A method for decoding auxiliary data from media signals in multimedia content decodes watermarks from different media signals and uses the watermarks to control processing of the multimedia content. A copy control method decodes a watermark from one of the media signals in multimedia content, and uses the watermark to control processing of the multimedia content. Another method uses a watermark decoded from a first media signal to decode a second media signal. Yet another method uses a watermark decoded from a media signal to decode metadata associated with the media signal. Finally, another method forms a key for decoding data from at least first and second watermarks extracted from first and second media signals.

Integrating digital watermarks in multimedia content

This paper considers externally linear, time-invariant systems which can be internally nonlinear and/or time-varying. Several examples of the synthesis of such systems are presented, and it is argued that they offer advantages in comparison to internally linear, time-invariant systems. In particular, their use in signal processing is considered, and it is shown that they can be designed to reduce the undesirable effects of overloading and noise corruption through the use of companding. It is shown that a variety of previously proposed schemes for this purpose are all related. Several practical problems are discussed, as are performance criteria and types of measurements needed for the evaluation of the systems discussed.

Externally linear, time-invariant systems and their application to companding signal processors

A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

Scalable and embedded codec for speech and audio signals

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal

Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method generating a third downmix signal by combining a first downmix signal extracted from a first audio signal and a second downmix signal extracted from a second audio signal; generating third object-based side information by combining first object-based side information extracted from the first audio signal and second object-based side information extracted from the second audio signal; converting the third object-based side information into channel-based side information; and generating a multi-channel audio signal using the third downmix signal and the channel-based side information.

Methods and apparatuses for encoding and decoding object-based audio signals

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.

Encoder/decoder for multidimensional sound fields

The invention concerns a dynamic range compressor type encoder or expander type decoder, in which a main signal component in a main path is boosted or bucked by a further signal component derived from a point in the main path by a further path having the characteristics of so restricting the further signal component that the boosting or bucking action is only appreciable below a low level threshold. In the present invention the further signal component is a difference signal formed between a direct signal derived from a point in the main path and a delayed version of either the same signal or of another signal derived from another point in the main path. At the frequency equal to the reciprocal of the delay, and at harmonics of this frequency, the direct and delayed signals cancel. The compressor or expander action, and hence noise reduction action, takes place only at intervening frequencies. The invention enables carrier components or other repetitive components of signals to be excluded from the compressor or expander action (which they would otherwise choke).

Signal compressors and expanders

In a stereophonic reproduction system, especially for a cinema, a center channel signal is derived by combining left and right channel signals. In order to harden the center image, particularly in speech, the left and right channel signals are compared and, when they include significant amounts of correlated information, the relative gain of the center channel is enhanced by boosting the gain of that channel and/or cutting the gain of the left and right channels. The test for correlation of information is preferably restricted to speech frequencies and the enhancement of relative gain may be performed only at middle frequencies.

Centre channel derivation for stereophonic cinema sound

Apparatus for scanning variable area optical sound tracks wherein a plurality of discrete photo sensors examine small fractional portions of the sound track width. Selection of the number of sensors in relation to photographic grain dimensions results in converting annoying quantizing noise to more tolerable random noise while using a practical number of sensors. Electronic processing of the sensors' outputs reduces impulse noise from dirt and scratches.

Apparatus for scanning variable area optical sound tracks by parallel processing discrete photosensor outputs

Three related embodiments of audio signal dynamic range circuits for consumer and semi-professional audio signal compressors, expanders, and compressor/expander noise reduction systems incorporate certain basic features of professional Spectral Recording (SR) while adapting SR for use with consumer recording media (particularly audio cassette tape) and reducing the cost and complexity of the SR circuitry In addition, the three embodiments, and particularly the preferred embodiment, provide substantial compatibility with the existing B-type and C-type consumer and semi-professional noise reduction systems Subjective performance substantially equivalent to 16-bit PCM audio systems is achieved with analog audio cassette tape media In all three embodiments there are two cascaded stages: (1) a high-level stage a high-frequency fixed-band/sliding band action-substitution sub-stage (324A) and a low-frequency fixed-band sub-stage (328A) and (2) a low-level stage having only a high-frequency fixed-band/sliding band action-substitution sub-stage (326A) The dynamic action regions of the high-frequency action-substitution sub-stages are staggered Each of the high-frequency sub-stages provides up to 12 dB of dynamic action (together providing up to 24 dB of dynamic action) and operates in a frequency band defined by a single-pole filter having a high-pass shelving response with a corner frequency of 400 Hz The low-frequency sub-stage provides up to 10 dB of dynamic action and operates in a frequency band defined buy a single-pole filter having a low-pass shelving response with a corner frequency of 200 Hz The single-pole filter characterisctics provide a broad and smooth overlap of the high-frequency and low-frequency characteristics Particular forms of spectral skewing and antisaturation enhance the performance of the circuits with consumer recording media

Ray Milton Dolby

Papers

Encoder/decoder for multidimensional sound fields

Signal compressors and expanders

Centre channel derivation for stereophonic cinema sound

Apparatus for scanning variable area optical sound tracks by parallel processing discrete photosensor outputs

Audio compressor, expander, and noise reduction circuits for consumer and semi-professional use