CN114333867A - Audio data processing method and device, call method, audio processing chip, electronic device and computer readable storage medium - Google Patents

Abstract

The present application discloses an audio data processing method and apparatus, a calling method, an audio processing chip, an electronic device, and a computer-readable storage medium. The method includes: performing linear filtering processing on the first audio data and the second audio data to obtain linear echo data; determining linear output data according to the second audio data and the linear echo data; audio data, determine the first state data and the second state data; determine the weight factor according to the first state data and the second state data, so as to perform weighted filtering processing on the linear output data, and obtain the third audio frequency sent to the first calling party data. Therefore, the present application can perform weighted filtering based on the current call state or adopt a corresponding suppression scheme for processing, so that the echo residual suppression processing can be performed considering the component characteristics of the echo residual under different call states, and the echo residual suppression effect can be improved effectively. Improve call quality.

Description

Audio data processing method and apparatus, calling method, audio processing chip, electronic device and computer readable storage medium

technical field

The present application relates to the technical field of audio data processing, and in particular, to an audio data processing method and apparatus, a calling method, an audio processing chip, an electronic device, and a computer-readable storage medium.

Background technique

As the application scenarios of audio call technology become more and more extensive, people have higher and higher requirements for call quality. During a normal call, when one party on the call speaks a voice, the voice of the other party is collected and transmitted to the other party of the call and played by the voice playback device of the other party's calling device, so that the other party of the call can hear. In this process, when the voice audio of one party on the call is played on the voice playback device of the other party's calling device, an echo will be generated in the space where the other party is located, that is, the played voice audio is played by the voice and audio in the space. The surface reflection of various walls or objects, and further when the other party responds to the voice of the calling party, it is collected by the voice acquisition device of the other party's communication device, and is regarded as the other party of the call. The voice is sent back to Party B of the call. Therefore, one party to the call may receive audio sounds whose own voice is transmitted to the other party of the call while speaking, that is, a call echo is generated, and such a call echo seriously affects the call experience of the call party.

In the prior art, the energy of the echo residual is usually estimated according to the time delay estimation result of the audio data and the output of the linear filter, so as to adjust the spectral gain of the signal after the linear echo processing. However, the existing echo processing solution only considers the residual energy of the echo in the audio data during the call. However, with the increasingly diversified application scenarios of the audio call technology, the echo will show different characteristics in different scenarios and environments. Therefore, using The unified residual energy is used as the benchmark to suppress echoes, which is difficult to meet people's requirements for audio call quality in changing and complex scenarios.

SUMMARY OF THE INVENTION

Embodiments of the present application provide an audio data processing method and apparatus, a calling method, an audio processing chip, an electronic device, and a computer-readable storage medium, so as to solve the defect of poor echo residual elimination effect in the prior art.

In order to achieve the above purpose, an embodiment of the present application provides an audio data processing method, including:

Perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data, wherein the first calling party and the second calling party are in the same During a call activity;

determining linear output data according to the second audio data collected by the second calling party and the linear echo data;

Determine first state data and second state data for identifying the state of an audio call between the first caller and the second caller based on the first audio data and the second audio data , wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the first state data is the linear echo data and the The average value of the ratio of the second audio data in each sub-band;

According to the first state data and the second state data, a weighting factor related to the call state is determined, so as to perform weighted filtering processing on the linear output data to obtain a third call sent to the first caller. audio data.

The embodiment of the present application also provides an audio data processing method, including:

Perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data, wherein the first calling party and the second calling party are in the same During a call activity;

determining linear output data according to the second audio data collected by the second calling party and the linear echo data;

First state data and second state data for identifying the state of an audio call between the first caller and the second caller are determined according to the first audio data and the second audio data , wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the first state data is the linear echo data and the The average value of the ratio of the second audio data in each sub-band;

selecting, according to the first state data and the second state data, a signal reduction amplitude value corresponding to the first state data and the second state data;

The signal amplitude reduction operation is performed on the linear output audio data according to the signal reduction amplitude value, so as to obtain audio data sent to the first calling party.

The embodiment of the present application also provides a method for calling, including:

receiving first audio data;

playing the first audio data;

performing audio capture processing to generate second audio data, wherein the second audio data includes at least audio data collected when the first audio data is played;

performing linear filtering processing on the second audio data to obtain linear echo data;

determining linear output data according to the second audio data and the linear echo data;

According to the first audio data and the second audio data, first state data and second state data for identifying the audio call state are determined, wherein the first state data is the first audio data and the the average value of the correlation coefficient of the second audio data in each sub-band, and the first state data is the average value of the ratio of the linear echo data and the second audio data in each sub-band;

According to the first state data and the second state data, determine a weighting factor related to the talking state, so as to perform weighted filtering processing on the linear output data to obtain third audio data;

The third audio data is output to the calling party making the call.

The embodiment of the present application also provides an audio processing chip, including:

an audio receiving module for receiving the first audio data;

audio output for playing the first audio data;

a sound pickup module, configured to perform audio collection processing to generate second audio data, wherein the second audio data at least includes audio data collected by the sound pickup module when playing the first audio data;

a filtering module, configured to perform linear filtering processing on the second audio data to obtain linear echo data;

a processing module, configured to determine linear output data according to the second audio data and the linear echo data, and determine first state data for identifying the audio call state according to the first audio data and the second audio data and second state data, wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the first state data is the linear an average value of the ratios of the echo data and the second audio data in each sub-band; and determining a weighting factor related to the call state according to the first state data and the second state data,

Wherein, the filtering module is configured to perform weighted filtering processing on the linear output data to obtain third audio data, and

The audio output module is used for outputting the third audio data to the calling party making the call.

The embodiment of the present application also provides an audio data processing device, including:

A filtering module, configured to perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data, wherein the first calling party and the The two call parties are in the same call activity;

a linear output module, configured to determine linear output data according to the second audio data collected by the second calling party and the linear echo data;

a state determination module for determining, according to the first audio data and the second audio data, a first state for identifying the state of an audio call between the first call party and the second call party data and second state data, wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in respective sub-bands, and the first state data is the the average value of the ratio of the linear echo data to the second audio data in each sub-band;

A suppression module, configured to determine a weighting factor related to the call state according to the first state data and the second state data, so as to perform weighted filtering processing on the linear output data, and obtain a result that is sent to the first The third audio data of the calling party.

