Method for tuning and adapting sound field uniformity based on audio processor

Abstract

The present invention discloses a method for tuning and adapting the uniformity of the sound field based on an audio processor. After the tuning is started, it is realized by automatic program control until the tuning is finished. During the tuning, the audio processor gradually sets an inverting combination and controls the loudspeaker to play pink noise, and the microphone collects the signal to form a closed-loop system. On the basis of adjusting the input and output gains, the inverting combination is gradually set to preliminarily obtain an inverting configuration suitable for the current uniformity of the sound field. After fine-tuning the input and output gains, the inverting combination is gradually set to obtain the final inverting configuration. The method of the present invention can make the high-pitched debugging results all be obtained according to the audio algorithm, and the consistency of the final result is strong.

Description

Technical Field

[0001] The present invention relates to the technical field of audio processing, and in particular to a method for tuning and adapting sound field uniformity based on an audio processor. Background Art

[0002] As a professional audio processing device, the audio processor has complex audio algorithms such as echo cancellation, intelligent noise reduction, feedback suppression, and a rich set of input and output hardware interfaces. When using audio processors for interactive teaching and local sound reinforcement, due to the complexity of the building environment, the diversity of input and output audio devices, and the inconsistency of the connection methods, there will be inconsistencies in the uniformity of the ambient sound field. To solve the above problems, China's CN113747336A patent discloses a method for tuning and adapting the sound field based on an audio processor, which can automatically adjust the hardware gain and algorithm parameters suitable for the sound field, etc., which solves the above problems to a certain extent, but still fails to achieve a more consistent output effect. Summary of the invention

[0003] The purpose of the present invention is to provide a method for tuning and adapting the uniformity of the sound field based on an audio processor in order to solve the problems in the prior art. The method of the present invention can ensure that the debugging results are obtained according to the audio algorithm, and the consistency of the final results is relatively strong.

[0004] The technical solution adopted to achieve the purpose of the present invention is:

[0005] A method for tuning and adapting the uniformity of a sound field based on an audio processor, after the tuning is started, is implemented by automatic program control until the tuning is completed, comprising the following steps;

[0006] S1. The audio processor controls all speakers to play white noise and collects signals from the microphone. It determines whether a microphone is connected and the type of microphone connected based on the amplitude of the collected signal of each microphone channel, and sets different hardware gains for different microphones.

[0007] S2. The audio processor makes each output channel play a sinusoidal signal of a predetermined frequency one by one according to the output channel number, collects signals with all connected microphones, and determines whether each output channel is connected to a speaker according to the maximum amplitude and harmonic distortion rate of the collected signals of all connected microphones, and determines the number of speaker channels connected to the audio processor;

[0008] S3. Based on the number of connected microphones and speakers, the audio processor makes each channel connected to the speaker play white noise collected by all connected microphones one by one, and compares the maximum signal amplitude collected by all microphones with the preliminary debugging threshold of the speaker gain to obtain the difference between the actual collected amplitude and the preliminary debugging threshold of the speaker gain, and uses the difference to adjust the hardware gain of the corresponding speaker one by one;

[0009] S4. The audio processor controls all detected speakers to play white noise simultaneously, compares the amplitude of the signal collected by the detected microphone with the overall debugging threshold of the speaker gain, and readjusts the hardware gain of all speakers;

[0010] S5. The audio processor sets the inverting configuration of each output channel connected to the speaker to form an inverting configuration group of multiple speaker output channels; pink noise is played through all speakers at the same time and collected by all microphones, and the overall amplitude of the collected signal and the current inverting configuration group are recorded, and the inverting configuration of the output channel is gradually modified to form multiple inverting configuration groups, and the overall amplitude of the microphone collected signals of different inverting configuration groups is recorded. According to the size of the overall amplitude of the collected signals of all recorded microphones, the combination with the smallest overall amplitude is obtained, and the corresponding inverting configuration is set to the audio processor;

[0011] S6. Enter the microphone gain recheck program, control all speakers to play white noise at the same time, compare the amplitude of the signal collected by each microphone with the microphone recheck threshold, and adjust the microphone gain accordingly;

[0012] S7. Enter the speaker inversion re-test procedure, repeat step S5, re-derive the inversion configuration group, and obtain the final inversion configuration group.

[0013] When the number of speakers is equal to 1, no inversion operation is performed; when the number of speakers is greater than 1, inversion settings are gradually performed starting from the second speaker to form multiple inversion configuration groups, and each inversion configuration group plays pink noise collected by a microphone.

[0014] The setting of different hardware gains for different microphones is the gain inside the audio processor.

