| DOLBY INTERNATIONAL AB Patent applications |
| Patent application number | Title | Published |
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| 20120128151 | Authentication of Data Streams - The present invention relates to techniques for authentication of data streams. Specifically, the invention relates to the insertion of identifiers into a data stream, such as a Dolby Pulse, AAC or HE AAC bitstream, and the authentication and verification of the data stream based on such identifiers. A method and system for encoding a data stream comprising a plurality of data frames is described. The method comprises the step of generating a cryptographic value of a number N of successive data frames and configuration information, wherein the configuration information comprises information for rendering the data stream. The method then inserts the cryptographic value into the data stream subsequent to the N successive data frames. | 05-24-2012 |
| 20120065983 | Efficient Combined Harmonic Transposition - The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular; a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described, The system may comprise an analysis filter bank ( | 03-15-2012 |
| 20120002818 | Advanced Stereo Coding Based on a Combination of Adaptively Selectable Left/Right or Mid/Side Stereo Coding and of Parametric Stereo Coding - The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant. | 01-05-2012 |
| 20110305352 | Cross Product Enhanced Harmonic Transposition - The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal. | 12-15-2011 |
| 20110302230 | LOW DELAY MODULATED FILTER BANK - The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a (64) channel filter bank using a prototype filter length of (640) coefficients and a system delay of (319) samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filterbanks used in high frequency reconstruction (HFR) or parametric stereo systems. | 12-08-2011 |
| 20110274281 | Method for Determining Inverse Filter from Critically Banded Impulse Response Data - A method for determining an inverse filter for altering the frequency response of a loudspeaker so that with the inverse filter applied in the loudspeaker's signal path the inverse-filtered loudspeaker output has a target frequency response, and optionally also applying the inverse filter in the signal path, and a system configured (e.g., a general or special purpose processor programmed and configured) to determine an inverse filter. In some embodiments, the inverse filter corrects the magnitude of the loudspeaker's output. In other embodiments, the inverse filter corrects both the magnitude and phase of the loudspeaker's output. In some embodiments, the inverse filter is determined in the frequency domain by applying eigenfilter theory or minimizing a mean square error expression by solving a linear equation system. | 11-10-2011 |
| 20110261966 | Method and Apparatus for Applying Reverb to a Multi-Channel Audio Signal Using Spatial Cue Parameters - A method and system for applying reverb to an M-channel down-mixed audio input signal indicative of X individual audio channels, where X is greater than M. Typically, the method includes steps of: in response to spatial cue parameters indicative of spatial image of the downmixed input signal, generating Y discrete reverb channel signals, where each of the reverb channel signals at a time, t, is a linear combination of at least a subset of values of the individual audio channels at the time, t, and individually applying reverb to each of at least two of the reverb channel signals, thereby generating Y reverbed channel signals. Preferably, the reverb applied to at least one of the channel signals has a different reverb impulse response than does the reverb applied to at least one other one of the channel signals, t, is a linear combination of at least a sub-set of values of the individual audio channels at the time, t, and individually applying reverb to each of at least two of the reverb channel signals, thereby generating Y reverbed channel signals. Preferably, the reverb applied to at least one of the channel signals has a different reverb impulse response than does the reverb applied to at least one other one of the channel signals. | 10-27-2011 |
| 20110208528 | SIGNAL CLIPPING PROTECTION USING PRE-EXISTING AUDIO GAIN METADATA - The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values ( | 08-25-2011 |
| 20110178810 | AUDIO SIGNAL ENCODING OR DECODING - Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of downsampled subband signals. Further, decoding is provided wherein an encoded audio signal comprising an encoded single channel audio signal and a set of spatial parameters is decoded by decoding the encoded single channel audio channel to obtain a plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, and deriving two audio channels from the spatial parameters, the sub-subband signals and those downsampled subband signals that are not further subband filtered. | 07-21-2011 |
| 20110004479 | HARMONIC TRANSPOSITION - The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L | 01-06-2011 |
| 20100286991 | AUDIO ENCODER AND DECODER - The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit. | 11-11-2010 |
| 20100286990 | AUDIO ENCODER AND DECODER - The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal. | 11-11-2010 |
| 20100246832 | METHOD AND APPARATUS FOR GENERATING A BINAURAL AUDIO SIGNAL - An apparatus for generating a binaural audio signal includes a de-multiplexer and decoder which receives audio data comprising an audio M-channel audio signal which is a downmix of an N-channel audio signal and spatial parameter data for upmixing the M-channel audio signal to the N-channel audio signal. A conversion processor converts spatial parameters of the spatial parameter data into first binaural parameters in response to at least one binaural perceptual transfer function. A matrix processor converts the M-channel audio signal into a first stereo signal in response to the first binaural parameters. A stereo filter generates the binaural audio signal by filtering the first stereo signal. The filter coefficients for the stereo filter are determined in response to the at least one binaural perceptual transfer function by a coefficient processor. The combination of parameter conversion/processing and filtering allows a high quality binaural signal to be generated with low complexity. | 09-30-2010 |
