729 results about "Delay" patented technology

Embodiments of the invention enable the synchronization of clocks across packet switched networks, such as the Internet, sufficient to drive a jitter buffer and other quality-of-service related buffering. Packet time stamps referenced to a local clock create a phase offset signal. A shortest-delay offset generator uses a moving-window filter to select the samples of the phase offset signal having the shortest network propagation delay within the window. This shortest network propagation delay filter minimizes the effect of network jitter under the assumption that queuing delays account for most of the network jitter. The addition of this filtered phase offset signal to a free-running local clock creates a time reference that is synchronized to the remote clock at the source thus allowing for the transport of audio, video, and other time-sensitive real-time signals with minimal latency.

In an improved system for receiving digital voice signals from a data network, a jitter buffer manager monitors packet arrival times, determines a time varying transit delay variation parameter and adaptively controls jitter buffer size in response to the variation parameter. A speed control module responds to a control signal from the jitter buffer manager by modifying the rate of data consumption from the jitter buffer, to compensate for changes in buffer size, preferably in a manner which maintains audio output with acceptable, natural human speech characteristics. Preferably, the manager also calculates average packet delay and controls the speed control module to adaptively align the jitter buffer's center with the average packet delay time.

A method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

The embodiment of the invention discloses a microphone-array speech-beam forming method, which comprises a digital-signal converting step, a frequency-region signal-obtaining step, a time-delay obtaining step, a time-delay compensating step and a weighted stacking step, wherein time-delay estimation based on phase conversion is adopted particularly in the time-delay obtaining step, thereby a beam-forming signal pointed to a target sound-source spatial position after enhancement processing is obtained. Compared with the prior art, the positioning precision and the directivity of a three-dimensional space are enhanced, the long-distance sound-picking capability in a complicated acoustics environment is greatly enhanced, high-quality voice signals are obtained, and noise and other interferences are reduced. The embodiment of the invention also discloses a speech-signal processing device and a system.

The invention discloses an audio processing method for a bone conduction headset, the bone conduction headset and an audio playback device based on the bone conduction headset. The bone conduction headset comprises a human bone and tissue model modeling module, a digital pre-corrector, a delay estimation unit, a digital-to-analog converter, an analog-digital converter, a first low-pass filter, a second low-pass filter, an audio amplifier, an audio driving amplifier, at least one bone conduction microphone, and at least one bone conduction vibrator. In the method, human bone and tissue attenuation effect information of different users is detected in real time; a compensating transfer function is generated based on the attenuation effect information; and an input audio signal is subjected to digital pre-correction via the compensating transfer function, and then the processed input audio signal is transmitted in human bones and tissues. Through adoption of the method, a problem that experience of the bone conduction headset is different caused by different tissue thickness of the different users is solved.

Generally speaking, there are provided systematic techniques for increasing and decreasing jitter buffer delay. The disclosed techniques typically utilize various combinations of: evaluating received data over a specified interval, increasing a recommended buffer delay if the interval delay exceeds a first threshold and decreasing the recommended buffer delay if the interval delay is less than a second threshold, causing the recommended buffer delay to decrease over time until an underflow condition is identified, and/or increasing the recommended buffer delay in response to identifying the underflow condition.

The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (410) that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit (404) that codes the first downmix signal; a first t-f converting unit (401) that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit (409) that generates spatial information for generating a multi-channel audio signal from a downmix signal.

A high quality digital 3D sound rendering is implemented using high resolution interaural time delays formed from two delay lines: a first delay line providing a rough estimate of the desired interaural time delay for a particular audio sample, and a second delay line in series with the first delay line providing a more finely resolved fractional delay. In the disclosed embodiment, the first delay module, i.e., the integer delay module, is formed from a first-in, first-out (FIFO) buffer with appropriate selection control of a desired sample as it passes through the FIFO buffer with each clock cycle based on the sampling rate. The second delay module (i.e., the fractional delay module) is formed from a plurality of polyphase (FIR) filters. The number of polyphase filters is determined based on the desired resolution of the interaural time delay.

An apparatus and method for canceling crosstalk between 2-channel speakers and two ears of a listener in a stereo sound generation system. The crosstalk canceling apparatus includes a first signal processing unit to cross-mix first and second channel signals with gain and delay-adjusted first and second channel signals, a second signal processing unit to adjust frequency characteristics of the signals mixed in the first signal processing unit.

The invention discloses a method for compensating delay and frequency response characteristics of multi-output channel sound system and system implementation thereof. The method comprises the following steps that firstly, phase difference between channels of a multi-channel output device is measured, so that the situation that the multi-channel output device itself is in the state of synchronously outputting signals is ensured; then the delay of the signals which are output by the multi-channel output device and pass through a signal processing system, a loudspeaker box and a space transmission routine to reach a specific audition area is estimated through a delay estimation method, and delay compensation is conducted on the channels by comparing delay difference between the channels; finally, an FIR frequency response compensating filter is designed and achieved according to actually detected frequency response curves of a signal transmission routine, and the part above the low and medium frequency in the frequency response curves of the multiple channels is compensated into a straight line through the filter as far as possible. By means of the method, when sound waves transmitted out of each loudspeaker box reach the audition area, the phases of the sound waves are basically the same, and the frequency response characteristics of the channels are basically the same.