The embodiment of the present application also provides an electronic device, including:

memory for storing programs;

The processor is configured to run the program stored in the memory, and when the program runs, the audio data processing method provided by the embodiment of the present application is executed.

Embodiments of the present application further provide a computer-readable storage medium on which a computer program executable by a processor is stored, wherein when the program is executed by the processor, the audio data processing method provided by the embodiments of the present application is implemented.

The audio data processing method and device, the calling method, the audio processing chip, the electronic device, and the computer-readable storage medium provided by the embodiments of the present application can be determined according to the audio data sent by the first calling party and the data collected by the second calling party. The first state data and the second state data used to identify the current audio call state; and then, according to the weighting coefficient or the corresponding suppression scheme related to the current audio call state determined by the first state data and the second state data. Targeted suppression processing is performed on the data collected by the microphone after linear filtering. Therefore, weighted filtering can be performed based on the current call state or a corresponding suppression scheme can be used for processing, so that the echo residual suppression processing can be performed considering the component characteristics of the echo residual under different call states, which can improve the echo residual suppression effect and effectively improve the call quality. .

The above description is only an overview of the technical solution of the present application. In order to be able to understand the technical means of the present application more clearly, it can be implemented according to the content of the description, and in order to make the above-mentioned and other purposes, features and advantages of the present application more obvious and easy to understand , and the specific embodiments of the present application are listed below.

Description of drawings

Various other advantages and benefits will become apparent to those of ordinary skill in the art upon reading the following detailed description of the preferred embodiments. The drawings are for purposes of illustrating preferred embodiments only and are not to be considered limiting of the application. Also, the same components are denoted by the same reference numerals throughout the drawings. In the attached image:

1 is a schematic diagram of an application scenario of an audio data processing method provided by an embodiment of the present application;

2 is a flowchart of an embodiment of an audio data processing method provided by the application;

3 is a flowchart of another embodiment of an audio data processing method provided by the present application;

4 is a schematic structural diagram of an embodiment of an audio data processing apparatus provided by the present application;

FIG. 5 is a schematic structural diagram of an embodiment of an electronic device provided by the present application.

Detailed ways

Exemplary embodiments of the present disclosure will be described in more detail below with reference to the accompanying drawings. While exemplary embodiments of the present disclosure are shown in the drawings, it should be understood that the present disclosure may be embodied in various forms and should not be limited by the embodiments set forth herein. Rather, these embodiments are provided so that the present disclosure will be more thoroughly understood, and will fully convey the scope of the present disclosure to those skilled in the art.

Example 1

The solutions provided by the embodiments of the present application can be applied to any communication system with audio processing capability, such as a communication device installed with an audio processing module, and so on. FIG. 1 is a schematic diagram of an application scenario of an audio data processing method provided by an embodiment of the present application. The scenario shown in FIG. 1 is only one example of a scenario to which the technical solution of the present application can be applied.

With the development of audio technology, the application scenarios of audio call technology have become more and more extensive, not only in daily audio calls, but also in the business field. Especially with the recent rise of telecommuting, more and more users use video conferencing or audio conferencing to communicate. Therefore, users have higher and higher requirements for call quality.

In the usual two-party call process, when one end of the call, such as the other end opposite to the local place, also called the far end, speaks the voice, which is collected by the other party or the far-end calling device and then transmitted to the The local place of the call, such as the near-end, is played by the audio playback device of the calling device at the near-end, so that the near-end can listen to the voice and audio sent by the far-end.

In this process, when the voice audio from the far end is played on the voice playback device of the near-end calling device, an echo will be generated in the space where the near-end is located, that is, the played voice and audio are played by the space where the near-end is located. and then is collected by the voice acquisition device of the near-end communication device, so as to be sent back to the far-end as the near-end voice audio. In this case, the far-end is usually still talking at this time, that is Continue to send out speech, and the near-end is actually in the listening state at this time and does not speak. However, since the echo generated by the audio from the far-end is echoed back to the far-end calling device, it will The echoed far-end own voice is played, so that the far-end can hear its own voice when speaking, that is, the echo of what was said before, that is, a call echo is generated at the far end, and such a call echo seriously affects improve the call experience of the caller.

For this reason, it has been proposed in the prior art to estimate the energy of the echo residual according to the information such as the time delay estimation result of the transmitted audio data and the output of the linear filter, so as to perform spectral gain adjustment on the signal after the linear echo processing, in order to Implements the suppression of the residual echo in the call audio. Specifically, in the prior art, for example, as shown in FIG. 1 , the downlink data from the far end, that is, the audio data that already contains the audio data sent by the far end, is played at the near end through, for example, a loudspeaker in the near end communication device. After the device is played, it can be collected by an audio acquisition device such as a microphone of the near-end communication device along the echo path shown in FIG. Perform delay alignment with the signal including the echo of the far-end audio data collected by the audio collection device such as a microphone in the near-end communication device, so as to obtain the delay estimation result of the far-end downlink data at the near end, and then Input the delay-adjusted signal and the signal collected by the audio acquisition device into the linear filter for linear echo estimation processing, so as to finally estimate the echo residual energy in the near-end uplink data based on the delay estimation result and the linear echo estimation result , so as to suppress the echo in the uplink data.

However, in actual use, there are many changes in the situation of both parties in the call. For example, one party may use headphones, the other party may use speakers, or both parties are talking, etc. In these cases, the collected audio signals are also very different. For example, in the scenario shown in Fig. 1, the echo residual collected by the microphone is very small when the earphone is used at the near end. Therefore, in the technical solution of the prior art, delay estimation and linear filtering estimation are still used. To estimate the echo energy may mistakenly identify the near-end speech as an echo. In this case, the near-end speech may be erroneously suppressed, thereby affecting the far-end listening to the near-end speech audio.

Therefore, with the increasingly diverse application scenarios of audio call technology, the echo components will show different characteristics in different scenarios and environments. Therefore, using a unified residual energy as a benchmark to suppress echoes is difficult to meet the changing and complex scenarios. meet people's requirements for audio call quality.