[0015] Among them, judging whether a microphone is connected and the type of microphone connected according to the amplitude of the collected signal of each microphone channel includes:

[0016] The signal amplitude collected by each microphone is compared with the detection threshold of all microphones. If the signal amplitude is greater than the detection threshold of all microphones, it means that the microphone is connected. Otherwise, there is no microphone or speaker and the tuning fails. If the signal amplitude is greater than the far-field microphone detection threshold, it is judged that the channel is connected to the far-field pickup microphone. If it is greater than the detection threshold of all microphones and less than the far-field microphone detection threshold, it is judged that a handheld microphone is connected.

[0017] The method of determining whether each output channel is connected to a speaker according to the maximum amplitude and harmonic distortion rate of all connected microphone acquisition signals includes:

[0018] When the maximum amplitude of the signals collected by all connected microphones corresponding to an output channel is greater than the speaker detection threshold, and the harmonic distortion rate is greater than the harmonic distortion rate threshold, it is determined that the output channel is connected to a speaker, otherwise it is confirmed that there is no speaker and the tuning is invalid.

[0019] The hardware gain of the speaker corresponding to the adjustment is the gain inside the audio processor.

[0020] The audio processor is provided with a one-touch tuning key, and when the one-touch tuning key is pressed, the internal tuning program of the audio processor is started and the tuning program is directly entered.

[0021] The present invention is based on a method for tuning an audio processor to adapt to the uniformity of the sound field. After adjusting the phase of the audio processor output signal according to the uniformity of the sound field of the environment, the volume of the speaker playback signal collected by the microphone can be minimized, which can effectively improve the echo cancellation effect of the audio processor in remote interactive applications and the feedback suppression effect in local sound reinforcement applications. BRIEF DESCRIPTION OF THE DRAWINGS

[0022] Figure 1 It is a schematic diagram of the architecture of the audio processor, microphone and speaker tuning system of the present invention.

[0023] Figure 2 It is a schematic diagram of the arrangement of the microphone and the loudspeaker in the sound field and signal transmission of the present invention.

[0024] Figure 3 The present invention is a flow chart of a method for tuning and adapting sound field uniformity based on an audio processor. DETAILED DESCRIPTION

[0025] The present invention will be further described in detail below in conjunction with the accompanying drawings and specific embodiments. It should be understood that the specific embodiments described herein are only used to explain the present invention and are not used to limit the present invention.

[0026] Figure 1The audio processor of the embodiment of the present invention has multiple microphone access channels and multiple speaker access channels, and is connected to the speaker and microphone through its multiple signal interfaces through signal lines or connecting lines. Figure 1 As shown, the Figure 1 It is shown that the audio processor has multiple interfaces, including input interfaces and output interfaces, such as 1-8 channels, to facilitate the connection of corresponding microphones or speakers. When tuning, the test signal is played through the speaker and collected by the microphone.

[0027] Figure 2 Schematic diagram of the spatial position of microphones and speakers in an embodiment of the present invention. The speakers and microphones are placed at different positions in a space. For example, four speakers are arranged in a classroom. Two speakers are arranged at the back end of the classroom with intervals, and the other two are arranged on both sides of the middle. The microphones are arranged on both sides in front of the speakers at the front end of the classroom. Students are located in a sound field environment surrounded by microphones and speakers. This is a typical classroom layout environment. The audio processing configuration has a one-touch tuning key. When tuning is required, the one-touch tuning key is pressed, and the tuning program inside the audio processor is started, and the tuning program is directly entered. Press Figure 3 The process shown performs automatic tuning processing.

[0028] When the one-key tuning key is pressed to start the tuning program for tuning, the audio processor gradually sets the inverting combination and controls the loudspeaker to play pink noise, and the microphone collects signals to form a closed-loop system; on the basis of adjusting the input and output gains by the one-key tuning, the inverting combination is gradually set to preliminarily obtain an inverting configuration suitable for the current sound field uniformity, and after fine-tuning the input and output gains, the inverting combination is gradually set to obtain the final inverting configuration.

[0029] The method for tuning and adapting the sound field uniformity based on the audio processor of the embodiment of the present invention refers to Figure 3 As shown, the Figure 3 The flow chart of the method for tuning and adapting the sound field uniformity based on the audio processor according to the embodiment of the present invention is shown. After the tuning program of the audio processor is started, the method comprises the following steps:

[0030] S1. The audio processor controls all speakers to play white noise at the same time. The microphone collects signals. The microphone detection threshold is compared with the signal amplitude collected by the microphone of each channel to determine whether the microphone is connected and the type of microphone connected. Different hardware gains are given to different microphone types.

