Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Methods, systems, and computer program products that provide streaming capabilities to audio input and output devices are disclosed. An audio processing device connects an upstream device to a downstream device. The upstream device is plugged into an input port of the audio processing device. The audio processing device intercepts a signal from the upstream device to the downstream device. The audio processing device converts the signal to digital data and streams the digital data to a server. The digital data can include metadata, e.g., an input gain. The audio processing device can adjust the input gain in response to instructions from the server. The audio processing device feeds a pass-through copy of the audio signal to an output port. A user can connect the downstream device in a usual signal chain into the output port of the audio processing device. The streaming does not affect the user's workflow.

Abstract: A method of coding at least one image comprising the steps of splitting the image into a plurality of blocks, of grouping said blocks into a predetermined number of subsets of blocks, of coding each of said subsets of blocks in parallel, the blocks of a subset considered being coded according to a predetermined sequential order of traversal. The coding step comprises, for a current block of a subset considered, the sub-step of predictive coding of said current block with respect to at least one previously coded and decoded block, and the sub-step of entropy coding of said current block on the basis of at least one probability of appearance of a symbol.

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: Methods and apparatus for decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or soundfield. The method may include receiving a bit stream containing the compressed HOA representation and decoding, based on a determination that there are multiple layers, the compressed HOA representation from the bitstream to obtain a sequence of decoded HOA representations. A first subset of the sequence of decoded HOA representations is determined based only on corresponding ambient HOA components. A second subset of the sequence of decoded HOA representations is determined based on corresponding ambient HOA components and corresponding predominant sound components.

Abstract: A handheld imaging device has a data receiver that is configured to receive reference encoded image data. The data includes reference code values, which are encoded by an external coding system. The reference code values represent reference gray levels, which are being selected using a reference grayscale display function that is based on perceptual non-linearity of human vision adapted at different light levels to spatial frequencies. The imaging device also has a data converter that is configured to access a code mapping between the reference code values and device-specific code values of the imaging device. The device-specific code values are configured to produce gray levels that are specific to the imaging device. Based on the code mapping, the data converter is configured to transcode the reference encoded image data into device-specific image data, which is encoded with the device-specific code values.

Abstract: Distortion reducing multi-band compressor with timbre preservation is provided. Timbre preservation is achieved by determining a time-varying threshold in each of a plurality frequency bands as a function of a respective fixed threshold for the frequency band and, at least in part, an audio signal level and a fixed threshold outside such frequency band. If a particular frequency band receives significant gain reduction due to being above or approaching its fixed threshold, then a time-varying threshold of one or more other frequency bands are also decreased to receive some gain reduction. In a specific embodiment, time-varying thresholds can be computed from an average difference of the audio input signal in each frequency band and its respective fixed threshold.

Abstract: A method for encoding an image having been cut up into partitions. The method includes: predicting data of a current partition based on an already encoded and then decoded reference partition, generating a predicted partition; determining residual data by comparing data relating to the current partition with the predicted partition, the residual data associated with various digital data items. Prior producing a signal containing the encoded information, performing the following steps: determining, from the predetermined residual data, a subset containing residual data capable of being modified; calculating the value of a function representative of the residual data; comparing the calculated value with a value of at least one of the digital data items; based on the comparison, modification or non-modification of at least one of the residual data items of the subset; and, in the event of a modification, entropy encoding the at least one modified residual data item.

Abstract: The present disclosure relates to a method of converting an audio signal in an intermediate signal format to a set of speaker feeds suitable for playback by an array of speakers. The audio signal in the intermediate signal format is obtainable from an input audio signal by means of a spatial panning function. The method comprises determining a discrete panning function for the array of speakers, determining a target panning function based on the discrete panning function, wherein determining the target panning function involves smoothing the discrete panning function, and determining a rendering operation for converting the audio signal in the intermediate signal format to the set of speaker feeds, based on the target panning function and the spatial panning function. The present disclosure further relates to a corresponding apparatus and a corresponding computer-readable storage medium.

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: Systems and methods are described for detecting and remedying potential incongruence in a video conference. A camera of a video conferencing system may capture video images of a conference room. A processor of the video conferencing system may identify locations of a plurality of participants within an image plane of a video image. Using face and shape detection, a location of a center point of each identified participant's torso may be calculated. A region of congruence bounded by key parallax lines may be calculated, the key parallax lines being a subset of all parallax lines running through the center points of each identified participant. When the audio device location is not within the region of congruence, audio captured by an audio device may be adjusted to reduce effects of incongruence when the captured audio is replayed at a far end of the video conference.

