Abstract: Provided is a signal separation system including a rendering unit which receives a first and a second input signal and positions the first input signal according to rendering information.

Abstract: The invention relates to the field of methods and devices for analyzing of psychophysiological reactions of a person to verbal tests. The invented device (1) for carrying out the inventive method for determining a stress state comprises a receiving unit for receiving a voice signal, for example, from a microphone (5); a processing unit for determining a level of the stress state according to one dimensionless parameter based on spectral characteristics such as a base frequency, intensity, median and width of a spectrum; and a display unit for displaying a stress state, consisting, for example, a light-emitting device (6) or a device for generating vibrations (7), wherein a length of light wave or vibration frequency depends on the level of a stress state.

Abstract: A medicament delivery device includes a housing, a medicament container disposed within the housing, an activation mechanism, a cover and an electronic circuit system. The activation mechanism includes an energy storage member configured to produce a force to deliver the dose of a medicament and/or vaccine. The cover is configured to receive at least a portion of the housing. The electronic circuit system is coupled to the housing such that a protrusion of the cover electrically isolates a battery from a portion of the electronic circuit system when the portion of the housing is received by the cover. The electronic circuit system is configured to be electrically coupled to the battery and to produce a recorded speech output when the portion of the housing is at least partially removed from the cover. The electronic circuit system configured to produce a signal when the activation mechanism is actuated.

Abstract: Computer implemented speech processing is disclosed. First and second voice segments are extracted from first and second microphone signals originating from first and second microphones. The first and second voice segments correspond to a voice sound originating from a common source. An estimated source location is generated based on a relative energy of the first and second voice segments and/or a correlation of the first and second voice segments. A determination whether the voice segment is desired or undesired may be made based on the estimated source location.

Abstract: A pitch detection method and apparatus are disclosed. The method includes: performing pitch detection on an input signal in a signal domain, and obtaining a candidate pitch; performing linear prediction (LP) on the input signal, and obtaining an LP residual signal; setting a candidate pitch range that includes the candidate pitch; searching the candidate pitch range for the LP residual signal, and obtaining a selected pitch.

Abstract: Method and processing system for measuring the intelligibility of a degraded output signal (Y(t)) from an audio transmission system (10) in response to a reference input signal (X(t)). A measurement device (11) is arranged for outputting a measure (I) for the speech intelligibility of the output signal (Y(t)). The measurement device (11) executes processing of the input signal (X(t)) and output signal (Y(t)) to obtain a disturbance density function (D(f)n). The disturbance density function (D(f)n) is corrected by multiplying it with a correction function for each frame derived from a correlation calculation of the compensated pitch power densities (PPX?(f)n) associated with the input signal (X(t)) of a present frame (n) and an independent previous frame (n?2). The corrected disturbance density function (D?(f)n) is aggregated over frequency and time to obtain a measure (I) for the speech intelligibility of the output signal (Y(t)).

Abstract: An information interactive kit adapted to an information interactive device is provided. The information interactive device is for generating an interactive event to express the content of at least one predetermined information accordingly. The information interactive kit includes a storing unit and a communicating interface. The storing unit is for storing a theme database. The communicating interface, when connected to the information interactive device, transmits the content of the theme database to the information interactive device, so that the information interactive device determines how to change the generated interactive event according to the content of the theme database.

Abstract: It is an object to provide a mounted-on-a-car instrument configured to prevent duplicate reproduction of the same contents. A mounted-on-a-car instrument (10) is provided with a DSRC unit (3) that receives contents information from a center device (30) through a roadside apparatus (20) and a car navigation unit (1) that stores guide information on driving of a car. The mounted-on-a-car instrument (10) is further provided with an audio output unit (5) that outputs the contents information and the guide information, a memory unit (3c) that stores information of a position where the roadside apparatus (20) is arranged, and a control unit (4) that makes the audio output unit (5) output either the contents information or the guide information based on the position information stored in the memory unit (3c) in the case where the contents information and the guide information meet an output condition.

