Abstract: Audio watermarking is the process of embedding watermark information items into an audio signal in an in-audible manner. In a first embodiment, in case the original audio signal has parts of low signal energy, an alternative signal having a level or strength given by the psycho-acoustic model is combined with the original audio signal. The combined signal is watermarked with watermark data to be embedded. In a second embodiment, in case the original audio signal has parts of low signal energy, an alternative signal having a level or strength given by the psycho-acoustic model is watermarked with watermark data to be embedded, and the audio signal is watermarked with the watermark data to be embedded. The watermarked alternative signal is combined with the watermarked audio signal.

Abstract: Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.

Abstract: A sound generation system for a vehicle includes a sound generator for operating a speaker of the vehicle to produce an audible sound. A controller detects a vehicle operation event and controls the sound generator to generate an audible indication of the event, an association of the audible indication with the event being programmable by a user.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: A communication system particularly for managing voice, video and data services between the station of an operator and the station of a user, the system including at least one device controlled by the operator to receive telephone calls forwarded by a call routing center and at least one unit that is controlled by the user and provided with elements for entering and displaying information and generating telephone calls. The device further includes elements for disabling the audio component of a telephone call generated by the at least one unit and elements for establishing with the unit a video call associated with the telephone call.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises selecting a long-term filter based on an echo tail length detection or an echo reverberation time detection of an microphone input signal; a reference signal is pre-processed with the selected long-term filter; the pre-processed reference signal is used to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo and/or acoustic echo tail; the replica signal of acoustic echo and/or acoustic echo tail is subtracted from a microphone input signal to suppress the acoustic echo and/or acoustic echo tail in the microphone input signal. The echo tail length or the echo reverberation time is detected by analyzing and comparing the microphone input signal and a received signal which is sent to a speaker.

Abstract: A Wireless Caption Communication Service (“WCCS”) System includes a relay center, a wireless caption communication device, and a wireless captioning service server. The wireless caption communication device has a voice collecting device and a wireless caption communication terminal. Text entered by a first user is transmitted to the wireless captioning service server and converted into a speech. Then, the speech is transmitted to the voice collecting device and the sound of the speech comes out of a speaker of the voice collecting device so that a second user can hear the speech. The voice of the second user is transmitted to the wireless captioning service server and then to the relay center. The voice is converted into a caption data and transmitted to the wireless caption communication device, and the caption data is displayed on the wireless caption communication terminal so that the first user can read the caption data.

Abstract: Streaming audio is received. The streaming audio includes a frame having plurality of samples. An energy estimate is obtained for the plurality of samples. The energy estimate is compared to at least one threshold. In addition, a band pass estimate of the signal is determined. An energy estimate is obtained for the band-passed plurality of samples. The two energy estimates are compared to at least one threshold each. Based upon the comparison operation, a determination is made as to whether speech is detected.

Abstract: A method for processing speech, comprising semantically parsing a received natural language speech input with respect to a plurality of predetermined command grammars in an automated speech processing system; determining if the parsed speech input unambiguously corresponds to a command and is sufficiently complete for reliable processing, then processing the command; if the speech input ambiguously corresponds to a single command or is not sufficiently complete for reliable processing, then prompting a user for further speech input to reduce ambiguity or increase completeness, in dependence on a relationship of previously received speech input and at least one command grammar of the plurality of predetermined command grammars, reparsing the further speech input in conjunction with previously parsed speech input, and iterating as necessary. The system also monitors abort, fail or cancel conditions in the speech input.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for generating a synthetic voice. A system configured to practice the method combines a first database of a first text-to-speech voice and a second database of a second text-to-speech voice to generate a combined database, selects from the combined database, based on a policy, voice units of a phonetic category for the synthetic voice to yield selected voice units, and synthesizes speech based on the selected voice units. The system can synthesize speech without parameterizing the first text-to-speech voice and the second text-to-speech voice. A policy can define, for a particular phonetic category, from which text-to-speech voice to select voice units. The combined database can include multiple text-to-speech voices from different speakers. The combined database can include voices of a single speaker speaking in different styles. The combined database can include voices of different languages.

Abstract: A method and system for assessing a student's reading ability is disclosed. An image-capturing device detects, from a worksheet comprising a position-identifying pattern, a first mark in a first region of the worksheet. The first mark is in a first indicator portion of the position-identifying pattern contained within a first indicator region that is associated with a first word. The image-capturing device detects a first note in a note region of the worksheet. Based on whether the first mark, the first note, or both indicates that the first word was read incorrectly or correctly, a processor determines a first reading assessment result for the first word and stores, in a memory, a digital document file comprising the first reading assessment result.