Abstract: A method for encoding a LUT defined as a lattice of vertices is disclosed. At least one value is of each vertex of the lattice. The method comprises for a current vertex: predicting the at least one value of said current vertex from another value which is for example obtained from reconstructed values of neighboring vertices; and encoding in a bitstream at least one residue computed between the at least one value of the current vertex and its prediction in a bitstream.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: Personalized audio metadata is generated based on audio program elements and presentation configuration metadata to specify personalized audio presentations for a media presentation. The personalized audio metadata is transmitted to an adaptive streaming client in response to receiving a media presentation request by the adaptive streaming client for the media presentation. Audio program elements of a specific personalized audio presentation are transmitted to the adaptive streaming client in response to receiving subsequent media streaming requests for the audio program elements.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: Apparatus and methods for generating an encoded audio bitstream, including by including program loudness metadata and audio data in the bitstream, and optionally also program boundary metadata in at least one segment (e.g., frame) of the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, e.g., including by performing adaptive loudness processing of the audio data of an audio program indicated by the bitstream, or authentication and/or validation of metadata and/or audio data of such an audio program. Another aspect is an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method.

Abstract: In some embodiments, an encoder device is disclosed to generate single-channel standard dynamic range/high dynamic range content predictors. The device receives a standard dynamic range image content and a representation of a high dynamic range image content. The device determines a first mapping function to map the standard dynamic range image content to the high dynamic range image content. The device generates a single channel prediction metadata based on the first mapping function, such that a decoder device can subsequently render a predicted high dynamic range image content by applying the metadata to transform the standard dynamic range image content to the predicted high definition image content.

Abstract: Apparatus, computer readable media and methods for processing a multi-channel, spatial audio format input signal. For example, one such method comprises determining object location metadata based on the received spatial audio format input signal; and extracting object audio signals based on the received spatial audio format input signal, wherein the extracting object audio signals based on the received spatial audio format input signal includes determining object audio signals and residual audio signals.

Abstract: Embodiments of source separation for reverberant environment are disclosed. According to a method, first microphone signals for each individual one of at least one source are captured respectively by at least two microphones for a period during which only the individual one produces sounds. Mixing parameters for modeling acoustic paths between the at least one source and the at least two microphones are learned by a processor based on the first microphone signals. Second microphone signals are captured respectively by the at least two microphones for a period during which all of the at least one source produce sounds. The reconstruction model is estimated by the processor based on the mixing parameters and second microphone signals. The processor performs the source separation by applying the reconstruction model.

Abstract: Example embodiments disclosed herein relate to separated audio analysis and processing. A system for processing an audio signal is disclosed. The system includes an audio analysis module configured to analyze an input audio signal to determine a processing parameter for the input audio signal, the input audio signal being represented in time domain. The system also includes an audio processing module configured to process the input audio signal in parallel with the audio analysis module. The audio processing module includes a time domain filter configured to filter the input audio signal to obtain an output audio signal in the time domain, and a filter controller configured to control a filter coefficient of the time domain filter based on the processing parameter determined by the audio analysis module. Corresponding method and computer program product of processing an audio signal are also disclosed.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A projector system comprising a laser light source, a collimating lens, a fly-eye lens, an integrating rod and a first modulator is disclosed. The light from a laser light source/fiber illuminates a collimator to substantially collimate the light and then is transmitted through a fly's-eye lens. The fly's-eye lens provides a desired angular/spatial light distribution for further processing to a first modulator of the projector system.

Abstract: In a method to improve the coding efficiency of high-dynamic range (HDR) images, a decoder parses sequence processing set (SPS) data from an input coded bitstream to detect that an HDR extension syntax structure is present in the parsed SPS data. It extracts from the HDR extension syntax structure post-processing information that includes one or more of a color space enabled flag, a color enhancement enabled flag, an adaptive reshaping enabled flag, a dynamic range conversion flag, a color correction enabled flag, or an SDR viewable flag. It decodes the input bitstream to generate a preliminary output decoded signal, and generates a second output signal based on the preliminary output signal and the post-processing information.

