WO2018206093A1

WO2018206093A1 - System and method for tuning audio response of an image display device

Info

Publication number: WO2018206093A1
Application number: PCT/EP2017/061074
Authority: WO
Inventors: Alper OZEL; Mehmet Emin CEPNI; Mustafa Ilker Uzun; Cengiz Berkay; Osman Osman
Original assignee: Arcelik AS
Current assignee: Arcelik AS
Priority date: 2017-05-09
Filing date: 2017-05-09
Publication date: 2018-11-15
Anticipated expiration: 2019-11-09
Also published as: TR201801622A2

Abstract

The present invention relates to a method for operating an image display device (1), said device comprising a plurality of speakers (2) and microphones (3) placed on a rear surface portion opposite a front display portion of said image display device (1), said method is characterized by separately producing pink noise by each speaker (2) in order and detection of the produced noise by each microphone (3), converting detected noise to an electrical signal, performing FFT analysis for each microphone (3), checking whether FFT analysis results fall within predetermined values and calculating equalizer settings to obtain said predetermined values.

Description

SYSTEM AND METHOD FOR TUNING AUDIO RESPONSE OF AN IMAGE DISPLAY DEVICE

The present invention relates to a system and method for tuning and optimizing audio response of an electronic device, such as a TV, according to the material behind it.

It is well-known that in addition to video, audio is an important feature of TVs. For this reason, the audio performances of TVs are highly important. TVs can have different audio performances in different environments. For example, the size of the room the TV is in, the material behind the TV (wood, glass, concrete, etc.), the furniture in said room and existing noise in said room all have an effect on the audio performance of the TV. As sound waves propagate, they become distorted due to interactions with the objects in the room depending on respective dampening characteristics of objects.

Another factor is the group delay caused by the mechanical features of the speakers. Group delay is the time delay between the electrical signal reaching the speaker and the speaker producing a sound wave and it can vary for each frequency band. This has a negative impact on sound quality.

Therefore, if the equalizer is adjusted according to the first material off which the sound wave is reflected, the first step where sound is most distorted can be compensated and the ambient sound quality can be improved. In addition, if the electrical input signal and the output signal from the speaker are evaluated according to frequency and time, time delay can be improved.

Due to trends in technology, the thickness of displays is decreased. For this reason, it has become difficult to integrate the speakers to the body of the monitors and speakers are increasingly fitted on the back of the displays. Therefore, it is certain that sound waves leaving the speaker facing the opposite direction to the display will be reflected off the material behind the screen before reaching the ear of the user.

Among others, a prior art publication in the technical field of the invention may be referred to as US8082051, which discloses an audio system installed in a listening space comprising a signal processor and a plurality of speakers. The audio system may be tuned with an automated audio tuning system to optimize the sound output of the speakers within the listening space. The automated audio tuning system may provide automated processing to determine at least one of a plurality of settings, such as channel equalization settings, delay settings, etc. The settings may be generated by the automated audio tuning system based on an audio response produced by the loudspeakers in the audio system. Another prior art publication may be referred to as US2002196951.

The present invention, on the other hand, addresses the situation where the sound waves reflecting off the material behind the TV are automatically improved with the use of speakers and microphones present on the body thereof. In this manner, the present invention aims to provide an improved ambient sound performance and a more comfortable viewing experience for the user.

To this end, the present invention provides a measurement and modification system to present an improved sound performance to the users. The method comprises acoustic feedback in order to optimize the performance of the speakers for different environments and obtain higher sound quality.

The present invention provides a system and method for tuning and optimizing audio of an electronic device according to the material behind it as provided by the characterizing features defined in Claim 1.

Primary object of the present invention is to provide a system and method for tuning and optimizing audio of an electronic device, such as a TV, according to the material behind it.

Accompanying drawings are given solely for the purpose of exemplifying an audio tuning system and method, whose advantages over prior art were outlined above and will be explained in brief hereinafter.

The drawings are not meant to delimit the scope of protection as identified in the Claims, nor should they be referred to alone in an effort to interpret the scope identified in said Claims without recourse to the technical disclosure in the description of the present invention.

The drawings are only exemplary in the sense that they do not necessarily reflect the actual dimensions and relative proportions of the respective components of the system if not otherwise explicitly stated.

Fig. 1 demonstrates a simplified general schematic diagram representation of an audio tuning system according to the present invention.

Fig. 2 demonstrates a back view of a TV monitor of an audio tuning system according to the present invention.

Fig. 3 demonstrates an ideal peak frequency vs time graph belonging to an audio tuning system according to the present invention.

Fig. 4 demonstrates a peak frequency vs time graph belonging to an audio tuning system according to the present invention.

Fig. 5 demonstrates a Bode plot and an inverse Bode plot of a microphone belonging to an audio tuning system according to the present invention.

Fig. 6 demonstrates the convolution of electrical response and normalized inverse Bode plot of a microphone belonging to an audio tuning system according to the present invention.

Fig. 7 demonstrates the frequency scan data of a speaker belonging to an audio tuning system according to the present invention.

Fig. 8 demonstrates the frequency scan data divided in the time domain and the performed FFT analysis of a speaker belonging to an audio tuning system according to the present invention.

