Audio format identification method for HIFI music playing based on multi-parameter analysis

Abstract

The invention relates to the technical field of audio format recognition, and particularly discloses a HIFI music playing audio format recognition method based on multi-parameter analysis, which comprises the following steps: S1, carrying out preprocessing and multi-dimensional parameter recognition on audio data, and obtaining basic and decoding parameters such as format, sampling rate, digit and sound channel by analyzing a file header; s2, converting the audio data into a signal, performing spectrum and bit rate fluctuation analysis, and accurately judging that the audio has a lossy, lossless or pseudo-lossless attribute; and S3, establishing a three-level parameter chain of an original file, a decoding process and an output signal, generating an intuitive graphical display interface accordingly, representing a sampling rate, a sampling digit and sound channel configuration by using the height of a histogram, the number of square blocks and loudspeaker icons respectively, distinguishing audio attributes by colors, monitoring parameter changes in real time in a playing process, updating and displaying, and displaying the parameters in real time. And the user can comprehensively and intuitively master the audio state in real time.

Description

TECHNICAL FIELD

[0001] The present application relates to the technical field of audio format identification, and particularly relates to an audio format identification method for HIFI music playing based on multi-parameter analysis. BACKGROUND

[0002] With the rapid development of digital audio technology, HIFI audio formats have evolved from early lossy compression formats to lossless formats, and even high-resolution formats. The parameter system of HIFI audio formats is also becoming increasingly complex, including sampling rate, sampling bit number, channel mode, encoding attribute and multiple dimensions. These parameters directly affect the audio restoration quality and are an important basis for users to evaluate the performance of the playing system.

[0003] Existing audio format identification and display schemes mainly include software playing systems and hardware decoding devices. The software scheme usually extracts basic parameters by analyzing file header metadata, but cannot identify special parameters of high-resolution formats, and lacks the ability to verify the authenticity of the signal. The hardware scheme can detect the basic parameters of the decoded signal, but cannot associate the original file information, the display content is fragmented, and lacks full-link parameter association. In addition, the existing technology has the problems of display delay and high identification error rate during dynamic switching, and cannot meet the high-level needs of HIFI users for parameter transparency and real-time performance. SUMMARY

[0004] The purpose of the present application is to provide an audio format identification method for HIFI music playing based on multi-parameter analysis, which solves the following technical problems.

[0005] The purpose of the present application can be achieved by the following technical solutions: The audio format identification method for HIFI music playing based on multi-parameter analysis comprises the following steps: Step S1: Obtain audio data and pre-process the audio data, the pre-processing including integrity check and cache preprocessing; perform multi-dimensional parameter identification on the pre-processed audio data, the process of multi-dimensional parameter identification including analyzing the file header information of each audio file in the audio data, the file header information including format type, sampling rate, sampling bit number, channel information and decoding process parameters; Step S2: Convert the audio data into an audio signal, perform spectrum analysis and bit rate fluctuation analysis on the audio signal to determine the attributes of the audio signal, the attributes including lossy files, lossless files and pseudo-lossless files; Step S3: establishing a three-level parameter chain, generating graphical display content based on the three-level parameter chain, the graphical display content representing the sampling rate by column height, the sampling bit number by the number of squares, the channel information by a speaker icon, and different colors being used to distinguish different attributes; in the process of playing the audio signal, the changes of the three-level parameters are monitored in real time, and the graphical display content is updated in real time.

[0006] As a further scheme of the present application, the process of analyzing the file header information of the audio file comprises: If the audio file is in FLAC format, the STREAMINFO block of the audio file is analyzed to obtain the sampling rate, the sampling bit number and the frame structure information; if the audio file is in DSD format, the DSD_HEADER structure of the audio file is analyzed to obtain the sampling frequency and the bit depth; if the audio file is in MQA format, the MQA_INFO block of the audio file is analyzed to obtain the original sampling rate and calculate the number of folding layers.