Abstract: This disclosure falls into the field of voice communication systems, more specifically it is related to the field of voice quality estimation in a packet based voice communication system. In particular the disclosure provides a method and device for reducing a prediction error of the voice quality estimation by considering the content of lost packets. Furthermore, this disclosure provides a method and device which uses a voice quality estimating algorithm to calculate the voice quality estimate based on an input which is switchable between a first and a second input mode.

Abstract: Techniques for driving a dual modulation display include generating backlight drive signals to drive individually-controllable illumination sources. The illumination sources emit first light onto a light conversion layer. The light conversion layer converts the first light into second light. The light conversion layer can include quantum dots or phosphor materials. Modulation drive signals are generated to determine transmission of the second light through individual subpixels of the display. These modulation drive signals can be adjusted based on one or more light field simulations. The light field simulations can include: (i) a color shift for a pixel based on a point spread function of the illumination sources; (ii) binning difference of individual illumination sources; (iii) temperature dependence of display components on performance; or (iv) combinations thereof.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: Embodiments are directed to a method of rendering adaptive audio by receiving input audio comprising channel-based audio, audio objects, and dynamic objects, wherein the dynamic objects are classified as sets of low-priority dynamic objects and high-priority dynamic objects, rendering the channel-based audio, the audio objects, and the low-priority dynamic objects in a first rendering processor of an audio processing system, and rendering the high-priority dynamic objects in a second rendering processor of the audio processing system. The rendered audio is then subject to virtualization and post-processing steps for playback through soundbars and other similar limited height capable speakers.

Abstract: A method of processing audio data for replay on a mobile device with a first speaker and a second speaker, wherein the audio data comprises a respective audio signal for each of the first and second speakers, includes: determining a device orientation of the mobile device; if the determined device orientation is vertical orientation, applying a first processing mode to the audio signals for the first and second speakers; and if the determined device orientation is horizontal orientation, applying a second processing mode to the audio signals for the first and second speakers.

Abstract: Projection systems and/or methods comprising a blurring element are disclosed. In one embodiment, a blurring element may comprise a first plate having a pattern on a first surface and second plate. The first plate and the second plate may comprise material having a slight difference in their respective index of refraction. In another embodiment, a blurring element may comprise a first plate having a pattern thereon and a second immersing material. The blurring element may be placed in between two modulators in a dual or multi-modulator projector system. The blurring element may be configured to give a desired shape to the light transmitted from a first modulator to a second modulator.

Abstract: In a method to reconstruct a high dynamic range video signal, a decoder receives parameters in the input bitstream to generate a prediction function. Using the prediction function, it generates a first set of nodes for a first prediction lookup table, wherein each node is characterized by an input node value and an output node value. Then, it modifies the output node values of one or more of the first set of nodes to generate a second set of nodes for a second prediction lookup table, and generates output prediction values using the second lookup table. Low-complexity methods to modify the output node value of a current node in the first set of nodes based on computing modified slopes between the current node and nodes surrounds the current node are presented.

Abstract: Methods to improve the quality of coding high-dynamic range (HDR) signals in the ICtCp color space are presented. Techniques are described to a) generate optimum chroma Offset and scaling parameters, b) compute chroma mode decisions by optimizing mode selection distortion metrics based on chroma saturation and hue angle values, and c) preserving iso-luminance by maintaining in-gamut values during chroma down-sampling and chroma up-sampling operations.

Abstract: A target view to a 3D scene depicted by a multiview image is determined. The multiview image comprises multiple sampled views. Each sampled view comprises multiple texture images and multiple depth images in multiple image layers. The target view is used to select, from the multiple sampled views of the multiview image, sampled views. A texture image and a depth image for each sampled view in the selected sampled views are encoded into a multiview video signal to be transmitted to a downstream device.

Abstract: A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.

Abstract: Example embodiments disclosed herein relate to signal processing. A method for decomposing a plurality of audio signals from at least two different channels is disclosed. The method comprises obtaining a set of components that are weakly correlated, the set of components generated based on the plurality of audio signals. The method comprises extracting a feature from the set of components, and determining a set of gains associated with the set of components at least in part based on the extracted feature, each of the gains indicating a proportion of a diffuse part in the associated component. The method further comprises decomposing the plurality of audio signals by applying the set of gains to the set of components. Corresponding system and computer program product are also disclosed.

Abstract: This disclosure falls into the field of voice communication systems, more specifically it is related to the field of voice quality estimation in a packet based voice communication system. In particular the disclosure provides methods, computer program products and devices for reducing a prediction error of the voice quality estimation by considering forward error correction of lost voice packets.

