Abstract: An apparatus and method of loudspeaker equalization. The method combines default tunings (generated based on a default listening environment) and room tunings (generated based on an end user listening environment) to result in combined tunings that account for differences between the end user listening environment and the default listening environment.

Abstract: Embodiments are disclosed for hybrid near/far-field speaker virtualization.

Abstract: Shaped glasses have curved surface lenses with spectrally complementary filters disposed thereon. The filters curved surface lenses are configured to compensate for wavelength shifts occurring due to viewing angles and other sources. Complementary images are projected for viewing through projection filters having passbands that pre-shift to compensate for subsequent wavelength shifts. At least one filter may have more than 3 primary passbands. For example, two filters include a first filter having passbands of low blue, high blue, low green, high green, and red, and a second filter having passbands of blue, green, and red. The additional passbands may be utilized to more closely match a color space and white point of a projector in which the filters are used. The shaped glasses and projection filters together may be utilized as a system for projecting and viewing 3D images.

Abstract: A rendering mode may be determined for received audio data, including audio signals and associated spatial data. The audio data may be rendered for reproduction via a set of loudspeakers of an environment according to the rendering mode, to produce rendered audio signals. Rendering the audio data may involve determining relative activation of a set of loudspeakers in an environment. The rendering mode may be variable between a reference spatial mode and one or more distributed spatial modes. The reference spatial mode may have an assumed listening position and orientation. In the distributed spatial mode(s), one or more elements of the audio data may each be rendered in a more spatially distributed manner than in the reference spatial mode and spatial locations of remaining elements of the audio data may be warped such that they span a rendering space of the environment more completely than in the reference spatial mode.

Abstract: Novel methods and systems for adapting a voice cloning synthesizer for a new speaker using real speech data are disclosed. Utterances from one or more target speakers are parameterized and are used to initialize an embedding vector for use with a voice synthesizer, by means of clustering the utterance data and determining the centroid of the data, using a speaker identification neural network, and/or by finding the closest stored embedded vector to the utterance data.

Abstract: A method is provided, including: defining a plurality of frequency bins; sending, during a training phase, a test signal at different amplitude levels to one or more speakers, and gathering resulting test voltage (V) and current (I) points for the different amplitude levels and for each frequency bin; for each frequency bin, applying a linear regression algorithm to the gathered test voltage and current points for the different amplitudes to obtain a reference electrical impedance of said one or more speakers; sending, during a monitoring phase subsequent to said training phase, an audio signal to said one or more speakers, and gathering resulting new voltage and current points to obtain an operating electrical impedance for said one or more speakers for each frequency bin, determining a deviation between the operating and the reference electrical impedance, and, if the deviation exceeds a defined tolerance, reporting the deviation to a user.

Abstract: A method for estimating a user's location in an environment may involve receiving output signals from each microphone of a plurality of microphones in the environment. At least two microphones of the plurality of microphones may be included in separate devices at separate locations in the environment and the output signals may correspond to a current utterance of a user. The method may involve determining multiple current acoustic features from the output signals of each microphone and applying a classifier to the multiple current acoustic features. Applying the classifier may involve applying a model trained on previously-determined acoustic features derived from a plurality of previous utterances made by the user in a plurality of user zones in the environment. The method may involve determining, based at least in part on output from the classifier, an estimate of the user zone in which the user is currently located.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: According to the present invention, an adaptive scheme is applied to an image encoding apparatus that includes an inter-predictor, an intra-predictor, a transformer, a quantizer, an inverse quantizer, and an inverse transformer, wherein input images are classified into two or more different categories, and two or more modules from among the inter-predictor, the intra-predictor, the transformer, the quantizer, and the inverse quantizer are implemented to perform respective operations in different schemes according to the category to which an input image belongs. Thus, the invention has the advantage of efficiently encoding an image without the loss of important information as compared to a conventional image encoding apparatus which adopts a packaged scheme.

