Showing papers on "Latency (audio) published in 2010"

Activity-driven, event-based vision sensors

Abstract: The four chips [1–4] presented in the special session on "Activity-driven, event-based vision sensors" quickly output compressed digital data in the form of events. These sensors reduce redundancy and latency and increase dynamic range compared with conventional imagers. The digital sensor output is easily interfaced to conventional digital post processing, where it reduces the latency and cost of post processing compared to imagers. The asynchronous data could spawn a new area of DSP that breaks from conventional Nyquist rate signal processing. This paper reviews the rationale and history of this event-based approach, introduces sensor functionalities, and gives an overview of the papers in this session. The paper concludes with a brief discussion on open questions.

JackTrip: Under the Hood of an Engine for Network Audio

Abstract: The design of a platform for bi-directional musical performance using modern Wide area networks (WANs) poses several challenges that are different from related applications, e.g. synchronous local area network (LAN) studio systems or uni-directional WAN streaming. The need to minimize as much as possible audio latency and also maximize audio quality requires specific strategies which are informed, in part, by musical decisions. We present some of the key design elements of the JackTrip application which has evolved through several years of deployment in musical work over wide area networks.

Prediction of channel state for cognitive radio using higher-order hidden Markov model

Abstract: Spectrum sensing detects the availability of the radio frequency spectrum, which is essential and vital to cognitive radio Traditional techniques for spectrum sensing fail to take the latency between spectrum sensing and data transmission into consideration However, such latency does exist in hardware implementation Prediction can be utilized to diminish the negative effect of such latency In this paper, this latency is illustrated, and an approach for prediction of channel state using higher-order hidden Markov model (HMM) is proposed The predicted channel states are output together with corresponding likelihood probabilities that are helpful to subsequent decision making Wi-Fi signals have been recorded using a latest advanced ultra-performance digital phosphor oscilloscope (DPO), which are employed to evaluate the performance of the proposed approach Experimental results show that the proposed approach for prediction of channel state is effective The proposed approach for prediction of channel state can be used together with traditional spectrum sensing techniques for spectrum sensing with the latency taken into consideration And it can also be utilized to provide predictive information to upper-level modules of cognitive radio

Synchrophasor Broadcast Over Internet Protocol for Distributed Generator Synchronization

Abstract: Synchronous islanded operation involves continuously holding an islanded power network in virtual synchronism with the main power system to aid paralleling and avoid potentially damaging out-of-synchronism reclosure. This requires phase control of the generators in the island and the transmission of a reference signal from a secure location on the main power system. Global positioning system (GPS) time-synchronized phasor measurements transmitted via an Internet protocol (IP) are used for the reference signal. However, while offering low cost and a readily available solution for distribution networks, IP communications have variable latency and are susceptible to packet loss, which can make time-critical control applications difficult. This paper investigates the ability of the phase-control system to tolerate communications latency. Phasor measurement conditioning algorithms that can tolerate latency are used in the phase-control loop of a 50-kVA diesel generator.

System and method for low-latency multimedia streaming

Abstract: Systems and methods for transmitting a multimedia stream over a communication link on a network are disclosed. The systems and methods adaptively adjust encoding parameters based on monitoring changing conditions of the network. A transmitter includes an adaptive-rate encoder that adaptively adjusts a video encoding bit rate in response to changing conditions of the communication link. The encoder maintains tight rate control by utilizing slice processing and sub-frame rate adaptation, as well as maintaining a headroom between the channel bit rate and the video encoding bit rate. The adaptive-rate encoder also embeds intra-frame constraints in predictive frames traffic in order to reduce latency.

