Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Abstract: According to the present disclosure, lighting sources, having operating parameter(s) which is controllable in lighting sequence(s) as a function of a time code data set coupled with the sequence, are controlled: by providing a repository of operating data files for the lighting sources coupled with lighting sources, with each data file including time code data set(s) for lighting sequence(s) for one of the lighting sources, by detecting ambient lighting level signal(s) at the ambient where the lighting sources are located, by retrieving in the data repository operating data file(s) coupled with a selected one of the lighting sources, and by operating the selected lighting source by controlling the operating parameter(s) as a function of the operating data included in the operating data file retrieved and as a function of the ambient lighting level signal.

Abstract: Systems and methods for low-power single-wire communication are provided. In some embodiments, a method of operation of a transmitter to transmit a data word to a receiver using low-power single-wire communication includes receiving the data word to be transmitted to the receiver. The method also includes encoding the data word to be transmitted in a Pulsed Index Communication (PIC) format to produce a PIC data word and transmitting the PIC data word to the receiver. In this way, the transmitter may be able to transmit an increased amount of data while maintaining a simple communication protocol that uses low power and does not require a Clock-Data Recovery circuit.

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 audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Abstract: Higher Order Ambisonics (HOA) signals can be compressed by decomposition into a predominant sound component and a residual ambient component. The compressed representation comprises pre-dominant sound signals, coefficient sequences of the ambient component and side information. For efficiently combining HOA decompression and HOA rendering to obtain loudspeaker signals, combined rendering and decoding of the compressed HOA signal comprises perceptually decoding the perceptually coded portion and decoding the side information, without reconstructing HOA coefficient sequences. For reconstructing components of a first type, fading of coefficient sequences is not required, while for components of a second type fading is required. For each second type component, different linear operations are determined: one for coefficient sequences that in a current frame require no fading, one for those that require fading-in, and one for those that require fading-out.

Abstract: A method for processing a signal includes receiving a first frame of an input audio bitstream at a decoder. The first frame includes at least one signal associated with a frequency range. The method also includes decoding the at least one signal to generate at least one decoded signal having an intermediate sampling rate. The intermediate sampling rate is based on coding information associated with the first frame. The method further includes generating a resampled signal based at least in part on the at least one decoded signal. The resampled signal has an output sampling rate of the decoder.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

Abstract: A device for processing audio data obtains data representing quantized versions of a set of one or more spatial vectors. Each respective spatial vector of the set of spatial vectors corresponds to a respective audio signal of the set of audio signals. Each of the spatial vectors is in a Higher-Order Ambisonics (HOA) domain and is computed based on a set of loudspeaker locations. The device inverse quantizes the quantized versions of the spatial vectors.

Abstract: A schematic block diagram of a decoder for decoding an encoded audio signal is shown. The decoder includes an adaptive spectrum-time converter and an overlap-add-processor. The adaptive spectrum-time converter converts successive blocks of spectral values into successive blocks of time values, e.g. via a frequency-to-time transform. Furthermore, the adaptive spectrum-time converter receives a control information and switches, in response to the control information, between transform kernels of a first group of transform kernels including one or more transform kernels having different symmetries at sides of a kernel, and a second group of transform kernels including one or more transform kernels having the same symmetries at sides of a transform kernel. Moreover, the overlap-add-processor overlaps and adds the successive blocks of time values to obtain decoded audio values, which may be a decoded audio signal.

Abstract: Systems and method for requesting audio content that is stored in a cloud network and accessed by many different audio devices where each of the audio devices may have an optimal type of audio encoding, format, and characteristic in addition to being located in different types of listening environment that are considered by a cloud audio server when preparing audio content for transmission to a requesting audio device.

Abstract: Methods and apparatus are provided that relate to receiving a bit pattern that is representative of a binary data sequence; generating a multi-bit data byte sequence that is representative of the bit pattern by, for at least one bit in the bit pattern: outputting a larger-valued multi-bit data byte that is larger than a byte-sequencer reference data byte according to a first criteria, outputting a smaller-valued multi-bit data byte that is smaller than the byte-sequencer reference data byte according to a second criteria; and encoding the generated multi-bit data byte sequence by, for each generated multi-bit data byte: outputting a single bit whose value is indicative of whether the generated multi-bit data byte is larger or smaller than a current reference multi-bit data byte and updating the current reference multi-bit data byte based on the generated multi-bit data byte.