Abstract: Two corresponding color patches are displayed on two image displays until adjusted by a viewer to match visually to a common color. Two sets of code values rendered on the two corresponding color patches on the two image displays are identified. Two sets of tristimulus values for the viewer are determined based on the two sets of code values rendered on the two corresponding color patches on the two image displays. The viewer's color matching function are generated based on the two sets of tristimulus values. The viewer's CMF is used in image rendering operations on a target image display.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: A device and method for video rendering. The device includes a memory and an electronic processor. The electronic processor is configured to receive, from a source device, video data including multiple reference viewpoints, determine a target image plane corresponding to a target viewpoint, determine, within the target image plane, one or more target image regions, and determine, for each target image region, a proxy image region larger than the corresponding target image region. The electronic processor is configured to determine, for each target image region, a plurality of reference pixels that fit within the corresponding proxy image region, project, for each target image region, the plurality of reference pixels that fit within the corresponding proxy image region to the target image region, producing a rendered target region from each target image region, and composite one or more of the rendered target regions to create video rendering.

Abstract: Some disclosed methods involve encoding or decoding directional audio data. Some encoding methods may involve receiving a mono audio signal corresponding to an audio object and a representation of a radiation pattern corresponding to the audio object. The radiation pattern may include sound levels corresponding to plurality of sample times, a plurality of frequency bands and a plurality of directions. The methods may involve encoding the mono audio signal and encoding the source radiation pattern to determine radiation pattern metadata. Encoding the radiation pattern may involve determining a spherical harmonic transform of the representation of the radiation pattern and compressing the spherical harmonic transform to obtain encoded radiation pattern metadata.

Abstract: Methods for performing dynamic range compression (DRC) on audio in a manner intended to produce output audio for playback by systems or devices with limited power handling capabilities and preferably also to reduce or prevent undesirable artifacts (e.g., pumping and/or breathing) in the output audio. Some embodiments perform the DRC so as to maximize average loudness (while preventing loss of quieter elements) during playback, and also to reduce or prevent distortion. Other aspects are systems or devices configured to perform embodiments of the method. In some embodiments, reduced DRC is applied when average loudness of the input audio approaches (or matches or exceeds) a target (e.g., a knee point for DRC, or a signal level near to a maximum playback level of the intended playback system), since such input audio is assumed to have already been compressed, and otherwise applying full DRC to the input audio.

Abstract: A method for channel identification of a multi-channel audio signal comprising X>1 channels is provided. The method comprises the steps of: identifying, among the X channels, any empty channels, thus resulting in a subset of Y?X non-empty channels; determining whether a low frequency effect (LFE) channel is present among the Y channels, and upon determining that an LFE channel is present, identifying the determined channel among the Y channels as the LFE channel; dividing the remaining channels among the Y channels not being identified as the LFE channel into any number of pairs of channels by matching symmetrical channels; and identifying any remaining unpaired channel among the Y channels not being identified as the LFE channel or divided into pairs as a center channel.

Abstract: Methods for rendering audio for playback by two or more speakers are disclosed. The audio includes one or more audio signals, each with an associated intended perceived spatial position. Relative activation of the speakers may be a cost function of a model of perceived spatial position of the audio signals when played back over the speakers, a measure of proximity of the intended perceived spatial position of the audio signals to positions of the speakers, and one or more additional dynamically configurable functions. The dynamically configurable functions may be based on at least one or more properties of the audio signals, one or more properties of the set of speakers and/or one or more external inputs.

Abstract: An apparatus and method of pre-conditioning audio for machine perception. Machine perception differs from human perception, and different processing parameters are used for machine perception applications (e.g., speech to text processing) as compared to those used for human perception applications (e.g., voice communications). These different parameters may result in pre-conditioned audio that is worsened for human perception yet improved for machine perception.

Abstract: A method and a device for encoding and decoding infra prediction are disclosed. An image decoding method for performing intra prediction comprises the steps of: receiving a bitstream including data on prediction modes of a current block and a block adjacent to the current block; extracting the data from the received bitstream so as to confirm the prediction mode of the adjacent block; determining whether a boundary pixel within the adjacent block can be used as a reference pixel for the current block in consideration of the prediction mode of the adjacent block; obtaining the reference pixel of the current block according to the determined result; generating a prediction block predicted in the frame on the basis of the obtained reference pixel; and decoding the current block by using the generated prediction block.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

Abstract: Backward reshaping metadata prediction models are trained with training SDR images and corresponding training HDR images. Content creation user input to define user adjusted HDR appearances for the corresponding training HDR images is received. Content-creation-user-specific modified backward reshaping metadata prediction models are generated based on the trained prediction models and the content creation user input. The content-creation-user-specific modified prediction models are used to predict operational parameter values of content-creation-user-specific backward reshaping mappings for backward reshaping SDR images into mapped HDR images of at least one content-creation-user-adjusted HDR appearance.

