79 results about "Latency (audio)" patented technology

DTS Interactive provides low cost fully interactive immersive digital surround sound environment suitable for 3D gaming and other high fidelity audio applications, which can be configured to maintain compatibility with the existing infrastructure of Digital Surround Sound decoders. The component audio is stored and mixed in a compressed and simplified format that reduces memory requirements and processor utilization and increases the number of components that can be mixed without degrading audio quality. Techniques are also provided for “looping” compressed audio, which is an important and standard feature in gaming applications that manipulate PCM audio. In addition, decoder sync is ensured by transmitting frames of “silence” whenever mixed audio is not present either due to processing latency or the gaming application.

Codeword-position-caused encoder latency is reduced by avoiding the requirement for knowledge of the message prior to generating an error detecting or concealing codeword associated with the message. A pseudo error detecting or concealing codeword is inserted in place of the normal error detecting or concealing codeword appropriate for the segment of information to which the error detecting or concealing codeword relates. In order to satisfy the requirement of conventional decoders, the pseudo error detecting or concealing information must match or be appropriate for the segment so that the decoder sees the codeword and message segment as valid or error free. This is accomplished by modifying or perturbing at least a portion of the segment to which the pseudo codeword relates. The invention is particularly useful for maintaining the backward compatibility of audio data encoding formats in which the minimum latency is too long (e.g., computer games, where the player performs some operation leading to a sound, and that sound must not be perceptibly delayed with respect to the operation).

A method for synchronizing content undergoing play out at first and second stations commences by comparing audio within content undergoing play out on the first station to the audio within content undergoing play out on the second station to determine if a latency exists between the first and second stations. If such a latency exists, then at least one of a pause or jump operations is performed in connection with content playing out by at least one of the first and second stations for an interval corresponding to an interval by which one station leads the other station to bring the two stations into substantial synchronism

A computerized method of delivering data to one or more client computing devices includes receiving, by a server computing device, a data stream starting at a first time; processing, by the server computing device, the data stream, thereby creating a processed data stream; transmitting, by the server computing device, via a wireless network in electronic communication with the server computing device, the processed data stream to the one or more client computing devices; and interpreting, by an application installed on the one or more client computing devices, the processed data stream, thereby recovering the data stream for use by the one or more client computing devices. A latency between the first time and the second time is less than 100 milliseconds.

Embodiments are described for a method for localizing a set of speakers (106) and microphones (108), having only the times of arrival between each of the speakers and microphones. An autodiscovery process (107) uses an external input to set: a global translation (3 continuous parameters), a global rotation (3 continuous parameters), and discrete symmetries, i.e., an exchange of any axis pairs and/or reversal of any axis. Different time of arrival acquisition techniques may be used, such as ultrasonic sweeps or generic multitrack audio content. The autodiscovery algorithm is based in minimizinga certain cost function, and the process allows for latencies in the recordings, possibly linked to the latencies in the emission.

Management and control of the distribution of multiple, synchronous and asynchronous, simultaneously occurring audio and video data content in a communications network, as well as the bandwidth used for distributing the content on the network, is performed by all of the content source and content rendering devices operating in unison. The devices, by utilizing a method of establishing dedicated communication channels based on available bandwidth, data content, and network link reliability, can perform the distribution of content throughout the network in a manner that will reduce the latency and improve the reliability of the data distribution.

An in-ear device includes a housing shaped to hold the in-ear device in an ear of a user, and an audio package, disposed in the housing, to emit augmented sound. A first set of one or more microphones is positioned to receive external sound, and a controller is coupled to the audio package and the first set of one or more microphones. The controller includes a low-latency audio processing path, digital control parameters, and logic that when executed by the controller causes the in-ear device to perform operations. The operations may include receiving the external sound with the first set of one or more microphones to generate a low-latency sound signal; augmenting the low-latency sound signal by passing the low-latency sound signal through the low-latency audio processing path to produce an augmented sound signal; and outputting, with the audio package, the augmented sound based on the augmented sound signal.

Exemplary methods include a processor executing instructions stored in a memory for generating an electronic count-in, binding the electronic count-in to a first performance to generate a master clock and transmitting a first musician's first performance and first timing information to a network caching, storage, timing and mixing module. The first musician's first performance may be recorded locally at full resolution and transmitted to a full resolution media server and the first timing information may be transmitted to the master clock. The first musician's first performance is transmitted to a sound device of a second musician and the second musician creates a second performance, transmits it and second timing information to a network caching, storage, timing and mixing module. The first and second performances are mixed along with the first and the second timing information to generate a first mixed audio, which can be transmitted to a sound device of a third musician.

A first audio device exchanges first control signals with a source device and thereby receives first audio signals from the source device. A second audio device exchanges second control signals with the first audio device including information the second audio device needs to receive the first audio signals from the source device. The first control signals include communications using Bluetooth BR/EDR protocols and establish a first Bluetooth link for audio streaming from the source device to the first audio device. The second control signals include communications using Bluetooth Low Energy (BLE) protocols and transfer to the second audio device parameters of the first Bluetooth link, such that the second audio device can receive and decode an audio stream transmitted from the source device to the first audio device over the first Bluetooth link.