[0007] As a further scheme of the present application, the process of pre-processing the cache further comprises: If the audio file is a local audio file, the first 10 seconds of data of the local audio file are cached; if the audio file is a network audio file, the code rate B of the network audio file is obtained, the cache duration T is set to 2s, the cache size C is dynamically adjusted according to the code rate, and the network audio file is cached according to the cache size and the cache duration.

[0008] As a further scheme of the present application, the process of performing spectrum analysis and bit rate fluctuation analysis on the audio signal comprises: The fluctuation amplitude of the bit rate fluctuation of the audio signal is monitored, if the fluctuation amplitude is less than or equal to 5%, it is determined that the audio signal is fixed bit rate encoding, if the fluctuation amplitude is greater than 10%, it is determined that the audio signal is variable bit rate encoding; When the format type of the audio signal is lossless format, and the audio signal is variable bit rate encoding, if the high frequency energy ratio of the audio signal is greater than 0.3, it is determined that the attribute of the audio signal is a lossless file; When the format type of the audio signal is lossless format, and the audio signal is fixed bit rate encoding, the signal spectrum is obtained by performing fast Fourier transform on the audio signal, and the signal spectrum is analyzed every 10 milliseconds; the ratio of the energy of the 20kHz to 22kHz frequency band to the energy of the 18kHz to 20kHz frequency band is obtained, if the ratio is less than 0.05 and the format type of the audio signal is lossless format, it is determined that the attribute of the audio signal is a pseudo-lossless file.

[0009] As a further scheme of the present application, the process of establishing the three-level parameter chain comprises: The third-level parameter chain includes original file parameters, decoding process parameters and output signal parameters; the original file parameters are file header information, the decoding process parameters are format type, sampling rate and channel information fed back by a decoder in real time, and the output signal parameters include output interface type, level and signal-to-noise ratio information.

[0010] As a further scheme of the present application, the generation process of the graphical display content includes: The sampling rate is in a proportional relationship with the height of the column chart, and 44.1 kHz corresponds to one unit of height; 16 bits of sampling are represented by 16 squares; a blue border is used to identify a lossless file, an orange border is used to identify a lossy file, and a red border and a warning icon are used to identify a pseudo-lossless file.

[0011] As a further scheme of the present application, the process of updating the graphical display content in real time includes: When the third-level parameters change, an interrupt signal is generated, and a response is made within 10 milliseconds; the interrupt signal triggers a rendering update process; display rendering of the rendering update process adopts a double-buffering mechanism, and while the current frame is displayed, the next frame of content is pre-rendered based on the changed third-level parameters, so as to realize seamless switching of frame content, and the overall display update is delayed within 1 millisecond.

[0012] As a further scheme of the present application, it further includes accepting an interactive instruction of a user, and the interactive instruction is used to switch a display mode, and the display mode includes a concise mode and a professional mode; the concise mode only displays format type, sampling rate and channel information, and the professional mode completely displays the third-level parameters and all historical third-level parameters.

[0013] The present application has the following beneficial effects: The present application significantly improves the depth of audio format identification, the real-time display and the intuitiveness of user experience in a HIFI music playing system; the present application realizes high-precision extraction of deep parameters including DSD bit depth and MQA folding layer number by using a multi-dimensional identification mechanism combining file header analysis and signal feature verification, and can effectively identify pseudo-lossless audio, and the identification error rate can be reduced to below 1%, solving the problems of incomplete parameter identification and unverifiable authenticity in the prior art; the third-level parameter chain and the interrupt-triggered real-time synchronization architecture constructed by the present application ensure the associated display and dynamic update of the whole link parameters, and the display delay is less than 50 milliseconds, overcoming the defects of parameter display fragmentation and slow response in the traditional scheme; and the present application adopts a display mode combining graphical display and color coding, so that the user can intuitively and quickly understand the audio parameter attributes and differences, greatly improving the professional use experience and system reliability of HIFI users, and being compatible with local and streaming audio sources, and having good applicability and expansibility. BRIEF DESCRIPTION OF DRAWINGS

[0014] The application will be further described below with reference to the drawings.

