Abstract: An exemplary extended reality audio processing system accesses first and second audio streams representative, respectively, of first and second virtual sounds within an extended reality world being experienced by a user by way of a media player device. The system also accesses acoustic propagation data representative of characteristics that affect the propagation of the first and second virtual sounds to an avatar of the user. Based on the first and second audio streams and the real-time acoustic propagation data, the system renders a composite binaural audio stream representative of a composite virtual sound that comprises a combination of simulated reproductions of the first and second virtual sounds that are customized to account for the characteristics that affect the propagation. The system further transmits the composite binaural audio stream to the media player device for presentation to the user as the user experiences the extended reality world.

Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing moving spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream.

Abstract: A method of connecting a service between a device and at least one other device is provided. The method includes recording, by the device, a user voice input in a state where a voice command button has been input, outputting first information based on the recorded user voice when an input of the voice command button is cancelled, receiving, by the device, second information corresponding to the first information, recognizing a service type according to the first information and the second information, connecting the device to a subject device in an operation mode of the device determined according to the recognized service type, and performing a service with the connected subject device.

Abstract: The invention provides methods and apparatuses for encoding a stereo audio signal having a left channel and a right channel. The apparatus includes a prediction coefficient estimator, downmixer, and multiplexer. The downmixer is configured operate either in a prediction mode or a non-prediction mode.

Abstract: The invention provides methods and devices for outputting a stereo audio signal having a left channel and a right channel. The apparatus includes a demultiplexer, decoder, and upmixer. The upmixer is configured operate either in a prediction mode or a non-prediction mode based on a parameter encoded in the audio bitstream.

Abstract: Disclosed are a high-band encoding/decoding method and device for bandwidth extension. A high-band encoding method comprising the steps of: generating sub band-specific bit allocation information on the basis of a low-band envelope; determining, on the basis of the sub band-specific bit allocation information, the sub band requiring an envelope update in a high band; and generating, for the determined sub band, refinement data relating to the envelope update. A high-band decoding method comprising the steps of: generating sub band-specific bit allocation information on the basis of a low-band envelope; determining, on the basis of the sub band-specific bit allocation information, the sub band requiring an envelope update in a high band; and decoding, for the determined sub band, refinement data relating to the envelope update, thereby updating the envelope.

Abstract: Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and “notch” the so-called “singing” or “howling” frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons).

Abstract: A system for encoding and applying Dynamic Range Control/Compression (DRC) gain values to a piece of sound program content is described. In particular, a set of DRC gain values representing a DRC gain curve for the piece of content may be divided into frames corresponding to frames of the piece of content. A set of fields may be included with an audio signal representing the piece of content. The additional fields may represent the DRC gain values using linear or spline interpolation. The additional fields may include 1) an initial gain value for each DRC frame, 2) a set of slope values at particular points in the DRC curve, 3) a set of time delta values for each consecutive pair of slope values, and/or 4) one or more gain delta values representing changes of DRC gain values in the DRC gain curve between points of the slope values.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: An example method of decoding video data includes determining, by a video decoder and based on syntax elements in an encoded video bitstream, a plurality of values for a current block of the video data; performing, by the video decoder, a multi-pass non-separable inverse transformation on the plurality of values to derive residual data that represents pixel differences between the current block of the video data and a predictive block of the video data; and reconstructing, by the video decoder, the current block of the video data based on the residual data and the predictive block of the video data. In some examples, performing a pass of the multi-pass non-separable inverse transformation includes performing a plurality of Givens orthogonal transformations.

Abstract: A wireless communication device (UE) may perform wireless communications according to at least a first radio access technology (RAT) and a second RAT. When the UE is in idle-mode, it may determine whether to bias a RAT selection policy—which may be presently favoring networks operating according to the first RAT—towards the second RAT, based on one or more of the following: first information indicative of a quality of previous wireless communications performed by the UE according to the first RAT, second information indicative of an overall quality of wireless communications performed according to the first RAT by the UE over a specified period of time at a specific location where the UE is presently located, and/or third information indicative of an overall quality of wireless communications performed according to the first RAT by other UEs presently located at the specific location.

