阅读说明：本技术 音频处理装置与相关的音频处理方法 (Audio processing device and related audio processing method ) 是由 吴佳哲 于 2020-05-21 设计创作，主要内容包括：本发明揭露一种音频处理装置,其包括有一滤波器以及一输出电路。该滤波器用以接收一声音信号以产生一滤波后声音信号,其中该滤波器具有可调整的多个系数,以供改变该滤波器的响应的带宽、中心频率或是增益；以及该输出电路用以接收该滤波后声音信号以产生一输出声音信号至一扬声器；其中当该滤波器的该多个系数改变时,该滤波器降低该多个系数对该声音信号所造成的变化,且该输出电路持续接收该滤波后声音信号以产生该输出声音信号,以供该扬声器不中断地播放该输出声音信号。(The invention discloses an audio processing device, which comprises a filter and an output circuit. The filter is used for receiving an acoustic signal to generate a filtered acoustic signal, wherein the filter has a plurality of adjustable coefficients for changing the bandwidth, the center frequency or the gain of the response of the filter; the output circuit is used for receiving the filtered sound signal to generate an output sound signal to a loudspeaker; when the coefficients of the filter are changed, the filter reduces the change of the coefficients to the sound signal, and the output circuit continuously receives the filtered sound signal to generate the output sound signal, so that the loudspeaker can play the output sound signal without interruption.)

1. An audio processing apparatus comprising:

a filter for receiving an audio signal to generate a filtered audio signal, wherein the filter has a plurality of coefficients that are adjustable to vary a bandwidth, a center frequency, or a gain of a response of the filter; and

the output circuit is coupled with the filter and used for receiving the filtered sound signal to generate an output sound signal to the loudspeaker;

when the coefficients of the filter are changed, the filter reduces the change of the coefficients to the sound signal, and the output circuit continuously receives the filtered sound signal to generate the output sound signal, so that the loudspeaker can play the output sound signal without interruption.

2. The audio processing apparatus according to claim 1, wherein the filter gradually decreases the change of the plurality of coefficients to the sound signal before the plurality of coefficients of the filter change, so that the filtered sound signal gradually approaches the sound signal.

3. The audio processing apparatus according to claim 2, wherein the filter sequentially reduces the variation of the plurality of coefficients to the sound signal before the plurality of coefficients of the filter are changed, so that the filtered sound signal is equal to the sound signal.

4. The audio processing apparatus according to claim 2, wherein the filter gradually increases the variation of the plurality of coefficients to the sound signal to generate the filtered sound signal a period of time after the filtered sound signal approaches the sound signal.

5. The audio processing apparatus according to claim 1, wherein the transfer function of the filter is H (z) ═ 1+ H0 (1-a (z)), H0 is the adjustable gain value, and a (z) is a multi-step function.

6. The audio processing apparatus according to claim 5, wherein the filter gradually decreases the adjustable gain value to make the transfer function approach 1 before the plurality of coefficients of the filter change to cause the multi-order function to change.

7. The audio processing apparatus according to claim 6, wherein the filter gradually decreases the adjustable gain value to make the transfer function equal to 1 before the plurality of coefficients of the filter change to cause the multi-order function to change.

8. The audio processing apparatus according to claim 6, wherein the filter gradually raises the adjustable gain value to a target value for a period of time after the transfer function approaches 1.

9. An audio processing method, comprising:

using a filter to receive the sound signal to produce a filtered sound signal;

changing a plurality of coefficients of the filter to change a bandwidth, a center frequency, or a gain of a response of the filter, wherein a change of the sound signal caused by the plurality of coefficients of the filter is reduced when the plurality of coefficients are changed; and

the filtered sound signal is continuously received to generate an output sound signal, so that the speaker can play the output sound signal without interruption during the process of changing the plurality of coefficients of the filter.

10. The audio processing method of claim 9, wherein the step of reducing the variation of the sound signal caused by the coefficients of the filter when the coefficients are changed comprises:

before the coefficients of the filter are changed, the change of the coefficients to the sound signal is gradually reduced, so that the filtered sound signal gradually approaches the sound signal.

Technical Field

The present invention relates to an audio processing apparatus.

Background

In current audio processing devices, in order to allow a user to control the quality or characteristics of sound desired to be heard, a filter having a plurality of adjustable coefficients is typically provided for varying the bandwidth, center frequency and/or gain of the filter's response by varying the coefficients. However, since the filter may cause the response of the filter to diverge when the coefficient is changed directly, thereby causing abnormal sound, and even in the case that the response of the filter is not diverged, pop sound may be caused by too large a fall of the output sound signal, the conventional audio processing device may additionally provide a mute circuit to temporarily stop the speaker from playing sound when the coefficient of the filter is changed, thereby preventing the user from hearing the abnormal sound signal. However, although the use of the mute circuit can prevent the user from hearing the abnormal sound signal, the user still feels the sound interruption, which affects the overall listening quality.

