Abstract: In one embodiment, a method for providing voice quality assurance is provided. The method determines voice information for an end point in a voice communication system. The voice information may be from an ingress microphone. The method determines if the voice quality is considered degraded based on an analysis of the voice information. For example, the voice information may indicate that it is distorted, too loud, too soft, is subject to an external noise, etc. Feedback information is determined if the voice quality is considered degraded where the feedback information designed to improve voice quality at an ingress point for a user speaking. The feedback information is then outputted at the end point to the user using the end point.

Abstract: A telecommunications system includes a network; a plurality of user devices operably coupled to said network and including a microphone and a speaker; a collaboration server configured to cause a playing of one or more test tones at said plurality of user devices and compensate for feedback if said test tones are detected by microphones associated with said plurality of user devices.

Abstract: Audio processing systems, such as full duplex digital telephone systems, are provided in which analog automatic level control (ALC) circuitry is employed to automatically control and stabilize the microphone output audio signal levels to a constant level for optimal digital signal processing. For example, an audio communication device (10) includes a microphone (11) to generate an audio signal, an analog automatic level control (ALC) circuit (13), coupled to an output of the microphone (11), to control a level of the audio signal output from the microphone (11) and output a level-controlled audio signal, and a DSP (digital signal processing) circuit (14) to process the level-controlled audio signal output from the ALC circuit (13). The ALC (13) allows audio signals to be captured by the microphone (11) with wide dynamic range, while stabilizing the microphone output level within a reduced dynamic range compatible with the limited dynamic range of the DSP circuitry (14) for enhanced performance.

Abstract: An automatic handset lifter is used with a wireless communication system. The system attaches to a conventional telephone and creates a communication link between the telephone and a portable device such as a wireless headset. The system automatically detects the ringing of the telephone by sensing magnetic fields of the telephone speaker. Upon ring detection, the system creates a communications link with the remote device and deploys a mechanical lifting device to lift the handset. Lifting the handset activates the telephone cradle switch to answer the phone and thereby allows the remote device to communicate over the telephone.

Abstract: A method (200), and electronic communication device (100), for automatically selectively changing from one mode of operation to a speakerphone mode of operation after a call on the electronic communication device (100) is established. The method (200), and electronic communication device (100), perform establishing a call (210) using a first mode of operation that is a mode of operation other than the speakerphone mode of operation. Then there is performed changing automatically (235) from the first mode of operation to the speakerphone mode of operation when a signal amplitude provided by a microphone used in the first mode of operation does not reach a threshold level before an expiration of a predetermined time period.

Abstract: A method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones mi and (p) speakers hpj for the control of said sound-reproducing systems characterized in that: A: for each speaker hpj, at least one sound signal S is sent on the speaker hpj, for each microphone mi, a piece of information hpjmi is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hpj and the microphone mi, B: a reference matrix Qr is saved, this reference matrix being constituted by all the pieces of reference information hpjmi obtained following the sending of the sound signal S, C: as soon as a comparison is to be made, the step A is run with a sound signal S? to obtain current information on a matrix Q, D: the matrices Q and Qr are compared.

Abstract: In an on-board hands-free communication apparatus, a gain of an amplifier of a speaker for outputting an incoming voice signal from a telephone device varies with a volume set value of a speaker volume switch adjusted by an user. When the gain of the amplifier increases equal to or greater than a threshold value, a feature extraction amplification circuit performs a feature extraction amplification processing to apparently amplify output power of the incoming voice signal, and a control circuit fixes the gain of the amplifier to the threshold value regardless of the volume set value.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: A radio communication system having a plurality of mobile stations, at least one base station, and a control side apparatus connected to the base station. Each mobile station has a mode switching instruction reception unit for receiving an instruction for switching from a radio communication mode for performing a radio communication operation to a loudspeaker mode or a hot mic mode, a destination identifying information storage unit for storing information for identifying a communication apparatus which serves as a communication partner when the mode switching instruction reception unit receives the instruction for switching to the hot mic mode, and a destination identifying information transmission unit responsive to the received instruction for switching to the hot mic mode by the mode switching instruction reception unit for transmitting the information stored in the destination identifying information storage unit over the air.

Abstract: The invention relates to processing signal for improving a sound pick-up in telephone and/or videophone units. The inventive method consists in comparing (13) a signal transmitted by a unit microphone (Mic) with a threshold low value (THR_LOW) and in amplifying (14,15) said microphone-transmitted signal when it is less than the threshold low value. However, said steps depend on the test (10) carried out on a signal used for feeding the loud speaker (HP), in particular consisting in controlling said last signal (crete_HP) of a background noise order (BDFd) in such a way that the background noise is not amplified by an acoustical coupling between the microphone (Mic) and the loud speaker (HP).