Wide-area phasor power oscillation damping controller: A new approach to handling time-varying signal latency

Abstract: Using the present technology, latency associated with remote feedback signals can be determined from the time stamp information at both the phasor measurement unit (PMU) location and the control centre. This study illustrates how this latency could be accounted for in the implementation of a wide-area phasor power oscillation damping controller (phasor POD). The basic idea is to adjust the position of the rotating reference frame - used for phasor extraction - to account for the extra phase shift introduced because of latency. The oscillatory component of the original PMU measurement is retrieved out of the delayed signal received at the control centre. Thus, continuous compensation is achieved without requiring any Pade approximation and/or gain scheduling, unlike the techniques reported in the literature. With the proposed modification, a phasor POD is shown to continuously adapt to the actual latency and maintain the desired dynamic performance over a range of different operating conditions.

Low latency active noise cancellation system

Abstract: Systems and methods described herein provide for low latency active noise cancellation, which alleviates the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques that overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Communication system, network element and method for antenna array calibration

Abstract: A network element for a wireless communication system is locatable to couple at least one base station to an antenna array comprising a plurality of antenna elements. The network element comprises a plurality of independent transceiver circuits coupled to at least one of a plurality of respective antenna elements of the antenna array; and logic arranged to apply at least one complex digital signal to at least one transceiver signal path of a transceiver circuit of the plurality of independent transceiver circuits. A feedback path is arranged to provide feedback of the at least one complex digital signal such that it is capable of facilitating determination of latency mismatch error response between at least two transceiver signal paths. Adjustment means comprises delay logic arranged to receive a complex digital signal and provide a modified representation of the received complex digital signal in response to the latency mismatch error response of the at least two transceiver signal path.

Memory controller and device with data strobe calibration

Abstract: A memory controller comprises a DQ path, a DQS path, a delay element, a flip flop, and an adjustment unit The DQ path receives and passes a data signal, and outputs a delayed data signal The DQS path receives and passes a data strobe signal The delay element is coupled to the DQS path, receiving the data strobe signal to generate a compensated data strobe signal having a calibrated latency The calibrated latency is determined by an adjustment signal The flip flop is coupled to the data signal path and the delay element, sampling the delayed data signal by the compensated data strobe signal to generate an output data The adjustment unit generates the adjustment signal according to the output data The adjustment unit performs a calibration to adjust the adjustment signal, thus the calibrated latency is adjusted

Semiconductor device having additive latency

Abstract: A semiconductor device receives a command corresponding to a memory access operation and performs the memory access operation after an additive latency period. The additive latency period begins when the command is received. The semiconductor device comprises a phase controller for controlling a phase of a clock signal and outputting a phase-controlled clock signal, and a controller for generating and outputting a control signal for enabling the phase controller that is disabled, at a predetermined time in the additive latency period.

Streaming, low-latency communication in on-line trading systems

Abstract: This paper presents and evaluates the performance of a prototype of an on-line OPRA data feed decoder. Our work demonstrates that, by using best-in-class commodity hardware, algorithmic innovations and careful design, it is possible to obtain the performance of custom-designed hardware solutions. Our prototype system integrates the latest Intel Nehalem processors and Myricom 10 Gigabit Ethernet technologies with an innovative algorithmic design based on the DotStar compilation tool. The resulting system can provide low latency, high bandwidth and the flexibility of commodity components in a single framework, with an end-to-end latency of less then four microseconds and an OPRA feed processing rate of almost 3 million messages per second per core, with a packet payload of only 256 bytes.

Hearing instrument and method of operating the same

Abstract: A hearing instrument and a method of operating a hearing instrument are provided. The hearing instrument includes an electro-acoustic or electro-mechanical output transducer. When a low battery status of the hearing instrument is detected, a low power audio signal processing mode is selected to extend a lifetime of a battery. The low power audio signal processing mode includes reducing a gain at predefined frequencies; switching off an audio signal processing function for one or more audio frequencies; switching-off parts of a digital audio processor to reduce a duty cycle; and reducing power consumption of an audio signal preamplifier, an audio signal analog-to-digital converter or an audio signal digital-to-analog converter.

