Abstract: A voice quality estimation process may be triggered by receiving one or more alarms corresponding to one or more endpoint terminals being used during a teleconference. The alarm(s) may include uplink transmission alarms, downlink transmission alarms and/or acoustic quality alarms. The alarms may be based on evaluating transmission metrics and/or acoustic quality metrics. The voice quality estimation process may require a relatively greater computational burden than the processes of evaluating the transmission metrics and/or acoustic quality metrics for the purpose of potentially triggering an alarm. The accuracy and computational complexity of voice quality estimation may be adjusted by selecting times during which alarm detection will take place, alarm detector thresholds, alarm analyzer thresholds and/or levels of voice quality estimation.

Abstract: A method of encoding audio information for forward error correction reconstruction of a transmitted audio stream over a lossy packet switched network, the method including the steps of: (a) dividing the audio stream into audio frames; (b) determining a series of corresponding audio frequency bands for the audio frames; (c) determining a series of power envelopes for the frequency bands; (d) encoding the envelopes as a low bit rate version of the audio frame in a redundant transmission frame.

Abstract: A method of determining a near optimal forward error correction scheme for the transmission of audio data over a lossy packet switched network having preallocated estimated bandwidth, delay and packet losses, between at least a first and second communications devices, the method including the steps of: determining a first coding rate for the audio data; determining a peak redundancy coding rate for redundant versions of the audio data; determining an average redundancy coding rate over a period of time for redundant versions of the audio data; determining an objective function which maximizes a bitrate-perceptual audio quality mapping of the transmitted audio data including a playout function formulation; and optimising the objective function to produce a forward error correction scheme providing a high bitrate perceptual audio quality.

Abstract: Example embodiments disclosed herein relate to noise level estimation. A method for noise level estimation is disclosed. The method includes, responsive to an increase of a signal level of a noise signal, calculating an impulsive noise probability of the noise signal, the impulsive noise probability indicating a likelihood that the noise signal is an impulsive noise. The method also includes determining a variable smoothing factor for noise level estimation based on the impulsive noise probability, the variable smoothing factor being associated with a previous estimated level of the noise signal. The method further includes smoothing the noise signal with the variable smoothing factor so as to determine a current estimated level of the noise signal. Corresponding system and computer program products are also disclosed.

Abstract: Techniques for re-associating dynamic metadata with media data are provided. A media processing system creates, with a first media processing stage, binding information comprising dynamic metadata and a time relationship between the dynamic metadata and media data. The binding information may be derived from the media data. While the first media processing stage delivers the media data to a second media processing stage in a first data path, the first media processing stage passes the binding information to the second media processing stage in a second data path. The media processing system re-associates, with the second media processing stage, the dynamic metadata and the media data using the binding information.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Embodiments are described for a method and system of rendering and playing back spatial audio content using a channel-based format. Spatial audio content that is played back through legacy channel-based equipment is transformed into the appropriate channel-based format resulting in the loss of certain positional information within the audio objects and positional metadata comprising the spatial audio content. To retain this information for use in spatial audio equipment even after the audio content is rendered as channel-based audio, certain metadata generated by the spatial audio processor is incorporated into the channel-based data. The channel-based audio can then be sent to a channel-based audio decoder or a spatial audio decoder. The spatial audio decoder processes the metadata to recover at least some positional information that was lost during the down-mix operation by upmixing the channel-based audio content back to the spatial audio content for optimal playback in a spatial audio environment.

Abstract: Equalizer controller and controlling method are disclosed. In one embodiment, an equalizer controller includes an audio classifier for identifying the audio type of an audio signal in real time; and an adjusting unit for adjusting an equalizer in a continuous manner based on the confidence value of the audio type as identified.

Abstract: Essence of a media program can be encoded such that it carries essence identification information and a set of time points. Auxiliary content such as companion content and other auxiliary data can be associated with the essence of the media program by way of the set of time points. The set of time points and/or fingerprints used to retrieve/recover the set of time points can be delivered to one or more of a media device, a companion device/application, etc., in any of a variety of different methods. Through the set of time points and/or fingerprints, and user preferences, the auxiliary content can be retrieved by the one or more of the media device, the companion device/application, etc., presented and/or interacted with a user. In some embodiments, the auxiliary content as described herein can be customized for a specific sender and/or a specific recipient.

