Abstract: Techniques for handling large string values in a string dictionary within an in-memory database system are presented. The string dictionary can be a paged-column main dictionary. For example, a value block of a dictionary may be loaded into memory, where the dictionary stores values for dictionary compression. The value block may include a first portion of a large string value and one or more logical pointers to one or more large string pages containing a remainder of the large string value. The large string value may be materialized into a contiguous location in memory, and a pointer to the materialized large string value may be created.

Abstract: A system and method for multimedia content protection on elastic cloud infrastructures is presented. The system can be used to protect various multi-media contents, including regular 2D videos, new 3D videos, animated graphics, images, audios clips, songs, and music clips. The system can run on private clouds, public clouds, or any combination of public-private clouds. The system is scalable and cost effective.

Abstract: A text-to-speech (TTS) computing includes a processor and a memory. The TTS computing device is configured to generate a machine pronunciation of a text data according to at least one phonetic rule, and provide the machine pronunciation to a user interface of the TTS computing device such that the machine pronunciation is audibly communicated to a user of the TTS computing device. The TTS computing device is also configured to receive a pronunciation correction of the machine pronunciation from the user via the user interface, and store the pronunciation correction in a TTS data source. The TTS computing device is further configured to assign the pronunciation correction provided by the user to a user profile that corresponds to the text data.

Abstract: An autocorrelation calculating part calculates autocorrelation Ro(i) from an input signal. A predictive coefficient calculating part performs linear predictive analysis using modified autocorrelation R?o(i) obtained by multiplying the autocorrelation Ro(i) by a coefficient wo(i). Here, a case is comprised where, for at least part of each order i, the coefficient wo(i) corresponding to each order i monotonically decreases as a value having positive correlation with a pitch gain in an input signal of a current frame or a past frame increases.

Abstract: A method to design, implement and create distributed indexes for storing and comparing fingerprints of videos is presented. The method effectively utilizes cloud computing platforms that offer varying amounts of computing resources. The method enables the distributed index to scale to large numbers of data points and the distributed index is robust to failures within the computing resources maintaining the index. The method minimizes the memory required to maintain the distributed index and reduces the I/O operations needed to process operations performed on the index. The method improves the efficiency of the index to process queries.

Abstract: A user wearing headphones (e.g., to listen to music, to engage in a voice call, etc.) may speak while receiving an audio signal through the headphones, which may cause the user to produce Lombard speech. Because the Lombard effect is generally involuntary, the user may be unaware that he or she is producing Lombard speech. The Lombard speech may inconvenience proximate individuals and/or embarrass the user (e.g., in an office, in an airport, etc.). An apparatus may be configured to receive, through a microphone communicatively coupled to the apparatus, an audio signal. The apparatus may be configured to determine whether the audio signal indicates speech by a user. The apparatus may be further configured to alert the user based on the determination that the audio signal indicates Lombard speech by the user.

Abstract: Relative Excitation Features, in all conditions, are far superior to conventional acoustic features like Mel-Frequency Cepstrum (MFC) and Perceptual Linear Prediction (PLP), and provide much more speaker-independence, channel-independence, and noise-immunity. Relative Excitation features are radically different than conventional acoustic features. Relative Excitation method doesn't try to model the speech-production or vocal tract shape, doesn't try to do deconvolution, and doesn't utilize LP (Linear Prediction) and Cepstrum techniques. This new feature set is completely related to human hearing. The present invention is inspired by the fact that human auditory perception analyzes and tracks the relations between spectral frequency component amplitudes and the “Relative Excitation” name implies relative excitation levels of human auditory neurons. Described herein is a major breakthrough for explaining and simulating the human auditory perception and its robustness.

Abstract: An improved concept for coding sample values of a spectral envelope is obtained by combining spectrotemporal prediction on the one hand and context-based entropy coding the residuals, on the other hand, while particularly determining the context for a current sample value dependent on a measure of a deviation between a pair of already coded/decoded sample values of the spectral envelope in a spectrotemporal neighborhood of the current sample value. The combination of the spectrotemporal prediction on the one hand and the context-based entropy coding of the prediction residuals with selecting the context depending on the deviation measure on the other hand harmonizes with the nature of spectral envelopes.

Abstract: The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.

Abstract: In some aspects, a method of recognizing speech that comprises natural language and at least one word specified in at least one domain-specific vocabulary is provided. The method comprises performing a first speech processing pass comprising identifying, in the speech, a first portion including the natural language and a second portion including the at least one word specified in the at least one domain-specific vocabulary, and recognizing the first portion including the natural language. The method further comprises performing a second speech processing pass comprising recognizing the second portion including the at least one word specified in the at least one domain-specific vocabulary.