Fig. 9 demonstrates a sample FFT analysis of a speaker belonging to an audio tuning system according to the present invention.

Fig. 10 demonstrates a schematic diagram representation of an audio tuning system according to the present invention.

Fig. 11 demonstrates automatic equalizer setting calculation flow diagram according to the present invention.

Fig. 12 demonstrates group delay calculation flow diagram according to the present invention.

The following numerals are assigned to different part number used in the detailed description:

Image display device
Speaker
Microphone
Reflection surface

In accordance with the present invention, in addition to the speakers (2) already available on the TV displays/monitors (image display devices (1)), acute angle, linear and high quality microphones (3) are placed at different points on the body and added to the motherboard design or alternatively a separate signal processing unit is added to the motherboard. Each speaker (2) is adapted to produce pink noise by itself in order and the noise is detected by each microphone (3) as will be delineated hereinafter.

Microphones (3) are placed on the TV body in a manner that they can’t directly detect the sound produced by the speakers. However, at least one microphone (3) is placed substantially close to each speaker (2) in order to avoid highly reflected sound waves (by the reflection surface (4)). The angle of the microphones (3) is kept acute to prevent them from directly detecting the sound produced by the speakers (2). Also, as it is expected that the sound waves expected to reach the user most intensely will pass near the edges of the TV body, microphones (3) are placed on the edges of the TV body.

It is to be noted that sound quality can be optimized using the speakers (2) as microphones (3) without need for additional microphones (3). In order to avoid confusion, the speakers (2) being usable as receivers will also be referred to as microphones (3) in the present description.

After pink noise is produced by each speaker (2), noise detected by each microphone (3) will be converted to an electrical signal and transmitted to a DSP to be amplified and processed in order to determine the automatic equalizer settings. DSP analyzes and compares these signals and determines a nominal pink noise level for the signals received from each microphone (3). Whichever speaker (2) is producing pink noise at any given moment, the microphone (3) nearest to said speaker (2) has the highest priority as the microphone (3) nearest to said speaker (2) receives the least reflected noise.

DSP then designates the level falling above or below the pink noise level for different frequency bands as tolerance and saves onto a memory unit. After completing this procedure for each speaker (2) and each microphone (3), the recorded tolerance spectrum is analyzed and a certain standard deviation will be applicable. As a result, DSP determines which band of the equalizer should be filtered more and which should be filtered less and will automatically tune the audio. In this manner, the ambient sound performance for image display devices having speakers (2) facing directly behind them is greatly improved.

In an ideal case, the data received by the microphones (3) after pink noise is emitted by the speakers (2) will look like Figure 3. The time between t₀ when the speakers (2) begin to emit noise and the time when the sound is converted to an electrical signal by the microphone (3) is designated Δ_t. However, the actual received signal will look like Figure 4.

The first step is to analyze the real sound wave taking into account the Bode plot of the microphone with known characteristics. To do this, the nominal inverse Bode plot is determined. And the convolution of electrical response and normalized inverse Bode plot is determined (Figure 6). In This manner, the real ambient sound is determined. The real ambient sound is the sound reaching the user’s ear. Therefore, all modifications in the equalizer will be done according to this curve.

After pink noise is received for each speaker (2) and microphones (3) of the system and the collected data is analyzed, the equalizer is optimized and group delay tests are performed. In order to calculate group delay, a frequency scan is started at each speaker (2) (one speaker at a time) by the system. The starting instant of the frequency scan is known and each frequency is set to have the same power.

Then, the frequency scan started at t₀ is received by all microphones and the signal is recorded to be analyzed. The recorded data is divided in the time domain and FFT analysis is performed for each sample (Figure 8).

Therefore, as the starting time for broadcast is known, which frequency is broadcast at which instant is also known. As the group delay characteristics of the microphones (3) are also known, the group shift of the frequency value of the analyzed chunk can be determined by subtracting the broadcast time of the frequency from the speaker (2) and the group shift of the microphone (3) at that frequency from the time of the chunk.

As a result of the FTT analysis on chunk 2, the dominant frequency is determined to be f₀. The group delay of f₀ can be found by subtracting the group delay of the microphone of at that frequency (t_m) from the transmission time of chunk 2 ((2*Δ_t+3*Δ_t)/2=2.5*Δ_t). The delay caused by electrical transmission and processes can be neglected as it is negligible compared to mechanical delay. For the analysis of low frequencies such as 200 Hz, multiple chunks must be analyzed simultaneously. In this case, the chunks are analyzed as a whole. First, all chunks will be analyzed together and then the chunks will be analyzed in groups based on the obtained FTT data (Figure 9).

This process is repeated until group shifts are determined for all frequencies. After group shift data is obtained for each frequency, the system will be configured for video and audio synchronization.

Video and audio data received by the system in real time are first recorded to cache memory 1. How much data will be recorded to cache memory 1 is calculated based on the maximum group delay. After sufficient data is recorded to cache memory 1, subsequent data is recorded to cache memory 2. In this time period, the sound analysis of the audio data in cache memory 1 will be performed, the audio data will be rearranged according to group delay and recorded to the memory (shared with DSP). In this time period, the video on the screen will be shifted. The rearranged data will be recorded to memory and audio and video will be synchronized and transmitted (Figure 10).

