Adaptive error correction method and system in broadcast digital audio transmission

Abstract

The invention relates to the technical field of digital audio transmission, in particular to a self-adaptive error correction method and system in broadcast digital audio transmission, and the method comprises the steps: obtaining audio data in a broadcast digital audio transmission network; obtaining an error rate in the audio data transmission process; selecting a matched error correction mode based on the error rate; performing error correction processing on the audio data according to the error correction mode; audio data subjected to error correction processing are sent through a broadcast transmission channel, firstly, through real-time error rate monitoring and self-adaptive error correction mode selection, the method can quickly respond to changes of channel conditions, the most suitable error correction strategy is selected under different error rate levels, and the anti-interference capacity and adaptability of a system are greatly improved. This is particularly effective in mobile reception and complex urban environments, which can significantly reduce audio quality degradation and transmission disruption.

Description

Technical Field

[0001] The present invention relates to the technical field of digital audio transmission, and more particularly to an adaptive error correction method and system in broadcast digital audio transmission. Background Art

[0002] With the rapid development of digital technology, broadcast digital audio transmission has become an important part of modern broadcasting systems. However, in actual broadcasting environments, signal transmission often faces various complex interferences and fading, which poses serious challenges to audio quality and transmission reliability.

[0003] Traditional broadcast digital audio transmission systems usually use fixed error correction coding schemes, such as Reed-Solomon codes or convolutional codes. These methods can provide a certain degree of error protection in a stable transmission environment, but their performance is often unsatisfactory in actual broadcast scenarios with changing channel conditions. Especially in mobile reception, urban environments with severe multipath propagation, or in severe weather conditions, fixed error correction schemes are difficult to adapt to the rapid changes in channel conditions, resulting in a significant decrease in audio quality or even transmission interruption.

[0004] Some improved methods attempt to introduce adaptive mechanisms to dynamically adjust encoding parameters according to channel conditions. However, these methods can usually only switch between a limited number of preset modes and lack the ability to finely respond to channel conditions. At the same time, they often ignore the characteristics of the audio data itself and fail to adopt differentiated protection strategies for audio content of different importance.

[0005] In addition, most existing error correction methods focus on the coding layer, while ignoring the optimization of the transmission layer. In complex broadcast environments, error correction relying solely on the coding layer often cannot fully cope with problems such as multipath effects and frequency selective fading, limiting the overall performance of the system.

[0006] Therefore, there is an urgent need for an adaptive error correction method that can comprehensively consider channel conditions, audio data characteristics, and transmission technology to improve the quality and reliability of broadcast digital audio transmission. Summary of the invention

[0007] The present invention is an innovative solution to the above technical problems. The present invention provides an adaptive error correction method and system in broadcast digital audio transmission, which effectively solves the problems existing in the prior art through innovations in real-time monitoring of error rate, dynamic selection of error correction mode, consideration of audio data characteristics, and optimization of transmission technology.

[0008] The present invention provides an adaptive error correction method in broadcast digital audio transmission, comprising:

[0009] The acquisition steps include:

[0010] Acquire audio data in a broadcast digital audio transmission network;

[0011] Obtaining an error rate during the audio data transmission process;

[0012] Processing steps include:

[0013] Based on the error rate, selecting a matching error correction mode;

[0014] According to the error correction mode, performing error correction processing on the audio data;

[0015] Output steps include:

[0016] The audio data that has been processed for error correction is sent through a broadcast transmission channel.

[0017] Preferably, the obtaining the error rate in the audio data transmission process specifically includes:

[0018] Count the number of bits with transmission errors within a preset time window;

[0019] The error rate is obtained by calculating the ratio of the number of bits with transmission errors to the total number of bits transmitted.

[0020] Preferably, the selecting a matching error correction mode based on the error rate specifically includes:

[0021] When the error rate is lower than a first preset threshold, selecting a coding error correction mode;

[0022] When the error rate is higher than the first preset threshold and lower than the second preset threshold, selecting a mixed error correction mode of coding and interleaving;

[0023] When the error rate is higher than the second preset threshold, a segmented coding and interleaving mixed error correction mode is selected.

