Sub-Band Audio Decoding After Packet Loss With Time-Warping
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Solution Overview
Problem
Existing packet loss concealment methods are inadequate for sub-band predictive coders like the ITU-T G.722 wideband speech coder, as they fail to address sub-band-specific architectural issues and the challenges posed by backward-adaptive ADPCM coders.
Innovation Solution
A method and system that alter decoding parameters in response to packet loss, using waveform extrapolation and time-warping to generate a decoded audio signal, specifically designed for sub-band predictive coders, which updates the states of ADPCM decoders and smooths transitions after packet loss.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If prior-art packet loss concealment methods are applied to sub-band predictive coders, then general packet loss concealment can be achieved, but sub-band-specific architectural issues and backward-adaptive challenges are not addressed, resulting in poor audio quality
Solution Approach 1:
The patent applies local quality by implementing packet loss concealment specifically tailored for sub-band predictive coders. The method updates ADPCM decoder states and applies time-warping specifically to the sub-band structure, rather than using a generic PLC approach. This localized adaptation to the specific architectural needs of sub-band coders resolves the contradiction between general PLC effectiveness and sub-band-specific adaptability.
Solution Approach 2:
The patent changes parameters by altering decoding parameters in response to packet loss detection. When packet loss is detected, the system modifies the ADPCM decoder state update process and applies time-warping to the decoded audio signal. These parameter changes enable the system to adapt to packet loss conditions while maintaining compatibility with sub-band predictive coder architecture, thus resolving the adaptability issue.
2Ease of operation
If normal decoding is performed after packet loss, then decoding simplicity is maintained, but audible artifacts occur due to discontinuities in the decoded audio signal
Solution Approach 1:
The patent applies preliminary action by detecting packet loss and altering decoding parameters before the audio discontinuity can occur. The system monitors for lost packets and proactively modifies the ADPCM decoder state updates and applies time-warping to prevent audible artifacts, rather than attempting to fix them after they occur. This maintains decoding simplicity while preventing harmful artifacts.
Solution Approach 2:
The patent introduces time-warping as an intermediary process between packet loss detection and audio output. The time-warping operation smoothly transitions the decoded audio signal through the packet loss event, acting as a mediator that prevents direct discontinuities while maintaining overall decoding simplicity. This intermediary approach resolves the contradiction between operational simplicity and artifact prevention.
3Stability of the object's composition
If ADPCM decoder states are updated during packet loss, then decoder synchronization is improved, but complexity increases due to additional state management requirements
Solution Approach 1:
The patent applies dynamics by making the ADPCM decoder state update process adaptive rather than static. The system dynamically adjusts whether to update decoder states based on packet loss detection, and dynamically modifies the update process itself during packet loss events. This dynamic approach improves decoder synchronization while managing complexity through conditional execution rather than always-on complex state management.
Data Source
AI summary
A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.