Disclosure of Invention

Therefore, one of the objectives of the present invention is to provide an audio processing apparatus and an audio processing method, which can directly change the coefficients of the filter without interrupting the sound, and the user does not feel the sound abnormality and the pop sound, so as to solve the problems described in the prior art.

In one embodiment of the present invention, an audio processing apparatus is disclosed, which includes a filter and an output circuit. The filter is used for receiving an acoustic signal to generate a filtered acoustic signal, wherein the filter has a plurality of adjustable coefficients for changing the bandwidth, the center frequency or the gain of the response of the filter; the output circuit is used for receiving the filtered sound signal to generate an output sound signal to a loudspeaker; when the coefficients of the filter are changed, the filter reduces the change of the coefficients to the sound signal, and the output circuit continuously receives the filtered sound signal to generate the output sound signal, so that the loudspeaker can play the output sound signal without interruption.

In another embodiment of the present invention, an audio processing method is disclosed, which includes the following steps: using a filter to receive an audio signal to generate a filtered audio signal; changing a plurality of coefficients of the filter to change a bandwidth, a center frequency, or a gain of a response of the filter, wherein a change of the sound signal caused by the plurality of coefficients of the filter is reduced when the plurality of coefficients are changed; and continuously receiving the filtered sound signal to generate an output sound signal, so that a loudspeaker can play the output sound signal without interruption in the process of changing the coefficients of the filter.

Drawings

Fig. 1 is a schematic diagram of an audio processing apparatus according to an embodiment of the invention.

Fig. 2 is a schematic diagram of the frequency response change of the filter.

Fig. 3 shows an architecture diagram of a filter according to an embodiment of the invention.

Fig. 4 is a flowchart of an audio processing method according to an embodiment of the invention.

Fig. 5 is a diagram illustrating a frequency response change of a filter according to an embodiment of the invention.

Detailed Description

Fig. 1 is a schematic diagram of an audio processing apparatus 100 according to an embodiment of the invention. As shown in fig. 1, the audio processing apparatus 100 includes a filter 110, an output circuit 120 and a control circuit 130, and the audio processing apparatus 100 is configured to process an audio signal Din to generate an output audio signal Dout, and then transmit the output audio signal Dout to a speaker 104 for playing after being processed by a back-end processing circuit 102. Specifically, the filter 110 in the audio processing apparatus 100 may be an Infinite Impulse Response (IIR) filter or other digital filter, which is used to filter the sound signal Din to generate a filtered sound signal Din'; then, the output circuit 120 processes the filtered sound signal Din' to generate an output sound signal Dout, and the back-end processing circuit 102 performs back-end processing on the output sound signal Dout to generate a signal Vout to the speaker 104 for playing. In one embodiment, the output sound signal Dout may be a digital signal, and the back-end processing circuit 102 includes a digital-to-analog converter, an analog amplifier, and the like; in another embodiment, the output circuit 120 includes a digital-to-analog converter, i.e., the output sound signal Dout is an analog signal, and the back-end processing circuit 102 may only include an analog amplifier and related interface circuits. In addition, in the embodiment, the audio processing apparatus 100 can be applied to any electronic apparatus having a sound playing function, such as a headset, a speaker, a notebook computer, a desktop computer, a tablet computer, a mobile phone, a television, and so on.

In this embodiment, in order to allow the user to control the quality or characteristics of the sound desired to be heard, the control circuit 130 in the audio processing apparatus 100 generates a plurality of control signals Vc according to the instructions of other components or the input signals of the user, so as to change the bandwidth, center frequency and/or gain of the response (filter response) of the filter by changing a plurality of coefficients in the filter 110. Taking fig. 2 as an example for explanation, assuming that the original response of the filter 110 has a bandwidth BW1, a center frequency Fc1 and a gain G1 through its own coefficient setting, the control circuit 130 may change the coefficient of the filter 110 according to the input signal of the user, so that the response of the filter 110 is changed to have a different bandwidth BW2, center frequency Fc2 and gain G2. However, since a period of time for stabilizing the coefficient of the filter 110 is required to change, a muting mechanism is required to prevent the speaker 104 from playing sound when the coefficient of the filter 110 is changed, so as to prevent the user from hearing abnormal sound or pop sound, but the muting mechanism may cause the sound playing to be interrupted and the listening quality to be affected. Therefore, the transfer function (transfer function) of the filter 110 in this embodiment has different designs, and the control signal Vc generated by the control circuit 130 can additionally control the filter 110 to generate no abnormal sound during the coefficient change process, and the speaker 104 can play the sound signal without interruption to improve the above problem.