Abstract: The invention discloses a command control system including a light emitting unit, an image capturing unit, a storage unit, and a processing unit. The processing unit is coupled with the image capture unit and the storage unit. The light emitting unit emits light to form an illumination area. The image capture unit captures a plurality of pieces of image information in the illumination area. The storage unit stores different commands corresponding to the image information. The processing unit performs functions according to the commands corresponding to the image information.

Abstract: A digital speech enabled middleware module is disclosed that facilitates interaction between a large number of client devices and network-based automatic speech recognition (ASR) resources. The module buffers feature vectors associated with speech received from the client devices when the number of client devices is greater than the available ASR resources. When an ASR decoder becomes available, the module transmits the feature vectors to the ASR decoder and a recognition result is returned.

Abstract: A voice conversion training system, voice conversion system, voice conversion client-server system, and program that realize voice conversion to be performed with low load of training are provided. In a server 10, an intermediate conversion function generation unit 101 generates an intermediate conversion function F, and a target conversion function generation unit 102 generates a target conversion function G. In a mobile terminal 20, an intermediate voice conversion unit 211 uses the conversion function F to generate speech of an intermediate speaker from speech of a source speaker, and a target voice conversion unit 212 uses the conversion function G to convert speech of the intermediate speaker speech generated by the intermediate voice conversion unit 211 to speech of a target speaker.

Abstract: Architecture that employs a combination of in-band signaling (e.g., DTMF) with speech recognition to deliver usability improvements. The in-band signaling allows the user to indicate to the system when a barge-in operation is occurring and/or when to start listening to subsequent speech input and optionally, when to stop listening for further speech input. The in-band signaling can be utilized during a telephone call and using wireline and wireless telephones. Moreover, the architecture can be incorporated at the platform level requiring little, if any, application changes to support the new mode of operation.

Abstract: Provided are techniques which offer an advantage of reduced time and reduced storage capacity required to calculate the feature value of AAC-format song data. A feature extraction unit includes: an MDCT coefficient extraction unit which extracts MDCT coefficients from AAC-format song data; a classification unit which locates the MDCT coefficients thus extracted by the MDCT coefficient extraction unit on Mel frequency regions so as to uniformly classify the MDCT coefficients into classes, the number of which is the same as that of a predetermined number of Mel filter banks; an integrating unit which extracts the MDCT coefficients classified by the classification unit by applying a predetermined window function, and integrates the MDCT coefficients thus extracted, in increments of the Mel filter banks; and a feature calculation unit which calculates the feature value by performing logarithmic cosine conversion of the integrated results obtained by the integrating unit.

Abstract: A one-step correction mechanism for voice interaction is provided. Correction of a previous state is enabled simultaneously with recognition in a current or subsequent state. An application is decomposed into a set of tasks. Each task is associated with the collection of one piece of information. Each task may be in a different state. At any point during the interaction, while a task/state pair is active, the dialog manager may enable multiple other task/state pairs to be active in latent fashion. The application developer may then use those facilities or resources to the active task/state and the latent task/state pairs depending on contextual condition of the interaction state of the application.

Abstract: A method of controlling a mobile terminal, and which includes displaying, via a display on the mobile terminal, a captured or a preview image in a first display portion, displaying, via the display, the same captured or a preview image in a second display portion, zooming, via a controller on the mobile terminal, the captured or preview image displayed in the first display portion, and displaying, via the display, a zoom guide on the image displayed in the second display portion that identifies a zoomed portion of the image displayed in the first display portion.

Abstract: A method of encoding one or more parent blocks of values, the number of values being the length of each block, the method comprising for each parent block: (a) determining a first sum of values in the parent block; (b) splitting the parent block into smaller subblocks; (c) for at least one of the subblocks, determining a second sum of the values in the subblock, selecting a likelihood table from the plurality of likelihood tables based on said first sum of values in the parent block and encoding the second sum using the likelihood table; (d) designating each subblock a parent block; (e) carrying out steps (a), (b), (c) and (d) until at least one parent block reaches a predetermined condition.

Abstract: A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.