Abstract: Disclosed are techniques for generating an error concealment signal, where such techniques may include an LPC representation generator for generating a replacement LPC representation; a gain calculator for calculating a gain information from the LPC representations; a compensator for compensating a gain influence of the replacement LPC representation using the gain information; and an LPC synthesizer for filtering codebook information using the replacement LPC representation to obtain the error concealment signal, where the compensator is configured for weighting the codebook information or an LPC synthesis output signal.

Abstract: An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (a) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A residual scaling unit is configured to determine a scaling factor for a residual channel based on an inter-channel mismatch value. The inter-channel mismatch value is indicative of a temporal alignment between a reference channel and a target channel. The residual scaling unit is further configured to scale (e.g., attenuate) the residual channel by the scaling factor to generate a scaled residual channel. A residual channel encoder is configured to encode the scaled residual channel as part of a bitstream.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: A broadcast transmitting apparatus and method, and a broadcast playback apparatus and method for providing an object-based audio by encoding and decoding a multichannel audio signal are provided. The broadcast transmitting apparatus may generate audio identification information used to determine whether the multichannel audio signal is an object-based audio signal. When the multichannel audio signal is determined to be the object-based audio signal, based on the audio identification information, the broadcast playback apparatus may control and output the multichannel audio signal for each channel.

Abstract: A method includes generating a windowed time-domain mid channel by applying two first asymmetric windows to a first frame of a time-domain mid channel and applying two second asymmetric windows to a second frame of the time-domain mid channel. The method includes transforming the windowed time-domain mid channel to a transform domain to generate sets of transform-domain mid channel data including first transform-domain mid channel data corresponding to a first mid channel window of the first frame and second transform-domain mid channel data corresponding to a second mid channel window of the first frame. The method includes performing an up-mix operation using the sets of transform-domain mid channel data, stereo parameters from the bit stream, and an interpolated parameter determined using an unevenly weighted interpolation between a first stereo parameter value associated with the first frame and a second stereo parameter value associated with the second frame.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor.

Abstract: An audio/speech encoding apparatus/method and an audio/speech decoding apparatus/method are provided. The audio/speech encoding apparatus includes a memory that stores instructions, and a processor that performs operations. The operations include transforming a time domain input audio/speech signal to a frequency spectrum, dividing the frequency spectrum to a plural of bands, calculating norm factors, and quantizing the norm factors. The operations also include calculating differential indices between an Nth band index and an (N?1)th band index, and modifying a range of the differential indices for the Nth band when N is 2 or more. The operations further include replacing the differential index with the modified differential index, and not modifying a range of the differential indices for the Nth band when N is 1. The apparatus encodes the differential indices using a selected Huffman table, and transmits the encoded differential indices and a flag signal over a communication network.

Abstract: Systems and methods for dynamically causing a terminal to change audio codec modes during a session with another terminal. In one aspect, the terminal is forced use to switch to a more resilient audio codec mode when the quality of wireless channel being used by the terminal deteriorates below a threshold.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component ({tilde over (C)}AMB(k?1)). The ambient HOA component ({tilde over (C)}AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: The present disclosure discloses an audio processing method and apparatus based on artificial intelligence. A specific embodiment of the method comprises: converting a to-be-processed audio to a to-be-processed picture; extracting a content characteristic of the to-be-processed picture; determining a target picture based on a style characteristic and the content characteristic of the to-be-processed picture, the style characteristic being obtained from a template picture converted from a template audio; and converting the target picture to a processed audio. The present embodiment achieves the processing effect that the processed audio takes a template audio style, improves the efficiency and the flexibility of audio processing, while without changing the content of the to-be-processed audio.

Abstract: Present disclosure pertains to an audio signal processing method and device. A current frame of a frequency-domain audio signal has N frequency sub-bands. According to an energy attribute and a spectral attribute of a first subset of M sub-bands, whether to adjust envelope values of the M sub-bands is determined. Based on a determination that the energy envelope values of the M sub-bands need to be adjusted, the energy envelope values of the M sub-bands are modified individually to obtain modified envelope values of the M sub-bands. Encoding bits are allocated to each of the N sub-bands according to the adjusted envelope values of the M sub-bands and energy envelope information of K sub-bands of a second subset. The first subset and the second subset have no overlap in frequency, and N=M+K.