To this end, as shown in FIG. 1, a scenario is shown in which the call echo heard by the far end, ie, the other party at the local place of the call, is canceled. In the prior art, the first calling party at the far end can transmit the voice data sent out as downlink data to the near-end calling device in a wired or wireless manner. For example, in FIG. 1, the downlink data x(t) It is transmitted to the speaker and delay estimation module in the calling device of the second calling party as the near-end, so that while the voice audio of the far-end is played by the speaker of the near-end calling device, the downlink data x( t) and the audio data d(t) collected by the microphone of the near-end calling device are subjected to delay alignment processing to obtain a delay estimation result x(t'). Next, the delay estimation result x(t') can be further processed. ') together with the audio data d(t) collected by the microphone and input into the existing linear echo cancellation filter for linear echo estimation calculation, so as to obtain the linear echo estimation result y(t) and the corresponding linear output.

Different from the prior art, in the present application, a processing module may be provided in the communication device serving as the near-end second calling party or in the audio processing chip in the communication device, so that in the solution as in the prior art, a processing module may be provided. After obtaining the delay estimation result x(t') and the linear echo estimation result y(t) as in The second audio data d(t) collected by the microphone is used to further determine the current calling state of the second calling party. For example, in this embodiment of the present application, in the processing module, the first audio data x(t) transmitted by the first calling party and the audio data d(t) collected by the microphone of the near-end calling device may be used to determine the a state data and the second state data is determined according to the second audio data d(t) and the linear echo data y(t) transmitted by the second calling party.

For example, in the embodiment of the present application, in the solution shown in FIG. 1 , the first state data may be the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the second state data is the average value of the ratio of the linear echo data to the second audio data in each sub-band. Therefore, according to the embodiment of the present application, compared with the prior art, the current call state can be further determined according to the original output of the prior art, And according to the call state, the weighting factor of the call state is determined, and the weight factor is introduced in the echo suppression process in the prior art to consider the call state of the second call party, that is, according to the difference of the call state of the second call party. The echo residual components that may exist in the linear output results can be determined more clearly, so that further echo residual components can be suppressed for the linear output results of the solutions in the prior art, or in the embodiment of the present application, the echo residual components can also be suppressed according to the history Using empirical data to establish a mapping table between the call state and the gain adjustment scheme of the linear output, so that when the first state data and the second state data are determined, the pre-established query can be directly queried according to the determined first state data and the second state data. The corresponding adjustment scheme or gain adjustment factor is selected from the mapping table to directly perform echo residual processing on the linear output result.

Therefore, according to the present application, the first state data and the second state data for identifying the current audio call state can be determined according to the audio data sent by the first call party and the collected data of the second call party; The weighting coefficient related to the current audio call state determined by the state data and the second state data or a corresponding suppression scheme is used to perform targeted suppression processing on the linearly filtered microphone collection data. Therefore, weighted filtering can be performed based on the current call state or a corresponding suppression scheme can be used for processing, so that the echo residual suppression processing can be performed considering the component characteristics of the echo residual under different call states, which can improve the echo residual suppression effect and effectively improve the call quality. .

The foregoing embodiments are descriptions of the technical principles and exemplary application frameworks of the embodiments of the present application, and the specific technical solutions of the embodiments of the present application will be further described in detail below through multiple embodiments.

Embodiment 2

FIG. 2 is a flowchart of an embodiment of an audio data processing method provided by this application. The execution body of the method may be various Internet of Things terminals or devices with audio processing capabilities, or may be devices or chips integrated on these devices. . As shown in Figure 2, the audio data processing method includes the following steps:

S201: Perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data.

In the embodiment of the present application, when the first call of the far-end party at the far end opposite to the local place is directed to the second caller of the near-end in the same call activity, when the voice is sent as the first audio data, as shown in FIG. 1 display, at the local end, that is, the second calling party can play the first audio data when receiving the first audio data, for example, through the speaker of the near-end calling device, and through the microphone acquisition to obtain second audio data, so that linear filtering processing can be performed on the first audio data and the second audio data. For example, the received first audio data of the first calling party and the second audio data collected by the local calling device may be input into, for example, a linear AEC (Acoustic Echo Cancellation, acoustic echo cancellation) filtering module to obtain linearity echo data. That is, the linear echo data is separated by processing of a linear filter.

S202: Determine linear output data according to the second audio data and the linear echo data collected by the second calling party.

After the linear echo data is obtained in step S201, since the linear echo data can reflect the echo components related to the first audio data sent by the first calling party at the far end in the second audio data collected by the local communication device, Therefore, the linear output data can be further determined according to the second audio data collected by the second calling party that is the near end of the local place and the linear echo data obtained in S201.

S203: Determine, according to the first audio data and the second audio data, first state data and second state data for identifying the state of the audio call between the first call party and the second call party.

In this embodiment of the present application, since the first audio data is voice data sent by the first calling party at the far end, and the second audio data is data collected by, for example, a microphone of the communication device of the second calling party at the local place. Under normal circumstances, the local site as the near end may have three call states: using the headset to speak, using the loudspeaker to play the far party's call and the near end is not speaking, and using the speaker to play the far party's call and the near end Duan is talking.

For example, in the above three states, the second audio data collected by the near-end communication device may contain echo data of the first audio data. For example, when the near-end uses an earphone to listen to the far-end call or uses the earphone to speak, the second audio data collected by the microphone of the near-end communication device hardly includes the first audio data sent by the far end. When the near-end uses the speaker to broadcast the call of the far-end party and the near-end does not speak, since the near-end uses the speaker to play the far-end first audio data, the first audio data propagated in the space where the near-end is located will The second audio data collected by the near-end communication device and therefore included in the second audio data collected by the near-end communication device, especially in this state, since the near-end does not speak, the second audio data collected by the near-end communication device The data is almost always the first audio data. When the near-end uses a loudspeaker to amplify the far-end first audio data and the near-end is talking at the same time, not only the far-end first audio data but also the near-end is propagated in the space where the near-end is located. The voice data sent out in this space, therefore, the second audio data that can be collected by the near-end communication device can include not only the components of the far-end first audio data, that is, echo data, but also what the near-end is saying .

