Abstract: The simultaneous transmission and reproduction of a compressed audio signal and a linear PCM signal is satisfactorily achieved. An audio signal of a predetermined unit is sequentially transmitted via a predetermined transmission line to a reception side. The audio signal of the predetermined unit is a mixed signal of a compressed audio signal and a linear PCM signal. For example, the audio signal of the predetermined unit is an audio signal of a sub-frame unit. In this case, for example, in the audio signal of the sub-frame unit, the compressed audio signal is arranged on an upper-order bit side, and the linear PCM signal is arranged on a lower-order bit side.

Abstract: An audio system includes an audio/video receiver, a power supply/wireless audio distribution assembly connected to the audio/video receiver, speaker wire, and speakers compatible with the power supply/wireless audio distribution assembly.

Abstract: The present invention provides an audio control circuit comprising an USB interface and a processing circuit is disclosed. The USB interface is used to connect to a host device, and the processing circuit is configured to perform enumeration with the host device via the USB interface, and the processing circuit is further configured to determine if the host device operates in a BIOS stage or an operating system stage to generate a control signal according to packets of the enumeration. When the processing circuit determines that the host device operates in the BIOS stage, the processing circuit generates the control signal to enable a de-pop circuit; and when the processing circuit determines that the host device operates in the operating system stage, the processing circuit generates the control signal to disable the de-pop circuit.

Abstract: A receiver for a serial data link, including an analog front end (AFE) including a continuous-time linear equalizer (CTLE) configured to receive an input signal from a transmitter, the CTLE including a first output node; a second output node; a plurality of programmable tail current sources configured to adjust a direct current (DC) offset between the first output node and the second output node; and a calibration circuit including: a slicer configured to output a difference between a first average output voltage corresponding to the first output node and a second average output, voltage corresponding to the second output node; and a calibration counter configured to increment or decrement an offset count based on the difference, wherein the plurality of programmable tail current sources are adjusted based on a value of the offset count.

Abstract: A wireless speaker audio system configured to receive audio information wirelessly transmitted by an audio source including first and second wireless transceivers. The first wireless transceiver establishes a bidirectional secondary wireless link with the audio source for receiving and acknowledging receipt of the audio information. The first and second wireless transceivers communicate with each other via a primary wireless link. A wireless audio system including an audio source and first and second wireless transceivers. The first and second wireless transceivers communicate via a primary wireless link. The audio source communicates audio information to the first wireless transceiver via a secondary wireless link which is configured according to a standard wireless protocol. The first wireless transceiver is configured to acknowledge successful reception of audio information via the secondary wireless link. Other embodiments are also described and claimed.

Abstract: A method for demodulating a multiplexed stereophonic signal, the signal including a signal called the sum signal, a signal called the difference signal, and a pilot signal, the method including the following steps: removing the pilot frequency from the multiplexed stereophonic signal, the resulting signal being called the pilotless signal, and subtracting the sum signal from the pilotless signal.

Abstract: The method for adapting a sound converter to a reference sound converter includes the sound converter having a first linear transfer function with a first frequency response, a second linear transfer function with a second frequency response, and a trivial nonlinearity. The sound converter has a non-linear transfer function corresponding to the frequency response from combination of the first linear transfer function, the trivial nonlinearity, and the second linear transfer function. A first frequency spectrum of the reference sound converter is determined at a low input level. A second frequency spectrum of the reference sound converter is determined at a high input level. The second determined frequency spectrum is used as the second frequency response in the second linear transfer function, and the division of the first frequency spectrum by the second frequency spectrum is used as the first frequency response in the first linear transfer function.

Abstract: A method, computer system, and a computer program product for interest-based digital content caching is provided. Hints regarding content of interest associated with a user are determined and used to identify portions of streaming media that are uninteresting to the user. Then the streaming media is played for the user and a portion of the post-uninteresting content is stored in a child cache before the portion of uninterest is played.

Abstract: The present disclosure relates to methods and apparatus for audio coding. A method of encoding a portion of an audio signal comprises determining whether the portion of the audio signal is likely to contain dense transient events, and if it is determined that the portion of the audio signal is likely to contain dense transient events, quantizing the portion of the audio signal using a quantization 5 mode that applies a substantially constant signal-to-noise ratio over frequency for the portion of the audio signal. The present disclosure further relates to a method of detecting dense transient events in a portion of an audio signal.

Abstract: There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels.

