Cogi, Inc.

Digital Sound Corporation

Crystalvoice Communications

Mark Robert Cromack

Andreas Wittenstein

David J Brahm

Robert A Dolan

Shawn W Smith

Vijay R Raman

A process for remote control of audio streams, including capturing and digitizing a broadcasted audio stream for an event, sending the captured audio stream of the event to a server based program, providing control functions to users for processing the audio stream according to user preferences, providing Internet appliance applications to users for accessing the control functions on the server, and authenticating users to the server who are authorized to use the control functions on the server on the captured audio stream for the event.

Remote control of separate audio streams with audio authentication

This invention description details systems and methods for improving human conversations by enhancing conversation participants' ability to: &#x2014;Distill out and record core ideas of conversations. &#x2014;Classify and prioritize these key concepts. &#x2014;Recollect commitments and issues and take appropriate action. &#x2014;Analyze and uncover new insight from the linkage of these ideas with those from other conversations.

Systems and methods for conversation enhancement

A method and system for improving the efficiency of speech transcription by automating the management of a varying pool of human and machine transcribers having diverse qualifications, skills, and reliability for a fluctuating load of speech transcription tasks of diverse requirements such as accuracy, promptness, privacy, and security, from sources of diverse characteristics such as language, dialect, accent, speech style, voice type, vocabulary, audio quality, and duration.

Method and system for efficient management of speech transcribers

A method and system for improving the efficiency of real-time and non-real-time speech transcription by machine speech recognizers, human dictation typists, and human voicewriters using speech recognizers. In particular, the pacing with which recorded speech is presented to transcriptionists is automatically adjusted by monitoring the transcriptionists' output by comparing the output acoustically or phonetically to the presented recorded speech as well as monitoring the resulting transcription, and accordingly adjusting the pacing.

Method and system for efficient pacing of speech for transcription

In an improved system for receiving digital voice signals from a data network, a jitter buffer manager monitors packet arrival times, determines a time varying transit delay variation parameter and adaptively controls jitter buffer size in response to the variation parameter. A speed control module responds to a control signal from the jitter buffer manager by modifying the rate of data consumption from the jitter buffer, to compensate for changes in buffer size, preferably in a manner which maintains audio output with acceptable, natural human speech characteristics. Preferably, the manager also calculates average packet delay and controls the speed control module to adaptively align the jitter buffer's center with the average packet delay time.

Adaptive jitter buffer for internet telephony

An apparatus and method for echo cancellation in voice-messaging and voice-response systems, to enhance recognition of received DTMF and voice signals, comprising an efficient software echo canceller using adaptive digital filtering techniques. The voice messaging system includes analog telephone line interface modules which provide digitized voice data to a digital signal processor (DSP) chip. A transmit data line and a receive data line are each coupled to a cancel module with a cancel filter and an adapt/window module with an adaptive digital filter. The cancel filter causes echo cancellation on the receive data line; the adapt/window module monitors buffered transmit data in non-real time, without directly causing cancellation to occur, and selectively transfers an adjacent window of filter coefficients to the cancel filter under control of an adaptation control coupled to the adapt/window module. The control identifies a plurality of frames meeting a power criterion and passes the frames to the adaptive filter, which adapts on taps in frame segments during all available DSP real time, using a 'cycle steal' approach for testing whether additional DSP processor cycles are available to use for echo cancellation. A masked white noise burst may be used to initialize adaptation. A windowing function identifies the best taps of an adapted frame, and corresponding coefficients are copied or loaded into the cancel filter. All control parameters are configurable, enabling site-specific performance optimization.

Adaptive echo canceller for voice messaging system

An automatic gain controller for a digitized audio signal, comprising a buffer with a plurality of subframes. Each subframe contains digitized data samples of the signal, the subframes including at least one future subframe and a current subframe. Signal processing means (such as a DSP) is coupled to the memory for controlling gain of the audio signal represented by the current subframe in the buffer. The signal processing means includes means to control gain of the data samples using a stored program for computing a plurality of mean signal level values from the plurality of subframes, each mean signal level value in the plurality corresponding to one of the subframes. The program includes means for causing decay of gain on the signal represented by the current subframe when a first set of the mean values are each below a low threshold signal level. The program also has means for causing gain attack on the audio signal represented by the current subframe when a second set of the mean values are each above a high threshold signal level. The program also comprises means for shifting the data samples in one subframe in the future group into the current subframe for controlling gain of the audio signal after causing gain attack or decay. Thus, a voice messaging system under control of the program causes gain attack and decay on the audio signal represented by the subframes which attack and decay appear to occur without delay to a user of the audio signal. All control parameters are configurable. The AGC system is initialized using novel floor tracking and decay boost initialization methods which improve performance at the start of a call. Spike resistance can be set as high as desired.

Digital automatic gain control with lookahead, adaptive noise floor

In a digital messaging system (10), an apparatus (37) and method for determining whether a DTMF (Dual Tone Multi Frequency) digit is present on an input signal (11,12). A heuristics engine (37) performs a series of tests on the input signal (11,12) using a set of parameters that are stored in a RAM (38). There is one parameter for each of a set of input signal (11,12) characteristics and for each of a set of operating modes of the digital messaging system (10). All of the parameters are configurable. The configurability allows for the use of non-standard telephone equipment (13,14) and minimizes deleterious effects such as talkoff. The tests are from the set of sets comprising an absolute magnitude test, a frequency deviation test, a twist test, an echo test, a consistency test, and a temporal test.

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Santa Ynez, CA, United States of America

Mark Robert Cromack