A subband audio coder employs perfect/non-perfect reconstruction filters, predictive/non-predictive subband encoding, transient analysis, and psycho-acoustic/minimum mean-square-error (mmse) bit allocation over time, frequency and the multiple audio channels to encode/decode a data stream to generate high fidelity reconstructed audio. The audio coder windows the multi-channel audio signal such that the frame size, i.e. number of bytes, is constrained to lie in a desired range, and formats the encoded data so that the individual subframes can be played back as they are received thereby reducing latency. Furthermore, the audio coder processes the baseband portion (0-24 kHz) of the audio bandwidth for sampling frequencies of 48 kHz and higher with the same encoding/decoding algorithm so that audio coder architecture is future compatible.

Multi-channel predictive subband audio coder using psychoacoustic adaptive bit allocation in frequency, time and over the multiple channels

A technique for transmission of wireless signals across CDMA radio links. Bandwidth is allocated dynamically within a session to specific CDMA subscriber unit based upon data rate determinations. Specifically, a dynamic bandwidth allocation algorithm operates from limits calculated based upon available ports per subscriber, expected user bandwidth, and parallel user bandwidth versus throughput. Provisions for priority service, unbalanced forward and reverse spectrum utilization, voice prioritization, and band switching are also made.

Dynamic bandwidth allocation to transmit a wireless protocol across a code division multiple access (cdma) radio link

An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Encoding device and decoding device

Multiple channels of audio are combined either to a monophonic composite signal or to multiple channels of audio along with related auxiliary information from which multiple channels of audio are reconstructed, including improved downmixing of multiple audio channels to a monophonic audio signal or to multiple audio channels and improved decorrelation of multiple audio channels derived from a monophonic audio channel or from multiple audio channels. Aspects of the disclosed invention are usable in audio encoders, decoders, encode/decode systems, downmixers, upmixers, and decorrelators.

Multichannel audio coding

Multi-channel audio decoder

The invention relates in general to high-quality low bit-rate digital transform coding and decoding of information corresponding to audio signals such as music signals. More particularly, the invention relates to signal analysis/synthesis in coding and decoding. The invention can optimize the trade off in transform coders between time resolution and frequency resolution by adaptively selecting the transform block length for each sampled audio segment, and/or can optimize coding gain by adaptively selecting the transform and/or by adaptively selecting the analysis window or the analysis/synthesis window pair.

Adaptive-block-length, adaptive-transform, and adaptive-window transform coder, decoder, and encoder/decoder for high-quality audio

Cette invention se rapporte en general au codage et au decodage de transformation numerique a faible debit binaire et de haute qualite, d'informations correspondant a des signaux sonores tels que des signaux de musique. This invention relates generally to encoding and decoding digital transformation low bit rate and high quality of information corresponding to sound signals such as music signals. En particulier, cette invention se rapporte a l'analyse/la synthese de signaux dans le codage et decodage. In particular, this invention relates to the analysis / synthesis of signals in the encoding and decoding. L'invention peut optimiser le compromis dans des codeurs de transformation entre la resolution de temps et la resolution de frequence en selectionnant de maniere adaptative la longueur de bloc de transformation pour chaque segment sonore echantillonne, et/ou optimiser le gain de codage en selectionnant de maniere adaptative la transformation et/ou en selectionnant de maniere adaptative la fenetre d'analyse ou la paire de fenetres d'analyse/de synthese. The invention can optimize the compromise in coders transformation between time resolution and frequency resolution by adaptively selecting the transform block length for each sampled audio segment, and / or maximize the coding gain by selecting adaptively processing and / or by adaptively selecting the analysis window or the pair of analysis window / synthesis.

Davidson Grant Allen

Papers

Adaptive-block-length, adaptive-transform, and adaptive-window transform coder, decoder, and encoder/decoder for high-quality audio

Adaptive-block-length, adaptive-transform, and adaptive-window transform coder and decoder for high-quality audio