Abstract: Disclosed are a walkie-talkie and a walkie-talkie information transmission method.

Abstract: An echo canceller is disclosed in which audio signals of the playback content received by one or more of the microphones from a loudspeaker of the device may be used as the playback reference signals to estimate the echo signals of the playback content received by a target microphone for echo cancellation. The echo canceller may estimate the transfer function between a reference microphone and the target microphone based on the playback reference signal of the reference microphone and the signal of the target microphone. To mitigate near-end speech cancellation at the target microphone, the echo canceller may compute a mask to distinguish between target microphone audio signals that are echo-signal dominant and near-end speech dominant. The echo canceller may use the mask to adaptively update the transfer function or to modify the playback reference signal used by the transfer function to estimate the echo signals of the playback content.

Abstract: A system for enhancing tonal quality for ringback signals in a telecommunications network by adjusting audio processing during an alerting interval of a telecommunications session (e.g. during a telephone call). The system may be embodied in a telecommunications switching device and include a ringback manager process that provides the ability to adjust audio processing (e.g., by signaling to in-path equipment) which enhances the tonal quality of the ringback signal. As an example, a telephone system (e.g. equipment associated with establishment and operation of telephone communications sessions) can disable or attenuate operation of voice quality enhancement processing such as echo cancellation, noise reduction, adaptive level control, and the like, during the ringback or alerting interval or period of a telephone call so that the ringback tone provided to the calling party is not distorted by such processing.

Abstract: A call directing system receives an incoming call from a caller. The caller is prompted to speak, thus enabling a prosody analyzer to generate an analysis of a prosody of the caller's voice. This analysis provides a basis for generating a caller profile that describes caller preferences of the caller. Based on the analysis of the prosody of the caller's voice and the generated caller profile, the call is directed to a particular call recipient.

Abstract: Systems and methods for managing the volume of multiple VoIP streams are disclosed. The system includes a VoIP server configured to receive an input audio stream from a first VoIP handset, create separate output audio streams from the input audio stream for transmission to second and third VoIP handsets, and to connect to a communications network. The system also includes a volume control table coupled to the VoIP server, the volume control table including records of volume adjustments made during prior conversations between the two or more VoIP handsets. The VoIP server is further configured create the separate output audio streams such that one or more of the output streams has a volume that is different than input audio stream based on the records.

Abstract: Systems and methods for reducing squealing or feedback on amplified telephones are disclosed. The telephone system generally includes a receiver in communication with a receive signal path, a transmitter in communication with a transmit signal path, a receiver gain detector configured to detect a receiver gain on the receive signal path, and a controller configured to operate the telephone system in full duplex when the receiver gain is approximately less than a receiver stability level and to operate the telephone system in adaptive duplex when the receiver gain is approximately above the receiver stability level. The adaptive duplex mode is such that an adaptive attenuation level alternately applied on the receive and transmit signal paths is dependent upon the level by which the receiver gain exceeds the receiver stability level.

Abstract: An audio host device has an electrical interface having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a cold input and a hot input. The hot input is coupled to the microphone contact. A switched attenuator circuit is also provided that has first and second states. In the first state, the attenuator circuit couples a sense point of the reference contact to the cold input, while in the second state the cold input is isolated from the reference sense point. Other embodiments are also described and claimed.

Abstract: An audio host device has a connector having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a first cold input, a second cold input, and hot input. The first cold input is coupled to an output of the speaker amplifier, while the second cold input is coupled a sense point for the reference contact. The hot input is coupled to the microphone contact. Other embodiments are also described and claimed.

Abstract: An audio host device has an electrical interface having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a cold input and a hot input. The hot input is coupled to the microphone contact. A variable attenuator circuit is also provided having an input coupled to receive a signal from a sense point for the reference contact, and an output coupled to the cold input of the difference amplifier. A controller has an output coupled to control the variable attenuator. Other embodiments are also described and claimed.

Abstract: An audio host device has a headset electrical interface in which a reference contact is shared by a microphone and a speaker. A variable or programmable gain amplifier, PGA, has a signal input coupled to a sense point for the reference contact. A first difference amplifier has its inputs coupled to a microphone contact and the PGA through a first gain stage; a second difference amplifier has its inputs coupled to the microphone contact and to the output of the PGA, through a second gain stage having a different gain than the first gain stage; a third difference amplifier has its inputs coupled to the microphone contact and the output of the PGA through a third gain stage having a gain in between that of the first and second gain stages. A controller monitors outputs of the first and second difference amplifiers and changes a gain setting of the PGA. Other embodiments are also described and claimed.