Abstract: A method includes decoding a low-band portion of an encoded mid signal to generate a decoded low-band mid signal. The method also includes processing the decoded low-band mid signal to generate a low-band residual prediction signal and generating a low-band left channel and a low-band right channel based partially on the decoded low-band mid signal and the low-band residual prediction signal. The method further includes decoding a high-band portion of the encoded mid signal to generate a time-domain decoded high-band mid signal and processing the time-domain decoded high-band mid signal to generate a time-domain high-band residual prediction signal. The method also includes generating a high-band left channel and a high-band right channel based on the time-domain decoded high-band mid signal and the time-domain high-band residual prediction signal.

Abstract: An apparatus for encoding a multi-channel signal including at least two channels includes a time-spectral converter for converting sequences of blocks of sampling values of the at least two channels into a frequency domain representation having sequences of blocks of spectral values for the at least two channels; a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values to obtain at least one result sequence of blocks of spectral values including information related to the at least two channels; a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation including an output sequence of blocks of sampling values; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: An audio apparatus and an audio providing method thereof are provided. The audio providing method includes receiving an audio signal including a plurality of channels, applying an audio signal having a channel, from among the plurality of channels, giving a sense of elevation to a filter to generate a plurality of virtual audio signals to be respectively output to a plurality of speakers, applying a combination gain value and a delay value to the plurality of virtual audio signals so that the plurality of virtual audio signals respectively output through the plurality of speakers form a sound field having a plane wave, and respectively outputting the plurality of virtual audio signals, to which the combination gain value and the delay value are applied, through the plurality of speakers. The filter processes the audio signal to have a sense of elevation.

Abstract: Data streams may be stored in a chunk store in the form of stream maps and data chunks. Data chunks corresponding to a data stream may be stored in a chunk container, and a stream map corresponding to the data stream may point to the data chunks in the chunk container. Multiple stream maps may be stored in a stream container, and may point to the data chunks in the chunk container in a manner that duplicate data chunks are not present. Techniques are provided herein for localizing the storage of related data chunks in such chunk containers, for locating data chunks stored in chunk containers, for storing data streams in chunk stores in localized manners that enhance locality and decrease defragmentation, and for reorganizing stored data streams in chunks stores.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: Input audio data, including first microphone audio signals and second microphone audio signals output by a pair of coincident, vertically-stacked directional microphones, may be received. An azimuthal angle corresponding to a sound source location may be determined, based at least in part on an intensity difference between the first microphone audio signals and the second microphone audio signals. An elevation angle corresponding to a sound source location may be determined, based at least in part on a temporal difference between the first microphone audio signals and the second microphone audio signals. Output audio data, including at least one audio object corresponding to a sound source, may be generated. The audio object may include audio object signals and associated audio object metadata. The audio object metadata may include at least audio object location data corresponding to the sound source location.

Abstract: A method includes performing modifying, at a decoder, at least a portion of inter-channel phase difference (IPD) parameter values based on a mismatch value to generate modified IPD parameter values. The mismatch value is indicative of an amount of temporal misalignment between an encoder-side reference channel and an encoder-side target channel. The modified IPD parameter values are applied to a decoded frequency-domain mid channel during an up-mix operation.

Abstract: The present document discloses a method and apparatus for compensating for a lost frame in a transform domain, comprising: calculating frequency-domain coefficients of a current lost frame using frequency-domain coefficients of one or more frames prior to the current lost frame, and performing frequency-time transform to obtain an initially compensated signal; and performing waveform adjustment, to obtain a compensated signal. Alternatively, extrapolation is performed for all or part of frequency points of the current lost frame using phases and amplitudes of corresponding frequency points of a plurality of previous frames to obtain phases and amplitudes of the corresponding frequency points of the current lost frame, to obtain frequency-domain coefficients of the corresponding frequency points, and frequency-time transform is performed to obtain a compensated signal.

Abstract: An audio apparatus and an audio providing method thereof are provided. The audio providing method includes receiving an audio signal including a plurality of channels, applying an audio signal having a channel, from among the plurality of channels, giving a sense of elevation to a filter to generate a plurality of virtual audio signals to be respectively output to a plurality of speakers, applying a combination gain value and a delay value to the plurality of virtual audio signals so that the plurality of virtual audio signals respectively output through the plurality of speakers form a sound field having a plane wave, and respectively outputting the plurality of virtual audio signals, to which the combination gain value and the delay value are applied, through the plurality of speakers. The filter processes the audio signal to have a sense of elevation.

Abstract: A sound effect configuration method and a mobile terminal are provided. The method comprises: parameters of sound effect are loaded in a running memory of the mobile terminal; responsive to detection of a playing control instruction for an audio stream of a target application, it is determined whether the parameters of sound effect loaded in the running memory comprise a parameter of sound effect corresponding to the target application; responsive to determining that the parameters of sound effect loaded in the running memory comprise the parameter of sound effect corresponding to the target application, the parameter of sound effect corresponding to the target application is read from the running memory; and the audio stream of the target application is configured by virtue of the parameter of sound effect.

Abstract: Systems and methods to provide a response to advertising determine and/or obtain a commercial good that is related to an advertisement presented to a user, and further determine and/or obtain a commercial source for that commercial good. Information regarding the particular commercial good and/or the corresponding commercial source is presented to the user, for example using a graphical user interface. The user may subsequently be able to buy the commercial good from the commercial source.

Abstract: Systems and methods to guide a user response to advertising determine and/or obtain information identifying a commercial good that is related to an advertisement presented to a user, and further determine and/or obtain commercial sources for that commercial good. Information regarding the particular commercial sources is used to select a commercial source. Information regarding the particular commercial good and the selected commercial source is presented to the user, for example using a graphical user interface. The user may subsequently be able to buy the commercial good from the selected commercial source.

Abstract: There is provided a signal processing device, comprising: a frequency detecting means that detects a frequency satisfying a predetermined condition from an audio signal; an offset means that gives an offset to the detected frequency by the frequency detecting means in accordance with a frequency property at the detected frequency or around the detected frequency; a reference signal generating means that generates a reference signal by extracting a signal from the audio signal based on the detected frequency offset by the offset means; an interpolation signal generating means that generates an interpolation signal based on the generated reference signal; and a signal synthesizing means that performs high band interpolation by synthesizing the generated interpolation signal and the audio signal.

Abstract: An apparatus for generating an enhanced signal from an input signal, wherein the enhanced signal has spectral values for an enhancement spectral region, the spectral values for the enhancement spectral regions not being contained in the input signal, includes a mapper for mapping a source spectral region of the input signal to a target region in the enhancement spectral region, the source spectral region including a noise-filling region; and a noise filler configured for generating first noise values for the noise-filling region in the source spectral region of the input signal and for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from the first noise values or for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from first noise values in the source region.

Abstract: Embodiments of the present application provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.

Abstract: One embodiment of the present invention relates to a method whereby a device-to-device (D2D) terminal transmits a discovery signal in a wireless communication system, the discovery signal transmission method comprising the steps of: determining the size of a subperiod in a discovery period on the basis of a buffer size; and repeatedly transmitting the discovery signal in the subperiod corresponding to the determined size by using a hopping pattern applicable to the discovery period.

Abstract: An audio encoding apparatus to encode an audio signal using lossless coding or lossy coding and an audio decoding apparatus to decode an encoded audio signal are disclosed. An audio encoding apparatus according to an exemplary embodiment may include an input signal type determination unit to determine a type of an input signal based on characteristics of the input signal, a residual signal generation unit to generate a residual signal based on an output signal from the input signal type determination unit, and a coding unit to perform lossless coding or lossy coding using the residual signal.

Abstract: Context adaptive binary arithmetic coding (CABAC) techniques are generally described. Aspects of the techniques are generally directed to inheritance-based context initialization. An example video coding device includes a memory configured to store video data, and one or more processors. The processor(s) are configured to initialize context information for a current slice of a current picture by inheriting context information of a previously-coded block of a previously-coded picture of the stored video data as initialized context information for the current slice of the current picture. The processor(s) are further configured to code data of the current slice using the initialized context information.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: For playing an interactive audiovisual movie, stored in a data store of a computerized playback device is a movie episode data object which comprises a plurality of audio/video segments, separate audio tracks, and playback instructions. The playback device retrieves the playback instructions and, in accordance with the playback instructions, retrieves and plays audio/video segments, generates interface elements for receiving from a user decision commands while playing the audio/video segments, and retrieves and plays further audio/video segments depending on the user decision commands. The playback device further retrieves one of the separate audio tracks and plays the separate audio track uninterruptedly while playing an end of a first audio/video segment and a start of a second audio/video segment which is played subsequently to the first audio/video segment. Thus, an audio track is played uninterruptedly across the boundary of two consecutive audio/video segments.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