The invention provides an adaptive voice separating method based on sound source positioning, and relates to the technical field of information processing. The method includes steps: acquiring an audio signal of an observed environment, and confirming the number of sound sources and the direction of arrival of each sound source; generating a dimension reduction matrix P; generating a voice transfer matrix and a delay superposed wave beam coefficient; determining an active sound source of a frequency point and separating voice components; obtaining the obtained voice components and setting non-activated sound source components as zero; and obtaining time domain voice signals of the sound sources. According to the method, the number and the orientation of the sound sources in a current environment can be obtained through a sound source positioning technology, dimension reduction of each frequency band of the voice signal is performed with the cooperation of a PCA whitening technology toobtain an initial separation matrix, frequency components of each sound source channel are separated through the number of the activated sound sources at the frequency point by adaptive usage of the beam forming technology and the FDICA technology to restore the voice components, the obtained signal-to-noise ratio improvement characteristic is higher, better noise suppression performance is achieved, and the method is applicable to any sound source situations in the real voice environment.

The determination of user perceived delay in audio and video conferencing systems/services is described. Example embodiments consistent with the present invention measure delays in audio and/or video conferencing systems and services, by treating such systems and services as black boxes, thereby providing end-to-end delays as they would be perceived by users. For example, the user perceived round-trip delay in a video conference system/service may include time delays from each of video capture, video encoding, video transmission, video decoding and video rendering.

The invention provides an audio data generating method for voice synthesis. The audio data generating method comprises the following steps: extracting the text features in the text data to obtain textfeature data; carrying out the acceleration conversion processing on the text feature data through a neural network structure, and converting the text feature data into acoustic feature data; performing sound synthesis or selected splicing according to the acoustic feature data to obtain audio data. According to the invention, through the special anti-convolution structure is adopted, a good voice synthesis effect can be achieved on the premise that no any auto-regressive structure is included and few parameters are used, the calculation delay can be reduced while the prediction precision ofthe acoustic features can be guaranteed through the neural network structure, and the requirement for computing resources is reduced, the concurrent quantity is increased, the voice synthesis speed isincreased, and contribution is made to improvement of human-computer interaction experience.

A stereo audio decoder with low complexity is provided. A high stereo sound quality can be obtained with a limited computational power and is thus suitable for miniature and mobile equipment. The stereo decoder generates a set of stereo output channels (C1, C2) in response to a parametric audio input including signal parameters (S1) and stereo related parameters (X1). A parameter processor (M) generates two different set of parameters (P1, P2) based on the input signal parameters (S1) thus up-mixing the signal parameters (S1) by altering or manipulating the signal parameters (S1) corresponding to the stereo related parameters (X1). The two different parameters (P1, P2) are finally synthesized by separate signalsynthesizers (SS1, SS2) to form respective stereo output channels (C1, C2). Since the stereo decoding can be performed in the parameter domain instead of the spectral domain, the required computational burden is reduced compared to what is known in prior art. Preferably the signalsynthesizers (SS1, SS2) are sinusoidal synthesizers, and preferably the decoder also includes transient and noise synthesizers to generate transient and noise signal portions to be applied to the stereo output channels (C1, C2). Further, different transient and noise signal portions to the output channels (C1, C2) may be provided by applying different gains based on the stereorelated parameter (X1). In preferred embodiments the two parameters (P1, P2) are determined from a current as well as a previous signal parameter input, e.g. by means of an input delay line.

A filter coefficient of a graphic equalizer GEQ is corrected based on detection results of reproduced sounds generated by supplying a noise to all frequency band loudspeakers 6_FL to 6_RR and a low frequency band exclusively reproducing loudspeaker 6_WF via the graphic equalizer GEQ. Then, attenuation factors of channel-to-channel attenuators ATG₁ to ATG₅ are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6_FL to 6_RR via the graphic equalizer. Then, delay times of delay circuits DLY₁ to DLY_k are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6_FL to 6_WF via the graphic equalizer. Then, an attenuation factor of a channel-to-channel attenuator ATG_k is corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6_FL to 6_RR via the graphic equalizer and the detection result of the reproduced sound generated by supplying the noise to the loudspeaker 6_WF via the graphic equalizer, whereby levels of the reproduced sounds reproduced by the loudspeakers 6_FL to 6_WF are adjusted to be made flat over the audio frequency band.

The invention discloses a helmet type microphone array sound source positioning method based on low-frequency diffraction delay inequalities. The problems that a helmet type microphone array is poor in positioning accuracy and large in computation burden are mainly solved. The method includes the steps of firstly, determining backward microphones through high-frequency signal strength differences; secondly, estimating the diffraction path delay inequalities generated when low-frequency signals reach the different backward microphones; thirdly, giving a sound source azimuth angle linear closed type solution for the relation between the low-frequency diffraction delay inequalities and sound source azimuth angles according to the microphones, and estimating the sound source azimuth angles. By means of the method, the positioning accuracy of the helmet type microphone array is improved; the computation burden is reduced when the number of the microphones is large; the method can be used for a single-soldier portable counter-ambush sound source positioning system.