Showing papers on "Latency (audio) published in 2003"

System and method for telephonic presence via e-mail and short message service

Abstract: A telecommunications device includes a broker module that translates telephone, mail and presence status information into short coded plain-text strings suitable for transmission over low-speed, high latency, high-cost IP networks. The broker module further transmits and receives such messages, to allow a user to monitor voicemail, e-mail, IM, and presence status, as well as control various telephone functions remotely.

Method and system for reducing latency when initiating real-time media sessions

Abstract: A method and system for reducing latency in establishment of a real-time communication session, such as an instant chat session for instance. The method and system provides for buffering an initial real-time media signal until a transmission path exists to send the signal along its way toward a receiving station. Upon establishment of the transmission path, the signal may then be sent along its way. Further, the method and system may provide 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.

Non-causal speaker selection for conference multicast

Abstract: A method for non-causal speaker selection is provided. In accordance with a particular embodiment of the present invention the method includes receiving a plurality of video streams at a multipoint control unit, each of the plurality of video streams being associated with a respective endpoint of a multipoint conference. A plurality of audio streams may also be received at the multipoint control unit, and each audio stream may be associated with a respective one of the video streams. The audio streams are buffered in respective audio buffers, and the video streams are buffered in respective video buffers. First video data is copied from the video buffers to obtain a low latency video stream for distribution to active conference participants. In a particular embodiment, second video data may be copied from the video buffers to obtain a high latency video stream for distribution to passive conference participants, the high latency video streams being delayed in time with respect to the low latency video stream.

Position location using digital audio broadcast signals

Abstract: A method, apparatus, and computer-readable media for determining the position of a user terminal comprises receiving, at the user terminal, a digital audio broadcast signal; and determining a pseudo-range between the user terminal and a transmitter of the digital audio broadcast signal based on a known component of the digital audio broadcast signal; wherein the position of the user terminal is determined based on the pseudo-range between the user terminal and the transmitter of the digital audio broadcast signal and a location of the transmitter of the digital audio broadcast signal.

Audio blend method and apparatus 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 methods.

Media communications method and apparatus

Abstract: A method of outputting multimedia content such as a video clip on a plurality of user devices (11, 12, 13) allows users of their devices to view the video clip in substantially time synchronised manner. Devices (10,11,12) communicate with each other to exchange control signals over a telecommunications network via communications links (13,14,15). The video clip is provided to user devices in advance of viewing so that during subsequent viewing only small control instructions denoting 'play', 'stop', 'pause' and the like need to be communicated between devices, where such small instructions suffer less from bandwidth constraints and latency of the communications network.

Wireless digital audio system

Abstract: A wireless digital audio system includes a portable audio source with a digital audio transmitter operatively coupled thereto and an audio receiver operatively coupled to a headphone set. The audio receiver is configured for digital wireless communication with the audio transmitter. The digital audio receiver utilizes fuzzy logic to optimize digital signal processing. Each of the digital audio transmitter and receiver is configured for code division multiple access (CDMA) communication. The wireless digital audio system allows private audio enjoyment without interference from other users of independent wireless digital transmitters and receivers sharing the same space.

A low-latency lip-synchronized videoconferencing system

Abstract: Audio is presented ahead of video in some videoconferencing systems since audio requires less time to process. Audio could be delayed to synchronize with video to achieve lip synchronization; however, the overall audio latency might then become unacceptable. We built a videoconferencing system to achieve lip synchronization with minimal perceived audio latency. Instead of adding a fixed audio delay, our system time-stretches the audio at the beginning of each utterance until the audio is synchronized with the video. We conducted user studies and found that (1) audio could lead video by roughly 50 msec and still be perceived as synchronized; (2) audio could lead video by 300 msec and still be perceived as synchronized if the audio was time-stretched to synchronization within a short period; and (3) our algorithm appears to strike a favorable balance between minimizing audio latency and supporting lip synchronization.

Audio splice process for digital Ad insertion

Abstract: A system and method for audio splicing (insertion) of an Ad audio stream in the compressed domain, where variable early delivery of the Ad audio stream and variable bit rate are allowed, without creating audio distortion, glitches, or other digital artefacts or errors, in the resultant audio stream is disclosed. The present system and method provides for a splice delay buffer which delays the first five Ad audio frames until transmission of the last frame of the primary audio stream, but before the splice time. Subsequent Ad audio frames are delayed by a fixed amount, where the fixed amount is greater than the frame delay of the primary audio stream, to allow for ease of splice back to the primary audio stream.