[0015] Figure 1 is a structural schematic diagram of an audio format identification method of HIFI music playing based on multi-parameter analysis of the application. DETAILED DESCRIPTION

[0016] The technical solutions in the embodiments of the application will be apparently and completely described below with reference to the drawings in the embodiments of the application. Obviously, the described embodiments are only part of the embodiments of the application, rather than all the embodiments. Based on the embodiments in the application, all other embodiments obtained by those skilled in the art without creative labor fall within the protection scope of the application.

[0017] Please refer to Figure 1 The audio format identification method of HIFI music playing based on multi-parameter analysis of the application includes the following steps: Step S1: obtaining audio data and pre-processing the audio data, the pre-processing including integrity check and cache pre-processing; performing multi-dimensional parameter identification on the pre-processed audio data, the process of the multi-dimensional parameter identification including analyzing file header information of each audio file in the audio data, the file header information including format type, sampling rate, sampling bit number, channel information and decoding process parameters.

[0018] Specifically, audio data from local storage or network transmission is obtained, and the audio data is pre-processed, the pre-processing including analyzing file header structure to complete format verification and integrity check, wherein cyclic redundancy check is used to ensure data undamaged, and differential cache pre-processing is performed according to data sources; to ensure the continuity and real-time performance of subsequent identification processing; performing multi-dimensional parameter identification on the pre-processed audio data, the process of the multi-dimensional parameter identification including respectively analyzing file header information of audio files in different formats; for FLAC format, analyzing STREAMINFO block to obtain sampling rate, sampling bit number and frame parameters, for DSD format, analyzing DSD_HEADER to obtain sampling frequency and bit depth, for MQA format, analyzing MQA_INFO block to obtain original sampling rate and calculate folding layer number; the file header information including format type, sampling rate, sampling bit number, channel mode and decoding related metadata.

[0019] As a preferred embodiment of the application, the process of analyzing file header information of the audio file includes: If the audio file is in FLAC format, the STREAMINFO block of the audio file is parsed to obtain the sampling rate, sampling bit number and frame structure information; if the audio file is in DSD format, the DSD_HEADER structure of the audio file is parsed to obtain the sampling frequency and bit depth; if the audio file is in MQA format, the MQA_INFO block of the audio file is parsed to obtain the original sampling rate and calculate the number of folding layers.

[0020] It should be noted that, if the audio file is in FLAC format, the STREAMINFO block of the audio file is parsed to obtain the sampling rate, sampling bit number, minimum or maximum frame size and channel number information, and the obtained parameters are used to accurately identify the audio basic attributes, calculate the compression ratio, and provide frame structure basis for subsequent decoding preprocessing; if the audio file is in DSD format, the DSD_HEADER structure of the audio file is parsed to obtain the sampling frequency and 1-bit depth information, and the bit depth information is used to distinguish the DSD level and accurately represent the high sampling characteristics in the display; if the audio file is in MQA format, the MQA_INFO block of the audio file is parsed to obtain the original sampling rate, and the number of folding layers is calculated according to the current playing sampling rate, and the original sampling rate and the number of folding layers are used to reveal the original quality of the master audio and the encoding process to the user, and enhance the playing transparency.

[0021] As a preferred embodiment of the application, the process of the cache preprocessing further comprises: If the audio file is a local audio file, the first 10 seconds of data of the local audio file is cached; if the audio file is a network audio file, the code rate B of the network audio file is obtained, and the cache time T is set to 2s, the cache size C is dynamically adjusted according to the code rate, and the network audio file is cached according to the cache size and the cache time.

[0022] Specifically, if the audio file is a local audio file, the first 10 seconds of data of the local audio file is cached, and this pre-caching mechanism aims to avoid the influence of the subsequent continuity of real-time identification and decoding caused by the reading delay of the hard disk or storage medium; if the audio file is a network audio file, the current code rate B of the network audio file is obtained in real time, and the minimum cache time T is set to 2s, and the cache size is dynamically calculated and allocated according to the code rate, and this dynamic cache strategy can adapt to different network bandwidth and code rate fluctuation, effectively eliminating the influence of network jitter and transmission delay on the continuity of audio identification and playing, and providing stable and continuous data stream for subsequent multi-dimensional parameter identification and signal analysis.

