阅读说明：本技术 扩声系统、及麦克风通道数据选择方法 (Sound amplification system and microphone channel data selection method ) 是由 吴道远 迟景立 于 2021-06-04 设计创作，主要内容包括：本发明涉及音频数据处理领域,具体涉及一种扩声系统、及麦克风通道数据选择方法,极大地降低混响以及噪声的影响同时还有利于扩大覆盖范围,保证了扩声质量。本发明麦克风通道数据选择方法,包括：配置每个麦克风通道采集数据的采样率以及采样位数；根据配置的采样率以及采样位数将各个麦克风采集的模拟信号转换为数字信号,并将所述数字信号发送给音频处理模块；音频处理模块用对数字信号进行处理,根据处理结果选择对应麦克风通道数据,并将对应麦克风通道数据发送给音频输出模块；音频输出模块将处理后的数字信号转换为模拟信号后发送至音响设备进行输出。本发明适用于对教室、会议室、报告厅等场地的声音扩大。(The invention relates to the field of audio data processing, in particular to a sound amplification system and a microphone channel data selection method, which greatly reduce the influence of reverberation and noise, are beneficial to expanding the coverage range and ensure the sound amplification quality. The invention discloses a microphone channel data selection method, which comprises the following steps: configuring the sampling rate and sampling digit of data collected by each microphone channel; converting analog signals collected by each microphone into digital signals according to the configured sampling rate and sampling bit number, and sending the digital signals to an audio processing module; the audio processing module is used for processing the digital signals, selecting corresponding microphone channel data according to a processing result and sending the corresponding microphone channel data to the audio output module; and the audio output module converts the processed digital signals into analog signals and then sends the analog signals to the sound equipment for output. The invention is suitable for sound amplification in places such as classrooms, meeting rooms, reporting halls and the like.)

1. The system comprises a plurality of microphones and sound equipment, wherein the microphones are arranged at all positions of the site and are used for acquiring sound signals of speakers and sending the sound signals to the sound equipment for output;

the audio configuration module is used for configuring the sampling rate and the sampling bit number of the data acquired by each microphone channel;

the audio acquisition module is used for converting analog signals acquired by each microphone into digital signals according to the configured sampling rate and sampling bit number and sending the digital signals to the audio processing module;

the audio processing module is used for processing the digital signals, selecting corresponding microphone channel data according to a processing result and sending the corresponding microphone channel data to the audio output module;

and the audio output module converts the digital signals corresponding to the microphone channel data into analog signals and then sends the analog signals to the sound equipment for output.

2. The system of claim 1, wherein the processing of the digital signal frames by the audio processing module comprises:

the audio processing module performs digital operation on the digital signals to obtain the signal-to-noise ratio (SNR) of each frame of signal, and performs voice VAD (voice detection and volume) check on each frame of signal to obtain the reliability SCR for judging that the current frame of data is voice data;

and calculating the average signal-to-noise ratio within N frames of each microphone channel N is the number of data frames; calculating the number of frames of which the voice credibility SCR in each microphone channel N frame is greater than a set threshold, and recording the number of frames greater than the set threshold as VN;

calculating the frame number ratio V of the voice judged in the N frames of each microphone channel, wherein V is VN/N, and N is greater than 0;

according to the average signal-to-noise ratio in N frames of each microphone channelAnd calculating the frame ratio V to obtain a weight of each microphone channel, recording the weight as K, and selecting the microphone channel data corresponding to the maximum weight as the current output data.

3. The system according to claim 2, wherein the weight K is calculated as:ws is the weight of the signal-to-noise ratio, Wv is the weight of the human signal, and P is the data priority of the microphone.

4. The sound amplification system of claim 3, wherein the audio processing module is further configured to perform sound effect processing on the current output data, and after the sound effect processing is completed, the current output data is sent to the audio output module.

5. The sound enhancement system of claim 4 wherein the sound effect processing includes noise reduction, EQ, echo suppression, and gain adjustment.

6. Microphone channel data selection method for use in a sound amplification system according to any one of claims 1 to 5, comprising:

step 1, configuring the sampling rate and sampling digit of data collected by each microphone channel;

step 2, converting the analog signals collected by each microphone into digital signals according to the configured sampling rate and sampling bit number, and sending the digital signals to an audio processing module;

step 3, the audio processing module processes the digital signal, selects corresponding microphone channel data according to the processing result, and sends the corresponding microphone channel data to the audio output module;

and 4, converting the digital signals corresponding to the microphone channel data into analog signals by the audio output module, and then sending the analog signals to the sound equipment for output.

