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

Abstract

The application relates to the technical field of audio format identification, and specifically discloses an audio format identification method for HIFI music playing based on multi-parameter analysis, which comprises the following steps: step S1: pre-processing and multi-dimensional parameter identification of audio data, basic and decoding parameters such as format, sampling rate, bit number and sound channel are obtained by analyzing a file header; step S2: converting the audio data into signals and performing spectrum and bit rate fluctuation analysis to accurately determine whether the audio is lossy, lossless or pseudo-lossless; and step S3: establishing a three-level parameter chain of an original file, a decoding process and an output signal, generating an intuitive graphical display interface according to the three-level parameter chain, taking the height of a columnar chart, the number of blocks and a loudspeaker icon to represent sampling rate, sampling bit number and sound channel configuration respectively, distinguishing audio properties by color, monitoring parameter changes in real time and updating the display in the playing process, and ensuring that a user can comprehensively, timely and intuitively master the audio state.

Description

Technical Field

[0001] This invention relates to the field of audio format recognition technology, and more specifically to an audio format recognition method for HiFi music playback based on multi-parameter analysis. Background Technology

[0002] With the rapid development of digital audio technology, Hi-Fi audio formats have evolved from early lossy compression formats to lossless formats and even high-resolution formats. Their parameter systems have also become increasingly complex, including multiple dimensions such as sampling rate, bit depth, channel mode, and encoding attributes. These parameters directly affect audio reproduction quality and are important criteria for users to evaluate the performance of playback systems.

[0003] Existing audio format recognition and display solutions mainly consist of software playback systems and hardware decoding devices. Software solutions typically extract basic parameters by parsing file header metadata, but they cannot identify the special parameters of high-resolution formats and lack the ability to verify the authenticity of the signal. Hardware solutions, while capable of detecting the basic parameters of the decoded signal, cannot correlate them with the original file information, resulting in fragmented display content and a lack of end-to-end parameter correlation. Furthermore, existing technologies suffer from display latency and high recognition error rates during dynamic switching, failing to meet the advanced requirements of Hi-Fi users for parameter transparency and real-time performance. Summary of the Invention

[0004] The purpose of this invention is to provide an audio format identification method for HiFi music playback based on multi-parameter analysis, and to solve the following technical problems.

[0005] The objective of this invention can be achieved through the following technical solutions:

[0006] An audio format identification method for HiFi music playback based on multi-parameter analysis includes the following steps:

[0007] Step S1: Acquire audio data and preprocess the audio data, the preprocessing including integrity verification and cache preprocessing; perform multi-dimensional parameter identification on the preprocessed audio data, the multi-dimensional parameter identification process includes parsing the file header information of each audio file in the audio data, the file header information including format type, sampling rate, sampling bit depth, channel information and decoding process parameters;

[0008] 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;

[0009] 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.

[0010] As a further aspect of the present invention: the process of parsing the header information of an audio file includes:

[0011] 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.

[0012] As a further aspect of the present invention: the cache preprocessing process further includes:

[0013] 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, and the network audio file is cached according to the cache size and cache duration.

[0014] As a further aspect of the present invention, the process of performing spectrum analysis and bit rate fluctuation analysis on audio signals includes:

[0015] 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.

[0016] 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.

[0017] 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.

[0018] As a further aspect of the present invention: the process of establishing the three-level parameter chain includes:

[0019] 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.

[0020] As a further aspect of the present invention: the process of generating the graphical display content includes:

[0021] 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.

[0022] As a further aspect of the present invention: the process of updating the graphical display content in real time includes:

[0023] 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.

[0024] As a further aspect of the present invention, it also includes accepting user interaction commands, the interaction commands being used to switch display modes, the 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.

