Method and system for improving audio quality of remote user voice communication

Abstract

The invention provides a method and system for improving voice communication audio quality of a remote user, and belongs to the technical field of audio processing, and the method comprises the steps: determining target factors which influence the quality of a first audio signal, and the target factors comprise environment noise and network bandwidth; in response to the fact that the target factor is environment noise, the environment noise in the first audio signal is suppressed based on a second audio signal, a target audio signal is obtained, and the second audio signal is a signal which is equal to the environment noise in audio amplitude and opposite to the environment noise in phase; and in response to the target factor being the network bandwidth, adjusting the first audio signal based on the adjusted transmission rate to obtain a target audio signal, the adjusted transmission rate being obtained based on the network bandwidth. According to the method and the system for improving the voice communication audio quality of the remote user, the voice communication audio quality of the remote user can be improved.

Description

Technical Field

[0001] The present disclosure belongs to the technical field of audio processing, and more specifically, to a method and system for improving the audio quality of voice communication of a remote user. Background Art

[0002] Remote user voice communication has become an indispensable part of modern society. However, in actual applications, remote voice communication is often restricted by environmental noise and network bandwidth, resulting in reduced audio quality and serious impact on user experience. Environmental noise, such as background noise and wind noise, will be mixed into the audio signal, reducing the clarity of the voice; and network bandwidth limitations may cause problems such as freezes, delays, and distortion during audio transmission. These problems not only affect the fluency of voice communication, but also limit the efficiency and effectiveness of remote communication. Summary of the invention

[0003] The object of the present disclosure is to provide a method and system for improving the audio quality of voice communication of a remote user, so as to improve the audio quality of the remote voice communication.

[0004] According to a first aspect of an embodiment of the present disclosure, there is provided a method for improving the audio quality of voice communication of a remote user, comprising: Determining a target factor affecting the quality of the first audio signal, the target factor comprising environmental noise and network bandwidth; In response to the target factor being environmental noise, suppressing the environmental noise in the first audio signal based on a second audio signal to obtain a target audio signal, wherein the second audio signal is a signal having an audio amplitude equal to that of the environmental noise and an opposite phase; In response to the target factor being the network bandwidth, the first audio signal is adjusted based on an adjusted transmission rate to obtain a target audio signal, wherein the adjusted transmission rate is obtained based on the network bandwidth.

[0005] A second aspect of the embodiments of the present disclosure provides a system for improving the audio quality of voice communication of a remote user, including: A factor determination module, configured to determine a target factor affecting the quality of the first audio signal, wherein the target factor includes environmental noise and network bandwidth; a first quality adjustment module, configured to suppress the environmental noise in the first audio signal based on a second audio signal to obtain a target audio signal in response to the target factor being environmental noise, wherein the second audio signal is a signal having an audio amplitude equal to that of the environmental noise and an opposite phase; The second quality adjustment module is used for adjusting the first audio signal based on adjusting the transmission rate to obtain a target audio signal in response to the target factor being the network bandwidth, wherein the adjusted transmission rate is obtained based on the network bandwidth.

[0006] According to a third aspect of an embodiment of the present disclosure, an electronic device is provided, comprising a memory, a processor, and a computer program stored in the memory and running on the processor, wherein the processor implements the steps of the above-mentioned method for improving the audio quality of voice communication of a remote user when executing the computer program.

[0007] According to a fourth aspect of an embodiment of the present disclosure, a computer-readable storage medium is provided, wherein the computer-readable storage medium stores a computer program, and when the computer program is executed by a processor, the steps of the above-mentioned method for improving the audio quality of voice communication of a remote user are implemented.

[0008] The method and system for improving the audio quality of voice communication of remote users provided by the embodiments of the present disclosure have the following beneficial effects: the embodiments of the present disclosure accurately identify and suppress environmental noise, and use the principle of destructive interference to generate an audio signal opposite to the environmental noise for superposition, thereby effectively eliminating background noise and improving the clarity and signal-to-noise ratio of the audio. At the same time, the embodiments of the present disclosure can also dynamically adjust the transmission rate of the audio signal to adapt to the real-time network bandwidth conditions, ensuring that the audio can still be smooth when the bandwidth is limited, and improving the audio quality when the network conditions improve, thereby realizing intelligent optimization of the audio quality. BRIEF DESCRIPTION OF THE DRAWINGS

[0009] In order to more clearly illustrate the technical solutions in the embodiments of the present disclosure, the drawings required for use in the embodiments or the description of the prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present disclosure. For ordinary technicians in this field, other drawings can be obtained based on these drawings without creative work.

[0010] Figure 1 A flowchart of a method for improving the audio quality of voice communication of a remote user provided by an embodiment of the present disclosure; Figure 2 A structural block diagram of a system for improving the audio quality of voice communication of a remote user provided by an embodiment of the present disclosure; Figure 3 A schematic block diagram of an electronic device provided by an embodiment of the present disclosure. DETAILED DESCRIPTION

[0011] In the following description, specific details such as specific system structures and technologies are provided for the purpose of illustration rather than limitation, so as to provide a thorough understanding of the embodiments of the present disclosure. However, it should be clear to those skilled in the art that the present disclosure may be implemented in other embodiments without these specific details. In other cases, detailed descriptions of well-known systems, devices, circuits, and methods are omitted to avoid obstructing the description of the present disclosure with unnecessary details.

[0012] In order to make the purpose, technical solutions and advantages of the present disclosure more clear, specific embodiments will be described below in conjunction with the accompanying drawings.