[0031] Specifically, the different hardware gains assigned to different microphone types are gains inside the audio processor.

[0032] The following method is used to determine whether a microphone is connected and the type of microphone connected:

[0033] The signal amplitude collected by the microphone of each channel is compared with the detection threshold of all microphones to determine whether there is a microphone. If the signal amplitude is greater than the detection threshold of all microphones, it means that a microphone is connected. Otherwise, there is no microphone or speaker and the tuning fails. If the signal amplitude is greater than the far-field microphone detection threshold, it is determined that the channel is connected to a far-field pickup microphone. If it is greater than the detection threshold of all microphones and less than the far-field microphone detection threshold, it is determined to be a handheld microphone, thereby realizing the judgment and processing of the microphone type.

[0034] S2. The audio processor enables each output channel speaker to play a sinusoidal signal of a predetermined frequency, such as 1500Hz, one by one according to the output channel sequence number, and then uses all connected microphones to collect signals. According to the maximum amplitude and harmonic distortion rate of the signals collected by all connected microphones, it determines whether each output channel is connected to a speaker, and determines the number of speaker channels and the number of speakers connected to the audio processor.

[0035] Specifically, when the maximum amplitude of the signals collected by all connected microphones corresponding to an output channel is greater than the speaker detection threshold, and the harmonic distortion rate is greater than the harmonic distortion rate threshold, it is determined that the output channel is connected to a speaker, otherwise it is confirmed that there is no speaker and the tuning is invalid.

[0036] S3. Based on the determination of the number of microphones and speakers, the audio processor enables each connected speaker to play white noise collected by all connected microphones one by one, and compares the maximum signal amplitude collected by all microphones with the speaker single-step debugging threshold to obtain the difference between the actual collected amplitude and the speaker single-step debugging threshold, and then uses the difference to adjust the hardware gain of the corresponding speaker one by one.

[0037] The hardware gain of the loudspeaker corresponding to the above adjustment is the gain inside the audio processor.

[0038] S4. Control all detected speakers to play white noise at the same time, compare the debugging thresholds of all speakers according to the amplitude of the signal collected by the detected microphone, resynchronize and adjust the gains of all speakers, and increase or decrease the gains of all speakers at the same time;

[0039] S5. The audio processor will gradually modify the speaker output inversion to form an inverted combination of multiple output channels, and then play pink noise through all speakers to be collected by all microphones. According to the amplitudes collected by all microphones, a combination with the smallest amplitude is obtained, and the obtained inverted combination is set.

[0040] Among them, when the number of speakers is 1, the inversion operation will not be performed; when the number of speakers is greater than 1, the inversion setting will be gradually performed starting from the second speaker to form multiple inversion configuration groups, and each inversion configuration group plays pink noise collected by the microphone. Taking 4 speakers as an example, 8 inversion configuration groups will be formed: positive-positive-positive-positive, positive-positive-positive-negative, positive-positive-negative, positive-negative-positive, positive-negative-positive, positive-positive-negative-negative, positive-positive-negative-negative, positive-negative-negative-negative, positive-negative-negative-negative, positive-negative-negative-negative, positive-negative-negative-negative-negative, positive-negative-negative-negative-negative, and positive-negative ...

[0041] Among them, the anti-phase is a 180-degree reversal of the phase of the wave in physics, and positive means no reversal. Under theoretical circumstances, two speakers, one positive and one negative, will cancel each other out to 0 at the center point. In practice, they will not cancel each other out to 0, and the volume at each position in the room after cancellation will vary greatly.

[0042] The present invention refers to maximizing the degree of cancellation at the microphone position, that is, after adjusting the phase of the audio processor output signal according to the uniformity of the sound field of the environment, the volume of the speaker playback signal collected by the microphone is minimized, thereby effectively improving the audio processor's echo cancellation effect in remote interactive applications and feedback suppression effect in local sound reinforcement applications.

[0043] S6. After completing the above steps, enter the re-test procedure, control all speakers to play white noise at the same time, compare the amplitude of the sound collected by each microphone with the microphone re-test threshold, and adjust the microphone gain accordingly, please refer to step S1.

[0044] S7. Enter the re-check inversion setting procedure, that is, re-execute step S5 to obtain and set the final inversion combination parameters.

[0045] In the embodiment of the present invention, the one-touch tuning key is provided on the audio processor to facilitate tuning operations.

[0046] The present invention is automated, which can simplify the debugging steps and reduce time costs; since the present invention is automatically adjusted by a program, the debugging effect can be relatively stable, and the final effects of different spaces are consistent.

