EP3148215A1 - Procédé et système de modification de la fréquence d'un signal audio - Google Patents

Procédé et système de modification de la fréquence d'un signal audio Download PDF

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Publication number
EP3148215A1
EP3148215A1 EP15460077.9A EP15460077A EP3148215A1 EP 3148215 A1 EP3148215 A1 EP 3148215A1 EP 15460077 A EP15460077 A EP 15460077A EP 3148215 A1 EP3148215 A1 EP 3148215A1
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EP
European Patent Office
Prior art keywords
signal
modification
frequency
unit
sound
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Withdrawn
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EP15460077.9A
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German (de)
English (en)
Inventor
Bozena Kostek
Piotr Hoffmann
Andrzej Czyzewski
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Politechnika Gdanska
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Politechnika Gdanska
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Priority to EP15460077.9A priority Critical patent/EP3148215A1/fr
Publication of EP3148215A1 publication Critical patent/EP3148215A1/fr
Withdrawn legal-status Critical Current

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    • 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
    • H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
    • 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

Definitions

  • the present invention relates to a method for modifying the frequency of audio signal and system for modifying the frequency of audio signal, intended in particular for use in listening rooms and recording studios.
  • US Patent 8531602 discloses arrangement whereby the system detects playing a song, and then replaces it with the same as to the content but higher quality. According to this solution, the system searches the reference base and when it finds a suitable sound, it mixes the original channel with a higher resolution file. At the mixing step a suitable time delay is selected and the phase of smart signals matching occurs. The entire procedure is performed automatically and the user does not need to interfere in the process of improving sound quality.
  • the disadvantage of this solution is that the solution does not apply to mobile devices, where low-quality transducers and loudspeakers are used.
  • the embodiment relates to an intelligent sound amplification system.
  • the system listens to the playback contents using an external device, and then adjusts the parameters to improve the sound to the collected content.
  • the device receives audio signals and produces improved audio signals.
  • Amplifying the sound signal is based on sound samples from the speaker of the mobile device.
  • the main limitation of this system is the ability to improve the sound of mobile devices only when playing sound from the built-in speakers and in good acoustic conditions. When playing through headphone, the said embodiment is idle. If sounds are played in poor acoustic conditions, there is distortion in the intercepted signal, resulting in inaccurate signal modification and consequently introducing additional distortion to the signal.
  • US Pat 20130178963 discloses a method for recognizing musical instruments in audio recording and improving their sound.
  • the main application of the embodiment is to use it in large collections of digital works.
  • the Polish patent application P.408563 presents and embodiment to modify the audio signal frequency, which is characterized in that a sound sample is acquired from the electronic device, from which the three copies are created, the first copy of the audio signal sample is analyzed for genre, the second copy of the audio signal sample is subjected to modifications towards the best sound, while the third copy of the audio signal sample is treated as a reference signal.
  • a method of modifying the sound signal frequency, based on acquisition of sample sound signal from the device, from which two, preferably three copies are created is characterized according to the invention in that the control system implements a first algorithm that analyzes the frequency characteristic of the room, which is obtained in a phonic analysis and analyzing genre information acquired from the genre classification unit.
  • the first copy of sound signal sample after identifying the genre in a genre classification unit, is sent to the control unit, in which, depending on the frequency characteristics of the room, the parameters are adjusted to modify frequency of audio signal and a first control signal is generated containing information about frequency response of the room, and a second control signal containing information about the type of music.
  • the second copy of audio signal sample is modified in modification unit in which with using the first frequency modifier, using the first control signal, and using the implemented second algorithm for realizing method to control frequency multiband filter customized in accordance with a control signal 1, reference is made to the reference signal, and the acquired first modified audio signal is directed to the second frequency modifier.
  • the signal is tuned to the genre using a third algorithm implementing the method of controlling the multiband filter customized in accordance with a control signal 2, and reference is made to the reference signal.
  • the second modified sound signal is fed to the percentile analyzer, which measures the statistical characteristic.
  • the resulting signal of the statistical information together with a second modified sound signal is fed to a signal amplifier, wherein on the basis of statistical information SST the volume is leveled, and the resulting output signal is fed to exporting block.
  • a first frequency modifier using the second algorithm in a first frequency modifier using the second algorithm, it obtains a first reference signal containing information about the level of modification of the first modified audio signal and the second frequency modifier using the third algorithm, it obtains a second reference signal containing information about the level of modification of the second modified signal sound. From the amplifier a third reference signal is acquired containing information about the sound volume. Then, the first reference signal, the second reference signal and a third reference signal are fed to modification control unit. In the modification control unit, wherein using a fourth implemented algorithm for performing the control of level of signal modification, it analyzes the signal quality of the result. In case of insufficient quality, as a differential signal is fed to the control unit.
  • a second sample of the audio signal is directed first to the modification decision unit, wherein in the case of large content of audio content, the signal is fed to the tracks separator.
  • the audio signal is divided into n-tracks.
  • Each of the n-tracks is subjected to a separate modification in a modification unit, and a separate modification control unit.
  • N-modified tracks are merged in the integrator and the resulting signal is fed to export block.
  • a system for modifying audio signal frequency comprising a sound buffer, which is connected via genre classification unit with a control unit which is connected to a modification unit is characterized in that the audio buffer is connected to modification unit, and control unit is coupled with room phonic analysis unit.
  • the control unit is connected to modification unit via first frequency modifier compensating frequency characteristics acoustics of the room and via the second frequency modifier compensating the genre.
