Abstract: There is provided a method for use by an echo canceller to cancel a far echo at a variable delay and a near echo at a fixed delay. The method comprises constructing an echo signal model based on an incoming signal, determining a variable echo delay for a far echo caused by a far echo source, determining a fixed echo delay for a near echo caused by a near echo source, subtracting the echo signal model from an outgoing signal at a window placed around the variable echo delay to cancel far echo, e.g. when the echo canceller determines existence of the far echo, and subtracting the echo signal model from the outgoing signal at a window placed around the fixed echo delay to cancel near echo, e.g. regardless of existence of the near echo, wherein the fixed echo delay is smaller than the variable echo delay.

Abstract: There is provided a method of detecting and reporting poor voice quality for use by a gateway device. The method comprises facilitating a connection between a telephone and a remote telephone via a network, and detecting a poor voice quality indictor during the connection. The method further comprises capturing, for a pre-determined period of time, telephone voice data being exchanged between the gateway and the telephone, network voice data being exchanged between the gateway and the network, and gateway parameters. The method also comprises packetizing the telephone voice data, the network voice data and the gateway parameters into a plurality packets having a network address of a network storage, and transmitting the plurality packets destined for the network storage via the network. In one aspect, the poor voice quality indictor may be generated by a user of the telephone in response to a poor voice quality of the connection.

Abstract: A method of adjusting an echo canceller comprises obtaining a first cross-correlation between a far-end signal and an error signal, wherein the error signal is generated by subtracting an output signal of an adaptive filter from a local-end signal; determining whether the first cross-correlation is above a pre-determined threshold; relocating the adaptive filter by a few samples if the determining determines that the first cross-correlation is above a pre-determined threshold; calculating a first improvement indicator parameter, wherein the first improvement indicator parameter is calculated after the relocating the adaptive filter by the few samples; determining whether the first improvement indicator parameter indicates a performance improvement by the adaptive filter after the relocating the adaptive filter by the few samples; calculating a gain based on the local-end signal and the error signal if the determining does not determine the performance improvement; and multiplying the adaptive filter by the ga

Abstract: The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.

Abstract: There is provided a method of detecting and reporting poor voice quality for use by a gateway device. The method comprises facilitating a connection between a telephone and a remote telephone via a network, and detecting a poor voice quality indictor during the connection. The method further comprises capturing, for a pre-determined period of time, telephone voice data being exchanged between the gateway and the telephone, network voice data being exchanged between the gateway and the network, and gateway parameters. The method also comprises packetizing the telephone voice data, the network voice data and the gateway parameters into a plurality packets having a network address of a network storage, and transmitting the plurality packets destined for the network storage via the network. In one aspect, the poor voice quality indictor may be generated by a user of the telephone in response to a poor voice quality of the connection.

Abstract: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis.

Abstract: There is provided system and method for receiving speech signals via a plurality of speech lines and communicating the speech signals over a phone line. For example, the communication system comprises a modem, a multiplexor in data communication with the modem, and a plurality of speech codecs in data communication with the multiplexor. According to this aspect, each of the plurality of speech codecs receives a portion of the speech signals via one of the plurality of speech lines and encodes that portion of the speech signals to generate encoded speech signals, and wherein the encoded speech signals from each of the plurality of speech codecs are multiplexed by the multiplexor to generate multiplexed encoded speech signals, and wherein the modem transmits the multiplexed encoded speech signals over the phone line.

Abstract: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis.

Abstract: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis.

Abstract: There is provided a method of detecting and reporting poor voice quality for use by a gateway device. The method comprises facilitating a connection between a telephone and a remote telephone via a network, and detecting a poor voice quality indictor during the connection. The method further comprises capturing, for a pre-determined period of time, telephone voice data being exchanged between the gateway and the telephone, network voice data being exchanged between the gateway and the network, and gateway parameters. The method also comprises packetizing the telephone voice data, the network voice data and the gateway parameters into a plurality packets having a network address of a network storage, and transmitting the plurality packets destined for the network storage via the network. In one aspect, the poor voice quality indictor may be generated by a user of the telephone in response to a poor voice quality of the connection.