Abstract: A telephone terminal having a function of telephone communication is provided with a communication recording unit that records the telephone communication as communication data, an interruption controller that controls the communication recording unit to interrupt and to restart recording the telephone communication, the communication recording unit indexing a position, in the communication data, corresponding to the interruption of recording, and a replaying unit that replays the telephone communication as recorded based on the communication data, the replaying unit notifying that the recording was interrupted at the indexed position.

Abstract: A telephone headset system having a network interface device, a head set including a headphone and a microphone, and a foot operated, hands free call answering and call ending element. The network interface device, headset, and call answering and call ending element are electronically in communication with each other.

Abstract: A small-footprint hands-free speakerphone apparatus includes a loudspeaker, at least one main microphone, at least one reference microphone, and openings for receiving sound. The loudspeaker is disposed within a speaker chamber. Each microphone is mounted in a microphone boot formed by an acoustic opaque resilient material. Each main microphone may be omni-directional or directional. Each reference microphone is omni-directional. An opening is formed in front of each microphone for receiving sound for the microphone. An opening is formed in back of each directional main microphone for receiving sound and forming directivity for the microphone. An opening is formed in the microphone boot for each reference microphone to receive sound from the loudspeaker. An opening may also be formed in the loudspeaker chamber for passing sound to the reference microphone. The apparatus may include one or more interface units to provide interface to external devices.

Abstract: A method (200), and electronic communication device (100), for automatically selectively changing from one mode of operation to a speakerphone mode of operation after a call on the electronic communication device (100) is established. The method (200), and electronic communication device (100), perform establishing a call (210) using a first mode of operation that is a mode of operation other than the speakerphone mode of operation. Then there is performed changing automatically (235) from the first mode of operation to the speakerphone mode of operation when a signal amplitude provided by a microphone used in the first mode of operation does not reach a threshold level before an expiration of a predetermined time period.

Abstract: An improved telephone system and method of managing telephone calls. In one embodiment, a telephone system includes a speakerphone and a handset. A plurality of telephone lines or channels is available for conducting each of a corresponding plurality of telephone calls. In one instance, a first call may be routed to the handset, while a second call is routed to the speakerphone. The first and second telephone calls may be conducted simultaneously. An optional third telephone call may be conducted on a third line routed to a headset, such as to a Bluetooth device, so that all three telephone calls may be conducted simultaneously.

Abstract: The present invention provides an indication to the user of a voice communication system that an echo at the user's location has been detected by the system. In response to receiving this indication of echo, users may take any of several actions to reduce the echo, such as changing the position of the microphone or changing their own position with respect to the microphone. The need for potentially expensive echo-canceling hardware and software is reduced because the actions to eliminate the echo are performed by the user. The echo detector in the present invention may be implemented as part of a voice terminal, or as part of a communication switch or server that is used in conjunction with a voice terminal.

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.

Abstract: A speakerphone incorporates a portion defining a loudspeaker cavity, the cavity containing a loudspeaker. The speakerphone also includes a handset with an internal microphone in the handset. The speakerphone has a housing adapted to receive the handset, and when it is thus received, the handset microphone is acoustically coupled to the loudspeaker cavity, allowing the handset microphone signal to be used to control feed-back or feed-forward algorithms that enhance the performance of the speakerphone.

Abstract: A transceiver front-end provides an interface between a transmission medium and transmitter, and between a transmission medium and receiver. The transceiver front-end includes a hybrid circuit, a high-pass filter, and a gain stage, that permits the reduction or the complete elimination of buffer amplifiers. Buffer amplifiers can be eliminated because the hybrid circuit and/or the high-pass filter are adapted so that they can be directly connected to each other, without a loss in circuit performance. Furthermore, the high-pass filter and/or the gain stage are also adapted so they can be directly connected. As such, the transceiver front-end can be constructed using all passive components, reducing or eliminating excess heat generation.

Abstract: A communications terminal comprising means operable, in the absence of significant input to the terminal, to automatically terminate a call.

Abstract: This invention includes a circuit for the prevention of acoustic feedback between an electronic device and an audio accessory. In a preferred embodiment, the circuit prevents audio feedback between a cellular telephone and a speakerphone accessory. The circuit includes a current limiting device coupled serially in the receive (Rx) line. The current limiting device is actuated via a delay circuit coupled between the current limiting device and the transmit (Tx) line. When a bias current is presented to the Tx line, the bias propagates through the delay circuit, thereby actuating the current limiting device a predetermined time after the presentation of the bias. The circuit keeps the Rx line open long enough for the phone to deactivate its internal microphone.

