Abstract: A system for real time transmission of speech audio received from a text-to-speech (TTS) engine can include a TTS engine and a real time speech audio producer for receiving speech audio from the TTS engine over a network, and for producing formatted audio packets for transmission over the network according to the transmission interval. The transmission interval can be determined according to a packetization delay parameter.

Abstract: If there is a limitation imposed to the time period within which recording of certain content such as broadcast content is permitted, with the purpose of protection of copyrights of such content, a relationship between the accessible time and a current time is indicated so that a user can be informed of the remaining time before the deadline or the elapsed time since the deadline. In this way, the user can verify how a system is currently working to manage recording operation regarding the information having restricted or limited accessible time. In addition, automatic reproduction of digital material performed immediately before the deadline can help the user to avoid careless failure to access the provided content.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A fast code length search method for determining the length of a code in a codebook, wherein the method is especially suited for MPEG-compliant audio encoding. A code length table is created which stores pre-calculated code lengths, including any sign bits and linear extension bits necessary, for data value pairs or quadruples. In one embodiment, two code length tables are created, one for determining the code lengths of the codes used for the ones region, and a second code length table for the big values region. When a code length determination is made, the value is simply read from the table, instead of being calculated each time.

Abstract: Improved channel code configurations for use in transmission of digital audio or other types of information in a digital communication system. The channel code may include an outer channel code, e.g., a cyclic redundancy code (CRC), and an inner channel code, e.g., a complementary punctured pair convolutional (CPPC) code. In accordance with the invention, multiple code words of the outer code are associated with a given packet of the digital information, in accordance with a particular outer code configuration, so as to provide partial error flagging for different portions of the given packet. An information encoder, e.g., a PAC encoder, interacts with an outer code encoder to determine a bit allocation for transmission of packets at a particular bit rate, based at least in part on the outer code configuration.

Abstract: A method for transferring audio data and audio-related information includes generating second audio data from first audio data, transmitting second audio data and audio-related information associated with the second audio data, and receiving the second audio data and audio-related information which includes information on a sampling frequency of the first audio data.

Abstract: Preprocessing audio data to generate parameters associated with time scaling reduces the processing power required for real-time time scaling of the audio data. An augmented audio data structure includes the audio data and the parameters. The parameters for a frame of the audio data can identify best match blocks for time scaling or represent a plot of offset versus time scale that can be interpolated to determine an offset. The real-time time scaling uses the blocks that the parameters identify instead of performing a search for the best matching blocks. The parameters can also indicate which of the frames represent silence and can be scaled differently from frames that do not represent silence.

Abstract: A time scaling process for a multi-channel (e.g., stereo) audio signal uses a common time offsets for all channels and thereby avoids fluctuation in the apparent location of a sound source. In the time scaling process, common time offsets correspond to respective time intervals of the audio signal. Data for each audio channel is partitioned into frames corresponding to the time intervals, and all frames corresponding to the same interval use the same common time offset in the time scaling process. The common time offset for an interval can be derived from channel data collectively or from separate time offsets independently calculated for the separate channels. Preprocessing can calculate the common time offsets for inclusion in an augmented audio data structure that a low-processing-power presentation system uses for real-time time scaling operations.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: An apparatus and method for compression and expansion of a wave signal on a time axis. A memory device stores waveform data representative of a waveform for each sub-frequency band of each main-frequency band of a wave signal, in which the wave signal is divided into a plurality of the main-frequency bands, each of the main-frequency bands is divided into a plurality of the sub-frequency bands. A plurality of time axis compression and expansion devices are provided for each of the sub-frequency bands for performing time axis compression and expansion of the waveform. A mixing device mixes signals provided from the time axis compression and expansion devices.

Abstract: A device for detecting a predetermined waveform in a received signal and synchronizing the detected waveform to the predetermined waveform is disclosed. The device includes a memory element for storing a reference set of encoded values. The reference set representing an encoded version of the predetermined waveform. An encoder is used to PCM encode the signal to obtain sets of encoded values representing the received signal. A processor calculates the statistical correlation coefficient of the reference set and the signal sets. The processor then determines the maximum statistical correlation coefficient. The predetermined waveform is detected in the signal if the maximum statistical correlation coefficient is greater than or equal to a predetermined threshold value. The device provides a compact, inexpensive, and fast method for detecting a known reference waveform in a received signal.

