Audio Decoding Method for High Quality Low Delay
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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, for each frame, checks if an extended code with the same frame number as a monaural code from a high-priority communication line is present in a code string from a low-priority communication line. If present, it outputs the monaural code and the extended code; otherwise, it outputs the monaural code with an extended code having the closest frame number.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If the time limit is set to a long duration to obtain high sound quality in many frames, then the percentage of frames with high sound quality increases, but the delay time becomes excessively long causing discomfort during two-way conversation
Solution Approach 1:
The patent dynamically adjusts the time limit for waiting extended codes based on whether a monaural code has been received. If a monaural code is received, the time limit is set to a first value (longer duration); if not received, a second value (shorter duration) is used. This dynamic adjustment allows the system to adaptively balance between obtaining high sound quality and maintaining acceptable delay times for real-time conversation.
Solution Approach 2:
The patent changes the parameter of time limit duration based on reception conditions. By switching between two different time limit values (first value and second value) depending on whether monaural codes are received, the system optimizes the balance between sound quality and delay time without requiring a fixed extreme setting.
2Loss of time
If the time limit is set to a short duration to prevent discomfort during conversation, then delay time is reduced, but the percentage of frames with high sound quality becomes very small
Solution Approach 1:
The system uses dynamic time limit adjustment to prevent the need for a consistently short time limit. By extending the time limit to a first value when monaural codes are received (indicating good network conditions), the system can achieve high sound quality without permanently committing to short delay times that would compromise audio quality.
Solution Approach 2:
The patent performs preliminary assessment of network conditions by monitoring monaural code reception before determining the appropriate time limit. This preliminary action allows the system to prepare appropriate time limit settings in advance, ensuring both low delay and high sound quality are achieved when network conditions permit.
3Adaptability or versatility
If scalable encoding is performed for each frame with separate transmission of base layer and extension layer codes, then sound quality can be adapted to network conditions, but the system complexity increases due to multi-network packet management
Solution Approach 1:
The patent segments the audio code into two distinct parts: monaural codes (base layer) transmitted via first network and extended codes (extension layer) transmitted via second network. This segmentation allows independent management of each code type through appropriate networks, simplifying the overall packet management while maintaining adaptability to different network conditions.
Solution Approach 2:
The patent introduces a selection unit that acts as an intermediary to manage the complex packet routing logic. This intermediary component simplifies the system architecture by centralizing the decision-making process for code selection and combination, reducing the complexity distributed across the entire system while maintaining the ability to adapt to network conditions.
Data Source
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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 therethan, 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.