Abstract: A set of peaks in a reconstructed audio vector ? of a received audio signal is detected and a scaling mask ?(?) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output.

Abstract: An audio encoding apparatus that encodes audio signals of a plurality of channels, includes an adaptive bit allocation control unit that adaptively controls a number of encoding bits assigned to the audio signal of each channel in accordance with perceptual entropy of the audio signal of each of the channels, a fixed bit allocation control unit that fixedly controls the number of encoding bits assigned to the audio signal of each of the channels in predetermined allocations, and a channel encoding unit that encodes the audio signal of each of the channels based on the number of adaptive allocation bits assigned by the adaptive bit allocation control unit and the number of fixed allocation bits assigned by the fixed bit allocation control unit.

Abstract: The present invention has as its object providing a terminal capable of accurately searching for an apparatus installed near a user. The terminal connected to a network apparatus through a network includes a display section which performs a displaying operation and position searching means which, on receiving an instruction to make a search for the network apparatus, transmits a signal indicating specific content and then, on receiving a response signal from the network apparatus that receives the signal through the network, causes the display section to display information that shows that the network apparatus transmitting the response signal has received the signal.

Abstract: Many embodiments may comprise logic such as hardware and/or code to implement user interface for traversal of long sorted lists, via audible mapping of the lists, using sensor based gesture recognition, audio and tactile feedback and button selection while on the go. In several embodiments, such user interface modalities are physically small in size, enabling a user to be truly mobile by reducing the cognitive load required to operate the device. For some embodiments, the user interface may be divided across multiple worn devices, such as a mobile device, watch, earpiece, and ring. Rotation of the watch may be translated into navigation instructions, allowing the user to traverse the list while the user receives audio feedback via the earpiece to describe items in the list as well as audio feedback regarding the navigation state. Many embodiments offer the user a simple user interface to traverse the list without visual feedback.

Abstract: An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.

Abstract: A decoding method for MP3 bit streams, which replaces a buffer required in the decoding process by manipulating the order of data decoding. The decoding method includes reading the head and side information of the current frame, and calculating a main data's start address of the current frame. While decoding the main data, the head and side information of subsequent frames are skipped if the reading of the main data is not yet completed. The start address of the next frame is calculated and directly accessed after finished reading the main data of the current frame. An optimum method for accessing frequency lines utilizes the characteristics of the MP3 frequency line, instead of inserting a plurality of zeros in the rzero zone containing successive zeros, the initial boundary address of the rzero zone is memorized.

Abstract: We disclose useful components of a method and system that allow identification of music from the song or sound using only the sound of the audio being played. A system built using the method and device components disclosed processes inputs sent from a mobile phone over a telephone or data connection, though inputs might be sent through any variety of computers, communications equipment, or consumer audio devices over any of their associated audio or data networks.

Abstract: The present invention applies spectral flatness characteristic values to simplify psychoacoustic analysis of a sound signal. If the sound signal comprises a plurality of frames, the present invention calculates the energy of the sound signal in a frequency domain, calculates a plurality of spectral flatness, and decides to use a short-block or a long-block Modified Discrete Cosine Transform accordingly. If the sound signal comprises left and right channel signals, the present invention performs psychoacoustic analysis on the sound signal to count energy of the left and right channel signals in a frequency domain, counts spectral flatness of the left and right channel signals, and decides to use middle/side transform or left and right channel encoding to transform the left and right channel signals accordingly.

Abstract: An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.

Abstract: In an information processing apparatus, the information of a webpage acquired by a page information reception unit is analyzed for a tag and the like by a page information analysis unit, and a character string is extracted under an extraction condition that is set in advance. Multiple character string groups are extracted so that multiple character strings are concurrently perceived in an aural manner. The extracted character strings are converted into respective audio signals by a text-to-sound conversion unit. The multiple audio signals thus generated are processed and synthesized by an audio processing unit based on the allocation pattern set by a frequency band allocation unit, the localization set by a localization allocation unit, and the difference in time at which the audio signals are output set by a time allocation unit. The output unit outputs the synthesized sounds.

