Abstract: A method comprises: vector quantizing input vectors representative of audio into an original sequence including indices of codewords of a codebook; generating candidate sequences including the indices of the codewords of the codebook by evaluating, for each candidate sequence, transition costs for transitions between the indices based on (i) transition probabilities of the transitions, and (ii) distances between the codewords represented by the indices and the input vectors that corresponds to the indices; determining a preferred candidate sequence of the candidate sequences to replace the original sequence based on the transition costs for each candidate sequence; and transmitting the preferred candidate sequence in place of the original sequence.

Abstract: A decoder-side method for outputting several audio channels of a sound program is described. An audio channel of the sound program, a residual signal, a gain parameter, and a delay parameter are received, for example within a bitstream. The audio channel is adjusted in accordance with the gain parameter and the delay parameter, to produce an adjusted audio signal, and is then combined with the residual signal to produce a combined signal. The audio channel is output as a first audio channel of the sound program for playback, while the combined signal is output as a second audio channel of the sound program. Other aspects are also described and claimed.

Abstract: An inter-channel phase difference (IPD) parameter extraction method includes obtaining a parameter for obtaining an information extraction manner for a current frame of a multi-channel signal; obtaining an IPD parameter extraction manner for the current frame based on the parameter for obtaining the information extraction manner, where the obtained IPD parameter extraction manner is one of at least two preset IPD parameter extraction manners; and obtaining an IPD parameter of the current frame based on the obtained IPD parameter extraction manner for the current frame.

Abstract: This disclosure provides a encoding method, and an encoder for a multi-channel signal. The encoding method includes: obtaining a first ITD of a current frame of a multi-channel signal includes an initial left channel signal and an initial right channel signal; obtaining a second ITD of the current frame based on the first ITD and a third ITD of a previous frame of the multi-channel signal; performing delay alignment on the initial left channel signal and the initial right channel signal based on the second ITD, to obtain an aligned left channel signal and an aligned right channel signal; and encoding the aligned left channel signal and the aligned right channel signal.

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

Abstract: This application provides example psychoacoustics-based audio encoding methods and apparatuses. One example method includes receiving audio data. The audio data can be decoded. Auditory feature information of a user can be obtained, where the auditory feature information includes at least one of the following: personal information, listening test result information, or frequency response curve information. A psychoacoustics model parameter of the user can be calculated based on the auditory feature information of the user, where the psychoacoustics model parameter includes at least one of the following: an intra-band masking parameter, a slope of a low-frequency inter-band masking line, a slope of a high-frequency inter-band masking line, or a human ear quiet threshold curve. The decoded audio data can be encoded based on the psychoacoustics model parameter of the user.

Abstract: The present invention discloses a video processing circuit, which is coupled to a memory chip and includes an image processing circuit. The image processing circuit includes a first channel, a second channel and a compression circuit. The two channels process first image data and second image data to generate first processed image data and second processed image data, respectively. The compression circuit compresses the first processed image data and the second processed image data to generate first compressed image data and second compressed image data, respectively. A memory block in the memory chip is configured as a ring buffer shared by the first channel and the second channel so as to store the first compressed image data and the second compressed image data.

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: Disclosed is an apparatus and method for audio encoding/decoding that is robust against coding distortion in a transition section. An audio encoding method includes outputting a frequency domain signal by time-to-frequency (T/F) transform of an input signal, outputting a frequency domain residual signal in which a frequency axis envelope is removed from the frequency domain signal by applying frequency domain noise shaping (FDNS) encoding to the frequency domain signal, outputting a time domain residual signal in which a time axis envelope is removed by performing linear prediction coefficient (LPC) analysis based on the frequency domain residual signal, and quantizing and transmitting the time domain residual signal.

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

Abstract: This disclosure provides a network call method and apparatus, a computer device, and a storage medium, and belongs to the field of audio data processing. The method includes: performing time-frequency transformation on an acquired audio signal, to obtain a plurality of pieces of frequency domain information of the audio signal; determining a target bit rate corresponding to the audio signal according to the plurality of pieces of frequency domain information; and encoding the audio signal based on the target bit rate, and performing a network call based on the encoded audio signal.

