Audio signal encoding and decoding

Abstract

An audio codec is 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 canadapt 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.

Description

technical field [0001] The invention relates to the field of audio transmission and audio codecs. In particular, the invention relates to the field of wireless transmission of high-quality digital audio signals with low latency and high resistance to packet loss, for example for audio signals from microphones and musical instruments in live performances. Background technique [0002] Real-time two-way audio applications such as headphones, microphones, and monitor speakers often require communication systems with minimal latency. When digital transmission with a limited bit rate is desired, eg for wireless transmission, the restrictions on audio coding schemes are severe. For such applications, a sampling coding scheme based on Adaptive Differential Pulse Code Modulation (ADPCM) can provide remarkably low latency. However, wireless transmission systems are prone to transmission errors and data loss. Audio coding schemes must handle such events in an appropriate manner, wi...

AI Technical Summary

Problems solved by technology

By using less compression (i.e. more bits per sample), one obtains relatively poorer quality (i.e. lower SNR at a given bit rate), and suffers more from packet loss. Artifacts

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