162 results about "Digital Audio Tape" patented technology

A digital audio file search method and apparatus for digital audio files is provided that allows a user to navigate the audio files by generating speech sounds related to the information of the audio files to facilitate searching and playback. The digital audio file search method and apparatus searches for audio files in a portable digital audio player in combination with an automobile audio system through speech sounds by utilizing text-to-speech processing and by prompting response from a user in response to the generated speech sounds. The text-to-speech technology is utilized to generate the speech sound based on tag-data of the audio files. When hearing the speech sounds, the user gives instruction for searching the files without being distracted from driving the automobile.

A method and apparatus for audio compensation is disclosed. If audio input components and audio output components are not driven by a common clock (e.g., input and output systems are separated by a network, different clock signals in a single computer system), input and output sampling rates may differ. Also, network routing of the digital audio data may not be consistent. Both clock synchronization and routing considerations can affect the digital audio output. To compensate for the timing irregularities caused by clock synchronization differences and / or routing changes, the present invention adjusts periods of silence in the digital audio data being output. The present invention thereby provides an improved digital audio output.

A method is provided for synchronizing the playback of a digital audio broadcast on a plurality of network output devices by inserting a control track pulse in an audio stream of the digital audio broadcast. The method includes the steps of outputting a first control track pulse as part of an audio signal which has unique identifying characteristics and is regularly occurring, outputting a second control track pulse so that the time between the first and second control track pulses must be significantly greater than the latency between sending and receiving devices, and coordinating play of audio at the time of the occurrence of the transmission of the second control track pulse assuring the simultaneous output of the audio signal from multiple devices. The control track pulses have a value unique from any other portion of the audio stream. The digital audio broadcast from multiple receivers does not present to a listener any audible delay or echo effect.

An interactive television program guide is provided. The interactive television program guide provides a user with the opportunity to select a language for playing television programming and displaying program guide text. Television program audio in the desired language may be obtained from a SAP or digital audio track and played in the selected language. Television related information in the desired language may be obtained from a digital track. If television program audio or related information is not provided in the selected language, the program guide may use a default language. The program guide may coordinate program guide display screen text with languages available for television programs when the programs are broadcast to users.

Circuitry for transferring multiple digital data streams, e.g. digital audio data, over a single communications link such as a single wire. A pulse-length-modulator is responsive to a plurality of data streams to generate a series of data pulses with a single data pulse having a rising and falling edge in each of a plurality of transfer periods defined by a first clock signal. The timing of the rising and falling edge of each data pulse is dependent on a combination of the then current data samples from the plurality of data streams. The duration and position of the data pulse in the transfer window in effect defines a data symbol encoding the data. An interface receives the stream of data pulses, and data extraction circuitry samples the data pulse to determine which of the possible data symbols the pulse represents and determines a data value for at least one received data stream.

A server for a merchant computer system, the server comprising: a file store for storing a range of audio / video products in respective product files; a dialogue unit having a network connection and operable to invite and receive a client selection from among the products via the network connection; a product reader for reading the product files to generate a digital audio / video signal; a digital signal processing unit having an input connectable to receive the digital audio / video signal from the product reader, a processing core operable to apply a defined level of content degradation to the digital audio / video signal, and an output connected to output the degraded digital audio / video signal from the processing core to the network connection. It is therefore possible for a content provider to change the characteristics of an audio or video data stream supplied over a network to a potential purchaser in a controlled and variable manner. The amount of degradation is sufficient to enable a potential purchaser to appreciate the characteristics of the audio or video product, whilst reducing the perceived quality.

A configuration method is performed for controlling an audio apparatus having a main block configurable for processing audio signals such as modifying and mixing of the audio signals, and peripheral units of various types being equipped with components of various types and being connectable to the main block for inputting the audio signals to be processed and outputting the audio signals processed by the main block. A first detection step is carried out for detecting when a new peripheral unit is connected to the main block. A second detection step is carried out for detecting a type of the new peripheral unit. An inquiry step is conducted for inquiring the new peripheral unit upon detection thereof about a type of a component equipped in the new peripheral unit and obtaining a reply indicating the type from the new peripheral unit. Thus, the main block can be configured according to the detected type of the new peripheral unit and the type of the component thereof indicated by the reply.

