Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: A bit error concealment (BEC) system and method is described herein that detects and conceals the presence of click-like artifacts in an audio signal caused by bit errors introduced during transmission of the audio signal within an audio communications system. A particular embodiment of the present invention utilizes a low-complexity design that introduces no added delay and that is particularly well-suited for applications such as Bluetooth® wireless audio devices which have low cost and low power dissipation requirements.

Abstract: Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders.

Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

Abstract: A method and system are provided for synthesizing a number of corrupted frames output from a decoder including one or more predictive filters. The corrupted frames are representative of one segment of a decoded signal (sq(n)) output from the decoder. The method comprises determining a first preliminary time lag (ppfe1) based upon examining a predetermined number (K) of samples of another segment of the decoded signal and determining a scaling factor (ptfe) associated with the examined number (K) of samples when the first preliminary time lag (ppfe1) is determined. The method also comprises extrapolating one or more replacement frames based upon the first preliminary time lag (ppfe1) and the scaling factor (ptfe).

Abstract: The present invention is directed to a wireless telephone having a first microphone and a second microphone and a method for processing audio signal in a wireless telephone having a first microphone and a second microphone. The wireless telephone includes a first microphone, a second microphone, and a signal processor. The first microphone outputs a first audio signal, the first audio signal comprising a voice component and a background noise component. The second microphone outputs a second audio signal. The signal processor increases a ratio of the voice component to the noise component of the first audio signal based on the content of at least one of the first audio signal and the second audio signal to produce a third audio signal.

Abstract: A method of attempting to determine a pitch period of an audio signal using a correlation-based signal derived from the audio signal. The correlation-based signal has known peaks, having been quadratically interpolated and filtered with coefficients that are a function of the interpolation ratio, each corresponding to a respective one of known time lags. The method comprises: identifying a time lag among the time lags; determining if there exists another time lag (i) within a time lag range of a respective one of one or more integer multiples of the identified time lag, and (ii) corresponding to a peak exceeding a peak threshold; and if the determination of step (a) passes, then returning the identified time lag as a time lag indicative of the pitch period.

Abstract: The present invention is directed to improved speech intelligibility in telephones with multiple microphones. Such a telephone includes a first microphone, a second microphone, a voice activity detector (VAD), a receiver module, and a signal processor. The first microphone outputs a first audio signal, which comprises a voice component when a near-end user talks and a background noise component. The second microphone outputs a second audio signal. The VAD generates a voice activity signal responsive to a ratio between the first audio signal and the second audio signal. The voice activity signal identifies time intervals in which the voice component of the near-end user is present in the first audio signal. The receiver module receives a third audio signal, which comprises a voice component of a far-end user. The signal processor modifies the third audio signal responsive to the voice activity signal.

Abstract: A filter controller processes a decoded speech (DS) signal. The DS signal has a spectral envelope including a first plurality of formant peaks having different respective amplitudes. The controller produces, from the DS signal, a spectrally-flattened DS signal that is a time-domain signal. The spectrally-flattened time-domain DS signal has a spectral envelope including a second plurality of formant peaks. Each of the second plurality of formant peaks approximately coincides in frequency with a respective one of the first plurality of formant peaks. Also, the second plurality of formant peaks have approximately equal respective amplitudes. Next, the controller derives, from the spectrally-flattened time-domain DS signal, a set of filter coefficients representative of a filter response that is to be used to filter the DS signal.

Abstract: Codec structures for achieving two-stage prediction and two-stage noise spectral shaping at the same time, resulting in a Two-Stage Noise Feedback Coding (TSNFC) method. One approach combines two predictors into a single composite predictor; and derives appropriate filters for use in a conventional single-stage NFC codec structure. Another approach duplicates a conventional single-stage NFC codec structure in a nested manner, thereby decoupling the operations of the long-term prediction and long-term noise spectral shaping from the operations of the short-term prediction and short-term noise spectral shaping.