Wireless hearing assistance system and method

Abstract: A hearing assistance system having a plurality of transmission units, a microphone arrangement and a digital transmitter for transmitting audio signals as audio data packets via a wireless digital audio link; at least one user worn receiver unit; a relay unit with a mixing unit for producing a mixed audio signal from audio signals received and at least one digital transceiver for receiving audio signals from the transmission units via the digital audio link and for transmitting the mixed audio signal via the wireless digital audio link as audio data packets to a digital receiver of the receiver unit(s) for stimulating the hearing of the user(s) according to audio signals supplied from the receiver unit, the relay unit and each transmission unit transmitting each audio data packet in at least one allocated separate slot of a TDMA frame at a different frequency according to a frequency hopping sequence

System, method and apparatus for wireless synchronizing three-dimensional eyewear

Abstract: An application for transmission of a three-dimensional eyewear synchronization signal to synchronize the operation of shutters of three-dimensional eyewear uses an industry standard wireless transmission technique. To compensate for inherent latencies of such transmission techniques, the latencies are measured and monitored to determine expected latencies and the shutter synchronization signal is skewed by the latency. In some embodiments, the synchronization signal is further adjusted by a user skew control.

Apparatus, system and method for buffering audio data to allow low power states in a processing system during audio playback

Abstract: An audio data stream from a processing system may be buffered to allow low power states in the processing system during audio playback. An audio buffer may be provided external to the processing system and between the processing system and an audio codec. The audio buffer may also shift to an alternate audio data interface mode when the processing system is in the low power state. Of course, many alternatives, variations, and modifications are possible without departing from this embodiment.

Audio Latency Measurement for Desktop Operating Systems with Onboard Soundcards

Abstract: Using commodity computers in conjunction with live music digital audio workstations (DAW) has become increasingly more popular in recent years. The latency of these DAW audio processing chains for some application such as live audio monitoring has always been perceived as a problem when DSP audio effects are needed. With "High Definition Audio" being standardised as the onboard soundcard's hardware architecture for personal computers, and with advances in audio APIs, the low latency and multi-channel capability has made its way into home studios. This paper will discuss the results of latency measurements of current popular operating systems andhosts applications with different audio APIs and audio processing loads.

VoIP: making secure calls and maintaining high call quality

Abstract: Modern multimedia communication tools must have high security, high availability and high quality of service (QoS). Any security implementation will directly impact on QoS. This paper will investigate how end-to-end security impacts on QoS in Voice over Internet Protocol (VoIP). The QoS is measured in terms of lost packet ratio, latency and jitter using different encryption algorithms, no security and just the use of IP firewalls in Local and Wide Area Networks (LAN and WAN). The results of laboratory tests indicate that the impact on the overall performance of VoIP depends upon the bandwidth availability and encryption algorithm used. The implementation of any encryption algorithm in low bandwidth environments degrades the voice quality due to increased loss packets and packet latency, but as bandwidth increases encrypted VoIP calls provided better service compared to an unsecured environment.

Improving sound field reproduction based on higher-order ambisonics in a small room with a 157-loudspeaker array

Abstract: This article presents a case study of higher-order Ambisonics (HOA) for real-time sound field reproduction in a small room with a 157-loudspeaker array. It addresses a number of specific questions and practical issues on the system design and implementation, such as the reproduction room's acoustic, loudspeaker positioning and radiation patterns, distributed computing and audio channel synchronization, and in more general the achievable accuracy of sound field reproduction. In the current configuration of the system Ambisonics up to order n = 6 is applied and the decoders are rendered in parallel on a cluster of four computers. For this reason, synchronization and communication between the different computers becomes a challenging task for achieving a good system performance. The overall system latency and the inter-channel synchronicity have been measured using time-stretched pulse (TSP) signals. The measurement results have shown a maximum (unsigned) latency of 51 samples, which corresponds to t = 1.1 ms. It is obvious that the acoustic of the reproduction room has a strong effect on the accuracy of the Ambisonics sound field reproduction. To achieve semi-anechoic conditions sound absorption materials have been installed in the room. Finally, spatial filters have been applied to each individual loudspeaker to correct for different orientations with reference to the sweet spot. These filters have been derived from radiation pattern measurements in an anechoic chamber.