A digital video processor and operating method for a receiving raw audio and video data representing images and sound to be played. The processor includes a demultiplexer for receiving the raw audio and video data and providing demultiplexed audio and video data to a memory. A first system time clock provides first time values in response to being continuously clocked by the demultiplexer. A CPU decodes and plays back the audio and video data as a function of the audio and video PTS values. The processor further includes a second system time clock providing second time values in response to being periodically incremented by the CPU. The CPU periodically sets the second system time clock to a second time value equal to a current first time value of the first system time clock in response to a standard play mode, and the CPU periodically sets the first system time clock to a first time value equal to a current second time value of the second system time clock in response to a trick play mode.

A system and method support the exchange of media between friends, family members, and 3rd party media providers over a closed and secure media exchange network. The media may include, for example, digital video, digital audio, digital images, digital data, or any form of digital information.

A digital receiver contains a local video clock that controls the video presentation speed and a local audio clock that controls the audio presentation speed separately and independently of the video presentation speed. To control the video presentation speed, a comparator compares a program clock reference in a video packet with a local time. A video clock controller then speeds up or slows down the video clock as needed to be back on schedule. To control the audio presentation speed, a comparator compares a program clock reference in an audio packet with the local time. An audio clock controller then speeds up or slows down the audio clock as needed to be back on schedule.

A system and method support the exchange of media between friends, family members, and 3rd party media providers over a closed and secure media exchange network. The media may include, for example, digital video, digital audio, digital images, digital data, or any form of digital information.

The present invention relates to a method and device for playing digital audio and video at variable speeds. The method first receives and stores the audio and video data transmitted to the local area, and then the device / network status and the audio and video data stored in the local area, or to the cache of the system, or to the system The cache and local audio and video data are processed and analyzed, and then the parameters of audio and video speed change are determined, so that the speed of audio and video data is changed, and then the audio and video data after speed change are processed synchronously, and finally the synchronously processed Audio and video data is output to playback devices or storage devices. The method can solve the problems of delay, intermittent, stuck and other problems encountered in the current network audio and video playback and the resulting out-of-sync problem of audio and video data without changing the current network conditions and increasing network costs.

The invention discloses a method for sea-land vector map data integration and fusion, which includes the steps of data pre-treatment, data format conversion, data integration display, same-name entity matching, same-name entity combination and the like. A digital audio tape (DAT) data format put forward by the method enables map control information, index data and position coordinate data to be packaged on one file and stored with layers as the unit. The storage format is convenient for management of layers, and when data of a corresponding layer is read, the data file only needs to be opened once, so that data reading speed of data is greatly improved, and simultaneously the size of the map data file is reduced to a certain degree. In addition, the same-name surface entity matching method based on simplification of polygon is capable of replacing all point entities forming a surface entity with four points or eight points obtained through simplification to participate in a matching process, thereby being capable of reducing complexity, improving matching rate, and simultaneously being capable of storing simplified results to be in the mode of files to be convenient for secondary use.

A cognitive loudspeaker system including a control station that communicates wirelessly and bi-directionally with a plurality of sound production stations. The control station and the sound production stations are initially synchronized to a conductor clock. Configuration information is then transmitted from the sound production stations to the control station. In response, the control station generates playback executables, which are transmitted to the sound production stations. The control station also transmits digital audio information to the sound production stations. Within each sound production station, the previously received playback executable is used to control the decoding and processing of the received digital audio information. Each sound production station generates digital audio output samples, which are converted to analog output signals. These analog output signals are amplified and are used to drive loudspeakers.

The invention uses digital signal processing (DSP) techniques to synchronize an audio encoding process with a video synchronization signal. Namely, the encoder parameters of a DSP microchip are preset according to characteristics of an audio frame. A buffer temporarily stores the audio frame prior to sending it to an encoder. The buffer then transfers the frame in response to receiving a video synchronization signal in conjunction with authorization from a microprocessor. As such, the encoding sequence of the audio frame coincides with the video synchronization signal. Since the corresponding video frame is already slaved to the video synchronization signal, the audio samples are effectively processed in sequence with the video data. Prior to outputting the encoded audio frame to a multiplexor, the encoder sends a value to the microprocessor representing the difference between the end of the encoded audio frame and a second video synchronization signal. Those audio samples are ultimately discarded from the bitstream. Thus, synchronization is achieved by beginning and effectively ending the encoding processes of both the audio and video data, respectively, in sequence with a common video synchronization clock.