Abstract: A method for classifying lexical stress in an utterance includes generating a feature vector representing stress characteristics of a syllable occurring in the utterance, wherein the feature vector includes a plurality of features based on prosodic information and spectral information, computing a plurality of scores, wherein each of the plurality of scores is related to a probability of a given class of lexical stress, and classifying the lexical stress of the syllable based on the plurality of scores.

Abstract: An autocorrelation calculating part calculates autocorrelation Ro(i) from an input signal. A predictive coefficient calculating part performs linear predictive analysis using modified autocorrelation R?o(i) obtained by multiplying the autocorrelation Ro(i) by a coefficient wo(i). Here, it is assumed that a case where, for at least part of each order i, the coefficient wo(i) corresponding to each order i monotonically increases as a value having negative correlation with a fundamental frequency of an input signal in a current frame or a past frame increases and a case where the coefficient wo(i) monotonically decreases as a value having positive correlation with a pitch gain in a current frame or a past frame increases, are included.

Abstract: Systems, methods, and devices for intelligent speech recognition and processing. According to one embodiment, a method for improving intelligibility of a speech signal may include (1) at least one processor receiving an incoming speech signal comprising a plurality of sound elements; (2) the at least one processor recognizing a sound element in the incoming speech signal to improve the intelligibility thereof; (3) the at least one processor processing the sound element by at least one of modifying and replacing the sound element; and (4) the at least one processor outputting the processed speech signal comprising the processed sound element.

Abstract: Method and apparatus for diphone or concatenative synthesis to compensate for insufficient or missing diphones.

Abstract: A mechanism is provided in a data processing system for minimizing uncertainty envelopes in trajectories of evolving ensemble members. The mechanism generates a trajectory forecast of each member object of an ensemble based on an initial state-space and a model for predicting trajectories of the member objects to generate a plurality of trajectory forecasts. Each of the plurality of trajectory forecasts has an individual uncertainty envelope. The mechanism applies a classification algorithm on the plurality of trajectory forecasts to identify at least one group of member objects having similar trajectory forecasts, generates a reduced ensemble of member objects including the identified group of member objects, and reconfigures the state-space and the model for predicting trajectories. The mechanism generates an updated trajectory forecast of each member object of the reduced ensemble based on the reconfigured state-space and the reconfigured model for predicting trajectories of the member objects.

Abstract: Provided is an audio encoding method. The audio encoding method includes: acquiring envelopes based on a predetermined sub-band for an audio spectrum; quantizing the envelopes based on the predetermined sub-band; and obtaining a difference value between quantized envelopes for adjacent sub-bands and lossless encoding a difference value of a current sub-band by using a difference value of a previous sub-band as a context. Accordingly, the number of bits required to encode envelope information of an audio spectrum may be reduced in a limited bit range, thereby increasing the number of bits required to encode an actual spectral component.

Abstract: Generally, this disclosure provides systems, devices, methods and computer readable media for adaptation of language models and semantic tracking to improve automatic speech recognition (ASR). A system for recognizing phrases of speech from a conversation may include an ASR circuit configured to transcribe a user's speech to a first estimated text sequence, based on a generalized language model. The system may also include a language model matching circuit configured to analyze the first estimated text sequence to determine a context and to select a personalized language model (PLM), from a plurality of PLMs, based on that context. The ASR circuit may further be configured to re-transcribe the speech based on the selected PLM to generate a lattice of paths of estimated text sequences, wherein each of the paths of estimated text sequences comprise one or more words and an acoustic score associated with each of the words.

Abstract: Techniques of conducting an online meeting involve outputting ambient sound to a participant of an online meeting. Along these lines, in an online meeting during which a participant wears headphones, the participant's computer receives microphone input that contains both speech from the participant and ambient sound that the participant may wish to hear. In response to receiving the microphone input, the participant's computer separates low-volume sounds from high-volume sounds. However, instead of suppressing this low-volume sound from the microphone input, the participant's computer renders this low-volume sound. In most cases, this low-volume sound represents ambient sound generated in the vicinity of the meeting participant. The participant's computer then mixes the low-volume sound with speech received from other conference participants to form output in such a way that the participant may distinguish this sound from the received speech.

