CN117912475A - Center console compression sounding method, system, equipment and storage medium - Google Patents
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Abstract
The invention relates to a center console compression sounding method, which comprises the steps of obtaining an audio signal in an audio signal source, and preprocessing the audio signal through a center console center control model to obtain a preprocessed signal; judging whether the audio amplitude of the preprocessed signal obtained by preprocessing meets the audio amplitude requirement, and compressing the preprocessed signal according to a dynamic compression algorithm through a central control model to obtain a compressed audio signal when the preprocessed signal does not meet the audio amplitude requirement; judging whether the compressed audio signal meets the preset tone quality frequency range value or not, when the compressed audio signal does not meet the preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal according to a tone quality adjustment algorithm through a central control model, and outputting a balanced tone quality signal obtained by the adjustment processing to a sound system; and receiving feedback data generated by the sound system in real time, and updating the data of the central control model according to the feedback data. The invention can keep the audio undistorted.
Description
Technical Field
The present invention relates to the field of audio processing technologies, and in particular, to a console compression sounding method, system, device, and storage medium.
Background
The center console compressed sounding (Console Compressor/Limiter) is an audio processing technology, and is mainly used in the fields of audio recording, mixing, playback and the like. It originates from a recording studio and a broadcasting studio, and is initially used to control the dynamic range of the audio signal so that it is more balanced and consistent during playback. The console compression sounding reduces the difference between the stronger and weaker audio portions by compressing and limiting the dynamic range of the audio signal.
However, the parameter setting of the traditional console compressed sounding is fixed, and cannot be adjusted and optimized in real time according to different audio contents, and meanwhile, cannot process complex audio scenes and audio signals with high dynamic range, so that audio distortion and poor sound quality are caused. And the processing power and parameter settings of the traditional method are difficult to adapt to rapid changes of technology and industry, and limit the application of the traditional method in the field of continuously changing audio processing.
Disclosure of Invention
The main purpose of the invention is to provide a method, a system, equipment and a storage medium for compressing and sounding by a central console, which can be adjusted and optimized in real time according to different audio contents, can keep audio undistorted and improve tone quality in complex audio scenes and high dynamic range audio signals, and can update data in real time so as to adapt to rapid changes of technology and industry.
In order to achieve the above object, the present invention provides a console compression sounding method, including:
Acquiring an audio signal in an audio signal source, and preprocessing the audio signal through a central control model of a central control console to obtain a preprocessed signal;
Judging whether the audio amplitude of the preprocessed signal obtained by preprocessing meets the audio amplitude requirement, and compressing the preprocessed signal according to a dynamic compression algorithm through a central control model to obtain a compressed audio signal when the preprocessed signal does not meet the audio amplitude requirement;
Judging whether the compressed audio signal meets the preset tone quality frequency range value or not, when the compressed audio signal does not meet the preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal according to a tone quality adjustment algorithm through a central control model, and outputting a balanced tone quality signal obtained by the adjustment processing to a sound system;
And receiving feedback data generated by the sound system in real time, and updating the data of the central control model according to the feedback data.
Further, the preprocessing of the audio signal by the central control model of the central control console to obtain a preprocessed signal includes:
Converting the continuous audio signal into a discrete digital audio signal through the central control model, dividing the discrete digital audio signal into audio frames, and judging whether the overlapping area of the audio frames meets the preset frame overlapping proportion requirement;
When the overlapping area of the audio frames meets the requirement, carrying out frequency domain representation conversion on the audio frames according to a fast Fourier transform analysis method to obtain an audio frame frequency spectrogram, and judging whether noise exists in the audio frame frequency spectrogram;
When the audio frame frequency spectrogram has noise, extracting the frequency spectrum characteristics of the noise corresponding to the audio frame frequency spectrogram to obtain noise characteristics, analyzing the noise characteristics through the central control model to obtain the noise type of the audio frame, selecting a noise reduction algorithm corresponding to the noise type in a database through the central control model to perform noise reduction on the audio frame, and synthesizing the audio frame subjected to the noise reduction to obtain a preprocessing signal.
Further, the compressing the pre-processed signal by the central control model according to the dynamic compression algorithm to obtain a compressed audio signal includes:
Dividing the preprocessing signal to obtain signal frames, applying a window function to each signal frame, sequencing and converting a histogram of the signal frames applying the window function to obtain a histogram, obtaining a peak value of the histogram, calculating the signal frame with the maximum amplitude value according to the peak value, calculating the mean value and square root of the square sum of the signal frames according to the histogram, analyzing by combining the signal frame with the maximum amplitude value, the mean value and the square root of the square sum of the signal frames to obtain an analysis result, converting and calculating the analysis result to obtain a fluctuation amplitude value corresponding to the preprocessing signal, judging whether the fluctuation amplitude value meets the preset threshold range requirement or not, judging the type of the fluctuation amplitude value when the fluctuation amplitude value is not met, and lifting the preprocessing signal corresponding to the fluctuation amplitude value when the fluctuation amplitude value is lower than the threshold range to obtain a compressed audio signal; and when the fluctuation amplitude value is higher than the threshold range, compressing the preprocessed signal corresponding to the fluctuation amplitude value to obtain a compressed signal.
