ADPCM Audio Codec Recovery From Wireless Packet Loss
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Solution Overview
Problem
Current audio transmission systems face challenges in achieving low latency while maintaining high quality and immunity to packet loss, especially in wireless transmission, often requiring longer latency or poorer quality to handle transmission errors effectively.
Innovation Solution
The method employs Adaptive Differential Pulse Code Modulation (ADPCM) encoding with adaptive quantization and prediction, including additional data bits to represent internal values, allowing for faster adaptation to transmission errors and minimizing audible artifacts, using a processor with fixed-point architecture and transmitting via wireless radio frequency signals.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Loss of energy
If compression algorithms are used to reduce bit rate, then transmission efficiency is improved, but latency increases and audio quality deteriorates
Solution Approach 1:
The patent changes the parameter representation by including logarithmic encoded versions of internal ADPCM values (scaling factors, predictor coefficients) instead of transmitting full precision values. This reduces the number of bits required per data block while maintaining the ability to reconstruct audio samples with acceptable quality and latency characteristics.
Solution Approach 2:
The patent extracts only the essential internal state information (scaling factors and predictor coefficients) from the ADPCM encoding process and transmits these separately as additional data bits. This allows the receiver to reconstruct the audio signal with fewer bits while maintaining synchronization and quality, resolving the trade-off between compression efficiency and latency.
2Manufacturing precision
If more bits per sample are used to reduce compression, then audio quality is improved, but transmission bandwidth increases and packet loss impact worsens
Solution Approach 1:
The patent transmits logarithmic encoded versions of internal ADPCM parameters (scaling factors and predictor coefficients) as additional data bits. This allows the receiver to maintain accurate reconstruction of audio samples with fewer total bits transmitted, reducing the impact of packet loss while preserving audio quality through efficient parameter representation.
3Reliability
If internal ADPCM values are transmitted to improve error recovery, then robustness against packet loss is improved, but data block size increases
Solution Approach 1:
The patent applies logarithmic encoding to represent internal ADPCM values (scaling factors and predictor coefficients) using fewer bits than their full precision representation. This reduces the increase in data block size while still providing the receiver with sufficient information to quickly recover from packet loss and maintain audio quality, thus improving robustness with minimal overhead.
Data Source
AI summary
An audio codec suitable for robust wireless transmission of high quality audio with low latency, still at a moderate bit rate. The encoding and decoding methods are based on ADPCM and in addition to the encoded output bits APM, additional data QB are included in output data blocks, namely data QB representing an internal value of the adaptive quantization ADQ of the ADPCM encoding algorithm, especially a scaling factor encoded and truncated to such as 8 bits. Further, output data blocks preferably include data CFB representing an internal value of the predictor PR of the ADPCM encoding algorithm, especially data CFB representing coefficients of a lattice prediction FIR filter which, truncated to such as 8 bits, can be sequentially included in output data blocks. These additional data QB, CFB regarding internal values of the ADPCM encoding algorithm can be utilized at the encoder side to increase robustness against loss of data blocks in wireless transmission. Especially, the decoding algorithm may comprise comparing its current internal ADPCM decoding values corresponding to the received internal values QB, CFB from the encoder, and in case there is a difference, the decoder can adapt or overwrite its internal values to the ones received QB, CFB. This helps to ensure fast recovery after lost data blocks, thereby ensuring robustness against artefacts in the reconstructed signal, e.g. clicks in case of audio.