Showing papers on "Latency (audio) published in 2000"

Voice-over-ip interface for standard household telephone

Abstract: The present invention enables a traditional analog telephone to be used with VoIP applications. For example, the user could connect their standard 900 MHz telephone to this invention, establish a VoIP call and enjoy the freedom of movement their cordless telephone provides. The preferred embodiment of the present invention minimizes overhead to the host computer via a dual CODEC modem that incorporates a DSP capable of simultaneous communication with the two CODEC modules. This architecture facilitates a latency and communication overhead reduction as the analog voice signals effectively “stream” from the first CODEC to the second CODEC.

Method and apparatus for pre-fetching audio data

Abstract: An audio system includes a memory storing audio data and an audio signal processor for processing the audio data. Addressing circuitry addresses the memory and a pre-fetch storage area stores data for a current address and for one or more following addresses to hide memory access latency during address changes of the addressing circuitry.

In-band signaling for synchronization in a voice communications network

Abstract: Methods for determining a system latency of an audio call path of a voice communications network, and for synchronizing a remote unit (108) with a reference oscillator of a reference station (102) involve transmitting a reference signal (106) over the audio call path from the reference station (102) to the remote unit (108), where a reply signal (112) is generated and transmitted back to the reference station (102) over the call path after a preselected reply delay interval (tdet). A round-trip time difference (tRT) is used to determine total system latency, which is then taken into account in synchronizing the remote unit (108) with the reference oscillator. The reference and reply signals (106, 112) are generated as audio-frequency signals resembling human voice sounds to avoid destructive attenuation by the voice communications network. One embodiment includes a wireless telephone unit having an on-board SPS receiver. The SPS receiver includes an oscillator that can be synchronized using the method to improve performance of the SPS receiver. Convenient and efficient methods of synchronization and location data reporting within existing wireless communication network infrastructures are disclosed.

An efficient algorithm for sample rate conversion from CD to DAT

Abstract: This letter unveils an efficient algorithm for sampling rate conversion (SRC) technique from 44.1 kHz compact disc (CD) to 48 kHz digital audio tape (DAT). This method involves upsampling the input signal by two, and then passing the interpolated signal through a fractional delay filter that employs a simple decimation. This method can also be used for SRC from DAT to CD without changing the filter coefficients. The proposed algorithm is simulated in Matlab and can be implemented in a realtime digital signal processor (DSP). Compared with other existing methods, the proposed method has the advantage that it requires fewer million instructions per second (MIPS) and memory.

Method and apparatus for voice latency reduction in a voice-over-data wireless communication system

Abstract: A method and apparatus for reducing voice latency in a voice-over-data wireless communication system. In a transmitter (100), audio information to be transmitted is first encoded using a voice encoder (104) operating in a first mode of operation. At least one operating parameter in the transmitter is measured by a processor (118), the at least one operating parameter proportional to a latency between the transmitter (100) and a receiver. If the at least one operational parameter exceeds a predetermined threshold, the voice encoder (104) is instructed to operate in a second mode of operation. When the at least one operational parameter falls below a second predetermined threshold, the voice encoder (104) is instructed to operate in the first mode of operation.

A simplified approach to high quality music and sound over ip

Abstract: Present systems for streaming digital audio between devices connected by internet have been limited by a number of compromises. Because of restricted bandwidth and “best effort” delivery, signal compression of one form or another is typical. Buffering of audio data which is needed to safeguard against delivery uncertainties can cause signal delays of seconds. Audio is in general an unforgiving test of networking, e.g., one data packet arriving too late and we hear it. Trade-offs of signal quality have been necessary to avoid this basic fact and until now, have vied against serious musical uses. Beginning in late 1998, audio applications specifically designed for next-generation networks were initiated that could meet the stringent requirements of professional-quality music streaming. A related experiment was begun to explore the use of audio as a network measurement tool. SoundWIRE (sound waves over the internet from real-time echoes) creates a sonar-like ping to display to the ear qualities of bidirectional connections. Recent experiments have achieved coast-to-coast sustained audio connections whose round trip times are within a factor of 2 of the speed of light. Full-duplex speech over these connections feels comfortable and in an IIR recirculating form that creates echoes like SoundWIRE, users can experience singing into a transcontinental echo chamber. Three simplifications to audio streaming are suggested in this paper: Compression has been eliminated to reduce delay and enhance signal-quality. TCP/IP is used in unidirectional flo ws for its delivery guarantees and thereby eliminating the need for application software to correct transmission errors. QoS puts bounds on latency and jitter affecting long-haul bidirectional flo ws.