[0013] Please refer to Figure 1 , Figure 1 A flow chart of a method for improving the audio quality of voice communication of a remote user provided by an embodiment of the present disclosure, the method comprising: S101: Determine target factors affecting the quality of a first audio signal, where the target factors include environmental noise and network bandwidth.

[0014] In this embodiment, the "first audio signal" refers to the audio signal initially generated in the voice communication of the remote user. This audio signal may come from the speaker's voice input device, such as a microphone. During the transmission process, the quality of the "first audio signal" may be affected by various factors, such as environmental noise interference that reduces its clarity, or problems such as freezes and distortions due to network bandwidth limitations.

[0015] Environmental noise refers to all unwanted sounds or interferences other than the target sound during the audio signal acquisition process. These noises can come from various sound sources in the surrounding environment, such as traffic noise, wind noise, crowd noise, machine operation noise, etc. For the detection of environmental noise, the surrounding sound signals can be collected through audio input devices such as microphones and analyzed and processed.

[0016] The network bandwidth determines the speed and efficiency of data transmission. In the case of limited bandwidth, the first audio signal may be damaged due to compression or delay. For monitoring of network bandwidth, the current network data transmission rate, delay and other parameter information can be obtained in real time through the module related to the network connection.

[0017] In this embodiment, the target factor that currently has a major impact on the quality of the first audio signal can be determined through the analysis results of the environmental noise and the network bandwidth. If both the environmental noise and the network bandwidth have a significant impact on the audio quality, both can be determined as target factors at the same time, so that corresponding measures can be taken later to improve the audio quality.

[0018] S102: In response to the target factor being environmental noise, suppress the environmental noise in the first audio signal based on the second audio signal to obtain a target audio signal, wherein the second audio signal is a signal having an audio amplitude equal to that of the environmental noise and an opposite phase.

[0019] In remote voice communication, environmental noise may be mixed into the first audio signal, reducing the audio quality. In this embodiment, environmental noise may be detected first, and surrounding sound may be collected through an audio input device such as a microphone, and spectrum analysis may be performed to determine characteristics such as frequency and amplitude of the noise.

[0020] Sound waves are a form of fluctuation, with properties such as amplitude, frequency and phase. When two sound waves with the same frequency, equal amplitude but opposite phase meet, they will cancel each other out at certain points in space, forming a "quiet zone". In audio processing, this principle can be used to eliminate unwanted noise components.

[0021] In this embodiment, the ambient noise in the audio signal is accurately identified and separated, and the ambient noise can be evaluated based on a statistical model or machine learning technology. After determining that the target factor is ambient noise, this embodiment accurately calculates an audio signal with the same amplitude as the ambient noise but a phase difference of 180 degrees based on the detected ambient noise characteristics. Ensure that the generated second audio signal corresponds to the ambient noise at each frequency point. The generated second audio signal is superimposed on the first audio signal. Since the second audio signal has the same amplitude as the ambient noise and the opposite phase, according to the interference principle of sound waves, when the two are superimposed, the ambient noise part will be offset.

[0022] For example, assuming that the first audio signal contains the speaker's voice and environmental noise, the amplitude of the environmental noise at a certain frequency point is A, and the phase is At this time, the amplitude of the generated second audio signal at this frequency point is also A, but the phase is When the two are superimposed, the amplitude of the ambient noise at this frequency point becomes 0, thereby achieving the effect of suppressing ambient noise.

[0023] In this embodiment, the method of suppressing environmental noise in the first audio signal based on the second audio signal realizes effective suppression of noise components in the audio signal by utilizing the physical principle of destructive interference and advanced noise estimation algorithms, thereby improving the audio clarity and signal-to-noise ratio, thereby improving the audio quality of the remote user's voice communication.

[0024] S103: In response to the target factor being the network bandwidth, adjusting the first audio signal based on adjusting the transmission rate to obtain a target audio signal, wherein the adjusted transmission rate is obtained based on the network bandwidth.

[0025] In this embodiment, in remote voice communication, the network bandwidth determines the data transmission capacity. If the network bandwidth is limited, it may cause problems such as freezing and distortion of the audio signal during transmission. For example, low bandwidth may not be able to transmit high-bitrate audio data in time, resulting in audio interruption or quality degradation.

[0026] In this embodiment, the current network bandwidth status is monitored in real time. This can be achieved by measuring parameters such as available bandwidth, delay, packet loss rate, etc. in the network. The monitoring of network bandwidth is performed dynamically because network conditions may change with time, number of users, network load, etc.

[0027] When the target factor is determined to be the network bandwidth in this embodiment, the first audio signal can be adjusted based on adjusting the transmission rate. If the network bandwidth is low, this embodiment can reduce the transmission rate of the audio signal. This can be achieved by reducing the bit rate of the audio, for example, adjusting from a high bit rate audio format to a low bit rate format. This can reduce the amount of data transmitted to adapt to the limited network bandwidth and ensure the smoothness of the audio. If the network bandwidth condition improves, this embodiment can appropriately increase the transmission rate to obtain a higher quality audio signal. For example, the audio bit rate is increased to a higher level to increase the details and fidelity of the audio.

[0028] The first audio signal is processed accordingly according to the adjusted transmission rate. For example, if the transmission rate is reduced, the audio signal may be re-encoded to remove some unnecessary high-frequency components or reduce the sampling rate of the audio.

