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Low-complexity, low-delay, scalable and embedded speech and audio coding with adaptive frame loss concealment
Low-complexity, low-delay, scalable and embedded speech and audio coding with adaptive frame loss concealment
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机译:低复杂度,低延迟,可扩展的嵌入式语音和音频编码,具有自适应帧丢失隐藏功能
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摘要
High-quality, low-complexity and low-delay scalable and embedded system and method are disclosed for coding speech and general audio signals. The invention is particularly suitable in Internet Protocol (IP)-based multimedia communications. Adaptive transform coding, such as a Modified Discrete Cosine Transform, is used, with multiple small-size transforms in a given signal frame to reduce the coding delay and computational complexity. In a preferred embodiment, for a chosen sampling rate of the input signal, one or more output sampling rates may be decoded with varying degrees of complexity. Multiple sampling rates and bit rates are supported due to the scalable and embedded coding approach underlying the present invention. Further, a novel adaptive frame loss concealment approach is used to reduce the distortion caused by packet loss in communications using IP networks.
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