Cisco Technology, Inc.

Kevin J Connor

Ramanathan T Jagadeesan

Bich Nguyen

Luke K Surazski

Ilya Slain

Michael E Knappe

David R Oran

Rajesh Kumar

Xiaode Xu

Ilya Umansky

James C Frauenthal

Mohammed Taher Shaikh

Radhika Padmanabhan

Paul Volkaerts

An audio and/or visual transmission quality device estimates transmission quality as perceived by a user based on observation and processing of objective metrics. The device preferably includes a primary integrator to generate an instantaneous estimate of user annoyance based on occurrence of frame loss events in a received signal. The device further preferably includes secondary integrators to process the instantaneous estimate of user annoyance in selected time intervals and generate a plurality of time-varying estimates of user annoyance valid over the selected time intervals. The device may further include a plurality of scaling modules to scale the plurality of time-varying estimates and generate a corresponding plurality of estimates of transmission quality as perceived by the user. The device may further include a modulation network to modify a sensitivity of the primary integrator to the occurrence of frame loss events in the received signal.

Device and method for estimating audiovisual quality impairment in packet networks

A device and method provides a means for monitoring a media segment of a Real-time Transport Protocol (RTP) media stream without interfering with end-to-end monitoring of the RTP media stream. The device includes a media segment monitor to generate segment control messages associated with a selected segment of the RTP media stream transmitted between a source endpoint and a destination endpoint in a packet network. The device further includes an interface to transmit and receive the segment control messages; and a processor to process the segment control messages, the segment control messages including call quality metrics related to the selected segment of the RTP media stream.

Media segment monitoring

A method for a client to receive streaming media over a network includes receiving data having the streaming media encoded therein and storing the received data in a de-jitter buffer thereby increasing a fullness of the de-jitter buffer. The method further includes, when the fullness reaches a fill level, initiating play out of the stored data from the de-jitter buffer, and changing the fill level while playing out the stored data.

Devices and methods for minimizing start up delay in transmission of streaming media

An embodiment may include an apparatus comprising a dejitter buffer to receive packets containing audio data, a codec coupled with the dejitter buffer, the codec to receive coded audio frames from the dejitter buffer and decode them, and a concealed seconds meter coupled with the dejitter buffer, the concealed seconds meter to record concealment events by the decoder to provide an objective measure of media impairment. Another exemplary embodiment may be a method comprising receiving packets containing audio information at a dejitter buffer, decomposing the packets to coded audio frames, sending the coded audio frames to a decoder and decoding the frames, generating a concealment output stream if the decoder does not receive a valid frame from the dejitter buffer, and recording concealment events to provide an objective measure of media impairment.

Characterization of impaired intervals in a voice over packet session using audio frame loss concealment

Devices and methods are provided for minimizing the startup delay of streaming media transmitted through networks. A server maintains a portion of the media stream stored in an initial burst transmit buffer. At startup, the stored portion is transmitted at a rate higher than the fixed frame rate, exploiting the full available bandwidth. The initial burst transmission fills up the de-jitter receive buffer at the receiving end faster, thereby shortening the startup delay. Then transmission is switched to the regular rate, from the regular buffer. A variable bit rate transcoder is optionally used for the data of the initial transmission. The transcoder diminishes the size of these frames, so they can be transmitted faster. This shortens the start up delay even more. A receiver has a buffer with a fill level started at a value lower than a final value. This triggers the beginning of play out faster, further shortening the delay time. The fill level of the de-jitter receive buffer is then gradually increased to a desired final value.

A server is provided with a transmit buffer, a transcoder, and a redundancy encoder. The server is for transmitting streaming media to a client through a communication network. A control unit senses when the network is temporarily congested, and activates the transcoder and the redundancy encoder. The frames require fewer bits, and thus arrive in the client's de-jitter buffer without interruption, notwithstanding the network congestion. This prevents network jitter.

Server and method for transmitting streaming media to client through a congested network

A test system measures performance of telephone network echo cancellers using a primary criterion of estimated user annoyance due to audible returned echo. The invention generates live telephone calls, uses real speech samples as stimulus signals and provides tail-circuit emulation using actual measured telephone tail-circuit impulse responses. These features provide better 'real-life' test conditions for the echo canceller system under test than current ITU standard test methods. Two methods are employed for echo canceller performance evaluation via metrics of estimated user annoyance due to echo. Energy-based method employs point-by-point comparison of talker speech and talker echo signal energy envelopes and uses variable energy thresholds for estimation of echo audibility. A perceptual-model based method uses a Perceptual Speech Distortion Metric (PSDM), such as ITU P.861, in an unique configuration to estimate user annoyance due to audible echo. Echo canceller performance is tested under both single-talk and double-talk conditions. Innovative application of the PSDM method in double-talk tests allow estimation of quality of received double-talk speech.

Method and apparatus for testing echo canceller performance

A Perceptual Speech Distortion Metric (PSDM) generates perceptual distortion values for voice prompts received from a voice response system by comparing the received voice prompts with reference signals associated with the same states in the voice response system. The perceptual distortion values identify the voice prompts as either correct or incorrect responses to signal generator inputs and also quantify an amount of perceptual distortion in the voice prompts.

System for automated testing of perceptual distortion of prompts from voice response systems

Multiple audio streams are spatially separated with a context switching system to allow a listener to mentally focus on individual point sources of auditory information in the presence of other sound sources. The switching system simultaneously directs incoming sound sources to different spatial processors. Each spatial processor moves the received sound sources to different audibly perceived point sources. The outputs from the spatial processors are mixed into a stereo signal with left and right outputs and then output to the listener. Important sound sources are moved to a foreground point source for increased intelligibility while less important source sources are moved to a background point source.

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Sunnyvale, CA, United States of America

Kevin J Connor