[0029] It can be concluded from the above that this embodiment can effectively eliminate background noise and improve audio clarity and signal-to-noise ratio by accurately identifying and suppressing environmental noise and using the principle of destructive interference to generate an audio signal opposite to the environmental noise for superposition. At the same time, this embodiment can also dynamically adjust the transmission rate of the audio signal to adapt to the real-time network bandwidth conditions, ensuring that the audio can still be smooth when the bandwidth is limited, and improving the audio quality when the network conditions improve, thereby realizing intelligent optimization of audio quality.

[0030] In one embodiment of the present disclosure, it also includes: A first equalizer of a first device is adjusted based on a first audio feature of a first audio signal so that the first equalizer of the first device processes the first audio signal, and the first device is a device that receives the first audio signal.

[0031] In this embodiment, the first equalizer is a device for adjusting the audio spectrum, which can enhance or weaken the sound within a specific frequency range. By adjusting the relative size of different frequency components in the audio signal, the equalizer can change the timbre, sound quality and volume balance of the audio, thereby meeting the user's personalized needs for audio effects.

[0032] The first device is a device that receives the first audio signal, such as a mobile phone, a smart wearable device, etc.

[0033] In this embodiment, the first audio feature can be extracted from the first audio signal. The first audio feature can include frequency distribution, amplitude distribution, timbre characteristics, etc. For example, the energy distribution of the audio signal at different frequencies can be determined by spectrum analysis; the strength change of the audio signal can be understood by analyzing the amplitude.

[0034] Based on the extracted first audio feature, this embodiment adjusts the first equalizer of the first device. If the energy of the first audio signal at certain frequencies is too high or too low, the first equalizer can increase or decrease the volume of these frequencies accordingly. The adjusted first equalizer can process the first audio signal. The first equalizer can amplify, attenuate or keep unchanged the audio components of different frequencies according to the set parameters. In this way, the first audio signal can be played on the first device to better meet the user's auditory needs and improve the audio quality.

[0035] Exemplarily, Xiao Li uses a smart phone as a first device to receive a first audio signal of remote voice communication. In an important remote meeting, the meeting place is located in a noisy open office area. First, the smart phone automatically extracts the first audio feature from the received first audio signal. Through spectrum analysis, it is found that the energy of environmental noise in the frequency range of 200Hz-500Hz is too high, while the high-frequency part of human voice in the frequency range of 2kHz-4kHz is relatively weak. At the same time, amplitude analysis shows that the overall strength of the audio signal varies greatly and the stability is insufficient. Based on these audio features, the first equalizer on Xiao Li's mobile phone is adjusted. For the part with too high mid-low frequency, the equalizer appropriately reduces the volume in the frequency range of 200Hz-500Hz. At the same time, in order to enhance the clarity of human voice, the equalizer enhances the volume in the frequency range of 2kHz-4kHz. The adjusted first equalizer processes the first audio signal. In the processed audio signal, the environmental noise is effectively suppressed, the human voice is more prominent, and the high-frequency part is clearer. Moreover, the volume of different frequency components is more balanced, and the overall audio quality is improved.

[0036] From the above, it can be concluded that this embodiment can achieve personalized optimization processing of the first audio signal by dynamically adjusting the first equalizer of the first device according to the first audio feature of the first audio signal, thereby enhancing the audio listening experience, making the audio output more in line with the user's auditory preferences and scene requirements, and improving the applicability of the audio device and user experience.

[0037] In one embodiment of the present disclosure, adjusting a first equalizer of a first device based on a first audio feature of a first audio signal so that the first equalizer of the first device processes the first audio signal includes: adjusting a first equalizer of the first device based on a correlation of the first audio feature with the first parameter; The first parameters are parameters of the first equalizer, and the first parameters include frequency range and gain.

[0038] In one embodiment of the present disclosure, a deviation between a first audio feature and a reference audio feature is calculated; If the deviation between the first audio feature and the reference audio feature is greater than a first preset deviation, the correlation between the first audio feature and the frequency range and the gain is calculated.

[0039] Specifically, the frequency range includes multiple frequency ranges, and the frequency range may include a low frequency range, a medium frequency range, and a high frequency range; In response to the first parameter being a frequency range, calculating energy deviations between the first audio signal and the reference audio signal in each frequency range, and calculating correlations between the frequency feature and the frequency range based on the energy deviations between the first audio signal and the reference audio signal in each frequency range; In response to the first parameter being a gain, an amplitude deviation between the first audio signal and a reference audio signal is calculated, and a correlation between the frequency feature and the frequency range is calculated based on the amplitude deviation between the first audio signal and the reference audio signal.

[0040] In this embodiment, first, the first audio feature is extracted and the first parameter is determined: Extract the first audio feature from the first audio signal, for example, determine the energy distribution of the audio at different frequencies through spectrum analysis, and calculate the amplitude-related parameters of the audio signal. The frequency range is divided into multiple specific frequency intervals including a low frequency range, a mid-frequency range, and a high frequency range, and the gain is used to adjust the volume of different frequency intervals.

[0041] Secondly, calculate the deviation between the first audio feature and the reference audio feature: Determine the reference audio feature, which can be a pre-set ideal audio feature standard, or a feature extracted from a high-quality audio sample. Calculate the deviations between the first audio feature and the reference audio feature in frequency and amplitude respectively. For the frequency distribution feature, compare the energy differences between the first audio signal and the reference audio signal in each frequency range. For example, calculate the energy deviation in the low frequency, medium frequency, and high frequency ranges. For the amplitude distribution feature, calculate the amplitude deviation between the first audio signal and the reference audio signal, such as the difference in amplitude mean, amplitude variance, etc.

