Abstract: An audio signal processor forms gaps or guard bands in sequences of blocks conveying encoded audio information and time aligns the guard bands with video information. The guard bands are formed to allow for variations in processing or circuit delays so that the routing or switching of different streams of video information with embedded audio information does not result in a loss of any encoded audio blocks.

Abstract: In an encoder, multiple channels of audio information representing multidimensional sound fields are split into subband signals and the subband signals in one or more subbands are combined to form composite signals. The composite signals, the subband signals not combined into a composite signal and information describing the spectral levels of subband signals combined into composite signals are assembled into an encoded output signal. The spectral level information conveys either the amplitude or power of the combined subband signals or the apparent direction of the sound field represented by the combined subband signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signal.

Abstract: When an audio stream signal is transmitted, the bit stream information for determining the surround mode and special-use mode, including karaoke use, is sensed from the signal at a mode sensor. After the surround mode has been sensed, the audio data is converted by a decoder into a front left, front center, and front right main audio signals and a back left and back right surround audio signals in the surround mode, which are converted into analog signals and outputted to the corresponding speakers. When the special-use mode has been sensed, by using the central main audio signal of the front left, front center, and front right main audio signals and back left and back right surround audio signals converted at the decoder, a normally used first-type accompanying sound selectively made unused, for example, guide melody, is generated. In addition, by using the back left and back right surround audio signals, a normally unused second-type accompanying sound selectively used, for example, vocals, is generated.

Abstract: Several audio signal processing techniques may be used in various combinations to improve the quality of audio represented by an information stream formed by splice editing two or more other information streams. The techniques are particularly useful in applications that bundle audio information with video information. In one technique, gain-control words conveyed with the audio information stream are used to interpolate playback sound levels across a splice. In another technique, special filterbanks or forms of TDAC transforms are used to suppress aliasing artifacts on either side of a splice. In yet another technique, special filterbanks or crossfade window functions are used to optimize the attenuation of spectral splatter created at a splice. In a further technique, audio sample rates are converted according to frame lengths and rates to allow audio information to be bundled with, for example, video information.

Abstract: In an encoder, multiple channels of audio information representing multidimensional sound fields are split into subband signals and the subband signals in one or more subbands are combined to form composite signals. The composite signals, the subband signals not combined into a composite signal and information describing the spectral levels of subband signals combined into composite signals are assembled into an encoded output signal. The spectral level information conveys either the amplitude or power of the combined subband signals or the apparent direction of the sound field represented by the combined subband signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signal.

Abstract: Several audio signal processing techniques may be used in various combinations to improve the quality of audio represented by an information stream formed by splice editing two or more other information streams. The techniques are particularly useful in applications that bundle audio information with video information. In one technique, gain-control words conveyed with the audio information stream are used to interpolate playback sound levels across a splice. In another technique, special filterbanks or forms of TDAC transforms are used to suppress aliasing artifacts on either side of a splice. In yet another technique, special filterbanks or crossfade window functions are used to optimize the attenuation of spectral splatter created at a splice. In a further technique, audio sample rates are converted according to frame lengths and rates to allow audio information to be bundled with, for example, video information.

Abstract: When an audio stream signal is transmitted, the bit stream information for determining the surround mode and special-use mode, including karaoke use, is sensed from the signal at a mode sensor. After the surround mode has been sensed, the audio data is converted by a decoder into a front left, front center, and front right main audio signals and a back left and back right surround audio signals in the surround mode, which are converted into analog signals and outputted to the corresponding speakers. When the special-use mode has been sensed, by using the central main audio signal of the front left, front center, and front right main audio signals and back left and back right surround audio signals converted at the decoder, a normally used first-type accompanying sound selectively made unused, for example, guide melody, is generated. In addition, by using the back left and back right surround audio signals, a normally unused second-type accompanying sound selectively used, for example, vocals, is generated.

Abstract: A multi-channel signal compressor for compressing digital sound signals in the respective channels of a multi-channel sound system. The apparatus comprises a first-stage compression system and a second-stage compression system. In the first-stage compression system, a coupling circuit performs coupling between the digital sound signals of at least two of the channels to generate coupling-processed signals, one for each of the channels. A compressor circuit receives the coupling-processed signals from the coupling circuit and frequency divides each coupling-processed signal into frequency range signals in respective frequency ranges, and compresses the frequency range signals obtained by dividing each coupling-processed signal to generate a first-stage compressed signal.