Abstract: A device for providing dynamic speech processing services during variable network connectivity with a network server includes a connection determiner that determines the level of network connectivity of the client device and the network server; and a simplified speech processor that processes speech data and is initiated based on the determination from the connection determiner that the network connectivity is impaired or unavailable. The devices further includes a speech data storage that stores processed speech data from the simplified speech processor; and a transition unit that determines when to transmit the stored speech data and connects with the network server, based on the determination of the connection determiner.

Abstract: An interface. unit (1) for display of alphanumeric-data packets, includes an input connector and an output connector (3). The input connector (2) is connectable by a first connection to a first apparatus (11, 11?) for reproduction of an audio signal to which alphanumeric data are associated. The output connector (3) is connectable by a second connection to a second reproduction apparatus (12) having a display (13) and a telematic decoder of the data packets. The interface unit (1) further includes a treatment of the audio signal connected to the input connector (2). The treatment of the audio signal carries out a preamplification and a filtering in frequency of the audio signal. A processor (7, 8) is connected to the input connector (2). The processor (7, 8) extracts and carries out a telematic encoding of alphanumeric data. A signal-mixer unit (9) is connected to the treatment of the audio signal and to, the processor.

Abstract: A mobile terminal includes a projector module configured to project an image onto an external surface to display the image on the external surface and a controller configured to adjust at least one of an area or a position of the displayed image.

Abstract: Method for controlling user access to a service available in a data network and/or to information stored in a user database, in order to protect stored user data from unauthorized access, such that the method comprises the following: input of a user's speech sample to a user data terminal, processing of the user's speech sample in order to obtain a prepared speech sample as well as a current voice profile of the user, comparison of the current voice profile with an initial voice profile stored in an authorization database, and output of an access-control signal to either permit or refuse access, taking into account the result of the comparison step, such that the comparison step includes a quantitative similarity evaluation of the current and the stored voice profiles as well as a threshold-value discrimination of a similarity measure thereby derived, and an access-control signal that initiates permission of access is generated only if a prespecified similarity measure is not exceeded.

Abstract: Disclosed are an alarm method and system based on voice events, and a building method on behavior trajectory thereof The system comprises a signal sensor, a voice-event detector and notice and alarm element. In the method, voice signals are captured from a remote unit in an environment. The captured voice signals are classified into at least a voice event. As such, an emergent-event notice is automatically transmitted out if one of predefined emergent events is detected. In the building method on behavior trajectory, messages on voice events are continuously recorded. When the number of the recorded voice events reaches a threshold, a behavior trajectory is constructed, in which a behavior consists of two or more voice events or a single voice event.

Abstract: There is provided a communication system having at least one headset (100) for recording speech signals with a directional information signalling device (1) for detecting directional information. The communication system further has a communication unit (200) which has a direction evaluation unit (2) for evaluating the items of directional information produced by the directional information signalling device. The communication unit further has a channel selection unit (3) for receiving the recorded speech signals (c) and for selecting an output channel (OUT1-OUTi) in dependence on the items of directional information detected and evaluated by the direction evaluation unit (2).

Abstract: Provided are an apparatus and method for coding and decoding a multi object audio signal with multi channel. The apparatus includes: a multi channel encoding means for down-mixing an audio signal including a plurality of channels, generating a spatial cue for the audio signal including the plurality of channels, and generating first rendering information including the generated spatial cue; and a multi object encoding unit for down-mixing an audio signal including a plurality of objects, which includes the down-mixed signal from the multi channel encoding unit, generating a spatial cue for the audio signal including the plurality of objects, and generating second rendering information including the generated spatial cue, wherein the multichannel encoding unit generates a spatial cue for the audio signal including the plurality of objects regardless of a Coder-DECoder (CODEC) scheme the limits the multi channel encoding unit.

Abstract: An audio encoding method and a corresponding decoding method are provided according to the present invention. Accordingly, the pre-echo effect of the audio transient signal is eliminated and the distortion of the transient signal is mitigated. The technical solution includes performing time-domain processing on an input audio transient signal; dividing sampling points x1, x2, . . .

