pour mettre en oeuvre un traitement de correction de caractéristique de fréquence pour un signal d'entrée audio, comprenant :
    - à un stade initial pour la mise en oeuvre du procédé, la transmission d'au moins un signal d'étalonnage dans un environnement ambiant contenant un spectre de fréquences prédéterminées et la mesure d'une réponse de signal pour déterminer des fréquences (F1, F2) de modes ambiants ;
    - la mémorisation d'une table (21, 22) contenant les fréquences déterminées (F1, F2) de modes ambiants, la table associant un temps de réaction respectif prédéterminé et un temps de déclin respectif prédéterminé des modes ambiants à une fréquence correspondante desdites plusieurs fréquences ;
    - l'analyse d'une caractéristique de fréquence d'un signal d'entrée audio (20) pour détecter la présence d'une puissance du signal d'entrée audio à une ou plusieurs fréquences (F1, F2) des fréquences déterminées (F1, F2) de modes ambiants ;
    - la mise en oeuvre, dans un processus d'égalisation, d'un traitement de correction de caractéristique de fréquence pour le signal d'entrée audio (20) ;
    - l'activation et la désactivation dynamique du traitement de correction de caractéristique de fréquence à ladite au moins une fréquence d'un mode ambiant au cours du temps lorsque la présence de la puissance du signal d'entrée audio à ladite au moins une fréquence (F1, F2) d'un mode ambiant a été détectée ;
    - dans lequel les fréquences déterminées (F1, F2) de modes ambiants sont utilisées dans un stade opérationnel pour l'activation et la désactivation du traitement de correction de caractéristique de fréquence ;
    - dans lequel le traitement de correction de caractéristique de fréquence est activé pour lesdites une ou plusieurs fréquences d'un mode ambiant lorsque la durée de la détection de la présence de la puissance de signal est supérieure à un seuil de temps pour la fréquence respective d'un mode ambiant ;
    - l'alimentation à partir du traitement de correction de caractéristique de fréquence d'un signal de sortie audio à un haut-parleur pour une sortie audio à partir du haut-parleur.
  11. Procédé selon la revendication 10, englobant en outre les étapes dans lesquelles :
    - lorsque la présence de la puissance du signal d'entrée audio auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant a été détectée, on détermine la durée de la détection de la présence de la puissance de signal et on compare la durée à un temps d'activation du mode ambiant correspondant ;
    - lorsque la durée est égale ou supérieure à un seuil de temps prédéterminé, on active le traitement de correction de caractéristique de fréquence à la fréquence respective (F1, F2) du mode ambiant.
  12. Procédé selon la revendication 11, englobant en outre les étapes dans lesquelles :
    - on détermine une puissance du signal d'entrée audio auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant et on compare la puissance de signal déterminée à une valeur seuil de puissance de signal ;
    - lorsque le traitement de correction de caractéristique de fréquence auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant a été activé précédemment, on détermine le fait de savoir si la puissance de signal déterminée est inférieure à la valeur seuil de puissance de signal ;
    - et lorsque la puissance de signal déterminée est inférieure à la valeur seuil de puissance de signal, on désactive le traitement de correction de caractéristique de fréquence auxdites une ou plusieurs fréquences (F1, F2) d'un mode ambiant après un temps de déclin du mode ambiant correspondant, à compter à partir du moment où il a été déterminé que la puissance de signal déterminé est inférieure à la valeur seuil de puissance de signal.
EP08010723.8A 2008-06-12 2008-06-12 Dispositif de traitement audio et procédé pour effectuer un traitement de correction de caractéristiques de fréquence pour un signal d'entrée audio Expired - Fee Related EP2134105B1 (fr)

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JP5199915B2 (ja) * 2009-02-18 2013-05-15 キヤノン株式会社 音場補正方法及び音場補正装置
US9779755B1 (en) * 2016-08-25 2017-10-03 Google Inc. Techniques for decreasing echo and transmission periods for audio communication sessions
CH719150A1 (de) * 2021-11-17 2023-05-31 Rocket Science Ag Verfahren zum Eliminieren von Raummoden und digitaler Signalprozessor sowie Lautsprecher dafür.

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EP0236076A2 (fr) * 1986-03-03 1987-09-09 Ray Milton Dolby Circuits électroniques pour changer l'étendue dynamique de signaux audio

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JP2000261900A (ja) 1999-03-09 2000-09-22 Sony Corp 音場補正方法および音響装置。
US20040002781A1 (en) 2002-06-28 2004-01-01 Johnson Keith O. Methods and apparatuses for adjusting sonic balace in audio reproduction systems
WO2007068257A1 (fr) * 2005-12-16 2007-06-21 Tc Electronic A/S Procede pour realiser des mesures au moyen d'un systeme audio comprenant des haut-parleurs passifs

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0236076A2 (fr) * 1986-03-03 1987-09-09 Ray Milton Dolby Circuits électroniques pour changer l'étendue dynamique de signaux audio

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