741 results about "Delay" patented technology

A digital wavetable audio synthesizer is described. The synthesizer can generate up to 32 high-quality audio digital signals or voices, including delay-based effects, at either a 44.1 KHz sample rate or at sample rates compatible with a prior art wavetable synthesizer. The synthesizer includes an address generator which has several modes of addressing wavetable data. The address generator's addressing rate controls the pitch of the synthesizer's output signal. The synthesizer performs a 10-bit interpolation, using the wavetable data addressed by the address generator, to interpolate additional data samples. When the address generator loops through a block of data, the signal path interpolates between the data at the end and start addresses of the block of data to prevent discontinuities in the generated signal. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data. The data can be placed in one of sixteen fixed stereo pan positions, or left and right offsets can be programmed to place the data anywhere in the stereo field. The left and right offset values can also be programmed to control the overall volume. Zipper noise is prevented by controlling the volume increment. A synthesizer LFO generator can add LFO variation to: (i) the wavetable data addressing rate, for creating a vibrato effect; and (ii) a voice's volume, for creating a tremolo effect. Generated data to be output from the synthesizer is stored in left and right accumulators. However, when creating delay-based effects, data is stored in one of several effects accumulators. This data is then written to a wavetable. The difference between the wavetable write and read addresses for this data provides a delay for echo and reverb effects. LFO variations added to the read address create chorus and flange effects. The volume of the delay-based effects data can be attenuated to provide volume decay for an echo effect. After the delay-based effects processing, the data can be provided with left and right offset volume components which determine how much of the effect is heard and its stereo position. The data is then stored in the left and right accumulators.

A digital wavetable audio synthesizer is described. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data. The data can be placed in one of sixteen fixed stereo pan positions, or left and right offsets can be programmed to place the data anywhere in the stereo field. The left and right offset values can also be programmed to control the overall volume. Zipper noise is prevented by controlling the volume increment. A synthesizer LFO generator can ad LFO variation to: (i) the wavetable data addressing rate, for creating a vibrato effect; and (ii) a voice's volume, for creating a tremolo effect. Generated data to be output from the synthesizer is stored in left and right accumulators. However, when creating delay-based effects, data is stored in one of several effects accumulators. This data is then written to a wavetable. The difference between the wavetable write and read addresses for this data provides a delay for echo and reverb effects. LFO variations added to the read address create a chorus and flange effects. The volume of the delay-based effects data can be attenuated to provide volume decay for an echo effect. After the delay-based effects processing, the data can be provided with left and right offset volume components which determine how much of the effect is heard and its stereo position. The data is then stored in the left and right accumulators.

In a video-audio synchronizing method for synchronizing video and audio reproduced respectively by a monitor 300 and an amplifier 400 connected to a serial bus 100 to which video from a DVD player 200 and audio associated with the video are supplied, the monitor 300 sends out delay time information caused by an intrinsic delay element for video display to the amplifier 400 through the serial bus 100 as a control command, and the amplifier 400 delays the audio by time necessary for synchronization between the video and the audio according to the control command and outputs the audio.

Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network. The decoding device uses the relative delay to drive a playout splitter-once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference. An advantage of these embodiments is that a pseudo-stereo conference can be conducted with virtually the same bandwidth as a monophonic conference.

The invention relates to a live broadcast data processing method and device, computer equipment and a storage medium. The method comprises the following steps: acquiring live broadcast stream data, and writing the live broadcast stream data into a live broadcast cache region; wherein the live broadcast stream data comprise audio stream data and video stream data, when the cached data volume of thelive broadcast stream data in the live broadcast cache region satisfies a playing adjustment condition, adjusting the initial audio frame rate of the audio stream data in the live broadcast cache region according to the cached data volume to obtain a target audio frame rate, and playing the audio stream data according to the target audio frame rate, and synchronously playing the video stream datacorresponding to the audio stream data of which the audio frame rate is adjusted. By adopting the method, the playing delay of the playing end can be reduced.