Systems and methods for automatically adjusting the volume of an audio source

Abstract: A system adjusts audio levels so as to maintain a constant perceived audio level at a user's location. The system includes a sensor ( 110 ) and an audio device ( 120 ). The sensor ( 110 ) receives a first audio signal, receives at least one data packet ( 1000 ) comprising a second audio signal, determines whether a difference between an average volume level of the first audio signal and the second audio signal exceeds a threshold value, generates a response data packet ( 1200 ) including a volume adjustment command when the difference between the average volume level of the first audio signal and the second audio signal exceeds the threshold value, and transmits the response data packet ( 1200 ). An audio device ( 120 ) transmits the first audio signal, transmits the at least one data packet ( 1000 ) to the sensor ( 110 ), receives the response data packet ( 1200 ), and adjusts an audio level based on the response data packet ( 1200 ).

Method and apparatus for latency reduction in low power two way communications equipment applications in hybrid fiber coax plants

Abstract: A two way communication system is adapted to reduce latency while the communications system is operating in a low power mode. The two way communication system includes a local host having a first primary communication channel and a secondary out of band transmitter; and customer premise equipment having a primary communication channel for communicating with the first primary communication channel of the local host and a secondary low power out of band receiver that receives out of band control signals from the out of band transmitter during low power operation of the customer premise equipment.

Selection of alternate sources for audio broadcast signals responsive to embedded source identification information

Abstract: An apparatus, program product, and method utilize embedded source identity information within an audio broadcast signal to facilitate the reception of the audio broadcast signal from an alternate source. Such embedded information may be used, for example, to facilitate the automated selection of an alternate source of an audio broadcast signal, e.g., in response to poor reception of the primary source of the audio broadcast signal, so that a listener is less likely to miss any portion of an audio broadcast. In one particular implementation, an audio broadcast signal is a radio signal broadcast by a radio station, whereby suitable alternate sources might include another radio station that broadcasts the audio broadcast signal over a different frequency, a transmission device accessible over a telephone network, or a transmission device accessible over a computer network such as the Internet.

A 550 MSample/s 8-tap FIR digital filter for magnetic recording read channels

Abstract: With the advent of partial response maximum likelihood (PRML) equalization of magnetic recording channels, spectral shaping of read back signal is typically performed by combination of a continuous time and a digital FIR filter. Digital FIR filters are scalable with technology and can be adapted to provide a desired channel response typically using the least mean square (LMS) algorithm in commercial read channels. For these reasons, use of continuous time filters for this application is gradually diminishing. Efficient timing recovery in a read channel is critical for fast phase and frequency acquisition in addition to acceptable bit error rate (BER) performance. Therefore, it is also critical that these filters are implemented with as little latency as possible since their output is used to extract timing information.

Scalable Connectivity Processor for Computer Music Performance Systems

Abstract: Standard laptop computers are now capable of sizeable quantities of sound synthesis and sound processing, but low-latency, high quality, multichannel audio I/O has not been possible without a cumbersome external card cage. CNMAT has developed a solution using the ubiquitous 100BaseT Ethernet that supports up to 10 channels of 24-bit audio, 64 channels of sample-synchronous control-rate gesture data, and 4 precisely time-stamped MIDI I/O streams. Latency measurements show that we can get signals into and back out of Max/MSP in under 7 milliseconds. The central component in the device is a field programmable gate array (FPGA). In addition to providing a variety of computer interface capabilities, the device can function as a cross-coder for a variety of protocols including GMICS. This paper outlines the motivation, design, and implementation of the connectivity processor. 1. Context and Prior Work Hardware development for computer music performance systems has followed the standard pattern of technology evolution passing through the first two phases of design focus, function and price, to the final phase: usability. We have identified size and connectivity as primary usability issues for computer music performance systems. Laptop computers have recently become available with fast signal processing capabilities at a moderate cost and although their size makes them very attractive for musical performance, their constrained expansion capabilities limit connectivity. Currently there are no commercially available, low-latency, high-reliability, compact, multi-channel audio solutions for laptop computers. Furthermore the architecture of current laptops and most computers makes it impossible to synchronize acquired gestural data and sound I/O to satisfy the low latency/jitter needed for satisfactory reactive performance systems: 10±1ms (Freed, et al., 1997). The 10 ms latency criterion is not difficult to meet, but a maximum latency variation of ±1ms is difficult to achieve, especially when the stimulus gesture is represented as a MIDI event or a low rate signal from a nonsample-synchronous input source like a data acquisition card rather than a sample-synchronized audio input signal. The only computers with the requisite unified clock management and operating systems support for such tight synchronization are from Silicon Graphics. Unfortunately, even the smallest configurations of their machines, the O2 and Octane are too large and expensive for most performing musicians. One of our key design goals was to eliminate virtually all latency variation in low sample rate inputs like those from gestural input devices.

