Abstract: According to an example embodiment, a method for processing a multi-channel input audio signal representing a sound field into a multi-channel output audio signal representing said sound field in accordance with a predefined loudspeaker layout is provided, the method comprising the following for at least one frequency band: obtaining spatial audio parameters that are descriptive of spatial characteristics of said sound field; estimating a signal energy of the sound field represented by the multi-channel input audio signal; estimating, based on said signal energy and the obtained spatial audio parameters, respective output signal energies for channels of the multi-channel output audio signal according to said predefined loudspeaker layout; determining a maximum output energy as the largest of the output signal energies across channels of said multi-channel output audio signal; and deriving, on basis of said maximum output energy, a gain value for adjusting sound reproduction gain in at least one of said channe

Abstract: First and second pluralities of residual elements useable to reconstruct first and second respective parts of a representation of a signal are obtained. A transformation operation is performed to generate at least one correlation element. The transformation operation involves at least one residual element in the first plurality and at least one residual element in the second plurality. The at least one correlation element is dependent on an extent of correlation between the at least one residual element in the first plurality and the at least one residual element in the second plurality. The transformation operation is performed prior to the at least one correlation element being encoded.

Abstract: An apparatus for decoding an encoded signal includes: an audio decoder for decoding an encoded representation of a first set of first spectral portions to obtain a decoded first set of first spectral portions; a parametric decoder for decoding an encoded parametric representation of a second set of second spectral portions to obtain a decoded representation of the parametric representation, wherein the parametric information includes, for each target frequency tile, a source region identification as a matching information; and a frequency regenerator for regenerating a target frequency tile using a source region from the first set of first spectral portions identified by the matching information.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for performing residual quantization are described. One example of the methods includes performing a first residual quantization on a first target residual signal at a first bit rate to generate a first quantized residual signal. A second target residual signal is generated based at least on the first quantized residual signal and the first target residual signal. A second residual quantization is performed on the second target residual signal at a second bit rate to generate a second quantized residual signal, where the first bit rate is different from the second bit rate.

Abstract: Disclosed are a method and an apparatus for high frequency decoding for bandwidth extension. The method for high frequency decoding for bandwidth extension comprises the steps of: decoding an excitation class; transforming a decoded low frequency spectrum on the basis of the excitation class; and generating a high frequency excitation spectrum on the basis of the transformed low frequency spectrum. The method and apparatus for high frequency decoding for bandwidth extension according to an embodiment can transform a restored low frequency spectrum and generate a high frequency excitation spectrum, thereby improving the restored sound quality without an excessive increase in complexity.

Abstract: A signal processing method and device includes obtaining spectral coefficients of a current frame of an audio signal, in which N sub-bands of the current frame comprises at least one of the spectral coefficients. A total energy of M successive sub-bands of the N sub-bands, a total energy of K successive sub-bands of the N sub-bands, and an energy of a first sub-band are obtained to determine whether to modify original envelope values of the M sub-bands. When the original envelope values of the M sub-bands are modified, encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands.

Abstract: Various embodiments are disclosed for (possibly simultaneously) applying EQ and DRC to audio signals. In an embodiment, a method comprises: dividing an input audio signal into n frames, where n is a positive integer greater than one; dividing each frame of the input audio signal into Nb frequency bands, where Nb is a positive integer greater than one; for each frame n: computing an input level of the input audio signal in each band f, resulting in a input audio level distribution for the input audio signal; computing a gain for each band f based at least in part on a mapping of one or more properties of the input audio level distribution to a reference N audio level distribution computed from one or more reference audio signals; and applying each computed gain for each band f to each corresponding band f of the input audio signal.

Abstract: A device includes an encoder configured to determine, during a first period, that a first audio signal is a leading signal and that a second audio signal is a lagging signal. The encoder is also configured to generate a first frame of at least one encoded signal based on a first modified version of the second audio signal that is generated by adjusting the second audio signal based on a first mismatch value. The encoder is configured to determine, during a second period, that the first audio signal is the leading signal and that the second audio signal is the lagging signal. The encoder is configured to generate a second frame of the at least one encoded signal based on a second modified version of the second audio signal that is generated by adjusting the second audio signal based on the first mismatch value and a second mismatch value.