A system using the nonlinearity of a propagation medium to demodulate an ultrasonic wave having audio modulated onto the ultrasonic frequency, audio signal processing circuitry providing a delayed audio signal and an envelope signal which is a function of peaks of the audio signal over a predetermined interval. The delayed audio signal and the envelope signals are combined with the delay in the audio allowing the envelope signal to accurately be aligned with the audio signal in the combination.

The invention relates to the technical field of Bluetooth communication, in particular to a synchronous playing method and system for Bluetooth audio equipment, Bluetooth audio master equipment and Bluetooth audio slave equipment. The method comprises the steps of correspondingly determining the playing time of the audio data packet to be synchronously played at the Bluetooth audio main equipmentend; converting the playing time into a timestamp, wherein the timestamp comprises a Bluetooth clock value and a microsecond clock offset value; and sending the timestamp. At a Bluetooth audio slave device end, a timestamp is received, and an audio data packet to be synchronously played is received; and the Bluetooth audio slave device plays the audio data packet based on the timestamp, and playsthe audio data packet corresponding to the timestamp when both the value of the Bluetooth clock and the microsecond clock offset value arrive. According to the method, the playing time is determined based on the microsecond-level clock offset value, so that audio playing is started and carried out between each Bluetooth audio slave device or between the Bluetooth audio master device and the Bluetooth audio slave device based on the more accurate playing time, and the playing delay is at the microsecond level.

A method for stereo expansion includes a step to remove the effects of actual relative speaker to listener positioning and head shadow and a step to introduce an artificial effect based on a desired virtual relative speaker to listener positioning using the inter-aural delay and the head-shadow models for the virtual speakers at desired angles relative to the listener thereby creating the impression of a widened and centered sound stage and an immersive listening experience. Known methods drown out vocals and add mid-range coloration thereby defeating equalization. The present method includes the integration of a novel binaural listening model and speaker-room equalization techniques to provide widening while not defeating equalization.

The present invention relates to methods and devices for encoding and decoding digital audio signals, e.g. a speech signal. An audio coder and a decoder are provided wherein a modeller adds a first distribution model obtained from model parameters of past segments of the digital audio signal and a fixed distribution model, each of the models being multiplied by a weighting coefficient, for obtaining a combined distribution model. The weighting coefficients are selected to minimize a code length of a current segment of the digital audio signal. As the combined distribution model is a sum of several distribution models, wherein at least some of the models is based on the model parameters, flexibility is introduced in the signal model used to encode the digital audio signal. Thus, an audio coder and decoder providing a low bit rate in average, low bit rate variations and low error propagation are provided.

The invention relates to an improved sound source localization method based on a progressive serial orthogonalization blind source separation algorithm, and an implementation system for the improved sound source localization method. The improved sound source localization method comprises the steps of: 1, acquiring and storing sound signals; 2, separating the sound signals to obtain independent sound source signals; 3, selecting the independent sound source signal of sounds to be localized by adopting a pattern matching algorithm from the independent sound source signals; 4, and if the sound source is a single sound source, performing coarse localization at first according to a result of pattern matching calculating an envelope of the signals, performing low-resolution sampling, calculatingtime delay by adopting a generalized autocorrelation function method roughly, carrying out time domain shifting on the signals according to a point number of rough localization, then performing finelocalization, carrying out high-resolution sampling, calculating time delay by adopting the generalized autocorrelation function method to obtain precise time delay, and solving a position of the sound source; and if the sound sources are multiple, calculating time delay by adopting a TDOA algorithm and solving positions of the sound sources. Compared with the traditional TDOA algorithm, the improved sound source localization method can improve the precision to some extent, and can reduce the algorithm computation amount.

The invention discloses a data transmission method, and the method comprises the steps: receiving a screen projection request of a screen projection terminal, and building an audio and video data connection channel with the screen projection terminal; acquiring audio and video data through the audio and video data connection channel and playing the audio and video data; monitoring the network state of the audio and video data connection channel; and when the network state becomes poor, selecting and executing a corresponding transmission mode from a preset transmission strategy according to the index value of the network state. The invention further provides a data transmission device, computer equipment and a computer readable storage medium. According to the invention, a proper transmission mode can be dynamically selected according to the index value of the network state, so that the problems of time delay and jamming are solved, and a display effect is improved.