Therefore, in this embodiment of the present application, the call state of the near-end may be determined based on the first audio data and the second audio data in step S203, that is, the first state data and the second state that identify the call state of the near-end are determined data. For example, in this embodiment of the present application, the first state data may be obtained by performing frequency band averaging on the correlation coefficients of the first audio signal and the second audio signal in respective frequency bands. For example, in this embodiment of the present application, the second state data may be obtained by performing frequency band averaging on ratios of the separated linear echo data and the second audio data in each self-frequency band after being processed by a linear filter. That is, the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each subband, and the second state data is the average value of the ratios of the linear echo data and the second audio data in each subband . Therefore, the first state data and the second state data determined in this way can well distinguish the above-mentioned three call states of the second call party, so that weighted filtering can be performed based on the call state in subsequent processing.

S204 , according to the first state data and the second state data, determine a weighting factor related to the call state, so as to perform weighted filtering processing on the linear output data to obtain third audio data sent to the first caller.

Therefore, the weighting factor related to the current call state of the second calling party can be determined according to the first state data and the second state data determined in step S203, and the outputs of steps S201 and S202 can be directly used in step S204 to further The current call state, that is, the weighting factor of the call state, is subjected to weighted filtering processing.

In other words, the embodiments of the present application can introduce the weighting factor into the echo suppression process in the prior art to consider the call state of the second caller, that is, the linear output can be more clearly determined according to the difference of the caller state of the second caller The echo residual components that may exist in the result, so that the linear output result of the solution in the prior art can be further suppressed. The mapping table of the gain adjustment scheme of the linear output, so that when the first state data and the second state data are determined, the pre-established mapping table can be directly inquired according to the determined first state data and the second state data to select the corresponding Adjust the scheme or gain adjustment factor to directly perform echo residual processing on the linear output result.

Therefore, according to the present application, the first state data and the second state data for identifying the current audio call state can be determined according to the audio data sent by the first call party and the collected data of the second call party; The weighting coefficient related to the current audio call state determined by the state data and the second state data or a corresponding suppression scheme is used to perform targeted suppression processing on the linearly filtered microphone collection data. Therefore, weighted filtering can be performed based on the current call state or a corresponding suppression scheme can be used for processing, so that the echo residual suppression processing can be performed considering the component characteristics of the echo residual under different call states, which can improve the echo residual suppression effect and effectively improve the call quality. .

Embodiment 3

FIG. 3 is a flowchart of another embodiment of the audio data processing method provided by this application. The execution subject of the method may be various communication terminals or devices with audio processing capabilities, or may be devices or chips integrated on these devices. . As shown in Figure 3, the audio data processing method includes the following steps:

S301. Perform audio activity detection on the first audio data to determine whether the first audio data contains voice audio.

In the embodiment of the present application, since each module used for audio processing in the communication terminal consumes a large amount of power, the two parties usually do not talk all the time during the call. Perform audio activity detection processing (Voice Activity Detection, VAD) on the received first audio data of the remote end, so as to start the relevant audio processing module only when it is determined that the first audio data contains voice data, and at other times. The related audio processing modules are all in standby or dormant state, and only when the VAD processing in this step determines that the first audio data contains voice data, that is, when it is determined that the first communication party at the far end is speaking The audio processing module sends a wake-up signal or sends a notification signal to the central processing unit or controller of the communication device, so that the remote communication device can wake up the relevant audio processing module to perform processing on the first audio data containing the voice of the remote end. deal with.

S302: Perform delay alignment processing on the first audio data according to the second audio data.

In the case where it is determined in step S301 that the received first audio data contains the audio data of the far end, in step S302, the received first audio data can be compared according to the second audio data collected by the communication device of the near end. For example, delay alignment is performed by a delay estimation module, so as to perform delay adjustment on the received first audio. Under normal circumstances, since the local near-end plays the first audio data through a playback device such as a microphone of the communication device and collects the audio data propagated in the space through a collection module such as a speaker, that is, when the second audio data is obtained , the second audio data needs to propagate in the space where the near-end is located before it can be collected by the near-end communication device. Therefore, even if the near-end does not speak, the second audio data collected by the near-end communication device There is a time difference between the far-end speech component contained in the data and the first audio data, that is, the first audio data is played by the speaker of the near-end communication device, propagated through space, and then received by the microphone, which takes a period of time. The first audio data is not aligned on the timeline. Therefore, in step S302, the first audio data can be used to align the second audio data collected by the near-end communication device on the timeline, thereby facilitating the accuracy and efficiency of subsequent echo identification and comparison processing. .

S303: Perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data.

In this embodiment of the present application, when the far end that is located at the opposite end of the local place sends a voice to the near end as the first audio data, as shown in FIG. 1 , the local end, that is, the near end, can receive When the first audio data is used, the first audio data can be played, for example, through the speaker of the near-end calling device, and the second audio data can be obtained by collecting through the microphone, so that the first audio data and the The second audio data is subjected to linear filtering processing. That is, the first audio data is voice data sent by the first calling party to the second calling party during the calling activity, and the second audio data is audio data collected by the second calling party when playing the first audio data. For example, the received first audio data of the first calling party and the second audio data collected by the local calling device may be input into, for example, a linear AEC (Acoustic Echo Cancellation) filtering module to obtain linear echo data. That is, the linear echo data is separated by processing of a linear filter.

S304: Determine linear output data according to the second audio data and the linear echo data collected by the second calling party.

After the linear echo data is obtained in step S301, since the linear echo data can reflect the echo components related to the first audio data sent by the first calling party at the far end in the second audio data collected by the local communication device, Therefore, the linear output data can be further determined according to the second audio data collected by the second calling party that is the near end of the local place and the linear echo data obtained in S301.

S305: Determine the correlation coefficient of the first audio data and the second audio data in each sub-band, and determine the average value of each correlation coefficient as the first state data.

S306, determine the ratio of the linear echo data and the second audio data in each subband, and determine the average value of each ratio as the second state data.

In this embodiment of the present application, since the first audio data is voice data sent by the first calling party at the far end, and the second audio data is data collected by, for example, a microphone of the communication device of the second calling party at the local place. Under normal circumstances, the local site as the near end may have three call states: using the headset to speak, using the loudspeaker to play the far party's call and the near end is not speaking, and using the speaker to play the far party's call and the near end Duan is talking.