Digital audio and/or video streaming system

Abstract: A digital multimedia streaming system has an encoder having an input port that receives input digital multimedia (video and audio) signals and an output port that outputs encoded digital multimedia signals. The encoded digital multimedia signals are encoded from the input digital multimedia signals. The system also includes a player having an input port that receives the encoded digital multimedia signals and an output port that outputs an output digital multimedia signal. The output digital multimedia signals are decoded from the encoded digital multimedia signals. Latency between the input digital multimedia signals and the output digital multimedia signals are less than one second. The system also has a server having at least one input port, which receives the encoded digital multimedia signals from the encoder, operatively connected to the output port of the encoder, and at least one output port that outputs the encoded digital multimedia signals. A method for multimedia streaming is also disclosed.

Method for communicating information bundled in digital message packets

Abstract: A method for communicating information bundled in digital message packets via a digital network communication system is provided. The digital network communication system a sample source and each packet includes a header and a communication payload area. The method (700) includes sampling the source at a first sample rate, selecting at least one decimation of the samples based on at least one of a plurality of algorithmic data rates and a channel bandwidth, determining a packet rate based on a plurality of algorithmic latency requirements, and transmitting the digital message packet containing decimated data on the digital network.

Apparatus and method for processing multimedia audio signal for a voice call in a mobile terminal capable of receiving digital multimedia broadcasting service

Abstract: An apparatus and method for enabling a user to view a multimedia broadcasting, especially for processing a multimedia audio signal when a voice call is requested during the multimedia broadcasting. In the mobile terminal, an RF module converts a received RF signal for a voice call to a coded signal. An audio processor converts the coded signal to an electrical voice signal and outputs it through a speaker. A demultiplexer receives a digital multimedia signal and separates it into an audio signal and a video signal. A decoder decodes the audio and video signals and provides the decoded audio signal to the speaker and the decoded video signal to a display. A controller discontinues decoding the audio signal and outputs the voice signal from the audio processor through the speaker, if a voice call request is generated during receiving of the digital multimedia signal.

System and method for communicating audio data signals via an audio communications medium

Abstract: A system for communicating audio data signals comprises a source computer that performs an action, generates an event message corresponding to the action, converts the event message into an audio data signal, and communicates the audio data signal through its speaker. A source telephone receives a voice signal from a participant and the audio data signal through its microphone and communicates the audio data signal and voice as coherent sound via an audio communications medium. A recipient telephone receives the audio data signal from the coherent sound communicated via the audio communications medium and communicates the audio data signal via its speaker. A recipient computer receives the audio data signal through its microphone, extracts the event message from the audio data signal, and performs an action based on the event message from the audio data signal. The audio communications medium can comprise a telephone communications system or air.

Low latency digital audio over packet switched networks by setting buffer size at receiver

Abstract: Method and Apparatus for delivering audio signals from a source node to a destination node on a network. The apparatus uses a number of switches that transmit prioritized data on a packet network. The switches are coupled to a number of send/receive nodes for sending and receiving digital audio signals on the data network. The audio packet size and the receive buffers are sized to store a minimum possible number of audio samples to minimize latency in processing audio signals arriving at said receive node, but still ensure audio delivery without interruption due to packet data network delay. An additional feature of the invention is recovery of clock synchronization over the same data network by novel arrangement of transmission of timing packets on the network. By sending a multiplicity of packets at irregular intervals a minimum network transit delay can be determined by each of the receive nodes which allows the receive nodes to filter out packet network transit delay error and maintain accurate local clocks.

Audio signal processing method and apparatus device

Abstract: An audio signal processing method and apparatus in which the apparatus includes a plurality of digital filters, each supplied with an audio signal, and a speaker array. Outputs from the digital filters are supplied to speakers included in the speaker array to form a sound field. A predetermined delay time is set in each of the digital filters, to thereby form, in the sound field, a point where the sound pressure is higher than in the surrounding and a point where the sound pressure is lower than in the surrounding. A low-pass filter characteristic is given to the frequency response of the digital filters and a pseudo pulse train is used to enhance the setting resolution of the delay time.

System and method for digitally processing one or more audio signals

Abstract: A method for processing an audio signal is provided that includes receiving an audio signal (34a, 34b, 34c, 34d) and integrating the audio signal with a selected one of a plurality of sound effects (38). The method also includes generating an output (40) that reflects the integration of the audio signal and the selected sound effect. The output may then be communicated to a next destination.

