Circular Buffer Audio Dropout Prevention
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Solution Overview
Problem
Conference calls often experience audio dropout due to bandwidth congestion and high jitter ratios in intermediary networks, leading to lost audio portions that disrupt the meeting experience.
Innovation Solution
A computer-implemented method using circular buffers to detect and prevent audio dropout by replicating missing audio segments in real-time, leveraging speech-to-text and text-to-speech modules to generate and inject missing audio at appropriate points, ensuring seamless audio playback.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If audio packets are transmitted through intermediary networks with low throughput or high jitter ratios, then network coverage and accessibility are improved, but audio dropout occurs due to bandwidth congestion and packet loss
Solution Approach 1:
The system performs preliminary actions by detecting audio dropout conditions before complete audio loss occurs, using circular buffers to store audio data in advance. When packet loss is detected, the system can immediately replay buffered audio segments, preventing the audio dropout from manifesting to users.
Solution Approach 2:
The circular buffer acts as a cushion that absorbs network jitter and packet loss. By maintaining a buffer of audio data, the system creates a protective layer that compensates for network instability, allowing audio playback to continue smoothly even when packets are lost during transmission.
2Reliability
If circular buffers are used to detect and prevent audio dropout by replicating missing audio segments, then audio continuity is improved, but network overhead and processing complexity increase
Solution Approach 1:
The system creates copies of audio data in circular buffers and generates synthetic replacements for lost audio segments. By copying audio characteristics and using text-to-speech to recreate missing segments, the system can replace lost audio without requiring complex real-time analysis, simplifying the overall processing while maintaining audio continuity.
Solution Approach 2:
The system substitutes complex real-time audio reconstruction with a simpler mechanism: detecting packet loss, retrieving buffered audio data, and using pre-trained text-to-speech models to generate replacement segments. This replaces what would otherwise require complex real-time audio synthesis with more manageable data retrieval and synthesis operations.
Data Source
AI summary
Embodiments of the present invention provide methods, computer program products, and systems. Embodiments of the present invention detect an audio stream comprising one or more voice packets from a first computing system. Embodiments of the present invention can, in response to detecting an audio stream, dynamically prevent audio drop out on a second computing system using circular buffers based on network consistency.