[0042] Finally, determine whether the deviation is greater than the first preset deviation: The calculated deviation is compared with the first preset deviation. If the deviation is greater than the first preset deviation, it means that the first audio signal differs greatly from the reference audio feature and needs to be adjusted to improve the audio quality.

[0043] In this embodiment, when the first parameter is a frequency range, the energy deviation between the first audio signal and the reference audio signal in each frequency range is calculated. For example, the energy of the first audio signal and the reference audio signal in the low frequency, medium frequency, and high frequency ranges are calculated respectively. Then, the energy difference in each frequency range, i.e., the energy deviation, is calculated.

[0044] The correlation between the frequency feature and the frequency range is calculated based on the energy deviation. If the energy deviation in a certain frequency range is large, it means that the first audio signal is significantly different from the reference audio signal in this frequency range. For example, if the energy of the first audio signal in the low-frequency range is much higher than that of the reference audio signal, it means that the low-frequency part of the first audio signal is relatively strong. Based on this energy deviation, the adjustment direction of the frequency range can be determined. If the low-frequency energy deviation is positive and large, it may be necessary to reduce the gain of the first equalizer in the low-frequency range to reduce the volume of the low-frequency part so that the first audio signal is closer to the reference audio feature.

[0045] When the first parameter is gain, the amplitude deviation between the first audio signal and the reference audio signal is calculated. The amplitude mean and amplitude variance of the first audio signal and the reference audio signal are compared to obtain the amplitude deviation.

[0046] The correlation between the frequency feature and the gain is calculated based on the amplitude deviation. If the amplitude mean of the first audio signal is greater than that of the reference audio signal, it means that the overall volume of the first audio signal is large, and the gain needs to be reduced to reduce the volume; conversely, if the amplitude mean is small, the gain needs to be increased to increase the volume. Therefore, the gain can be adjusted according to the amplitude deviation to make the amplitude distribution of the first audio signal closer to the reference audio feature.

[0047] It can be concluded from the above that this embodiment can improve the quality of the audio signal by accurately analyzing the deviations between the first audio signal and the reference audio feature in terms of frequency and amplitude, and intelligently adjusting the frequency range and gain parameters of the first equalizer of the first device based on these deviations, so as to make it closer to the ideal or high-quality audio standard. This not only improves the efficiency of audio processing, but also ensures the accuracy and consistency of audio output, bringing users a better listening experience.

[0048] In one embodiment of the present disclosure, in response to the target factor being environmental noise, suppressing the environmental noise in the first audio signal based on the second audio signal to obtain the target audio signal includes: If the audio amplitude of the ambient noise is less than the first preset value, suppressing the ambient noise based on the second audio signal to obtain a target audio signal; If the audio amplitude of the ambient noise is greater than or equal to the first preset value, the ambient noise is suppressed based on the second equalizer of the first device and the second audio signal to obtain a target audio signal.

[0049] In this embodiment, the audio amplitude of the ambient noise is continuously monitored. The audio amplitude of the ambient noise can be obtained by analyzing the collected surrounding sound signals, reflecting the intensity of the ambient noise. The audio amplitude of the ambient noise is compared with the first preset value. The first preset value in this embodiment can be a signal generated by the first device that is equal to the maximum audio amplitude of the ambient noise and has an opposite phase. This means that the first preset value represents the upper limit of the ability of the first device to completely offset the ambient noise under ideal conditions. By setting such a value, ambient noise of different intensities can be clearly distinguished.

[0050] When the audio amplitude of the ambient noise is less than the first preset value, it indicates that the intensity of the ambient noise is relatively weak. In this case, the present embodiment directly suppresses the ambient noise based on the second audio signal to obtain the target audio signal. The second audio signal is a signal with an audio amplitude equal to that of the ambient noise and an opposite phase. By superimposing the second audio signal with the first audio signal, the ambient noise is offset, thereby improving the audio quality.

[0051] When the audio amplitude of the ambient noise is greater than or equal to the first preset value, it indicates that the ambient noise intensity is strong. At this time, the present embodiment first uses the second equalizer of the first device to weaken the ambient noise. The second equalizer can adjust the volume of different frequency ranges in a targeted manner according to the frequency characteristics of the ambient noise. For example, if the ambient noise is mainly concentrated in the low frequency band, the second equalizer can reduce the volume of the low frequency part, thereby reducing the overall intensity of the ambient noise. Through the adjustment of the second equalizer, the ambient noise is preliminarily weakened to a certain extent. Then, the second audio signal is used to further suppress the ambient noise processed by the equalizer. Since the ambient noise has been weakened by the equalizer, the second audio signal can play a more effective role at this time and offset the remaining ambient noise. Through the synergistic effect of the second audio signal and the second equalizer, strong ambient noise can be more effectively suppressed, and the audio quality can be significantly improved.

[0052] For example, Xiao Wang is a reporter who is conducting a remote interview at a busy market. He uses a professional recording device as the first device to receive the first audio signal. This device has powerful audio processing capabilities. The device continuously monitors the ambient noise, collects and analyzes the ambient sound signals through the built-in high-sensitivity microphone. After analysis, it is found that the audio amplitude of the current ambient noise is 45dB. At the same time, the first preset value determined by the device according to its own performance is 50dB. Since the audio amplitude of the ambient noise of 45dB is less than the first preset value of 50dB, the system determines that the intensity of the ambient noise is relatively weak. The device directly generates a second audio signal with the same audio amplitude and opposite phase as the ambient noise. This second audio signal is superimposed on the first audio signal, and the ambient noise is quickly offset. During Xiao Wang's remote interview, the other party can hear his voice clearly, and the audio quality is guaranteed.