[0024] Preferably, the coding error correction mode includes:

[0025] Extracting a check digit from the audio data;

[0026] Error correction coding is performed on the audio data to generate coded audio data.

[0027] Preferably, the coding and interleaving mixed error correction mode includes:

[0028] Splitting the audio data into a plurality of data segments;

[0029] Applying an (n, k) erasure code to each of the data segments, where n>k;

[0030] Add parity bits between adjacent data segments.

[0031] Preferably, the segmented coding and interleaved mixed error correction mode includes:

[0032] Splitting the audio data into a plurality of data frames;

[0033] Performing encoding and interleaving mixed error correction processing on each of the data frames;

[0034] Interleave between processed data frames.

[0035] As a preference, it also includes:

[0036] Based on the characteristics of the audio data, setting different error rate thresholds for data at different content locations;

[0037] According to the content location of the data, a corresponding error rate threshold is selected to select an error correction mode.

[0038] Preferably, sending the audio data after error correction through a broadcast transmission channel specifically includes:

[0039] Modulating the error-corrected audio data using an orthogonal frequency division multiplexing (OFDM) technique;

[0040] The modulated signal is sent through a broadcast transmission channel.

[0041] As a preference, it also includes:

[0042] monitoring a changing trend of the error rate;

[0043] When the error rate continues to exceed a preset threshold, a maintenance mode is triggered;

[0044] In the maintenance mode, the audio signal transmission link is reestablished.

[0045] An adaptive error correction system in broadcast digital audio transmission for implementing the method comprises:

[0046] An error rate acquisition module, used to acquire the error rate during the audio data transmission process in the broadcast digital audio transmission network;

[0047] An error correction mode selection module, used for selecting a matching error correction mode based on the error rate;

[0048] An error correction processing module, used for performing error correction processing on the audio data according to the error correction mode;

[0049] A data sending module, used for sending the audio data after error correction through a broadcast transmission channel;

[0050] Wherein, the error correction mode selection module includes:

[0051] A coding error correction unit, configured to perform coding error correction when the error rate is lower than a first preset threshold;

[0052] a hybrid error correction unit, configured to perform coding and interleaving hybrid error correction when the error rate is between the first preset threshold and the second preset threshold;

[0053] The segmented error correction unit is used to perform segmented coding and interleaving mixed error correction when the error rate is higher than the second preset threshold.

[0054] The beneficial effects of the present invention are mainly reflected in the following aspects:

[0055] The method of the present invention has significant technical effects. First, through real-time error rate monitoring and adaptive error correction mode selection, the method can quickly respond to changes in channel conditions and select the most appropriate error correction strategy at different error rate levels, greatly improving the anti-interference ability and adaptability of the system. This is particularly effective in mobile reception and complex urban environments, and can significantly reduce audio quality degradation and transmission interruptions.

[0056] Secondly, the present invention takes into account the characteristics of audio data and sets differentiated protection strategies for data in different content locations. This fine-grained error correction mechanism ensures that key audio information is better protected and optimizes the overall sound quality under limited system resources.

[0057] Furthermore, the present invention introduces OFDM technology in the transmission layer, which effectively addresses the multipath effect and frequency selective fading problems, which not only improves the spectrum utilization efficiency, but also enhances the stability of the system in complex electromagnetic environments.

[0058] Finally, the multi-level error correction mechanism and maintenance mode provided by the present invention further enhance the robustness of the system. Even under extremely harsh transmission conditions, the system can maintain service by automatically switching to a stronger error correction mode or rebuilding the transmission link.

