Abstract: Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

Abstract: Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

Abstract: Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

Abstract: Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

Abstract: Systems, devices, and methods are described for reducing degradation of a voice recognition input. An always listening device may always be listening for voice commands via a microphone and may experience interference from unwanted audio such as from the output audio of television speakers. The always listening device may receive data associated with the output audio over a first communications channel. The always listening device may also receive, on a second communications channel, timing information associated with data. The always listening device may adjust admission of the audio received by the microphone to enable it to arrive at approximately the same time as the data received via the first communications channel. The unwanted output audio included in the audio received via the microphone may then be determined and may be removed so that a voice command in the audio received by the microphone may be processed.

Abstract: A method that includes: (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.

Abstract: A method that includes; (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.

Abstract: In one implementation, a conference hosted on a conference bridge that includes an echo control device that classifies the echo return loss of a plurality of endpoints. The endpoints may include endpoints on internet protocol (IP) networks and public switched telephone networks (PSTN). The echo control of the endpoints is controlled independently. An echo classification module calculates the echo return loss value of the associated echo tail of the endpoint and chooses an appropriate echo control mode. If the echo return loss is above a high threshold, a pass through mode is activated. If the echo return loss is below a low threshold, a full echo cancellation mode is activated. If the echo return loss is between the low threshold and the high threshold, an echo suppression mode is activated.

Abstract: A method that includes: (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.

Abstract: In an example embodiment, a control connection is introduced between an adaptive jitter buffer (AJB) and an adaptive bulk delay (ABD) buffer of an echo canceller (ECAN) in an Internet Protocol (IP) conference bridge. The control connection allows the AJB to control the amount of delay inserted by the ABD in the ECAN convolution processor (CP) signal path. The adjustment in ABD delay restores the time alignment of the ECAN internal echo estimate and offsets variations in echo tail delay introduced by network induced AJB delay changes. Time-invariance is preserved in the echo tail path.

Abstract: A method that includes: (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.

Abstract: In an example embodiment, an example method is provided for echo mitigation in a conference call. In this method, a test audio signal is transmitted to a conference endpoint and as a result, an echo associated with the transmittal of the test audio signal is received. One or more parameters of the echo are then identified and an echo mitigation process is selected from multiple echo mitigation processes based on the identified parameters. The selected echo mitigation process is then applied.

Abstract: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.

Abstract: In one implementation, a conference hosted on a conference bridge that includes an echo control device that classifies the echo return loss of a plurality of endpoints. The endpoints may include endpoints on internet protocol (IP) networks and public switched telephone networks (PSTN). The echo control of the endpoints is controlled independently. An echo classification module calculates the echo return loss value of the associated echo tail of the endpoint and chooses an appropriate echo control mode. If the echo return loss is above a high threshold, a pass through mode is activated. If the echo return loss is below a low threshold, a full echo cancellation mode is activated. If the echo return loss is between the low threshold and the high threshold, an echo suppression mode is activated.

Abstract: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.