  • the first frequency modifier is connected to the second frequency modifier which is connected via the percentile analyzer to the signal amplifier.
  • Signal amplifier is connected to the export block.
  • the sound buffer is also connected directly to the first frequency modifier, the second frequency modifier and with a signal amplifier which is connected to the export block.
  • modification unit is connected to a modification control unit via first frequency modifier, second frequency modifier and signal amplifier, while the modification control unit is connected to control unit.
  • decision unit is located on the input for the modification unit, which is connected to track separator, and the track integrator is located on the output of the modification unit.
  • the invention is based on an intelligent use of the control sound utilizing the musical genres and improve the sound quality, depending on the device used.
  • Modification of the audio signal is carried out according to the genre and taking into account the acoustics of the room in which the sound is played.
  • the modification signal may take place in the entire audio signal or the individual instrumental tracks.
  • the system recognizes music genre from the processed audio signal, also analyzes the acoustic environment, adapting the process to the current acoustic environment in the room.
  • the present invention by utilizing the phenomena of perceptual frequency amplification, restores bands, which physically could not be restored.
  • the invention extends the possibility to improve the sound quality of intelligent diversification of gain parameters depending on the genre, making it possible to achieve less distortion in the audio processing.
  • sound source UE is connected to the audio buffer 1.
  • a sound signal frequency modification unit is composed of the sound buffer BD, which is connected via genre classification unit UG with the control unit US, which is connected to modification unit UM.
  • Genre classification unit UG consists of a filter block 1 that filters the signal in a manner appropriate to the applied parameters, parameterization block 2, the signal data redundancy reducer 3, genres classifier 4. The result of the recognition genre is transmitted to the control unit US.
  • Room phonic analysis unit UA is connected to control unit US is included in audio analysis system for UA rooms Room phonic analysis unit UA is comprised of the impulse response recorder 5 and Fourier transform analyzer 6.
  • Control unit US is connected with modification unit UM via the first frequency modifier 7 compensating acoustic characteristics of the room and through the second frequency modifier 8 compensating genre.
  • the first frequency modifier 7 is connected to a second frequency modifier 8 which is connected to the percentile analyzer 9 with a signal amplifier 10 which is connected to the export block BE.
  • the first frequency modifier 7, second frequency modifier 8 and amplifier 10 are also connected directly to the sound buffer BD.
  • Control unit US implements the first algorithm ALI analyzing frequency characteristics of the room and analyzing information acquired from the genre classification.
  • the first modifier implements a second algorithm AL2 implementing the method of controlling a multi-band frequency filter, customized in accordance with control signal 1.
  • the second modifier implements the third algorithm AL3 realizing the filter control frequency multiband customized in accordance with a control signal 2.
  • a sample audio signal SP is acquired from the electronic device EU, which is used to create two copies.
  • the first copy of the audio signal sample S1 is analyzed in terms of genre, a second copy of the audio signal sample S2 is subjected to modifications towards optimal sound.
  • Characteristics signal SCH is acquired from impulse response S01 which is acquired from impulse response recorder 5 and processed by Fourier transform analyzer 6.
  • the first copy of the audio signal sample SI is sent to the control unit US, where depending on the frequency characteristics of the room, acquired from in the form of signal characteristics SCH and an information signal S1I, frequency modification parameters of audio signal S2 are adjusted.
  • Control unit US generates a first control signal SS1 containing information about frequency response of the room, and the second control signal SS2, containing information about the genre of music.
  • the second copy of audio signal sample S2 is modified in a modification unit UM, in which, with the first frequency modifier 7, using the first control signal SS1 and through implementing a second algorithm AL2, reference is made to the reference signal S3.
  • the first resulting modified audio signal S2Z1 is directed to second frequency modifier 8.
  • the second frequency modifier 8, using the second control signal SS2, tunes the signal to the genre using a third algorithm AL3 and reference is made to the reference signal S3.
  • the second modified audio signal, S2Z2 is directed to the percentile analyzer 9, which measures statistical characteristic.
  • the resulting signal SST of statistical information together with a second modified sound signal S2Z2 are fed to signal amplifier 10, which in comparison to the reference signal S3 the intensity of the second modified sound signal S2Z2 is measured. Based on this measurement volume is leveled, and the resulting output signal SK is fed to the exporting block BE.
  • Embodiment 2 Embodiment 2
  • modification unit UM is connected to a modification control unit UK through the first frequency modifier 7, the second frequency modifier 8 and a signal amplifier 10, while modification control unit UK is connected to control unit US
  • Modification control unit UK implements the fourth algorithm AL4 that controls the level of the signal modification by analyzing the output signal quality.
  • the amplifier 10 provides the third reference signal S03 that contains information about sound level.
  • First reference signal S01, second reference signal S02 and third reference signal S03 are directed to modification control unit UK, wherein implemented uses fourth algorithm AL4 to analyze the quality of the output signal SK.
  • fourth algorithm AL4 to analyze the quality of the output signal SK.
  • a differential signal SR is fed to the control system the US.
  • this signal as the result signal SK is directed to exporting block BE.
  • the system is constructed as described in Embodiment 2, except that the input to the modification unit UM is located in the decision unit UD which is connected to the paths separator 11, and on the output of the modification unit UM the track integrator 12 is located.
  • a second sample of the audio signal S2 is directed in the first instance to the decision unit UD, in the case of wide audio content consisting of three instrumental tracks, this signal is fed to the paths separator 11, where it is divided into three tracks S2P.
  • Each of the separate tracks are subjected to separate modifications on the modification unit UM and a separate control modifications on modification control unit UK.
  • Modified tracks S2Z are merged in integrator 12 and fed to export block BE.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP15460077.9A 2015-09-23 2015-09-23 Procédé et système de modification de la fréquence d'un signal audio Withdrawn EP3148215A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP15460077.9A EP3148215A1 (fr) 2015-09-23 2015-09-23 Procédé et système de modification de la fréquence d'un signal audio