Further, the determining whether the compressed audio signal meets the preset tone quality frequency range value requirement includes:
And carrying out frequency analysis on the compressed audio signal to obtain a compressed frequency, judging whether the compressed frequency meets the requirement according to a preset threshold frequency, acquiring the audio signal when the compressed frequency meets the requirement, carrying out reduction processing on the compressed audio signal to obtain a reduction signal consistent with the format of the audio signal, carrying out time alignment on the reduction signal and the audio signal, and judging whether the reduction signal subjected to time alignment meets the requirement of a preset tone quality frequency range value.
Further, when the compressed audio signal does not conform to the preset tone quality frequency range value, the tone quality frequency adjustment processing is performed on the compressed audio signal according to the tone quality adjustment algorithm through the central control model, including: calculating the energy proportion relation between the restored signal and the audio signal to obtain a signal-to-noise ratio; calculating the proportional relation between the harmonic component and the fundamental frequency component in the restored signal to obtain the distortion degree; calculating the peak power or energy proportion relation between the original audio signal and the restored compressed audio signal to obtain a peak signal-to-noise ratio; calculating the difference between the frequency spectrum of the compressed audio signal and the frequency spectrum of the original audio signal to obtain a frequency spectrum distortion degree;
And detecting the signal-to-noise ratio, the distortion degree, the peak signal-to-noise ratio and the frequency spectrum distortion degree in sequence according to the tone quality frequency range value, when any one of the signal-to-noise ratio, the distortion degree, the peak signal-to-noise ratio and the frequency spectrum distortion degree does not accord with the tone quality frequency range value, carrying out frequency adjustment on the restored signal according to a tone quality adjustment algorithm through the central control model, adjusting frequency components exceeding the requirement to a proper range, clearing out frequency components failing to be adjusted, judging whether the restored signal after adjustment accords with the requirement of the tone quality frequency range value, and compressing the restored signal after adjustment when meeting the requirement to obtain the balanced tone quality signal.
Further, the receiving feedback data generated by the sound system in real time and updating the data of the central control model according to the feedback data includes:
after the sound system receives the balanced tone quality signals, the sound system operates according to the balanced tone quality signals and outputs corresponding operation audio signals, feedback data are generated according to the operation audio signals, the feedback data are sent to the central control model, the feedback data are analyzed through the central control model to obtain analysis results, the analysis results and the audio signals acquired from the audio signal source are subjected to data analysis to obtain deviation data of the operation audio signals and the audio signals, and the central control model is used for carrying out data updating according to the deviation data.
The invention also provides a center console compression sounding system, which comprises:
the acquisition module is used for acquiring the audio signals in the audio signal source, and preprocessing the audio signals through the central control model to obtain preprocessed signals;
The compression module is used for carrying out audio amplitude requirement detection on the preprocessed signals obtained through preprocessing according to the audio amplitude values, and carrying out compression processing on the preprocessed signals through a central control model according to a dynamic compression algorithm when the preprocessed signals do not meet the audio amplitude requirements to obtain compressed audio signals;
The balance module is used for carrying out tone quality frequency detection on the compressed audio signal through a preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal through a central control model according to a tone quality adjustment algorithm when the compressed audio signal does not accord with the preset tone quality frequency range value, and outputting a balanced tone quality signal obtained through the adjustment processing to the sound system;
And the processing module is used for receiving feedback data generated by the sound system in real time and updating the data of the central control model according to the feedback data.
The invention also provides a center console compression sounding device, comprising:
a memory for storing a program;
And the processor is used for executing the program to realize the steps of the method for compressing the sound production equipment by the center console.
The present invention also provides a storage medium storing computer instructions for causing a computer to perform any one of the console compression sounding methods.
The center console compression sounding method, system, equipment and storage medium provided by the invention have the following beneficial effects:
The preprocessing signals which do not meet the audio amplitude requirement are compressed by adopting a dynamic compression algorithm to obtain compressed audio signals, the audio signals can be adjusted and optimized in real time according to different audio contents, the amplitude of the audio signals is controlled while the dynamic range of the audio signals is maintained, the overlarge or overlarge volume fluctuation is avoided, the audibility and the comfort of the audio are improved, and the audio is kept undistorted and the tone quality is improved in complex audio scenes and high dynamic range audio signals. The compressed audio signal which does not meet the preset tone quality frequency range value requirement is subjected to tone quality frequency adjustment processing by adopting a tone quality adjustment algorithm, so that the frequency range of the audio signal meets the preset requirement, the tone quality of the audio is ensured to be in a reasonable range, and a clear and balanced sound effect is provided. And updating the data of the central control model according to the feedback data, so that the method can adapt to rapid changes of technology and industry.