Effects transitions in a music and audio playback system

Abstract: A music and audio playback system enables an operator to control an audio effect for transitioning an audio output signal from a first audio source to a second audio source by using a single cross-fader control. The transition audio effect combines the two incoming sources using a function that depends on the current position of the cross-fader control. Effects include cross-fading the frequency range, band-ducking, vocoder effects, and beat-cutting effects. The technique is especially advantageous in DJ performance systems in which audio effect-based transitions can add excitement and variety to a performance. Dedicated transition effect preset buttons located near the cross fader facilitate rapid selection by the DJ of the cross-fader controlled transition audio effect.

Circuit and method for generating on-die termination signal and semiconductor apparatus using the same

Abstract: Various embodiments of an on-die termination (ODT) signal generating circuit are disclosed. In one exemplary embodiment, the ODT signal generating circuit includes a latency unit and an ODT control signal generating unit. The latency unit is configured to receive a clock signal and an ODT signal. The latency unit is configured to delay the ODT signal by a predetermined time to generate a first ODT signal. The latency unit is also configured to delay the ODT signal by less than the predetermined time to generate a second ODT signal. The ODT control signal generating unit is configured to provide either one of the first and second ODT signals as an ODT control signal in response to a control signal.

Implementation of a high-definition 3D audio-visual display based on higher-order ambisonics using a 157-loudspeaker array combined with a 3D projection display

Abstract: We have implemented a high-definition 3D audiovisual reproduction system based on higher-order ambisonics (HOA) using a surrounding-157-loudspeaker array combined with a 3D projection display to reproduce information with high sense-of-presence. In this report, we introduce the system overview as well as an investigation to realize good system synchronization, which we are keenly devoted to. Results of the investigation show that all 157 channels of loudspeakers were completely synchronized within the one-sample level (48 kHz) controlled using a single PC with three Multichannel Audio Digital Interface (MADI) systems. Moreover, the latency among 157 audio signals and the video signal was only about 1.1 ms.

Electronic device audio volume reduction in response to audio communication request

Abstract: Presented is a method of facilitating use of an audio communication device, such as a telephone, in the presence of an electronic device capable of generating sound, such as a television or audio receiver. In the method, a message is received from the audio communication device, wherein the message indicates the audio communication device has received a request for an audio communication, such as a telephone call, from a second audio communication device. In response to receiving the message, an audio volume of the electronic device is reduced.

Voltage scaling systems

Abstract: A voltage scaling system is provided and includes a processor, a latency predictor, a controller, and a voltage supplier. The processor performs functions and includes a function unit with variable-latency. The function unit is divided into several power domains. When the processor performs the functions, the function unit generates a latency signal according to a current circuit execution speed. The latency predictor predicts performance of the processor according to the received latency signal to generate a predication signal. The controller compares a value of the predication signal with at least one reference value. The controller generates control signals according to the comparison result. The voltage supplier couples to a first voltage source providing a high voltage and a second voltage source providing a low voltage. The voltage supplier is switched to provide the high or low voltage to the power domains according to the control signals, respectively.

Video recording method and device

Abstract: The invention discloses a video recording method, which comprises the following steps of: respectively encoding acquired audio and video data and respectively allocating synchronous control identifiers to encoded audio frame data and encoded video frame data; and storing the audio frame data, the video frame data and the corresponding synchronous control identifiers in a cache, comparing synchronous control identifiers of the current audio frame and the current video frame in the cache and then storing the audio and video frame data in the cache according to a preset storage rule. The invention discloses a video recording device simultaneously. The method and the device of the invention can realize synchronization of the audio and video data acquired in the recording process so as to avoid asynchronism of the audio and video data caused by variation of frame rate of an image sensor or scheduling latency of a system and further promote user experience.