Abstract: The processing efficiency of a process used to decode frames of an enhanced AC-3 bit stream is improved by processing each audio block in a frame only once. Audio blocks of encoded data are decoded in block order rather than in channel order. Exemplary decoding processes for enhanced bit stream coding features such as adaptive hybrid transform processing and spectral extension are disclosed.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: Embodiments are described for a system for virtual rendering of object based audio through binaural rendering of each object followed by panning of the resulting stereo binaural signal between a plurality of cross-talk cancelation circuits feeding a corresponding plurality of speaker pairs. In comparison to prior art virtual rendering utilizing a single pair of speakers, the described embodiments improve the spatial impression for both listeners inside and outside of the cross-talk canceller sweet spot. Also described is an improved equalization technique for a crosstalk canceller that is computed from both the crosstalk canceller filters and the binaural filters and applied to a monophonic audio signal being virtualized. The described techniques improve timbre for listeners outside of the sweet-spot as well as a smaller timbre shift when switching from standard rendering to virtual rendering.

Abstract: A method for performing linear mixing on coupled Head-related transfer functions (HRTFs) to determine an interpolated HRTF for any specified arrival direction in a range (e.g., a range spanning at least 60 degrees in a plane, or a full range of 360 degrees in a plane), where the coupled HRTFs have been predetermined to have properties such that linear mixing can be performed thereon (to generate interpolated HRTFs) without introducing significant comb filtering distortion. In some embodiments, the method includes steps of: in response to a signal indicative of a specified arrival direction, performing linear mixing on data indicative of coupled HRTFs of a coupled HRTF set to determine an HRTF for the specified arrival direction; and performing HRTF filtering on an audio input signal using the HRTF for the specified arrival direction.

Abstract: Higher Order Ambisonics (HOA) represents three-dimensional sound. HOA provides high spatial resolution and facilitates analyzing of the sound field with respect to dominant sound sources. The invention aims to identify independent dominant sound sources constituting the sound field, and to track their temporal trajectories. Known applications are searching for all potential candidates for dominant sound source directions by looking at the directional power distribution of the original HOA representation, whereas in the invention all components which are correlated with the signals of previously found sound sources are removed. By such operation the problem of erroneously detecting many instead of only one correct sound source can be avoided in case its contributions to the sound field are highly directionally dispersed.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: Embodiments are directed to an interconnect for coupling components in an object-based rendering system comprising: a first network channel coupling a renderer to an array of individually addressable drivers projecting sound in a listening environment and transmitting audio signals and control data from the renderer to the array, and a second network channel coupling a microphone placed in the listening environment to a calibration component of the renderer and transmitting calibration control signals for acoustic information generated by the microphone to the calibration component. The interconnect is suitable for use in a system for rendering spatial audio content comprising channel-based and object-based audio components.

Abstract: Embodiments of the present invention relate to signal processing. Methods for enhancing intelligibility of speech content in an audio signal are disclosed. One of the methods comprises obtaining reference loudness of the audio signal. The method further comprises enhancing the intelligibility of the speech content by adjusting partial loudness of the audio signal based on the reference loudness and a degree of the intelligibility. Corresponding systems and computer program products are also disclosed.

Abstract: Example embodiments disclosed herein relate to modelling a frequency response characteristic of an electro-acoustic transducer. A method includes obtaining at least one measurement of the frequency response characteristic for at least one electro-acoustic transducer of the category. A model of a frequency response characteristic specific to a category of electro-acoustic transducers is generated at least in part based on perceptual importance of a frequency band, an averaged, normalized or microphone compensated measurement such that the distortion of the model is optimized. A further method for estimating a frequency response characteristic of an electro-acoustic transducer is based on the generated model and the sensitivity of the electro-acoustic transducer or headphone. Corresponding system and computer program product are also disclosed.

Abstract: A display management processor receives an input image with enhanced dynamic range to be displayed on a target display which has a different dynamic range than a reference display. The input image is first transformed into a perceptually-quantized (PQ) color space. A non-linear mapping function generates a tone-mapped intensity image in response to the characteristics of the source and target display and a measure of the intensity of the PQ image. After a detail-preservation step which may generate a filtered tone-mapped intensity image, an image-adaptive intensity and saturation adjustment step generates an intensity adjustment factor and a saturation adjustment factor as functions of the measure of intensity and saturation of the PQ image, which together with the filtered tone-mapped intensity image are used to generate the output image. Examples of the functions to compute the intensity and saturation adjustment factors are provided.

Abstract: A communication system for communicating with at least one loudspeaker is described where the loudspeaker is connected to audio equipment over standard two-wire speaker wire operable to carry an audio signal. The communication system includes a master node in electrical communication with a signal path carrying the audio signal between the audio equipment and the loudspeaker, the master node also including an interface with the audio equipment, a data encoder operable to encode data signals, a data transceiver operable to place the data signals onto the audio signal at frequencies above audio frequencies. The communication system also includes at least one slave node in electrical communication with the audio signal and each loudspeaker, the slave node including a data transceiver operable to receive data signals from the master node, a data decoder, and an interface able to communicate with the loudspeaker.

