Showing papers on "Latency (audio) published in 2002"

Method and system for reduction of quantization-induced block-discontinuities and general purpose audio codec

Abstract: A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms.

Digital audio signal filtering mechanism and method

Abstract: Essentially all of the processing parameters which control processing of a source audio signal to produce an encoded audio signal are stored in an audio processing profile. Multiple audio processing profiles are stored in a processing profile database such that specific combinations of processing parameters can be retrieved and used at a later time. Audio processing profiles are organized according to specific delivery bandwidths such that a sound engineer can quickly and efficiently encode audio signals for each of a number of distinct delivery media. Synchronized A/B switching during playback of various encoded audio signals allows the sound engineer to detect nuances in the sound characteristics of the various encoded audio signals.

Method of synchronizing the playback of a digital audio broadcast using an audio waveform sample

Abstract: A method is provided for synchronizing the playback of a digital audio broadcast on a plurality of network output devices by inserting an audio waveform sample in an audio stream of the digital audio broadcast. The method includes the steps of outputting a first unique signal as part of an audio signal which has unique identifying characteristics and is regularly occurring, outputting a second unique signal so that the time between the first and second unique signals must be significantly greater than the latency between sending and receiving devices, and coordinating play of audio by an audio waveform sample assuring the simultaneous output of the audio signal from multiple devices. An algorithm in hardware, software, or a combination of the two identifies the audio waveform sample in the audio stream. The digital audio broadcast from multiple receivers does not present to a listener any audible delay or echo effect.

Method of synchronizing the playback of a digital audio broadcast by inserting a control track pulse

Abstract: 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.

Method of synchronizing read timing in a high speed memory system

Abstract: The read latency of a plurality of memory devices in a high speed synchronous memory subsystem is equalized through the use of at least one flag signal. The flag signal has equivalent signal propagation characteristics read clock signal, thereby automatically compensating for the effect of signal propagation. After detecting the flag signal, a memory device will begin outputting data associated with a previously received read command in a predetermined number of clock cycles. For each of the flag signal, the memory controller, at system initialization, determines the required delay between issuing a read command and issuing the flag signal to equalize the system read latencies. The delay(s) are then applied to read transactions during regular operation of the memory system.

Method and system for selectively reducing call-setup latency through management of paging frequency and buffering of user speech in a wireless mobile station

Abstract: A method and system for selectively reducing call-setup latency. The method and system provides for selectively increasing the paging frequency used for paging certain mobile stations, so as to decrease the time that it takes to establish radio-link connectivity with those mobile stations. The method and system is particularly useful when establishing real-time communication sessions, such as instant chat sessions, but may be useful in other scenarios as well.

Echo cancellation in IP networks

Abstract: Voice transmission over IP networks imposes new DSP challenges. Package loss and latency caused by packet buffering have the fundamental influence on the speech quality. These features have also an impact on the line echo canceller (EC) performance. The constant latency greater than 60 milliseconds usually causes that an echo is very well audible even for a short echo path delay. Therefore, the EC must be deployed on each 2-wire/4-wire connection while on a conventional transmission only toll connections require EC. Such a large total delay requires short convergence time and sufficient Echo Return Loss Enhancement (ERLE). Further, package loss that is related to non-stationary intervals of speech imposes stringent requirements on EC performance in tracking statistical variations of the signal. Overally, IP telephony requires more robust and less expensive EC than conventional networks. In this paper we address the basic design issues in the EC for IP telephony. We show that classical Least Mean Square (LMS) algorithms are rather inappropriate and offer an alternative solution.

System and method for queuing and presenting audio messages

Abstract: A system and method is disclosed for queuing and presenting audio messages in a communication system (10) comprising an audio controller and audio message computer software. The audio controller receives overlapping audio messages that have portions that have been simultaneously received by the audio controller. The audio controller separately stores the audio messages in a queue in an audio buffer and then sequentially plays the messages. The audio controller may delay playing an audio message for a predetermined period of time or until the audio controller receives a control signal from a user. The user may select an audio message to be played from a list that displays the names of the senders of the audio messages. The audio controller may obtain a timestamp from a video program and associate an audio message to be played with the video program.

System and method for frame re-transmission in a broadcast communication system

Abstract: A method and apparatus for providing frame re-transmission in a broadcast communication system. Frame re-transmission is accomplished only when a predetermined number of negative acknowledgement messages are received with respect to one more data frames. The predetermined number may vary in accordance with various operating parameters, such as the latency of the transmission and/or the number of wireless communication devices currently receiving a broadcast transmission.