Abstract: A method for transmitting files based on an audio signal includes a first terminal and a second terminal, the first terminal receives a file identification of a target file transmitted by a server; obtains a first audio signal, hiding the file identification in the first audio signal to form a first signal, and plays the first signal; a second terminal samples at least a portion of the first signal, and gets the file identification according to the at least a portion of the first signal; transmits the file identification to the server; the server sends the target file to the second terminal according to the file identification.

Abstract: The invention discloses a system applied to optimize the perceived sound quality in virtual sound zones. The system includes a method to establish a threshold of acceptability for an interfering audio programme on a target audio programme. The method includes physical parameters like target programme and interferer programme which combined with the scenarios: information gathering, entertainment and reading and/or working, constitutes modes of operations that may be processed and controlled by a system controller.

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: A mobile apparatus and control method thereof are provided, which includes an audio data signal input unit arranged to receive an audio data signal. An audio output unit is arranged to output an audio signal according to the received audio data signal within a first reproduction frequency. A balanced armature is arranged to output an audio signal according to the received audio data signal within a second reproduction frequency band. The second reproduction frequency band is different from the first reproduction frequency band. An audio signal processor is arranged to adjust the first reproduction frequency band and the second reproduction frequency band such that a combination of the first reproduction frequency band and the second reproduction frequency band is wider than the first reproduction frequency band and the second reproduction frequency band individually.

Abstract: A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.

Abstract: A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

Abstract: An audio input device is provided which can include a number of features. In some embodiments, the audio input device includes a housing, a microphone carried by the housing, and a processor carried by the housing and configured to modify an input sound signal so as to amplify frequencies corresponding to a target human voice and diminish frequencies not corresponding to the target human voice. In another embodiment, an audio input device is configured to treat an auditory gap condition of a user by extending gaps in continuous speech and outputting the modified speech to the user. In another embodiment, the audio input device is configured to treat a dichotic hearing condition of a user. Methods of use are also described.

Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

Abstract: Gapless looping playback of an audio sound seamlessly without any silence between loops. An uncompressed digital audio file is divided and encoded using an MP3 encoder to produce two encoded files, A and B. The MP3 encoder adds a padding delay at the beginning and end of each file. Two timers are defined: T1=A?(a+b) and T2=B?(d+c), where a is the padding delay at the end of A, b is the delay at the beginning of B, c is the delay at the beginning of A, and d is the delay at the end of B. Simultaneously, a decoder starts to decode A on Track 1 and T1 is started. Immediately after T1 expires, a decoder starts to decode B on Track 2 and T2 is started. Thanks to the timers, the respective end and beginning of the sound portions of A and B temporally align so that no silence is heard.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.

Abstract: A method for embedding digital information into an audio signal, is provided. The method includes dividing the digital information into low-priority data and high-priority data; dividing the audio signal into first and second signal parts; embedding at least one echo signal into the first signal part; embedding a communication signal modulated with low-priority data, which has a spectrum according to psychoacoustic analysis of the second signal part, into the second signal part; and combining the embedded first and second signal parts.

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: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.

Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state 715 located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation 745 of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: A computing device identifies a first codeword in a first codebook to represent short-timescale information of frames in a time-based data item segmented at intervals and identifies a second codeword in a second codebook to represent long-timescale information of the frames. The computing device generates a third codebook based on the first codeword and the second codeword for the frames to add long-timescale information context to the short-timescale information of the frames.

Abstract: Adaptive Gain Control (AGC) is performed directly in a coded domain. A Coded Domain Adaptive Gain Control (CD-AGC) system modifies at least one parameter of a first encoded signal, resulting in corresponding modified parameter(s). The CD-VQE system replaces the parameter(s) of the first encoded signal with the modified parameter(s), resulting in a second encoded signal. In a decoded state, the second encoded signal approximates a target signal that is a function of two signals, including the first encoded signal and a third encoded signal, in at least a partially decoded states. Thus, the first encoded signal does not have to go through intermediate decode/re-encode processes, which can degrade overall speech quality. Computational resources required for a complete re-encoding are not needed. Overall delay of the system is minimized. The CD-AGC system can be used in any network in which signals are communicated in a coded domain, such as a Third Generation (3G) wireless network.