Abstract: A stereo sound encoding method and system, for encoding left and right channels of a stereo sound signal, down mix the left and right channels of the stereo sound signal to produce primary and secondary channels and encode the primary and secondary channels. Encoding the primary channel and encoding the secondary channel comprise determining a first bit budget to encode the primary channel and a second bit budget to encode the secondary channel. If the second bit budget is sufficient, the secondary channel is encoded using a four subframes model and, if the second bit budget is insufficient for using the four subframes model, the secondary channel is encoded using a two subframes model.

Abstract: A sound wave-based data sending method, apparatus, and electronic device, a sound wave-based data receiving method, apparatus, and electronic device, and a sound wave-based data transmission system. The sound wave-based data sending method includes: obtaining to-be-sent data; obtaining a sound wave frequency sequence corresponding to the to-be-sent data according to a preset sound wave frequency mapping table; generating a filtered sound wave signal sequence according to the sound wave frequency sequence and a preset sending duration, the sending duration being sending duration of a sound wave signal corresponding to a single character; and sending a sound wave signal corresponding to the to-be-sent data according to the filtered sound wave signal sequence. The method, apparatus, and electronic device of the present application provide an improved communication distance and identification accuracy.

Abstract: It is presented a method for managing a jitter buffer depth for receiving real-time communication. The method is performed in a receiver and comprises the steps of: determining an adaptive bitrate state of the receiver when a current capacity of a communication channel for receiving the real-time communication is below a maximum bitrate for receiving the real-time communication; and increasing a depth of a jitter buffer for receiving the real-time communication when the adaptive bitrate state is determined.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: According to the present disclosure, lighting sources having operating parameter(s) which is controllable in lighting sequence(s) as a function of a time code data set coupled with the sequence, are controlled in cooperation with virtual/augmented reality device(s), by: providing a repository of operating data files for the sources, coupled with the lighting sources, with each data file including time code data set(s) for lighting sequence(s) for a source among the lighting sources, retrieving in the data repository operating data file(s) coupled with a selected one of the lighting sources, and detecting a virtual/augmented reality signal from the virtual/augmented reality device(s), indicative of the fact that the virtual/augmented reality device is gazing at and/or is recognizing the selected lighting source, and operating the selected lighting source by controlling the operating parameter(s) as a function of the operating data included in the operating data file retrieved and of the virtual/augmented reali

Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

Abstract: A transmitting device comprising a transmit buffer for buffering a plurality of packets representing a live media stream, the packets having an order in the media stream from oldest to most recent. The transmitting device further comprising a transmitter for transmitting the packets from the buffer live over a network; and a controller arranged to measure an amount of data buffered for transmission in the transmit buffer, and to drop or compress the oldest packet or a predetermined number of the oldest packets on condition that the amount of data buffered for transmission exceeds or is likely to exceed a predetermined threshold.

Abstract: An audio decoder and method for providing a decoded audio information on the basis of an encoded audio information are disclosed. In one example, the audio decoder includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Abstract: According to the present disclosure, lighting sources, having operating parameter(s) which is controllable in lighting sequence(s) as a function of a time code data set coupled with the sequence, are controlled: by providing a repository of operating data files for the lighting sources coupled with lighting sources, with each data file including time code data set(s) for lighting sequence(s) for one of the lighting sources, by detecting ambient lighting level signal(s) at the ambient where the lighting sources are located, by retrieving in the data repository operating data file(s) coupled with a selected one of the lighting sources, and by operating the selected lighting source by controlling the operating parameter(s) as a function of the operating data included in the operating data file retrieved and as a function of the ambient lighting level signal.

Abstract: Systems and methods for low-power single-wire communication are provided. In some embodiments, a method of operation of a transmitter to transmit a data word to a receiver using low-power single-wire communication includes receiving the data word to be transmitted to the receiver. The method also includes encoding the data word to be transmitted in a Pulsed Index Communication (PIC) format to produce a PIC data word and transmitting the PIC data word to the receiver. In this way, the transmitter may be able to transmit an increased amount of data while maintaining a simple communication protocol that uses low power and does not require a Clock-Data Recovery circuit.