For example, in the above three states, the first audio data included in the second audio data collected by the near-end communication device, that is, the echo data, are different. For example, when the near-end uses an earphone to listen to the far-end call or uses the earphone to speak, the second audio data collected by the microphone of the near-end communication device hardly includes the first audio data sent by the far end. When the near-end uses the speaker to broadcast the call of the far-end party and the near-end does not speak, since the near-end uses the speaker to play the far-end first audio data, the first audio data propagated in the space where the near-end is located will The second audio data collected by the near-end communication device and therefore included in the second audio data collected by the near-end communication device, especially in this state, since the near-end does not speak, the second audio data collected by the near-end communication device The data is almost always the first audio data. When the near-end uses a loudspeaker to amplify the far-end first audio data and the near-end is talking at the same time, not only the far-end first audio data but also the near-end is propagated in the space where the near-end is located. The voice data sent out in this space, therefore, the second audio data that can be collected by the near-end communication device can include not only the components of the far-end first audio data, that is, echo data, but also what the near-end is saying .

Therefore, in this embodiment of the present application, the call state of the near-end may be determined based on the first audio data and the second audio data in steps S303 and S304, that is, in step S303, the first audio signal and the second audio The correlation coefficients of the audio signals in each sub-band are band-averaged to obtain the first state data Coh _XD , and in step S304, the ratio of the separated linear echo data and the second audio data in each sub-band can be processed by a linear filter. The frequency bands are averaged to obtain the second state data YDR. Therefore, in this case, the first state data obtained in step S303 and the second state data obtained in step S304 can be used to identify the state of the audio call between the first calling party and the second calling party. Therefore, the first state data and the second state data determined in this way can well distinguish the above-mentioned three call states of the second call party, so that weighted filtering can be performed based on the call state in subsequent processing.

S307: Determine a trade-off factor for controlling speech distortion according to the first state data and the second state data.

In this embodiment of the present application, the first state data obtained in step S303 and the second state data obtained in step S304 are used to determine the trade-off factor. In this embodiment of the present application, the following formula can be used to determine the trade-off factor.

Among them, _ΦYY and ΦEE are covariance matrices calculated using different frame signals. Therefore, in this embodiment of the present application, a trade-off factor that can be used to determine echo residual components can be obtained through the first state data and the second state data.

S308: Determine the inter-frame Wiener filter coefficients according to the trade-off factor, the linear echo data, and the linear output data.

After the trade-off factor is obtained in step S307, the inter-frame Wiener filter coefficient may be further determined in step S308 according to the trade-off factor in step S307 and the linear echo data and linear output data obtained in steps S303 and S304. For example, the following formulas can be used to determine the inter-frame Wiener filter coefficients.

where t is time, and f is the sound frequency, and e ₁ is the calculation parameter.

S309: Perform filtering processing on the linear output data according to the Wiener filter coefficients to obtain first output audio data.

In step S309, the Wiener filter coefficients determined in step S308 may be used to perform filtering processing on the linear output data obtained in step S304, so as to obtain the first output audio data. In the embodiment of the present application, the trade-off factor can be introduced into the echo suppression process in the prior art to consider the call state of the second call party, that is, according to the difference of the call state of the second call party, the linear output result can be more clearly determined. possible existing echo residual components, so that further echo residual components can be suppressed for the linear output result of the solution in the prior art.

Therefore, in the embodiment of the present application, the near-end talking state can be considered to introduce the judgment result of the current near-end talking state as a parameter when determining the filter coefficient, so as to improve the effect of echo suppression processing.

S310: Determine the average gain of the frequency band of the first output audio data.

S311, select a signal reduction amplitude value corresponding to the average gain of the frequency band.

S312 , performing an operation of reducing the signal amplitude on the first output audio data according to the signal reduction amplitude value.

After the first output audio data is obtained in step S309, the band average gain of the first output audio data obtained in step S309 may be further determined in step S310, and selected according to the band average gain obtained in step S309 in step S311 The corresponding signal reduces the amplitude value, so that in step S312, the selected signal reduction amplitude value is used to perform the operation of reducing the signal amplitude on the first output audio data obtained in step S309, so as to realize the operation of reducing the signal amplitude contained in the first output audio data. Further suppression of echo residuals.

In addition, in the embodiment of the present application, a mapping table between the call state and the gain adjustment scheme of the linear output can also be established according to historical experience data, so that when the first state data and the second state data are determined, the determined The first state data and the second state data of the first state data and the second state data query the pre-established mapping table to select the corresponding adjustment scheme or gain adjustment factor to directly perform echo residual processing on the linear output result.

For example, in this embodiment of the present application, instead of determining the trade-off factor, a signal reduction amplitude value corresponding to the first state data and the second state data may be selected, and an operation of reducing the signal amplitude is performed on the linear output audio data according to the signal reduction amplitude value , so that an incremental adjustment scheme for removing echo residues can be quickly determined according to historical experience data.

Therefore, according to the present application, the first state data and the second state data for identifying the current audio call state can be determined according to the audio data sent by the first call party and the collected data of the second call party; The weighting coefficient related to the current audio call state determined by the state data and the second state data or a corresponding suppression scheme is used to perform targeted suppression processing on the linearly filtered microphone collection data. Therefore, weighted filtering can be performed based on the current call state or a corresponding suppression scheme can be used for processing, so that the echo residual suppression processing can be performed considering the component characteristics of the echo residual under different call states, which can improve the echo residual suppression effect and effectively improve the call quality. .

Embodiment 4

FIG. 4 is a schematic structural diagram of an embodiment of an audio data processing apparatus provided by the present application, which can be used to execute the method steps shown in FIG. 2 and FIG. 3 . As shown in FIG. 4 , the audio data processing apparatus may include: a filtering module 41 , a linear output module 42 , a state determination module 43 and a suppression module 44 .

The filtering module 41 may be configured to perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data.

In the embodiment of the present application, when the first call of the far-end party at the far end opposite to the local place is directed to the second caller of the near-end in the same call activity, when the voice is sent as the first audio data, as shown in FIG. 1 display, at the local end, that is, the second calling party can play the first audio data when receiving the first audio data, for example, through the speaker of the near-end calling device, and through the microphone acquisition to obtain second audio data, so that linear filtering processing can be performed on the first audio data and the second audio data. For example, the received first audio data of the first calling party and the second audio data collected by the local calling device may be input into, for example, a linear AEC (Acoustic Echo Cancellation, acoustic echo cancellation) filtering module to obtain linearity echo data. That is, the linear echo data is separated by processing of a linear filter.