Abstract: A fast codebook search method for finding an optimal Huffman codebook from a group of Huffman codebooks, wherein the method is especially suited for MPEG-compliant audio encoding. In order to select an optimal codebook from among candidate codebooks for a given sub-region, a bit difference table is created, which for any given data pair contains a bit difference value. The bit difference value is the difference between the number of bits needed for a given data pair (or quadruple) in a first candidate codebook and a second candidate codebook [N bits−M bits]. By summing all such bit difference values for the data samples in a given sub-region, a quick determination can be made as to which codebook would encode the sub-region using the fewest bits (based on the size and/or sign of the sum(s)). For sub-regions having three candidate codebooks, two bit difference sums are calculated.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A method and a system is provided for embedding and detecting additional information, such as copyright information, in audio data, so that a modification in the sonic quality due to the embedding is imperceptible to human beings, and does not drastically deteriorate the sonic quality.

Abstract: A method and apparatus for synchronizing audio to an asynchronous clock while preserving pitch utilizes a phase-vocoder to implement time-scaling without pitch-shifting.

Abstract: A time-scale modification method or apparatus performs time-scale modification (i.e., compression or expansion with respect to time) on original audio signals having waveforms. Adjacent wave segments are divided and cut from the waves of the original audio signals by various lengths. A certain number of samples are thinned out from each of the adjacent waveform segments to provide a reduced amount of data. Calculations are performed on the reduced amount of data to sequentially produce similarities between the adjacent wave segments in response to the various lengths. The similarities are evaluated to determine a length that provides a best similarity within the various lengths as a basic period. The waves of the original audio signals are divided and cut into two waves by the basic period. Time-scale modification is effected on the two waves to produce a mixed wave.

Abstract: A sound signal processing apparatus including extracting means for extracting from a composite sound signal, representing multiple sound sequences, digital sound signals corresponding to a portion of the composite sound signal. Each of the digital sound signals is individually sampled. Also included is a signal converter for converting the digital sound signals which have been extracted into analog sound signals.

Abstract: A proxy server transcodes audio data on behalf of a client computer prior to transmitting the audio data to the client computer. In response to a request from the client computer, the proxy server obtains the audio data from a remote server. If the proxy server determines that transcoding is appropriate, the proxy server transcodes the audio data and transmits the transcoded audio data to the client computer. Transcoding can include changing the type of the audio file, compressing the audio data, reducing the number of audio channels, or reducing the data sampling rate. The proxy server determines the extent and type of transcoding based on file formats that the client computer can process, the size of the requested audio file, the memory capacity of the client computer, the bandwidth of the connection between the proxy server and the client computer, and the desired level of audio quality.

Abstract: The proceedings of a business meeting are recorded on a portable digital audio recorder. The resulting digitized speech signals are uploaded to a personal computer which is part of a data communication network. Users of the network may arrange to access the uploaded digitized speech signals for downloading to the user's location. The downloaded digitized speech signals are audibly reproduced at the location of the requesting user.

Abstract: The disclosed method and apparatus controls the rate of playback of audio data corresponding to a stream of speech. Using speech recognition, the rate of speech of the audio data is determined. The determined rate of speech is compared to a target rate. Based on the comparison, the playback rate is adjusted, i.e. increased or decreased, to match the target rate.

Abstract: A system and method for detecting beats in a compressed audio domain is disclosed where a beat detector functions as part of an error concealment system in an audio decoding section used in audio information transfer and audio download-streaming system terminal devices such as mobile phones. The beat detector includes a MDCT coefficient extractor, a band feature value analyzer, a confidence score calculator; and a converging and storage unit. The method provides beat detection by means of beat information obtained using both MDCT coefficients as well as window-switching information. A baseline beat position is determined using MDCT coefficients obtained from the audio bitstream which also provides a window-switching pattern. A window-switching beat position is compared with the baseline beat position and, if a predetermined condition is satisfied, the window-switching beat position is validated as a detected beat.