Abstract: An apparatus for decomposing an input signal having a number of at least three input channels includes a downmixer for downmixing the input signal to obtain a downmixed signal having a smaller number of channels. Furthermore, an analyzer for analyzing the downmixed signal to derive an analysis result is provided, and the analysis result is forwarded to a signal processor for processing the input signal or a signal derived from the input signal to obtain the decomposed signal.

Abstract: Disclosed is a method of synchronization signal correlation, comprising receiving a synchronization signal (“SS”) on a plurality of receive antennas; performing a signal revision on the SS received on a first receive antenna, the signal revision comprising performing an element-wise squaring of the SS; calculating a complex coefficient by summing the element-wise squared SS; estimating a phase angle of the complex coefficient; and performing an element-wise phase correction of the SS; the method synchronization signal correlation further comprising performing the signal revision on the SS received on a second receive antenna; conditioning the revised signals by constructively and destructively combining the revised signals; selecting an optimized conditioned signal, wherein the optimized conditioned signal is the constructive or destructive combination with the largest Euclidian norm; correlating the optimized conditioned signal with a series of candidate SSs; and selecting the correlation with the largest ma

Abstract: Example methods and apparatus to measure exposure to broadcast signals having embedded data are disclosed. An example broadcast signal exposure meter includes a first decoder to obtain an identifier of a broadcast station from an audio signal output by an end user broadcast receiver, a radio to tune to a broadcast signal from the broadcast station associated with the identifier of the broadcast station, a second decoder to obtain embedded data from the broadcast signal, the embedded data representing media contained in the broadcast signal, and an interface to provide the embedded data to a server, the server to determine audience measurement information for the media based on the provided embedded data.

Abstract: A method and associated systems for a resource-utilization monitor with self-adjusting sample size. A processor monitors availability of a resource-limited entity by analyzing a set of samples that each identify, at the time the sample was recorded, an unused amount of resource available to the entity. The processor computes a Chi-square statistic of the sample set. If the statistic reveals that the number of samples in the sample set is too small to produce valid results, the processor adjusts the sample size to specify a larger number of samples and repeats these steps. If the number of samples is large enough to produce statistically valid results, the processor analyzes the sample set to determine whether the current amount of remaining resource is undesirably low, is likely to become undesirably low during a short-term period of time, or is likely to become undesirably low during a longer-term period of time.

Abstract: Utilizing the systems disclosed herein, a network element (in a network) controls, within another network, the constraints of a service, timing of the creation of the service, and selection a service on which a packet is transmitted. For example, a first network element (located in a first network) receives a request associated with initiating a service. The request is received from a second network element located in a second network and includes at least one path constraint. The first network element controls creation of the service in the first network on behalf of the second network element located in the second network by, e.g., identifying a path based, at least in part, on the at least one path constraint; and binding an identifier and an interface to the path, wherein the interface is associated with one or more operation to perform on any traffic that is labeled with the identifier.

Abstract: An apparatus for processing an audio signal including a sequence of blocks of spectral values includes: a processor for processing the sequence of blocks using at least one modification values for a first block to obtain aliasing-reduced or aliasing-free first result signal in an overlap range and using at least one second different modification value for a second block of the sequence of blocks to obtain an aliasing-reduced or aliasing-free second result signal in the overlap range; and a combiner for combining the first result signal and the second result signal in the overlap range to obtain a processed signal for the overlap range.

Abstract: An apparatus includes a receiver configured to receive at least one encoded signal that includes inter-channel bandwidth extension (BWE) parameters. The device also includes a decoder configured to generate a mid channel time-domain high-band signal by performing bandwidth extension based on the at least one encoded signal. The decoder is also configured to generate, based on the mid channel time-domain high-band signal and the inter-channel BWE parameters, a first channel time-domain high-band signal and a second channel time-domain high-band signal. The decoder is further configured to generate a target channel signal by combining the first channel time-domain high-band signal and a first channel low-band signal, and to generate a reference channel signal by combining the second channel time-domain high-band signal and a second channel low-band signal. The decoder is also configured to generate a modified target channel signal by modifying the target channel signal based on a temporal mismatch value.

Abstract: Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element.