Abstract: According to an embodiment, a text-to-speech device includes a context acquirer, an acoustic model parameter acquirer, a conversion parameter acquirer, a converter, and a waveform generator. The context acquirer is configured to acquire a context sequence affecting fluctuations in voice. The acoustic model parameter acquirer is configured to acquire an acoustic model parameter sequence that corresponds to the context sequence and represents an acoustic model in a standard speaking style of a target speaker. The conversion parameter acquirer is configured to acquire a conversion parameter sequence corresponding to the context sequence to convert an acoustic model parameter in the standard speaking style into one in a different speaking style. The converter is configured to convert the acoustic model parameter sequence using the conversion parameter sequence. The waveform generator is configured to generate a voice signal based on the acoustic model parameter sequence acquired after conversion.

Abstract: A state predicting circuitry predicts a route showing a future change in the vehicle state from among a plurality of routes from a first node to a second node. The first node corresponds to the current vehicle state. The second node corresponds to the vehicle state after having transitioned a predetermined number of times from the first node. The state predicting circuitry predicts a route in which at least one of an accumulated value of the node that exists in the routes and an accumulated value of the link that exists in the routes is greatest, from among the plurality of routes.

Abstract: Speech transcoding in packet networks may be useful when both incoming and outgoing speech streams of the transcoding entity are packet based. This can be any transcoding entity having packet interfaces. A method can include omitting jitter buffering before decoding in a transcoder and omitting bad frame handling in a decoding stage of a transcoder. The method can also include freezing a decoder and the encoder when a packet is not received. The method can also include sending packet loss information from the decoder to the encoder as side information when the packet is not received. The method can further include setting an outgoing packet stream to permit detection of missing packets by a downstream decoder upon receiving a valid packet after the packet is not received.

Abstract: An audio decoder device for decoding a bitstream includes a bitstream receiver configured to receive the bitstream and to derive an encoded audio signal from the bitstream; a core decoder module configured for deriving a decoded audio signal in a time domain from the encoded audio signal; a temporal envelope generator configured to determine a temporal envelope of the decoded audio signal; a bandwidth extension module configured to produce a frequency domain bandwidth extension signal; a time-to-frequency converter configured to transform the decoded audio signal into a frequency domain decoded audio signal; a combiner configured to combine the frequency domain decoded audio signal and the frequency domain bandwidth extension signal in order to produce a bandwidth extended frequency domain audio signal; and a frequency-to-time converter configured to transform the bandwidth extended frequency domain audio signal into a bandwidth-extended time domain audio signal.

Abstract: An audio signal decoder for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information and in dependence on a rendering information has an object parameter determinator. The object parameter determinator is configured to obtain inter-object-correlation values for a plurality of pairs of audio objects. The object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether to evaluate individual inter-object-correlation bitstream parameter values to obtain inter-object-correlation values for a plurality of pairs of related audio objects, or to obtain inter-object-correlation values for a plurality of pairs of related audio objects using a common inter-object-correlation bitstream parameter value.

Abstract: A client device encodes data into an audio signal and communicates the audio data to an additional client device, which decodes the data from the audio signal. The data is partitioned into characters, which are subsequently partitioned into a plurality of sub-characters. Each sub-character is encoded into a frequency, and multiple frequencies that encode sub-characters are combined by the client device to generate an audio signal. Frequencies encoding sub-characters may be above 16 kilohertz, so the sub-characters are transmitted using frequencies that are inaudible to humans. The audio signal is communicated to an additional client device, which decodes frequencies from the audio signal to sub-characters, which are then combined into characters by the additional client device to generate the data.

Abstract: Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPIus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented.

Abstract: Proposed is a method and apparatus for determining a weighting function for quantizing a linear predictive coding (LPC) coefficient and having a low complexity. The weighting function determination apparatus may convert an LPC coefficient of a mid-subframe of an input signal to one of a immittance spectral frequency (ISF) coefficient and a line spectral frequency (LSF) coefficient, and may determine a weighting function associated with an importance of the ISF coefficient or the LSF coefficient based on the converted ISF coefficient or LSF coefficient.

Abstract: A system and a method for assessing a condition in a subject. Phones from speech of the subject are recognized, one or more prosodic or speech-excitation-source features of the phones are extracted, and an assessment of a condition of the subject, is generated based on a correlation between the features of the phones and the condition.

Abstract: A system for generating comfort noise for a stream of frames carrying an audio signal includes frame characterizing logic configured to generate a set of filter parameters characterising the frequency content of a frame; an analysis filter adapted using the filter parameters and configured to filter the frame so as to generate residual samples; an analysis controller configured to cause the residual samples to be stored in a store responsive to receiving an indication that the frame does not comprise speech; and a synthesis controller operable to select stored residual samples from the store and cause a synthesis filter, inverse to the analysis filter and adapted using filter parameters generated by the frame characterizing logic for one or more frames not comprising speech, to filter the selected residual samples so as to generate a frame of comfort noise.