[0025] The beneficial effects of this invention are:

[0026] This invention significantly improves the depth of audio format recognition, the real-time performance of display, and the intuitiveness of user experience in Hi-Fi music playback systems. Through a multi-dimensional recognition mechanism that integrates file header parsing and signal feature verification, it achieves high-precision extraction of deep parameters, including DSD bit depth and MQA folding layers, and effectively identifies pseudo-lossless audio, reducing the error rate to below 1%. This solves the problems of incomplete parameter recognition and inability to verify authenticity in existing technologies. The three-level parameter chain and its interrupt-triggered real-time synchronization architecture constructed in this invention ensure full-link parameter association display and dynamic updates, with a display latency of less than 50 milliseconds, overcoming the fragmented parameter display and slow response defects of traditional solutions. Furthermore, this invention adopts a display method combining graphical and color coding, enabling users to intuitively and quickly understand the attributes and differences of audio parameters, greatly improving the professional user experience and system reliability for Hi-Fi users. It is compatible with local and streaming media audio sources and has good applicability and scalability. Attached Figure Description

[0027] The invention will now be further described with reference to the accompanying drawings.

[0028] Figure 1 This is a schematic diagram of the audio format recognition method for HiFi music playback based on multi-parameter analysis according to the present invention. Detailed Implementation

[0029] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0030] Please see Figure 1 As shown, this invention is an audio format identification method for HiFi music playback based on multi-parameter analysis, comprising the following steps:

[0031] Step S1: Acquire audio data and preprocess the audio data, including integrity verification and cache preprocessing; perform multi-dimensional parameter identification on the preprocessed audio data, the multi-dimensional parameter identification process includes parsing the file header information of each audio file in the audio data, the file header information including format type, sampling rate, sampling bit depth, channel information and decoding process parameters.

[0032] Specifically, the process involves acquiring audio data from local storage or network transmission and preprocessing it. This preprocessing includes parsing the file header structure to perform format and integrity verification, using cyclic redundancy check to ensure data integrity, and performing differentiated caching preprocessing based on the data source to ensure the continuity and real-time performance of subsequent recognition processing. Multi-dimensional parameter recognition is then performed on the preprocessed audio data. This process includes parsing the file header information of different audio file formats; for FLAC format, parsing the STREAMINFO block to obtain the sampling rate, bit depth, and frame parameters; for DSD format, parsing the DSD_HEADER to obtain the sampling frequency and bit depth; and for MQA format, parsing the MQA_INFO block to obtain the original sampling rate and calculate the number of folds. The file header information includes format type, sampling rate, bit depth, channel mode, and decoding-related metadata.

[0033] In a preferred embodiment of the present invention, the process of parsing the header information of an audio file includes:

[0034] 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.

[0035] It should be noted that if the audio file is in FLAC format, its STREAMINFO block is parsed to obtain information on the sampling rate, bit depth, minimum or maximum frame size, and number of channels. The obtained parameters are used to accurately identify the basic audio attributes, calculate the compression ratio, and provide a frame structure basis for subsequent decoding preprocessing. If the audio file is in DSD format, its DSD_HEADER structure is parsed to obtain the sampling frequency and 1-bit bit depth information. The bit depth information is used to distinguish DSD levels and accurately represent its high sampling characteristics in the display. If the audio file is in MQA format, its MQA_INFO block is parsed to obtain the original sampling rate, and the number of folding layers is calculated based on the current playback sampling rate. 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, enhancing playback transparency.

[0036] In a preferred embodiment of the present invention, the cache preprocessing process further includes:

[0037] 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, and the network audio file is cached according to the cache size and cache duration.

[0038] Specifically, if the audio file is a local audio file, the first 10 seconds of data of the local audio file are cached. This pre-caching mechanism aims to avoid the impact of hard disk or storage medium read delays on the continuity of subsequent real-time recognition and decoding. If the audio file is a network audio file, the current bitrate B of the network audio file is obtained in real time, and the minimum cache duration T=2s is set. The cache size is dynamically calculated and allocated according to the bitrate. This dynamic caching strategy can adapt to different network bandwidth and bitrate fluctuations, effectively eliminating the impact of network jitter and transmission delay on the continuity of audio recognition and playback, and providing a stable and continuous data stream for subsequent multi-dimensional parameter recognition and signal analysis.

[0039] 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.

[0040] In a preferred embodiment of the present invention, the process of performing spectrum analysis and bit rate fluctuation analysis on the audio signal includes:

[0041] 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.

[0042] 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.

[0043] 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.

[0044] Specifically, this invention effectively overcomes the shortcomings of relying solely on file header metadata being easily tampered with through a dual verification mechanism, significantly improving the accuracy and reliability of lossless audio authenticity identification, and providing a reliable signal attribute basis for subsequent parameter association and user display.

