Institution
Dolby Laboratories
Company•Amsterdam, Netherlands•
About: Dolby Laboratories is a company organization based out in Amsterdam, Netherlands. It is known for research contribution in the topics: Audio signal & Audio signal flow. The organization has 956 authors who have published 1726 publications receiving 29456 citations.
Papers published on a yearly basis
Papers
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07 Apr 2000TL;DR: In this article, a method and apparatus for image compression using temporal and resolution layering of compressed image frames is proposed, which allows a form of modularized decomposition of an image that supports flexible application of a variety of image enhancement techniques.
Abstract: A method and apparatus for image compression using temporal and resolution layering of compressed image frames. In particular, layered compression allows a form of modularized decomposition of an image that supports flexible application of a variety of image enhancement techniques. Further, the invention provides a number of enhancements to handle a variety of video quality and compression problems. Most of the enhancements are preferably embodied as a set of tools which can be applied to the tasks of enhancing images and compressing such images. The tools can be combined by a content developer in various ways, as desired, to optimize the visual quality and compression efficiency of a compressed data stream, particularly a layered compressed data stream. Such tools include improved image filtering techniques, motion vector representation and determination, de-interlacing and noise reduction enhancements, motion analysis, imaging device characterization and correction, an enhanced 3-2 pulldown system, frame rate methods for production, a modular bit rate technique, a multi-layer DCT structure, variable length coding optimization, an augmentation system for MPEG-2 and MPEG-4, and guide vectors for the spatial enhancement layer.
353 citations
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TL;DR: In this article, a low bit-rate (192 kBits per second) transform encoder/decoder system (44.1 kHz or 48 kHz sampling rate) for high quality music applications employs short time-domain sample blocks (128 samples/block) so that the system signal propagation delay is short enough for real-time aural feedback to a human operator.
Abstract: A low bit-rate (192 kBits per second) transform encoder/decoder system (44.1 kHz or 48 kHz sampling rate) for high-quality music applications employs short time-domain sample blocks (128 samples/block) so that the system signal propagation delay is short enough for real-time aural feedback to a human operator. Carefully designed pairs of analysis/synthesis windows are used to achieve sufficient transform frequency selectivity despite the use of short sample blocks. A synthesis window in the decoder has characteristics such that the product of its response and that of an analysis window in the encoder produces a composite response which sums to unity for two adjacent overlapped sample blocks. Adjacent time-domain signal samples blocks are overlapped and added to cancel the effects of the analysis and synthesis windows. A technique is provided for deriving suitable analysis/synthesis window pairs. In the encoder, a discrete transform having a function equivalent to the alternate application of a modified Discrete Cosine Transform and a modified Discrete Sine Transform according to the Time Domain Aliasing Cancellation technique or, alternatively, a Discrete Fourier Transform is used to generate frequency-domain transform coefficients. The transform coefficients are nonuniformly quantized by assigning a fixed number of bits and a variable number of bits determined adaptively based on psychoacoustic masking. A technique is described for assigning the fixed bit and adaptive bit allocations. The transmission of side information regarding adaptively allocated bits is not required. Error codes and protected data may be scattered throughout formatted frame outputs from the encoder in order to reduce sensitivity to noise bursts.
341 citations
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TL;DR: In this article, the tradeoff between time resolution and frequency resolution is optimized by adaptively selecting the transform block length for each sampled audio segment, and/or can optimize coding gain by adapting the transform and analysis window or the analysis/synthesis window pair.
Abstract: 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.
275 citations
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08 Jan 1992TL;DR: In this paper, the authors propose a subband encoding and decoding scheme for multidimensional sound fields, in which the encoded signals may be carried by multiple discrete signals and/or a composite signal with a control signal conveying either the relative levels of encoded signals, or the apparent direction of the sound field represented by the encoded signal.
Abstract: The invention relates in general to the recording, transmitting, and reproducing of multidimensional sound fields intended for human hearing. More particularly, the invention relates to subband encoding and decoding of signals representing such sound fields, wherein the encoded signals may be carried by multiple discrete signals and/or a composite signal with a control signal conveying either the relative levels of the encoded signals, or the apparent direction of the sound field represented by the encoded signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signals.
235 citations
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27 Jun 2012TL;DR: In this article, the authors describe an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams, which are associated with metadata that specifies whether the stream is a channel-based or object-based stream.
Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the sound correctly using the available speaker positions in a room that is set up to play the adaptive audio content.
231 citations
Authors
Showing all 959 results
Name | H-index | Papers | Citations |
---|---|---|---|
Wolfgang Heidrich | 64 | 312 | 15854 |
Rabab K. Ward | 56 | 549 | 14364 |
Lorne A. Whitehead | 42 | 232 | 6661 |
Scott J. Daly | 41 | 230 | 5543 |
Michael E. Miller | 40 | 225 | 5264 |
Alireza Marandi | 39 | 140 | 6116 |
Wolfgang Stuerzlinger | 35 | 230 | 5192 |
Lars Villemoes | 33 | 180 | 2815 |
Joan Serrà | 31 | 139 | 4046 |
Dong Tian | 31 | 116 | 3621 |
Peng Yin | 30 | 133 | 2454 |
Ning Xu | 28 | 117 | 2705 |
Nicolas R. Tsingos | 28 | 110 | 2749 |
Panos Nasiopoulos | 27 | 271 | 3706 |
Zhibo Chen | 27 | 344 | 3385 |