Abstract: A likely source language of a media item can be identified by attempting an initial language identification of the media item based on intrinsic or extrinsic factors, such as words in the media item and languages known by the media item author. This initial identification can generate a list of most likely source languages with corresponding likelihood factors. Translations can then be performed presuming each of the most likely source languages. The translations can be performed for multiple output languages. Each resulting translation can receive a corresponding score based on a number of factors. The scores can be combined where they have a common source language. These combined scores can be used to weight the previously identified likelihood factors for the source languages of the media item.

Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

Abstract: A method includes obtaining a signal type of an audio signal and a low frequency band signal of the audio signal, where the audio signal includes the low frequency band signal and a high frequency band signal; obtaining a frequency envelope of the high frequency band signal according to the signal type; predicting an excitation signal of the high frequency band signal according to the low frequency band signal; and restoring the high frequency band signal according to the frequency envelope of the high frequency band signal and the excitation signal of the high frequency band signal. By using the technical solutions of the embodiments of the present invention, an error existing between a high frequency band signal obtained by prediction and an actual high frequency band signal can be effectively reduced, and an accuracy rate of the predicted high frequency band signal can be increased.

Abstract: A mobile device, such as a media player, a telephone, or a gaming console, concurrently produces multiple content streams based on differing playlists. The mobile device includes a memory, a controller, and at least two output interfaces. The controller processes concurrently the content stored in the memory to produce a first content stream according to a first playlist and a second content stream according to a second playlist. More content streams may be produced based on additional playlist, all of which playlist differ from each other. The content streams may include audio content, video content, or a combination thereof. For each content stream, an output interface is available to convey the content stream. Also disclosed is a method of processing content to produce and send content streams to output interfaces.

Abstract: Methods and apparatus for audio watermarking are disclosed. Example methods disclosed herein include obtaining a watermarked noise signal having energy in a first frequency band, the watermarked noise signal being associated with a first operating state of a device. Disclosed example methods also include adjusting the watermarked noise signal based on an ambient audio level sensed by an audio sensor. Disclosed example methods further include emitting the adjusted watermarked noise signal via a speaker in response to determining that the device is in the first operating state.

Abstract: Embodiments of the present invention provide systems and methods for internationalization of real-world features during navigation. The method includes receiving a request for navigation to a landmark in a particular language. The method further includes retrieving keywords from a database associated with the landmark, and scoring the keywords based on their level of recognition to a user. The highest ranked keywords are then sent to the user to navigate to the landmark.

Abstract: Embodiments of the present invention provide systems and methods for internationalization of real-world features during navigation. The method includes receiving a request for navigation to a landmark in a particular language. The method further includes retrieving keywords from a database associated with the landmark, and scoring the keywords based on their level of recognition to a user. The highest ranked keywords are then sent to the user to navigate to the landmark.

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: An electronic device which includes a plurality of microphones and an audio data processing module is provided. The plurality of microphones is operatively coupled to the electronic device, and the audio data processing module is capable of being implemented with at least one processor. The audio data processing module recognizes a specified command, based on first audio data collected using a portion of the plurality of microphones and executes a function or an application corresponding to second audio data collected using all the plurality of microphones, when the specified command is recognized.

Abstract: The present disclosure includes a hybrid waveform system that displays a hybrid waveform to a user. In general, the hybrid waveform system provides a hybrid waveform to a user that uses converted readable text and waveforms to represent an audio segment. By providing a user with a hybrid waveform, the hybrid waveform system offers users with a number of benefits, such as providing an audio display that enables a user to quickly ascertain context information and audio information typically missing from audio transcriptions.

Abstract: The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value (311, 312) for a first frequency subband (205) of an audio signal is described. The first banded tonality value (311, 312) is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal.

Abstract: The present invention provides a system comprising a user interface configured to receive text input by a user, a text prediction engine comprising a plurality of language models and configured to receive the input text from the user interface and to generate concurrently text predictions using the plurality of language models, and wherein the text prediction engine is further configured to provide text predictions to the user interface for display and user selection. An analogous method and an interface for use with the system and method are also provided.

