Dolby Laboratories Licensing Corporation

Dolby Laboratories Corporation

Kenneth James Gundry

Alan Jeffrey Seefeldt

Jeffrey Charles Riedmiller

Mark Stuart Vinton

Charles Quito Robinson

Grant Allen Davidson

Michael John Smithers

Craig C Todd

Steven Joseph Venezia

Louis Dunn Fielder

Michael Mead Truman

Matthew Conrad Fellers

Rongshan Yu

James W Fosgate

Stephen Decker Vernon

Roger Wallace Dressler

Marina Bosi-Goldberg

John Iles

Eric Benjamin

Michael J Smithers

Stephen D Vernon

Eric M Benjamin

An audio dynamics processor or processing method that uses a reset mechanism or process in order to adapt quickly to content changes in the audio signal. A reset signal may be generated by analyzing the audio signal itself or the reset may be triggered from an external event such as a channel change on a television set or an input selection change on an audio/visual receiver. In the case of an external trigger, one or more indicators of the state of the dynamics processor for a current audio source may be saved and associated with that audio source before switching to a new audio source. Then, if the system switches back to the first audio source, the dynamics processor may be reset to the state previously stored or an approximation thereof.

Audio dynamics processing using a reset

An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.

Method for adaptive control and equalization of electroacoustic channels

Information useful for modifying the dynamics of an audio signal is derived from one or more devices or processes operating at one or more respective nodes of each of a plurality of hierarchy levels, each hierarchical level having one or more nodes, in which the one or more devices or processes operating at each hierarchical level takes a measure of one or more characteristics of the audio signal such that the one or more devices or processes operating at each successively lower hierarchical level takes a measure of one or more characteristics of progressively smaller subdivisions of the audio signal.

Hierarchical control path with constraints for audio dynamics processing

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.

Controlling loudness of speech in signals that contain speech and other types of audio material

A single control determines both volume and the degree of compression in the reproduction of motion picture soundtracks. For some settings of the control, compression or limiting reduces the highest levels on a movie soundtrack, leaving the dialogue level substantially unchanged, thus removing the reason for complaints from the audience without the danger of making the dialogue too quiet for intelligibility.

Volume and compression control in movie theaters

Methods for splicing PCM audio frames form modified frames by appending to each frame either a portion of the next preceding frame or the next following frame. According to a first approach, splices are obtained by fading up and fading down a frame end and a frame appendage only at a splice point, and overlapping and combining to provide a crossfade at the splice. Alternatively, every frame end and frame appendage is faded up and faded down, overlapped and combined. A subtractive method of providing complementary fade-up and fade-down reduces rounding errors.

Frame-based audio transmission/storage with overlap to facilitate smooth crossfading

A multidirectional audio decoder using an 'adaptive' audio matrix derives at least one of a plurality of output audio signals from two or more directionally-encoded audio input signal streams (S(&#x3b1;), S(&#x3b1;), . . . SN(&#x3b1;), wherein &#x3b1; is the encoded angle of a source audio signal. Each output signal is associated with a principal direction &#x3b2;. In order to generate each output signal, a pair of intermediate signals ('antidominant' signals) are generated, constituting the antidominant signal for each of the two adjacent principal output directions of the decoder. The antidominant signal for any arbitrary principal (or &#x201c;dominant&#x201d;) direction is the combination of input signals having coefficients such that the combination goes to zero for that dominant direction. Amplitude control is applied to the two antidominant signals to deliver a pair of signals having substantially equal magnitudes that are additively or subtractively combined to provide the output audio signal associated with a principal direction. The pair of signals that are combined include passive matrix components. In an alternative embodiment, passive matrix components for the output signal are derived instead by applying the input audio signals to a passive matrix and by combining the passive matrix components with pairs of amplitude controlled versions of the antidominant signals that do not include passive matrix components. For sources from one direction at a time, there is little or no unwanted crosstalk into outputs that should be silent (ie., there is substantially no signal in outputs other than the two representing directions adjacent to the desired direction, except when the desired direction happens to correspond to that of an output, in which case there is a signal in substantially only that output).

Method and apparatus for deriving at least one audio signal from two or more input audio signals

Three surround sound channels are provided within the current formats of the Dolby Digital, Sony SDDS and DTS digital soundtrack systems in a manner that provides compatibility with conventional two surround channel playback in standard 5.1 channel and 7.1 channel systems while allowing the soundtrack preparer to send the same signal to all surround sound channels and preserving the ability to pan among the three matrix decoded surround sound channels in an arrangement that employs an active matrix decoder to provide the three surround sound channels.