Abstract: Methods and systems encoding a stereo audio signal having a left channel and a right channel are disclosed. The system includes a downmixer for generating a downmix signal and a residual signal from the stereo audio signal in selected frequency bands representing only part of a used audio frequency range of the stereo audio signal, and a decision module for selecting, in a time variant manner, either left/right perceptual encoding or mid/side perceptual encoding. The system also includes a parameter estimator for estimating stereo parameters for reconstructing a stereo image of a portion of the stereo audio signal, and a perceptual encoder for performing either left/right perceptual encoding or mid/side perceptual encoding based on the selecting to generate an encoded output signal. Finally, the system includes a bitstream generator for creating a bitstream signal comprising the encoded output signal.

Abstract: Methods for communicating audio and voice signals as well as hearable devices therefor are disclosed. A disclosed method includes creating an audio and voice Bluetooth communication link between a computing device and a hearable device. The audio and voice Bluetooth communication link includes a downlink and an uplink. The method further includes configuring the downlink with a downlink audio codec and the uplink with an uplink voice codec and receiving, by the hearable device through the downlink, a channel of a downlink audio signal from the computing device. The channel is encoded with the downlink audio codec. The method further includes transmitting, to the computing device, an uplink voice signal by the hearable device through the uplink. The uplink voice signal is encoded with the uplink voice codec.

Abstract: An audio rendering system includes a processor that combines audio input signals with personalized spatial audio transfer functions preferably including room responses. The personalized spatial audio transfer functions are selected from a database having a plurality of candidate transfer function datasets derived from in-ear microphone measurements for a plurality of individuals. Alternatively, the personalized transfer function datasets are derived from actual in-ear measurements of the listener. Foreground and background positions are designated and matched with transfer function pairs from the selected dataset for the foreground and background direction and distance. Two channels of input audio such as voice and music are processed. When a voice communication such as a phone call is accepted the music being rendered is moved from a foreground to a background channel corresponding to a background spatial audio position using the personalized transfer functions.

Abstract: An audio rendering device includes: at least one wireless communications interface configured to receive a first plurality of audio packets from a wireless communication device, the first plurality of audio packets including frames of audio data for a first audio channel from a set of one or more audio channels; and a processing unit; wherein the audio rendering device is configured to release, at respective first buffer release times, the frames of the audio data for the first audio channel from a first buffer for rendering by the processing unit; wherein the audio rendering device is configured to receive one or more messages from an other audio rendering device; and wherein the audio rendering device is configured to synchronize the first buffer release times with second buffer release times associated with the other audio rendering device based on the received one or more messages.

Abstract: A system, method and program product for obfuscating audible messages in a listening space A system is provided that includes an orchestrator having: an invocation detection system that triggers an obfuscation event; a system for selecting injector nodes in the listening space for the obfuscation event; and a key management system that distributes keys, derived from a natural interface key, to the injector nodes to cause the injector nodes to inject sounds into the listening space to obfuscate an audible message broadcast by a source node for a target node; and a machine learning system that calculates the natural interface key based on interactions captured from the source node in the listening space.

Abstract: An input transducer (11, 12) of a hearing aid (10) converts audio from the environment into an electric input signal. A signal processor (13) amplifies and conditions the electric input signal according to a preset set of audio processing parameters, and an output transducer (14) reproduces output audio. The hearing aid (10) is connected with a personal communication device (20) via a short range radio communication link (15, 29). The personal communication device (20) identifies music included in the background noise. The personal communication device (20) retrieves a music stream based on the identified music, and streams the music stream to the hearing aid (10). The signal processor (13) of the hearing aid (10) combines the input signal from the input transducer (11, 12) and music signal in order to reduce the presence of the music included in the background noise from the audio signal delivered by the output transducer (14).

Abstract: Methods and systems encoding a stereo audio signal having a left channel and a right channel are disclosed. The system includes a downmixer for generating a downmix signal and a residual signal from the stereo audio signal in selected frequency bands representing only part of a used audio frequency range of the stereo audio signal, and a decision module for selecting, in a time variant manner, either left/right perceptual encoding or mid/side perceptual encoding. The system also includes a parameter estimator for estimating stereo parameters for reconstructing a stereo image of a portion of the stereo audio signal, and a perceptual encoder for performing either left/right perceptual encoding or mid/side perceptual encoding based on the selecting to generate an encoded output signal. Finally, the system includes a bitstream generator for creating a bitstream signal comprising the encoded output signal.

Abstract: Some embodiments include a system for delivery of binaural audio to a consumer including a computing device with a processor, a non-transitory computer readable medium having stored instructions that when executed by the computing device, cause the computing device to perform operations for delivery of the binaural audio to the consumer through the consumer device. The operations can enable a delivery server to associate the consumer device with a binaural audio file associated with the event attended by the consumer. The operations can enable the consumer device to retrieve and download a binaural audio file from the delivery server, and to receive synchronization information associated with the event and the binaural audio file. The operations can also include processing the binaural audio file and delivering audio through an audio emitting device coupled to the consumer device, where the delivered audio is synchronized with the event.

