ADPCM Audio Codec State Feedback for Wireless Packet Loss
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Solution Overview
Problem
Current audio transmission systems face challenges in achieving low latency while maintaining high-quality digital audio transmission over wireless networks, especially in environments prone to packet loss, which often result in increased latency or artifacts.
Innovation Solution
The proposed method involves an ADPCM encoding scheme that includes additional data bits representing internal values of adaptive quantization and prediction, allowing the decoder to adapt to transmission errors and packet loss, thereby reducing artifacts and maintaining low latency.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Loss of time
If compression algorithms are used to reduce latency, then transmission speed is improved, but audio quality deteriorates with more artifacts caused by packet loss
Solution Approach 1:
The encoder transmits side information about internal decoder state (quantizer state, predictor coefficients) in advance within the same packet or previous packets. This preliminary transmission of state information enables the decoder to quickly recover from packet loss without requiring lengthy re-synchronization, thus maintaining both low latency and high reliability
Solution Approach 2:
The system uses feedback by transmitting encoder state information (quantizer state, predictor coefficients) to the decoder. This feedback mechanism allows the decoder to maintain accurate internal state even when packets are lost, enabling rapid recovery and maintaining audio quality without increasing latency
2Reliability
If more bits per sample are used to improve audio quality, then SNR is improved, but transmission bandwidth consumption increases
Solution Approach 1:
The audio coding information is segmented into essential ADPCM data and supplementary side information (quantizer state, predictor coefficients). This segmentation allows the system to transmit only the necessary state information rather than redundant audio data, maintaining audio quality while optimizing bandwidth usage in wireless transmission
Solution Approach 2:
The system dynamically adjusts transmission parameters by sending side information about internal decoder state only when necessary for recovery. This parameter change approach allows adaptive bandwidth consumption - using more bits only when packet loss occurs and recovery is needed, rather than continuously transmitting at high bitrate
3Reliability
If packet loss handling mechanisms are added to improve robustness, then reliability is improved, but device complexity increases
Solution Approach 1:
The decoder uses its own internal state information (quantizer state, predictor coefficients) that was transmitted by the encoder to automatically recover from packet loss. This self-service approach enables robust packet loss handling without requiring complex external correction mechanisms or additional processing complexity
Data Source
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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.