The linear output module 42 is configured to determine the linear output data according to the second audio data and the linear echo data collected by the second calling party.

After the filtering module 41 obtains the linear echo data, since the linear echo data can reflect the echo components related to the first audio data sent by the first calling party at the far end in the second audio data collected by the local communication device, Therefore, the linear output module 42 can be further utilized to determine the linear output data according to the second audio data collected by the second calling party as the local near-end and the linear echo data obtained by the filtering module 41 .

The state determination module 43 is configured to determine, according to the first audio data and the second audio data, the first state data and the second state data for identifying the state of the audio call between the first calling party and the second calling party.

In this embodiment of the present application, since the first audio data is voice data sent by the first calling party at the far end, and the second audio data is data collected by, for example, a microphone of the communication device of the second calling party at the local place. Under normal circumstances, the local site as the near end may have three call states: using the headset to speak, using the loudspeaker to play the far party's call and the near end is not speaking, and using the speaker to play the far party's call and the near end Duan is talking.

Therefore, in this embodiment of the present application, the state determination module 43 may determine the call state of the near-end based on the first audio data and the second audio data, that is, determine the first state data and the second state that identify the call state of the near-end data. For example, in this embodiment of the present application, the first determination unit 431 of the state determination module 43 may obtain the first state data by performing frequency band averaging on the correlation coefficients of the first audio signal and the second audio signal in respective frequency bands. For example, in this embodiment of the present application, the second determination unit 432 of the state determination module 43 may perform frequency band averaging on the ratio of the linear echo data and the second audio data separated by the linear filter processing in the respective frequency bands to obtain the second status data.

The suppression module 44 may be configured to determine a weighting factor related to the call state according to the first state data and the second state data, so as to perform weighted filtering processing on the linear output data to obtain third audio data sent to the first caller.

Therefore, the weighting factor related to the current call state of the second calling party can be determined according to the first state data and the second state data determined by the state determination module 43 , and the suppression module 44 can directly use the filtering module 41 and the linear output module 42 The output is further weighted and filtered according to the current call state, that is, the weight factor of the call state.

In other words, the embodiments of the present application can introduce the weighting factor into the echo suppression process in the prior art to consider the call state of the second caller, that is, the linear output can be more clearly determined according to the difference of the caller state of the second caller The echo residual components that may exist in the result, so that the linear output result of the solution in the prior art can be further suppressed. The mapping table of the gain adjustment scheme of the linear output, so that when the first state data and the second state data are determined, the pre-established mapping table can be directly inquired according to the determined first state data and the second state data to select the corresponding Adjust the scheme or gain adjustment factor to directly perform echo residual processing on the linear output result.

For example, in this embodiment of the present application, the suppression module 44 may include: a third determination unit 441 , a fourth determination unit 442 , and a filtering unit 443 .

The third determining unit 441 may be configured to determine a trade-off factor for controlling speech distortion according to the first state data and the second state data.

In this embodiment of the present application, the trade-off factor is determined by using the first state data obtained by the first determining unit 431 and the second state data obtained by the second determining unit 432 . In this embodiment of the present application, the following formula can be used to determine the trade-off factor μ.

Among them, Φ _YY and Φ _EE are covariance matrices determined using different frame signals. Therefore, in this embodiment of the present application, such a trade-off factor that can be used to determine echo residual components can be obtained through the first state data and the second state data.

The fourth determination unit 442 may be configured to determine the inter-frame Wiener filter coefficients according to the trade-off factor, the linear echo data and the linear output data.

After the third determining unit 441 obtains the trade-off factor, the fourth determining unit 442 may further use the trade-off factor determined by the third determining unit 441, the linear echo data obtained by the filtering module 41, and the linear output data obtained by the linear output module 42. to determine the inter-frame Wiener filter coefficients. For example, the following formula can be used to determine the inter-frame Wiener filter coefficients w(t,f).

where t is time, and f is the sound frequency, and e ₁ is the calculation parameter.

The filtering unit 443 may be configured to perform filtering processing on the linear output data according to the Wiener filter coefficients to obtain the first output audio data

In this embodiment of the present application, the filtering unit 443 may use the Wiener filter coefficients determined and obtained by the third determining unit 441 to perform filtering processing on the linear output data obtained by the linear output module 42 by using the Wiener filter, so as to obtain the first Output audio data. Therefore, in this embodiment of the present application, the trade-off factor can be introduced into the echo suppression processing in the prior art to consider the call state of the second call party, that is, according to the difference of the call state of the second call party, it can be more clearly defined The echo residual components that may exist in the linear output result are determined, so that further echo residual components can be suppressed for the linear output results of the solution in the prior art.

In addition, the audio processing apparatus in this embodiment of the present application may further include: a gain determination module 47 , a selection module 48 and a signal amplitude adjustment module 49 .

The gain determination module 47 can be used to determine the average gain of the frequency band of the first output audio data;

The selection module 48 can be used to select the signal reduction amplitude value corresponding to the average gain of the frequency band;

The signal amplitude adjustment module 49 may be configured to perform an operation of reducing the signal amplitude on the first output audio data according to the signal amplitude reduction value.

After the filtering unit 443 obtains the first output audio data, the gain determination module 47 may be used to further determine the average gain of the frequency band of the first output audio data obtained by the filtering unit 443, and the selection module 48 may be used to determine the frequency band obtained by the gain determination module 47 according to the The average gain is used to select the corresponding signal reduction amplitude value, so that the signal amplitude adjustment module 49 is used to use the selected signal reduction amplitude value to perform the operation of reducing the signal amplitude on the first output audio data obtained by the filtering unit 443, thereby realizing the first output audio data. A further suppression of echo residues contained in the output audio data.

In addition, the suppression module 44 may further include: a selection unit 444 and a signal amplitude adjustment unit 445 .

The selection unit 444 may be configured to select a signal reduction amplitude value corresponding to the first state data and the second state data.

The signal amplitude adjustment unit 445 may be configured to perform the signal amplitude reduction operation on the linear output audio data according to the signal reduction amplitude value.