Abstract: A time-axis compression/expansion system compresses or expands a multichannel signal at a specified compression/expansion rate. Waveform segments are sequentially cut out from each channel signal. A cutting starting point of a leading end portion of a waveform segment following each preceding waveform segment of the cut out waveform segments is determined commonly between the channel signals, based on two portions of a waveform of a synthesized signal formed by synthesizing the channel signals within a range of predetermined search parameters of the waveform of the synthesized signal. The two portions correspond to a time period over which cross-fading is to be carried out and are most similar to each other. The preceding waveform segment and the following waveform segment cut from each channel signal are spliced together by cross-fading a trailing end portion of the preceding waveform segment and the leading end portion of the following waveform segment.

Abstract: Described is a method of compressing an analogue signal, e.g. a voice signal, the signal function being continuously sampled, quantized and encoded into data words, the difference between each two successive data words determined and each difference value quantized and encoded.

Abstract: A lossless decoding method that restores losslessly compression encoded audio data on a real-time basis. The encoded audio data is divided into first data having a data amount exceeding the maximum bitrate and second data having a data amount less than the maximum bitrate, where the first data is divided into third data being the encoded audio data having a data amount of the maximum bitrate and fourth data being the encoded data of the portion exceeding the maximum bitrate, and where the encoded audio data is output so that the fourth data is output together with the second data. The method includes buffering the fourth data in a buffer, combining the buffered fourth data with the corresponding third data, and outputting the combined third and fourth data as restored audio data.

Abstract: Media encoding, transmission, and playback processes and structures employ a multi-channel architecture with different audio channels corresponding to different playback rates for a presentation to be transmitted over a network. Audio frames in the various audio channels all correspond to the same amount of time in the original presentation and have frame indexes that identify in the different audio channels the frames corresponding to the same time interval in the presentation. A user can make a real-time change in playback rate causing selection of a channel corresponding to the new playback rate and a frame required for prompt and smooth transition in the playback rate of the presentation. The architecture can additionally provide channels for graphics data such as image data that are displayed according to the index of the audio, and different audio channels with the same playback rate but different compression schemes for use according to available bandwidth on the network.

Abstract: A method and apparatus for balancing the forward link capacity of a wireless communication system with the reverse link capacity of the system is presented. Speech coders with selectable modes are implemented in both links so that a forward link speech coder will not be operating with a mode set that is identical to the mode set used by a reverse link speech coder. Since the reverse link has a higher user capacity than a forward link, the reverse link speech coder can operate with a higher average data rate. Hence, the mode set used by the reverse link speech coder can be implemented without low average data rate modes. Elimination of modes from the mode set reduces complexity of the speech coder.

Abstract: A lossless encoding apparatus encodes audio data and a lossless decoding apparatus restores the losslessly compression encoded audio data on a real-time basis, and a method therefor. The lossless encoding apparatus includes a lossless compression unit which losslessly compression encodes the audio data stored in an input buffer in units of predetermined data and outputs the encoded data in sequence, and an output buffer which stores the encoded audio data output from the lossless compression unit.

Abstract: A lossless decoding apparatus restores the losslessly compression encoded audio data on a real-time basis, and a method therefor. The lossless decoding apparatus includes a restorer which restores losslessly compression encoded audio data. The encoded audio data has been divided into first data having a data amount exceeding the maximum bitrate and second data having a data amount less than the maximum bitrate, where the first data is divided into third data being the encoded audio data having a data amount of the maximum bitrate and fourth data being the encoded data of the portion exceeding the maximum bitrate, and where the fourth data is output together with the second data. A bitrate controller controls the input and output of the fourth data into or out of a buffer to combine the fourth data with the corresponding third data to be output to the restorer to be restored with the remaining restored audio data in sequence.