Abstract: In some implementations, a method of encoding a low-frequency effect (LFE) channel comprises: receiving a time-domain LFL channel signal; filtering, using a low-pass filter, the time-domain LFE channel signal; converting the filtered time-domain LFE channel signal into a frequency-domain representation of the LFE channel signal that includes a number of coefficients representing a frequency spectrum of the LFL channel signal; arranging coefficients into a number of subband groups corresponding to different frequency bands of the LFE channel signal; quantizing coefficients in each subband group according to a frequency response curve of the low-pass filter; encoding the quantized coefficients in each subband group using an entropy coder tuned for the subband group; and generating a bitstream including the encoded quantized coefficients; and storing the bitstream on a storage device or streaming the bitstream to a downstream device.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: A device includes an electronic processor configured to define a first set of sample pixels from a set of sample pixels determined from received video data according to a first electro-optical transfer function (EOTF) in a first color representation of a first color space; convert the first set of sample pixels to a second EOTF via a mapping function, producing a second set of sample pixels according to the second EOTF; convert the first and second set of sample pixels from the first color representation to a second color representation of the first color space; determine a backward reshaping function by repeatedly applying and adjusting a sample backward reshaping function so as to minimize a difference between predicted pixel values obtained by applying the sample backward reshaping function to the pixels of the converted first set and the pixels of the converted second set.

Abstract: A projector controller includes an object detector and control electronics, and is configured to protect audience members from intense light imposing an exclusion zone in front of a projector. The object detector is configured to optically sense a presence of an object in a detection region beneath the exclusion zone and above the audience members. The control electronics is configured to control the projector when the object detector indicates the presence of the object in the detection region. A method for protecting audience members from intense light imposing an exclusion zone in front of an output of a projector includes: (i) optically sensing a presence of an object in a detection region between the exclusion zone and the audience members, and (ii) controlling the projector when the presence of the object is sensed in the detection region.

Abstract: Three dimensional (3D) glasses suited for wearers with varying facial geometries are disclosed. A particular embodiment includes a frame adapted to position spectrally filtering lenses at a particular distance from the eyes of the wearer. In a more particular embodiment, the 3D glasses include a means for adjusting the distance between the lenses and the eyes of the wearer. In another particular embodiment, the lenses include positive runout.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: A method for generating mastered audio content, the method comprising obtaining an input audio content comprising a number, M1, of audio signals, obtaining rendered presentation of the input audio content, the rendered presentation comprising a number, M2, of audio signals, obtaining a mastered presentation generated by mastering the rendered presentation, comparing the mastered presentation with the rendered presentation to determine one or more indications of differences between the mastered presentation and the rendered presentation, modifying one or more of the audio signals of the input audio content based on the indications of differences to generate the mastered audio content. With this approach, conventional, typically stereo, channel-based mastering tools can be used to provide a mastered version of any input audio content, including object-based immersive audio content.

Abstract: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: An apparatus and method of rendering audio objects with multiple types of renderers. The weighting between the selected renderers depends upon the position information in each audio object. As each type of renderer has a different output coverage, the combination of their weighted outputs results in the audio being perceived at the position according to the position information.

Abstract: Encoding/decoding an immersive voice and audio services (IVAS) bitstream comprises: encoding/decoding a coding mode indicator in a common header (CH) section of an IVAS bitstream, encoding/decoding a mode header or tool header in the tool header (TH) section of the bitstream, the TH section following the CH section, encoding/decoding a metadata payload in a metadata payload (MDP) section of the bitstream, the MDP section following the CH section, encoding/decoding an enhanced voice services (EVS) payload in an EVS payload (EP) section of the bitstream, the EP section following the CH section, and on the encoder side, storing or streaming the encoded bitstream, and on the decoder side, controlling an audio decoder based on the coding mode, the tool header, the EVS payload, and the metadata payload or storing a representation of same.