Abstract: An apparatus for combining three or more audio signals is described. The apparatus includes a segmentation block for segmenting each audio signal into segments, a weight determination block, which is configured to determine a weight value for each of the temporally weighted audio signal segments, a combination block for combining the temporally weighted audio signal segments of each audio signal, and a synthesis block for generating an output audio signal. A method for combining three or more audio signals and a computer program product are also described.

Abstract: Techniques and solutions are described for encoding digital video files, such as for streaming applications. Data associated with the digital video file forms a dataset that can be characterized by a measure of the dataset's center, such as an average, and a spread of the dataset, such as a deviation, with respective to a bitrate over a duration of the digital video file. The measure of center and spread are used to calculate a deviation-adjusted bitrate. A deviation adjusted bitrate can be calculated for the entire digital video file, or for particular subsets of the digital video file, such as for segments of a duration forming units of video streaming Disclosed techniques can provide various advantages, including using a reduced bitrate for video or video portions as compared with an average or static bitrate, for lower-complexity video, or using a higher bitrate for video or video portions for higher-complexity video.

Abstract: To reduce a processing load at a reception side in a case where a plurality kinds of audio data items are transmitted. A container in a predetermined format having a predetermined number of audio streams including coded data items of a plurality of groups is transmitted. For example, the coded data items of the plurality of groups include either or both of channel coded data items and object coded data items. Attribute information indicating respective attributes of the coded data items of the plurality of groups is inserted into a layer of the container and/or a layer of an audio stream. For example, stream correspondence relation information indicating which audio stream includes each of the coded data items of the plurality of groups is further inserted.

Abstract: There is inter alia disclosed an apparatus for spatial audio encoding comprising: means for determining at least two of a type of spatial audio parameter for one or more audio signals, wherein a first of the type of spatial audio parameter is associated with a first group of samples in a domain of the one or more audio signals and a second of the type of spatial audio parameter is associated with a second group of samples in the domain of the one or more audio signals; and means for merging the first of the type of spatial audio parameter and the second of the type of spatial audio parameter into a merged spatial audio parameter.

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

Abstract: Disclosed is an apparatus and method for processing a multichannel audio signal. A multichannel audio signal processing method may include: generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; and generating a stereo audio signal by performing binaural rendering of the N-channel audio signal.

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: The technology generally relates to spatial audio communication between devices. For example, a first device and a second device may be connected via a communication link. The first device may capture audio signals in an environment through two or more microphones. The first device may encode the captured audio with spatial configuration data. The first device may transmit the encoded audio via the communication link to the second device. The second device may decode the encoded audio into binaural or ambisonic audio to be output by one or more speakers of the second device. The binaural or ambisonic audio may be converted into spatial audio to be output. The second device may output the binaural or spatial audio to create an immersive listening experience.

Abstract: An audio signal coding method is provided that includes: obtaining a current frame of an audio signal; obtaining a coding parameter based on a power spectrum ratio of a current frequency in a current frequency area of at least a part of signals of the current frame, where the coding parameter indicates tonal component information of the at least a part of signals, the tonal component information includes at least one of location information of a tonal component, quantity information of tonal components, amplitude information of the tonal component, or energy information of the tonal component, and the power spectrum ratio of the current frequency is a ratio of a value of a power spectrum of the current frequency to a mean value of power spectrums of the current frequency area; and performing bitstream multiplexing on the coding parameter to obtain a coded bitstream.

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: An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.

Abstract: A vehicle control system includes a transmitting electronic control unit for transmitting multiplexed protocol data units. The transmitting electronic control unit is enabled to transmit first and second protocol data units with corresponding first and second selector field values. A receiving electronic control unit is in electronic communication with the transmitting electronic control unit. The receiving electronic control unit is enabled to be programmed to operate in a first mode of operation or in a second mode of operation. The receiving electronic control unit is enabled to receive and process the first protocol data unit based on receipt of the first selector field value when operating in the first mode of operation and is enabled to receive and process the second protocol data unit based on receipt of the second selector field value when operating in the second mode of operation.