[0045] 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.

[0046] 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 parsing result in step S1, the decoding process parameters are obtained in real time from the decoder's working status, and the output signal parameters are monitored from the physical output interface. Based on the three-level parameter chain, integrated graphical display content is generated, where the sampling rate is intuitively represented by the height of a bar chart; for example, 44.1kHz corresponds to a reference height, and 192kHz is scaled up to approximately 4.3 times. The sampling bit depth is visualized by the number of squares; for example, 16-bit displays 16 squares. The system displays 24 blocks in a 24-bit format. Channel configuration is clearly indicated by speaker icon layout; for example, 2.0, 5.1, and 7.1 channels correspond to different numbers and arrangements of icons. A blue, orange, and red three-color encoding system is used to clearly identify lossless, lossy, and pseudo-lossless attributes. During audio playback, an interrupt trigger mechanism monitors changes in any parameter in the three-level parameter chain in real time, and relies on double-buffered rendering technology to dynamically update the graphical display content, with update latency controlled within 50ms. This provides users with end-to-end, visual, real-time, and accurate audio parameter status feedback.

[0047] In a preferred embodiment of the present invention, the process of establishing the three-level parameter chain includes:

[0048] 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.

[0049] It should be noted that the original file parameters are derived from the parsing results of the file header in step S1, including format type, sampling rate, sampling bit depth, channel mode, and high-resolution format-specific parameters, which are used to truly reflect the original attributes of the audio source; the decoding process parameters are the current processing status fed back by the decoder in real time through the hardware register interface, including the actual decoding format, working sampling rate, and number of channels, which are used to characterize the actual processing environment of the audio signal in the decoding stage; the output signal parameters are obtained by monitoring the output interface circuit, including output interface type, output level, and signal-to-noise ratio, which are used to characterize the quality of the final output electrical signal; by establishing a dynamic correlation and mapping relationship between the three, the full-link parameters and their historical records are stored in the parameter correlation database.

[0050] In a preferred embodiment of the present invention, the process of generating the graphical display content includes:

[0051] 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.

[0052] In a preferred embodiment of the present invention, the process of updating the graphical display content in real time includes:

[0053] 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.

[0054] Specifically, when any parameter in the three-level parameter chain changes, the decoding module or output monitoring circuit generates a high-priority interrupt signal through the GPIO interface and responds to this interrupt within 10 milliseconds. The interrupt signal immediately triggers the display rendering update process, in which the display rendering adopts a double buffering mechanism: while maintaining the display content of the current frame, the next frame of graphical content, including the updated bar chart, block array, and color encoding, is pre-rendered in real time in the background buffer based on the latest three-level parameters; seamless connection between the previous and next frames is achieved through hardware-level switching; this mechanism, combined with the 100MHz clock source shared with the decoding module for synchronous control, ensures that the overall delay from parameter change to graphical display update is strictly controlled within 1 millisecond.

[0055] In a preferred embodiment of the present invention, the method further includes receiving user interaction instructions, the interaction instructions being used to switch display modes, the 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.

[0056] This method also includes receiving user interaction commands via touchscreen, physical buttons, or infrared remote control. These commands are used to dynamically switch display modes during playback. The display modes include a simplified mode for general users and a professional mode for HiFi enthusiasts. The simplified mode focuses on core parameters, dynamically displaying only the current audio format, sampling rate, and channel information, with simplified interface elements to reduce interference. The professional mode fully presents all the details of the three-level parameter chain, including original file parameters, decoding process parameters, and output signal parameters. It also provides a historical three-level parameter record and comparison function for the most recent 100 songs. Users can backtrack via the timeline or select specific tracks to compare parameters, thereby assisting them in deeply analyzing sound quality differences, diagnosing system status, or verifying the quality consistency of different audio sources.

[0057] The foregoing has provided a detailed description of one embodiment of the present invention, but this description is merely a preferred embodiment and should not be construed as limiting the scope of the invention. All equivalent variations and modifications made within the scope of the present invention should still fall within the scope of the invention.

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. 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; 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. 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.

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, 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.

Images