Abstract: Downmixing multi-channel audio signals to target channels by pre-downmixing frequency coefficients that are encoded using a most frequently used block type in stereo channels in the frequency domain, thereby reducing an amount of calculations and an amount of power required to downmix the multi-channel audio signals.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A method, a device and a system for voice encoding/decoding are disclosed in the present invention. The method includes: assembling an input pulse code modulation signal into one signal according to a designated time slot and assembly manner; and encoding the assembled signal according to a designated encoding manner to output an encoded voice signal. In the present invention, because a process of assembling or splitting the signal may be implemented through software, in the case that hardware in a current network does not need to be replaced, an effect of encoding/decoding voice with a 7 K spectrum may be achieved in the current network.

Abstract: Encoding and decoding of residual signals are provided. In a method of encoding a residual signal of an audio signal, the residual signal is divided into a plurality of sections having different sizes, based on a change of the residual signal. Then, section division information representing information about the divided sections and section-by-section residual signal information representing characteristics of the sections of the residual signal are acquired. Thereafter, the residual signal is encoded based on the section division information and the section-by-section residual signal information.

Abstract: A method for minimizing message collision in a device is presented. Two or more overlapping real-time streaming simplex audio messages are received. One of the audio messages is forwarded to be reproduced while the other is stored. Forwarding of the delayed audio message is delayed such that overlapping reproduction of the audio messages is minimized. Reproduction is delayed until a predetermined clock time expires or is dependent on one or more of: the length of the second audio message or the amount of overlap of the first and second audio messages.

Abstract: Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal.

Abstract: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: An apparatus for calculating bandwidth extension data of an audio signal in a bandwidth extension system, in which a first spectral band is encoded with a first number of bits and a second spectral band different from the first spectral band is encoded with a second number of bits, the second number of bits being smaller than the first number of bits, has a controllable bandwidth extension parameter calculator for calculating bandwidth extension parameters for the second frequency band in a frame-wise manner for a sequence of frames of the audio signal. Each frame has a controllable start time instant. The apparatus additionally includes a spectral tilt detector for detecting a spectral tilt in a time portion of the audio signal and for signaling the start time instant for the individual frames of the audio signal depending on spectral tilt.

Abstract: An uplink or downlink audio processor contains a multi band compressor that receives an input, uplink or downlink, audio signal. The multi-band compressor has a band splitter that splits the input audio signal into a number of different band signals. Each band signal is input to a respective compressor block, which is independently programmable so that its audio frequency response (a) differs from a linear response in at least two non-overlapping windows of its input signal, and (b) differs from the frequency response of another one of the compressor blocks. Other embodiments are also described and claimed.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information.

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag value according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag value is a first value; obtaining a second-domain predictive signal according to the LP processing result when the long-term flag value is a second value; converting the second-domain predictive signal into a first-domain predictive signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract: A hierarchical audio coding, decoding method and system are provided. The method includes dividing frequency domain coefficients of an audio signal after MDCT into a plurality of coding sub-bands, quantizing and coding amplitude envelope values of coding sub-bands; allocating bits to each coding sub-band of the core layer, quantizing and coding core layer frequency domain coefficients to obtain coded bits of core layer frequency domain coefficients; calculating the amplitude envelope value of each coding sub-band of the core layer residual signal; allocating bits to each coding sub-band of the extended layer, quantizing and coding the extended layer coding signal to obtain coded bits of the extended layer coding signal; multiplexing and packing amplitude value envelope coded bits of each coding sub-band composed by core layer and extended layer frequency domain coefficients, core layer frequency coefficients coded bits, and extended layer coding signal coded bits, then transmitting to the decoding end.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Abstract: A method and apparatus for decoding portions of a data stream, wherein each portion comprises a plurality of samples. The method comprises storing portions of the data stream, decoding portions of the data stream to form decoded portions, and storing the decoded portions. The method further comprises identifying that a portion of the data stream is degraded. Following identifying that a portion of the data stream is degraded, the method generates a decoded portion for the degraded portion of the data stream using the stored decoded portions. The method also updates a state of a decoder by: estimating a pitch period of the degraded portion; selecting a group of successive samples of the stored portions of the data stream, the group of successive samples offset from the degraded portion in the data stream by a multiple of the estimated pitch period; and decoding the selected samples at the decoder.

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.

Abstract: An audio decoder includes an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically encoded representation of the spectral values, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates a hash table, entries of which define both significant state values and boundaries of intervals of numeric context values, in order to select the mapping rule. A mapping rule index value is individually associated to a numeric context value being a significant state value.

Abstract: Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC.