Audio Encoder Time Information Segmentation for Seamless Switching
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Existing methods for transmitting audio content over communication networks face challenges in synchronizing audio streams between different devices, leading to potential delays and discontinuities during reproduction, especially when switching between main and backup encoders, as they require precise timing information and external synchronization.
Innovation Solution
Generating time information for each sample within audio frames, rather than just the first sample of a block, allows encoders to synchronize independently, ensuring consistent output streams without the need for external communication or synchronization between devices.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Device complexity
If time information is provided only for the first sample of each block, then the device complexity is reduced, but the measurement precision of time information for individual samples deteriorates
Solution Approach 1:
The block of 192 samples is segmented into multiple groups of 16 samples each. Time information is provided for the first sample of each group rather than only for the first sample of the entire block. This segmentation allows precise time tracking for every 16th sample while maintaining manageable encoder complexity.
Solution Approach 2:
The encoder pre-calculates and inserts time information markers at regular intervals (every 16 samples) within each block during the encoding process. This preliminary placement of time markers eliminates the need for complex real-time time calculation at the decoder, reducing overall system complexity while ensuring precise time information is available where needed.
2Measurement precision
If GPS modules are used to associate time data with each sound sample, then the measurement precision of time information is improved, but the device complexity and cost increase
Solution Approach 1:
The encoder system generates and embeds its own time information internally without requiring external GPS modules. The encoder uses its internal clock and block structure to create time markers at regular intervals, making the system self-sufficient for time stamping while avoiding the complexity and cost of external synchronization hardware.
Solution Approach 2:
The encoder is designed to perform multiple functions: audio encoding, time information generation, and time marker insertion all within a single device. This multi-functionality eliminates the need for separate GPS modules or external time synchronization devices, reducing system complexity while maintaining precise time information.
3Reliability
If communication between encoders is implemented to synchronize processing, then the reliability of synchronized output streams is improved, but the device complexity and ease of operation worsen
Solution Approach 1:
Each encoder independently generates time information based on its own internal processing of audio blocks without requiring communication with other encoders. The time markers are self-generated at regular intervals within each block, making the encoder self-sufficient and eliminating the need for inter-encoder communication infrastructure.
Solution Approach 2:
The encoder changes its operational parameters to generate time information at fixed intervals (every 16 samples) rather than requiring continuous synchronization communication. This parameter-based approach to time stamping ensures reliable time information generation independently, without complex communication protocols between devices.
4Productivity
If packets are transmitted quickly to maintain constant reproduction speed, then the productivity is improved, but the loss of time for buffering increases
Solution Approach 1:
Time information markers are pre-inserted into the audio data stream at regular intervals during encoding. This preliminary time stamping allows the decoder to process packets sequentially without requiring large buffering capacity, as each packet contains its own time reference for immediate processing at the correct reproduction rate.
Solution Approach 2:
The audio stream is segmented into small groups of 16 samples with time markers inserted between groups. This fine-grained segmentation allows for efficient processing with minimal buffering, as the decoder can process each small group independently using its embedded time information, reducing the need for large buffer memory.
Data Source
Figure 1~2
Figure 3
Figure 4
AI summary
The present invention relates to a method for generating a stream of digital packets by means of an apparatus. The apparatus receives data packets sequentially transmitted in an isochronous stream and containing a series of data samples, e.g. sound data. The apparatus transmits a predetermined number of samples in frames that encode a series of samples starting with a first sample. The apparatus determines the value of a number identifying each sample contained in the input stream. The value of the number of said first sample in each frame is a multiple, possibly zero, of the predetermined number of samples. In this manner, regardless of the encoder that produces the frames, the latter contain the same samples without any shift from one encoder to the next. The switching between the output streams transmitted by a plurality of encoders can be carried out without any artifacts due to the transmission of the same samples.