Abstract: Audio processing methods may involve receiving audio data corresponding to a plurality of audio channels. The audio data may include a frequency domain representation corresponding to filterbank coefficients of an audio encoding or processing system. A decorrelation process may be performed with the same filterbank coefficients used by the audio encoding or processing system. The decorrelation process may be performed without converting coefficients of the frequency domain representation to another frequency domain or time domain representation. The decorrelation process may involve selective or signal-adaptive decorrelation of specific channels and/or specific frequency bands. The decorrelation process may involve applying a decorrelation filter to a portion of the received audio data to produce filtered audio data. The decorrelation process may involve using a non-hierarchal mixer to combine a direct portion of the received audio data with the filtered audio data according to spatial parameters.

Abstract: A system, a method, logic embodied in a computer-readable medium, and a computer-readable medium comprising instructions that when executed carry out a method. The method processes: (a) a plurality of input signals, e.g., signals from a plurality of spatially separated microphones; and, for echo suppression, (b) one or more reference signals, e.g., signals from or to be rendered by one or more loudspeakers and that can cause echoes. The method processes the input signals and one or more reference signals to carry out in an integrated manner simultaneous noise suppression and out-of-location signal suppression, and in some versions, echo suppression.

Abstract: A method for generating a reconstructed audio signal having a baseband portion and a highband portion is disclosed. The method includes deformatting an encoded audio signal into a first part and a second part and decoding the first part to obtain a decoded baseband audio signal. The method also includes extracting an estimated spectral envelope of the highband portion and a noise parameter from the second part and filtering the decoded baseband audio signal to obtain a plurality of subband signals. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and adjusting a spectral envelope of the high-frequency reconstructed signal based on the estimated spectral envelope of the highband portion to obtain an envelope adjusted high-frequency signal.

Abstract: A method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes decoding an encoded audio signal to obtain a decoded baseband audio signal, filtering the decoded baseband audio signal to obtain subband signals, and generating a high-frequency reconstructed signal by copying a number of consecutive subband signals. The method also includes adjusting a spectral envelope of the high-frequency reconstructed signal based on an estimated spectral envelope of the highband portion extracted from the encoded audio signal to obtain an envelope adjusted high-frequency signal, generating a noise component based on a noise parameter extracted from the encoded audio signal, and adding the noise component to the envelope adjusted high-frequency signal to obtain a noise and envelope adjusted high-frequency signal.

Abstract: A method including steps of decoding an encoded audio signal indicative of encoded audio content (e.g., audio content captured during a teleconference) to generate a decoded signal indicative of a decoded version of the audio content, and performing adaptive quantization noise filtering on the decoded signal. The filtering is performed adaptively in the frequency domain in response to data indicative of signal to noise values in turn indicative of a post-quantization signal-to-quantization noise ratio for each frequency band of each of at least one segment of the encoded audio content. In some embodiments, each signal to noise value is a bit allocation value equal to the number of mantissa bits of an encoded audio sample of a frequency band of a segment of the encoded audio content. Other aspects are decoder, or post-filter coupled to receive a decoder's output, configured to perform an embodiment of the adaptive filtering.

Abstract: 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.

Abstract: 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).

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A system, a method, logic embodied in a computer-readable medium, and a computer-readable medium comprising instructions that when executed carry out a method. The method processes: (a) a plurality of input signals, e.g., signals from a plurality of spatially separated microphones; and, for echo suppression, (b) one or more reference signals, e.g., signals from or to be rendered by one or more loudspeakers and that can cause echoes. The method processes the input signals and one or more reference signals to carry out in an integrated manner simultaneous noise suppression and out-of-location signal suppression, and in some versions, echo suppression.

Abstract: 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.

Abstract: 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.

Abstract: A method and system for encoding and decoding an input signal in relation to the most perceptually relevant aspects of the input signal. A two-dimensional (2D) transform is applied to the input signal to produce a magnitude matrix and a phase matrix that can be inverse quantized by a decoder. A first column of coefficients of the magnitude matrix represents a mean spectral density (MSD) function of the input signal. Relevant aspects of the MSD function are encoded at a beginning of a data packet. The MSD function is also processed through a core perception model to determine bit allocation. The matrices are then quantized and priority ordered into a data packet, with the least perceptually relevant information at the end of the packet so that it may be ignored or truncated for scalability to the channel data rate capacity.

Abstract: A method and system for encoding and decoding an input signal in relation to the most perceptually relevant aspects of the input signal. A two-dimensional (2D) transform is applied to the input signal to produce a magnitude matrix and a phase matrix that can be inverse quantized by a decoder. A first column of coefficients of the magnitude matrix represents a mean spectral density (MSD) function of the input signal. Relevant aspects of the MSD function are encoded at a beginning of a data packet. The MSD function is also processed through a core perception model to determine bit allocation. The matrices are then quantized and priority ordered into a data packet, with the least perceptually relevant information at the end of the packet so that it may be ignored or truncated for scalability to the channel data rate capacity.