[0059] In summary, the method of the present invention has made significant progress in improving the quality and reliability of broadcast digital audio transmission. It can not only adapt to various complex broadcast environments, but also achieve optimal performance under limited system resources. This has important practical significance and application value for improving the quality of broadcast services, improving user experience, and promoting the development of digital audio broadcasting technology. BRIEF DESCRIPTION OF THE DRAWINGS

[0060] Figure 1 The figure is a flow chart of the method of the present invention.

[0061] Figure 2It is a logic block diagram of the error rate acquisition module of the present invention.

[0062] Figure 3 It is a logic block diagram of the error correction mode selection module of the present invention.

[0063] Figure 4 It is a logic block diagram of the error correction processing module of the present invention.

[0064] Figure 5 It is a logic block diagram of the data sending module of the present invention. DETAILED DESCRIPTION

[0065] Please refer to Figure 1-5 The present invention provides an adaptive error correction method and system in broadcast digital audio transmission. The method can effectively improve the quality and reliability of broadcast digital audio transmission, and is particularly suitable for broadcast environments with variable channel conditions.

[0066] First, the method of the present invention includes an acquisition step, a processing step and an output step. In the acquisition step, the system acquires audio data in a broadcast digital audio transmission network and simultaneously acquires the error rate of the audio data during the transmission process. The error rate here is a key indicator for measuring transmission quality and directly affects the subsequent error correction strategy selection.

[0067] In the processing step, the method selects a matching error correction mode based on the obtained error rate. This adaptive selection mechanism is one of the core innovations of the present invention, and the error correction strategy can be dynamically adjusted according to the real-time channel conditions. After selecting the appropriate error correction mode, the system will perform corresponding error correction processing on the audio data. This processing can effectively reduce transmission errors and improve audio quality.

[0068] Finally, in the output step, the error-corrected audio data is sent out through the broadcast transmission channel. This ensures that the final transmitted audio data has higher quality and reliability.

[0069] Furthermore, the present invention describes in detail a method for obtaining an error rate. Specifically, the system counts the number of bits of transmission errors within a preset time window, and then calculates the ratio of these error bits to the total number of transmission bits, thereby obtaining an error rate. This method can quickly and accurately reflect the current transmission quality status. Preferably, the size of the time window can be adjusted according to the actual application scenario, for example, it can be set to between 100ms and 1s. A shorter time window can respond to channel changes more quickly, while a longer time window can obtain more stable statistical results.

[0070] An important feature of the present invention is to select a matching error correction mode based on the error rate. Specifically, when the error rate is lower than a first preset threshold, the system selects a coding error correction mode; when the error rate is higher than the first preset threshold but lower than a second preset threshold, the system selects a coding and interleaving mixed error correction mode; when the error rate is higher than the second preset threshold, the system selects a segmented coding and interleaving mixed error correction mode.

[0071] This multi-level error correction mechanism can adopt the most suitable error correction strategy according to different error rate situations, which not only ensures the error correction effect but also avoids unnecessary computing overhead. For example, the first preset threshold can be set to 1%, and the second preset threshold can be set to 5%. The selection of these thresholds is based on actual broadcast audio transmission experience and can be fine-tuned according to specific application scenarios.

[0072] It is worth noting that the method of the present invention has shown significant advantages in practical applications. For example, in a field broadcast test, when the vehicle-mounted receiving equipment moves in a mountainous area, the channel conditions change frequently. The method can quickly adapt to these changes and select the most appropriate error correction mode at different error rate levels, ensuring the continuity and quality of audio transmission.

[0073] In general, the adaptive error correction method provided by the present invention greatly improves the anti-interference ability and reliability of broadcast digital audio transmission by real-time monitoring of the error rate and dynamically selecting the error correction mode. This has important practical significance for improving the quality of broadcasting, especially in complex and changeable transmission environments. In a preferred embodiment of the present invention, the specific implementation of the coding error correction mode includes extracting check bits from audio data, performing error correction coding on the audio data, and generating encoded audio data. This method can effectively detect and correct random errors in the transmission process.