Abstract: A long term signal level of an audio signal is computed at a local device, wherein the audio signal was transmitted from a remote device and received by the local device. An automatic gain control (AGC) gain is computed at the local device based on the long term signal level. A noise factor indicative of a level of ambient noise at the local device is computed at the local device. A dynamic range compression (DRC) gain is computed at the local device based on the noise factor. An amplitude of the audio signal is adjusted at the local device based on the AGC gain and the DRC gain.

Abstract: The invention provides a dynamic range control module installed in a speech processing apparatus. In one embodiment, the dynamic range control module comprises a buffer, a voice activity detector, a peak calculation module, and an amplitude adjusting module. The buffer buffers a speech signal to obtain a delayed speech signal. The voice activity detector determines a syllable from the delayed speech signal. The peak calculation module calculates peak amplitude of the syllable. The amplitude adjusting module determines an attenuation factor corresponding to the syllable according to the peak amplitude in the syllable, and adjusts amplitude of the whole syllable with the same gain according to the attenuation factor to obtain an adjusted speech signal.

Abstract: An information retrieval system comprises: a speech input unit for inputting speech; an information storage unit for storing information with which speech information, of a length with which text degree of similarity is computable, is associated as a retrieval tag; an information selection unit for comparing a feature of each spoken content item extracted from each item of said speech information, with a feature of spoken content extracted from said input speech, to select information with which speech information similar to input speech is associated. The system further comprises an output unit for outputting information selected by said information selection unit, as information associated with input speech.

Abstract: Universal Video Computer Vision Input Virtual Space Mouse-Keyboard Control Panel Robot has computer system use video vision camera sensors, logical vision sensor programming as trainable computer vision seeing objects movements X, Y, Z dimensions' definitions to recognize users commands by their Hands gestures and/or enhance symbols, colors objects combination actions to virtually input data, and commands to operate computer, and machines. The robot has automatically calibrated working space into Space Mouse Zone, Space Keyboard zone, and Hand-Sign Languages Zone between user and itself. The robot automatically translate the receiving coordination users' hand gesture actions combinations on the customizable puzzle-cell positions of working space and mapping to its software mapping lists for each of the puzzle-cell position definition and calibrate these user hand and/or body gestures' virtual space actions into entering data and commands to computer meaningful computer, machine, home appliances operations.

Abstract: An apparatus and method of creating a face character which corresponds to a voice of a user is provided. To create various facial expressions with fewer key models, a face character is divided in a plurality of areas and a voice sample is parameterized corresponding to pronunciation and emotion. If the user's voice is input, a face character image corresponding to divided face areas is synthesized using key models and data about parameters corresponding to the voice sample to synthesize an overall face character image using the synthesized face character image corresponding to the divided face areas.

Abstract: SYSTEM FOR VOICE-BASE INTERACTION ON WEB PAGES, of type that permits the incorporation of voice-handling functions on a Web page, in which from a Terminal (1) a Web page (3) of a Web site that is structured under the DOM (Domain Object Model), or any of its extensions, and a networked Voice Service Server (5), by means of a downloadable module (6) for further incorporation in a Web browser, the system including the operating procedures for enabling said module to act as a transparent gateway in a dialogue between said Voice Service Server (5) and said Web page (3), said Web browser permitting to handle said Voice Services of said Server (5) through script functions incorporated in said Web page (3).

Abstract: An apparatus and method for deciding an adaptive noise level for bandwidth extension are provided. The apparatus includes a noise level decider for deciding a high-band noise level for bandwidth extension according to tonality of an input signal, a pitch frequency analyzer for detecting a pitch frequency of the input signal and analyzing correlation between the detected pitch frequency and a frequency channel, and a noise level controller for adaptively controlling the decided high-band noise level based on the analyzed correlation of the pitch frequency and the frequency channel.