[0053] Xiao Wang moved to a construction site under construction, and the ambient noise increased dramatically. The device monitored the ambient noise again and analyzed that the audio amplitude of the ambient noise at this time was 60dB, which was greater than the first preset value of 50dB. The device first used the second equalizer to weaken the ambient noise. After analyzing the frequency characteristics of the ambient noise, it was found that the noise was mainly concentrated in the medium and low frequency bands. Therefore, the second equalizer weakened the volume of the medium and low frequency parts and reduced the volume of the frequency range of 200Hz-800Hz by 8dB. After adjustment, the overall intensity of the ambient noise decreased. Then, the device generates a second audio signal with the same audio amplitude and opposite phase as the ambient noise processed by the second equalizer. This second audio signal is superimposed on the first audio signal to further offset the remaining ambient noise. Through the synergy of the second audio signal and the second equalizer, the strong ambient noise is effectively suppressed.

[0054] In one embodiment of the present disclosure, suppressing environmental noise based on a second equalizer of a first device and a second audio signal to obtain a target audio signal includes: adjusting the second equalizer based on the environmental noise characteristics of the environmental noise so that the second equalizer processes the environmental noise to obtain the target environmental noise; The target environmental noise in the first audio signal is suppressed based on the second audio signal to obtain a target audio signal.

[0055] In this embodiment, the environmental noise is deeply analyzed to extract the environmental noise features, which may include information on the frequency distribution, amplitude change, duration, periodicity, etc. of the noise.

[0056] For example, spectrum analysis can be used to determine the energy distribution of environmental noise at different frequencies and understand the frequency ranges in which the noise is mainly concentrated.

[0057] By analyzing the amplitude change, we can know the intensity fluctuation of the noise. The second equalizer is adjusted based on these environmental noise characteristics. If the environmental noise is mainly concentrated in the low frequency band, the gain of the low frequency part of the second equalizer can be appropriately reduced to reduce the impact of the low frequency noise. If the noise has a prominent peak at a certain frequency, the frequency can also be attenuated in a targeted manner. By adjusting the different parameters of the second equalizer, it can be targeted to deal with specific environmental noise, thereby reducing the intensity of the environmental noise.

[0058] The adjusted second equalizer starts to process the environmental noise. The second equalizer can amplify, attenuate or keep unchanged the different frequency components in the environmental noise according to the set parameters. Then, this embodiment generates a second audio signal with the same audio amplitude and opposite phase as the target environmental noise. This second audio signal is specially generated for the target environmental noise after being processed by the second equalizer. The second audio signal is superimposed on the first audio signal. Since the second audio signal is in opposite phase to the target environmental noise, the target environmental noise is further offset when the two are superimposed. Finally, the target audio signal is obtained. The environmental noise in this target audio signal is greatly reduced, and the audio quality is significantly improved, which is more in line with the user's auditory needs.

[0059] In one embodiment of the present disclosure, in response to the target factor being the network bandwidth, adjusting the first audio signal based on adjusting the transmission rate to obtain the target audio signal includes: If the network bandwidth is greater than a second preset value, adjusting the first audio signal based on the first transmission rate to obtain a target audio signal; If the network bandwidth is less than or equal to a second preset value, adjusting the first audio signal based on the second transmission rate to obtain a target audio signal; The first transmission rate is greater than the second transmission rate.

[0060] In this embodiment, a second preset value is introduced as a criterion for judging the network status. The second preset value can be set according to the requirements of audio transmission and the experience value under different network environments. The currently monitored network bandwidth is compared with the second preset value. If the network bandwidth is greater than the second preset value, it means that the network condition is relatively good and there is enough bandwidth to support high-quality audio transmission. If the network bandwidth is less than or equal to the second preset value, it indicates that the network condition may be relatively tense and a more conservative transmission strategy needs to be adopted.

[0061] When the network bandwidth is greater than the second preset value, this embodiment selects to adjust the first audio signal based on the first transmission rate. Since the network bandwidth is relatively abundant at this time, the first transmission rate is relatively high.

[0062] High transmission rates can support the transmission of richer audio information. For example, a higher audio encoding bit rate can be used to retain more audio details and improve audio quality and fidelity. Or a more complex audio encoding algorithm can be used to obtain better audio effects. At high transmission rates, audio signals can be transmitted quickly and stably in the network without freezing or packet loss, thus providing users with a high-quality audio experience.

[0063] When the network bandwidth is less than or equal to the second preset value, the second transmission rate can be used in this embodiment to adjust the first audio signal. Due to the limited network bandwidth, the second transmission rate is relatively low. In order to adapt to the limited network bandwidth, the audio encoding bit rate can be reduced to reduce the size of the audio file and reduce the demand for network bandwidth. Alternatively, a simpler audio encoding algorithm can be selected, which can ensure the stable transmission of the audio signal in the network, although it will sacrifice the audio quality to a certain extent.

[0064] In this embodiment, the first transmission rate is greater than the second transmission rate, which is set according to different network conditions. When the network condition is good, the high transmission rate can give full play to the advantages of the network and provide high-quality audio; when the network condition is poor, the low transmission rate can ensure the basic transmission of audio and avoid serious freezes and interruptions.