Abstract: A call directing system receives an incoming call from a caller. The caller is prompted to speak, thus enabling a prosody analyzer to generate an analysis of a prosody of the caller's voice. This analysis provides a basis for generating a caller profile that describes caller preferences of the caller. Based on the analysis of the prosody of the caller's voice and the generated caller profile, the call is directed to a particular call recipient.

Abstract: An audio host device has an electrical interface having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a cold input and a hot input. The hot input is coupled to the microphone contact. A switched attenuator circuit is also provided that has first and second states. In the first state, the attenuator circuit couples a sense point of the reference contact to the cold input, while in the second state the cold input is isolated from the reference sense point. Other embodiments are also described and claimed.

Abstract: A signal processing device includes: a speaker amplifier for amplifying a reception voice; a reception voice attenuator for attenuating the reception voice inputted to the speaker amplifier during a voice transmission mode; a reception voice detection circuit for detecting the reception voice outputted from the speaker amplifier; a microphone amplifier for amplifying a transmission voice inputted to a microphone; a transmission voice attenuator for attenuating the transmission voice outputted by the microphone amplifier during a voice reception mode; a transmission voice detection circuit for detecting the transmission voice outputted by the microphone amplifier; and a discriminator for discriminating the mode to be operated among the voice transmission mode and the voice reception mode on the basis of detection outputs from the reception voice detection circuit and the transmission voice detection circuit.

Abstract: Embodiments of the method and system provide for automatic balancing of audio signal strength levels on voice paths between first and second parties in a telecommunication system. The method in an embodiment may have the steps of: checking for a subscription to audio signal strength level balancing by at least one of the first and second parties; and balancing, if at least one of the first and second parties has the subscription, audio signal strength levels on voice paths between the first and second parties. The system implements the method.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: A half duplex device and signal processing method thereof. First, a signal detection unit of a signal detector of a half duplex device obtains a short-term power and long-term power. Next, a data update unit of the signal detector sets a short-term power threshold as a default threshold, determines whether the long-term power is greater than the default threshold, sets the short-term power threshold value equal to the long-term power value if the long-term power value is greater than the default threshold value, and provides smooth performance in a noisy environment.

Abstract: The amplitudes of a signal on the microphone input of a telephone and a signal on the line input of the telephone are compared with each other and with two thresholds. The comparisons are sampled and the samples control an up-down counter. The count in the counter is compared with two thresholds and the results of the comparisons are used to control at least two accumulators. A first accumulator counts consecutive comparisons having the same result and a second accumulator counts up or down in accordance with the comparisons. The counts in the first and second accumulators are logically analyzed to determine whether to receive or transmit. A plurality of samples defines a window, at the end of which the up-down counter is reset. The accumulators store data from a plurality of windows.

Abstract: A system and method for adjusting the volume level of a communications device in response to a voice volume/ambient noise relationship. Ambient noise and voice volume are sampled and compared to a predetermined ambient noise to voice volume relationship. Voice volume is adjusted up or down in response to a control signal that is generated in view of the comparison between the sampled voice volume to ambient noise relationship and the predetermined ambient noise to voice volume relationship.

Abstract: A plurality of Goertzel filters whose operating frequencies are distributed across the voice baseband are used to detect voice and control tones in a signal. Filters operating at frequencies of control tones and detecting that most of the signal energy occurs at those frequencies indicates presence of the control tones. At least three of the filters detecting that about 10% to 20% of the signal energy occurs at each of their operating frequencies indicate presence of voice. The total energy detected in the signal being below a noise threshold indicates presence of noise or silence.

Abstract: A phone reception system that provides differential volume at different frequencies is made available on demand by application of a multiband dynamic range compression filter to enhance the normally audible spectrum of a signal. Audio components that lie above a person's sensitive hearing regions are amplified so that they can hear and understand them. The modifications are available on demand through a message from any remote customer premises equipment or business phone to the softswitch.

