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Showing papers by "Dolby Laboratories published in 1997"


Patent
16 Jul 1997
TL;DR: A split-band coding system combines multiple channels of input signals into various forms of composite signals and generates spatial-characteristic signals representing soundfield spatial characteristics in a plurality of frequency subbands.
Abstract: A split-band coding system combines multiple channels of input signals into various forms of composite signals and generates spatial-characteristic signals representing soundfield spatial characteristics in a plurality of frequency subbands. The spatial-characteristics signals may be generated in either or both of two forms. In a first form, the signal represents measures of signal levels for subband signals from the input channels. In a second form, the signal represents one or more apparent directions for the soundfield. The type of the spatial-characteristics signal may be adapted dynamically in response to a variety of criteria including input signal characteristics. Temporal smoothing and spectral smoothing of the spatial-characteristics signals may be applied in an encoder. Temporal smoothing and spectral smoothing may be applied to gain factors derived from the spatial-characteristics signals in a decoder.

123 citations


Patent
21 Feb 1997
TL;DR: In this article, the phase shift is achieved by applying a signal to two phase-shifting processes, producing two signals whose relative phase difference is sufficiently close to the desired phase shift over at least a substantial part of the frequency band of interest.
Abstract: A surround sound encoder, intended for implementation in software, runs in real time on a personal computer using low mips and a small fraction of available CPU cycles. In the principal application for the encoder, the Lt and Rt signals of the encoder are mixed with the Lt and Rt signals of a pre-recorded source (e.g., computer game soundtrack, CD ROM, Internet audio, etc.). Alternatively, the encoder may be used by itself or with one or more other virtual encoders to provide a totally user-generated soundfield. The encoder is implemented in either of two ways: the signal being encoded may be panned to one or more of the four inputs of a surround-sound fixed matrix encoder or the signal may be encoded by applying the signal to a surround-sound variable-matrix encoder. Phase shifting, required in the encoder, is achieved by applying a signal to two phase-shifting processes, producing two signals whose relative phase difference is sufficiently close to the desired phase shift over at least a substantial part of the frequency band of interest. Satisfactory audible results may be achieved, using very low computer processing power, when one of the phase shifting processes is implemented by a first-order all-pass filter and the other phase shifting process is implemented by only a short time delay, which also has an all-pass characteristic.

65 citations


Patent
TL;DR: In this article, 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, and 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, which are converted into analog signals and outputted to the corresponding speakers.
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. The first- and second-type accompanying sounds, together with the front left and front right main audio signals, are reproduced selectively. Consequently, a system that transmits surround sound using a plurality of channels can maintain compatibility with a surround system easily, when the plurality of channels are applied to special use, such as karaoke, not being restricted to surround use.

58 citations


Patent
19 Mar 1997
TL;DR: In this article, an efficient implementation of oddly-stacked critically-sampled single sideband analysis/synthesis filter banks is achieved by application of a set of functions to time-domain and frequency-domain values before and after transformation.
Abstract: An efficient implementation of oddly-stacked critically-sampled single sideband analysis/synthesis filter banks is achieved by application of a set of functions to time-domain and frequency-domain values before and after transformation. In one embodiment of an analysis filter bank, a forward pre-transform function groups blocks of N samples into blocks of 1/4N modified samples, a discrete transform generates frequency-domain coefficients in response to the modified samples, and a forward post-transform function generates spectral information in response to the frequency-domain transform coefficients. In one embodiment of a synthesis filter bank, an inverse pre-transform function groups spectral information into blocks of 1/4N frequency-domain transform coefficients, a discrete transform generates blocks of 1/4N time-domain transform coefficients in response to the frequency-domain transform coefficients, and an inverse post-transform function generates blocks of N time-domain samples in response to the time-domain transform coefficients. An implementation of an oddly-stacked Time Domain Aliasing Cancellation transform permits the length of the transformation to be adaptively selected.

31 citations


Patent
23 May 1997
TL;DR: In this paper, multiple channels of audio information representing multidimensional sound fields are split into subband signals, and the subbands signals in one or more subbands are combined to form composite signals.
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.

18 citations


Patent
14 Feb 1997
TL;DR: In this paper, a mode detector 163 is used to detect the bit stream information which decides a surrounding mode and a specific application mode such as KARAOKE application, and a first type of fundamentally used subordinate voice set is generated by using a center main audio signal out of left, center and right front main audio signals and left and right rear surrounding audio signals.
Abstract: PROBLEM TO BE SOLVED: To provide an audio system capable of easily keeping inter-changeability with a surrounding system when it is used in a specific application such as KARAOKE, etc. SOLUTION: When an audio stream signal is transmitted, bit stream information which decides a surrounding mode and a specific application mode such a KARAOKE application, etc., is detected by a mode detector 163. When the surrounding mode is detected, ordinary surrounding reproduction is performed. When the specific application mode is detected, a first type of fundamentally used subordinate voice set as a selectively fundamentally disused one is generated by using a center main audio signal out of left, center and right front main audio signals and left and right rear surrounding audio signals, and a second type of fundamentally disued subordinate voice used selectively by using the left and right rear surrounding audio signals is generated. The first and second types of subordinate voices are reproduced selectively with the main audio signal.

12 citations


Journal ArticleDOI
TL;DR: In this paper, the authors have developed a three-way system for both small and large venues, and an installation of five new screen channels has been made in the Samuel Goldwyn Theater at the Academy of Motion Picture Arts and Sciences.
Abstract: Digital soundtracks have the capability of delivering lower distortion program signals with flat power bandwidth in the motion picture theater. Traditional two-way theater loudspeaker systems are presently being pushed far beyond their original performance limits with the new soundtracks, and three-way systems are now considered to be their future replacement. The authors have developed such systems for both small and large venues, and an installation of five new screen channels has been made in the Samuel Goldwyn Theater at the Academy of Motion Picture Arts and Sciences. System evolution, design, and performance will be discussed in detail in this paper.

1 citations