[0047] The above is only a preferred embodiment of the present invention. It should be pointed out that, for ordinary technicians in this technical field, several improvements and modifications can be made without departing from the principle of the present invention. These improvements and modifications should also be regarded as the scope of protection of the present invention.

Claims

1. A method for tuning and adapting the sound field uniformity based on an audio processor, after the tuning is started, it is realized by automatic program control until the tuning is completed, characterized in that: Includes steps: S1. The audio processor controls all speakers to play white noise, and the microphone collects white noise signals. It determines whether a microphone is connected and the type of microphone connected based on the amplitude of the collected signal of each microphone channel, and sets different hardware gains for different microphones. S2. The audio processor makes each output channel play a sine signal of a predetermined frequency one by one according to the output channel number, collects the sine signal with all connected microphones, and determines whether each output channel is connected to a speaker and the number of speaker channels connected to the audio processor according to the maximum amplitude and harmonic distortion rate of the collected signal of all connected microphones; S3. Based on the number of connected microphones and speakers, the audio processor plays white noise in each channel connected to the speaker one by one, and all connected microphones collect white noise signals. The maximum signal amplitude collected by all microphones is compared with the preliminary debugging threshold of the speaker gain to obtain the difference between the actual collected amplitude and the preliminary debugging threshold of the speaker gain, and the difference is used to adjust the hardware gain of the corresponding speaker one by one; S4. The audio processor controls all detected speakers to play white noise simultaneously, and all connected microphones collect white noise signals. The signal amplitude collected by the detected microphones is compared with the overall debugging threshold of the speaker gain, and the hardware gain of all speakers is readjusted; S5. The audio processor sets the inverting configuration of each output channel connected to the speaker to form an inverting configuration group of multiple speaker output channels; all speakers play pink noise at the same time, and all connected microphones collect pink noise signals, record the overall amplitude of the collected signals and the current inverting configuration group, gradually modify the inverting configuration of the output channel to form multiple inverting configuration groups, record the overall amplitude of the microphones collecting signals of different inverting configuration groups, and obtain the combination with the smallest overall amplitude based on the overall amplitude of all recorded microphones collecting signals, and set the corresponding inverting configuration to the audio processor; S6. Enter the microphone gain recheck program, control all speakers to play white noise at the same time, collect white noise signals from all connected microphones, compare the amplitude of the signal collected by each microphone with the microphone recheck threshold, and adjust the microphone gain accordingly; S7. Enter the speaker inversion re-test procedure, repeat step S5, re-derive the inversion configuration group, and obtain the final inversion configuration group.

2. The method for tuning and adapting the sound field uniformity based on an audio processor according to claim 1, characterized in that: When the number of speakers is equal to 1, no inversion operation is performed; when the number of speakers is greater than 1, inversion settings are gradually performed starting from the second speaker to form multiple inversion configuration groups, each of which plays pink noise, and the pink noise signal is collected by a microphone.

3. The method for tuning and adapting the sound field uniformity based on an audio processor according to claim 1, characterized in that: The setting of different hardware gains for different microphones refers to the gain inside the audio processor.

4. The method for tuning and adapting the sound field uniformity based on an audio processor according to claim 1, characterized in that: Determine whether a microphone is connected and the type of microphone connected based on the amplitude of the collected signal of each microphone channel, including: Compare the signal amplitude collected by each microphone with the detection threshold of all microphones. If the signal amplitude is greater than the detection threshold of all microphones, it means that the microphone is connected. Otherwise, there is no microphone and the tuning fails. If the signal amplitude is greater than the far-field microphone detection threshold, it is judged that the channel is connected to the far-field pickup microphone. If it is greater than the detection threshold of all microphones and less than the far-field microphone detection threshold, it is judged that a handheld microphone is connected.

5. The method for tuning and adapting the sound field uniformity based on an audio processor according to claim 1, characterized in that: The step of determining whether each output channel is connected to a speaker according to the maximum amplitude and harmonic distortion rate of all connected microphone acquisition signals includes: When the maximum amplitude of the signals collected by all connected microphones corresponding to an output channel is greater than the speaker detection threshold, and the harmonic distortion rate is greater than the harmonic distortion rate threshold, it is determined that the output channel is connected to a speaker, otherwise it is confirmed that there is no speaker and the tuning is invalid.

6. The method for tuning and adapting the sound field uniformity based on an audio processor according to claim 1, characterized in that: The hardware gain of the loudspeaker corresponding to the adjustment is the gain inside the audio processor.

7. The method for tuning and adapting the sound field uniformity based on an audio processor according to claim 1, characterized in that: The audio processor is configured with a one-touch tuning key. When the one-touch tuning key is pressed, the internal tuning program of the audio processor is started and the tuning program is directly entered.

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