7. The method as claimed in claim 6, wherein the specific method for processing the digital signal by the audio processing module in step 3 comprises:

the audio processing module performs digital operation on the digital signals to obtain the signal-to-noise ratio (SNR) of each frame of signal, and performs voice VAD (voice detection and volume) check on each frame of signal to obtain the reliability SCR for judging that the current frame of data is voice data;

and calculating the average signal-to-noise ratio within N frames of each microphone channel N is the number of data frames; calculating the number of frames of which the voice credibility SCR in each microphone channel N frame is greater than a set threshold, and recording the number of frames greater than the set threshold as VN;

calculating the frame number ratio V of the voice judged in the N frames of each microphone channel, wherein V is VN/N, and N is greater than 0;

according to the average signal-to-noise ratio in N frames of each microphone channelAnd calculating the frame ratio V to obtain a weight of each microphone channel, recording the weight as K, and selecting the microphone channel data corresponding to the maximum weight as the current output data.

8. The method of claim 7, wherein the weight value K is calculated by the following formula:ws is the weight of the signal-to-noise ratio, Wv is the weight of the human signal, and P is the data priority of the microphone.

9. The method as claimed in claim 8, wherein the audio processing module is further configured to perform sound effect processing on the current output data, and after the sound effect processing is completed, the current output data is sent to the audio output module.

10. The method of claim 9 wherein the sound effects processing includes noise reduction, EQ, echo suppression, and gain adjustment.

Technical Field

The invention relates to the field of audio data processing, in particular to a sound amplification system and a microphone channel data selection method.

Background

In the existing sound amplifying system, when a plurality of microphones collect data, due to the characteristic of the remote microphone, when the required voice data is collected, the noise of a plurality of microphones is collected at the same time, so that the noise is increased when the voice data is fused and output, and meanwhile, the reverberation of the fused voice data is large due to the time difference and the azimuth difference of the data collected by the plurality of microphones.

The existing solution is to adopt technologies like microphone arrays and the like, and based on a microphone array algorithm or directly fuse and output multi-path microphone data.

However, the coverage range is limited due to the microphone array-based mode, and the coverage range is not easily expanded due to the influence of the hardware array; and the multi-path microphones are directly fused, so that not only can reverberation not be eliminated, but also noise increase and reverberation aggravation are brought, and the sound amplification effect is poorer than that of a single-path microphone.

Disclosure of Invention

The invention aims to provide a sound amplification system and a microphone channel data selection method, which greatly reduce the influence of reverberation and noise, are beneficial to expanding the coverage range and ensure the sound amplification quality.

The invention adopts the following technical scheme to realize the aim, and the sound amplifying system is used for amplifying the sound of a speaker in a field, comprises a plurality of microphones and sound equipment, wherein the microphones are arranged at all positions of the field and used for acquiring the sound signal of the speaker and sending the sound signal to the sound equipment for output, and further comprises an audio configuration module, an audio acquisition module, an audio processing module and an audio output module;

the audio configuration module is used for configuring the sampling rate and the sampling bit number of the data acquired by each microphone channel;

the audio acquisition module is used for converting analog signals acquired by each microphone into digital signals according to the configured sampling rate and sampling bit number and sending the digital signals to the audio processing module;

the audio processing module is used for processing the digital signals, selecting corresponding microphone channel data according to a processing result and sending the corresponding microphone channel data to the audio output module;

and the audio output module converts the digital signals corresponding to the microphone channel data into analog signals and then sends the analog signals to the sound equipment for output.

Further, the processing of the digital signal frame by the audio processing module includes:

the audio processing module performs digital operation on each frame of digital signals to obtain the signal-to-noise ratio (SNR) of each frame of signals, and performs voice VAD (voice detection and volume) check on each frame of signals to obtain the reliability SCR for judging that the current frame of data is voice data;

and calculating the average signal-to-noise ratio within N frames of each microphone channel N is the number of data frames; calculating the number of frames of which the voice credibility SCR in each microphone channel N frame is greater than a set threshold, and marking the number of frames greater than the set threshold as VN, wherein N is greater than 0;

calculating the frame number ratio V of the voice judged in the N frames of each microphone channel, wherein V is VN/N;

according to the average signal-to-noise ratio in N frames of each microphone channelAnd calculating the frame ratio V to obtain a weight of each microphone channel, recording the weight as K, and selecting the microphone channel data corresponding to the maximum weight as the current output data.