Drawings
FIG. 1 is a flow chart of a console compression sounding method of the present invention;
FIG. 2 is a block diagram of a console compression sounding system of the present invention;
Fig. 3 is a block diagram of a console compressed sound generating device of the present invention.
The achievement of the objects, functional features and advantages of the present invention will be further described with reference to the accompanying drawings, in conjunction with the embodiments.
Detailed Description
The present invention will be described in further detail with reference to the drawings and examples, in order to make the objects, technical solutions and advantages of the present invention more apparent. It should be understood that the specific embodiments described herein are for purposes of illustration only and are not intended to limit the scope of the invention.
The invention will be further described with reference to the drawings and detailed description.
Referring to fig. 1, the invention provides a center console compression sounding method, which comprises the following steps:
step 1: acquiring an audio signal in an audio signal source, and preprocessing the audio signal through a central control model of a central control console to obtain a preprocessed signal;
Step 2: judging whether the audio amplitude of the preprocessed signal obtained by preprocessing meets the audio amplitude requirement, and compressing the preprocessed signal according to a dynamic compression algorithm through a central control model to obtain a compressed audio signal when the preprocessed signal does not meet the audio amplitude requirement;
Step 3: judging whether the compressed audio signal meets the preset tone quality frequency range value or not, when the compressed audio signal does not meet the preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal according to a tone quality adjustment algorithm through a central control model, and outputting a balanced tone quality signal obtained by the adjustment processing to a sound system;
Step 4: and receiving feedback data generated by the sound system in real time, and updating the data of the central control model according to the feedback data.
As described above, the specific steps are as follows:
Step 1: an audio signal is obtained from an audio signal source, wherein the audio signal source is any one of a microphone, a recording device and an audio file. After the audio signal is obtained, it is sampled and converted, converting the continuous analog audio signal into a discrete digital audio signal, which can be converted by an analog-to-digital converter (ADC). The obtained digital audio signals are sent to a central control model of a central control console for preprocessing, and finally the preprocessed audio signals are taken as output, wherein the central control model of the central control console is a deep learning model for preprocessing the audio signals and is responsible for receiving, processing and managing the audio signals and performing various preprocessing operations on the audio signals so as to meet specific requirements and application scenes
Step 2: and acquiring a pre-processing signal, performing audio amplitude analysis on the pre-processing signal, and judging whether the pre-processing signal meets the audio amplitude requirement, wherein the audio amplitude can be evaluated by calculating indexes such as peak value, root Mean Square (RMS) and the like of the signal. Comparing the audio amplitude obtained by analysis with a preset audio amplitude requirement, and if the audio amplitude of the pre-processing signal exceeds the required range (too large or too small), performing compression processing on the pre-processing signal. The preprocessing signal is compressed by a dynamic compression algorithm in the central control model, wherein the dynamic compression algorithm generally comprises a compressor, a limiter and other components, and is used for adjusting the dynamic range of the audio signal to be within a required range. According to the actual requirements, proper compression parameters such as compression ratio, threshold value, attack time and release time are set, and the parameters influence the compression effect and the perception characteristic of the audio. And applying the compression parameters to the preprocessed signals, performing compression processing through a central control model to obtain compressed audio signals, and outputting the compressed audio signals.
Step 3: and analyzing the tone quality frequency range of the compressed audio signal, and judging whether the tone quality frequency range meets the preset tone quality frequency range requirement, wherein the tone quality frequency range can be evaluated through methods such as frequency spectrum analysis and the like. Comparing the analyzed tone quality frequency range with a preset tone quality frequency range requirement, and if the tone quality frequency range of the compressed audio signal exceeds the required range (too wide or too narrow), performing tone quality frequency adjustment processing on the compressed audio signal. The compressed audio signal is subjected to a timbre frequency adjustment process by a timbre adjustment algorithm in the central control model, wherein the timbre adjustment algorithm can comprise an equalizer, a filter and other components for adjusting the frequency response of the audio signal to be within a required range. And sets appropriate tone quality adjustment parameters such as frequency gain, center frequency, bandwidth, etc., which will affect the tone quality adjustment effect and perceptual characteristics of the audio. And applying the tone quality adjustment parameters to the compressed audio signals, and performing tone quality frequency adjustment processing through the central control model to obtain balanced tone quality signals, and directly outputting the balanced tone quality signals to a sound system for playing.