Abstract: Disclosed herein are system, method, and computer program product embodiments for transforming data from a first version, for example an initial version of a database, to a second version, for example a subsequent version of a database. An embodiment operates by modifying the metadata of the data to include transformational clauses, each of which describes how a portion of the data in the first version is transformed to data required by the second version.

Abstract: A method for calculating a similarity of audio files includes constituting a pitch sequence of a first audio file and a pitch sequence of a second audio file; calculating an eigenvector of the first audio file according to the pitch sequence of the first audio file, and calculating an eigenvector of the second audio file according to the pitch sequence of the second audio file; calculating a similarity between the first audio file and the second audio file according to the eigenvector of the first audio file and the eigenvector of the second audio file.

Abstract: A method, medium, and apparatus encoding and/or decoding an audio signal to surround data. While encoding spatial information, which can up-mix an audio signal to a surround signal, to extension data, a length of a payload corresponding to the spatial information is encoded and a payload of the spatial information is decoded using the length of the payload. Accordingly, compatibility of the spatial information can be provided, and the spatial information can be transmitted by effectively embedding the spatial information.

Abstract: Features are disclosed for determining an element of a user utterance or user intent in conjunction with one or more related elements of the user utterance or user intent. A user utterance may be transcribed by an automatic speech recognition (“ASR”) module, and the results may be provided to a natural language understanding (“NLU”) module. The NLU module may perform named entity recognition, intent classification, and/or other processes on the ASR results. In addition, the NLU module may determine or verify the values associated with the recognized named entities using a data store of known values. When two or more named entities are related, their values may be determined and/or verified in conjunction with each other in order to preserve the relationship between them.

Abstract: Systems and methods are provided for assigning a difficulty score to a speech sample. Speech recognition is performed on a digitized version of the speech sample using an acoustic model to generate word hypotheses for the speech sample. Time alignment is performed between the speech sample and the word hypotheses to associate the word hypotheses with corresponding sounds of the speech sample. A first difficulty measure is determined based on the word hypotheses, and a second difficulty measure is determined based on acoustic features of the speech sample. A difficulty score for the speech sample is generated based on the first difficulty measure and the second difficulty measure.

Abstract: The current invention discloses methods, applications, and devices for audio transmission from a mobile terminal. After receiving an audio signal transmission request from a user, the mobile terminal may initiate a recording session to record audio signals into audio frames. During the recording session, the terminal may adjust the audio codecs used for encoding the audio frames based on the workload and the performance of the terminal. By measuring and evaluating the encoding time, the terminal may change between using a floating-point AMR audio codec and a fixed-point AMR audio codec. The encoded audio frames are transmitted to a remote server. The current invention provides a flexible and efficient approach for audio signal encoding and transmission, balancing signal integrity and encoding speed at the same time.

Abstract: Methods and arrangements in a codec for supporting bandwidth extension, BWE, of an harmonic audio signal. The method in the decoder part of the codec comprises receiving a plurality of gain values associated with a frequency band b and a number of adjacent frequency bands of band b. The method further comprises determining whether a reconstructed corresponding frequency band b? comprises a spectral peak. When the band b? comprises a spectral peak, a gain value associated with the band b? is set to a first value based on the received plurality of gain values; and otherwise the gain value is set to a second value based on the received plurality of gain values. The suggested technology enables bringing gain values into agreement with peak positions in a bandwidth extended frequency region.

Abstract: Exemplary embodiments of methods and apparatuses for automatic speech recognition are described. First model parameters associated with a first representation of an input signal are generated. The first representation of the input signal is a discrete parameter representation. Second model parameters associated with a second representation of the input signal are generated. The second representation of the input signal includes a continuous parameter representation of residuals of the input signal. The first representation of the input signal includes discrete parameters representing first portions of the input signal. The second representation includes discrete parameters representing second portions of the input signal that are smaller than the first portions. Third model parameters are generated to couple the first representation of the input signal with the second representation of the input signal. The first representation and the second representation of the input signal are mapped into a vector space.

Abstract: A method and apparatus for managing an advertisement application in a mobile advertising system is provided. When an advertisement application for representing an advertisement is installed in an advertisement-receiving terminal, a registration request for the installed advertisement application is made. The advertisement-receiving terminal assigns an application Identifier (ID) to the advertisement application in response to the registration request, and stores a profile of the advertisement application in association with the assigned application ID.