Parametric Packet-Layer Model for Evaluation Audio Quality in Multimedia Streaming Services

Abstract: We propose a parametric packet-layer model for monitoring audio quality in multimedia streaming services such as Internet protocol television (IPTV). This model estimates audio quality of experience (QoE) on the basis of quality degradation due to coding and packet loss of an audio sequence. The input parameters of this model are audio bit rate, sampling rate, frame length, packet-loss frequency, and average burst length. Audio bit rate, packet-loss frequency, and average burst length are calculated from header information in received IP packets. For sampling rate, frame length, and audio codec type, the values or the names used in monitored services are input into this model directly. We performed a subjective listening test to examine the relationships between these input parameters and perceived audio quality. The codec used in this test was the Advanced Audio Codec-Low Complexity (AAC-LC), which is one of the international standards for audio coding. On the basis of the test results, we developed an audio quality evaluation model. The verification results indicate that audio quality estimated by the proposed model has a high correlation with perceived audio quality.

Semiconductor device having ODT function and data processing system including the same

Abstract: To include an AL counter that outputs a second ODT signal after counting a clock signal by an additive latency after receiving a first ODT signal, and a counter control circuit that controls the AL counter such that the second ODT signal having the same logic value as a logic value of the first ODT signal at a time of shifting from an asynchronous mode to a synchronous mode is output during a period until when at least the clock signal is input by an additive latency after the shifting. With this configuration, an interruption of an CDT operation can be prevented without separately providing a CKE counter. Therefore, the circuit scale can be reduced and the power consumption can be also reduced.

Method and apparatus for controlling power supply of audio amplifier

Abstract: A method of controlling power supply voltage of an audio amplifier delays an input audio signal; estimates (STEP 440), with a digital signal processor, an audio output level of the delayed input audio signal based on correlations between the delayed input audio signal level and audio level change factors; sets (STEPS 450-476) a value of power supply voltage supplied to a power switching circuit unit in correspondence with the estimated audio output level prior to outputting the delayed input audio signal on which the estimated audio output level is based; and amplifies (STEP 480) the delayed audio input signal to provide an audio output signal by variably controlling the power supply voltage supplied to the power switching circuit unit according to the set value of power supply voltage.

Low latency transmitter path data link interface

Abstract: A data link interface can include a programmable delay chain configured to provide an amount of delay to a first clock signal that clocks a first portion of a data path. The data link interface can include a phase interpolator configured to determine an amount of phase offset applied to a second clock signal that clocks a second portion of the data path. The data link interface further can include a latency detector coupled to the programmable delay chain and the phase interpolator. The latency detector can measure a phase difference between the first and second clock signals and vary the amount of delay applied to the first clock signal and/or the amount of phase offset on the second clock signal responsive to the phase difference.

A Real-Time Framework for Video Time and Pitch Scale Modification

Abstract: A framework is presented which addresses the issues related to the real-time implementation of synchronized video and audio time-scale and pitch-scale modification algorithms. It allows for seamless real-time transition between continually varying, independent time-scale and pitch-scale parameters arising as a result of manual or automatic intervention. We illuminate the problems which arise in a real-time context as well as provide novel solutions to prevent artifacts, minimize latency, and improve synchronization. The time and pitch scaling approach is based on a modified phase vocoder with optional phase locking and an integrated transient detector which enables high-quality transient preservation in real-time. A novel method for audio/visual synchronization was implemented in order to ensure no perceptible latency between audio and video while real-time time scaling and pitch shifting is applied. Evaluation results are reported which demonstrate both high audio quality and minimal synchronization error.

Communication of audio control signals for wireless audio output devices

Abstract: A wireless audio headphone communication system has an audio input for receiving an audio signal from an audio source. A wireless transceiver receives and transmits radio frequency communications with a wireless audio headphone, the communications including an audio control signal that conforms to a wireless audio control protocol of a wireless communication protocol. A converter converts the audio control signal between the wireless audio control protocol transceived by the wireless transceiver and a local control protocol for controlling the audio source.