Audio Buffering Dynamics for Jitter Compensation
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Solution Overview
Problem
Existing audio processing technologies face challenges in maintaining audio fluency and reducing delay due to network jitter, as they rely on buffering audio data packets of fixed long time lengths, which can lead to significant delays before sound is heard by the user.
Innovation Solution
An audio processing method that accumulates the actual total playing time length and total sampling time length of audio data packets, suspending playback when the sum of sampling time lengths of unread packets in the buffer is less than a threshold, until it exceeds an estimated network jitter value based on these accumulated times, dynamically addressing network delay cumulative effects.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If audio data packets of fixed long time length are buffered to eliminate impact of network jitter on fluency, then audio fluency is improved, but playback delay increases
Solution Approach 1:
The patent applies dynamics by transitioning from fixed-time buffering to adaptive buffering. The buffering duration is dynamically adjusted based on real-time network jitter estimation, allowing the system to optimize between fluency and delay trade-offs adaptively rather than using a static fixed duration approach
Solution Approach 2:
The patent implements feedback mechanisms by continuously monitoring actual playback timing versus sampling timing, estimating network jitter based on accumulated time differences, and using this feedback to dynamically adjust the buffering duration to maintain optimal audio fluency while minimizing delay
2Measurement precision
If network jitter is estimated based on accumulated actual playing time and sampling time, then buffering accuracy is improved, but system complexity increases
Solution Approach 1:
The system uses self-service by leveraging its own internal timing data (actual playing time and sampling time) to estimate network jitter, eliminating the need for external complex measurement systems or additional hardware, thereby achieving accurate estimation with minimal added complexity
Data Source
AI summary
Audio processing methods and apparatus are provided. An audio processing method may include: receiving audio data packets; buffering the audio data packets to a buffer; reading the audio data packets from the buffer and playing the audio data packets; accumulating an actual total playing time length and a total sampling time length of the audio data packets that currently have been read from the buffer and have been played; and suspending reading and playing, when a sum of sampling time lengths of audio data packets that are buffered and unread in the buffer is less than or equal to a first threshold, until a sum of sampling time lengths of the audio data packets in the buffer that are unread is greater than or equal to a current network jitter estimated value.