Abstract: An audio decoding device for decoding audio information with multiple channels includes a coded information memory section for storing the coded audio information; an information transmission section for reading the coded audio information stored at an arbitrary position in the coded information memory section; and an audio decoding section for decoding the coded audio information read by the information transmission section and outputting the resultant audio information in accordance with a time axis, wherein the information transmission section includes a buffer memory for retaining an address of an actual pointer for reading the coded audio information in the coded information memory section so as not to be reread, an address of a temporary pointer for reading the coded audio information in the coded information memory section so as to be reread, actual pointer data read by the actual pointer, and temporary pointer data read by the temporary pointer.

Abstract: A wide band audio signal encoding apparatus, a wide band audio signal decoding apparatus, a wide band audio signal encoding and decoding apparatus and a wide band audio signal recording medium, each having a low bit rate, a wide band, a low distortion factor and a wide dynamic range are provided. Wide band audio data is divided into signal data of a predetermined natural number N of sub-bands, and the number of bits for quantization for sub-sampling is determined based on noise floor information of the above wide band audio data. The signal data of the N sub-bands are sub-sampled by the respective numbers of bits for quantization, and encoded data obtained by multiplexing the signal data of the sub-sampled N sub-bands are recorded on a wide band audio signal recording medium. Therefore, both he characteristics of an extremely wide dynamic range and a wide band can be concurrently achieved at a relatively low bit rate per channel.

Abstract: A speech encoding system for use with a digital cellular communication device and a receiving station, includes a mechanism for determining whether a voice communications packet needs to be treated as a data communications packet; a voice recognition mechanism for receiving instructions by voice command; and a control mechanism for responding to said voice command and controlling a controlled entity. A method for encoding a voice command generated on a digital cellular communication device and transmitted over a wireless communication network to a receiving station for controlling a controllable entity includes recognizing a voice command; determining whether the voice command needs to be treated as a data communications packet; encoding the voice command; connecting the voice command to a voice recognition mechanism; and controlling a controlled entity with the voice command.

Abstract: An audio decoding device for decompressing an audio signal that was compressed in accordance with a given compression method having a program-controlled signal processor which receives the compressed audio signal and from it produces a decompressed audio signal under the control of a decompression program; a loadable program memory that is connected to the signal processor for storing the decompression program; and a management device which is connected to the program memory and is controlled by the compressed audio signal, and which manages decompression programs corresponding to at least two different compression methods in order to determine the respectively used compression method from the compressed audio signal, to select the pertinent decompression program and to load the pertinent decompression program into the program memory.

Abstract: An apparatus and method is disclosed for converting an input signal having frequency related information sustained over a first duration of time into an output signal sustained over a second duration of time at substantially the same first frequency by adding or subtracting to the effective wave length of the output signal. Preferably, the signals are converted in digital form with samples added or subtracted to frequency convert the signal.

Abstract: Loss-less data compression/decompression especially useful in a limited resource environment such as a handheld portable video game system allows graphics and/or attribute data to be efficiently and quickly decompressed on an as-needed basis in real time response to interactive user inputs. A two-level run-length-encoding is used to encode redundant patterns and redundant symbols. A common sentinel field format encodes whether data following the field is non-redundant data, a symbol run, or a pattern run. Compression ratios of 60% for representative symbol-mapped video display graphics/attribute files can be achieved.

Abstract: This invention comprises a two part audio system in which all of the processing power is allocated to a small, lightweight satellite part that is the face unit. Mass storage, amplification, and wired power for recharging the face unit is provided by the other part, the base. The base runs either from a 120 volts AC source or 12 volts DC. The face unit contains a small amount of flash memory making it capable of carrying music normally stored on two or more compact discs (CDS).

Abstract: An audio information bit stream including audio control bits and audio data bits is processed for transmission in a communication system. The audio data bits are first separated into n classes based on error sensitivity, that is, the impact of errors in particular audio data bits on perceived quality of an audio signal reconstructed from the transmission. Each of the n different classes of audio data bits is then provided with a corresponding one of n different levels of error protection, where n is greater than or equal to two. The invention thereby matches error protection for the audio data bits to source and channel error sensitivity. The audio control bits may be transmitted independently of the audio data bits, using an additional level of error protection higher than that used for any of the n classes of the audio data bits. Alternatively, the control bits may be combined with one of the n classes of audio data bits and provided with the highest of the n levels of error protection.