Inter-channel communication in a multi-channel digital hearing instrument

Abstract: A multi-channel digital hearing instrument is provided that includes a microphone, an analog-to-digital (A/D) converter, a sound processor, a digital-to-analog (D/A) converter and a speaker. The microphone receives an acoustical signal and generates an analog audio signal. The A/D converter converts the analog audio signal into a digital audio signal. The sound processor includes channel processing circuitry that filters the digital audio signal into a plurality of frequency band-limited audio signals and that provides an automatic gain control function that permits quieter sounds to be amplified at a higher gain than louder sounds and may be configured to the dynamic hearing range of a particular hearing instrument user. The D/A converter converts the output from the sound processor into an analog audio output signal. The speaker converts the analog audio output signal into an acoustical output signal that is directed into the ear canal of the hearing instrument user.

Controlling data strobe output

Abstract: A method and apparatus is provided for controlling a data strobe. A latency of operation relating to a data operation of a device is determined. A delay lock loop signal is generated for the data operation of the device. The delay lock loop signal is compared to an external clock to determine a phase difference. A determination is made as to whether the delay lock loop signal is early based upon the phase difference. A signal is adjusted in response to the latency of operation is performed based upon the determination that the delay lock loop signal is early.

Continuous bandwidth assessment and feedback for voice-over-internet-protocol (VoIP) comparing packet's voice duration and arrival rate

Abstract: A voice-over-Internet-Protocol (VoIP) application estimates bandwidth and congestion of the reception path to the VoIP application from a sending VoIP application. Packet arrivals are timed and the inter-packet delay is compared to the voice duration of the data contained in the more recent packet. When the inter-packet delay is longer than the voice duration the network is slowing and the bandwidth estimate is reduced. The bandwidth estimate is increased when inter-packet delay is smaller than the voice duration. Packet latencies are the difference in send and receive times and are compared to a moving average latency. When the current packet's latency is longer than the moving average, congestion is detected. When the current packet's latency equals the moving average, the network has recovered from congestion and the congestion estimate is reduced. Congestion and bandwidth estimates are added to packets sent out to provide feedback to the other VoIP application.

Resolution scalable video coder for low latency

Abstract: A method in a scalable encoder for reducing the average latency associated with randomly accessing an encoded digital video signal is disclosed. This method may include converting a digital video signal into a first layer having a first degree of quality and a second layer having a second degree of quality that is higher than the first degree of quality. The method may also include encoding the first layer at a first intra-frame rate, and encoding the second layer at a second intra-frame rate that is lower than the first intra-frame rate. A method in a scalable decoder for reducing the average latency associated with randomly accessing an encoded digital video signal is also provided. The method includes accessing a first layer of a digital video signal. The first layer includes the digital video signal encoded at a first degree of quality and a first intra-frame rate. The method also includes accessing a second layer of the digital video signal. The second layer includes the digital video signal encoded at a second degree of quality higher than the first degree of quality and a second intra-frame rate lower than the first intra-frame rate.

Method and system for controlling the rate of transmission for data packets over a computer network

Abstract: The present invention provides a window-based method for controlling the rate of data transmission for computer software applications transmitting data using packet switched protocols. The method is suitable for real-time applications, such as Internet video conferencing, which require low transmission latency, and which can tolerate some level of packet loss.

Semiconductor memory device having preamble function

Abstract: A semiconductor memory device includes a data controller for generating a data signal in response to data generated at an internal circuit of the semiconductor memory device when a latency signal, which sets the latency of the semiconductor memory device, is activated. The device includes an output driver for generating a data strobe signal in response to the data signal, a preamble controller for outputting a preamble control signal in response to a read command input to the semiconductor memory device, and a preamble unit for preambling the data strobe signal by changing an output signal of the output driver from a logic high level to a logic low level, when the preamble control signal is activated. Data output from the semiconductor memory device has a satisfactory preamble section.

Semiconductor memory device with detection circuit

Abstract: A detection circuit in a semiconductor memory device includes a first latch circuit and a second latch circuit. The first latch circuit latches a data strobe signal at a rise of a clock signal after a write latency passes. The second latch circuit receives an output signal of the first latch circuit at a rise of a clock signal to output a detection signal. Circuits in the semiconductor memory device are controlled by a detection signal. With such an operation applied, the semiconductor memory device grasps a correct phase difference between a data strobe signal and a clock signal, thereby enabling a normal operation.

