Adaptive Audio Time Scaling for VoIP Jitter Compensation
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Solution Overview
Problem
Existing audio signal time-scaling methods in packet-switched networks face challenges in maintaining optimal audio quality due to varying network jitter, which introduces irregular packet arrival times and delays, making it difficult to balance buffering delay and delayed frames, especially in real-time audio playback applications like VoIP.
Innovation Solution
A method and system that dynamically adjust the time-scaling of audio signals by detecting changes in delay and determining the appropriate amount and duration of time scaling, using a windowed approach to perform time scaling in small steps within a specified time window, allowing for immediate adjustments in response to network changes while minimizing audio quality degradation.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If a fixed delay jitter buffer is used to cover worst case jitter scenarios, then the number of delayed frames is kept in control, but the end-to-end delay becomes too long to enable natural conversation
Solution Approach 1:
The patent applies dynamics by transitioning from a fixed delay jitter buffer to an adaptive jitter buffer that dynamically adjusts its delay parameter based on observed network conditions. The buffer delay is continuously monitored and adjusted to match actual jitter levels, allowing the system to use minimal delay during low jitter periods and increase delay only when network conditions deteriorate, thus resolving the contradiction between maintaining reliability and minimizing end-to-end delay.
Solution Approach 2:
The patent changes the delay parameter of the jitter buffer dynamically based on observed packet arrival patterns and jitter measurements. By adjusting the delay parameter adaptively rather than keeping it fixed, the system optimizes the balance between preventing frame delays and minimizing overall latency, directly addressing the technical contradiction presented.
2Loss of time
If the buffering delay is reduced to minimize end-to-end delay, then the overall delay is minimized, but the audio signal needs to be shortened which complicates the buffer adjustment
Solution Approach 1:
The patent introduces time scaling as an intermediary mechanism that mediates between the jitter buffer and the audio playback component. When buffer delay needs adjustment, the time scaling unit modifies the audio signal duration accordingly, allowing the buffer to operate at optimal delay settings without directly complicating the buffer adjustment logic. This intermediary handles the complexity of signal duration adjustment separately from buffer management.
Solution Approach 2:
The patent segments the audio processing system into distinct functional blocks: jitter buffer, decoder, time scaling unit, and playback component. This segmentation allows each component to operate independently with well-defined interfaces, simplifying the overall buffer adjustment complexity by localizing the time scaling function to a dedicated unit rather than requiring complex coordination across the entire system.
3Reliability
If the buffering delay is increased to meet worsening network conditions, then the number of delayed frames is reduced, but the audio signal has to be lengthened to compensate
Solution Approach 1:
The time scaling unit serves as an intermediary that automatically handles the complexity of audio signal length adjustment when buffer delay changes. Whether the delay is increased or decreased, the time scaling unit mediates the corresponding signal duration changes, maintaining a clear separation between buffer management logic and signal processing complexity.
4Loss of time
If time scaling is applied to compensate for changing buffer delay, then the audio signal length is adjusted, but audio quality degradation occurs especially in active speech parts
Solution Approach 1:
The patent applies local quality by implementing different time scaling strategies for different parts of the audio signal. During active speech portions, the system uses pitch-synchronous time scaling that operates at the pitch period level to minimize quality degradation. During non-speech or comfort noise portions, more aggressive time scaling can be applied. This localized approach to time scaling preserves audio quality in critical regions while still providing effective buffer delay compensation.
Solution Approach 2:
The time scaling operation is made dynamic by adapting the scaling factor and method based on the current audio signal characteristics. The system continuously monitors whether the signal contains active speech, comfort noise, or silence, and dynamically adjusts the time scaling approach accordingly, using finer-grained scaling during speech and coarser scaling during non-speech segments to maintain audio quality.
Data Source
AI summary
For controlling a time-scaling of an audio signal, the audio signal being distributed to a sequence of frames that are received via a packet switched network, a change in a delay of received frames is detected. Moreover, an amount of time scaling that is to be applied to received frames for compensating for the detected change is determined. Further, a kind of the change is determined. Further, a length of a time window within which a time scaling of the determined amount is to be completed is determined depending on the determined kind of the change.