[0023] Step S2: converting the audio data into an audio signal, performing spectrum analysis and bit rate fluctuation analysis on the audio signal to determine the attribute of the audio signal, the attribute including lossy file, lossless file and pseudo lossless file.

[0024] As a preferred embodiment of the present application, the process of performing spectrum analysis and bit rate fluctuation analysis on the audio signal comprises: Monitoring the fluctuation amplitude of the bit rate fluctuation of the audio signal, if the fluctuation amplitude is less than or equal to 5%, it is determined that the audio signal is fixed bit rate encoded, if the fluctuation amplitude is greater than 10%, it is determined that the audio signal is variable bit rate encoded; When the format type of the audio signal is lossless format, and the audio signal is variable bit rate encoded, if the high frequency energy ratio of the audio signal is greater than 0.3, it is determined that the attribute of the audio signal is a lossless file; When the format type of the audio signal is lossless format, and the audio signal is fixed bit rate encoded, the audio signal is subjected to fast Fourier transform to obtain a signal spectrum, the signal spectrum is analyzed in units of 10 milliseconds, the ratio of the energy of the 20kHz to 22kHz frequency band to the energy of the 18kHz to 20kHz frequency band is obtained, if the ratio is less than 0.05 and the format type of the audio signal is lossless format, it is determined that the attribute of the audio signal is a pseudo lossless file.

[0025] Specifically, the present application effectively overcomes the defect that only relying on file header metadata is easy to be tampered with by a double verification mechanism, significantly improves the accuracy and reliability of the true and false identification of lossless audio, and provides a reliable signal attribute basis for subsequent parameter association and user display.

[0026] Step S3: Establishing a three-level parameter chain, generating graphical display content based on the three-level parameter chain, the graphical display content representing the sampling rate by the height of a column, representing the sampling bit number by the number of squares, representing the channel information by a loudspeaker icon, and using different color coding to distinguish different attributes; in the process of playing the audio signal, the changes of the three-level parameters are monitored in real time, and the graphical display content is updated in real time.

[0027] Specifically, a three-level parameter chain is established between the original file parameters, decoding process parameters and output signal parameters, the original file parameters are derived from the file header analysis result of step S1, the decoding process parameters are obtained in real time from the decoder working state, and the output signal parameters are monitored from the physical output interface; integrated graphical display content is generated based on the three-level parameter chain, in which the sampling rate is directly represented by the height of the column chart, for example, 44.1 kHz corresponds to the reference height, and 192 kHz is proportionally expanded to about 4.3 times; the sampling bit number is visualized by the number of squares, for example, 16 squares are displayed for 16 bits, and 24 squares are displayed for 24 bits; the channel configuration is clearly indicated by the layout of the speaker icon, for example, 2.0, 5.1 and 7.1 channels correspond to different numbers and arrangements of icons; and a blue, orange and red three-color coding system is used to clearly identify the lossless, lossy and pseudo-lossless properties; during audio playback, the changes of any parameter in the three-level parameter chain are monitored in real time through an interrupt triggering mechanism, and the graphical display content is dynamically updated relying on the double-buffering rendering technology, and the update delay is controlled within 50 ms, thereby providing the user with full-link, visual, real-time and accurate audio parameter state feedback.

[0028] As a preferred embodiment of the present application, the establishment process of the three-level parameter chain includes: The three-level parameter chain includes original file parameters, decoding process parameters and output signal parameters; the original file parameters are file header information, the decoding process parameters are the format type, sampling rate and channel information fed back by the decoder in real time, and the output signal parameters include output interface type, level and signal-to-noise ratio information.