In a nutshell, the present invention proposes 1) A method for operating an image display device (1), said device comprising a plurality of speakers (2) and microphones (3) placed on a rear surface portion opposite a front display portion of said image display device (1), said method is characterized by separately producing pink noise by each speaker (2) in order and detection of the produced noise by each microphone (3), converting detected noise to an electrical signal, performing FFT analysis for each microphone (3), checking whether FFT analysis results fall within predetermined values and calculating equalizer settings to obtain predetermined values.

In one embodiment of the present invention, the method further comprises the steps of comparing electrical signal detected by each microphone (3) to determine a nominal pink noise level for the signals received, designating the level falling above or below the pink noise level for different frequency bands as tolerance for each speaker (2) and each microphone (3), analyzing the obtained tolerance spectrum to determine which band of the equalizer is filtered more and which band is filtered less to automatically determine equalizer settings.

In a further embodiment of the present invention, the method further comprises the steps of extracting FFT (Fast Fourier Transform) of received signal, determining the nominal pink noise line on FFT plot, calculating the RSSI (Received Signal Strength Indicator) error for each frequency band and passing error information of each microphone (3) to a control circuit.

In a further embodiment of the present invention, the method further comprises the steps of orderly starting frequency sweep at each speaker (2) to be received by all microphones (3), performing peak frequency time analysis for all microphones (3), comparing obtained data with speaker input signals and calculating maximum group delay for each speaker (2).

In a further embodiment of the present invention, the method further comprises the steps of orderly starting a frequency scan at each speaker (2) to be received by all microphones (3), dividing received data in the time domain and performing FFT analysis for each sample, determining group shift of the frequency value of an analyzed chunk by subtracting the broadcast time of the frequency from the speaker (2) and the group shift of the microphone (3) at that frequency from the time of the chunk.

In a further embodiment of the present invention, the method further comprises the steps of determining group shifts for all frequencies and calculating maximum frame number to be stored on cache memory depending on maximum group delay and separately storing audio and video data on cache memory.

In a further embodiment of the present invention, said microphones are placed in acute angles so as to be forming an acute angle by linear projection lines extending towards a reflection surface (4) in between a speaker (2) and a microphone (3).

In a further embodiment of the present invention, at least one microphone (3) is placed substantially close to each speaker (2).

In a further embodiment of the present invention, at least one microphone (3) is placed along the edges portions on the rear surface of the image display device (1).

In a further embodiment of the present invention, each frequency is set to have the same power.

Claims

A method for operating an image display device (1), said device comprising a plurality of speakers (2) and microphones (3) placed on a rear surface portion opposite a front display portion of said image display device (1), said method is characterized by separately producing pink noise by each speaker (2) in order and detection of the produced noise by each microphone (3), converting detected noise to an electrical signal, performing FFT analysis for each microphone (3), checking whether FFT analysis results fall within predetermined values and calculating equalizer settings to obtain said predetermined values.
A method as in Claim 1, the method further comprising the steps of comparing electrical signal detected by each microphone (3) to determine a nominal pink noise level for the signals received, designating the level falling above or below the pink noise level for different frequency bands as tolerance for each speaker (2) and each microphone (3), analyzing the obtained tolerance spectrum to determine which band of the equalizer is filtered more and which band is filtered less to automatically determine equalizer settings.
A method as in Claim 1 or 2, the method further comprising the steps of extracting FFT of received signal, determining the nominal pink noise line on FFT plot, calculating the RSSI error for each frequency band and passing error information of each microphone (3) to a control circuit.
A method as in Claim 2 or 3, the method further comprising the steps of orderly starting frequency sweep at each speaker (2) to be received by all microphones (3), performing peak frequency time analysis for all microphones (3), comparing obtained data with speaker input signals and calculating maximum group delay for each speaker (2).
A method as in Claim 2, 3 or 4, the method further comprising the steps of orderly starting a frequency scan at each speaker (2) to be received by all microphones (3), dividing received data in the time domain and performing FFT analysis for each sample, determining group shift of the frequency value of an analyzed chunk by subtracting the broadcast time of the frequency from the speaker (2) and the group shift of the microphone (3) at that frequency from the time of the chunk.
A method as in Claim 4 or 5, the method further comprising the steps of determining group shifts for all frequencies and calculating maximum frame number to be stored on cache memory depending on maximum group delay and separately storing audio and video data on cache memory.
An image display device (1) performing the method of Claim 1, characterized in that said microphones (3) are placed in acute angles so as to be forming an acute angle between linear projection lines from a speaker (2) and a microphone (3) extending towards a reflection surface (4).
An image display device (1) as in Claim 7, characterized in that at least one microphone (3) is placed substantially close to each speaker (2).
An image display device (1) as in Claim 7 or 8, characterized in that at least one microphone (3) is placed along the edges portions on the rear surface of the image display device (1).
An image display device (1) performing the method of Claim 4 or 5, characterized in that each frequency is set to have the same power.