Step 4: by establishing a feedback data channel between the center console and the sound system, feedback data generated by the sound system can be received in real time, and the feedback data can be realized through network connection, sensors and the like. Feedback data received from the sound system is monitored in real-time, wherein the feedback data includes information such as parameters of the audio signal, sound system status, user feedback, etc. And the received feedback data is analyzed and processed, and the feedback data can be analyzed and processed through data analysis, feature extraction and statistical analysis. And according to the feedback data obtained by analysis, carrying out data updating on the central control model, including operations of updating model parameters, adjusting algorithm parameters, optimizing model structures and the like, so as to improve the performance and adaptability of the central control model. And retraining the model according to the updated data, and adjusting the weight and parameters of the model by taking the feedback data as a training sample so as to improve the accuracy and effect of the model. And verifying and testing the updated central control model to ensure the performance and stability of the central control model in practical application. And finally, applying the updated central control model to the audio processing process.
According to the center console compression sounding method provided by the invention, the audio frequency amplitude is judged and the compression processing is carried out, so that the audio frequency signal is ensured to be in the preset amplitude range, the negative influence of the oversized or undersized audio frequency signal on a sound system is avoided, and the definition and the balance of the audio frequency can be maintained. The audio signal is processed through the dynamic compression algorithm, and the dynamic range of the audio is adjusted, so that the volume change of the audio in the playing process is more stable and consistent, and excessive peak value of the audio and inaudibility of a low volume part can be avoided. The frequency response of the audio signal is ensured to be in a preset range by judging the tone quality frequency range and adjusting the tone quality frequency range, and the situation that the audio frequency is too strong or too weak in different frequency ranges is avoided, so that the balance and naturalness of the audio frequency are maintained. The feedback data of the sound system is received in real time and the data of the central control model is updated, so that the performance and the adaptability of the model are optimized according to different audio contents and user feedback in practical application, and the central control model can be continuously improved, so that the central control model is better adapted to different audio processing requirements.
In one embodiment, preprocessing an audio signal by a central control model of a central control console to obtain a preprocessed signal includes:
And converting the analog audio signal into a digital audio signal according to two steps of sampling and quantizing through a central control model of the central control console. The discrete digital audio signal is divided into audio frames, where the audio frames are small segments of the audio signal, typically of a length of a few milliseconds to a few tens of milliseconds, and the overlapping area between frames needs to be considered during the division. After the audio frames are divided, judging whether the overlapping area of the audio frames meets the preset frame overlapping proportion requirement, wherein the frame overlapping proportion refers to the proportion of the overlapping area between adjacent frames, and is usually used for signal smooth transition in subsequent processing.
For the audio frames meeting the frame overlapping proportion requirement, the audio frames are converted into frequency domain representation by using methods such as Fast Fourier Transform (FFT) and the like, so that a spectrogram of the audio frames is obtained, and the spectrogram shows the energy distribution condition of the audio signals on different frequencies.
And judging whether noise exists in the audio frame frequency spectrogram or not by analyzing the characteristics such as energy distribution, peak value, spectrum morphology and the like in the spectrogram.
When the audio frequency frame frequency spectrogram has noise, extracting the frequency spectrum characteristics of the audio frequency frame frequency spectrogram corresponding to the noise, wherein the frequency spectrum characteristics comprise the information such as the frequency range and the energy distribution of the noise. The extracted noise characteristics are analyzed through the central control model to obtain the noise type of the audio frame, and the central control model can identify different types of noise, such as white noise, background noise and the like through training and learning. And selecting a noise reduction algorithm corresponding to the noise type in the database according to the central control model to carry out noise reduction processing on the audio frame, wherein the noise reduction algorithm is a combination of a series of signal processing technologies and is used for inhibiting or reducing noise components. And synthesizing the audio frames subjected to the noise reduction treatment to obtain a preprocessing signal, wherein the synthesis treatment is to carry out overlapping synthesis on the audio frames according to a certain rule and algorithm so as to keep the continuity and smooth transition of the audio.
According to the embodiment, the spectrogram and the noise characteristics of the audio frame are analyzed, the central control model identifies and reduces noise components, so that the influence of background noise on the audio quality is reduced, and the definition and the audibility of the audio are improved. The central control model can carry out enhancement processing on the audio signals according to the characteristics of the audio frames and the central control algorithm, so that the audio is plump and bright, the details and the sound effects in the audio are highlighted, and the quality and the immersion sense of the audio are improved. And controlling the dynamic range of the audio frame through the central control model, namely adjusting the volume difference of the audio signal. By compressing the dynamic range of the audio, the audio is made to have a more consistent volume level across different playback devices, avoiding the problem of the audio being too noisy or too quiet. And (3) carrying out equalization processing on the frequency spectrum of the audio frame through the central control model, namely adjusting the volume balance on different frequencies, so as to improve the balance between tone color and audio elements of the audio and enable the sounds in different frequency ranges to be clearer and balanced.