The invention discloses a low-latency on-board Ethernet audio stream processing system. The system comprises an audio module, an FPGA module, and an interface module, wherein the audio module is connected with the interface module through the FPGA module, the audio module receives and plays an analog audio processed by the FPGA module, and collects and sends an analog audio to the FPGA module, theFPGA module receives and decodes the analog audio sent by the audio module so as to obtain a digital audio, packages the digital audio according to an ARINC628P3 protocol and then sends out through the interface module, receives an audio data packet sent by the interface module, unpacks according to an ARINC628P3 protocol and then processes to obtain an analog audio and sends the analog audio tothe audio module, and the interface module provides an interface between the external environment and the FPGA module, an interface circuit between the FPGA module and periphery, and provides power for devices. Through adoption of the low-latency on-board Ethernet audio stream processing system, the problem that audio broadcast freezes or has noise due to overlarge fluctuation of processing latency is solved.

To synchronize sound information with corresponding picture information for digital cinema compositions at different frame rates in a play list during play out of the digital cinema compositions, associated audio latency settings are first established for the corresponding picture information of the digital cinema compositions in the play list in accordance with the digital cinema composition frame rates. The timing between the sound information and the picture information is then adjusted during play out of the digital cinema compositions in accordance with the associated audio latency settings for the corresponding digital cinema composition frame rates.

A method of playing back audio/video signals by means of an audio/video system includes a decoder appliance connected to a display screen and to a wireless speaker including a buffer storing audio frames so as to allow latency between receiving encoded audio frames and supplying decoded audio frames. The method comprises the steps of: determining a filling level of the buffer; comparing the filling level of the buffer with a filling target for the buffer and deducing a filling difference therefrom; and increasing or decreasing the number of audio frames that are stored as a function of the filling difference so as to adjust latency.

In some embodiments, a multistage filter whose biquad filter stages are combined with latency between the stages, a system (e.g., an audio encoder or decoder) including such a filter, and methods for multistage biquad filtering. In typical embodiments, all biquad filter stages of the filter are operable independently to perform fully parallelized processing of data. In some embodiments, the inventive multistage filter includes a buffer memory, at least two biquad filter stages, and a controller coupled and configured to assert a single stream of instructions to the filter stages. Typically, the multistage filter is configured to perform multistage filtering of a block of input samples in a single processing loop with iteration over a sample index but without iteration over a biquadratic filter stage index.

Various embodiments of the disclosure provide a method and a device, the device including: a communication module; a memory; and a processor operatively coupled to the communication module and the memory, wherein the processor is configured to: monitor a first device latency between a wireless audio output device and the electronic device connected using a first communication scheme through the communication module; predict a second device latency in connection with the wireless audio output device via a wearable display device connected using a second communication scheme through the communication module; and control connection of the wireless audio output device based on the first device latency and the second device latency. Various embodiments are possible.

The invention discloses an audio and video conference method based on a 5G low-delay spanning tree, and the method employs a terminal video source module, a terminal video viewing module, a terminal audio module, a terminal sound mixing module, a terminal distribution module, a terminal detection module, a terminal 5G communication module, a terminal D2D communication module, a 5G base station, anedge detection module, an edge distribution module, an edge sound mixing module, a 5G bearer network, a 5G core network, a central conference module, a central distribution module, a central sound mixing module, a central delay spanning tree module and a central detection module. According to the invention, a 5G network centralized core network is changed into a distributed optimization and 5G terminal ad hoc network and has forwarding capability, the delay of video forwarding and audio mixing is reduced by using a delay spanning tree and the self-forwarding of an edge server and a terminal,and the joining of an audio and video conference is realized through the forwarding of the terminal by using the signal difference between a 5G D2D technology terminal and a base station.

A method for wireless RF transmission of short audio data blocks, e.g. 0.5 ms to 2 ms blocks, with low latency. The method involves a fixed part (FP) serving as synchronization master, and one or more portable parts (PP) being synchronization slaves in a time division scheme with fixed transmission intervals, and with a fixed and limited payload capacity of the RF transmission channel, such as 1.5-3 times the capacity required to transmit the audio data blocks in real- time. Short length transmission and receiving queues (TQ, RQ), e.g. having each 2-8 spaces for audio data blocks, for the audio data blocks are used to allow retransmission of blocks in—response to a missing acknowledge response from the portable part (PP). The queuing is operated so as to result in a fixed latency C determined e.g. by the transmission and receiving queue (TQ, RQ) lengths. A two-way audio scheme can be implemented following the same principle and utilizing the same RF transmission principles. The method provides a robust and low latency wireless audio interface suitable for dedicated audio devices and/or combined audio and Human Interface Devices (HIDs), e.g. for gaming equipment.