Abstract: A method of masking a residual echo signal by an echo canceller is provided. The method comprises receiving a far-end signal, adjusting filter coefficients of an adaptive filter in response to the far-end signal, generating an echo model signal based on the far-end signal using the adaptive filter, receiving a near-end signal, subtracting the echo model signal from the near-end signal to generate an output signal, defining a spectral mask based on the near-end signal, wherein the spectral mask is indicative of near-end spectral peaks and near-end spectral valleys, de-emphasizing the output signal in spectral regions of the near-end spectral peaks, and emphasizing the output signal in spectral regions of the near-end spectral valleys, wherein the de-emphasizing occurs during filter coefficients determination for the adaptive filter. A weighted filter may perform the de-emphasizing and the emphasizing operations, where the weighted filter uses medium term spectral characteristics of the near-end signal.

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: A method of using music detection to enhance an operation of an echo canceller is provided, wherein the echo canceller includes an adaptive filter and a nonlinear processor. The method comprises receiving an input signal including an echo signal by the echo canceller from a near end device, filtering the input signal using the adaptive filter to eliminate linear components of the echo signal in the input signal and generate an error signal, analyzing the error signal using a music detector to determine existence of a music signal in the error signal, bypassing the nonlinear processor if the analyzing determines the music signal exists in the error signal, and eliminating nonlinear components of the echo signal from the error signal using the nonlinear processor if the analyzing determines the music signal does not exist in the error signal.

Abstract: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1?k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

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: There is provided a method of transcoding an Enhance Full Rate (EFR) 12.2 Kbps encoded frame into an Adaptive Multi-Rate (AMR) 12.2 Kbps encoded frame, where the method comprises receiving the EFR 12.2 Kbps encoded frame from a first codec; determining if the EFR 12.2 Kbps encoded frame is a Silence Insertion Descriptor (SID) frame; if the EFR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the EFR SID frame. There is also provided a method of transcoding an EFR 12.2 Kbps encoded frame into an AMR 12.2 Kbps encoded frame, where the method comprises receiving the AMR 12.2 Kbps encoded frame from a first codec; determining if the AMR 12.2 Kbps encoded frame is an SID frame; if the AMR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the AMR SID frame.

Abstract: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1?k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

Abstract: There is provided a communication method for the receiving a first call-initiation request, generating a second call-initiation request in response to the first call-initiation request and generating a third call-initiation request in response to the first call-initiation request. Moreover, there is provided a communication method comprising receiving call-session information, separating the call-session information into core audio information and into supplementary information, routing core audio information on a first path and routing supplemental information on a second path. Further, there is provided a communication method, comprising receiving core audio information on a first communication path, receiving audio-enhancement information on a second communication path, uncompressing audio-enhancement information, combining core audio information with audio-enhancement information to generate combined audio information and providing combined audio information to audio terminals.

Abstract: There is provided a method for use by a speech encoder to encode an input speech signal.

Abstract: There is provided a method of updating a noise state of a voice activity detector (VAD) for indicating an active voice mode and an inactive voice mode. The method comprises receiving an input signal having a plurality of frames, determining an elapsed time since the last update of the noise state, updating the noise state of the VAD if the elapsed time exceeds a predetermined time, determining an average minimum energy based on two or more of the plurality of frames, determining a current minimum energy based on a current frame of the plurality of frames, updating the noise state of the VAD if the average minimum energy is less than the current minimum energy, and updating the noise state of the VAD if the average minimum energy is greater than the current minimum energy plus a first predetermined value.

Abstract: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1-k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

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 voice activity detection method for indicating an active voice mode and an inactive voice mode. The method comprises receiving an input signal having a plurality of frames, determining whether each of the plurality of frames includes an active voice signal or an inactive voice signal, determining a second reflection coefficient for each frame determined to include the inactive voice signal, comparing the second reflection coefficient with a reflection threshold, and selecting the active voice mode if the second reflection coefficient is greater than the reflection threshold. The method may further comprise selecting the inactive voice mode if the second reflection coefficient is not greater than the reflection threshold. The method may also comprise analyzing the input signal to determine an energy level of the input signal, and selecting the active voice mode if the energy level is greater than an energy threshold.

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 are provided methods and devices for generating excitation values for a speech signal. In one aspect, an example method comprises obtaining one or more characteristics of a first speech frame of the speech signal, deriving a first seed value based on the one or more characteristics of the first speech frame, providing the first seed value to a Gaussian time series generator; and using the Gaussian time series generator to generate an excitation values for the first frame. The one or more characteristics may include a spectrum information of the first frame, an energy information of the first frame, or a gain information of the first frame.

Abstract: Speech coding systems include multi-rate speech codecs having an encoder and a decoder. Silence description coding for multi-rate speech coding systems that employ discontinued transmission is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes.

Abstract: There is provided a voice activity detection method for indicating an active voice mode and an inactive voice mode. The method comprises receiving a first portion of an input signal; determining that the first portion of the input signal includes an active voice signal; indicating the active voice mode in response to the determining that the first portion of the input signal includes the active voice signal; receiving a second portion of the input signal immediately following the first portion of the input signal; determining that the second portion of the input signal includes an inactive voice signal; extending the indicating the active voice mode for a period of time after determining that the second portion of the input signal includes the inactive voice signal, wherein the period of time varies based on one or more conditions; and indicating the inactive voice mode after expiration of the period of time.

Abstract: There is provided a method of updating a noise state of a voice activity detector (VAD) for indicating an active voice mode and an inactive voice mode. The method comprises receiving an input signal having a plurality of frames, determining an elapsed time since the last update of the noise state, updating the noise state of the VAD if the elapsed time exceeds a predetermined time, determining an average minimum energy based on two or more of the plurality of frames, determining a current minimum energy based on a current frame of the plurality of frames, updating the noise state of the VAD if the average minimum energy is less than the current minimum energy, and updating the noise state of the VAD if the average minimum energy is greater than the current minimum energy plus a first predetermined value.

Abstract: A multi-channel speech processor for encoding speech in a packet network environment is disclosed. In one illustrative aspect, a complexity resource manager (CRM) is executed by a controller or processor. The CRM manages the level of complexity of encoding which is used by a signal processing unit (SPU) to convert the speech signal into packet data. In general, the CRM determines the level of complexity of encoding based on a calculated complexity budget, where the complexity budget is determined based on the time required to process prior speech signal channels and the time available to process the remaining channels. In this way, the CRM is able to control the overall complexity of the speech processor through its ability to signal the SPU to encode speech signal in a complexity reduced mode based on the calculated complexity budget under certain conditions.

Abstract: There is provided a method of adjusting power consumption by a wireless device having a codec and a battery, where the codec has a plurality of states of operation and the battery has a plurality of power levels. The method comprises operating the wireless device at a first power level of the plurality of power levels and a first state of the plurality of states of operation, while in communication with a remote device; receiving a power level indication indicative of a second power level of the plurality of power levels; selecting a second state of the plurality of states of operation for the codec corresponding to the second power level of the plurality of power levels; and modifying the codec to operate at the second state of the plurality of states of operation.