Abstract: A source device comprising a memory and a processor may be configured to perform techniques described in this disclosure. The memory may store at least a portion of the audio data. The processor may obtain, from a compressed version of the audio data, a symbol, and obtain a plurality of intervals, each having a same bit length. The processor may obtain a portion of the symbol within the bit length and an excess portion of the symbol over the bit length, and specify, in a first interval, the portion of the symbol. The processor may also specify, in a second interval, the excess portion of the symbol, and apply, to the first interval and the second interval, error resiliency. The processor may specify, in a bitstream representative of the compressed version of the audio data, the first error resilient interval and the second error resilient interval.

Abstract: A headset, which may be configured to process audio for a player participating in an online multiplayer game, may support auto-pairing with audio basestations. In this regard, an audio basestation may be utilized in communicating game and chat audio content from an associated game console to a paired headset and in communicating microphone audio from the paired headset to the console. The auto-pairing may comprise automatically exchanging signals with an audio basestation, in response to the headset coming within a determined distance of the audio basestation; and pairing with the audio basestation based on processing of the exchanged signals. The pairing may result in establishing one or more wireless links for game and chat audio to be communicated to the headset and microphone audio to be communicated from the headset. The determined distance may correspond to physical contact (e.g., tapping) between the headset and the basestation.

Abstract: An apparatus for encoding an audio or image signal, includes: a controllable windower for windowing the audio or image signal to provide the sequence of blocks of windowed samples; a converter for converting the sequence of blocks of windowed samples into a spectral representation including a sequence of frames of spectral values; a transient location detector for identifying a location of a transient within a transient look-ahead region of a frame; and a controller for controlling the controllable windower to apply a specific window having a specified overlap length to the audio or image signal in response to an identified location of the transient, wherein the controller is configured to select the specific window from a group of at least three windows, wherein the specific window is selected based on the transient location.

Abstract: Methods and apparatus to perform windowed sliding transforms are disclosed.

Abstract: An audio encoding apparatus and method that encodes hybrid contents including an object sound, a background sound, and metadata, and an audio decoding apparatus and method that decodes the encoded hybrid contents are provided. The audio encoding apparatus may include a mixing unit to generate an intermediate channel signal by mixing a background sound and an object sound, a matrix information encoding unit to encode matrix information used for the mixing, an audio encoding unit to encode the intermediate channel signal, and a metadata encoding unit to encode metadata including control information of the object sound.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: A method of processing an audio signal includes receiving an audio bitstream encoded via MPEG Surround 212 (MPS212); generating an internal channel (IC) signal for a single channel pair element (CPE), based on the received audio bitstream, equalization (EQ) values for MPS212 output channels defined in a format converter, and gain values for the MPS212 output channels; and generating stereo output channels, based on the generated IC signal.

Abstract: A portable device uses a microphone to listen to ambient audio, ascertains a corresponding identifier, and uses the identifier to enable one or more further functions. One of these can be internet search. Such functionality can also be based on digital data—without requiring a microphone-equipped device.

Abstract: Apparatus including: an audio capture application configured to determine separate microphones from a plurality of microphones and identify a sound source direction of at least one audio source within an audio scene by analyzing respective two or more audio signals from the separate microphones, wherein the audio capture application is further configured to adaptively select, from the plurality of microphones, two or more respective audio signals based on the determined direction and furthermore configured to select, from the two or more respective audio signals, a reference audio signal also based on the determined direction; and a signal generator configured to generate a mid signal representing the at least one audio source based on a combination of the selected two or more respective audio signals and with reference to the reference audio signal.

Abstract: An audio providing apparatus and method are provided. The audio providing apparatus includes: an object renderer configured to render an object audio signal based on geometric information regarding the object audio signal; a channel renderer configured to render an audio signal having a first channel number into an audio signal having a second channel number; and a mixer configured to mix the rendered object audio signal with the audio signal having the second channel number.

Abstract: Some embodiments include a system for delivery of binaural audio to a consumer including a computing device with a processor, a non-transitory computer readable medium having stored instructions that when executed by the computing device, cause the computing device to perform operations for delivery of the binaural audio to the consumer through the consumer device. The operations can enable a delivery server to associate the consumer device with a binaural audio file associated with the event attended by the consumer. The operations can enable the consumer device to retrieve and download a binaural audio file from the delivery server, and to receive synchronization information associated with the event and the binaural audio file. The operations can also include processing the binaural audio file and delivering audio through an audio emitting device coupled to the consumer device, where the delivered audio is synchronized with the event.

