Dynamic Audio Buffer Adaptation for Packet Loss Recovery
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Solution Overview
Problem
Existing network-based audio communication systems either prioritize audio quality or conversational quality, but not both dynamically, leading to delays or compromised quality during real-time interactions due to packet loss handling methods like retransmission and forward error correction.
Innovation Solution
A system that dynamically adapts to communication conditions by increasing buffer size during non-interactive periods to mask packet loss recovery delays and reducing it during interactive periods to minimize playback delay, using techniques like packet retransmission and forward error correction based on interactivity mode detection.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If packet retransmission is used to guarantee packet delivery, then audio quality is improved, but playback delay increases
Solution Approach 1:
The system dynamically adjusts the buffer size based on the detected interactivity mode. During non-interactive periods, the buffer size is increased to allow time for packet retransmission and recovery, improving audio quality without significantly impacting user experience. During interactive periods, the buffer size is reduced to minimize playback delay and maintain real-time conversational quality. This dynamic adaptation resolves the contradiction by making the buffer size flexible rather than fixed.
2Reliability
If buffer size is increased to recover lost packets, then audio quality is improved, but conversational quality deteriorates
Solution Approach 1:
The system uses interactivity mode detection to dynamically control buffer size. When the system detects non-interactive mode (e.g., single speaker), it increases buffer size to recover lost packets through retransmission requests, improving audio quality. When interactive mode is detected (e.g., multiple speakers taking turns), the system reduces buffer size to minimize delay and maintain conversational quality. This dynamic adjustment resolves the contradiction by adapting buffer size to actual communication needs.
3Reliability
If forward error correction is applied, then packet loss is corrected, but channel bandwidth increases
Solution Approach 1:
Instead of applying forward error correction uniformly throughout the communication session, the system selectively applies error correction techniques based on the detected interactivity mode and packet loss conditions. During non-interactive periods with detected packet loss, the system requests retransmission or applies error correction. During interactive periods, the system prioritizes low latency and reduces or eliminates error correction overhead. This localized, conditional application of error correction resolves the contradiction by using bandwidth-intensive techniques only when necessary.
Data Source
AI summary
When packet loss is detected during a communication session, a current interactivity mode is checked to determine whether to increase the amount of received audio data stored in a buffer of the receiving device. If the current interactivity mode indicates a low level of interactivity between participants in the communication session, then the total amount of received audio data stored in the buffer is increased, in order to increase the delay between receipt of audio data by the electronic device, and outputting of the audio data by the electronic device. The increased output delay is then used to recover lost packets, so that audio quality is increased while the level of interactivity between participants is low. When the current interactivity mode subsequently indicates higher participant interactivity, the amount of received audio data stored in the buffer may be reduced.