Showing papers by "Dolby Laboratories published in 1998"

Frame-based audio coding with video/audio data synchronization by dynamic audio frame alignment

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. In yet a further technique, audio blocks are dynamically aligned so that proper synchronization can be maintained across a splice. An example for 48 kHz audio with NTSC video is discussed.

Multidirectional audio decoding

Abstract: An acoustic crossfeed canceller, intended for implementation in software, such that when run in real time on a personal computer, the canceller has very low mips requirements and uses a small fraction of available CPU cycles. Thus, for example, the program could be included with video games, CD-ROMs, Internet audio and the like, rendering surround sound images outside the space between left and right computer multimedia loudspeakers when the audio from such sources is reproduced.

Frame-based audio coding with gain-control words

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. In yet a further technique, audio blocks are dynamically aligned so that proper synchronization can be maintained across a splice. An example for 48 kHz audio with NTSC video is discussed.

Frame-based audio coding with additional filterbank to attenuate spectral splatter at frame boundaries

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. In yet a further technique, audio blocks are dynamically aligned so that proper synchronization can be maintained across a splice. An example for 48 kHz audio with NTSC video is discussed.

Frame-based audio coding with additional filterbank to suppress aliasing artifacts at frame boundaries

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. In yet a further technique, audio blocks are dynamically aligned so that proper synchronization can be maintained across a splice. An example for 48 kHz audio with NTSC video is discussed.

Frame-based audio coding with video/audio data synchronization by audio sample rate conversion

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. In yet a further technique, audio blocks are dynamically aligned so that proper synchronization can be maintained across a splice. An example for 48 kHz audio with NTSC video is discussed.

Method and apparatus for encoding and decoding multiple audio channels at low bit rates

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.

Are Movies Too Loud

Abstract: Over the past few years, the subject of movies getting louder has generated increasing concern within the film production community. In addition, it has become commonplace for moviegoers to claim that movies are too loud. Many theatres now project films at an audio fader setting below the calibration setting that would match that in the dubbing theatre, presumably because of audience complaints. The following material discusses some of the issues that can be described as sound-track loudness. What is a measure of loudness for a movie? Why do theatres turn the sound down? Have new sound formats (Dolby SR and more recently the digital formats, Dolby SR.D, Sony SDDS, and DTS) exacerbated the perceived loudness problem? This paper describes an investigation aimed at defining perceived long-term sound-track loudness with a meter reading, and some data are presented.

Selecting analog or digital motion picture sound tracks

Abstract: A storage medium carries symbols representing digital information encoded in two dimensions. A method and apparatus recovers the symbols using oversampling in two dimensions and derives the information represented by the symbols. In one embodiment, the symbols are carried on motion picture film between the sprocket holes.

Avoiding forbidden data patterns in coded audio data

Abstract: Any of several information processing techniques may be used in various information storage and transmission applications to prevent the occurrence of certain “forbidden” bit patterns. According to an encoding technique, a reversible coding process is used to generate an encoded representation of an information stream that cannot contain any forbidden data patterns. This may be accomplished by partitioning the information stream into segments and encoding each segment according to a respective encoding key that is selected such that the results of the coding process cannot contain a forbidden data pattern. According to one substitution technique, all occurrences of forbidden data patterns are replaced with permissible data patterns that do not otherwise occur in the information stream. This may be accomplished by partitioning the information stream into segments, identifying an unused data pattern in a respective segment, and carrying out the replacement of all occurrences of the forbidden data pattern in that segment. According to another substitution technique, all occurrences of a forbidden data pattern are replaced by any permissible data pattern. This may be accomplished by partitioning the information stream into segments, identifying occurrences of the substitution data pattern and the forbidden data pattern in a respective segment, constructing a flag for each occurrence, and replacing all occurrences of the forbidden data pattern in that segment with the substitution data pattern. Decoding keys, substitution data patterns, substitution flags or any other information needed to recover the original information is assembled with the modified information in a form that does not equal the forbidden data pattern.

Compatible digital soundtracks for 70 mm motion picture film

Abstract: Dolby Digital and Sony SDDS digital signal carrying optical symbols on large-format motion picture film (e.g., film having any of the various 70 mm formats) are recorded in such a way as to permit existing 35 mm optical readers and signal decoders to recover the soundtracks with little or no modification. The normal format of a motion picture soundtrack in the form of a two-dimensional array of photographically recorded symbols (10) representing digital information intended for application to 35 mm motion picture film is modified by stretching the symbols (10) in the direction in which the motion picture film is transported when played, the degree of stretching being commensurate with the increase in film speed of the large-format motion picture film with respect to said 35 mm motion picture film. The stretched format soundtrack is applied to the large-format motion picture film, whereby when said large-format motion picture film is played the information recovered is essentially the same as the information recovered from 35 mm motion picture film to which the normal format of said motion picture soundtrack is applied.

Metadata Issues for ATSC Audio

Abstract: The audio decoder in a consumer digital television (DTV) receiver conforming to the Advanced Television Systems Committee (ATSC) specification needs two types of data in order to work properly. The data stream it receives must contain coded audio as well as audio metadata. Audio metadata is a group of parameters that describes the audio program data being transmitted and controls some aspects of how the audio program is presented in the consumer's receiver. This paper will provide an overview of the audio metadata parameters. It will discuss the purpose of the parameters and how they are chosen. The paper will also discuss the effect that each of these parameters has on domestic receiver operation. The operational implications of authoring the different parameters at different points in the production or distribution chain will be examined. Some different methods for authoring the metadata will also be proposed. These methods will include those necessary during the transition period when proper metadata authoring and monitoring equipment may not exist.