摘要 |
<p>A high-quality, low-complexity and low-delay scalable embedded method is disclosed for coding speech and audio signals suitable for Internet Protocol (IP) based multimedia communications. MDCT Processor (10) produces multiple small sized adaptive transforms in a frame of input signal S(n) which reduces the coding delay and complexity of the output bit stream produced by Multiplexer (80). In a preferred embodiment, where for a given input sampling rate, one or more output sampling rates are decoded with varying degrees of complexity by MDCT Coefficient Decoder (140), by Log-gain Decoder (100) or by Adaptive Bit Allocation processor (110). Further, a novel Adaptive Frame Loss Concealment Processor (160) reduces distortion of communications caused by packet loss.</p> |