Abstract: Methods, apparatuses and program logic in non-transitory media to process video data for quality enhancement. Information is accepted from a resource constrained device, e.g., a wireless portable device related to the quality enhancement and/or environmental quantities such as ambient lighting for the device. The video data is processed to achieve quality enhancement using at least some of the accepted information to generate processed output. The processing of the video data includes processing when or where one or more resources sufficient for the processing are available.

Abstract: Ambience signal components are obtained from source audio signals, matrix-decoded signal components are obtained from the source audio signals, and the ambience signal components are controllably combined with the matrix-decoded signal components. Obtaining ambience signal components may include applying at least one decorrelation filter sequence. The same decorrelation filter sequence may be applied to each of the input audio signals or, alternatively, a different decorrelation filter sequence may be applied to each of the input audio signals.

Abstract: Methods, apparatuses and program logic in non-transitory media to process video data for quality enhancement. Information is accepted from a resource constrained device, e.g., a wireless portable device related to the quality enhancement and/or environmental quantities such as ambient lighting for the device. The video data is processed to achieve quality enhancement using at least some of the accepted information to generate processed output. The processing of the video data includes processing when or where one or more resources sufficient for the processing are available.

Abstract: During production, at least one audio signal is processed in order to derive instructions for channel reconfiguring it. The at least one audio signal and the instructions are stored or transmitted. During consumption, the at least one audio signal is channel reconfigured in accordance with the instructions. Channel reconfiguring includes upmixing, downmixing, and spatial reconfiguration. By determining the channel reconfiguration instructions during production, processing resources during consumption are reduced.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

Abstract: Ambience signal components are obtained from source audio signals, matrix-decoded signal components are obtained from the source audio signals, and the ambience signal components are controllably combined with the matrix-decoded signal components. Obtaining ambience signal components may include applying at least one decorrelation filter sequence. The same decorrelation filter sequence may be applied to each of the input audio signals or, alternatively, a different decorrelation filter sequence may be applied to each of the input audio signals.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness and dynamic range of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

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 process that allocates bits for quantizing spectral components in a perceptual coding system is performed more efficiently by obtaining an accurate estimate of the optimal value for one or more coding parameters that are used in the bit allocation process. In one implementation for a perceptual audio coding system, an accurate estimate of an offset from a calculated psychoacoustic masking curve is derived by selecting an initial value for the offset, calculating the number of bits that would be allocated if the initial offset were used for coding, and estimating the optimum value of the offset from a difference between this calculated number and the number of bits that are actually available for allocation.

Abstract: During production, at least one audio signal is processed in order to derive instructions for channel reconfiguring it. The at least one audio signal and the instructions are stored or transmitted. During consumption, the at least one audio signal is channel reconfigured in accordance with the instructions. Channel reconfiguring includes upmixing, downmixing, and spatial reconfiguration. By determining the channel reconfiguration instructions during production, processing resources during consumption are reduced.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness and dynamic range of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

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: 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: Techniques like Huffman coding can be used to represent digital audio signal components more efficiently using non-uniform length symbols than can be represented by other coding techniques using uniform length symbols Unfortunately, the coding efficiency that can be achieved by Huffman coding depends on the probability density function of the information to be coded and the Huffman coding process itself requires considerable processing and memory resources. A coding process that uses gain-adaptive quantization according to the present invention can realize the advantage of using non-uniform length symbols while overcoming the shortcomings of Huffman coding. In gain-adaptive quantization, the magnitudes of signal components to be encoded are compared to one or more thresholds and placed into classes according to the results of the comparison. The magnitudes of the components placed into one of the classes are modified according to a gain factor that is related to the threshold used to classify the components.

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.