Abstract: A time-domain stereo parameter encoding method and a related product are provided. The time-domain stereo parameter encoding method includes: determining a channel combination scheme for a current frame; determining a time-domain stereo parameter of the current frame based on the channel combination scheme for the current frame; and encoding the determined time-domain stereo parameter of the current frame, where the time-domain stereo parameter includes at least one of a channel combination ratio factor or an inter-channel time difference.

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 decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: There is inter alia disclosed an apparatus for spatial audio encoding which can receive or determine for one or more audio signals (102), spatial audio parameters (106) on a sub band basis for providing spatial audio reproduction, the spatial audio parameters can comprise a coherence value (112) for each sub band of a plurality of subbands (202) of a frame. The apparatus then determines a significance measure for the coherence values (401) of the plurality of sub bands of the frame and uses the significance measure to determine whether to encode (403) the coherence values of the plurality of sub bands of the frame.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of ¼ of the frame length.

Abstract: An embodiment method comprises applying domain transformation processing to a time-series of signal samples, received from a sensor coupled to a dynamical system, to produce a dataset of transformed signal samples therefrom, buffering the transformed signal samples, obtaining a data buffer having transformed signal samples as entries, computing statistical parameters of the data buffer, producing a drift signal indicative of the evolution of the dynamical system as a function of the computed statistical parameters, selecting transformed signal samples buffered in the data buffer as a function of the drift signal, applying normalization processing to the buffered transformed signal samples, applying auto-encoder artificial neural network processing to a dataset of resealed signal samples, and producing a dataset of reconstructed signal samples and calculating an error of reconstruction.

Abstract: An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of ¼ of the frame length.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of ¼ of the frame length.

Abstract: An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

Abstract: Methods and apparatus provide digital media with spatial audio to a blockchain. The blockchain network executes a decentralized application (Dapp) with a user interface (UI) that enables a user to select audio for spatialization and uploading to the blockchain. The spatial audio transmits to the blockchain network to reduce processing and transmission of network data.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of 1/4 of the frame length.

Abstract: A system and method code an object-based audio signal comprising audio objects in response to audio streams with associated metadata. In the system and method, a metadata processor codes the metadata and generates information about bit-budgets for the coding of the metadata of the audio objects. An encoder codes the audio streams while a bit-budget allocator is responsive to the information about the bit-budgets for the coding of the metadata of the audio objects from the metadata processor to allocate bitrates for the coding of the audio streams by the encoder.

Abstract: Some implementations involve receiving a content stream that includes audio data, receiving at least one type of level adjustment indication relating to playback of the audio data and controlling a level of the input audio data, based on the at least one type of level adjustment indication, to produce level-adjusted audio data. Some examples involve determining, based at least in part on the type(s) of level adjustment indication, a multiband limiter configuration, applying the multiband limiter to the level-adjusted audio data, to produce multiband limited audio data and providing the multiband limited audio data to one or more audio reproduction transducers of an audio environment.

Abstract: A method of processing speech includes: providing a first set of audio data having audio features in a first bandwidth; down-sampling the first set of audio data to a second bandwidth lower than the first bandwidth; producing, by a high frequency reconstruction network (HFRN), an estimate of audio features in the first bandwidth for the first set of audio data, based on at least the down-sampled audio data; inputting, into the HFRN, a second set of audio data having audio features in the second bandwidth; producing, by the HFRN, based on a second set of audio data having audio features in the second bandwidth, an estimate of audio features in the first bandwidth for the second set of audio data; and training a speech processing system (SPS) using the estimates of audio features in the first bandwidth for the first and second sets of audio data.