Abstract: The present document relates to a method of layered encoding of a frame of a compressed higher-order Ambisonics, HOA, representation of a sound or sound field. The compressed HOA representation comprises a plurality of transport signals. The method comprises assigning the plurality of transport signals to a plurality of hierarchical layers, the plurality of layers including a base layer and one or more hierarchical enhancement layers, generating, for each layer, a respective HOA extension payload including side information for parametrically enhancing a reconstructed HOA representation obtainable from the transport signals assigned to the respective layer and any layers lower than the respective layer, assigning the generated HOA extension payloads to their respective layers, and signaling the generated HOA extension payloads in an output bitstream.

Abstract: Methods, systems, and bitstream syntax are described for canvas size, single layer or multi-layer, sealable decoding, with support for regions of interest (ROI). using a decoder supporting reference picture resampling. Offset parameters for a region of interest in a current picture and offset parameters for an ROI in a reference picture are taken into consideration when computing scaling factors to apply reference picture resampling Syntax elements for supporting ROI regions under reference picture resampling arc also presented.

Abstract: Methods for generating an AV bitstream (e.g., an MPEG-2 transport stream or bitstream segment having adaptive streaming format) such that the AV bitstream includes at least one video I-frame synchronized with at least one audio I-frame, e.g., including by re-authoring at least one video or audio frame (as a re-authored I-frame or a re-authored P-frame). Typically, a segment of content of the AV bitstream which includes the re-authored frame starts with an I-frame and includes at least one subsequent P-frame. Other aspects are methods for adapting such an AV bitstream, audio/video processing units configured to perform any embodiment of the inventive method, and audio/video processing units which include a buffer memory which stores at least one segment of an AV bitstream generated in accordance with any embodiment of the inventive method.

Abstract: Described herein is a method for generating a modified bitstream on a source device, wherein the method includes the steps of: a) receiving, by a receiver, a bitstream including coded media data; b) generating, by an embedder, payload of additional media data and embedding the payload in the bitstream for obtaining, as an output from the embedder, a modified bitstream including the coded media data and the payload of the additional media data; and c) outputting the modified bitstream to a sink device. Described is further a method for processing said modified bitstream on a sink device. Described are moreover a respective source device and sink device as well as a system of a source device and a sink device and respective computer program products.

Abstract: A quantization parameter signalling mechanism for both SDR and HDR content in video coding is described using two approaches. The first approach is to send the user-defined Qpc table directly in high level syntax. This leads to more flexible and efficient QP control for future codec development and video content coding. The second approach is to signal luma and chroma QPs independently. This approach eliminates the need for Qpc tables and removes the dependency of chroma quantization parameter on luma QP.

Abstract: A method of enhancing dialog intelligibility in an audio signal, comprising determining a speech confidence score that the audio content includes speech content, determining a music confidence score that the audio content includes music correlated content, in response to the speech confidence score, and applying a user selected gain of selected frequency bands of the audio signal to obtain a dialogue enhanced audio signal. The user selected gain is smoothed by an adaptive smoothing algorithm, an impact of past frames in said smoothing algorithm being determined by a smoothing factor, the smoothing factor being calculated in response to the music confidence score, and having a relatively higher value for content having a relatively higher music confidence score and a relatively lower value for speech content having a relatively lower music confidence score, so as to increase the impact of past frames on the dialogue enhancement of music correlated content.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

Abstract: Described herein is a method of encoding an audio signal. The method comprises: generating a plurality of subband audio signals based on the audio signal; determining a spectral envelope of the audio signal; for each subband audio signal, determining autocorrelation information for the subband audio signal based on an autocorrelation function of the subband audio signal; and generating an encoded representation of the audio signal, the encoded representation comprising a representation of the spectral envelope of the audio signal and a representation of the autocorrelation information for the plurality of subband audio signals. Further described are methods of decoding the audio signal from the encoded representation, as well as corresponding encoders, decoders, computer programs, and computer-readable recording media.