Sound signal input/output device

Abstract: PROBLEM TO BE SOLVED: To realize an output function for an audio sound signal and a hands- free function to properly cancel echo in the condition of mixture of the audio sound signal and a voice signal. SOLUTION: A received sound signal has the sampling rate equalized to that of the audio sound signal by a first sampling rate conversion part 11 and has the sound volume controlled in a channel gain multiplication part 12 by channels corresponding to respective channels (L, R, etc.), of the audio sound signal and is mixed with the audio sound signal, and a mixed sound signal is outputted from the speakers. On the basis of the mixed sound signal converted to a low sampling rate, echo is removed from a transmission sound signal inputted from a microphone M by an echo canceller 20. The learning timing of parameters for echo removal is controlled on the basis of levels of transmission and voice sound signals and the audio sound signal by a controller 30. COPYRIGHT: (C)2003,JPO

A 2Gbps 21ch low-latency transceiver circuit for inter-processor communication

Abstract: A 20-data-channel transceiver with a control channel allows uncoded data transfer with 13ns latency. A digital DLL (Delay Locked Loop) with a ring-interpolator tracks phase with 20ps resolution. A pre-emphasis driver enables 2Gbps transmission per channel over a 7m cable at 1.5V supply. The effective full-duplex bandwidth reaches 10GB/s.

Wireless audio and data interactive system and method

Abstract: An audio and data interactive system and method includes at least one base unit and at least one remote unit. The at least one base unit has a first microprocessor-based control and the at least one student unit has a second microprocessor-based control, a speaker system and a wireless transceiver. The first control combines digital data with at least one audio signal within the human hearing range and supplies the combined digital data and audio signal to the first transceiver. The second control separates the combined digital data and audio signal from the second transceiver into the digital data in the audio signal. The second control supplies the audio signal to the speaker system and causes the second wireless transceiver to transmit in response to the digital data.

Method and apparatus for improving automotive audio system performance

Abstract: An adaptive audio system that determines the status of various portal covers such as vehicle doors, windows, sun roofs, hatchbacks, and the like and uses this status information to modify properties of the audio output signal. By adapting the audio output signal of the audio system to the position or status of various vehicle portal covers, the quality and performance of the audio system can be further optimized.

Data-driven software architecture for digital sound processing and equalization

Abstract: A digital sound processing design system for a vehicle audio system includes a computer and a design tool that is run by the computer. The design tool allows a user to define sound processing criteria that is stored in a template file. An audio signal processor is connected to the first and second real channel inputs of an audio source. Memory that is coupled to the audio signal processor stores the template file. The sound processing engine that is coupled to the audio signal processor and the memory reads the template file at run-time to obtain the sound processing criteria. The sound processing engine applies the sound processing criteria to the first and second real channel inputs. The design tool allows a user to create virtual channel inputs and outputs that are based, in part, on the first and second real channel inputs.

Synchronizing voice and video transmitted over separate channels

Abstract: Reproduction of voice signals and video signals in a video telephony call are synchronized at the receiving end. Voice signals are transmitted from a first telephone device of a first party to a second telephone device of a second party via a public switched telephone network (PSTN) connection. The video signals are transmitted from a first computer of the first party to a second computer of the second party via packets in a computer data network. The video signals comprise successive video frames. A video latency time is determined for transmission of the video signals over the computer data network. If the video latency time is in a first predetermined range then transmission of the voice signals is delayed from the first telephone device to the second telephone device by a selected delay in response to the video latency time. The selected delay is no greater than a predetermined maximum voice delay.

Audio signal processor

Abstract: An audio signal processor performs a signal processing on the first audio signal and the second audio signal in correspondence with an operation to produce an output. The processor comprises and operation device, a signal processing device and a control device. The operation device is operatively movable between one end and the other end. The one end is associated with the first audio signal and the other end is associated with the second audio signal. The signal processing device performs a signal processing on any one of the first audio signal and the second audio signal to provide at least one kind of predetermined effect. The control device controls the signal processing device so that the effect is provided for any one of the first audio signal and the second audio signal at a predetermined timing based on a position of the operation device and a moving direction thereof.