Abstract: The present invention comprises methods and systems for a dynamic audio processor for processing an audio signal prior to encoding. The audio signal is pre-processed to provide analog-to-analog modification that creates a preferred analog format that is effectively and efficiently ready to be converted into a digital data stream by an encoder, such as an A-to-D converter or codec. One possible application includes, for example, a web streamer that provides digital data streams from a server. The inventive arrangements are preferably carried by functional stand-alone components, integral components of a PC or other computing platform, or carried as part of the encoder. Preferably, the pre-processing comprises receiving the audio signal, nominalizing the signal to provide a level input, and compressing and equalizing the signal for outputting thereof.

Abstract: A lossless encoding apparatus encodes audio data and a lossless decoding apparatus restores the losslessly compression encoded audio data on a real-time basis, and a method therefor. The lossless encoding apparatus includes a lossless compression unit which losslessly compression encodes the audio data stored in an input buffer in units of predetermined data and outputs the encoded data in sequence, and an output buffer which stores the encoded audio data output from the lossless compression unit.

Abstract: In a speech communications network, continuous play of audio packets is achieved using a jitter buffer in a receiver. Audio packets are stored in the jitter buffer before decoding the audio packets into an audible output. When the level of stored audio packets approaches the full capacity of the jitter buffer, the rate at which the audio packets are played out of the jitter buffer is increased signaling a compression operation in the decoder. When the level of stored audio packets approaches an empty level of the jitter buffer, the rate which the audio packets are played out of the jitter buffer is reduced signaling an expansion operation in the decoder. Audio packets are not modified when the level of stored audio packets is within a predetermined range. A speed controller is provided to instruct the decoder to decode the audio packets according to either a compressed, expanded or normal audio packet status.

Abstract: A solid-state audio recording unit, capable of checking whether normal audio recording is performed or not on a real-time basis, includes one input buffer for receiving incoming audio data operating at a standard speed, another input buffer for writing audio data into a memory 9 operating at a high speed, one output buffer for receiving audio data from the memory 9 operating at a high speed, and another output buffer for delivering audio data as output operating at a standard speed. As such, it is possible to write/read data into/from the memory at a high speed; and thus operation is ensured enabling, on appearance, parallel processing of input and output, even though, in reality, a single memory is shared by the input and output ports; and thus it becomes possible to deliver audio data stored in a memory as an output on a real-time basis.

Abstract: A processing system retrieves selected compressed audio data from a compact disc and produces sound based on the retrieved data. In this regard, the processing system utilizes a disc storage mechanism, a processing element, a system manager, and an audio output device. A compact disc is inserted into the disc storage mechanism. The system manager receives inputs from a user and, in response to the user inputs, reads organizational structure information stored on the compact disc. Based on the inputs and the organizational structure information, the system manager identifies a set of compressed data stored on the compact disc. The system manager retrieves instructions of a decompression application from the compact disc and transmits these instructions to the processing element. The processing element executes the received instructions, thereby decompressing the set of compressed data into uncompressed data.

Abstract: Entropy encoding and decoding of data with a code book containing variable length entropy-type codes that are assigned to variable length input symbol groupings. The variable length input sequences are identified by scanning an input channel, such as a live broadcast, non-volatile data storage, or network connection (e.g., LAN, WAN, Internet). Each time a symbol grouping is recognized, a corresponding entropy-type code is output as a replacement for the input stream. Decoding is the inverse process of encoding, where a code word is looked up in the code book and the corresponding original input is obtained.

Abstract: An embodiment of the present invention is a method for generating a listener-interest-filled work for an audio or audio-visual work, which method comprises the steps of: (a) generating one or more average speed contours for one or more audio or audio-visual works for one or more categories of users; (b) converting the one or more average speed contours to one or more conceptual speed association data structures; and (c) forming a listener-interest-filtered conceptual speed association data structure from the one or more conceptual speed association data structures.