In one embodiment, compressing the preprocessed signal by the central control model according to a dynamic compression algorithm to obtain a compressed audio signal comprises:
The pre-processed signal is divided into a series of signal frames, where a signal frame is a small segment of the pre-processed signal, typically a length of a few milliseconds to a few tens of milliseconds. A window function is applied to each signal frame, the window function being a mathematical function for reducing amplitude jumps across the signal frames to avoid introducing discontinuities between the signal frames. The row sorts the signal frames to which the window function is applied and converts them into a histogram showing the distribution of the signal frames over different amplitude ranges. The peak value, i.e. the amplitude value corresponding to the pillar with the largest amplitude, is obtained from the histogram. And analyzing according to the information such as the signal frame with the maximum amplitude value, the average value of the square sum of the signal frames, the square root and the like to obtain an analysis result, and performing conversion calculation on the analysis result to obtain a fluctuation amplitude value corresponding to the preprocessing signal. Judging whether the fluctuation amplitude value meets the preset threshold range requirement, and carrying out corresponding compression processing adjustment on the preprocessing signal according to the type of the fluctuation amplitude value and the threshold range. When the fluctuation amplitude value is lower than the threshold range, the preprocessing signal corresponding to the fluctuation amplitude value is lifted to obtain a compressed audio signal; and when the fluctuation amplitude value is higher than the threshold range, compressing the preprocessed signal corresponding to the fluctuation amplitude value to obtain a compressed audio signal.
According to the embodiment, the preprocessing signals are analyzed and adjusted, so that the compression of the dynamic range is realized, the intensity difference of the audio signals is smaller, the problems of overhigh peak value and overlook details are reduced, and the audio is more balanced and consistent in the playing process. The fluctuation amplitude value is analyzed and adjusted according to the threshold range, so that the fluctuation amplitude of the audio signal can be effectively controlled, the audio signal is prevented from being too severe or distorted, and a more stable and comfortable hearing experience is provided. And the preprocessing signals are enhanced or compressed according to the analysis result, so that the amplitude and the dynamic range of the audio signals are adjusted, the audio signals are more suitable for specific playing environments and requirements, and the definition, the volume balance and the hearing effect of the audio are optimized.
In one embodiment, determining whether the compressed audio signal meets a predetermined timbre frequency range value requirement includes:
And carrying out frequency analysis on the compressed audio signal by applying a Fourier transform method to obtain the energy distribution condition of the audio signal on different frequencies, thereby obtaining the specific characteristics of the compressed frequency.
And judging whether the compression frequency meets the preset tone quality frequency range value requirement or not by comparing the relation between the compression frequency and the threshold frequency. When the compression frequency is satisfactory, an audio signal is acquired, which may include the original audio signal and the preprocessed signal, for use as a reference for the restoration process.
And restoring the compressed audio signal, namely decompressing, removing distortion or noise introduced by compression and the like, and restoring the quality of the original audio signal to obtain a restored signal consistent with the format of the audio signal. The time alignment of the restored signal with the audio signal will be ensured by synchronizing the starting points of the restored signal with the audio signal to ensure that they remain consistent in time, while time stretching or compression techniques can also be applied to time align the restored signal with the audio signal.
And (3) carrying out frequency analysis on the time-aligned restored signals, judging whether the restored signals meet the preset tone quality frequency range value requirement, and particularly determining whether the restored signals meet the requirement or not by comparing the energy distribution of the restored signals on different frequencies with the preset tone quality frequency range.
According to the embodiment, through frequency analysis of the compressed audio signal and the restored signal, the energy distribution condition of the audio signal on different frequencies is evaluated, and whether the compressed audio signal is in a preset tone quality frequency range can be judged, so that the high-frequency and low-frequency performance of the audio can meet the requirements. The quality of the original audio signal can be recovered by restoring the compressed audio signal, the distortion or noise introduced by compression is reduced, and the fidelity and audibility of the audio are improved. By time aligning the restored signal with the audio signal, ensuring that they remain consistent in time, helps to eliminate the problem of time shifting or out of synchronization due to compression and processing, providing a more accurate and consistent audio experience.