Abstract: Provided is a mobile device including: an input reception unit for receiving at least one input through a user interface; a control information determination unit for determining control information corresponding to an operation of an external device based on the at least one input, wherein the control information includes a first symbol and a second symbol; a sound wave data generation unit for generating sound wave data corresponding to the control information; and an output unit for outputting a sound wave corresponding to the sound wave data generated by a sound wave generation apparatus. In this case, the sound wave data generation unit determines a first frequency corresponding to the first symbol and a second frequency corresponding to the second symbol, and generates the sound wave data based on the first frequency and the second frequency, and the sound wave data includes a portion corresponding to the first frequency and a portion corresponding to the second frequency.

Abstract: Methods of data encoding using trees formed with logic gates are described which lead to spatial compression of image data. Data encoding is achieved using a five-level wavelet transform, such as the Haar or the 2/10 transform. A dual transform engine is used, the first and engine being used for the first part of the first-level transform, the second part of the first-level transform and the subsequent-level transforms being performed by the second transform engine within a time interval which is less than or equal to the time taken by the first transform engine to effect the part-transform. Each bit plane of the resulting coefficients is then encoded by forming a tree structure from the bits and OR logical combinations thereof. Redundant data are removed from the resulting tree structure, and further data can be removed by using a predetermined compression profile.

Abstract: An audio encoding apparatus and method that encodes hybrid contents including an object sound, a background sound, and metadata, and an audio decoding apparatus and method that decodes the encoded hybrid contents are provided. The audio encoding apparatus may include a mixing unit to generate an intermediate channel signal by mixing a background sound and an object sound, a matrix information encoding unit to encode matrix information used for the mixing, an audio encoding unit to encode the intermediate channel signal, and a metadata encoding unit to encode metadata including control information of the object sound.

Abstract: A signal processing device includes a selector configured to select one of a plurality of methods, in accordance with which a signal generation process is performed for generating a transfer signal in which additional information is added to an audio signal; a signal processor configured to execute the signal generation process of adding the additional information to the audio signal in accordance with the method selected by the selector; and a transferrer configured to transfer to a reproduction device the transfer signal generated by the signal processor.

Abstract: A kinesthetic speaker usage system provides a speaker connected to a housing about a cavity into which the sound from the speaker is directed. The housing has an exterior surface in tactile communication with a retainer which can direct vibration directly or directly to a user, such as a person. Many embodiments utilize a pipe for the housing, such as a flexible pipe.

Abstract: A method for extending the range of isochronous USB data transfer, by providing a local extender and a remote extender, coupled respectively to the USB host and the USB device, and coupled to each other via a non-USB cable. After receiving a first In-token from the host, the local extender transmits a first data packet to the host, which contains a data payload having a non-zero length; the data is either generated by the first extender, or previously received from the USB device and stored in a buffer. The local extender forwards the first In-token to the USB device via the remote extender; in response, the USB device generates a second data packet containing requested data and transmits it to the local extender via the remote extender. The local extender stores the second data packet and transmits it to the host in response to a second In-token from the host.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: An example apparatus to generate media monitoring information includes a processor to collect a first data record corresponding to a first media presentation session, and collect a second data record corresponding to a second media presentation. The example apparatus also includes a memory in circuit with the processor, the memory to: store a first end timestamp in the first data record indicative of a first time at which the first media presentation session ended; store a second end timestamp in the second data record indicative of a second time at which the second media presentation session ended; and store the first end timestamp in the second data record.

Abstract: Example embodiments disclosed herein relate to an evaluation of perceptual delay impact on a conversation. A method of evaluating perceptual delay impact on a conversation is disclosed. The method includes detecting a conversational event between a first participant and a second participant, the conversational event including a plurality of speech segments; in response to the conversational event being detected, obtaining at least one conversational parameter for the first and second participants, the at least one conversational parameter indicating a relationship between the plurality of speech segments in the conversational event; and determining a quality of the conversation according to a model based on the at least one conversational parameter, the model characterizing the quality with the at least one conversational parameter. A method of obtaining a model for evaluating perceptual delay impact on a conversation is also disclosed. Corresponding systems and computer program products are also disclosed.

Abstract: A method for determining a connection order of nodes on a powered audio bus by having the nodes draw power from the powered audio bus, measure the power consumed downstream, determine who is the last node on the bus, and transmit an identifier of the last node to the bus driver.

Abstract: An audio encoding apparatus and method that encodes hybrid contents including an object sound, a background sound, and metadata, and an audio decoding apparatus and method that decodes the encoded hybrid contents are provided. The audio encoding apparatus may include a mixing unit to generate an intermediate channel signal by mixing a background sound and an object sound, a matrix information encoding unit to encode matrix information used for the mixing, an audio encoding unit to encode the intermediate channel signal, and a metadata encoding unit to encode metadata including control information of the object sound.

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.