Further, the calculation formula of the weight K is as follows:ws is the weight of the signal-to-noise ratio, Wv is the weight of the human signal, and P is the data priority of the microphone.

Further, the audio processing module is further configured to perform sound effect processing on the current output data, and send the current output data to the audio output module after the sound effect processing is completed.

Further, the sound effect processing includes noise reduction, EQ, echo suppression, and gain adjustment.

The microphone channel data selection method is applied to the sound amplification system and comprises the following steps:

step 1, configuring the sampling rate and sampling digit of data collected by each microphone channel;

step 2, converting the analog signals collected by each microphone into digital signals according to the configured sampling rate and sampling bit number, and sending the digital signals to an audio processing module;

step 3, the audio processing module processes the digital signal, selects corresponding microphone channel data according to the processing result, and sends the corresponding microphone channel data to the audio output module;

and 4, converting the digital signals corresponding to the microphone channel data into analog signals by the audio output module, and then sending the analog signals to the sound equipment for output.

Further, in step 3, the specific method for processing the digital signal by the audio processing module includes:

the audio processing module performs digital operation on the digital signals to obtain the signal-to-noise ratio (SNR) of each frame of signal, and performs voice VAD (voice detection and volume) check on each frame of signal to obtain the reliability SCR for judging that the current frame of data is voice data;

and calculating the average signal-to-noise ratio within N frames of each microphone channel N is the number of data frames; calculating the number of frames of which the voice credibility SCR in each microphone channel N frame is greater than a set threshold, and recording the number of frames greater than the set threshold as VN;

calculating the frame number ratio V of the voice judged in the N frames of each microphone channel, wherein V is VN/N, and N is greater than 0;

and calculating to obtain a weight value of each microphone channel according to the average signal-to-noise ratio SNR in the N frames of each microphone channel and the frame ratio V, recording the weight value as K, and selecting the microphone channel data corresponding to the maximum weight value as the current output data.

Further, the calculation formula of the weight K is as follows:ws is the weight of the signal-to-noise ratio, Wv is the weight of the human signal, and P is the data priority of the microphone.

Further, the audio processing module is further configured to perform sound effect processing on the current output data, and send the current output data to the audio output module after the sound effect processing is completed.

Further, the sound effect processing includes noise reduction, EQ, echo suppression, and gain adjustment.

The invention does not directly perform fusion processing on the multi-path microphone data any more, and directly reduces the probability of the noise data fusion; the method comprises the steps that a sampling rate and sampling bit number of data collected by each microphone channel are configured before data collection, analog signals collected by each microphone are converted into digital signals according to the configured sampling rate and sampling bit number, digital signal frames are sent to an audio processing module, the audio processing module calculates a weight of each microphone channel according to the reliability of human voice data, the average signal-to-noise ratio in N frames of each microphone channel and the frame number ratio of human voice in N frames of each microphone channel, the weight is large, and the probability of representing human voice is large; therefore, the microphone channel data with the largest weight is selected as output data, and the noise data source is further reduced; and the invention does not adopt the technology of a microphone array, thereby greatly reducing the influence of reverberation and noise and being beneficial to expanding the coverage area and ensuring the sound amplification quality.

Drawings

FIG. 1 is a flow chart of the method of the present invention.

Detailed Description

The invention discloses an acoustic amplification system, which is used for amplifying the sound of a speaker in a field, and comprises a plurality of microphones and acoustic equipment, wherein the microphones are arranged at all positions of the field and used for acquiring the sound signal of the speaker and sending the sound signal to the acoustic equipment for output;

for example, the sound amplification device can be used for sound amplification in classrooms, meeting rooms, report halls and the like, and microphones can be installed in places such as classrooms, meeting rooms, report halls and the like; the speaker does not need to hold a microphone or wear a loudspeaker or only stand at a fixed location to speak.

The audio configuration module is used for configuring the sampling rate and the sampling bit number of the data acquired by each microphone channel;

the audio acquisition module is used for converting the analog signals acquired by each microphone into digital signals according to the configured sampling rate and sampling bit number and sending the digital signals to the audio processing module;

the audio processing module is used for processing the digital signals, selecting corresponding microphone channel data according to a processing result and sending the corresponding microphone channel data to the audio output module;

and the audio output module converts the digital signals corresponding to the microphone channel data into analog signals and then sends the analog signals to the sound equipment for output.