Method for dynamic adjustment of audio input gain in a speech system

Abstract: A method for adjusting audio input signal gain in a speech system can include seven steps. First, an upper and a lower threshold can be predetermined in which the upper and lower threshold define an optimal range of audio data signal amplitude measurements. Second, a frame of unpredicted digital audio data samples can be received. Each sample can indicate an amplitude measurement of the audio data signal at a particular point in time. Third, a maximum signal amplitude can be calculated for a configurable measurement percentile of the unpredicted digital audio data samples. Fourth, the audio input signal gain can be incrementally adjusted downward if the maximum signal amplitude exceeds the upper threshold. Conversely, fifth, the audio input signal gain can be incrementally adjusted upward if the maximum signal amplitude falls below the lower threshold. Sixth, additional frames of unpredicted digital audio data samples can be received. Finally, seventh, each of the third through the sixth steps can be repeated with the received additional frames until the calculated maximum signal amplitude falls within the optimal range of audio signal amplitude.

Method and apparatus for active latency characterization

Abstract: Briefly, in accordance with one embodiment of the invention, a method for actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least two signal streams for a waveform are created in the audio channel. The presence of the first signal sample stream for the waveform and the second signal sample stream for the waveform is detected at a point in the audio channel. The time between the detections of the signal sample streams is measured. Briefly, in accordance with another embodiment of the invention, a method of actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least a first and a second waveform are created in the audio channel. The presence of the first and the second waveform are detected at a point in the audio channel. The time between the detections of the waveforms is measured.

Personalized incoming call signal for communication devices

Abstract: Apparatus and methods are presented to allow the creation of a personalized audio signal for a communication device., An option to record audio input and create a call signal audio file is selected via an input mechanism (203). Audio input is recorded when a record button (204) is pressed and the recording is terminated when the record button (204) is pressed a second time. Processing circuitry (220) optionally applies audio compression, filtering and encoding algorithms to said audio input and creates a call signal audio file. The call signal audio file is then stored in the memory circuitry designated for call signal audio files (210). Additional audio output circuitry (207) plays the call signal audio file when an incoming call is detected by the transceiver (201).

The AHI: an audio and haptic interface for contact interactions

Abstract: We have implemented a computer interface that renders synchronized auditory and haptic stimuli with very low (0.5ms) latency. The audio and haptic interface (AHI) includes a Pantograph haptic device that reads position input from a user and renders force output based on this input. We synthesize audio by convolving the force profile generated by user interaction with the impulse response of the virtual surface. Auditory and haptic modes are tightly coupled because we produce both stimuli from the same force profile. We have conducted a user study with the AHI to verify that the 0.5ms system latency lies below the perceptual threshold for detecting separation between auditory and haptic contact events. We discuss future applications of the AHI for further perceptual studies and for synthesizing continuous contact interactions in virtual environments.

Audio transmission apparatus

Abstract: In an audio transmission apparatus used in an asynchronous communication network, a receiving buffer stores temporarily an audio packet received in a network interface section. This audio data is decoded in an audio decoder, and is passed through a D/A converting speed changer, and the digital audio data is converted into analog in a D/A converter. Concurrently, in a delay time fluctuation measuring section, the delay time fluctuation of audio packet received is measured in the network interface section, and a buffer controller determines the data storage amount of the receiving buffer on the basis of this delay time fluctuation. Accordingly, the buffer controller changes the D/A converting speed on the basis of the delay time fluctuation, thereby adjusting the audio data flow from the receiving buffer. Accordingly, a real-time audio transmission apparatus capable of avoiding lack of audio data in the buffer and reproducing audio continuously is presented.