Compatible matrix-encoded surround-sound channels in a discrete digital sound format

Codeword-position-caused encoder latency is reduced by avoiding the requirement for knowledge of the message prior to generating an error detecting or concealing codeword associated with the message. A pseudo error detecting or concealing codeword is inserted in place of the normal error detecting or concealing codeword appropriate for the segment of information to which the error detecting or concealing codeword relates. In order to satisfy the requirement of conventional decoders, the pseudo error detecting or concealing information must match or be appropriate for the segment so that the decoder sees the codeword and message segment as valid or error free. This is accomplished by modifying or perturbing at least a portion of the segment to which the pseudo codeword relates. The invention is particularly useful for maintaining the backward compatibility of audio data encoding formats in which the minimum latency is too long (e.g., computer games, where the player performs some operation leading to a sound, and that sound must not be perceptibly delayed with respect to the operation).

Low latency data encoder

A signal processing system comprising components such as split-band perceptual coders can receive a peak amplitude limited input audio signal and can process the signal in such a manner that the processed signal preserves the apparent loudness of the input signal but is no longer peak-amplitude limited. In one embodiment, upsampling is used in estimating the resultant peak amplitude and gain factors established in response to the estimated peak amplitude are applied to one or more frequency subbands of the processed signal.

Digital audio limiter

One-bit analog-to-digital converters (ADCs) and digital-analog-converters (DACs) employ an adaptive filter. The filter has two regimes of operation: variable gain within the passband of the filter under low-level signal conditions and fixed gain but a variable filter cutoff frequency (sliding band) under high-level signal conditions. Thus, excessive low-frequency gain under no-signal conditions is avoided and, when implemented using a voltage controlled amplifier (VCA), the arrangement does not demand less offset from the VCA under low-level signal conditions.

One-bit analog-to-digital converters and digital-to-analog converters

An encoder converts an analog input signal to a digital signal. The encoder comprises a circuit for generating a control signal with a predetermined bandwidth from the input analog signal. The control signal indicates the desired optimum step-size to be applied in the encoder. The encoder further comprises a converter for converting the analog input signal to a digital signal. The converter is responsive to the control signal representing step-size so that the digital signal will be generated in accordance with the step-size information. Since the generation of a limited bandwidth control signal requires time, the encoder also includes a circuit for introducing time delay to the analog input signal, so that the delayed input signal reaches the converter when the control signal is available for digital encoding. A decoder generates an analog output signal from a digital signal received from an encoder through a transmission medium. The digital signal received includes a digital representation of an analog input signal to the encoder and step-size information which was used to digitally encode the analog signal in the encoder. The decoder comprises a converter responsive to a second control signal of limited bandwidth derived from the step-size information for generating from the digital representation an analog output signal. The analog output signal so generated is similar to the input analog signal.

A-D encoder and D-A decoder system

The complementary pre-processing and post-processing circuits of this inventiona re advantageous for reducing noise introduced into a signal by a process or medium where the noise introduced rises with the signal level. The pre-processing circuit analyzes the spectral composition of an input signal and generates an emphasis control signal to so indicate. The emphasis control signal is limited by a limitation circuit to a desired bandwidth. The pre-processing circuit includes an emphasis applying circuit for amplifying or attenuating the input signal as a function of the emphasis control signal. A delay circuit delays the input signal relative to the emphasis control signal so that the input signal reaches the emphasis applying circuit when the emphasis control signal is available for controlling the amplification or attenuation applied. The post-processing circuit restores the spectral composition of the signal. The post-processing circuit comprises a circuit for deriving a de-emphasis control signal of limited bandwidth from the emphasis control signal. A de-emphasis applying circuit attenuates or amplifies the decoded signal as a function of the de-emphasis control signal to restore its spectral composition., Since the noise introduced by the process or medium is a function of the signal level, the pre-processing circuit alters the input signal in such manner that the noise introduced is reduced. The emphasis and de-emphasis applied in pre- and post-processing change slowly and the system is tolerant of component and timing errors.

Spectral emphasis and de-emphasis

The time-base corrector of this invention corrects the time-base of audio signals reproduced from video tapes without correcting the time-base of video signals which are associated with the audio signals. Reference signals such as the horizontal synchronizing pulses which accompany video information are used for correcting the time-base of audio signals if such reference signals have been subjected to substantially the same time compression or expansion as the audio signals. In the preferred embodiment, the reproduced audio signals are sampled and written into a sampled data delay and read out of it by means of a control circuit. The control circuit receives as input the undistorted reference signal and the reproduced reference signal which has undergone time compression or expansion. The control circuit derives a write clock signal with a frequency equal to k times the frequency of the distorted reference signal. The reproduced (and therefore time compressed or expanded) audio signal is sampled and read into the sampled data delay by the write clock signal. The audio signal stored in the sampled data delay is then read by the control circuit at a frequency of a read clock signal with a frequency equal to k times the frequency of the undistorted reference signal. The audio signal read in the above manner will be time-base corrected.

Growing community of inventors

San Francisco, CA, United States of America

Kenneth James Gundry