A record wave pattern modeling method for a fault record device of an intelligent transformer substation belongs to the technical field of power supply systems, builds pattern configuration files by standardizing information pattern files of fault record wave equipment to supply uniform configuration patterns for automatic pattern building and generates intermediate-range nuclear forces (INF) patter files. When fault happens, a scheduling end acquires fault nitrification to obtain the fault record wave files containing control flow graph (CFG) configuration files generated in online mode, digital audio tape data files generated in online mode and INF pattern files from the fault record device, and the pattern information is utilized to extract channel information for multiple stations, multiple device and multiple files according to the same system fault to form virtual record wave files containing complete information of the fault. The record wave pattern modeling method for the fault record device of the intelligent transformer substation achieves treatment of a plurality of automatic fault record wave data based on storage and transmission of distributed type record wave pattern files of a transformer substation at the scheduling end, thereby greatly improving efficiency of fault diagnosis and fault analysis of power grids.

A wireless communication system and in particular to a wireless communication system for digital audio players that provides for increased functionality, such as communication, interaction and synchronization between a computing platform and various mobile, portable or fixed digital audio players, as well as providing a communication link between the various digital audio players themselves. The computing platform may act, for example, through a wireless network or wireless communication platform, to control the digital audio players; to act as a cache of digital audio data for the digital audio players; as well as provide a gateway to the Internet to enable the digital audio players to access additional digital audio content and other information. The computing platform may also be used to automatically update digital audio content on the digital audio players; synchronize digital audio content and playlists between digital audio players; and automatically continue a particular playlist as the user moves from one digital audio player to another.

Device includes a disc jockey mixing console with analog controls, crossfader and scratchpad and is used to mix audio tracks from compressed digital audio data sound recordings. One device substitutes for mixing console, turntables and records. For compressed digital audio data recordings, analog controls allow far superior manual dexterity for adjustment of volume and speed when mixing audio tracks as compared to digital mouse. Includes two audio outputs—a headphone and a main speaker output—having to analog convertors, analog controls in the form of knobs and sliders, a touch screen LCD panel for selecting and queuing songs and a computer with a processor, ROM storage means, RAM storage means, software and a hard disc to store audio tracks files. Optional interface between device and personal computer to upload songs, audio input and CD ROM drive for converting audio to compressed digital audio data.

A bi-phase decoder suitable for use in a broadcast router and an associated method for extracting subframes of digital audio data from a stream of digital audio data. Logical circuitry within the bi-phase decoder extracts subframes of the digital audio data by constructing a transition window from an estimated bit time, sampling the stream of digital audio data using a fast clock and applying the sampled stream of digital audio data to the transition window to identify transitions indicative of preambles of the subframes of digital audio data.

A store and forward receiver for a satellite communication system employs a multi-threaded command interpreter, and an associated reduced complexity audio control language (ACL) to define commands for controlling actions at different receiver sites, each of which is programmable for its own local programming purposes. Upon receipt of a relatively simple command from the headend, the interpreter accesses and executes an associated sequence of potentially locally unique, previously stored commands, causing performance of a sequence of actions, e.g., play back of potentially locally unique, previously stored information files, interleaved with portions of a headend-source commercial audio program being rebroadcast at the receiver site. This allows different affiliate stations to produce separate potentially locally unique complex sequences of actions from a single headend-sourced command for all receivers in a network grouping.

A gain control at the input monitors an input signal and a supply voltage, which drives an output. The gain control adjusts the gain to compress the input signal when the supply voltage decreases in magnitude and / or the input signal is of such magnitude to cause the supply voltage to decrease.

A portable media device (PMD) can produce an isochronous audio / video experience when the PMD provides a digital audio signal to the accessory while displaying analog video on an accessory-independent display. The accessory can communicate audio latency information to the PMD. The PMD can delay a video portion of a presentation, relative to providing the digital audio signal to the accessory, based on the audio latency information communicated by the accessory. As a result, the user may perceive an isochronous presentation of the audio and video portions of the presentation.