Abstract: A method for creating a head-related impulse response (HRIR) for use in rendering audio for playback through headphones comprises receiving location parameters for a sound including azimuth, elevation, and range relative to a head of a listener, applying a spherical head model to the azimuth, elevation, and range input parameters to generate binaural HRIR values, computing a pinna model using the azimuth and elevation parameters to apply to the binaural HRIR values to pinna modeled HRIR values, computing a torso model using the azimuth and elevation parameters to apply to the pinna modeled HRIR values to generate pinna and torso modeled HRIR values, and computing a near-field model using the azimuth and range parameters to apply to the pinna and torso modeled HRIR values to generate pinna, torso and near-field modeled HRIR values.

Abstract: Novel methods and systems for non-backward compatible video encoding are disclosed. The bitrates of the base layer and enhancement layer are dynamically assigned based on features found in scenes in the video compared to a machine learned quality classifier.

Abstract: A system and method of mixing narration into content. The system automatically reduces the volume of the content according to a threshold value and a knee value. In this manner, the audio of the content does not overwhelm the narration.

Abstract: In one embodiment, a dual modulator display systems and methods for rendering target image data upon the dual modulator display system are disclosed where the display system receives target image data, possible HDR image data and first calculates display control signals and then calculates backlight control signals from the display control signals. This order of calculating display signals and then backlight control signals later as a function of the display systems may tend to reduce clipping artifacts. In other embodiments, it is possible to split the input target HDR image data into a base layer and a detail layer, wherein the base layer is the low spatial resolution image data that may be utilized as for backlight illumination data. The detail layer is higher spatial resolution image data that may be utilized for display control data.

Abstract: Methods and systems for generating and applying scene-stable metadata for a video data stream are disclosed herein. A video data stream is divided or partitioned into scenes and a first set of metadata may be generated for a given scene of video data. The first set of metadata may be any known metadata as a desired function of video content (e.g., luminance). The first set of metadata may be generated on a frame-by-frame basis. In one example, scene-stable metadata may be generated that may be different from the first set of metadata for the scene. The scene-stable metadata may be generated by monitoring a desired feature with the scene and may be used to keep the desired feature within an acceptable range of values. This may help to avoid noticeable and possibly objectionably visual artifacts upon rendering the video data.

Abstract: Power levels of a backlight are adjusted in a number of ways and based on a number of criteria. The adjustments result in a lower power consumption and, in some cases, may enhance audience attention to important objects in a scene. The adjustments comprise, for example, a combination of ramping down power (lowering final display brightness) in concert with corresponding compensatory LCD adjustments (increasing final display brightness). The adjustments may also include, for example, system dimming after ramp down/LCD adjustments are exhausted, or the shifting of an LDR2HDR curve.

Abstract: A method is provided for coding at least one image split up into partitions, a current partition to be coded containing data, at least one data item of which is allotted a sign. The coding method includes, for the current partition, the following steps: calculating the value of a function representative of the data of the current partition with the exclusion of the sign; comparing the calculated value with a predetermined value of the sign; as a function of the result of the comparison, modifying or not modifying at least one of the data items of the current partition, in the case of modification, coding the at least one modified data item.

Abstract: An image decoding apparatus that includes a motion compensation prediction circuit configured to conduct motion compensation prediction for each of blocks to be decoded by using the reconstructed image, an inverse transformation circuit configured to conduct inverse orthogonal transformation for the data of the blocks to be decoded, and a determination circuit configured to determine a filtering strength and whether or not to conduct filtering, with respect to each of the boundaries. In addition, the determining circuit is configured to determine filtering is conducted when at least one of the two adjacent blocks is intra-coded, and filtering is not conducted when both of the two adjacent blocks are not intra-coded, a non-zero transformation coefficient is not coded in both of the two adjacent blocks, the two adjacent blocks are predicted by the same reference frame, and an absolute value of a difference between motion vectors of the two adjacent blocks is smaller than a specified threshold value.

Abstract: 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.

Abstract: Novel methods and systems for inverse tone mapping are disclosed. A scene can be analyzed to obtain highlight detection from bright light sources and specular reflections. An inverse tone mapping curve can be calculated based on the lower dynamic range and higher dynamic range displays. Multi-scale filtering can be applied to reduce noise or artifacts.