In one embodiment, when the compressed audio signal does not conform to the preset tone quality frequency range value, performing tone quality frequency adjustment processing on the compressed audio signal according to a tone quality adjustment algorithm through the central control model, including: the signal-to-noise ratio, which reflects the ratio of the signal to the noise, is obtained by calculating the energy proportional relationship between the restored signal and the audio signal, and is used to evaluate the sharpness and noise level of the audio signal. The distortion degree is obtained by calculating the proportional relation between the harmonic component and the fundamental frequency component in the restored signal, reflects the degree of nonlinear distortion in the signal and is used for evaluating the distortion degree of the audio signal. And calculating the peak power or energy proportional relation between the original audio signal and the restored compressed audio signal to obtain a peak signal-to-noise ratio, wherein the peak signal-to-noise ratio reflects the signal-to-noise ratio level of the signal at the peak value and is used for evaluating the dynamic range and the peak retention capacity of the audio signal. And calculating the difference between the frequency spectrum of the compressed audio signal and the frequency spectrum of the original audio signal to obtain a frequency spectrum distortion degree, wherein the frequency spectrum distortion degree reflects the distortion condition of the signal on a frequency domain and is used for evaluating the frequency spectrum fidelity of the audio signal.
And detecting the signal-to-noise ratio, the distortion degree, the peak signal-to-noise ratio and the frequency spectrum distortion degree in sequence, and when any one of the 4 parameters does not accord with the tone quality frequency range value, carrying out frequency adjustment processing on the restored signal according to a tone quality adjustment algorithm through a central control model, wherein the frequency adjustment processing comprises the steps of adjusting the frequency components which exceed the meeting requirements to be within a proper range, and eliminating the frequency components which fail to be adjusted. By comparing the energy distribution of the adjusted restoring signal on different frequencies with the preset tone quality frequency range, whether the adjusted restoring signal meets the requirement of the tone quality frequency range value can be judged, so that whether the requirement is met can be determined.
When the adjusted restored signal meets the requirement, the restored signal is compressed to obtain a balanced tone quality signal, wherein the compression processing comprises the technologies of dynamic range control, equalization processing, noise elimination and the like so as to optimize the quality and balance of the audio.
According to the embodiment, the frequency components exceeding the tone quality frequency range can be adjusted to be within a proper range by carrying out frequency adjustment on the restored signal, the frequency components failing to be adjusted are removed, the balance and definition of the audio signal can be kept, and the tone quality frequency range which meets the requirements better is provided. By calculating indexes such as signal-to-noise ratio, distortion degree, peak signal-to-noise ratio, spectrum distortion degree and the like, and performing corresponding frequency adjustment, the definition, dynamic range, distortion degree and spectrum fidelity of the audio signal are optimized, so that the quality and hearing experience of the audio are improved. The adjusted restored signal is compressed to obtain a balanced tone quality signal, and the dynamic range, the balance and the noise level of the audio are optimized through the compression, so that the audio playback device is more suitable for specific playback environments and requirements.
In one embodiment, receiving feedback data generated by the sound system in real time, and updating data of the central control model according to the feedback data includes:
When the sound system receives the balanced tone quality signal, the sound system operates according to the received balanced tone quality signal and generates a corresponding operation audio signal, wherein the operation audio signal is a signal which can be used for music playing, video playing or other audio applications. The sound system generates feedback data from the operating audio signal. The feedback data comprises information such as frequency spectrum distribution, peak power, distortion degree and the like of the audio signal, the information is used for evaluating the quality and performance of the audio, and the sound system sends the generated feedback data to the central control model. The central control model analyzes the received feedback data and extracts key information therein, wherein the key information comprises frequency spectrum characteristics, dynamic range, distortion degree and the like of the audio signal. And the central control model performs data analysis on the analysis result and the original audio signal acquired from the audio signal source, and obtains deviation data of the operation audio signal and the audio signal by comparing the difference between the analysis result and the original audio signal. And the central control model updates parameters of the central control model according to the deviation data, wherein the parameters comprise parameters of a compression algorithm, an algorithm for optimizing audio processing or other related parameters.
According to the embodiment, the parameters of the central control model are adjusted and optimized according to the characteristics and performance performances of the current running audio signal by receiving feedback data generated by the sound system in real time, and the system can be adaptively adjusted according to actual conditions by updating the parameters of the central control model in real time so as to adapt to different environments and application scenes. The feedback data is received in real time and updated, so that potential problems and adverse phenomena can be found and corrected, stability and reliability of the sound system are improved through timely adjustment and optimization, and distortion, noise or other quality problems possibly occurring in the audio processing process are reduced.