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP15460077.9A EP3148215A1 (fr) 2015-09-23 2015-09-23 Procédé et système de modification de la fréquence d'un signal audio

Publications (1)

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EP3148215A1 true EP3148215A1 (fr) 2017-03-29

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EP15460077.9A Withdrawn EP3148215A1 (fr) 2015-09-23 2015-09-23 Procédé et système de modification de la fréquence d'un signal audio

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Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20090182563A1 (en) * 2004-09-23 2009-07-16 Koninklijke Philips Electronics, N.V. System and a method of processing audio data, a program element and a computer-readable medium
EP2384025A2 (fr) 2009-11-16 2011-11-02 Harman International Industries, Incorporated Système audio avec dispositif portable d'amélioration audio
US20130163784A1 (en) 2011-12-27 2013-06-27 Dts Llc Bass enhancement system
US20130178963A1 (en) 2012-01-08 2013-07-11 Oliver Bergmann Audio system with adaptable equalization
US8531602B1 (en) 2011-10-19 2013-09-10 Google Inc. Audio enhancements for media

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20090182563A1 (en) * 2004-09-23 2009-07-16 Koninklijke Philips Electronics, N.V. System and a method of processing audio data, a program element and a computer-readable medium
EP2384025A2 (fr) 2009-11-16 2011-11-02 Harman International Industries, Incorporated Système audio avec dispositif portable d'amélioration audio
US8531602B1 (en) 2011-10-19 2013-09-10 Google Inc. Audio enhancements for media
US20130163784A1 (en) 2011-12-27 2013-06-27 Dts Llc Bass enhancement system
US20130178963A1 (en) 2012-01-08 2013-07-11 Oliver Bergmann Audio system with adaptable equalization

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