Abstract: An apparatus for generating real-valued output audio samples is disclosed. The apparatus includes a memory that stores complex-valued input subband samples, real-valued demodulated samples, and the real-valued output audio samples. The apparatus also incudes a phase shifter that shifts a phase of the complex-valued input subband samples by an amount equal to a previously added phase shift and a complex-valued synthesis filter bank that generates the real-valued output audio samples in response to the complex-valued input subband samples, the real-valued demodulated samples, and prototype filter coefficients.

Abstract: Video data with both ultra-high definition (UHD) resolution and high or enhanced dynamic range (EDR) data are coded in a backward-compatible layered stream which allows legacy decoders to extract an HD standard dynamic range (SDR) signal. In response to a base layer HD SDR signal, a predicted signal is generated using separate luma and chroma prediction models. In the luma predictor, luma pixel values of the predicted signal are computed based only on luma pixel values of the base layer, while in the chroma predictor, chroma pixel values of the predicted signal are computed based on both the luma and the chroma pixel values of the base layer. A residual signal is computed based on the input UHD EDR signal and the predicted signal. The base layer and the residual signal are coded separately to form a coded bitstream. A compatible dual-layer decoder is also presented.

Abstract: Methods to reduce chroma-related artifacts during video coding of high dynamic range images are presented. Given an input signal in a color space comprising a luma component and two chromaticity components, a processor determines the original white point chromaticity coordinates (Du, Dv) of a white point in the color space of the input signal. The input signal is translated using a chromaticity translation function to a second signal in a translated color space comprising two translated chromaticity components, wherein the chromaticity translation function shifts the original white point chromaticity coordinates to a predetermined second set of coordinates in the translated chromaticity color space. The second signal is encoded to generate a coded bit stream.

Abstract: A method and system for reconstructing an original audio signal is disclosed. The original audio signal has a baseband up to a cutoff frequency and high-frequency components not included in the baseband above the cutoff frequency. The system includes a bitstream deformatter that extracts a representation of the baseband, an estimated spectral envelope, and noise-blending parameters from an audio bitstream. The system also includes a spectral component regenerator that copies or translates all or at least some of the baseband spectral components to non-overlapping frequency ranges of the high-frequency components not included in the baseband to generate regenerated spectral components. The system further includes a gain adjuster that modifies a spectral envelope of the regenerated spectral components based at least in part on the estimated spectral envelope and the noise-blending parameters to generate gain-adjusted regenerated spectral components.

Abstract: Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

Abstract: A method of rendering an audio program by generating one or more loudspeaker channel feeds based on the dynamic trajectory of each audio object in the audio program, wherein the parameters of the dynamic trajectory may be included explicitly in the audio program, or may be derived from the instantaneous location of audio objects at two or more points in time. Embodiments include rendering audio by defining a nominal loudspeaker map of loudspeakers used for playback of the audio program, determining a trajectory of an auditory source corresponding to one or more audio objects through 3D space, generating loudspeaker signals feeding the loudspeakers based on the one or more audio object trajectories, and rendering the one or more audio objects based on object location to match the trajectory of the auditory source as perceived by a listener.

Abstract: The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications.

Abstract: In some embodiments, a method for monitoring speakers within an audio playback system (e.g., movie theater) environment. In typical embodiments, the monitoring method assumes that initial characteristics of the speakers (e.g., a room response for each of the speakers) have been determined at an initial time, and relies on one or more microphones positioned in the environment to perform a status check on each of the speakers to identify whether a change to at least one characteristic of any of the speakers has occurred since the initial time. In other embodiments, the method processes data indicative of output of a microphone to monitor audience reaction to an audiovisual program. Other aspects include a system configured (e.g., programmed) to perform any embodiment of the inventive method, and a computer readable medium (e.g., a disc) which stores code for implementing any embodiment of the inventive method.

Abstract: An audio processing method and apparatus are described. In one embodiment, at least one first sub-band of a first audio signal is suppressed to obtain a reduced first audio signal with reserved sub-bands; suppressing at least one second sub-band of the at least one second audio signal to obtain at least one reduced second audio signal with reserved sub-bands; and mixing the reduced first audio signal and at least one reduced second audio signal. Alternatively, a first spatial auditory property is assigned to a first audio signal so that the first audio signal may be perceived as originating from a first position. Alternatively, rhythmic similarity between at least two audio signals is detected, and time scaling is applied to an audio signal in response to relatively high rhythmic similarity between the audio signal and the other audio signal(s); and then at least two audio signals are mixed.