[0065] In one embodiment of the present disclosure, if the network bandwidth is less than or equal to the second preset value, the method for improving the audio quality of the remote user voice communication further includes: Calculate the rate of change of network bandwidth per unit time; If the change rate of the network bandwidth is greater than the first change rate, adjusting the sampling rate of the first device for the first audio signal based on the first step length; If the change rate of the network bandwidth is less than or equal to the second change rate, adjusting the sampling rate of the first device for the first audio signal based on the second step size; The first change rate is greater than the second change rate.

[0066] In this embodiment, when the network bandwidth is less than or equal to the second preset value, it indicates that the network condition is poor and the audio signal may not be transmitted with high quality. In this case, further measures need to be taken to adjust the audio transmission to improve the audio quality.

[0067] In this embodiment, the rate of change of the network bandwidth per unit time is calculated. This can be obtained by measuring the network bandwidth multiple times within a period of time and calculating the ratio of its change amplitude to time. The rate of change of the network bandwidth reflects the dynamic changes of the network status. If the rate of change is large, it means that the network status is unstable and may fluctuate greatly in a short period of time; if the rate of change is small, it means that the network status is relatively stable.

[0068] When the rate of change of the network bandwidth is greater than the first rate of change, it means that the network conditions change dramatically and may be experiencing large fluctuations. In this case, the present embodiment adjusts the sampling rate of the first device for the first audio signal based on the first step length. The first step length is usually large so as to quickly adapt to changes in network conditions. Lowering the sampling rate can reduce the amount of audio data, thereby reducing the demand for network bandwidth. For example, if the original sampling rate is 44.1kHz, it may be reduced to 22.05kHz after adjustment based on the first step length. In this way, when the network conditions are unstable and the bandwidth is limited, the audio signal can continue to be transmitted. Although the audio quality will be reduced, serious freezes or interruptions can be avoided.

[0069] When the rate of change of the network bandwidth is less than or equal to the second rate of change, it means that the network condition is relatively stable and the change is small. At this time, this embodiment adjusts the sampling rate of the first device for the first audio signal based on the second step size. The second step size is usually small because the network condition is relatively stable and does not require a large adjustment. The sampling rate can be appropriately lowered or increased according to the specific situation of the network bandwidth. If the network bandwidth improves slightly, the sampling rate can be slightly increased to improve the audio quality; if the network bandwidth is still tight, the sampling rate can continue to be lowered to ensure the stability of the transmission.

[0070] It can be concluded from the above that, on the one hand, when the network condition is poor, this embodiment can take timely measures to adjust the audio transmission, ensure that the audio signal can continue to be transmitted under the limited network bandwidth, avoid serious freezing or interruption, and ensure the basic smoothness of the remote user's voice communication. On the other hand, the sampling rate is finely adjusted according to the dynamic changes of the network condition, and the network condition changes drastically. When the change is fast, the sampling rate is quickly adapted with a larger step size, and when the network is relatively stable, the network is fine-tuned with a smaller step size. When the network is unstable and the bandwidth is limited, the amount of audio data can be reduced to maintain transmission, and when the network condition is slightly improved, the sampling rate can be appropriately increased to improve the audio quality, thereby maximizing the audio quality of the remote user's voice communication under different network conditions.

[0071] Corresponding to the method for improving the audio quality of voice communication of a remote user in the above embodiment, Figure 2 This is a block diagram of a system for improving the audio quality of voice communication of a remote user provided by an embodiment of the present disclosure. For ease of explanation, only the parts related to the embodiment of the present disclosure are shown. Figure 2 The system 20 for improving the audio quality of voice communication of a remote user includes: a factor determination module 21, a first quality adjustment module 22, and a second quality adjustment module 23.

[0072] The factor determination module 21 is used to determine the target factors affecting the quality of the first audio signal, the target factors including environmental noise and network bandwidth; A first quality adjustment module 22 is used for suppressing the environmental noise in the first audio signal based on the second audio signal in response to the target factor being environmental noise, so as to obtain a target audio signal, wherein the second audio signal is a signal having an audio amplitude equal to that of the environmental noise and an opposite phase; The second quality adjustment module 23 is used for adjusting the first audio signal based on adjusting the transmission rate to obtain a target audio signal in response to the target factor being the network bandwidth, wherein the adjusted transmission rate is obtained based on the network bandwidth.

[0073] In one embodiment of the present disclosure, the system 20 for improving the audio quality of voice communication of a remote user further includes: a third quality adjustment module; the third quality adjustment module is specifically used for: A first equalizer of a first device is adjusted based on a first audio feature of a first audio signal so that the first equalizer of the first device processes the first audio signal, and the first device is a device that receives the first audio signal.

[0074] In one embodiment of the present disclosure, the third quality adjustment module is further configured to: adjusting a first equalizer of the first device based on a correlation of the first audio feature with the first parameter; The first parameters are parameters of the first equalizer, and the first parameters include frequency range and gain.

[0075] In one embodiment of the present disclosure, the first quality adjustment module 22 is specifically used to: If the audio amplitude of the ambient noise is less than the first preset value, suppressing the ambient noise based on the second audio signal to obtain a target audio signal; If the audio amplitude of the ambient noise is greater than or equal to the first preset value, the ambient noise is suppressed based on the second equalizer of the first device and the second audio signal to obtain a target audio signal.