Abstract: A method for steering binauralization of audio is provided. The method comprises steps of: receiving (410) an audio input signal, calculating (430) a confidence value indicating a likelihood that a current audio frame of the audio input signal comprises binauralized audio; determining (450) a state signal based on the confidence value; determining (460) a steering signal, based on the first confidence value, the state signal and an energy value of the audio frame; and generating (470) an audio output signal with steered binauralization by processing the audio input signal according to the steering signal.

Abstract: Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.

Abstract: A method may involve receiving output signals from each microphone of a plurality of microphones in the environment, each of the plurality of microphones residing in a microphone location of the environment, the output signals corresponding to an utterance of a person. The method may involve determining, based at least in part on the output signals, a zone within the environment that has at least a threshold probability of including the person's location and generating a plurality of spatially-varying attentiveness signals within the zone. Each attentiveness signal may be generated by a device located within the zone. Each attentiveness signal may indicate that a corresponding device is in an operating mode in which the corresponding device is awaiting a command and may indicate a relevance metric of the corresponding device.

Abstract: Audio perception in local proximity to visual cues is provided. A device includes a video display, first row of audio transducers, and second row of audio transducers. The first and second rows can be vertically disposed above and below the video display. An audio transducer of the first row and an audio transducer of the second row form a column to produce, in concert, an audible signal. The perceived emanation of the audible signal is from a plane of the video display (e.g., a location of a visual cue) by weighing outputs of the audio transducers of the column. In certain embodiments, the audio transducers are spaced farther apart at a periphery for increased fidelity in a center portion of the plane and less fidelity at the periphery.

Abstract: Described is a method of processing position information indicative of an object position of an audio object, wherein the object position is usable for rendering of the audio object, that comprises: obtaining listener orientation information indicative of an orientation of a listener's head; obtaining listener displacement information indicative of a displacement of the listener's head; determining the object position from the position information; modifying the object position based on the listener displacement information by applying a translation to the object position; and further modifying the modified object position based on the listener orientation information. Further described is a corresponding apparatus for processing position information indicative of an object position of an audio object, wherein the object position is usable for rendering of the audio object.

Abstract: When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (?e) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to ?e=?log2(?log2(?{square root over (KMAX)}·O)?+1)?.

Abstract: Described is a method of processing position information indicative of an object position of an audio object, wherein the object position is usable for rendering of the audio object, that comprises: obtaining listener orientation information indicative of an orientation of a listener's head; obtaining listener displacement information indicative of a displacement of the listener's head; determining the object position from the position information; modifying the object position based on the listener displacement information by applying a translation to the object position; and further modifying the modified object position based on the listener orientation information. Further described is a corresponding apparatus for processing position information indicative of an object position of an audio object, wherein the object position is usable for rendering of the audio object.

Abstract: Described herein is a method (30) of rendering an audio signal (17) for playback in an audio environment (27) defined by a target loudspeaker system (23), the audio signal (17) including audio data relating to an audio object and associated position data indicative of an object position. Method (30) includes the initial step (31) of receiving the audio signal (17). At step (32) loudspeaker layout data for the target loudspeaker system (23) is received. At step (33) control data is received that is indicative of a position modification to be applied to the audio object in the audio environment (27). At step (38) in response to the position data, loudspeaker layout data and control data, rendering modification data is generated. Finally, at step (39) the audio signal (17) is rendered with the rendering modification data to output the audio signal (17) with the audio object at a modified object position that is between loudspeakers within the audio environment (27).

Abstract: A multi-stream rendering system and method may render and play simultaneously a plurality of audio program streams over a plurality of arbitrarily placed loudspeakers. At least one of the program streams may be a spatial mix. The rendering of said spatial mix may be dynamically modified as a function of the simultaneous rendering of one or more additional program streams. The rendering of one or more additional program streams may be dynamically modified as a function of the simultaneous rendering of the spatial mix.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A method of processing an audio signal includes the following operations. A system receives an audio signal. The system determines that a first frame of the audio signal includes a sparse transient signal. The system determines that a second frame of the audio signal includes a dense transient signal. The system compresses/expands (compands) the audio signal using a companding mle that applies a first companding exponent to the first frame of the audio signal and applies a second companding exponent to the second frame of the audio signal, each companding exponent being used to derive a respective degree of dynamic range compression and expansion for a corresponding frame. The system then provides the companded audio signal to a downstream device.