Abstract:
A dynamic optical microphone system may include an acoustic microphone that receives an audio signal and a laser microphone that transmits a laser beam and receives optical feedback from a human struck by the laser beam. The system may include a depth sensor that determines a distance to the human and a camera that tracks human faces. A processor may be communicatively coupled to the acoustic microphone, laser microphone, depth sensor, camera, and a memory storing computer executable instructions. The processor may determine a direction to a human, direct the laser beam at a voice box of the human, determine a distance to the human using the depth sensor, adjust an intensity of the laser beam based on the distance, receive optical feedback and isolate a voice signal through the optical feedback from background noise in the audio signal.
Abstract:
A two-way conversation assisting device includes a first microphone that enters a first voice, a first loudspeaker that outputs the first voice, a second microphone that enters a second voice, a second loudspeaker that outputs the second voice, and a first echo and crosstalk canceller. The first echo and crosstalk canceller estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and removes the calculated first interference signal from an output signal of the first microphone.
Abstract:
In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem in the specific situation involving store-and-forward systems (e.g. voicemail), the invention provides a novel method and apparatus including a plurality of different vocoders that can be selectively invoked to process the voice signal so as to reduce signal degradation. Also, the apparatus has the capability to bypass the vocoder bank when exchanging data with a remote signal processor capable of accepting data frames in compressed format.
Abstract:
Existing automatic log on/log off systems in telephone systems determine whether or not a workstation is occupied and whether calls can be directed to that workstation by determining whether or not the amplifier that provides the interface between the telephone system and the occupant's headset is plugged into the workstation. Recent headsets have included a connector in the cord between the headset and the amplifier, enabling the occupant to leave the workstation without unplugging the amplifier. However, this defeats the sensing mechanism of the existing automatic log on/log off system. A wireless telephone headset system according to the present invention replaces the existing wired amplifier and headset assembly and includes a detector for determining whether a wireless communication link exists between the headset and the amplifier, and an activator for activating the existing automatic log on/log off system in the telephone system. In the preferred embodiment, the detector senses both interruption and reestablishment of the wireless communication link between the headset and the amplifier, and the activator activates both current-sensing and voltage/resistance-sensing automatic log on/log off systems.
Abstract:
An audio communications interface 102 provides substantially automatic audio communications between a remote agent telephone 107 and a telephonic switch 104 through a telephonic console 106 connected to the telephonic switch 104. A remote agent dials a specified telephone number on the agent telephone 107 to access the audio communications interface 102. The audio communications interface 102 receives the dialed number from a telephone network 108 connected to the agent telephone 107 and generates a call signal which is transmitted to the telephonic switch 104 through the telephonic console 106. In response to the call signal, the telephonic switch 104 establishes audio communications with the agent telephone 107 through the console 106 and audio communications interface 102. A method for establishing audio communications through the audio communications interface 102 and data communications through an agent terminal 112, a computer network and a host computer 110 with the telephonic switch 104 is provided. In addition, a communications system for providing audio and data communications between a remote agent and the telephonic switch 104 is provided.
Abstract:
A voice level controller comprises a discriminator for determining whether speech is transmitted through an analog line or not, and a control circuit for controlling a speech level in accordance with the output of the discriminator.
Abstract:
A telecommunications network service overcomes the annoying effects of transmitted noise by a signal processing which filters out the noise using interactive estimations of a linear predictive coating speech model. The speech model filter uses an accurate updated estimate of the current noise power spectral density, based upon incoming signal frame samples which are determined by a voice activity detector to be noise-only frames. A novel method of calculating the incoming signal using the linear predictive coating model provides for making intraframe iterations of the present frame based upon a selected number of recent past frames and up to two future frames. The processing is effective notwithstanding that the noise signal is not ascertainable from its source.
Abstract:
A telecommunications network (120) is provided with adaptive gain control (AGC) of the voice signals in network (120). Network (120)includes an input (12) for receiving a voice signal, an output (14) for receiving the voice signal, and a coupling between input (12) and output (14) including at least one switch (124) or (126). Network (120) also includes voice enhancer (10) including power averager (18) for measuring and determining the average power of an input signal. Voice enhancer (10) also includes bass band equalizer (16) to attenuate a predetermined portion of the input signal to provide an equalized input signal. From the average power of the input signal is determined a scaling factor from a gain/attenuation look-up table (28). Voice enhancer (10) also includes output scaler (30) coupled to output (14), output scaler (30) scales the equalized input signal with the scaling factor and provides the scaled signal to output (14). Voice enhancer (10) also includes bass to treble power comparator (20) for detecting tandem enhancement and voice-band data detector (22) which cause enhancer (10) to be disabled appropriately.
Abstract:
An automated method for modifying a speech signal in a telephone network by applying a gain factor which is a function of the level of background noise at a given destination, and transmitting the modified speech signal to the destination. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands. The individual modified subband signals may then be combined to form the resultant modified speech signal.
Abstract:
A processor controlled test set is disclosed for testing special service circuits of a telecommunication system. A microprocessor controls the overall operation of the test set, while a digital signal processor provides high speed timing signals to the various test circuits for generating the wave forms used in testing, as well as analyzes the test result signals that are converted into digital signals. A calibration of the test generator signals as well as the signal measuring path is carried out prior to the test sequence. The digital signal processor also provides gain control over a talking path to maintain stability thereof. An I/O circuit of the test set provides plural communication paths between remote equipment and the test set to initiate and carry out various tests. Processors in the I/O module are effective to convert the various protocols of the serial data, by way of software, to digital bit streams usable by the test set.