Abstract: A method for decoding video includes receiving quantized coefficients representative of a block of video representative of a plurality of pixels. The quantized coefficients are dequantized based upon a function of a remainder. The dequantized coefficients are inverse transformed to determine a decoded residue.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

Abstract: A method for decoding a video bitstream is disclosed. The method comprises: entropy decoding a first portion of a video bitstream, wherein first portion of video bitstream is associated with a video frame, thereby producing a first portion of decoded data; entropy decoding a second portion of video bitstream, wherein second portion of video bitstream is associated with video frame, thereby producing a second portion of decoded data, wherein entropy decoding second portion of video bitstream is independent of entropy decoding first portion of video bitstream; and reconstructing a first portion of video frame associated with video bitstream using first portion of decoded data and second portion of decoded data.

Abstract: A plurality of utterances of a first user from the language of the first user is translated into a language of a second user. The confidence scores associated with the translated utterances are compared with a confidence threshold. A predetermined utterance gap is adjusted based on the comparison. The predetermined utterance gap is a duration of time that occurs between utterances.

Abstract: Methods, systems, and computer program products for rending an audio object having an apparent size are disclosed. An audio processing system receives audio panning data including a first grid mapping first virtual sound sources in a space and speaker positions to speaker gains. The first grid specifies first speaker gains of the first virtual sound sources in the space. The audio processing system determines a second grid of second virtual sound sources in the space, including mapping the first virtual sound sources into the second virtual sound sources of the second virtual sources. The audio processing system selects at least one of the first grid or second grid for rendering an audio object based on an apparent size of the audio object. The audio processing system renders the audio object based on the selected grid or grids.

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

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.

Abstract: Overlapped block disparity estimation and compensation is described. Compensating for images with overlapped block disparity compensation (OBDC) involves determining if OBDC is enabled in a video bit stream, and determining if OBDC is enabled for one or more macroblocks that neighbor a first macroblock within the video bit stream. The neighboring macroblocks may be transform coded. If OBDC is enabled in the video bit stream and for the one or more neighboring macroblocks, predictions may be made for a region of the first macroblock that has an edge adjacent with the neighboring macroblocks. OBDC can be causally applied. Disparity compensation parameters or modes may be shared amongst views or layers. A variety of predictions may be used with causally-applied OBDC.

Abstract: Methods and systems for adaptive chroma reshaping are discussed. Given an input image, a luma-reshaped image is first generated based on its luma component. For each chroma component of the input image, the range of the pixel values in the luma reshaped image is divided into bins, and for each bin a maximal scale factor is generated based on the chroma pixel values in the input image corresponding to the pixels of the luma reshaped image in the bin. A forward reshaping function is generated based on a reference reshaped function and the maximal scale factors, and reshaped chroma pixel values for the chroma component are generated based on the forward reshaping function and the corresponding pixel values in the luma reshaped image. Implementations options using look-up tables for mobile platforms with limited computational resources are also described.

Abstract: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system configured to generate a bitstream indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system comprises a downmix processing unit configured to generate the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein the multi-channel input signal comprises n channels; n, m being integers with m<n. Furthermore, the system comprises a parameter processing unit configured to determine the spatial metadata from the multi-channel input signal.

Abstract: Methods and systems for generating a speaker feed for driving a sealed speaker, including: generating feedback indicative of excursion of the speaker; in response to the feedback, equalizing an input signal to generate the speaker feed, such that the speaker feed is equalized for driving the speaker with: a desired frequency response above the speaker's resonance frequency; and an at least substantially flat frequency response in a variable sub-resonance equalization band without exceeding the speaker's excursion limit, where the sub-resonance equalization band extends from the resonance frequency down to a variable cutoff frequency, and the cutoff frequency is determined in response to the feedback.

Abstract: A motion vector coding unit 117 executes processing including a neighboring block specification step (S100) of specifying a neighboring block which is located in the neighborhood of a current block; a judgment step (Steps S102, S104) of judging whether or not the neighboring block has been coded using a motion vector of another block; a prediction step (S106, S108) of deriving a predictive motion vector of the current block using a motion vector calculated from the motion vector of the other block as a motion vector of the neighboring block; and a coding step (S110) of coding the motion vector of the current block using the predictive motion vector.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.