Abstract: Techniques for canceling echo and suppressing noise using an array microphone and signal processing. In one system, at least two microphones form an array microphone and provide at least two microphone input signals. Each input signal may be processed by an echo canceller unit to provide a corresponding intermediate signal having some echo removed. An echo cancellation control unit receives the intermediate signals and derives a first gain used for echo cancellation. A noise suppression control unit provides at least one control signal used for noise suppression based on background noise detected in the intermediate signals. An echo cancellation and noise suppression unit derives a second gain based on the control signal(s), cancels echo in a designated intermediate signal based on the first gain, and suppresses noise in this intermediate signal based on the second gain. The signal processing may be performed in the frequency domain.

Abstract: An apparatus and method are presented for dynamically disabling a first audio input/output (“I/O”) connector when an audio I/O device is coupled to a second audio I/O connector, locating the disablement point to reduce significantly any spurious noise coupled onto the primary audio I/O connector and its associated electrical components and cabling. An advantage is that it can be implemented with components that provide low resistance from the first audio I/O connector to ground, thoroughly grounding, and significantly reducing spurious noise coupled onto, the first audio I/O connector and associated electronics and cabling. This invention therefore also significantly reduces the spurious noise processed with, and therefore interfering with, the signal associated with a device coupled to a second audio I/O connector, significantly increasing the quality of the input or output signal from that secondary device.

Abstract: A pocket speakerphone adapter for a landline-based common telephone to provide the common telephone with speakerphone capabilities. The conventional handset of a conventional common telephone is disconnected from the handset jack of its base unit, and the pocket speakerphone is temporarily connected in its place to the handset jack of the base unit using a conventional handset cord. The conventional common telephone base unit provides interconnection with the public switched telephone network (PSTN), while the pocket speakerphone in accordance with the principles of the present invention provides a temporary adaptation of the land line common telephone to operate as a speakerphone.

Abstract: The present invention is directed to the provision of a personalized speaker phone and hands-free telephony. In particular, the present invention allows communications to be output along a narrowly defined path, rather than being broadcast. In this way, a private voice communication signal can be provided to a user, even though the user is not holding the output device to the user's ear. Furthermore, by providing audible signals along narrowly defined paths, different audible signals may be provided to users at the same location, without interfering with one another.

Abstract: To achieve an improved convergence behaviour for, e.g., echo cancellation there is provided a digital adaptive filter which includes a filter coefficient update means to successively update filter coefficients in accordance with an input signal, an estimated power of the input signal, and an error signal between the input signal filtered in the digital adaptive filter and the input signal propagated along an external path being modelled by the digital adaptive filter. Here, an input signal power estimation means is adapted to perform recursive smoothing for an increasing input power and a decreasing input power in an asymmetric fashion with different smoothing factors. In case the estimation is carried out in the frequency domain the step size for the update of filter coefficients may be calculated for each frequency band individually.

Abstract: In a speakerphone system, the signal that is directed to a loudspeaker is filtered by a dynamically controllable highpass filter. The filter characteristics are adjusted on the basis of the power of the signal and the degree of recent microphone activity. The filtered signal is a far-talker signal, while the microphone activity is represented as a near-talker signal that includes an echo component as a consequence of loudspeaker-to-microphone coupling. The degree of microphone activity is determined by a subband analyzer. A noise floor estimator may be used to determine the degree to which the near-talker signal is comprised of background noise, so that the noise can be eliminated in the process of controlling the highpass filter.

Abstract: Voice activity is detected by comparing a signal with two thresholds and producing data representing the energy of the signal. The data, in binary form, is compared with thresholds to determine voice activity. In accordance with another aspect of the invention, the thresholds are adjusted based upon statistical information. In accordance with another aspect of the invention, the data can be weighted to provide an indication of the quasi-RMS energy of an input signal. In accordance with another aspect of the invention, voice activity detectors, individually weighted, are provided at each input and each output of a telephone for reliably controlling echo cancelling circuitry within the telephone.

Abstract: A speaker-phone that is operable to perform system change detection, double-talk detection and if desired, optimal step-size control. The invention differentiates between system change and double-talk and is operable to these two situations differently. The invention does not rely solely on the energy measurements that are employed by conventional speaker-phones. Highly reliable system change detection is offered when compared to conventional speaker-phones. The direction and the amplitude of the system change is measured with very good detection probability in the invention. For double-talk detection, the invention does not rely on assumptions that the far-end echoes are of lower amplitude that the near-end signal in performing double-talk detection. The invention performs double-talk detection within a large diversity of echo-paths and background noise levels. A designer may tune the speaker-phone system to ensure proper double-talk detection near low levels of the near-end signal.