Abstract: A method and a device for training an acoustic language model, include: conducting word segmentation for training samples in a training corpus using an initial language model containing no word class labels, to obtain initial word segmentation data containing no word class labels; performing word class replacement for the initial word segmentation data containing no word class labels, to obtain first word segmentation data containing word class labels; using the first word segmentation data containing word class labels to train a first language model containing word class labels; using the first language model containing word class labels to conduct word segmentation for the training samples in the training corpus, to obtain second word segmentation data containing word class labels; and in accordance with the second word segmentation data meeting one or more predetermined criteria, using the second word segmentation data containing word class labels to train the acoustic language model.

Abstract: A method of synchronizing robot motion with a social interaction. The method comprises storing in the robot a map that associates keywords with at least one robot motion, composing by the robot a dialogue based on a context of a social interaction with a human being, searching the dialogue for keywords, parsing the dialogue to determine its syntax, and analyzing the syntax. The method further comprises generating, by the robot, a robot motion script synchronized with the dialogue based on mapping one or more keywords located in the dialogue to robot motions, based on the syntax of the dialogue, and based on a physical cadence, wherein the robot motion script comprises a sequence of separate robot motions. The method further comprises playing aloud the dialogue by the robot and performing the robot motion script by the robot in synchronization with the playing aloud of the dialogue.

Abstract: Systems and method are provided for rendering different speech-based services to a plurality of users. A service-providing system may be accessed via a plurality of connectivity ports. Each of the connectivity ports may be associated with at least one of a plurality of different speech-related services. The connectivity ports may be associated with the different speech-related services may be performed before receiving user service requests. The service-providing system may comprise a plurality of processing components, each of which may be configurable to provide one or more of a plurality of different speech-related services. The service-providing system may further comprise a connection component, which may be operable to establish a connection between the respective connectivity port and a processing component having a configuration of suitable for performing a service requested through the respective connectivity port.

Abstract: A method and apparatus for speech translation. The method includes: receiving a source speech; extracting non-text information in the source speech; translating the source speech into a target speech; and adjusting the translated target speech according to the extracted non-text information so that the target speech preserves the non-text information in the source speech. The apparatus includes: a receiving module for receiving source speech; an extracting module for extracting non-text information in the source speech; a translation module for translating the source speech into a target speech; and an adjusting module for adjusting the translated target speech according to the extracted non-text information so that the target speech preserves the non-text information in the source speech.

Abstract: Provided are a coding device, a communication processing device, and a coding method, whereby processing operation load (computational load) is significantly reduced for a configuration which computes either frame energy or sub-frame energy of an input signal, using auto-correlation operations, without causing a decline in the precision of either the frame energy or the sub-frame energy. In a coding device (101), a sub-frame energy computation unit (201) computes the sub-frame energy by substituting the sum of input signal auto-correlation operations in a first range with the sum of auto-correlation operations in a second range which differs at least partially from the first range.

Abstract: Spread level parameter correcting means 501 receives a contour parameter as information representing the contour of a feature sequence (a sequence of features of a signal considered as the object of generation) and a spread level parameter as information representing the level of a spread of the distribution of the features in the feature sequence. The spread level parameter correcting means 501 corrects the spread level parameter based on a variation of the contour parameter represented by a sequence of the contour parameters. Feature sequence generating means 502 generates the feature sequence based on the contour parameters and the corrected spread level parameters.

Abstract: Techniques and systems for training an acoustic model are described. In an embodiment, a technique for training an acoustic model includes dividing a corpus of training data that includes transcription errors into N parts, and on each part, decoding an utterance with an incremental acoustic model and an incremental language model to produce a decoded transcription. The technique may further include inserting silence between a pair of words into the decoded transcription and aligning an original transcription corresponding to the utterance with the decoded transcription according to time for each part. The technique may further include selecting a segment from the utterance having at least Q contiguous matching aligned words, and training the incremental acoustic model with the selected segment. The trained incremental acoustic model may then be used on a subsequent part of the training data. Other embodiments are described and claimed.

Abstract: Systems, methods, and computer-readable media that may be used to modify a voice action system to include voice actions provided by advertisers or users are provided. One method includes receiving electronic voice action bids from advertisers to modify the voice action system to include a specific voice action (e.g., a triggering phrase and an action). One or more bids may be selected. The method includes, for each of the selected bids, modifying data associated with the voice action system to include the voice action associated with the bid, such that the action associated with the respective voice action is performed when voice input from a user is received that the voice action system determines to correspond to the triggering phrase associated with the respective voice action.