Wherein the audio processing module processes the digital signal frame, including:

the audio processing module performs digital operation on each frame of digital signals to obtain the signal-to-noise ratio (SNR) of each frame of signals, and performs voice VAD (voice activity Detection) check on each frame of signals to obtain the reliability SCR for judging that the current frame of data is voice data;

and calculating the average signal-to-noise ratio within N frames of each microphone channel N is the number of data frames; and calculating that the human voice credibility SCR in N frames of each microphone channel is greater than a set threshold valueAnd the number of frames greater than the set threshold is recorded as VN, N is greater than 0;

the threshold may be set to 0.5;

calculating the frame number ratio V of the voice judged in the N frames of each microphone channel, wherein V is VN/N;

according to the average signal-to-noise ratio in N frames of each microphone channelAnd calculating the frame ratio V to obtain a weight of each microphone channel, recording the weight as K, and selecting the microphone channel data corresponding to the maximum weight as the current output data.

The calculation formula of the weight K is as follows:ws is the weight of the signal-to-noise ratio, Wv is the weight of the human signal, and P is the data priority of the microphone.

The audio processing module is also used for carrying out sound effect processing on the current output data, and sending the current output data to the audio output module after the sound effect processing is finished.

Sound processing includes noise reduction, eq (equal), echo suppression, and gain adjustment.

The EQ basically functions to adjust the timbre by performing gain or attenuation on one or more frequency bands of the sound.

EQ generally includes the following three parameters: frequency-this is a parameter used to set the Frequency point you are to adjust; gain, Gain-a parameter used to adjust the Gain or attenuation at your set F value; quantize-a parameter used to set the "width" of the band you want to gain or attenuate. Here, note that: the frequency band you handle is wider when you set the smaller the Q value you set, and narrower when you set the larger the Q value you handle.

A method for selecting microphone channel data, a flow chart of which is shown in fig. 1, includes:

step 101, configuring the sampling rate and sampling bit number of data collected by each microphone channel;

step 102, converting analog signals collected by each microphone into digital signals according to the configured sampling rate and sampling bit number, and sending the digital signals to an audio processing module;

103, the audio processing module processes the digital signal, selects corresponding microphone channel data according to a processing result, and sends the corresponding microphone channel data to the audio output module;

and step 104, converting the digital signals corresponding to the microphone channel data into analog signals by the audio output module, and then sending the analog signals to the sound equipment for output.

In step 103, the specific method for processing the digital signal by the audio processing module includes:

the audio processing module performs digital operation on the digital signals to obtain the signal-to-noise ratio (SNR) of each frame of signal, and performs voice VAD (voice detection and volume) check on each frame of signal to obtain the reliability SCR for judging that the current frame of data is voice data;

and calculating the average signal-to-noise ratio within N frames of each microphone channel N is the number of data frames; calculating the number of frames of which the voice credibility SCR in each microphone channel N frame is greater than a set threshold, and recording the number of frames greater than the set threshold as VN;

calculating the frame number ratio V of the voice judged in the N frames of each microphone channel, wherein V is VN/N, and N is greater than 0;

according to the average signal-to-noise ratio in N frames of each microphone channelAnd calculating the frame ratio V to obtain a weight of each microphone channel, recording the weight as K, and selecting the microphone channel data corresponding to the maximum weight as the current output data.

Calculation of weight KThe formula is as follows:ws is the weight of the signal-to-noise ratio, Wv is the weight of the human signal, and P is the data priority of the microphone.

The audio processing module is also used for carrying out sound effect processing on the current output data, the current output data is sent to the audio output module after the sound effect processing is finished, and the sound effect processing comprises noise reduction, EQ, echo suppression and gain adjustment.

The invention does not adopt the microphone array technology, avoids the reduction of the coverage method, does not directly carry out fusion processing on multi-path microphone data any more, and directly reduces the probability of noise data fusion; the audio processing module calculates a weight of each microphone channel according to the reliability of the voice data, the average signal-to-noise ratio in N frames of each microphone channel and the frame number ratio of the voice in the N frames of each microphone channel, wherein the larger the weight is, the larger the probability expressed as the voice is; therefore, the microphone channel data with the largest weight is selected as the output data, and the noise data source is further reduced.

In conclusion, the invention greatly reduces the influence of reverberation and noise, is beneficial to expanding the coverage range and ensures the sound amplification quality.