Since music signals are encoded by a voice encoding method optimized to human voice signals such as EVRC (Enhanced Variable Rate Coding) in a cellular communication system, the music signals are often distorted by such encoding method, and listeners experience pauses in music caused by such voice-optimized encoding method. To improve the perceptual sound quality of music, a method for preprocessing digital audio data is provided in order to prevent the problem of pause in music signals in a cellular phone. In particular, AGC (Automatic Gain Control) preprocessing and PHE (Pitch Harmonics Enhancement) is performed to the digital audio data having low dynamic range. By this method, the number of pauses in music signal is reduced, and the perceptual sound quality of the music is improved.

Methods and systems are provided for time altering one or more discrete digital audio / video program segments, each program segment having distinct In time and Out time code points. Data from the digital program segments are received from a data source and decoded. The decoded data are modulated as a serial data stream. The modulated decoded data are provided to a time altering processor to remove or duplicate frame positions to time alter the frame sequence. The resulting time altered serial data stream is demodulated to provide buffered program segment data. The buffered and time altered program segments are encoded and provided in a desired file or streaming format. Audio synchronization with the video is maintained by duplication or removal of audio samples corresponding to the duplicated or removed video frames.

The invention discloses a multipath data fusion device, a realization method thereof and a multipath audio data processing system, wherein the method comprises the following steps of: in each sampling period, simultaneously receiving N paths of digital audio data and respectively distributing the digital audio data received in the first half period and the digital audio data received in the second half period to a first data buffer and a second data buffer for storage, wherein N is not less than 2; fusing and packing all digital audio data distributed to the first data buffer and the second data buffer in the current sampling period; and sequentially transmitting the fused and packed digital audio data stored in the first data buffer and the second data buffer to an audio and video processor. The invention realizes no-delay and no-loss transmission and processing of the audio data.

An offset correction circuit removes DC offset from an analog audio output signal by comparing transitions of digital audio values to which the analog output signal is related to the output of a monitor that monitors the analog output signal. The monitor may be a zero-crossing detector and the transitions of the digital signal that are compared may be transitions of the most-significant bit (MSB) of the digital audio values. A filtering algorithm or filter circuit may be used to average a result of the comparison of the transitions, so that the offset is slowly and accurately removed. A chopped or autozero comparator may be used to further reduce error in the offset determination.

A video decoding apparatus capable of controlling presentation of sub-pictures includes a first decoder, a second decoder, a first scaler, a second scaler and a combiner. The first decoder and the second decoder respectively decode a digital audio / video signal to generate a decoded video and a decoded sub-picture. In accordance with an output picture size, the decoded video size, the decoded sub-picture size and a sub-picture aspect ratio, the first scaler and the second scaler generate a first scaling factor and a second scaling factor. The combiner combines the decoded video and the decoded sub-picture according to the first scaling factor and the second scaling factor and outputs the combination thereof. A video decoding method and a digital audio / video playback system capable of controlling presentation of sub-pictures are also disclosed.

The invention discloses a digital audio and video stage monitoring system. The digital audio and video stage monitoring system comprises a main control unit, wherein the main control unit is communicated with a plurality of display units; the display units are used for displaying a stage video acquired by a camera; the main control unit is further communicated with a wristband type terminal, a control room terminal and a field reminding terminal for monitoring. By adopting the digital audio and video stage monitoring system, visual scheduling and centralized control in regions such as stages and waiting regions are realized, and all-dimensional stage conversation and video monitoring and controller are ensured reliably; point-to-point, point-to-multipoint and multipoint-to-multipoint real-time audio and video talkback communication functions among personnel in the departments of light, stage, cosmetic and the like are realized; the mobile comprehensive command scheduling capabilities of personnel such as stage supervisors and directors are increased, limitations on distances, signal interference and signal source blind zones are eliminated, and seamless joint of commands is ensured.

A wireless communication system and in particular to a wireless communication system for digital audio players that provides for increased functionality, such as communication, interaction and synchronization between a computing platform and various mobile, portable or fixed digital audio players, as well as providing a communication link between the various digital audio players themselves. The computing platform may act, for example, through a wireless network or wireless communication platform, to control the digital audio players; to act as a cache of digital audio data for the digital audio players; as well as provide a gateway to the Internet to enable the digital audio players to access additional digital audio content and other information. The computing platform may also be used to automatically update digital audio content on the digital audio players; synchronize digital audio content and playlists between digital audio players; and automatically continue a particular playlist as the user moves from one digital audio player to another.