Referring to fig. 2, the present invention further provides a console compression sounding system, including:
The acquisition module is used for acquiring the audio signals in the audio signal source, and preprocessing the audio signals through the central control model to obtain preprocessed signals;
The compression module is used for carrying out audio amplitude requirement detection on the preprocessed signals obtained through preprocessing according to the audio amplitude values, and carrying out compression processing on the preprocessed signals through a central control model according to a dynamic compression algorithm when the preprocessed signals do not meet the audio amplitude requirements to obtain compressed audio signals;
the balance module is used for carrying out tone quality frequency detection on the compressed audio signal through a preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal through a central control model according to a tone quality adjustment algorithm when the compressed audio signal does not accord with the preset tone quality frequency range value, and outputting a balanced tone quality signal obtained through the adjustment processing to the sound system;
The processing module is used for receiving feedback data generated by the sound system in real time and updating the data of the central control model according to the feedback data.
According to the center console compression sounding system, the audio frequency amplitude is judged and is subjected to compression processing, so that the audio frequency signals are ensured to be in a preset amplitude range, the audio frequency signals which are too large or too small are prevented from negatively affecting the sound system, and the definition and the balance of the audio frequency can be maintained. The audio signal is processed through the dynamic compression algorithm, and the dynamic range of the audio is adjusted, so that the volume change of the audio in the playing process is more stable and consistent, and excessive peak value of the audio and inaudibility of a low volume part can be avoided. The frequency response of the audio signal is ensured to be in a preset range by judging the tone quality frequency range and adjusting the tone quality frequency range, and the situation that the audio frequency is too strong or too weak in different frequency ranges is avoided, so that the balance and naturalness of the audio frequency are maintained. The feedback data of the sound system is received in real time and the data of the central control model is updated, so that the performance and the adaptability of the model are optimized according to different audio contents and user feedback in practical application, and the central control model can be continuously improved, so that the central control model is better adapted to different audio processing requirements.
Referring to fig. 3, the present invention further provides a console compression sounding device, including:
a memory for storing a program;
And the processor is used for executing the program to realize the steps of the method for compressing the sound production equipment by the center console.
In this embodiment, the processor and the memory may be connected by a bus or other means. The memory may include volatile memory, such as random access memory; the memory may also include non-volatile memory, such as read-only memory, flash memory, a hard disk, or a solid state disk. The processor may be a general-purpose processor, such as a central processing unit, a digital audio signal processor, an application specific integrated circuit, or one or more integrated circuits configured to implement embodiments of the present invention.
The present invention also provides a storage medium storing computer instructions for causing a computer to perform any one of the console compression sounding methods.
The foregoing description is only of the preferred embodiments of the present invention and is not intended to limit the scope of the invention, and all equivalent structures or equivalent processes using the descriptions and drawings of the present invention or direct or indirect application in other related technical fields are included in the scope of the present invention.
Claims (9)
1. A compression sounding method for a center console is characterized by comprising the following steps:
Acquiring an audio signal in an audio signal source, and preprocessing the audio signal through a central control model of a central control console to obtain a preprocessed signal;
Judging whether the audio amplitude of the preprocessed signal obtained by preprocessing meets the audio amplitude requirement, and compressing the preprocessed signal according to a dynamic compression algorithm through a central control model to obtain a compressed audio signal when the preprocessed signal does not meet the audio amplitude requirement;
Judging whether the compressed audio signal meets the preset tone quality frequency range value or not, when the compressed audio signal does not meet the preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal according to a tone quality adjustment algorithm through a central control model, and outputting a balanced tone quality signal obtained by the adjustment processing to a sound system;
And receiving feedback data generated by the sound system in real time, and updating the data of the central control model according to the feedback data.
2. The method for compressing and sounding a console according to claim 1, wherein the preprocessing the audio signal by the console model to obtain a preprocessed signal comprises:
Converting the continuous audio signal into a discrete digital audio signal through the central control model, dividing the discrete digital audio signal into audio frames, and judging whether the overlapping area of the audio frames meets the preset frame overlapping proportion requirement;
When the overlapping area of the audio frames meets the requirement, carrying out frequency domain representation conversion on the audio frames according to a fast Fourier transform analysis method to obtain an audio frame frequency spectrogram, and judging whether noise exists in the audio frame frequency spectrogram;
When the audio frame frequency spectrogram has noise, extracting the frequency spectrum characteristics of the noise corresponding to the audio frame frequency spectrogram to obtain noise characteristics, analyzing the noise characteristics through the central control model to obtain the noise type of the audio frame, selecting a noise reduction algorithm corresponding to the noise type in a database through the central control model to perform noise reduction on the audio frame, and synthesizing the audio frame subjected to the noise reduction to obtain a preprocessing signal.