System for compression and decompression of audio signals for digital transmission

Abstract: A audio digital CODEC is provided with various parameters that when changed affect the quality of the resultant audio. These psycho-acoustic parameters include the standard ISO parameters and additional parameters to aid in effecting a pure resulting audio quality. The psycho-acoustic parameters located in the audio digital CODEC can be monitored and controlled by the user. The parameters can be monitored by a speaker associated with the CODEC or headphones. The user can control the adjustment of the psycho-acoustic parameters through the use of knobs present on the front panel of the CODEC or graphic or digital representations. Adjustment of the parameters will provide real time change of the resulting audio sound that the user can monitor through the speaker or the headphones. Selections may also be made to connect to a plurality of transmission facilities.

Apparatus and method for reducing latency and buffering associated with multiple access communications systems

Abstract: A multiple access communications system includes a communications unit, such as satellite or a mobile link base station, and an antenna disposed thereon. The antenna has the capability to direct an antenna beam at a selected user on a packet by packet basis thereby reducing the latency, buffering, and non-uniform gain distribution associated with conventional transmission systems.

Audio blend method, transmitter and receiver for am and fm in band on channel digital audio broadcasting

Abstract: A method is provided for processing a composite digital audio broadcast signal to mitigate intermittent interruptions in the reception of said digital audio broadcast signal. The method includes the steps of separating an analog modulated portion of the digital audio broadcast signal from a digitally modulated portion of the digital audio broadcast signal, producing a first plurality of audio frames having symbols representative of the analog modulated portion of the digital audio broadcast signal, and producing a second plurality of audio frames having symbols representative of the digitally modulated portion of the digital audio broadcast signal. The first plurality of audio frames is then combined with the second plurality of audio frames to produce a blended audio output. A method is also provided for transmitting a composite digital audio broadcast signal having an analog portion and a digital portion to mitigate intermittent interruptions in the reception of said digital audio broadcast signal. The method comprises the steps of arranging symbols representative of the digital portion of the digital audio broadcast signal into a plurality of audio frames, producing a plurality of modem frames, each of the modem frames including a predetermined number of the audio frames, and adding a frame synchronization signal to each of the modem frames. The modem frames are then transmitted along with the analog portion of the digital audio broadcast signal, with the analog portion being delayed by a time delay corresponding to an integral number of the modem frames. The invention also encompasses radio receivers and transmitters which process signals according to the above method.

Low latency for start of initial automatic gain control for data in digital signal processing

Abstract: An apparatus and method for controlling the AGC in a receiver is described. Samples of the input signal are compared to the upper and lower threshold values which are defined by the dynamic range of the A-to-D converter. These samples are recorded and used in determining whether to count-up or count-down in counters prior to the time the signal is detected. These counts provide, in effect, a history of what has occurred prior to signal detection and are used in computing an AGC gain. The gain can be computed more quickly since there is zero latency in starting the calculation for correcting the AGC.

Voice quality improvement for voip connections on low loss network

Abstract: The type of audio stored in the payload of a data packet transmitted over a data network is identified as speech audio or non-speech audio through the use of a non-speech identifier included in a header in the data packet. Upon detection of data packet containing non-speech audio, the receiver of the data packet may modify jitter buffer latency while the non-speech audio is being received. Modifying the jitter buffer latency while non-speech audio is being received minimizes the loss of spoken words during jitter buffer latency modification.

Digital speech processing system

Abstract: In a speech processing system ( 10 ) characterized by a finite range of audio levels, the speech processing system ( 10 ) receiving an incoming audio signal, the speech processing system amplifying ( 12 ) the incoming audio signal by an audio gain factor, the speech processing system ( 10 ) representing the amplified audio signal by the finite range of audio levels, a method for adjusting the audio gain factor, including the steps of: decreasing the audio gain factor when detecting clipping of the amplified audio signal, maintaining the audio gain factor for a hold time period, and increasing the gain factor when detecting that the result of amplification of the incoming sound levels by the audio gain factor, is lower than the highest level of the finite range of audio levels.

Device for bi-directional transmission of audio and/or video signals

Abstract: Proposed by the invention is a device for the bidirectional transfer of audio and/or video signals, in particular in the context of sound and/or image reports, with: at least one means (6) for providing an audio input signal; a first mixing device (10), which is connected to the means (6) for providing the audio input signal and which is designed to output a mixed audio transmission signal; a transmission and/or reception device coupled to the first mixing device (10) for transmitting the mixed audio transmission signal and/or receiving an audio reception signal; a control device (22) coupled to the first mixing device (10) for controlling the first mixing device (10); a compression and/or decompression device (12, 32) for compression of the mixed audio transmission signal or, as the case may be, for decompression of the audio reception signal, which compression and/or decompression device is connected to the first mixing device (10) for taking up the mixed audio transmission signal or, as the case may be, to at least the second mixing device (30) for delivering a decompressed audio reception signal, and connected to the transmission and/or reception device for delivering a compressed audio transmission signal or, as the case may be, for taking up the audio reception signal; and at least one means (26), connected to the second mixing device (30), for reproducing an audio output signal, which in particular contains the decompressed audio reception signal; in which device provision is made for at least one mobile-radio and/or mobile-telephone-network channel as a transfer channel.