[0029] It should be noted that the original file parameters are derived from the analysis result of the file header in step S1, including the format type, sampling rate, sampling bit number, channel mode and high-resolution format specific parameters, which are used to truly reflect the original properties of the audio source; the decoding process parameters are the current processing state fed back by the decoder through the hardware register interface in real time, including the actual decoding format, working sampling rate and channel number, which are used to represent the real processing environment of the audio signal in the decoding stage; and the output signal parameters are obtained by monitoring the output interface circuit, including the output interface type, output level and signal-to-noise ratio, which are used to represent the quality of the final output electrical signal; by establishing the dynamic association and mapping relationship among the three, and storing the full-link parameters and their historical records in the parameter association database.

[0030] As a preferred embodiment of the present application, the generation process of the graphical display content includes: The sampling rate is proportional to the height of the column chart, and 44.1 kHz corresponds to a unit of height; 16-bit sampling is represented by 16 squares; a lossless file is marked with a blue border, a lossy file is marked with an orange border, and a pseudo-lossless file is marked with a red border and a warning icon.

[0031] As a preferred embodiment of the present application, the process of updating the graphical display content in real time includes: When the tertiary parameters change, an interrupt signal is generated, and the response is within 10 milliseconds; the interrupt signal triggers the rendering update process; the display rendering of the rendering update process uses a double buffering mechanism, and while the current frame is displayed, the next frame content is pre-rendered based on the changed tertiary parameters to achieve seamless switching of frame content, and the overall display update is delayed within 1 millisecond.

[0032] Specifically, when any parameter in the tertiary parameter chain changes, a high-priority interrupt signal is generated by the decoding module or the output monitoring circuit through the GPIO interface, and the response to this interrupt is within 10 milliseconds; the interrupt signal immediately triggers the display rendering update process, in which the display rendering uses a double buffering mechanism: while maintaining the display content of the current frame, the next frame of graphical content, including the updated column chart, square array and color coding, is pre-rendered in the background buffer in real time based on the latest tertiary parameters; seamless connection between the front and back frames is achieved through hardware-level switching; this mechanism is synchronized and controlled by a 100MHz clock source shared with the decoding module, ensuring that the overall delay from parameter change to graphical display update is strictly controlled within 1 millisecond.

[0033] As a preferred embodiment of the present application, the method further includes receiving a user's interactive instruction, the interactive instruction being used to switch display modes, the display modes including a concise mode and a professional mode; the concise mode only displays format type, sampling rate and channel information, and the professional mode completely displays the tertiary parameters and all historical tertiary parameters.

[0034] The method further includes receiving a user's interactive instruction through a touch screen, physical buttons or infrared remote control, the interactive instruction being used to dynamically switch display modes during playback, the display modes including a concise mode for ordinary users and a professional mode for HIFI enthusiasts; the concise mode focuses on core parameters and only dynamically displays the format type, sampling rate and channel information of the current audio, with simplified interface elements to reduce interference; the professional mode completely presents all details of the tertiary parameter chain, including original file parameters, decoding process parameters, output signal parameters, and provides historical tertiary parameter records and comparison functions for the last 100 songs, so that users can backtrack or select specific tracks to compare parameters, thereby assisting them in in-depth analysis of sound quality differences, diagnosis of system status or verification of the quality consistency of different sound sources.

[0035] The above detailed description has shown, by way of example, an embodiment of the application. It is specifically contemplated that the application is not limited to the embodiments described herein, but rather, the application is intended to cover all modifications and equivalents thereof falling within the scope of the application.

