Dynamic Voice Codec Negotiation for Adaptive Call Quality
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Solution Overview
Problem
Conventional voice compression techniques in wireless communication systems fix voice bandwidth and bit-rate, failing to adapt to individual preferences or background noise, leading to suboptimal call quality.
Innovation Solution
A method and apparatus for dynamically adjusting the sampling rate and bit-rate of voice signals between terminals during a call, allowing for a wider range of options and adapting to user preferences and background noise by negotiating and combining different sampling and bit-rate combinations.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If voice bandwidth and bit-rate are fixed in conventional voice compression techniques, then device complexity is reduced and ease of operation is improved, but call quality cannot be optimized for individual preferences or background noise conditions
Solution Approach 1:
The patent implements dynamic switching between different voice codecs (AMR-NB, AMR-WB, EVS) with varying sampling rates (8kHz, 16kHz, 32kHz, 48kHz) and bit-rates based on real-time network conditions and user preferences. The system transitions from fixed codec configuration to adaptive codec selection, allowing call quality optimization while managing complexity through automated negotiation protocols.
Solution Approach 2:
The system changes multiple codec parameters simultaneously including sampling rate (8kHz to 48kHz), bit-rate (various levels), and voice bandwidth (narrowband to fullband) to optimize call quality. These parameter changes are negotiated between terminals through capability exchange and condition-based selection mechanisms.
2Adaptability or versatility
If multiple sampling rates and bit-rates are supported for optimal call quality, then adaptability improves, but device complexity and negotiation overhead increase
Solution Approach 1:
The patent implements a universal codec framework where a single terminal can handle multiple voice codecs (AMR-NB, AMR-WB, EVS) with different sampling rates and bit-rates. This multi-functional capability allows the system to adapt to various network conditions and user preferences while maintaining a unified negotiation and control mechanism.
Solution Approach 2:
The system uses feedback mechanisms where terminals exchange capability information and negotiate optimal codec parameters based on current network conditions and user preferences. The feedback loop includes capability advertisement, condition assessment, and parameter selection to manage complexity while maintaining flexibility.
3Adaptability or versatility
If voice codecs process a fixed bandwidth range, then manufacturing precision is improved, but the ability to accommodate different listener preferences and noise conditions deteriorates
Solution Approach 1:
The system dynamically adjusts voice bandwidth processing from narrowband (300-3400 Hz) to wideband (50-7000 Hz) to fullband (50-14000 Hz or higher) based on network conditions and user preferences. This dynamic bandwidth adaptation allows the codec to process different frequency ranges while maintaining a unified design framework.
Data Source
AI summary
Provided are a device and a method for transmitting and receiving voice data in a wireless communication system. A method for operating a transmission terminal for transmitting a voice signal comprises the steps of: generating sampling and bitrate request information including sampling information for determining a sampling rate of the voice signal and bitrate information for determining a bitrate of the voice signal, and transmitting the generated sampling and bitrate request information to a reception terminal; receiving, from the reception terminal, combined determination information obtained by at least one combination of the sampling rate determined on the basis of the sampling information and the bitrate determined on the basis of the bitrate information; and compressing the voice signal according to the received combined determination information, and transmitting the compressed voice signal to the reception terminal.