Specifically, fig. 3 shows an architecture diagram of the filter 110 according to an embodiment of the present invention, wherein the architecture shown in fig. 3 is expressed mathematically, i.e., wherein the filter 110 includes a multiple-order function a (z), two adders 320 and 340, and a multiplier 330. The transfer function H (z) of the filter 110 can be represented as H (z) 1+ H0 (1-a (z)) in fig. 3, where H0 is an adjustable gain value, and the multi-order function a (z) can be represented as H (z)Or other form of multi-order function, and b0, b1, b2, a1, a2 are the coefficients of the filter 110. Reference is also made to the flow shown in fig. 4. In step 400, the process begins and the control circuit 130 receives commands from other components or user input signals to request a change in the frequency response (i.e., bandwidth, center) of the filter 110Frequency and/or gain). In step 402, the control circuit 130 generates the control signal Vc to gradually decrease the adjustable gain value H0 (e.g., step-down) shown in fig. 3, i.e., control the filter 110 to gradually decrease the variation of the sound signal Din caused by the coefficients, so that the filtered sound signal Din' gradually approaches the sound signal Din. In detail, when the adjustable gain value H0 decreases, the change H0 a (z) caused by the plurality of coefficients to the sound signal Din becomes smaller, so the transfer function H (z) of the filter 110 gradually approaches "1", and the filtered sound signal Din' gradually approaches the sound signal Din. In one embodiment, the adjustable gain value H0 may be finally decreased to 0, i.e., the filtered sound signal Din' is substantially equal to the sound signal Din.

In one embodiment, the amount of the gain change that the user cannot hear can be designed as the amount of the gain change that the user cannot hear each time the adjustable gain value H0 is decreased in the step-down process of the adjustable gain value H0.

In step 404, the control circuit 130 may temporarily stop controlling the filter 110, i.e. wait/delay for a short time, to take account of some timing errors or circuit delays.

In step 406, the control circuit 130 generates the control signal Vc to directly change the coefficients of the filter 100, i.e., to change the coefficients in the multi-step function a (z). In one embodiment, the coefficients of the filter 110 are directly replaced (e.g., the coefficients are directly replaced with the final target values), without any indirect or sequential change, to speed up the setting and operation of the filter 110.

In step 408, the control circuit 130 may temporarily stop controlling the filter 110, i.e. wait/delay for a short time, to take account of some timing error or circuit delay.

In step 410, the control circuit 130 generates the control signal Vc to gradually increase the adjustable gain value H0 (e.g., step-up) to a target value, i.e., controls the filter 110 to gradually increase the variation of the sound signal Din caused by the coefficients, so as to complete the process of changing the coefficients of the filter 110. In one embodiment, the amount of gain change that can not be heard by the user can be designed as the amount of gain change that can not be heard by the user every time the adjustable gain value H0 is increased in the step-up process of the adjustable gain value H0.

It should be noted that the above steps 404, 408 are optional steps, i.e. the steps 404, 408 can be removed from the flow without affecting the main operation of the present invention.

Fig. 5 is a diagram illustrating a flow of the change of the coefficient of the filter 110 shown in fig. 4. In fig. 5, the frequency response of the filter 110 originally has a bandwidth BW1, a center frequency Fc1 and a gain G1, and before the coefficients of the filter 110 need to be changed, the frequency response is close to "1" by gradually decreasing the adjustable gain value H0; then, after the coefficients of the filter 110 are set, the adjustable gain value H0 is gradually increased, so that the frequency response of the filter 110 has a bandwidth BW2, a center frequency Fc2 and a gain G2.

In the above process of changing the coefficient of the filter 110, the output circuit 120 will continuously generate the output sound signal Dout to the back-end processing circuit 102 for generating the signal Vout to the speaker 104 for playing continuously, and since the filtered sound signal Din' outputted by the filter 110 when the coefficient is changed to be in an unstable state is substantially equal to the sound signal Din, the normal sound quality can be provided to the user in the whole process of changing the coefficient without any muting mechanism, so that the problems in the prior art can be effectively solved.

The above-mentioned embodiments are merely preferred embodiments of the present invention, and all equivalent changes and modifications made by the claims of the present invention should be covered by the scope of the present invention.

[ notation ] to show

100: audio processing device

102: back-end processing circuit

104: loudspeaker

110: filter with a filter element having a plurality of filter elements

120: output circuit

130: control circuit

320. 340, and (3): adder

330: multiplier and method for generating a digital signal

400-410: step (ii) of

A (z): function of multiple orders

BW1, BW 2: bandwidth of

Fc1, Fc 2: center frequency

G1, G2: gain of

H0: adjustable gain value

Din: sound signal

Din': filtered sound signal

Dout: outputting a sound signal

Vc: control signal

Vout: signal