Abstract: In a first aspect, arbitrary shaping of the temporal envelope of noise is provided in spectral domain coding systems without the need of side-information. In the encoding, a filtered measure of quantization error is applied as a feedback signal to the frequency-domain representation of a discrete time-domain signal prior to quantization, so that the filtering parameters of said filtering affect the shaping of quantization noise in the time domain of the quantized frequency-domain representation of the discrete time-domain signal when it is inversely transformed from the frequency domain back to the time domain in decoding. This may be accomplished with respect to each of a plurality of frequency bins or groups of bins. In another aspect, frequency-domain noise-feedback quantizing in digital audio encoding is provided.

Abstract: An audio upmixer, such as a two-channel to three-channel upmixer, employs a difference in a measure of sound at the ears of a listener in accordance with first and second models, one based on a reproduction of the original channels and the other based on a reproduction of the upmixed channels. The difference is minimized while simultaneously causing a, portion of one or more of the stereophonic channels to be applied to the center loudspeaker under some conditions of the signals in the stereophonic channels, the portion being commensurate with the value of a weighting factor, such that the weighting factor controls a balance between two opposing conditions, one in which no signals are applied to the center loudspeaker and another in which no signals are applied to the left and right loudspeakers.

Abstract: An audio encoder or encoding method receives a plurality of input channels and generates one or more audio output channels and one or more parameters describing desired spatial relationships among a plurality of audio channels that may be derived from the one or more audio output channels, by detecting changes in signal characteristics with respect to lime in one or more of the plurality of audio input channels, identifying as auditory event boundaries changes in signal characteristics with respect to lime in the one or more of the plurality of audio input channels, an audio segment between consecutive boundaries constituting an auditory event in the channel or channels, and generating all or some of the one or more parameters al least partly in response to auditory events and/or the degree of change in signal characteristics associated with the auditory event boundaries. An auditory-event-responsive audio upmixer or upmixing method is also disclosed.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

Abstract: A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal including temporal shape. Spectral components are synthesized having the one or more assessed characteristics, integrated with the subband signals and passed through a synthesis filterbank to generate an output signal.

Abstract: A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal including temporal shape. Spectral components are synthesized having the one or more assessed characteristics, integrated with the subband signals and passed through a synthesis filterbank to generate an output signal.

Abstract: Analysis and synthesis filter banks such as those used in audio and video coding systems are each implemented by a hybrid transform that comprises a primary transform in cascade with one or more secondary transforms. The primary transforms for the filter banks implement an analysis/synthesis system in which time-domain aliasing artifacts are cancelled. The secondary transforms, which are in cascade with the primary transforms, are applied to blocks of transform coefficients. The length of the blocks is varied to adapt the time resolution of the analysis and synthesis filter banks.

Abstract: Mechanisms are known that allow receivers to control loudness of speech in broadcast signals but these mechanisms require an estimate of speech loudness be inserted into the signal. Disclosed techniques provide improved estimates of loudness. According to one implementation, an indication of the loudness of an audio signal containing speech and other types of audio material is obtained by classifying segments of audio information as either speech or non-speech. The loudness of the speech segments is estimated and this estimate is used to derive the indication of loudness. The indication of loudness maybe used to control audio signal levels so that variations in loudness of speech between different programs is reduced. A preferred method for classifying speech segments is described.

Abstract: Audio coding processes like quantization can cause spectral components of an encoded audio signal to be set to zero, creating spectral holes in the signal. These spectral holes can degrade the perceived quality of audio signals that are reproduced by audio coding systems. An improved decoder avoids or reduces the degradation by filling the spectral holes with synthesized spectral components. An improved encoder may also be used to realize further improvements in the decoder.

Abstract: A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal. Spectral components are synthesized having the assessed characteristics. The synthesized spectral components are integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. In one implementation, the assessed characteristic is temporal shape and noise-like spectral components are synthesized having the temporal shape of the audio signal.

Abstract: Blocks of audio information are arranged in groups that share encoding control parameters to reduce the amount of side information needed to convey the control parameters in an encoded signal. The configuration of groups that reduces the distortion of the encoded audio information may be determined by any of several techniques that search for an optimal or near optimal solution. The techniques include an exhaustive search, a fast optimal search and a greed merge, which allow the search technique to tradeoff the reduction in distortion against the bit rate of the encoded signal and/or the computational complexity of the search technique.

Abstract: Each of N audio signals are filtered with a unique decorrelating filter (38) characteristic, the characteristic being a causal linear time-invariant characteristic in the time domain or the equivalent thereof in the frequency domain, and, for each decorrelating filter characteristic, combining (40, 44, 46), in a time and frequency varying manner, its input (Zi) and output (Z-i) signals to provide a set of N processed signals (X i). The set of decorrelation filter characteristics are designed so that all of the input and output signals are approximately mutually decorrelated. The set of N audio signals may be synthesized from M audio signals by upmixing (36), where M is one or more and N is greater than M.

Abstract: A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal. Spectral components are synthesized having the assessed characteristics. The synthesized spectral components are integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. In one implementation, the assessed characteristic is temporal shape and noise-like spectral components are synthesized having the temporal shape of the audio signal.

Abstract: A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.

Abstract: The perceived quality of an audio signals obtained from very low bit-rate audio coding system is improved by using expanding quantizers and arithmetic coding in a transmitter and using complementary compression and arithmetic decoding in a receiver. An expanding quantizer is used to control the number of signal components that are quantized to zero and arithmetic coding is used to efficiently code the quantized-to-zero coefficients. This allows a wider bandwidth and more accurately quantized baseband signal to be conveyed to the receiver, which regenerates an output signal by synthesizing the missing components.

Abstract: A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.

Abstract: An indication of the loudness of an audio signal containing speech and other types of audio material is obtained by classifying segments of audio information as either speech or non-speech. The loudness of the speech segments is estimated and this estimate is used to derive the indication of loudness. The indication of loudness may be used to control audio signal levels so that variations in loudness of speech between different programs is reduced. A preferred method for classifying speech segments is described.

Abstract: The perceived quality of an audio signals obtained from very low bit-rate audio coding system is improved by using expanding quantizers and arithmetic coding in a transmitter and using complementary compression and arithmetic decoding in a receiver. An expanding quantizer is used to control the number of signal components that are quantized to zero and arithmetic coding is used to efficiently code the quantized-to-zero coefficients. This allows a wider bandwidth and more accurately quantized baseband signal to be conveyed to the receiver, which regenerates an output signal by synthesizing the missing components.

Abstract: Audio coding processes like quantization can cause spectral components of an encoded audio signal to be set to zero, creating spectral holes in the signal. These spectral holes can degrade the perceived quality of audio signals that are reproduced by audio coding systems. An improved decoder avoids or reduces the degradation by filling the spectral holes with synthesized spectral components. An improved encoder may also be used to realize further improvements in the decoder.

Abstract: A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal. Spectral components are synthesized having the assessed characteristics. The synthesized spectral components are integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. In one implementation, the assessed characteristic is temporal shape and noise-like spectral components are synthesized having the temporal shape of the audio signal.