Abstract: A decoding system (100) reconstructs an n-channel audio signal on the basis of an input signal (A) representing the audio signal either by parametric coding or as n discretely coded channels. Parametric decoding proceeds on the basis of a core signal and mixing parameters (a) controlling a spatial synthesis stage (150), which is supplied with a downmix signal from a downmix stage (140). A selector (170) controls the components of the decoding system, in steady-state parametric and discrete decoding mode and transitions between these. The downmix stage realizes a projection on the downmix signal based on an n-channel signal, either an n-channel input signal or a core signal padded with neutral values. The downmix stage is active in each time frame in which the input signal represents the audio signal by parametric coding and in at least the first time frame after the last time frame in each episode of parametrically coded time frames.

Abstract: RFID data is utilized for billing and management of 3D glasses in a cinema theater or group of theaters. RFID pedestals identify privately owned or rented 3D glasses brought into a theater by a customer. An invoice is automatically generated billing the theater operator for the use of the 3D glasses. Usage of the glasses along with other data collected about the customers is utilized for targeted advertising (e.g., e-mails, texts, selected advertise in pre-show, etc.). The invention reduces up front cost in installing a reusable 3D glasses system.

Abstract: Techniques are provided for vector noise cancellation. Different value combinations for a plurality of weighting factors may be established for a plurality of selection regions. Each value combination for the plurality of weighting factors may correspond to a different combination of a set of input signals. One or more characteristics of input signals may be used to select a particular selection region. A particular value combination of the set of weighting factors may be chosen to attenuate or amplify the input signals to generate one or more output signals.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: Dual or multi-modulation display systems comprising a first modulator and a second modulator are disclosed. The first modulator may comprise a plurality of analog mirrors (e.g. MEMS array) and the second modulator may comprise a plurality of mirrors (e.g., DMD array). The display system may further comprise a controller that sends control signals to the first and second modulator. The display system may render highlight features within a projected image by affecting a time multiplexing scheme. In one embodiment, the first modulator may be switched on a sub-frame basis such that a desired proportion of the available light may be focused or directed onto the second modulator to form the highlight feature on a sub-frame rendering basis.

Abstract: A display management unit configured to provide a modified video signal for display on a target display over an electronic distribution network. The unit may access information regarding the target display and at least one input. The unit comprises a database interface configured to retrieve display characteristics corresponding to the information regarding the target display from a characteristics database, and a mapping unit configured to map at least one of tone and color values from the at least one input to corresponding mapped values based at least in part on the retrieved display characteristics to produce the modified video signal.

Abstract: Embodiments are described for a method of rendering an audio program by receiving, in a renderer of a playback system, the audio program and a target response representing desired characteristics of the playback environment, deriving a playback environment response based on characteristics of the playback environment, comparing the target response to the playback environment response to generate a set of correction settings, and applying the correction settings to the audio program so that the audio program is rendered according to the characteristics of the target response. The target response may be based on audio characteristics in a creation environment.

Abstract: A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module.

Abstract: In an audio conferencing mixing system of the type taking a plurality of audio input streams of input audio information of conference participants, including mixing transition events and outputting a plurality of audio output streams including output audio information, a method of mixing the audio output streams so as to reduce the detectability of the mixing transition events, the method including the steps of (a) determining that a transition event is to occur; (b) determining that a masking trigger is to occur; (c) scheduling the transition event to substantially occur when the masking event occurs. Change blindness mechanism to mask changes in audio conference mix and maintain perceptual continuity.

Abstract: In one aspect, the invention provides an audio encoding method characterized by a decision being made as to whether the device which will decode the resulting bit stream should apply post filtering including attenuation of interharmonic noise. Hence, the decision whether to use the post filter, which is encoded in the bit stream, is taken separately from the decision as to the most suitable coding mode. In another aspect, there is provided an audio decoding method with a decoding step followed by a post-filtering step, including interharmonic noise attenuation, and being characterized in a step of disabling the post filter in accordance with post filtering information encoded in the bit stream signal. Such a method is well suited for mixed-origin audio signals by virtue of its capability to deactivate the post filter in dependence of the post filtering information only, hence independently of factors such as the current coding mode.

Abstract: An apparatus and method are disclosed for filtering and performing high frequency reconstruction of an audio signal. The apparatus includes an analysis filter bank, a high frequency reconstructor, and a synthesis filter bank. The analysis filterbank receives real-valued time domain input audio samples and generates complex valued subband samples. The high frequency reconstructor modifies at least some of the complex valued subband samples. The synthesis filter bank receives the modified complex valued subband samples and generates time domain output audio samples. The analysis filter bank comprises analysis filters that are complex exponential modulated versions of a prototype filter with an arbitrary phase shift.