Abstract: A digitized input signal is sampled, to obtain a reference input signal. The reference input signal is spuriously whitened, to obtain a pseudo whitened signal. An echo replica signal is generated on the basis of the pseudo whitened signal and an adaptive filter coefficient. A cancellation error signal is obtained on the basis of an echo signal and the echo replica signal. The adaptive filter coefficient is updated on the basis of the cancellation error signal, the adaptive filter coefficient, and the pseudo whitened signal.

Abstract: A full duplex speaker-phone that performs adaptive filtering to provide increased loop stability between users of the speaker-phone. The invention employs a sliding filter that performs adaptive frequency dependent attenuation of the speech signals within a speaker-phone. By detecting the mode of operation of the speaker-phone in real time, the speaker-phone performs adaptive filtering to ensure a high perceptual quality for the users of the speaker-phone. The sliding filtering is performed using a sliding low pass filter (LPF) in certain embodiments of the invention. Perceptually, the speaker-phone does not surrender the appearance of a full duplex speaker-phone from the perspective of the users. In addition, by performing frequency dependent filtering of the speech signals, the speaker-phone offers improved stability in the speaker-phone loop and reduces residual echoes significantly over conventional technology employed in speaker-phones.

Abstract: In a speakerphone device identification of signals (i.e., voice input or speaker output) in a process for reducing acoustic feedback, in a communication device, is accomplished by adding a signature noise (i.e., an identification mark) to output signals radiated by the speaker to enable these signals to be separated from speech input to the microphone. Having identified the signal (i.e., speech output) likely to cause a “singing” phenomenon, appropriate insertion loss to reduce the feedback may be added to the appropriate speech path within the communication device, to reduce a probability of singing.

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: The present invention relates to an apparatus and a method for volume control in digital telephones such as second generation mobile telephone terminals, third generation IMT 2000 image mobile telephone terminals, and ISDN (integrated service digital network) terminals. In particular, an apparatus and a method in accordance with an embodiment of the present invention measure background noise and adjust volume of received sound and/or ring signal accordingly. An embodiment of the present invention measures background noise before ringing ring signal and adjusts volume of ring signal. While communicating, the apparatus automatically adjusts volume of received sound in response to the measured background noise. A telephone terminal in accordance with an embodiment of the present invention includes voice/non-voice determination part, volume measurement part, and gain control part. The voice/non-voice determination part separates pure background noise from composite signal (voice+background noise).

Abstract: A voice-switching device for use in a telephone terminal which applies an optimal amount of switched loss to a plurality of distinct receive transducers. The voice-switching device detects speech signals on the transmit and receive paths to determine a mode of operation. Based on the mode of operation the device applies optimal amount of loss to the transmit path and each output path separately. This system eliminates excessive attenuation of non-dominant receive transducers as well as optimizes the transition time from active to quiescent mode for all receive transducers.

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: Headset for connection to a telephone apparatus, said headset comprising a capsule with a built-in receiver, and on which there is mounted a boom with a microphone. The headset is connected to a telephone apparatus by means of a wire with associated jack connection, and the headset has associated amplification and adjustment circuits for both the microphone and receiver, manual operating elements for the setting of said amplification and adjustment circuits, and switch elements for changeover depending partly on the type of microphone in the telephone apparatus and partly on the polarity of the telephone apparatus' microphone and receiver wires. The amplification and adjustment circuits are built into the headset capsule, so that the use of a separate connection box is avoided, and the manual operation elements are placed on the capsule. The switching elements consist of a multi-position switch, which is placed in the capsule in such a manner that it is accessible for operation.

Abstract: The conventional speakerphone using the voice switch, when combined with the echo canceler, involves a problem that the threshold for switching the transmit/receive state of the voice switch cannot comply smoothly and stably with the performance variation of the echo canceler, thereby obstructing echo canceling in a manner that the voice switch cooperates with the echo canceler. The speakerphone of the invention estimates the performance variation of the adaptive filter by using the integrated value of a power of the receive signal or the transmit signal referred in the past when the adaptive filter learns, and varies the threshold in accordance with the performance variation. Thus, the speakerphone of the invention achieves a stable communication system that approaches to the fill duplex with the voice switching system and the echo canceling system combined.

Abstract: There are provided a voice switching system and a voice switching method which are capable of carrying out highly accurate voice detection and performing a switching action properly even in the presence of an unstationary noise or under circumstances where the S/N ratio is low.