Abstract: Methods and devices are disclosed for enabling improved transmission performance on a multi-SIM wireless communication device. The wireless device may detect a voice communication in a held state on a modem stack associated with the first SIM and an active voice communication on a modem stack associated with the second SIM. The wireless device may detect a conflict between at least one silence descriptor (SID) frame scheduled for transmission by the modem stack associated with the first SIM and a transmit opportunity for the modem stack associated with the second SIM. Once the wireless device identifies a SID frame transmission rate for the modem stack associated with the first SIM, the wireless device may apply a reduction scheme to the SID frames scheduled to be transmitted by the modem stack associated with the first SIM.

Abstract: A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector.

Abstract: Systems and methods for bio-phonetic multi-phrase speaker identity verification are disclosed. Generally, a speaker identity verification engine generates a dynamic phrase including at least one dynamically-generated word. The speaker identity verification engine prompts a user to speak the dynamic phrase and receives a dynamic phrase utterance. The speaker identity verification engine extracts at least one voice characteristic from the dynamic phrase utterance and compares the at least one voice characteristic with a voice profile the generate a score. The speaker identity verification engine then determines whether to accept a speaker identity claim based on the score.

Abstract: A constrained variable rate coding technique limits the number of bits used in an encoding process. A quality setting indicates a maximum level of quality to be used in the encoding process which limits the number of bits used in the encoding process. A bandwidth reclamation factor which indicates an amount of bandwidth to conserve may also be used with the quality setting. The constrained variable rate coding technique using a lower quality encoding process for less complex video data and a higher quality encoding technique for higher quality video data.

Abstract: A television (1) including a function of a call using Internet Protocol (IP) includes: a communicating section to transmit and receive a call signal over an IP communication network; an incoming call destination identifying section (11) to identify a user who is designated as an incoming call destination; a judging section to detect a person who is present around the television (1); and a communication control section (13) to transfer the call signal to a mobile phone (4) of the user in a case where a plurality of persons containing the user has been detected. This offers a television capable of a call and ensuring the privacy of the call.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

Abstract: An electronic device includes a display module and a dictionary storage module which stores dictionary data that causes a plurality of entry words including compound words obtained by connecting a plurality of words to correspond to explanatory information on the entry words. When the user retrieves a dictionary, entry words for compound words are retrieved from the entry words in the dictionary storage module and words common to the retrieved compound words are listed and displayed on the display module. Entry words for compound words connecting with a word specified by a user operation in the displayed list are read from the dictionary data and displayed in list form on the display module.

Abstract: Methods and apparatus to generate a speech recognition library for use by a speech recognition system are disclosed. An example method comprises identifying a plurality of video segments having closed caption data corresponding to a phrase, the plurality of video segments associated with respective ones of a plurality of audio data segments, computing a plurality of difference metrics between a baseline audio data segment associated with the phrase and respective ones of the plurality of audio data segments, selecting a set of the plurality of audio data segments based on the plurality of difference metrics, identifying a first one of the audio data segments in the set as a representative audio data segment, determining a first phonetic transcription of the representative audio data segment, and adding the first phonetic transcription to a speech recognition library when the first phonetic transcription differs from a second phonetic transcription associated with the phrase in the speech recognition library.

Abstract: A method and computer program product for providing a random linear coding approach to distributed data storage is presented. A file is broken into a plurality of pieces. For every peer (peer means storage-location with limited storage space), the number of coded-pieces the peer can store is determined. Each of the coded-piece is determined by taking random linear combination of all the pieces of the entire file. The associate code-vector is stored for every coded-piece. The file is retrieved by collecting code-vectors and the coded-pieces from the peers and viewing the collected code-vectors as a matrix. When a dimension of the matrix is equal to the number of pieces of the file, the file is recovered using the collection of code vectors in the matrix.

Abstract: A device for operating an automated machine for handling, assembling or machining workpieces, comprising: a display apparatus having a screen for displaying a graphic user interface for controlling and/or monitoring machine functions of the machine, an operating apparatus for inputting command-triggering operator actions for controlling functions of the machine and controlling functions of the graphic user interface, and a controller for implementing input command-triggering operator actions into control commands for controlling functions of the machine and/or functions of the graphic user interface. The operating apparatus comprises an apparatus for inputting manual operator actions and an apparatus for inputting contact-free operator actions.

Abstract: A vehicle audio system that includes a source of audio signals, which may include both entertainment audio signals and announcement audio signals, speakers for radiating audio signals, and spatial enhancement circuitry comprising circuitry to avoid applying spatial enhancement processing to the announcement audio signals.