Claims

1. An audio format identification method for HiFi music playback based on multi-parameter analysis, characterized in that, Includes the following steps: Step S1: Acquire audio data and preprocess the audio data, the preprocessing including integrity verification and cache preprocessing; Multi-dimensional parameter identification is performed on the preprocessed audio data. The process of multi-dimensional parameter identification includes parsing the file header information of each audio file in the audio data. The file header information includes format type, sampling rate, sampling bit depth, channel information and decoding process parameters. Step S2: Convert the audio data into an audio signal, and perform spectrum analysis and bit rate fluctuation analysis on the audio signal to determine the attributes of the audio signal, including lossy file, lossless file and pseudo lossless file; Step S3: Establish a three-level parameter chain, and generate graphical display content based on the three-level parameter chain. The graphical display content uses the height of the bar chart to represent the sampling rate, the number of squares to represent the sampling bit depth, and the speaker icon to represent the channel information. Different attributes are distinguished by different color codes. During the playback of the audio signal, the changes of the three-level parameters are monitored in real time, and the graphical display content is updated in real time.

2. The audio format recognition method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, In step S1, the process of parsing the header information of the audio file includes: If the audio file is in FLAC format, its STREAMINFO block is parsed to obtain the sampling rate, sampling bit depth, and frame structure information; if the audio file is in DSD format, its DSD_HEADER structure is parsed to obtain the sampling frequency and bit depth; if the audio file is in MQA format, its MQA_INFO block is parsed to obtain the original sampling rate and calculate the number of folding layers.

3. The audio format recognition method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, In step S1, the cache preprocessing process further includes: If the audio file is a local audio file, then the first 10 seconds of the local audio file are cached; if the audio file is a network audio file, then the bitrate B of the network audio file is obtained, and the cache duration T=2s is set. The cache size C=BT is dynamically adjusted according to the bitrate B, and the network audio file is cached according to the cache size and cache duration.

4. The audio format recognition method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, In step S2, the process of performing spectrum analysis and bit rate fluctuation analysis on the audio signal includes: The fluctuation amplitude of the bit rate of the audio signal is monitored. If the fluctuation amplitude is less than or equal to 5%, the audio signal is determined to be fixed bit rate encoded. If the fluctuation amplitude is greater than 10%, the audio signal is determined to be variable bit rate encoded. When the audio signal is in a lossless format and is encoded with a variable bit rate, if the high-frequency energy ratio of the audio signal is greater than 0.3, then the audio signal is determined to be a lossless file. When the audio signal is in a lossless format and is encoded at a fixed bit rate, a Fast Fourier Transform is performed on the audio signal to obtain the signal spectrum. The signal spectrum is analyzed every 10 milliseconds. The ratio of the energy in the 20kHz to 22kHz frequency band to the energy in the 18kHz to 20kHz frequency band is obtained. If the ratio is less than 0.05 and the audio signal is in a lossless format, the audio signal is determined to be a pseudo-lossless file.

5. The audio format recognition method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, In step S3, the process of establishing the three-level parameter chain includes: The three-level parameter chain includes original file parameters, decoding process parameters, and output signal parameters; the original file parameters are file header information, the decoding process parameters are format type, sampling rate, and channel information fed back by the decoder in real time, and the output signal parameters include output interface type, level, and signal-to-noise ratio information.

6. The audio format recognition method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, In step S3, the process of generating the graphical display content includes: The sampling rate is directly proportional to the height of the bar chart, with 44.1kHz corresponding to one unit of height; the sampling bit depth is 16 bits, represented by 16 squares; lossless files are marked with a blue border, lossy files with an orange border, and pseudo-lossless files with a red border and a warning icon.

7. The audio format recognition method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, In step S3, the process of updating the graphical display content in real time includes: When the level 3 parameters change, an interrupt signal is generated and a response is made within 10 milliseconds; the interrupt signal triggers the rendering update process; the display rendering of the rendering update process adopts a double buffering mechanism, which pre-renders the content of the next frame based on the changed level 3 parameters while the current frame is being displayed, so as to achieve seamless switching of frame content and control the overall display update delay to within 1 millisecond.

8. The audio format identification method for HiFi music playback based on multi-parameter analysis according to claim 1, characterized in that, It also includes receiving user interaction commands, which are used to switch display modes, including a simplified mode and a professional mode; the simplified mode only displays the format type, sampling rate, and channel information, while the professional mode fully displays the three-level parameters and all historical three-level parameters.

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