[0074] Specifically, the extraction of the check bit can adopt a cyclic redundancy check (CRC) method. For example, a CRC-32 algorithm can be used, which can detect all burst errors less than 32 bits. For error correction coding, the present invention preferably adopts Reed-Solomon coding. Reed-Solomon coding has a strong error correction capability and is particularly suitable for resisting burst errors, which are very common in broadcast environments.

[0075] In practical applications, appropriate encoding parameters can be selected according to the characteristics of the audio data. For example, for CD-quality audio with a sampling rate of 44.1kHz and 16-bit quantization, RS (255, 223) encoding can be selected, that is, each 255-byte encoding block contains 223 bytes of data and 32 bytes of checksum. This configuration can correct up to 16 bytes of errors in each encoding block, or detect 32 bytes of errors.

[0076] Another embodiment of the present invention relates to a hybrid error correction mode of coding and interleaving. In this mode, the system first divides the audio data into multiple data segments, and then applies (n,k) erasure codes to each data segment, where n>k. Finally, parity bits are added between adjacent data segments. This hybrid method can not only correct random errors, but also effectively cope with burst errors.

[0077] Preferably, a (255, 223) Reed-Solomon code can be selected as an erasure code. The size of the data segment can be determined according to the actual audio frame size, for example, it can be set to 1024 bytes. In this way, each data segment can be encoded into 1152 bytes (including 128 bytes of checksum). Between adjacent encoded data segments, an 8-bit parity bit can be inserted to further enhance the error correction capability.

[0078] A significant advantage of this method is that by properly setting the data segment size and erasure code parameters, a good balance can be achieved between error correction capability and coding efficiency. For example, in an experiment, a system using this hybrid error correction mode can still maintain good audio quality under a 5% bit error rate environment, while the traditional single encoding method will show a significant drop in sound quality under the same conditions.

[0079] In the case of higher error rates, the present invention provides a segmented coding and interleaved hybrid error correction mode. This mode first divides the audio data into multiple data frames, then encodes and interleaves each data frame for hybrid error correction, and finally interleaves the processed data frames. This multi-level error correction structure can effectively deal with various types of errors, including long-term burst errors.

[0080] In actual implementation, the audio data can be divided into frames of fixed size, such as 4096 bytes per frame. Each frame is first Reed-Solomon encoded and then block interleaved. The depth of block interleaving can be set to 16, which means that every 16 encoded symbols will be scattered to 16 different positions. Finally, interleaving is performed again at the frame level to mix the data of adjacent frames together. This method can disperse burst errors into multiple small random errors, greatly improving the system's anti-interference ability.

[0081] The method of the present invention also takes into account the characteristics of audio data and sets different error rate thresholds for data at different content locations. This is because in an audio stream, some parts (such as key frames) have a greater impact on the overall quality. Therefore, the system will select the corresponding error rate threshold to select the error correction mode based on the content location of the data.

[0082] For example, for MPEG audio streams, the error rate threshold of I frames (key frames) can be set lower, such as 0.5%, while the thresholds of P frames and B frames can be relatively high, such as 1% and 2%. This differentiated threshold setting can ensure that key audio information is better protected, thereby optimizing the overall sound quality under limited computing resources.

[0083] In general, the error correction modes and strategies provided by the present invention form a comprehensive adaptive error correction system. By real-time monitoring of the error rate, dynamically selecting the most suitable error correction mode, and considering the characteristics of the audio data, the present method can provide high-quality, high-reliability digital audio transmission services in various complex broadcast environments. This has important practical application value for improving broadcast quality, especially under challenging conditions such as mobile reception and bad weather. In an important embodiment of the present invention, when the audio data that has been processed for error correction is sent through a broadcast transmission channel, orthogonal frequency division multiplexing (OFDM) technology is used for modulation. The use of OFDM technology is an important innovation of the present method at the transmission level, which can effectively deal with the multipath effect and frequency selective fading problems in the broadcast environment.