In addition, after the first state data and the second state data have been obtained in this embodiment of the present application, the selection unit 444 can also directly identify the current call state of the near-end identified by the first state data and the second state data through the selection unit 444. The signal reduction amplitude value corresponding to the state data is inquired from, for example, a preset table, and the linear output audio obtained by the linear output module 42 is down signaled by the signal amplitude adjustment unit 445 using the signal reduction amplitude value selected by the step selection unit 444 Amplitude operation.

The voice detection module 45 may be configured to perform audio activity detection on the first audio data to determine whether the first audio data contains voice audio.

In the embodiment of the present application, since each module used for audio processing in the communication terminal consumes a large amount of power, the two parties usually do not talk all the time during the call. Use the voice detection module 45 to perform audio activity detection processing (VoiceActivity Detection, VAD) on the received first audio data of the far end, so as to activate the relevant audio processing module only when it is determined that the first audio data contains voice data, and At other times, the relevant audio processing modules are all in a standby or dormant state, and until the VAD processing in this step determines that the first audio data contains voice data, that is, when it is determined that the first communication party at the far end is speaking, Only send a wake-up signal to the relevant audio processing module or send a notification signal to the central processing unit or controller of the communication device, etc., so that the remote communication device can wake up the relevant audio processing module to respond to the first voice containing the remote voice. An audio data is processed.

The delay alignment module 46 may be configured to perform delay alignment processing on the first audio data according to the second audio data.

In the case that the voice detection module 45 determines that the received first audio data contains the audio data of the far end, the delay alignment module 46 can be used to align the received first audio data according to the second audio data collected by the near end communication device. For example, a delay aligning of audio data is performed by a delay estimation module, so as to perform delay adjustment on the received first audio. Under normal circumstances, since the local near-end plays the first audio data through a playback device such as a microphone of the communication device and collects the audio data propagated in the space through a collection module such as a speaker, that is, when the second audio data is obtained , the second audio data needs to propagate in the space where the near-end is located before it can be collected by the near-end communication device. Therefore, even if the near-end does not speak, the second audio data collected by the near-end communication device There is a time difference between the far-end speech component contained in the data and the first audio data, that is, the first audio data is played by the speaker of the near-end communication device, propagated through space, and then received by the microphone, which takes a period of time. The first audio data is not aligned on the timeline. Therefore, for example, the delay alignment module 46 of the near-end communication device can use the first audio data to align the second audio data collected by the near-end communication device on the timeline, so as to facilitate subsequent echoes Accuracy and efficiency of identification and comparison processing.

Therefore, according to the audio data processing apparatus of the embodiment of the present application, according to the present application, it is possible to determine the first audio call state for identifying the current audio call state according to the audio data sent by the first call party and the collected data of the second call party. state data and second state data; further, according to the weighting coefficient related to the current audio call state determined by the first state data and the second state data or the corresponding suppression scheme, the linear filtering of the microphone collection data is carried out. Sexual inhibition treatment. Therefore, weighted filtering can be performed based on the current call state or a corresponding suppression scheme can be used for processing, so that the echo residual suppression processing can be performed considering the component characteristics of the echo residual in different call states, which can improve the echo residual suppression effect and effectively improve the call quality. .

Embodiment 5

The internal functions and structures of the audio data processing apparatus are described above, and the apparatus can be implemented as an electronic device. FIG. 5 is a schematic structural diagram of an embodiment of an electronic device provided by the present application. As shown in FIG. 5 , the electronic device includes a memory 51 and a processor 52 .

The memory 51 is used to store programs. In addition to the above-described programs, the memory 51 may also be configured to store various other data to support operations on the electronic device. Examples of such data include instructions for any application or method operating on the electronic device, contact data, phonebook data, messages, pictures, videos, etc.

Memory 51 may be implemented by any type of volatile or non-volatile storage device or combination thereof, such as static random access memory (SRAM), electrically erasable programmable read only memory (EEPROM), erasable Programmable Read Only Memory (EPROM), Programmable Read Only Memory (PROM), Read Only Memory (ROM), Magnetic Memory, Flash Memory, Magnetic or Optical Disk.

The processor 52 is not limited to a central processing unit (CPU), but may also be a graphics processing unit (GPU), a field programmable gate array (FPGA), an embedded neural network processor (NPU), or an artificial intelligence (AI) chip and other processing chips. The processor 52, coupled with the memory 51, executes the program stored in the memory 51, and when the program runs, the audio data processing methods of the second and third embodiments above are executed.

Further, as shown in FIG. 5 , the electronic device may further include: a communication component 53 , a power supply component 54 , an audio component 55 , a display 56 and other components. Only some components are schematically shown in FIG. 5 , which does not mean that the electronic device only includes the components shown in FIG. 5 .

The communication component 53 is configured to facilitate wired or wireless communication between the electronic device and other devices. Electronic devices can access wireless networks based on communication standards, such as WiFi, 3G, 4G or 5G, or a combination thereof. In one exemplary embodiment, the communication component 53 receives broadcast signals or broadcast related information from an external broadcast management system via a broadcast channel. In an exemplary embodiment, the communication component 53 also includes a near field communication (NFC) module to facilitate short-range communication. For example, the NFC module may be implemented based on radio frequency identification (RFID) technology, infrared data association (IrDA) technology, ultra-wideband (UWB) technology, Bluetooth (BT) technology and other technologies.

The power supply assembly 54 provides power to various components of the electronic device. Power supply components 54 may include a power management system, one or more power supplies, and other components associated with generating, managing, and distributing power to electronic devices.

Audio component 55 is configured to output and/or input audio signals. For example, audio component 55 includes a microphone (MIC) that is configured to receive external audio signals when the electronic device is in operating modes, such as calling mode, recording mode, and voice recognition mode. The received audio signal may be further stored in the memory 51 or transmitted via the communication component 53 . In some embodiments, audio assembly 55 also includes a speaker for outputting audio signals.

Display 56 includes a screen, which may include a liquid crystal display (LCD) and a touch panel (TP). If the screen includes a touch panel, the screen may be implemented as a touch screen to receive input signals from a user. The touch panel includes one or more touch sensors to sense touch, swipe, and gestures on the touch panel. The touch sensor may not only sense the boundaries of a touch or swipe action, but also detect the duration and pressure associated with the touch or swipe action.