Abstract: Systems and processes for operating an intelligent automated assistant are provided. An example process for determining user intent includes receiving a natural language input and detecting an event. The process further includes, determining, at a first time, based on the natural language input, a first value for a first node of a parsing structure; and determining, at a second time, based on the detected data event, a second value for a second node of the parsing structure. The process further includes in accordance with a determination that the first time and the second time are within the predetermined time: determining, using the parsing structure, the first value, and the second value, a user intent associated with the natural language input; initiating a task based on the determined intent; and providing an output indicative of the task.

Abstract: A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Abstract: An apparatus for encoding directional audio coding parameters comprising diffuseness parameters and direction parameters having a parameter calculator (100) for calculating the diffuseness parameters with a first time or frequency resolution and for calculating the direction parameters with a second time or frequency resolution; and a quantizer and encoder processor (200) for generating a quantized and encoded representation of the diffuseness parameters and the direction parameters.

Abstract: A method for spatial audio signal encoding comprising: obtaining a plurality of audio direction parameters, wherein each parameter comprises an elevation value and an azimuth value and wherein each parameter has an ordered position; deriving for each of the plurality of audio direction parameters a corresponding derived audio direction parameter (SP) comprising an elevation and an azimuth value, corresponding derived audio direction parameters (SP) being arranged in a manner determined by a spatial utilization defined by the elevation values and the azimuth values of the plurality of audio direction parameters; rotating each derived audio direction parameter (SP) by the azimuth value (?0) of an audio direction parameter in the first position of the plurality of audio direction parameters and quantizing the rotation to determine for each a corresponding quantized rotated derived audio direction parameter; changing the ordered position of an audio direction parameter to a further position coinciding with a posi

Abstract: Disclosed is a dialogue emotion correction method based on a graph neural network, including: extracting acoustic features, text features, and image features from a video file to fuse them into multi-modal features; obtaining an emotion prediction result of each sentence of a dialogue in the video file by using the multi-modal features; fusing the emotion prediction result of each sentence with interaction information between talkers in the video file to obtain interaction information fused emotion features; combining, on the basis of the interaction information fused emotion features, with context-dependence relationship in the dialogue to obtain time-series information fused emotion features; correcting, by using the time-series information fused emotion features, the emotion prediction result of each sentence that is obtained previously as to obtain a more accurate emotion recognition result.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: Embodiments are disclosed for channel-based audio (CBA) (e.g., 22.2-ch audio) to object-based audio (OBA) conversion. The conversion includes converting CBA metadata to object audio metadata (OAMD) and reordering the CBA channels based on channel shuffle information derived in accordance with channel ordering constraints of the OAMD. The OBA with reordered channels is rendered in a playback device using the OAMD or in a source device, such as a set-top box or audio/video recorder. In an embodiment, the CBA metadata includes signaling that indicates a specific OAMD representation to be used in the conversion of the metadata. In an embodiment, pre-computed OAMD is transmitted in a native audio bitstream (e.g., AAC) for transmission (e.g., over HDMI) or for rendering in a source device. In an embodiment, pre-computed OAMD is transmitted in a transport layer bitstream (e.g., ISO BMFF, MPEG4 audio bitstream) to a playback device or source device.

Abstract: In a stereo encoding method, a channel combination encoding solution of a current frame is first obtained, and then a quantized channel combination ratio factor of the current frame and an encoding index of the quantized channel combination ratio factor are obtained based on the obtained channel combination encoding solution, so that an obtained primary channel signal and secondary channel signal of the current frame meet a characteristic of the current frame.

Abstract: An encoder for encoding a parametric spectral representation (f) of auto-regressive coefficients that partially represent an audio signal. The encoder includes a low-frequency encoder configured to quantize elements of a part of the parametric spectral representation that correspond to a low-frequency part of the audio signal. It also includes a high-frequency encoder configured to encode a high-frequency part (fH) of the parametric spectral representation (f) by weighted averaging based on the quantized elements ({circumflex over (f)}L) flipped around a quantized mirroring frequency ({circumflex over (f)}m), which separates the low-frequency part from the high-frequency part, and a frequency grid determined from a frequency grid codebook in a closed-loop search procedure. Described are also a corresponding decoder, corresponding encoding/decoding methods and UEs including such an encoder/decoder.