Abstract: There is provided a method of decoding speech data generated from a speech signal. The method comprises receiving the speech data having at least one main pulse in a subframe of the speech data; generating a first predicted pulse, based on the at least one main pulse, on one side of the main pulse in the subframe of the speech data, wherein the first predicted pulse has a lower gain than the main pulse; generating a second predicted pulse, as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data; reconstructing the speech signal using the at least one main pulse, the first predicted pulse and the second predicted pulse.

Abstract: A method comprises analyzing each frame of a plurality of frames of the speech signal to determine one or more speech parameters for the speech signal; deciding, for each frame of the plurality of frames of the speech signal, based on the one or more speech parameters of the speech signal, to select one of a plurality of encoding modes including a first encoding mode and a second encoding mode for encoding each frame of the plurality of frames of the speech signal; encoding each frame of the plurality of frames of the speech signal according to the selected one of the plurality of encoding modes for each frame of the plurality of frames in the deciding; the first encoding mode supports a first encoding rate and the second encoding mode supports a second encoding rate, wherein the first encoding rate is the same encoding rate as the encoding rate.

Abstract: There is provided a method of encoding an input speech signal. The method comprises identifying a fixed codebook vector from a fixed codebook; identifying an adaptive codebook vector from a adaptive codebook; calculating an adaptive codebook gain; reducing the adaptive codebook gain by an amount; optimally selecting a fixed codebook gain based on the adaptive codebook gain while both the fixed codebook vector and the adaptive codebook vector remain fixed; and converting the input speech signal into an encoded speech using the fixed codebook gain, the adaptive codebook gain, the fixed codebook vector and the adaptive codebook vector. The amount of reducing the adaptive codebook gain may be varied.

Abstract: There is provided a method of using an adaptive tilt compensation by a speech decoder. The method comprises receiving a bit stream including a plurality of parameters representative of a speech signal; identifying an adaptive code vector and a fixed code vector using the plurality of parameters; scaling the adaptive code vector and the fixed code vector to generate a scaled adaptive code vector and a scaled fixed code vector; summing the scaled adaptive code vector and the scaled fixed code vector to generate a synthesized output; calculating a first reflection coefficient based on the plurality of parameters representative of the speech signal; multiplying the first reflection coefficient by a factor to generate a tilt factor; and applying the tilt factor to the synthesized output based on an encoding bit rate.

Abstract: There is provided a method of using a processing circuitry for selecting a preferential pitch lag value from a plurality of pitch lag values, including a first pitch lag value and a second pitch lag value, for coding an input speech signal. The method comprises determining a first timing relationship between a previous pitch lag value and at least one of the plurality of pitch lag values; determining a second timing relationship between the first pitch lag value and the second pitch lag value; favoring one of the first pitch lag value and the second pitch lag value based on the first timing relationship and the second timing relationship to select one of the first pitch lag value and the second pitch lag value as the preferential pitch lag value; and converting the input speech signal into an encoded speech using the preferential pitch lag value.

Abstract: There is provided a method of selecting a pitch lag value for a portion of a speech signal, the method comprising: computing a weighted correlation function of the portion of the speech signal for a range of delay times, wherein the weighting of the correlation function depends on both the delay time and a characteristic of one or more previous portions of the speech signal; and selecting the pitch lag value based on a delay time from the range of delay times that maximizes the weighted correlation function.

Abstract: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis.

Abstract: There is provided a method of selecting a pitch lag value from a plurality of pitch lag candidates for coding a speech signal. The method comprises identifying the plurality of pitch lag candidates from a frame of the speech signal using correlation; classifying the speech signal to obtain a voice classification; determining whether one or more of the plurality of pitch lag candidates are in a temporal neighborhood of one or more previous pitch lag values; favoring the one or more of the plurality of pitch lag candidates determined to be in the temporal neighborhood of the one or more previous pitch lag values, by adaptive weighting, over other ones of the plurality of pitch lag candidates; and selecting the pitch lag value based on the voice classification and the one or more of the plurality of pitch lag candidates favored by the adaptive weighting.

Abstract: The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.

Abstract: The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.