Those of ordinary skill in the art can understand that all or part of the steps of implementing the above method embodiments may be completed by program instructions related to hardware. The aforementioned program can be stored in a computer-readable storage medium. When the program is executed, the steps including the above method embodiments are executed; and the foregoing storage medium includes: ROM, RAM, magnetic disk or optical disk and other media that can store program codes.

Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention, but not to limit them; although the present invention has been described in detail with reference to the foregoing embodiments, those of ordinary skill in the art should understand that: The technical solutions described in the foregoing embodiments can still be modified, or some or all of the technical features thereof can be equivalently replaced; and these modifications or replacements do not make the essence of the corresponding technical solutions deviate from the technical solutions of the embodiments of the present invention. scope.

Claims (12)

1. A method for processing audio data, comprising: Perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data, wherein the first calling party and the second calling party are in the same During a call activity; determining linear output data according to the second audio data collected by the second calling party and the linear echo data; First state data and second state data for identifying the state of an audio call between the first caller and the second caller are determined according to the first audio data and the second audio data , wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the second state data is the linear echo data and the The average value of the ratio of the second audio data in each sub-band; According to the first state data and the second state data, a weighting factor related to the call state is determined, so as to perform weighted filtering processing on the linear output data to obtain a third call sent to the first caller. audio data. 2. The audio data processing method according to claim 1, wherein the first audio data is voice data sent by the first calling party to the second calling party in the calling activity, and the The second audio data is audio data collected by the second calling party when the first audio data is played. 3. The audio data processing method according to claim 1 or 2, wherein the weighting factor related to the call state is determined according to the first state data and the second state data, so as to determine the weighting factor related to the call state. The linear output data is subjected to weighted filtering processing to obtain third audio data sent to the first calling party, including: determining a trade-off factor for controlling speech distortion according to the first state data and the second state data; determining inter-frame Wiener filter coefficients according to the trade-off factor, the linear echo data, and the linear output data; The linear output data is filtered according to the Wiener filter coefficients to obtain first output audio data as the third audio data. 4. The audio data processing method according to claim 3, wherein the method further comprises: determining a frequency band average gain of the first output audio data; selecting a signal reduction amplitude value corresponding to the average gain of the frequency band; A signal amplitude reduction operation is performed on the first output audio data according to the signal reduction amplitude value. 5. The audio data processing method according to claim 1, wherein, in the said first state data and said second state data, determining a weighting factor related to the talking state, so that the linear Before performing weighted filtering processing on the output data to obtain third audio data sent to the first calling party, the method further includes: Audio activity detection is performed on the first audio data to determine whether the first audio data contains speech audio. 6 . The audio data processing method according to claim 1 , wherein, in said determining according to the first audio data and the second audio data, a method for identifying the connection between the first calling party and the second audio data is determined. 7 . Before the first state data and the second state data of the audio call state between the calling parties, the method further includes: Delay alignment processing is performed on the first audio data according to the second audio data. 7. An audio data processing method, comprising: Perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data, wherein the first calling party and the second calling party are in the same During a call activity; determining linear output data according to the second audio data collected by the second calling party and the linear echo data; First state data and second state data for identifying the state of an audio call between the first caller and the second caller are determined according to the first audio data and the second audio data , wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the first state data is the linear echo data and the The average value of the ratio of the second audio data in each sub-band; selecting, according to the first state data and the second state data, a signal reduction amplitude value corresponding to the first state data and the second state data; The signal amplitude reduction operation is performed on the linear output audio data according to the signal reduction amplitude value, so as to obtain audio data sent to the first calling party. 8. A calling method, comprising: receiving first audio data; playing the first audio data; performing audio capture processing to generate second audio data, wherein the second audio data includes at least audio data collected when the first audio data is played; performing linear filtering processing on the second audio data to obtain linear echo data; determining linear output data according to the second audio data and the linear echo data; According to the first audio data and the second audio data, first state data and second state data for identifying the audio call state are determined, wherein the first state data is the first audio data and the the average value of the correlation coefficient of the second audio data in each sub-band, and the first state data is the average value of the ratio of the linear echo data and the second audio data in each sub-band; According to the first state data and the second state data, determine a weighting factor related to the talking state, so as to perform weighted filtering processing on the linear output data to obtain third audio data; The third audio data is output to the calling party making the call. 9. An audio processing chip, comprising: an audio receiving module for receiving the first audio data; an audio output module for playing the first audio data; a sound pickup module, configured to perform audio collection processing to generate second audio data, wherein the second audio data at least includes audio data collected by the sound pickup module when playing the first audio data; a filtering module, configured to perform linear filtering processing on the second audio data to obtain linear echo data; a processing module, configured to determine linear output data according to the second audio data and the linear echo data, and determine first state data for identifying the audio call state according to the first audio data and the second audio data and second state data, wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in each sub-band, and the first state data is the linear an average value of the ratios of the echo data and the second audio data in each sub-band; and determining a weighting factor related to the call state according to the first state data and the second state data, Wherein, the filtering module is configured to perform weighted filtering processing on the linear output data to obtain third audio data, and The audio output module is used for outputting the third audio data to the calling party making the call. 10. An audio data processing device, comprising: A filtering module, configured to perform linear filtering processing on the first audio data sent by the first calling party and the second audio data collected by the second calling party to obtain linear echo data, wherein the first calling party and the The two call parties are in the same call activity; a linear output module, configured to determine linear output data according to the second audio data collected by the second calling party and the linear echo data; a state determination module for determining, according to the first audio data and the second audio data, a first state for identifying the state of an audio call between the first call party and the second call party data and second state data, wherein the first state data is the average value of the correlation coefficients of the first audio data and the second audio data in respective sub-bands, and the first state data is the the average value of the ratio of the linear echo data to the second audio data in each sub-band; A suppression module, configured to determine a weighting factor related to the call state according to the first state data and the second state data, so as to perform weighted filtering processing on the linear output data, and obtain a result that is sent to the first The third audio data of the calling party. 11. An electronic device comprising: memory for storing programs; A processor, configured to run the program stored in the memory, and execute the method according to any one of claims 1 to 8 when the program runs. 12. A computer-readable storage medium having stored thereon a computer program executable by a processor, wherein the program, when executed by the processor, implements the method of any one of claims 1 to 8.

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