Adaptive De-Jitter Buffer for Voice Calls
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Solution Overview
Problem
Mobile devices face challenges in maintaining optimal speech quality during voice calls due to varying network conditions, such as jitter and underflow rates, which affect the de-jitter buffer's performance across different wireless networks like LTE and Wi-Fi, leading to inconsistent mouth-to-ear delays.
Innovation Solution
An adaptive de-jitter buffer system that adjusts its target delay range based on estimated jitter and underflow rates, allowing for dynamic adjustment of lower and upper bounds to optimize audio packet playback and reduce mouth-to-ear delay, and includes the option to drop packets based on a relative play out delay metric to maintain quality.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Device complexity
If a fixed target delay value is used in the de-jitter buffer, then the device complexity is reduced, but the adaptability to different network conditions deteriorates
Solution Approach 1:
The patent implements dynamic adjustment of the target delay value based on real-time network conditions. The system monitors packet arrival intervals and dynamically modifies the target delay parameter to adapt to varying jitter characteristics across different networks (LTE, Wi-Fi, handover scenarios), transforming a static buffer into an adaptive system that optimizes speech quality for each network context
Solution Approach 2:
The system changes the target delay parameter based on detected network conditions. By monitoring packet arrival patterns and calculating appropriate delay adjustments, the system modifies the playback delay parameter to match current network characteristics, enabling optimal performance across diverse wireless environments without increasing fundamental system complexity
2Reliability
If the target delay is increased to reduce underflow rate, then speech quality is improved, but the mouth-to-ear delay increases
Solution Approach 1:
The system employs feedback mechanisms to monitor packet arrival patterns and underflow conditions. By continuously measuring the interval between successive packet arrivals and comparing this against the current target delay, the system adjusts the delay parameter dynamically - increasing it when underflow risk is detected and decreasing it when network conditions improve, thus optimizing the trade-off between speech quality and latency
Solution Approach 2:
The target delay parameter is transformed from a static value to a dynamic variable that responds to real-time network conditions. The system adapts the delay duration based on observed packet arrival patterns, allowing it to increase delay only when necessary to prevent underflow, rather than maintaining a consistently high delay that would unnecessarily increase mouth-to-ear latency
3Device complexity
If the de-jitter buffer uses a fixed target delay range, then the device complexity is reduced, but the adaptability to network handovers deteriorates
Solution Approach 1:
The system implements dynamic adjustment of the target delay parameter specifically optimized for handover scenarios. By detecting network transitions and adapting the delay value based on handover-specific packet arrival patterns, the system maintains speech quality continuity during network switches between LTE and Wi-Fi, demonstrating adaptability to transient network condition changes
Solution Approach 2:
The target delay parameter is modified based on handover detection and network transition patterns. The system identifies handover events through packet arrival analysis and adjusts the delay parameter accordingly, enabling the de-jitter buffer to maintain optimal performance during network handovers without requiring complex handover-specific control logic
Data Source
AI summary
In a particular aspect, an apparatus includes a de-jitter buffer configured to store a plurality of audio packets received from a source device. The apparatus includes de-jitter buffer control circuitry configured to adjust at least one bound of a target delay range associated with the de-jitter buffer based at least in part on an estimated jitter associated with the plurality of audio packets, an estimated underflow rate associated with the plurality of audio packets, or both. The apparatus includes a processor configured to generate an audio signal based on the plurality of audio packets stored at the de-jitter buffer. The apparatus includes a receiver configured to receive the plurality of audio packets via a network from the source device. The apparatus further includes a speaker configured to output the audio signal.