Computer system with wireless audio signal transmitter module

Abstract: A computer system provides a wireless audio signal transmitter module which is capable of transmitting audio signal wirelessly from a host computer to at least one remote wireless signal receiver. The wireless audio signal transmitter module is connected to a sound effect interface for receiving audio signal from the computer. The audio signal is processed, modulated, and then transmitted out by an antenna. The wireless signal is received and processed by the audio signal receiver located within an effective transmission distance. The signal is then transmitted to a user via a microphone.

Switchable receiver with reduced amount of audible distortions

Abstract: A receiver and a method for receiving a digital signal and an analogue signal is described, whereby both signals transmit the same audio program In dependence of the reception situation, the receiver switches or slides from said digital signal to said analogue signal and back Each time the receiver switches between the digital signal, which has an audio bandwidth of about 12 kHz, and the analogue signal, which has an audio bandwidth of only 45 kHz, the sound of the received audio output signal changes from a rich sound to a rather flat sound, or vice versa To avoid this, the audio bandwidth of the digital signal is reduced in dependence of the reception situation In case of frequent occurrence of switching, or in case of a bad reception of the digital signal, the bandwidth of the digital signal is reduced

Loss concealment for multi-channel streaming audio

Abstract: With the advent of high-speed networks such as Internet2, high quality uncompressed transmission of multi-channel audio streams has become possible. For interactive applications, such as a distributed musical performance, minimizing latency is of paramount importance. Given the strict latency requirements, error recovery (via either retransmission or FEC) may not always be successful, and thus concealment is frequently required. In this paper we propose a novel concealment algorithm, based on inter-channel redundancy, for multi-channel, professional quality, uncompressed audio streams, with particular emphasis on an experimental 10.2 audio standard, which provides an immersive experience for the audience and the players in a performance. We also propose a smoothing method based on Bezier curves. We focus on interactive applications; thus we investigate concealment techniques that can be performed in real time. Our algorithms are implemented in a testbed capable of streaming up to 24 uncompressed audio channels with end-to-end latency of less than 6 ms. Our results show that our techniques outperform existing methods. We expect that our protocol will become an important part of a distributed immersive musical performance system currently being developed at our university.

Mobile communication terminal having a three-dimensional surround sound effect and its control method

Abstract: A mobile communication terminal with three-dimensional surround-sound effect system has a data processing section for coding and modulating a transmitting signal or demodulating and decoding a receiving signal, and an audio processing section for reproducing an audio signal input from the data processing section or transferring the audio signal input from a microphone to the data processing section. The audio signal has a first audio signal for one speaker and a second audio signal for the other speaker. A first audio path has a first amplifier for amplifying the first audio signal and a first delay for delaying in phase the first audio signal, and a second audio path has a second amplifier for amplifying the second audio signal and a second delay for delaying in phase the second audio signal. A voltage measurer measures the voltage of the first and second audio signals. A comparator compares the first and the second audio signals. An audio controller transfers at least one of the first audio signal and the second audio signal for audio reproduction on the basis of the voltages measured in the first and second audio signals.

Radio transmitter-receiver, quality-of-service control apparatus, and quality-of-service control method of transmitting data

Abstract: PROBLEM TO BE SOLVED: To improve data throughput by shortening the latency of data transmission caused by quality-of-service (QoS) control in a radio transmitter-receiver of an adaptive modulation scheme wherein an information transmission rate to a radio zone is variable. SOLUTION: In the radio transmitter-receiver having a QoS control part 11 for outputting transmitting data in the order of QoS, a QoS control mode is switched in accordance with the information transmission rate in the radio zone and when the information transmission rate in the radio zone is equal with or higher than a threshold, the QoS control of the transmitting data is omitted to supply the transmitting data to a modulation part in the order of data inputs. COPYRIGHT: (C)2004,JPO

Latency time circuit for an S-DRAM

Abstract: Latency time circuit for an S-DRAM, which is clocked by a high-frequency clock signal for producing a delayed data enable control signal for synchronous data transfer through a data path of the S-DRAM, having at least one controllable latency time generator for delaying a decoded data enable control signal with an adjustable latency time, characterized by at least one comparison circuit, which compares the cycle time of the high-frequency clock signal with a predetermined decoding time and by a signal delay circuit which can be switched on by means of the comparison circuit in order to delay the decoded data enable control signal with a predetermined delay time, in which the signal delay circuit is switched on by the comparison circuit when the cycle time of the clock signal is in a limit time region which is located about the predetermined decoding time.