3. The method for generating sound by console compression according to claim 1, wherein the compressing the pre-processed signal by the console model according to the dynamic compression algorithm to obtain the compressed audio signal comprises:
Dividing the preprocessing signal to obtain signal frames, applying a window function to each signal frame, sequencing and converting a histogram of the signal frames applying the window function to obtain a histogram, obtaining a peak value of the histogram, calculating the signal frame with the maximum amplitude value according to the peak value, calculating the mean value and square root of the square sum of the signal frames according to the histogram, analyzing by combining the signal frame with the maximum amplitude value, the mean value and the square root of the square sum of the signal frames to obtain an analysis result, converting and calculating the analysis result to obtain a fluctuation amplitude value corresponding to the preprocessing signal, judging whether the fluctuation amplitude value meets the preset threshold range requirement or not, judging the type of the fluctuation amplitude value when the fluctuation amplitude value is not met, and lifting the preprocessing signal corresponding to the fluctuation amplitude value when the fluctuation amplitude value is lower than the threshold range to obtain a compressed audio signal; and when the fluctuation amplitude value is higher than the threshold range, compressing the preprocessed signal corresponding to the fluctuation amplitude value to obtain a compressed signal.
4. The console compression sounding method of claim 1, wherein the determining whether the compressed audio signal meets the preset tone quality frequency range value requirement comprises:
And carrying out frequency analysis on the compressed audio signal to obtain a compressed frequency, judging whether the compressed frequency meets the requirement according to a preset threshold frequency, acquiring the audio signal when the compressed frequency meets the requirement, carrying out reduction processing on the compressed audio signal to obtain a reduction signal consistent with the format of the audio signal, carrying out time alignment on the reduction signal and the audio signal, and judging whether the reduction signal subjected to time alignment meets the requirement of a preset tone quality frequency range value.
5. The method for generating compressed audio signals by console according to claim 4, wherein when the compressed audio signals do not conform to the preset tone quality frequency range value, the tone quality frequency adjustment processing is performed on the compressed audio signals by the console model according to a tone quality adjustment algorithm, comprising: calculating the energy proportion relation between the restored signal and the audio signal to obtain a signal-to-noise ratio; calculating the proportional relation between the harmonic component and the fundamental frequency component in the restored signal to obtain the distortion degree; calculating the peak power or energy proportion relation between the original audio signal and the restored compressed audio signal to obtain a peak signal-to-noise ratio; calculating the difference between the frequency spectrum of the compressed audio signal and the frequency spectrum of the original audio signal to obtain a frequency spectrum distortion degree;
And detecting the signal-to-noise ratio, the distortion degree, the peak signal-to-noise ratio and the frequency spectrum distortion degree in sequence according to the tone quality frequency range value, when any one of the signal-to-noise ratio, the distortion degree, the peak signal-to-noise ratio and the frequency spectrum distortion degree does not accord with the tone quality frequency range value, carrying out frequency adjustment on the restored signal according to a tone quality adjustment algorithm through the central control model, adjusting frequency components exceeding the requirement to a proper range, clearing out frequency components failing to be adjusted, judging whether the restored signal after adjustment accords with the requirement of the tone quality frequency range value, and compressing the restored signal after adjustment when meeting the requirement to obtain the balanced tone quality signal.
6. The method of claim 1, wherein the receiving feedback data generated by the sound system in real time and updating the data of the central control model according to the feedback data comprises:
after the sound system receives the balanced tone quality signals, the sound system operates according to the balanced tone quality signals and outputs corresponding operation audio signals, feedback data are generated according to the operation audio signals, the feedback data are sent to the central control model, the feedback data are analyzed through the central control model to obtain analysis results, the analysis results and the audio signals acquired from the audio signal source are subjected to data analysis to obtain deviation data of the operation audio signals and the audio signals, and the central control model is used for carrying out data updating according to the deviation data.
7. A center console compression sounding system, comprising:
the acquisition module is used for acquiring the audio signals in the audio signal source, and preprocessing the audio signals through the central control model to obtain preprocessed signals;
The compression module is used for carrying out audio amplitude requirement detection on the preprocessed signals obtained through preprocessing according to the audio amplitude values, and carrying out compression processing on the preprocessed signals through a central control model according to a dynamic compression algorithm when the preprocessed signals do not meet the audio amplitude requirements to obtain compressed audio signals;
The balance module is used for carrying out tone quality frequency detection on the compressed audio signal through a preset tone quality frequency range value, carrying out tone quality frequency adjustment processing on the compressed audio signal through a central control model according to a tone quality adjustment algorithm when the compressed audio signal does not accord with the preset tone quality frequency range value, and outputting a balanced tone quality signal obtained through the adjustment processing to the sound system;
And the processing module is used for receiving feedback data generated by the sound system in real time and updating the data of the central control model according to the feedback data.
8. A center console compression sounding device, comprising:
a memory for storing a program;
A processor for executing the program to perform the steps of a method for compressing sound generating devices on a console as claimed in any one of claims 1 to 6.
9. A storage medium having stored thereon computer instructions for causing a computer to perform the method according to any one of claims 1 to 6.
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