[0076] In one embodiment of the present disclosure, the first quality adjustment module 22 is further configured to: adjusting the second equalizer based on the environmental noise characteristics of the environmental noise so that the second equalizer processes the environmental noise to obtain the target environmental noise; The target environmental noise in the first audio signal is suppressed based on the second audio signal to obtain a target audio signal.

[0077] In one embodiment of the present disclosure, the second quality adjustment module 23 is specifically used to: If the network bandwidth is greater than a second preset value, adjusting the first audio signal based on the first transmission rate to obtain a target audio signal; If the network bandwidth is less than or equal to a second preset value, adjusting the first audio signal based on the second transmission rate to obtain a target audio signal; The first transmission rate is greater than the second transmission rate.

[0078] In one embodiment of the present disclosure, the second quality adjustment module 23 is further configured to: Calculate the rate of change of network bandwidth per unit time; If the change rate of the network bandwidth is greater than the first change rate, adjusting the sampling rate of the first device for the first audio signal based on the first step length; If the change rate of the network bandwidth is less than or equal to the second change rate, adjusting the sampling rate of the first device for the first audio signal based on the second step size; The first change rate is greater than the second change rate.

[0079] See also Figure 3 , Figure 3 A schematic block diagram of an electronic device provided by an embodiment of the present disclosure. Figure 3 The electronic device 300 in the embodiment shown may include: one or more processors 301, one or more input devices 302, one or more output devices 303 and one or more memories 304. The processors 301, input devices 302, output devices 303 and memories 304 communicate with each other via a communication bus 305. The memory 304 is used to store computer programs, which include program instructions. The processor 301 is used to execute the program instructions stored in the memory 304. The processor 301 is configured to call the program instructions to execute the functions of the modules in the above-mentioned system embodiments, such as Figure 2 The functions of modules 21 to 23 are shown.

[0080] It should be understood that in the embodiment of the present disclosure, the processor 301 may be a central processing unit (CPU), and the processor may also be other general-purpose processors, digital signal processors (DSP), application-specific integrated circuits (ASIC), field-programmable gate arrays (FPGA) or other programmable logic devices, discrete gate or transistor logic devices, discrete hardware components, etc. A general-purpose processor may be a microprocessor or the processor may also be any conventional processor, etc.

[0081] The input device 302 may include a touch panel, a fingerprint collection sensor (for collecting the user's fingerprint information and fingerprint direction information), a microphone, etc., and the output device 303 may include a display (LCD, etc.), a speaker, etc.

[0082] The memory 304 may include a read-only memory and a random access memory, and provide instructions and data to the processor 301. A portion of the memory 304 may also include a non-volatile random access memory. For example, the memory 304 may also store information about the device type.

[0083] In a specific implementation, the processor 301, input device 302, and output device 303 described in the embodiments of the present disclosure can execute the implementation methods described in the first and second embodiments of the method for improving the audio quality of remote user voice communication provided in the embodiments of the present disclosure, and can also execute the implementation methods of the electronic device described in the embodiments of the present disclosure, which will not be repeated here.

[0084] In another embodiment of the present disclosure, a computer-readable storage medium is provided, wherein the computer-readable storage medium stores a computer program, wherein the computer program includes program instructions, and when the program instructions are executed by the processor, all or part of the processes in the above-mentioned embodiment method are implemented, and the computer program can also be completed by instructing the relevant hardware through the computer program. The computer program can be stored in a computer-readable storage medium, and when the computer program is executed by the processor, the steps of each of the above-mentioned method embodiments can be implemented. Among them, the computer program includes computer program code, and the computer program code can be in source code form, object code form, executable file or some intermediate form, etc. The computer-readable medium can include: any entity or system capable of carrying computer program code, recording medium, U disk, mobile hard disk, magnetic disk, optical disk, computer memory, read-only memory (ROM, Read-Only Memory), random access memory (RAM, Random Access Memory), electric carrier signal, telecommunication signal and software distribution medium, etc.

[0085] The computer-readable storage medium may be an internal storage unit of the electronic device of any of the aforementioned embodiments, such as a hard disk or memory of the electronic device. The computer-readable storage medium may also be an external storage device of the electronic device, such as a plug-in hard disk, a smart memory card (Smart Media Card, SMC), a secure digital (SecureDigital, SD) card, a flash card (Flash Card), etc. equipped on the electronic device. Furthermore, the computer-readable storage medium may also include both an internal storage unit of the electronic device and an external storage device. The computer-readable storage medium is used to store computer programs and other programs and data required by the electronic device. The computer-readable storage medium may also be used to temporarily store data that has been output or is to be output.

[0086] Those of ordinary skill in the art will appreciate that the units and algorithm steps of each example described in conjunction with the embodiments disclosed herein can be implemented in electronic hardware, computer software, or a combination of the two. In order to clearly illustrate the interchangeability of hardware and software, the composition and steps of each example have been generally described in terms of function in the above description. Whether these functions are performed in hardware or software depends on the specific application and design constraints of the technical solution. Professional and technical personnel can use different methods to implement the described functions for each specific application, but such implementation should not be considered to be beyond the scope of this disclosure.

[0087] Those skilled in the art can clearly understand that, for the convenience and brevity of description, the specific working processes of the electronic devices and units described above can refer to the corresponding processes in the aforementioned method embodiments, and will not be repeated here.

[0088] In the several embodiments provided in the present application, it should be understood that the disclosed electronic device and method can be implemented in other ways. For example, the system embodiments described above are only schematic. For example, the division of units is only a logical function division. There may be other division methods in actual implementation. For example, multiple units or components can be combined or integrated into another system, or some features can be ignored or not executed. In addition, the mutual coupling or direct coupling or communication connection shown or discussed can be an indirect coupling or communication connection through some interfaces or units, or it can be an electrical, mechanical or other form of connection.

