Patent application number | Description | Published |
20120128159 | Decorrelator for Upmixing Systems - An improved decorrelator is disclosed that processes an input audio signal in two separate paths. In one path, a banded phase-flip filter is applied to lower frequencies of the input audio signal. In a second path, a frequency-dependent delay is applied to higher frequencies of the input audio signal. Signals from the two paths are combined to obtain an output signal that is psychoacoustically decorrelated with the input audio signal. The decorrelated signal can be mixed with the input audio signal without generating audible artifacts. | 05-24-2012 |
20130179175 | Method and System for Encoding Audio Data with Adaptive Low Frequency Compensation - A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method. | 07-11-2013 |
20140214431 | SAMPLE RATE SCALABLE LOSSLESS AUDIO CODING - A transmitter in an audio coding system generates an encoded audio signal that conveys a losslessly encoded representation of an audio signal at a first sample rate and losslessly encoded representations of related audio information at other sample rates. A companion receiver with limited computational resources can generate a high-quality output audio signal at a desired sample rate by losslessly decoding the encoded representation of the audio signal and possibly other portions of the encoded audio signal as needed to obtain an output signal at one of the other sample rates. | 07-31-2014 |
20140324441 | METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION - A method for determining mantissa bit allocation of audio data values of frequency domain audio data to be encoded. The allocation method includes a step of determining masking values for the audio data values, including by performing adaptive low frequency compensation on the audio data of each frequency band of a set of low frequency bands of the audio data. The adaptive low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set of low frequency bands has prominent tonal content; and performing low frequency compensation on the audio data in each frequency band in the set of low frequency bands having prominent tonal content as indicated by the compensation control data, but not performing low frequency compensation on the audio data in any other frequency band in the set of low frequency bands. | 10-30-2014 |
20140358554 | AUDIO ENCODING METHOD AND SYSTEM FOR GENERATING A UNIFIED BITSTREAM DECODABLE BY DECODERS IMPLEMENTING DIFFERENT DECODING PROTOCOLS - In a class of embodiments, an audio encoding system (typically, a perceptual encoding system that is configured to generate a single (“unified”) bitstream that is compatible with (i.e., decodable by) a first decoder configured to decode audio data encoded in accordance with a first encoding protocol (e.g., the multichannel Dolby Digital Plus, or DD+, protocol) and a second decoder configured to decode audio data encoded in accordance with a second encoding protocol (e.g., the stereo AAC, HE AAC v1, or HE AAC v2 protocol). The unified bitstream can include both encoded data (e.g., bursts of data) decodable by the first decoder (and ignored by the second decoder) and encoded data (e.g., other bursts of data) decodable by the second decoder (and ignored by the first decoder). In effect, the second encoding format is hidden within the unified bitstream when the bitstream is decoded by the first decoder, and the first encoding format is hidden within the unified bitstream when the bitstream is decoded by the second decoder. The format of the unified bitstream generated in accordance with the invention may eliminate the need for transcoding elements throughout an entire media chain and/or ecosystem. Other aspects of the invention are an encoding method performed by any embodiment of the inventive encoder, a decoding method performed by any embodiment of the inventive decoder, and a computer readable medium (e.g., disc) which stores code for implementing any embodiment of the inventive method. | 12-04-2014 |
20150025896 | Enabling Sampling Rate Diversity In A Voice Communication System - An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams. | 01-22-2015 |
Patent application number | Description | Published |
20090138267 | Audio Coding System Using Temporal Shape of a Decoded Signal to Adapt Synthesized Spectral Components - A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal including temporal shape. Spectral components are synthesized having the one or more assessed characteristics, integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. | 05-28-2009 |
20090144055 | Audio Coding System Using Temporal Shape of a Decoded Signal to Adapt Synthesized Spectral Components - A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal including temporal shape. Spectral components are synthesized having the one or more assessed characteristics, integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. | 06-04-2009 |
20090192806 | Broadband Frequency Translation for High Frequency Regeneration - An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal. | 07-30-2009 |
20090222272 | Controlling Spatial Audio Coding Parameters as a Function of Auditory Events - An audio encoder or encoding method receives a plurality of input channels and generates one or more audio output channels and one or more parameters describing desired spatial relationships among a plurality of audio channels that may be derived from the one or more audio output channels, by detecting changes in signal characteristics with respect to lime in one or more of the plurality of audio input channels, identifying as auditory event boundaries changes in signal characteristics with respect to lime in the one or more of the plurality of audio input channels, an audio segment between consecutive boundaries constituting an auditory event in the channel or channels, and generating all or some of the one or more parameters al least partly in response to auditory events and/or the degree of change in signal characteristics associated with the auditory event boundaries. An auditory-event-responsive audio upmixer or upmixing method is also disclosed. | 09-03-2009 |
20090304189 | Rendering Center Channel Audio - An audio upmixer, such as a two-channel to three-channel upmixer, employs a difference in a measure of sound at the ears of a listener in accordance with first and second models, one based on a reproduction of the original channels and the other based on a reproduction of the upmixed channels. The difference is minimized while simultaneously causing a, portion of one or more of the stereophonic channels to be applied to the center loudspeaker under some conditions of the signals in the stereophonic channels, the portion being commensurate with the value of a weighting factor, such that the weighting factor controls a balance between two opposing conditions, one in which no signals are applied to the center loudspeaker and another in which no signals are applied to the left and right loudspeakers. | 12-10-2009 |
20100094637 | ARBITRARY SHAPING OF TEMPORAL NOISE ENVELOPE WITHOUT SIDE-INFORMATION - In a first aspect, arbitrary shaping of the temporal envelope of noise is provided in spectral domain coding systems without the need of side-information. In the encoding, a filtered measure of quantization error is applied as a feedback signal to the frequency-domain representation of a discrete time-domain signal prior to quantization, so that the filtering parameters of said filtering affect the shaping of quantization noise in the time domain of the quantized frequency-domain representation of the discrete time-domain signal when it is inversely transformed from the frequency domain back to the time domain in decoding. This may be accomplished with respect to each of a plurality of frequency bins or groups of bins. In another aspect, frequency-domain noise-feedback quantizing in digital audio encoding is provided. | 04-15-2010 |
20100177903 | Hybrid Derivation of Surround Sound Audio Channels By Controllably Combining Ambience and Matrix-Decoded Signal Components - Ambience signal components are obtained from source audio signals, matrix-decoded signal components are obtained from the source audio signals, and the ambience signal components are controllably combined with the matrix-decoded signal components. Obtaining ambience signal components may include applying at least one decorrelation filter sequence. The same decorrelation filter sequence may be applied to each of the input audio signals or, alternatively, a different decorrelation filter sequence may be applied to each of the input audio signals. | 07-15-2010 |
20120128177 | Circular Frequency Translation with Noise Blending - An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal. | 05-24-2012 |
20120328121 | Reconstructing an Audio Signal By Spectral Component Regeneration and Noise Blending - An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal. | 12-27-2012 |
20140161283 | Reconstructing an Audio Signal By Spectral Component Regeneration and Noise Blending - An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal. | 06-12-2014 |