Abstract: A system for sharing and processing traffic information includes a number of traffic information computer systems within individual vehicles or devices and a virtual traffic information server on a mobile network. The traffic information computer systems are each connected through a peer-to-peer radio, cellular, Wi-Fi, or other similar types of communications network, and which each operate with a database for displaying road maps, with a database storing average speed data for directions of travel along roadways, and with a location sensor used to determine the location and average speed of the vehicle or device, which are transmitted to other vehicles. The virtual server returns average speed data for road segments, which is displayed on the road maps. The system includes, sharing average speed data calculated as well average speed data received from the plurality of vehicles to other vehicles, thereby enhancing the real-time communication of traffic data.

Abstract: An exemplary system and method are directed at receiving an audio signal and process the audio signal into a remapped audio signal based on a plot profile. The plot profile may include at least one of an identified range of audio frequencies. The processing may comprise retrieving an identified range of audio frequencies from the plot profile; determining a range of impaired audio frequencies in the audio signal based on the identified range of audio frequencies; shifting the frequency of at least a portion of the impaired audio frequencies to outside of the identified range; and continuing to retrieve identified ranges of audio frequencies from the plot profile. The shifting of the impaired audio frequencies of the audio signal may be performed until no further identified ranges of audio frequencies are available for consideration.

Abstract: Synthesizing a set of digital speech samples corresponding to a selected voicing state includes dividing speech model parameters into frames, with a frame of speech model parameters including pitch information, voicing information determining the voicing state in one or more frequency regions, and spectral information. First and second digital filters are computed using, respectively, first and second frames of speech model parameters, with the frequency responses of the digital filters corresponding to the spectral information in frequency regions for which the voicing state equals the selected voicing state. A set of pulse locations are determined, and sets of first and second signal samples are produced using the pulse locations and, respectively, the first and second digital filters. Finally, the sets of first and second signal samples are combined to produce a set of digital speech samples corresponding to the selected voicing state.

Abstract: In order to reduce the problem that sound cuts out due to overflow of audio output data caused by a delay in shifting to an audio reproducing process, an audio coding and reproducing apparatus includes: an input data storage unit in which PCM audio signals are stored; an output data storage unit in which data to be outputted is stored; an audio output unit configured to output the audio data; an audio coding unit configured to code the audio data; a coded data storage unit configured to store the audio data coded by the audio coding unit; a bitrate control unit configured to control a bitrate at which the coded data is outputted, based on an amount of free space of the output data storage unit; and a data memory unit configured to retain the coded data.

Abstract: An apparatus for processing an audio input signal is provided and includes an audio processing circuit and an audio compressing circuit. The audio processing circuit receives the audio input signal, and enhances a first frequency part of the audio input signal to output a bass-enhancement signal. The audio compressing circuit is coupled to the audio processing circuit, and reduces a gain of a second frequency part of the bass-enhancement signal to output an audio output signal.

Abstract: By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and/or cancellation protocols at the appropriate time via analog and/or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

Abstract: A logarithmic frequency spectrum within a predetermined time range is calculated from a speech signal. The logarithmic frequency spectrum has a frequency element at equal intervals along a logarithmic frequency axis. A logarithmic frequency spectrogram is calculated by connecting a plurality of logarithmic frequency spectrums. A value of the frequency element along a straight line on the logarithmic frequency spectrogram is voted onto a Hough plane. The Hough plane has a voted value in correspondence with a gradient of the straight line. The voted value above a threshold and the gradient corresponding to the voted value are extracted from the Hough plane. A fundamental frequency change is calculated using the voted value and the gradient extracted.

Abstract: The invention relates to the field of audio encoding. In particular, it relates to the transcoding of audio metadata between different audio coding schemes. It describes a method and a system for transcoding audio gain metadata related to dynamic range control from first gain metadata of a first audio coding scheme to second gain metadata of a second audio coding scheme, wherein the first and second audio coding schemes use coding blocks and wherein each coding block has at least one associated gain value. The method and the system select a gain value of the second gain metadata based on the gain values of the first gain metadata such that within a time interval around the time instance associated with the gain value of the second gain metadata, the minimum gain value of the first gain metadata is selected.