Audio signal processing

Abstract: A processor for processing an audio signal can have a receiving unit configured to receive an audio signal, an expansion unit configured to expand a bandwidth of the audio signal, and a processing unit configured to process the audio signal having an expanded bandwidth for spatial reproduction.

Audio signal processing apparatus and method

Abstract: An audio signal processing apparatus (1) comprises an audio input (3) for an entered audio signal, an audio output (5) for outputting an outgoing audio signal, and a processor (9) for performing a transformation (2) to improve the intelligibility of speech present in the entered audio signal. The transformation (2) transforms the entered audio signal into the outgoing audio signal, by modeling at least one aspect of the Lombard effect, based upon a noise level value (7). The Lombard effect is a specific way in which people change their speech, when speaking in noisy environments. The audio signal processing apparatus can be applied in a television receiver and a radio program receiver.

Method for reducing latency in a push-to-talk set-up

Abstract: In the method for reducing latency in push-to-talk set up, a calling party communicates with a wireless network such that the wireless network initiates a push-to-talk operation and origination of a traffic channel for the calling party in parallel.

Distributed multimedia plays with QoS guarantees over IP

Abstract: Musical collaboration over telecommunication networks has marvellous possibilities when it comes to musical education, practise and performance. The deployment and use of high-capacity digital networks makes it possible to obtain end-to-end audio and video latency of 5-20 milliseconds, which is similar to the audio time delay typically experienced between musicians on a stage. The main challenge of reaching this latency budget over digital packet switched network such as the Internet, is how to control the queuing delay of the IP packet experienced at each router on the network paths between each participating musician. Other important aspects are video and audio codec latency, and error resilient tools needed to cope with situations where IP packets arrive too late or have been lost in the network. We give an overview of resent research (work-in-progress) in this field, including network simulations, conducted at the NTNU, and suggest how these techniques can be utilized under live concert performances.

Subband-based acoustic shock limiting algorithm on a low-resource DSP system.

Abstract: Acoustic Shock describes a condition where sudden loud acoustic signals in communication equipment causes hearing damage and discomfort to the users To combat this problem, a subband-based acoustic shock limiting (ASL) algorithm is proposed and implemented on an ultra low-power DSP system with an input-output latency of 65 msec This algorithm processes the input signal in both the time and frequency domains This approach allows the algorithm to detect sudden increases in sound level (time-domain), as well as frequencyselectively suppressing shock disturbances in frequency domain The unaffected portion of the sound spectrum is thus preserved as much as possible A simple ASL algorithm calibration procedure is proposed to satisfy different sound pressure level (SPL) limit requirements for various communication equipment Acoustic test results show that the ASL algorithm limits acoustic shock signals to below specified SPL limits while preserving speech quality

User controlled multi-channel audio conversion system

Abstract: A multi-channel audio conversion system is described, comprising audio mode converting means having a signal input and a signal output for converting audio input signals to audio output signals representing audio in an audio output mode. The audio mode converting means are arranged for user controlled conversion from the audio input signals to the audio output signals. Advantageously the influencing of the conversion from audio input signals to audio output signals may take place by the listener. This results in a smooth and personalized freedom of choice regarding the amount of multi-channel stereo and/or surround effects in the audio output signals of the system.

Method and system for buffering media to reduce apparent latency in initiating a packet-based real-time media session

Abstract: A method and system for buffering media at an initiating station. In response to a user request to initiate a packet-based real-time media session, such as a “push-to-talk” session for instance, the initiating station acquires a data connection. The station then determines that it has acquired a data connection, such as by determining that it has received incoming packet-data, and responsively begins receiving and buffering media, such as voice, from a user. Once the station successfully establishes the requested media session, or a leg of the session, the station then begins transmitting the buffered media to a remote endpoint.

System and method for communication channel and device control via an existing audio channel

Abstract: A system and method for communication channel and device control via an existing audio channel. A combined communication signal is received from a communication device comprising an audio signal and a modulated carrier signal via an audio channel. The audio signal is separated from the carrier signal. Digital data is extracted from the carrier signal. Device control is performed utilizing the extracted digital data.