Abstract: An echo canceler provided with a noise detecting section which holds signals showing a presupposed ambient noise such as a horn of a vehicle. The noise detecting section is adapted to monitor a sending signal received through a microphone and, while detecting such ambient noises in the sending signal, an adaptive filter outputs a pseudo echo existing immediately before the ambient noise is detected. Even during the generation of ambient noises, an echo noise can be eliminated and the generation of a foreign voice (noise) can be avoided as well.

Abstract: A communication apparatus having a one-way speakerphone function including a handset connected to the communication apparatus by a transmit-receive line, wherein the transmit-receive line is enabled when the handset of the communication apparatus is off-hook; a speaker outputting a sound of the handset communication; an input key part including a selection key for the one-way speakerphone function and a plurality of dial keys; and a central control device controlling the one-way speakerphone function; wherein a conversation signal is transmitted through the speaker when the handset is off-hook, the one-way speakerphone function selection key signal is input, and it is determined that a dialing is completed.

Abstract: An echo canceller is provided in which a down-sampling circuit converts a 16-kHz sampling frequency of a wide-band voice signal output to an 8-kHz sampling frequency of a narrow-band voice signal supplied at an input terminal, an adaptive filter estimates, from the wide-band voice signal whose sampling frequency has been down-sampled to 8 kHz in the down-sampling circuit, an echo signal coming from a speaker to a microphone and having an echo path characteristic imparted to the echo signal by an echo path filter, and a subtraction circuit subtracts from the microphone input signal the echo signal having been estimated by the adaptive filter.

Abstract: A method of adapting an audio unit to a computer terminal or digital signal processing unit, in particular a telecommunications device, with unknown system parameters, comprises the following steps:

Abstract: An apparatus and method for joint music production wherein a remotely separated plurality of musicians are joined by phone lines of a public telephone system. The public telephone system has a signal frequency cutoff limiting transmission to a low range which is incompatible with music. Each of musicians produces a local music signal (MSL) with full audio range. This signal is impressed on the public telephone system which creates a low-end outbound music signal (LEL). Also, separately, the MSL signal is filtered through a first high-pass filter and mixed with a mixer signal to produce a sum/difference signal (SDL). The SDL signal is impressed onto the second phone line of the public telephone system as a high-end outbound music signal (HEL). A second mixer circuit receives at least one remotely produced sum/difference signal (SDR) and the mixer signal and this is again filtered to obtain the high end signal only (HER).

Abstract: A headset or hand-free operating mode selecting and switching system for telephone is provided. The system mainly includes a circuit for a main unit of the telephone into which a headset connection detecting system and a single hand-free/headset button switch are added. By depressing the one single hand-free/headset button switch, the headset connection detecting system is automatically enabled to detect whether a headset is connected to the circuit of the main unit of the telephone. The telephone is automatically set to either the headset or the hand-free operating mode depending on a detected result from the headset connection detecting system. And, depressing of the one single hand-free/headset button switch again automatically disables the previously enabled hand-free or headset mode.

Abstract: A method of displaying alternating transmitting and receiving phases of voice communication in a mobile phone with a display when it is operated in the speakerphone mode includes the steps of comparing the intensity of a received voice signal from a caller with that of a transmitted voice signal, outputting the received voice signal through the speaker of the mobile phone while displaying a visual indication representing the receiving phase on the display if the intensity of the received voice signal is stronger than that of the transmitted voice signal, and sending the transmitted voice signal to the caller while displaying a visual indication representing the transmitting phase on the display if the intensity of the received voice signal is weaker than that of the transmitted voice signal.

Abstract: A wireless user terminal operates in a mode that is selected based on an indication of the orientation of the wireless user terminal from a gravitational sensor in the wireless user terminal. When the wireless user terminal is in a first orientation, the wireless user terminal operates in a speakerphone mode. When the wireless user terminal is in a second orientation, the wireless user terminal operates in a handset mode. The mode of operation of the wireless user terminal transitions based on movement of the wireless user terminal from the first orientation to the second orientation. For example, when the wireless user terminal placed in a horizontal orientation, one of the speakerphone modes is selected. When the user picks up the wireless user terminal, the wireless user terminal transitions to handset mode.

Abstract: A hand-free operator can be externally dialed via a dialing device. It is more particularly related to an external dialing device, which can be applied to all kinds of cellular phones. Thereby people do not have to keep telephone numbers in memory, they can make direct dials on a hand-free operator by means of telephone numbers stored in advance in a memory unit of a dialing device. It mainly houses a hand-free operator circuit and a dial processing unit in a hand-free operator embodiment that operate in collaboration with a signal cable and a dialing device (DTMF acoustic generator). It enables people to make a dial on cellular phones of all kinds via a hand-free operator without memorizing phone numbers or resorting to phone books, effectively facilitating the use of cellular phones and advancing driving safety.