[0089] The units described as separate components may or may not be physically separated, and the components shown as units may or may not be physical units, that is, they may be located in one place or distributed on multiple network units. Some or all of the units may be selected according to actual needs to achieve the purpose of the embodiments of the present disclosure.

[0090] In addition, each functional unit in each embodiment of the present disclosure may be integrated into one processing unit, or each unit may exist physically separately, or two or more units may be integrated into one unit. The above-mentioned integrated unit may be implemented in the form of hardware or in the form of software functional units.

[0091] The above are only specific embodiments of the present disclosure, but the protection scope of the present disclosure is not limited thereto. Any technician familiar with the technical field can easily think of various equivalent modifications or replacements within the technical scope disclosed in the present disclosure, and these modifications or replacements should be included in the protection scope of the present disclosure. Therefore, the protection scope of the present disclosure should be based on the protection scope of the claims.

Claims

1. A method for improving the audio quality of voice communication of a remote user, characterized in that: include: Determining a target factor affecting the quality of the first audio signal, the target factor comprising environmental noise and network bandwidth; In response to the target factor being environmental noise, suppressing the environmental noise in the first audio signal based on a second audio signal to obtain a target audio signal, wherein the second audio signal is a signal having an audio amplitude equal to that of the environmental noise and an opposite phase; In response to the target factor being the network bandwidth, the first audio signal is adjusted based on an adjusted transmission rate to obtain a target audio signal, wherein the adjusted transmission rate is obtained based on the network bandwidth.

2. The method for improving the audio quality of voice communication of a remote user as claimed in claim 1, characterized in that: Also includes: A first equalizer of a first device is adjusted based on a first audio feature of the first audio signal so that the first equalizer of the first device processes the first audio signal, wherein the first device is a device that receives the first audio signal.

3. The method for improving the audio quality of voice communication of a remote user as claimed in claim 2, characterized in that: The adjusting a first equalizer of the first device based on the first audio feature of the first audio signal so that the first equalizer of the first device processes the first audio signal comprises: adjusting a first equalizer of the first device based on a correlation of the first audio feature with the first parameter; The first parameters are parameters of the first equalizer, and the first parameters include frequency range and gain.

4. The method for improving the audio quality of voice communication of a remote user as claimed in claim 1, characterized in that: In response to the target factor being environmental noise, suppressing the environmental noise in the first audio signal based on the second audio signal to obtain a target audio signal includes: If the audio amplitude of the environmental noise is less than a first preset value, suppressing the environmental noise based on the second audio signal to obtain a target audio signal; If the audio amplitude of the environmental noise is greater than or equal to a first preset value, the environmental noise is suppressed based on a second equalizer of the first device and the second audio signal to obtain a target audio signal.

5. The method for improving the audio quality of voice communication of a remote user as claimed in claim 4, characterized in that: Suppressing the environmental noise based on the second equalizer of the first device and the second audio signal to obtain a target audio signal includes: adjusting the second equalizer based on the environmental noise characteristics of the environmental noise so that the second equalizer processes the environmental noise to obtain the target environmental noise; The target environmental noise in the first audio signal is suppressed based on the second audio signal to obtain a target audio signal.

6. The method for improving the audio quality of voice communication of a remote user as claimed in claim 1, characterized in that: In response to the target factor being the network bandwidth, adjusting the first audio signal based on adjusting the transmission rate to obtain a target audio signal includes: If the network bandwidth is greater than a second preset value, adjusting the first audio signal based on a first transmission rate to obtain a target audio signal; If the network bandwidth is less than or equal to a second preset value, adjusting the first audio signal based on a second transmission rate to obtain a target audio signal; The first transmission rate is greater than the second transmission rate.

7. The method for improving the audio quality of voice communication of a remote user as claimed in claim 6, characterized in that: If the network bandwidth is less than or equal to the second preset value, the method for improving the audio quality of the remote user voice communication further includes: Calculate the rate of change of network bandwidth per unit time; If the change rate of the network bandwidth is greater than the first change rate, adjusting the sampling rate of the first device for the first audio signal based on the first step length; If the change rate of the network bandwidth is less than or equal to the second change rate, adjusting the sampling rate of the first device for the first audio signal based on the second step size; The first change rate is greater than the second change rate.

8. A system for improving the audio quality of voice communication of a remote user, characterized in that: include: A factor determination module, configured to determine a target factor affecting the quality of the first audio signal, wherein the target factor includes environmental noise and network bandwidth; a first quality adjustment module, configured to suppress the environmental noise in the first audio signal based on a second audio signal to obtain a target audio signal in response to the target factor being environmental noise, wherein the second audio signal is a signal having an audio amplitude equal to that of the environmental noise and an opposite phase; The second quality adjustment module is used for adjusting the first audio signal based on adjusting the transmission rate to obtain a target audio signal in response to the target factor being the network bandwidth, wherein the adjusted transmission rate is obtained based on the network bandwidth.

9. An electronic device comprising a memory, a processor, and a computer program stored in the memory and running on the processor, characterized in that: When the processor executes the computer program, the steps of the method according to any one of claims 1 to 7 are implemented.

10. A computer-readable storage medium storing a computer program, characterized in that: When the computer program is executed by a processor, the steps of the method according to any one of claims 1 to 7 are implemented.

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