Data processing method, apparatus and system

By performing text-to-speech processing on the reconstructed text data at the receiving end to generate second audio data and merging it with the reconstructed audio data, the problem of audio communication quality under poor network conditions is solved, and the integrity and fluency of audio data are improved in weak network environments.

WO2026045580A1 Publication Date: 2026-03-05HUAWEI TECH CO LTD
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Patent Information

Application Number
PCT/CN2025/104770
Authority / Receiving Office
WO · WO
Patent Type
Applications
Current Assignee / Owner
Priority Date
2024-08-30
Filing Date
2025-06-27
Publication Date
2026-03-05

AI Technical Summary

Technical Problem

In real-time communication scenarios, when network conditions deteriorate, packet loss leads to a decrease in audio communication quality, affecting fluency and clarity.

Method used

Text-to-speech technology is used to process the reconstructed text data at the receiving end to generate second audio data, which is then fused with the reconstructed audio data to generate third audio data, supplementing the parts lost in the network and ensuring the integrity and fluency of the audio data.

Benefits of technology

Even in poor network conditions, text-to-speech processing can restore complete audio data, reduce stuttering and silence, and improve the smoothness and clarity of audio communication.

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Abstract

The present application provides a data processing method, apparatus and system. The method comprises: receiving a first code stream and a second code stream, the first code stream comprising data obtained by encoding text corresponding to first audio data, and the second code stream comprising data obtained by encoding the first audio data; when reconstructed audio data is incomplete, performing text-to-speech processing on reconstructed text data to obtain second audio data; and fusing the second audio data and the reconstructed audio data to obtain third audio data, and outputting the third audio data, the reconstructed text data being obtained by decoding the first code stream, and the reconstructed audio data being generated on the basis of the second code stream. In this way, even if a network state becomes poor, a receiving end can still recover complete audio data, thereby reducing problems such as stuttering, dropped audio frames, and even silence, and improving the fluency of audio in a real-time communication scenario.
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