Abstract: A disclosed speech processor includes a front end to receive a speech input and generate a feature vector indicative of a portion of the speech input and a Gaussian mixture (GMM) circuit to receive the feature vector, model any one of a plurality of GMM speech recognition algorithms, and generate a GMM score for the feature vector based on the GMM speech recognition algorithm modeled. In at least one embodiment, the GMM circuit includes a common compute block to generate feature a vector sum indicative of a weighted sum of differences squares between the feature vector and a mixture component of the GMM speech recognition algorithm. In at least one embodiment, the GMM speech recognition algorithm being modeled includes a plurality of Gaussian mixture components and the common compute block is operable to generate feature vector scores corresponding to each of the plurality of mixture components.

Abstract: In a speech coding scheme based on a speech production model, such as a CELP-based scheme, an object of the present invention is to provide a decoding method that can reproduce natural sound even if the input signal is a noise-superimposed speech. The decoding method includes a speech decoding step of obtaining a decoded speech signal from an input code, a noise generating step of generating a noise signal that is a random signal, and a noise adding step of outputting a noise-added signal, the noise-added signal being obtained by summing the decoded speech signal and a signal obtained by performing, on the noise signal, a signal processing that is based on at least one of a power corresponding to a decoded speech signal for a previous frame and a spectrum envelope corresponding to the decoded speech signal for the current frame.

Abstract: Systems and methods are described for personifying communications. According to at least one embodiment, the computer-implemented method for personifying a natural-language communication includes observing a linguistic pattern of a user. The method may also include analyzing the linguistic pattern of the user and adapting the natural-language communication based at least in part on the analyzed linguistic pattern of the user. In some embodiments, observing the linguistic pattern of the user may include receiving data indicative of the linguistic pattern of the user. The data may be one of verbal data or written data. Written data may include at least one of a text message, email, social media post, or computer-readable note. Verbal data may include at least one of a recorded telephone conversation, voice command, or voice message.

Abstract: Embodiments are directed towards generating a representative sampling as a subset from a larger dataset that includes unstructured data. A graphical user interface enables a user to provide various data selection parameters, including specifying a data source and one or more subset types desired, including one or more of latest records, earliest records, diverse records, outlier records, and/or random records. Diverse and/or outlier subset types may be obtained by generating clusters from an initial selection of records obtained from the larger dataset. An iteration analysis is performed to determine whether a sufficient number of clusters and/or cluster types have been generated that exceed at least one threshold and when not exceeded, additional clustering is performed on additional records. From the resultant clusters, and/or other subtype results, a subset of records is obtained as the representative sampling subset.

Abstract: A phase vocoder executes a fast-Fourier transform (FFT) with respect to an input audio data stream to generate an array of frequency-domain values corresponding to respective frequencies that are nominally uniformly distributed across a frequency range of interest, each of the frequency-domain values being representative of amplitude and phase of a spectral component of the input audio data stream at the respective frequency. The phase vocoder scales the nominally uniform distribution of the respective frequencies to reduce a cumulative error across the frequency distribution resulting from finite precision of a digital representation and then implements at least one of a time-stretching operation or a pitch-shifting operation with respect to the input data stream by manipulating the frequency-domain values with respect to one another within the array.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for intelligently cancelling user inputs. In one aspect, a requests input by a user is received by a dialog engine. A prompt or notification regarding the request is output by the dialog engine. That the user has taken an action in response to the prompt or notification is determined by the dialog engine. Based on the action taken by the user, that the response corresponds to a potential cancellation command is determined by the dialog system.

Abstract: Embodiments of systems and methods are described for generating layered audio such that computing devices can request a variable amount of data based on criteria such as their available bandwidth, device capability, or user selection. A base layer and one or more enhancement layers that incrementally enhance the previous layers may be generated. A computing device may retrieve the base layer and/or one or more enhancement layers, adjusting, in real-time or near real-time, which layers are retrieved based on fluctuations in the available bandwidth among other possible criteria.

Abstract: The present invention provides a method for obtaining an accurate detected value of a similarity, such as an hitting sound.

Abstract: An apparatus for generating a frequency enhancement signal, includes: a signal generator for generating an enhancement signal from a core signal, the enhancement signal including an enhancement frequency range not included in the core signal, wherein a time portion of the enhancement signal includes subband signals for a plurality of subbands; a synthesis filterbank for generating the frequency enhanced signal using the enhancement signal, wherein the signal generator is configured for performing an energy limitation in order to make sure that the frequency enhanced signal obtained by the synthesis filterbank is so that an energy of a higher band is, at the most, equal to an energy in a lower band or is greater than an energy of a higher band, at the most, by a predefined threshold.

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.