[0084] Specifically, OFDM technology divides a broadband signal into multiple orthogonal narrowband signals for parallel transmission. In a preferred embodiment of the present invention, an OFDM system with a 2048-point FFT can be selected, which means that the total bandwidth is divided into 2048 subcarriers. Among them, 1705 subcarriers are used for data transmission, and the remaining subcarriers are used as guard intervals and pilots. This configuration can achieve an effective data transmission rate of about 19Mbps within a 6MHz channel bandwidth, which is sufficient to support high-quality digital audio broadcasting.

[0085] In practical applications, the selection of OFDM parameters needs to be optimized according to the specific broadcast environment. For example, in urban environments, multipath delay spread is usually large, and a longer cyclic prefix, such as 1 / 4 symbol period, can be selected. In suburban or rural areas, a shorter cyclic prefix, such as 1 / 8 or 1 / 16 symbol period, can be selected to improve spectrum efficiency.

[0086] The method of the present invention also includes monitoring the error rate change trend. When the error rate continues to exceed a preset threshold, the system will trigger to enter a maintenance mode, in which the audio signal transmission link is reestablished. This mechanism can effectively cope with drastic changes in channel conditions or system failures.

[0087] Preferably, multiple warning levels can be set. For example, when the error rate exceeds 10% for 30 consecutive seconds, the system enters the first alert state and starts to increase the redundancy of the error correction code. If the error rate exceeds 20% for 60 consecutive seconds, it enters the second alert state, at which time the system will try to switch to a backup frequency or adjust the transmission power. Only when the error rate exceeds 30% for 120 consecutive seconds will the system fully enter maintenance mode and re-establish the transmission link.

[0088] This hierarchical response mechanism can ensure transmission quality while avoiding frequent service interruptions due to transient interference. In actual broadcast systems, this feature significantly improves system stability and user experience. For example, in a cross-city mobile reception test, the system successfully coped with multiple signal quality fluctuations caused by terrain changes and electromagnetic interference, maintaining the continuity of audio transmission.

[0089] Finally, the present invention also provides an adaptive error correction system in broadcast digital audio transmission for implementing the above method. The system includes multiple functional modules, each module is responsible for a specific step or function in the method.

[0090] The error rate acquisition module 1 is used to obtain the error rate during the audio data transmission process in the broadcast digital audio transmission network. The module can obtain the error rate by analyzing the received data packets in real time and calculating the proportion of error bits. Preferably, the module can adopt a sliding window technology, for example, updating the error rate data every 100ms, so as to timely reflect the changes in channel conditions.

[0091] The error correction mode selection module 2 selects a matching error correction mode based on the error rate data provided by the error rate acquisition module 1. The module contains a decision algorithm that can quickly switch different error correction strategies based on a preset threshold and the trend of the current error rate.

[0092] The error correction processing module 3 performs specific error correction processing on the audio data according to the output of the error correction mode selection module 2. This module integrates multiple error correction algorithms, such as Reed-Solomon coding, interleaving, etc., and can flexibly combine these algorithms to achieve the best error correction effect.

[0093] The data transmission module 4 is responsible for transmitting the error-corrected audio data through the broadcast transmission channel. This module not only implements OFDM modulation, but also includes functions such as power control and spectrum shaping to ensure that the transmitted signal complies with relevant broadcast standards and regulatory requirements.

[0094] In particular, the error correction mode selection module 2 also includes several important sub-units: the coding error correction unit 21 is used to perform basic coding error correction when the error rate is low; the mixed error correction unit 22 performs mixed error correction of coding and interleaving when the error rate is at a medium level; and the segmented error correction unit 23 performs more complex mixed error correction of segmented coding and interleaving when the error rate is high.

[0095] This modular design makes the system easy to expand and maintain. For example, if a new error correction algorithm needs to be added in the future, it is only necessary to add a new functional unit to the corresponding module without changing the architecture of the entire system. At the same time, this design also facilitates parallel processing and load balancing of the system, which is conducive to improving the overall processing efficiency.