Abstract: Systems and methods are described for dynamically suppressing non-linear distortion for a device, such as a speakerphone. A device may receive a signal, where the device has non-linear distortion at a predetermined frequency. The received signal may be analyzed to compute a tone strength parameter and a band level. The received signal may be filtered such that a spectrum of the input signal is dynamically limited by reducing suppression of the non-linear distortion when the tone strength parameter is in a lower portion of a predetermined range and increasing suppression of the non-linear distortion when the tone strength parameter is in an upper portion of the predetermined range, the predetermined range of the tone strength parameter corresponding to a loudness range of the device.

Abstract: A user of a network-based system may correspond to a user profile that describes the user. The user profile may describe the user using one or more descriptors of items that correspond to the user (e.g., items owned by the user, items liked by the user, or items rated by the user). In some situations, such a user profile may be characterized as a “taste profile” that describes an array or distribution of one or more tastes, preferences, or habits of the user. Accordingly, the user profile machine within the network-based system may generate the user profile by accessing descriptors of items that correspond to the user, clustering one or more of the descriptors, and generating the user profile based on one or more clusters of the descriptors.

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 echo canceller device includes a band adjuster including a filter for performing band adjustment by applying a gain to a signal in a specific frequency band of an input signal, an adaptive filter for inputting a speech reception signal, updating a filter coefficient, and generating a pseudo echo signal using the updated filter coefficient, a band corrector including a filter for correcting a frequency characteristic of a signal obtained by subtracting the pseudo echo signal from an input signal, to a frequency characteristic of the input signal prior to the band adjustment, and a residual echo suppressor for suppressing an echo component that remains in a signal, by a suppression amount, and the residual echo suppressor sets the suppression amount for a signal in a specific frequency band that has been subjected to the band adjustment.

Abstract: A system configured to improve speech quality by performing residual echo suppression (RES). The system may detect when double-talk conditions are present in individual frequency bands during a voice conversation and may determine gain values for the individual frequency bands. The system may determine whether double-talk conditions are present based on a normalized cross power spectral density function in a frequency domain. If double-talk conditions are present in a frequency band or far end energy is low, the system may determine a gain value that passes audio data in the frequency band, whereas if double-talk conditions are not present, the system may determine a gain value that attenuates audio data in the frequency band. The system may determine binary gain values using a decision threshold value or continuous gain values using a mapping function. The system may control an amount of suppression by selecting different mapping functions and/or parameters.

Abstract: Systems and methods for holistically modelling audio feedback and removing the entire feedback signal corresponding thereto. The systems can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume), than those conventional systems which seek to remove singing frequencies with PEQs. The systems are an improvement over traditional audio feedback elimination systems which attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling, and those feedback elimination systems which simply employ adaptive notch filtering to detect and “notch” the so-called “singing” or “howling” frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

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: An audio signal decoding method includes: obtaining mode information of a high frequency band signal of an audio signal and indices of a low frequency band signal of the audio signal by parsing a received bitstream; obtaining the low frequency band signal based on the indices; predicting an excitation signal of a high frequency band signal based on the low frequency band signal; and reconstructing the high frequency band signal based on the frequency envelope and the excitation signal. A manner for obtaining the frequency envelope of the high frequency band signal when mode information indicates the high frequency band signal is a harmonic type signal is different from a manner for obtaining the frequency envelope of the high frequency band signal when the mode information indicates the high frequency band signal is not a harmonic type signal.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: Methods, systems and computer program products that allow for the determination of human presence in a room where content is being presented. The audio that is associated with the content may be captured, along with the audio that is being generated collectively by whatever sources may be in the room including the presentation of the content. Features may be extracted from both the content audio and the room audio. These features may then be compared, and the differences may be quantified. If the differences are significant, then human presence may be inferred.

Abstract: A method includes generating a high-band residual signal based on a high-band portion of an audio signal. The method also includes generating a harmonically extended signal at least partially based on a low-band portion of the audio signal. The method further includes determining a mixing factor based on the high-band residual signal, the harmonically extended signal, and modulated noise. The modulated noise is at least partially based on the harmonically extended signal and white noise.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: An audio encoder device includes an audio encoder configured for producing an encoded audio bitstream from an audio signal having consecutive audio frames; a dynamic range control encoder configured for producing an encoded dynamic range control bitstream from an dynamic range control sequence corresponding to the audio signal and having consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames has one or more nodes, wherein each node of the one or more nodes has gain information for the audio signal and time information indicating to which point in time the gain information corresponds.