Abstract: A method for adaptive long-term filtering of an audio signal, such as a decoded speech signal. The method includes measuring a smoothed periodicity of an audio signal segment, such as an audio frame, wherein the smoothed periodicity is measured by low-pass filtering an instantaneous periodicity of the audio signal segment. The periodicity of the audio signal segment is then increased in a manner that depends upon whether the smoothed periodicity is less than a predetermined threshold. By utilizing a smoothed periodicity measurement in this fashion, more accurate control of the post-filter is provided as compared to conventional solutions. Additionally, the method includes deriving filters by interpolating between filter responses of adjacent audio signal segments to minimize distortion at segment boundaries.

Abstract: A system is described that performs frame erasure concealment (FEC) to generate frames of an output speech signal corresponding to erased frames of encoded bit-stream in a manner that conceals the quality-degrading effects of such erased frames. An embodiment of the invention advantageously does not introduce additional delay, has lower state memory requirement than the FEC technique specified in G.711 Appendix I, and produces better speech quality than the FEC technique specified in G.711 Appendix I while still allowing for reduced computational complexity and code size.

Abstract: A modified synchronized overlap add (SOLA) algorithm for performing high-quality, low-complexity audio time scale modification (TSM) is described. The algorithm produces good output audio quality with a very low complexity and without producing additional audible distortion during dynamic change of the audio playback speed. The algorithm may achieve complexity reduction by performing the maximization of normalized cross-correlation using decimated signals. By updating the input buffer and the output buffer in a precise sequence with careful checking of the appropriate array bounds, the algorithm may also achieve seamless audio playback during dynamic speed change with a minimal requirement on memory usage.

Abstract: A system and method of concealing bit errors in a signal are provided. An exemplary method detects bit errors in an input signal having at least a current signal segment and a previous signal segment. The previous signal segment has a log-gain value qlg(m?1) and immediately precedes the current signal segment. The method comprises estimating a level lvl(m?1) of the input signal and determining a log-gain value qlg(m) of the current signal segment within the input signal. The method also comprises determining a difference between the gain value of the current signal segment and the previous signal segment and determining whether the difference exceeds a threshold. wherein the threshold is adaptive to the input signal level.

Abstract: A low-complexity sampling rate conversion (SRC) method and apparatus for the processing of digital audio signals. A first stage upsamples an input audio signal to generate an upsampled audio signal. For example, the first stage may perform 1:2 upsampling using a halfband filter. A second stage re-samples the upsampled audio signal from the first stage at a target sampling rate. For example, re-sampling may be achieved using linear interpolation.

Abstract: A method to eliminate discontinuities in an adaptively filtered signal includes filtering a beginning portion of a current signal frame using a past set of filter coefficients, thereby producing a first filtered frame portion. The method also includes filtering the beginning portion of the current signal frame using a current set of filter coefficients, thereby producing a second filtered frame portion. The method also includes modifying the second filtered frame portion with the first filtered frame portion so as to smooth a possible filtered signal discontinuity between the second filtered frame portion and a past filtered frame produced using the past filter coefficients.

Abstract: A system and method for reducing end-to-end delay associated with telephone calls carried over a voice over packet network in a manner that avoids overloading of the network. The system and method involves monitoring one or more parameters indicative of an amount of bandwidth being utilized on the voice over packet network, responsive to the monitoring, determining that a level of bandwidth utilization on the voice over packet network has changed, responsive to the determination that the level of bandwidth utilization on the voice over packet network has changed, issuing a command to a telephony device, and responsive to receipt of the command by the telephony device, changing the size of packets used for carrying frames of encoded voice signals associated with a telephone call from a first packet size to a second packet size.

Abstract: A technique is described herein for updating a state of a decoder configured to decode a series of frames representing an encoded audio signal. In accordance with the technique, an output audio signal associated with a lost frame in the series of frames is synthesized. The decoder state is set to align with the synthesized output audio signal at a frame boundary. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with a first received frame after the lost frame in the series of frames, wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoder state is then reset based on the time lag.