[0096] In general, the adaptive error correction system provided by the present invention can effectively cope with various complex transmission situations in the broadcasting environment through its flexible modular structure and advanced algorithms, and provide users with high-quality and high-reliability digital audio broadcasting services. This has important practical significance and application value for improving the overall technical level of the broadcasting industry and promoting the popularization and development of digital audio broadcasting.

[0097] It should be noted that the above description is only a preferred embodiment of the present invention and is not intended to limit the present invention. Any modification, replacement, and improvement made within the principles of the present invention should be included in the protection scope of the present invention.

Claims

1. An adaptive error correction method in broadcast digital audio transmission, characterized in that include: The acquisition steps include: Acquire audio data in a broadcast digital audio transmission network; Obtaining an error rate during the audio data transmission process; Processing steps include: Based on the error rate, selecting a matching error correction mode; According to the error correction mode, performing error correction processing on the audio data; Output steps include: The audio data that has been processed for error correction is sent through a broadcast transmission channel.

2. The method according to claim 1, characterized in that The obtaining of the error rate in the audio data transmission process specifically includes: Count the number of bits with transmission errors within a preset time window; The error rate is obtained by calculating the ratio of the number of bits with transmission errors to the total number of bits transmitted.

3. The method according to claim 1, characterized in that: The selecting a matching error correction mode based on the error rate specifically includes: When the error rate is lower than a first preset threshold, selecting a coding error correction mode; When the error rate is higher than the first preset threshold and lower than the second preset threshold, selecting a mixed error correction mode of coding and interleaving; When the error rate is higher than the second preset threshold, a segmented coding and interleaving mixed error correction mode is selected.

4. The method according to claim 3, characterized in that The coding error correction mode includes: Extracting a check digit from the audio data; Error correction coding is performed on the audio data to generate coded audio data.

5. The method according to claim 3, characterized in that: The coding and interleaving mixed error correction mode includes: Splitting the audio data into a plurality of data segments; Applying an (n, k) erasure code to each of the data segments, where n>k; Add parity bits between adjacent data segments.

6. The method according to claim 3, characterized in that The segmented coding and interleaved mixed error correction mode includes: Splitting the audio data into a plurality of data frames; Performing encoding and interleaving mixed error correction processing on each of the data frames; Interleave between processed data frames.

7. The method according to claim 1, characterized in that Also includes: Based on the characteristics of the audio data, setting different error rate thresholds for data at different content locations; According to the content location of the data, a corresponding error rate threshold is selected to select an error correction mode.

8. The method according to claim 1, characterized in that: The sending of the error-corrected audio data through a broadcast transmission channel specifically includes: Modulating the error-corrected audio data using orthogonal frequency division multiplexing (OFDM) technology; The modulated signal is sent through a broadcast transmission channel.

9. The method according to claim 1, characterized in that: Also includes: monitoring a changing trend of the error rate; When the error rate continues to exceed a preset threshold, a maintenance mode is triggered; In the maintenance mode, the audio signal transmission link is reestablished.

10. An adaptive error correction system in broadcast digital audio transmission for implementing the method according to any one of claims 1 to 9, characterized in that: include: An error rate acquisition module, used to acquire the error rate during the audio data transmission process in the broadcast digital audio transmission network; An error correction mode selection module, used for selecting a matching error correction mode based on the error rate; An error correction processing module, used for performing error correction processing on the audio data according to the error correction mode; A data sending module, used for sending the audio data after error correction through a broadcast transmission channel; Wherein, the error correction mode selection module includes: A coding error correction unit, configured to perform coding error correction when the error rate is lower than a first preset threshold; a hybrid error correction unit, configured to perform coding and interleaving hybrid error correction when the error rate is between the first preset threshold and the second preset threshold; The segmented error correction unit is used to perform segmented coding and interleaving mixed error correction when the error rate is higher than the second preset threshold.

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