Abstract: In a communications system, multi-rate coding in accordance with the invention is implemented to generate multiple representations of an audio signal at different rates. These representations contain equivalent and/or various amounts of audio information. In an illustrative embodiment, at least one of the representations is a core representation containing core audio information. The remaining representations are enhancement representations containing enhancement audio information. The core representation is necessary for recovering the audio signal with minimal acceptable quality. Such quality is enhanced when the core representation, together with one or more of the enhancement representations, is used to recover the audio signal.

Abstract: A rounding method to increase the precision of an audio decoder during arithmetic and/or shifting operations is disclosed. The most significant bit of the discarded bits is evaluated for a rounding up operation.

Abstract: The invention enables the apparent display rate of an audiovisual display to be varied. The invention can modify an original set of audio data in accordance with a target display rate, then modify a related original set of video data to conform to the modifications made to the audio data set, such that the modified audio and video data sets are synchronized. When the modified audio and video data sets so produced are used to generate an audiovisual display, the audiovisual display has an apparent display rate that approximates the target display rate. The target display rate can be faster or slower than a normal display rate at which an audiovisual display system generates an audiovisual display from the original sets of audio and video data. The target display rate can be established solely by a user instruction, by analysis of the audiovisual data, or by modification of a user-specified nominal target display rate based upon analysis of the audiovisual data.

Abstract: A data conversion device is provided for storing digital data in a DAT (332) at a 16-bit word length and then recovering the data at a 24-bit word length with an overall reduction in truncation noise that would be inherently associated with data at the 16-bit word length This is facilitated by noise shaping the data at the 16-bit word length prior to storage in the DAT (332) with a noise-shaping filter (324) This results in truncation noise in the lower portion of the frequency band being shifted to the higher portion of the band When the data is recovered, it is converted to a 24-bit word length and then processed through a bandpass filter to filter out the higher frequency noise to yield a signal that has a maximum noise equal to or less than that in the lower portion of the band stored in the DAT (332) Since the truncation noise was shifted from the lower band to the upper band, this is a lower noise level than that inherently associated with the 16-bit word length.

Abstract: A data conversion device is provided for storing digital data in a DAT (332) at a 16-bit word length and then recovering the data at a 24-bit word length with an overall reduction in truncation noise that would be inherently associated with data at the 16-bit word length. This is facilitated by noise shaping the data at the 16-bit word length prior to storage in the DAT (332) with a noise-shaping filter (324). This results in truncation noise in the lower portion of the frequency band being shifted to the higher portion of the band. When the data is recovered, it is converted to a 24-bit word length and then processed through a bandpass filter to filter out the higher frequency noise to yield a signal that has a maximum noise equal to or less than that in the lower portion of the band stored in the DAT (332). Since the truncation noise was shifted from the lower band to the upper band, this is a lower noise level than that inherently associated with the 16-bit word length.

Abstract: A method and apparatus for decoding a bitstream (100) of transform coded multi-channel audio data. The bitstream is subjected to a block decoding process (101) to obtain for each input audio channel within the multi-channel audio data a corresponding block of frequency coefficients (102). Each block of frequency coefficients (102) is assigned a higher precision inverse transform or a lower precision inverse transform according to predetermined characteristics of the audio data represented by the block. The blocks of frequency coefficients are subsequently subjected to the assigned transform (105, 106) and an output audio signal (108) is generated in response to each of the higher and lower precision inverse transform processes.

Abstract: The present invention relates to a system and method for automatic synchronization in multimedia presentations. According to an embodiment of the present invention, when a data stream is compressed, delay which would normally be compressed out is replaced by a delay token which indicates a length of time of the delay. When a data stream is decompressed and presented, the delay tokens may either be used or ignored. In particular, when data streams are presented together in a multimedia presentation the delay tokens may be used to synchronize the various data streams of the multimedia presentation. Otherwise, when data streams are presented alone without the other data streams of a multimedia presentation or are not part of a multimedia presentation, the delay tokens may be ignored. In such cases when the delay token is ignored, any data stream delay is simply skipped since there is no need to synchronize with other data streams.