Adaptive Jitter Buffer for VoIP Audio Continuity
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Solution Overview
Problem
Voice-over-internet-protocol (VOIP) communication faces challenges due to network jitter and frame losses in packet-switched networks, leading to discontinuities in audio playback and increased end-to-end delay, which degrade communication quality.
Innovation Solution
An adaptive jitter buffer system that includes an adaptive audio sample buffer, resampler, and fill quantity controller, which estimates and adjusts the buffer fill quantity based on jitter estimation and target fill determination to maintain optimal buffer levels, reorder packets, and conceal data losses, ensuring minimal delay and preserving binaural cues.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If a fixed-size jitter buffer is used to compensate for network jitter, then audio playback discontinuities are reduced, but end-to-end delay increases
Solution Approach 1:
The jitter buffer size is made dynamic rather than fixed. The controller continuously monitors network jitter conditions and adjusts the buffer size accordingly - expanding it when jitter is high to prevent playback discontinuities, and contracting it when jitter is low to minimize end-to-end delay. This dynamic adaptation resolves the contradiction by making the buffer responsive to actual network conditions rather than using a static size.
Solution Approach 2:
The key parameter being changed is the jitter buffer size. By varying this parameter based on measured network conditions (jitter estimates), the system optimizes the trade-off between playback continuity and delay. When network conditions deteriorate, the buffer size parameter is increased; when conditions improve, it is decreased, thus adapting to changing requirements.
2Reliability
If a larger jitter buffer is used to handle high network jitter, then audio playback discontinuities are reduced, but the buffer fill quantity control becomes more complex
Solution Approach 1:
The system implements a feedback control mechanism where the controller monitors the actual buffer fill quantity and compares it against a target fill quantity. Based on this feedback, the controller dynamically adjusts the buffer size and packet reception rate to maintain optimal operation. This closed-loop feedback simplifies the control complexity by providing a systematic, automated approach rather than requiring complex manual tuning or multiple separate control mechanisms.
Solution Approach 2:
The jitter buffer controller performs self-adjustment based on its own monitoring of network conditions and buffer state. It autonomously determines when to expand or contract the buffer without requiring external intervention or complex coordination with other system components. This self-service capability reduces overall system complexity by making the buffer management independent and self-regulating.
3Reliability
If packets are reordered to compensate for network jitter, then audio playback discontinuities are reduced, but processing complexity increases
Solution Approach 1:
The system performs preliminary actions by pre-buffering incoming packets before they are needed for playback. Instead of waiting for packets to arrive in strict sequence and then processing them, the system anticipates potential out-of-order arrivals and buffer underruns by maintaining a sufficient buffer fill quantity in advance. This preliminary buffering simplifies the processing complexity by decoupling packet arrival timing from playback timing.
Data Source
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AI summary
The present invention relates to an adaptive jitter buffer (10) for buffering audio data received via a packet-switched network. The adaptive jitter buffer (10) comprises an adaptive audio sample buffer (50), which in turn comprises an adaptive resampler for receiving a number of audio samples of the audio data and for outputting a first number of audio samples, which are resampled from the received number of audio samples according to a resampling factor, an audio sample buffer for buffering audio samples, wherein the outputted first number of audio samples are written to the audio sample buffer during an input access event and a second number of audio samples are read from the audio sample buffer during an output access event, and an audio sample buffer fill quantity controller for controlling the fill quantity of the audio sample buffer, wherein the audio sample buffer fill quantity controller is adapted to control the fill quantity of the audio sample buffer based on controlling the resampling factor of the adaptive resampler.