Audio/video speedup system and method in a server-client streaming architecture

Abstract: A system and a corresponding method for temporal modification of audio signals, to increase or reduce the playback rates of an audio and/or a video file in a client-server environment. The system and method improve the efficiency of serving streaming media to a client so that the client can select an arbitrary time-speedup factor. The speedup system performs many of the pre-calculations once, at the server, so that the bandwidth needs are reduced and the client's computational load is minimized. The final time-scale-modification can be either done completely on the server, thus reducing the client's needs, or partly on the client's computer to minimize latency, and to reduce on-the-fly computational load from the server that serves multiple clients concurrently.

A 1.3 GSample/s 10-tap full-rate variable-latency self-timed FIR filter with clocked interfaces

Abstract: A 6 b 10-tap digital FIR filter has a self-timed datapath, clocked interfaces, and variable latency. The architecture of the filter is full rate, distributed arithmetic with signed-digit offset binary (SDOB) number representation. The 0.45 mm/sup 2/ circuit, in 0.18 μm CMOS technology, is operational from 1.2 V to 2.1 V power supply, and has 80 mW dissipation at 300 MSample/s and 4 cycles of latency, and 500 mW at 1.3 GSample/s and 7 cycles of latency.

Audio video reproduction apparatus, audio video reproduction method, program, and medium

Abstract: An audio video reproduction apparatus having: an audio speed conversion circuit of performing audio speed conversion on a generated audio signal in such a manner that the information on an A-PTS is maintained, depending on an executing reproduction speed. An audio output buffer circuit of accumulating the audio signal having undergone the audio speed conversion and of outputting the signal according to the executing reproduction speed. A video output buffer circuit of accumulating and outputting a generated video signal. A comparison circuit of comparing the A-PTS with a V-PTS by using the maintained information on the A-PTS; wherein using the result of the comparison, the video output buffer circuit outputs the accumulated video signal according to the executing reproduction speed.

Arbitrated audio communication with reduced latency

Abstract: An arbitrated communication system provides reduced audio latency by transmitting audio with transmission of an access request. To obtain access to a broadcast link, a request for access is transmitted to an arbitration controller. The user does not wait for an indication that the access request has been granted. Instead, the audio is transmitted following or integrated with transmission of the request. If the request is denied, the audio is discarded. In most instances, however, the request will be granted. As a result, the audio can be transmitted relatively immediately, significantly reducing latency in the system.

An Audio Architecture Integrating Sound and Live Voice for Virtual Environments

Abstract: : The purpose behind this thesis was to design and implement audio system architecture, both in hardware and in software, for use in virtual environments The hardware and software design requirements were aimed at implementing acoustical models, such as reverberation and occlusion, and live audio streaming to any simulation employing this architecture, Several free or open-source sound APIs were evaluated, and DirectSound3DTM was selected as the core component of the audio architecture, Creative Technology Ltd, Environmental Audio Extensions (EAXTM 3,0) were integrated into the architecture to provide environmental effects such as reverberation, occlusion, obstruction, and exclusion, Voice over IP (VoIP) technology was evaluated to provide live, streaming voice to any virtual environment DirectVoice was selected as the voice component of the VoIP architecture due to its integration with DirectSound3DTM, However, extremely high latency considerations with DirectVoice, and any other VoIP application or software, required further research into alternative live voice architectures for inclusion in virtual environments Ausim3D's GoldServe Audio System was evaluated and integrated into the hardware component of the audio architecture to provide an extremely low-latency, live, streaming voice capability.

Methods and apparatus for selectively including an audio signal component within an audio output signal

Abstract: An apparatus and method selectively includes an audio signal component, such as hold music or a hold tone, within an audio output signal, such as a composite audio signal from a multi-party teleconference, based upon a condition of an audio delivery system. In one arrangement, the condition is based upon a presence characteristic, indicating the number of active endpoint devices in the audio delivery system and upon an endpoint signal, indicating non-participation of one of the active endpoint devices. Based upon the condition of the audio delivery system, the audio delivery system either omits the audio signal component from the audio output signal or includes the audio signal component with the audio output signal distributed within the system.

Method and apparatus for latency specific duty cycle correction

Abstract: The illustrated embodiments relate to a control circuit that uses a latency signal to generate an output signal. The latency is used to create a control signal that is dependent on the latency signal. The control signal is used to select from among multiple input sources. The selected input source is used to create an output signal.