An audio processor card for special sound effects

Abstract: The design of an audio processor card for generating special sound effects is presented. This is designed as an add-on card that plugs directly into the ISA bus of a PC. The card uses a Motorola DSP processor, DSP56001, for audio signal processing. External SRAM modules are used for program and data storage. A codec chip is employed to handle the digital interfacing between the DSP processor and the analog audio world.

Auto latency test tool

Abstract: A system and method for measuring latency in voice communications is presented. In an exemplary embodiment, latency is measured by establishing a call between a first and a second telephony device, and measuring a latency between a signal originating at the first telephony device and the signal as it arrives at the second telephony device.

Information processing system, wireless transmitter and electronic equipment

Abstract: PROBLEM TO BE SOLVED: To provide an information processing system employing a wireless signaling transmission with a simple and easy-to-use configuration and a reduced cost. SOLUTION: A transmission device 2 functions as a remote commander of an audio unit 3 in a stand-alone operation, while functions in an operation by connecting to a personal computer 1 as a wireless transmitter for transmitting an ultra-red signal formed of a transmission signal including audio data, control data, and text data being output from the personal computer 1. The remote control signal or the transmission signal transmitted from transmission device 2 is received by a photodetector 3PD of the audio unit. The remote control signal is used for controlling the audio unit, etc. The transmission signal is separated into audio data, control data and text data to forward to respective processes.

Method and apparatus for noise suppression of received audio signal in a cellular telephone

Abstract: In a cellular telephone, a method of reducing background noise of an audio signal resulting from a received signal. Each digital sample of the audio signal is used to calculate an average energy of the audio signal over a number of samples. When the average energy is less than a threshold, a gain factor is calculated using the average energy and the threshold. The amplitude of the audio sample is then adjusted according to the gain factor.

Multiplexing device and method, and decoding device and method

Abstract: PROBLEM TO BE SOLVED: To select whether the transmission delay time of an audio signal is shortened or not by outputting a video signal and the audio signal synchro nously or asynchronously with each other. SOLUTION: A transmitter 2 multiplexes a video stream and an audio stream to generate a transport stream. In such a case, the differential time ΔTx between video encode delay time ΔT1 and audio encode delay time ΔT2 is obtained and transmitted as PSI/SI information. At a receiver side, a synchronous reproduction mode that the video signal and the audio signal are outputted synchronously, and an audio preference mode that the video signal and the audio signal are outputted asynchronously are set. For instance, when the audio preference mode is set, the PTS of the audio signal is corrected on the basis of the differential time ΔTx, and the reproduction time of the audio signal is hastened. COPYRIGHT: (C)2002,JPO

Portable electronic device updated via broadcast channel

Abstract: A system is provided for transmitting generally static media. The system ( 10 ) comprising an electronic device ( 12 ) and a digital audio transmitter ( 14 ). The electronic device ( 12 ) has a CPU ( 16 ), a storage medium ( 18 ), a display ( 20 ), a user interface ( 22 ), and a digital audio broadcast receiver ( 26 ). The digital audio transmitter ( 14 ) has a specialized broadcast server ( 30 ). The digital audio broadcast receiver ( 26 ) receives and decodes the digital audio signal transmitted by the digital audio transmitter ( 14 ).

Synchronous semiconductor memory device including an output controlling circuit with reduced occupying area

Abstract: A read enable signal OEMF activated in response to an input command is applied to an N minus 2 clock shift circuit included in an output control circuit for implementation of ZCAS latency. An output signal of the N minus 2 clock shift circuit and an internal mask instructing signal activated in response to an external mask instructing signal are logically processed and applied to a one-clock shift circuit. According to an output signal OEMQM of one-clock shift circuit, a data output enable signal OEM controlling activation/inactivation of an output buffer circuit is activated/inactivated. Data output controlling portion occupying area of a synchronous dynamic random access memory is reduced and timings of activation/inactivation of data output by different commands are made the same.