Abstract: In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end.

Abstract: Provided is an audio processing device comprising: a feature data generation unit which generates, for each unit section of an audio signal, section feature data expressing features of the audio signal in the unit section; a feature variation calculation unit which calculates, for each unit section, a feature variation value quantifying temporal variation of the features in the unit section, by setting the unit section as a target section and using section feature data of unit sections close to the target section; and a section judgment unit which judges, for each unit section, whether the unit section is a feature unit section including a variation point of the features, based on comparison of a threshold value and the feature variation value. Through the above, the audio processing device can detect feature unit sections from an audio signal of an AV content or the like.

Abstract: The application relates to a method for encoding time marking information within audio data. According to the method, time marking information is encoded as audio metadata within the audio data. The time marking information indicates at least one section of an audio object encoded in the audio data. E.g. the time marking information may specify a start position and an end position of the section or only a start position. The at least one section may be a characteristic part of the audio object, which allows instant recognition by listening. The time marking information encoded in the audio data enables instantaneous browsing to a certain section of the audio object. The application further relates to a method for decoding the time marking information encoded in the audio data.

Abstract: A network component comprising a communications portion and a processor portion is disclosed. The communications portion may be configured to detect a signal indicative of a call associated with a mobile device. The processor portion may be configured to detect at least one record cue in the signal. The processor portion may be also be configured to respectively capture at least one portion of the call upon the at least one record cue being detected. The processor portion may also be configured to respectively associate at least one identifier with the at least one captured portion of the call. The identifier may respectively identify the at least one captured portion of the call.

Abstract: A user device presents passages of an electronic publication. The user device tracks a user's access behavior for the passages of the electronic publication. The user device identifies the user's favorite passages of the electronic publication based on the user's access behavior and stores an identification of the user's favorite passages.

Abstract: In a method of improving perceived loudness and sharpness of a reconstructed speech signal delimited by a predetermined bandwidth, performing the steps of providing (S10) the speech signal, and separating (S20) the provided signal into at least a first and a second signal portion. Subsequently, adapting (S30) the first signal portion to emphasize at least a predetermined frequency or frequency interval within the first bandwidth portion. Finally, reconstructing (S40) the second signal portion based on at least the first signal portion, and combining (S50) the adapted first signal portion and the reconstructed second signal portion to provide a reconstructed speech signal with an overall improved perceived loudness and sharpness.

Abstract: In prediction of a speech quality evaluation score such as a phone speech, even when a background noise exists, a subjective opinion score is predicted with high precision. A speech quality evaluation system that outputs a predicted value of the subjective opinion score for an evaluation speech such as a far-end speech of a phone, includes a speech distortion calculation unit that conducts, after calculating frequency characteristics of the evaluation speech, a process of subtracting given frequency characteristics from frequency characteristics of the evaluation speech, and calculates the speech distortion on the basis of the frequency characteristics after the subtracting process has been conducted, and a subjective evaluation prediction unit that calculates the predicted value of the subjective opinion score on the basis of the speech distortion.

Abstract: Systems, methods and apparatus are described herein for continuously measuring voice clarity and speech intelligibility by evaluating a plurality of telecommunications channels in real time. Voice clarity and speech intelligibility measurements may be formed from chained, configurable DSPs that can be added, subtracted, reordered, or configured to target specific audio features. Voice clarity and speech intelligibility may be enhanced by altering the media in one or more of the plurality of telecommunications channels. Analytics describing the measurements and enhancements may be displayed in reports, or in real time via a dashboard.

Abstract: The invention relates to a method for determining a quality indicator representing a perceived quality of an output signal of an audio device with respect to a reference signal. Such audio device may for example be a speech processing system. In the method the reference signal and the output signal are processed and compared. The processing includes dividing the reference signal and the output signal into mutually corresponding time frames. The processing further includes scaling the reference signal towards a fixed intensity level. Time frames of the output signal are selected based on measurements performed on the scaled reference signal. Then, a noise contrast parameter is calculated based on the selected time frames of the output signal. A noise suppression is applied on at least one of the reference signal and the output signal based on the noise contrast parameter.

Abstract: Methods and apparatus to monitor media exposure in vehicles are disclosed. An example implementation includes collecting audience measurement data with a media monitoring device fixed in a vehicle and transmitting the audience measurement data from the media monitoring device to a shuttle located within the vehicle, the shuttle being incapable of collecting audience measurement data independent of the media monitoring device.