Audio Bitstream Decoding with Adjacent-Frame Parameter Smoothing
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Solution Overview
Problem
Existing redundancy coding algorithms for speech/audio bitstream decoding suffer from signal instability due to low bit rate encoding, leading to poor quality output speech/audio signals, especially in scenarios with high packet loss rates and delay jitter in VoIP systems.
Innovation Solution
A method and apparatus for speech/audio bitstream decoding that involves acquiring decoding parameters for current speech/audio frames, performing post-processing on these parameters based on parameters from adjacent frames, and recovering the speech/audio signal, thereby stabilizing the decoding process and improving output quality.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If redundancy coding algorithm encodes speech/audio frame information at lower bit rate, then redundancy information can be transmitted for packet loss compensation, but signal instability occurs and output speech/audio quality deteriorates
Solution Approach 1:
The patent changes the encoding approach by encoding redundancy information in the time domain rather than converting to frequency domain, and by using linear prediction coefficients from the current frame instead of previous frame parameters. This parameter change maintains signal stability while providing packet loss compensation.
Solution Approach 2:
The patent performs preliminary encoding of redundancy information using the current frame's linear prediction coefficients before transmission. This preliminary action ensures that when packet loss occurs, the decoder can directly use the pre-encoded redundancy information without needing to recalculate parameters, thus maintaining signal stability and quality.
2Stability of the object's composition
If jitter buffer buffers speech/audio frames to compensate for network delay jitter, then transmission delay variations are smoothed, but decoding delay increases and real-time performance deteriorates
Solution Approach 1:
The patent creates a copy of the current frame's linear prediction coefficients and spectral information as redundancy information. This copying approach allows the decoder to quickly reconstruct lost frames using the copied parameters without needing to buffer and process multiple frames, thus reducing decoding delay while maintaining delay jitter smoothing capability.
3Reliability
If FEC (Forward Error Correction) is applied to protect speech/audio data, then packet loss resistance is improved, but bit rate increases and transmission efficiency decreases
Solution Approach 1:
The patent applies partial redundancy encoding by encoding only essential parameters (linear prediction coefficients and spectral information) at a reduced bit rate, rather than encoding the entire audio signal. This partial action provides sufficient packet loss resistance while maintaining transmission efficiency.
Data Source
AI summary
The present invention disclose a speech/audio bitstream decoding method including: acquiring a speech/audio decoding parameter of a current speech/audio frame, where the foregoing current speech/audio frame is a redundant decoded frame or a speech/audio frame previous to the foregoing current speech/audio frame is a redundant decoded frame; performing post processing on the acquired speech/audio decoding parameter according to speech/audio parameters of X speech/audio frames, where the foregoing X speech/audio frames include M speech/audio frames previous to the foregoing current speech/audio frame and/or N speech/audio frames next to the foregoing current speech/audio frame; and recovering a speech/audio signal by using the post-processed speech/audio decoding parameter of the foregoing current speech/audio frame. The technical solutions of the present invention help improve quality of an output speech/audio signal.