Sound Signal Decoding Method for Dual Network Delay Management
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Solution Overview
Problem
Existing techniques for decoding sound signals in terminal apparatuses connected to two communication networks with different priority levels face challenges in achieving high sound quality without significantly increasing delay time, leading to discomfort during two-way conversations.
Innovation Solution
A sound signal receiving and decoding method that determines whether the average difference in reception times of code strings from two communication networks is within a predetermined time limit, and based on this, selects either the monaural code from the first communication line or a corresponding extended code from the second communication line to produce decoded digital sound signals of multiple channels.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If the time limit is set longer to obtain high sound quality in many frames, then the percentage of frames with high quality decoded signals increases, but the delay time becomes so long that a sense of discomfort occurs during two-way conversation
Solution Approach 1:
The patent applies dynamics by making the time limit adjustable rather than fixed. The time limit is dynamically changed based on the reception state of packets from the second communication line. When packet reception is good, a longer time limit is used to achieve high sound quality. When packet reception is poor or delay is excessive, the time limit is shortened to maintain conversational comfort. This dynamic adjustment resolves the contradiction between reliability and time loss.
2Loss of time
If the time limit is set to 0 to prevent discomfort during conversation, then delay time is minimized, but the percentage of frames with high sound quality becomes very small
Solution Approach 1:
The patent dynamically adjusts the time limit based on actual packet reception conditions. Instead of using a fixed time limit of 0, the system monitors the reception state and sets the time limit adaptively. This allows the system to achieve high sound quality in frames where packets are received in time while maintaining conversational comfort in frames where delay would be problematic.
Solution Approach 2:
The patent changes the parameter of time limit based on reception conditions. By monitoring whether packets are received within expected timeframes, the system adjusts the time limit parameter to optimize the balance between sound quality and delay. This parameter change strategy resolves the contradiction by allowing the system to operate at different points on the quality-delay tradeoff curve depending on current conditions.
3Reliability
If packets from the second communication line are used to enhance sound quality, then decoded sound signal quality improves, but the complexity of managing multiple code strings and synchronization increases
Solution Approach 1:
The patent segments the processing of code strings from different communication lines. It separately manages the first code string from the first communication line and the second code string from the second communication line. The system determines whether to use codes from the second line based on reception conditions, effectively segmenting the decision-making process. This segmentation reduces complexity by treating each code string independently rather than managing them as a unified complex system.
Solution Approach 2:
The patent extracts only the necessary components from the second code string based on reception conditions. Instead of always processing and managing all elements of the second code string, the system selectively extracts and uses only those code elements that can be reliably received and synchronized. This extraction approach reduces the complexity of managing multiple code strings while still achieving enhanced sound quality when conditions permit.
Data Source
AI summary
Provided is a technique according to which it is possible to obtain a decoded sound signal of high sound quality without significantly increasing the delay time compared to a configuration in which only a decoded sound signal of the minimum necessary sound quality is obtained. In a terminal apparatus connected to a first communication line and a second communication line with a lower priority level there than, sound signals of multiple channels are obtained and output based on a monaural code included in a first code string input from the first communication line and an extended code included in a second code string with the closest frame number to that of the monaural code among extended codes included in the second code string input from the second communication line.