Abstract: In a Noise Feedback Coding (NFC) system operable in a ZERO-STATE condition and a ZERO-INPUT condition, the NFC system including at least one filter having a filter memory, a method of updating the filter memory. The method comprises: (a) producing a ZERO-STATE contribution to the filter memory when the NFC system is in the ZERO-STATE condition; (b) producing a ZERO-INPUT contribution to the filter memory when the NFC system is in the ZERO-INPUT condition; and (c) updating the filter memory as a function of both the ZERO-STATE contribution and the ZERO-INPUT contribution.

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 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: 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 and system are provided for removing discontinuities associated with synthesizing a corrupted frame output from a decoder including one or more predictive filters. The corrupted frame is representative of one segment of a decoded signal. The method comprises copying a first number of stored samples of the decoded signal in accordance with a time lag and a scaling factor, and calculating a first number of ringing samples output from at least one of the filters.

Abstract: A technique for performing frame erasure concealment (FEC) in a speech decoder. One or more non-erased frames of a speech signal are decoded in a block-independent manner. When an erased frame is detected, a short-term predictive filter and a long-term predictive filter are derived based on previously-decoded portions of the speech signal. A periodic waveform component is generated using the short-term predictive filter and the long-term predictive filter. A random waveform component is generated using the short-term predictive filter. A replacement frame is generated for the erased frame. The replacement frame may be generated based on the periodic waveform component, the random waveform component, or a mixture of both.

Abstract: A method of performing an excitation Vector Quantization (VQ) in a Noise Feedback Coding environment involves reorganizing a calculation of an energy of an error vector for each of a plurality of candidate excitation vectors of a codebook. The energy of the error vector is a cost function that is minimized during a search of the codebook for a best candidate excitation VQ vector. The reorganization includes expanding a Mean Squared Error (MSE) term of the error vector, excluding an energy term that is invariant to the candidate excitation vector, and pre-computing energy terms of ZERO-STATE responses of the candidate excitation vectors that are invariant to sub-vectors of a subframe. Another method searches a signed codebook. Both methods use correlation techniques.

Abstract: A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.

Abstract: The present invention is directed to a multiple description transmission system that provides redundancy to combat transmission channel impairments. The multiple description transmission system includes a first and second wireless telephone. The first wireless telephone includes the following: an array of microphones, wherein each microphone in the array of microphones is configured to receive voice input from a user and to produce a voice signal corresponding thereto; an encoder coupled to the microphone array and configured to encode each of the voice signals; and a transmitter coupled to the encoder and configured to transmit each of the encoded voice signals. The second wireless telephone includes the following: a receiver configured to receive the transmitted signals; a decoder coupled to the receiver and configured to decode the signals received by the receiver, thereby producing an output signal; and a loudspeaker that receives the output signal and produces a pressure sound wave corresponding thereto.

Abstract: The present invention is directed to a telephone equipped with multiple microphones that provides improved performance during operation of the telephone in a speaker-phone mode. For example, the multiple microphones can be used to improve voice activity detection, which in turn, can improve echo cancellation. In addition, the multiple microphones can be configured as an adaptive microphone array and used to reduce the effects of (i) room reverberation, when a near-end user is speaking, and/or (ii) acoustic echo, when a far-end user is speaking.

Abstract: 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, wherein at least one of the first microphone and the second microphone is a unidirectional microphone. The first microphone outputs a first audio signal that includes 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 wireless telephone having an array of microphones and a digital signal processor (DSP) and a method of processing audio signals from a wireless telephone having an array of microphones and a DSP. The wireless telephone includes an array of microphones and a DSP. Each microphone in the array is configured to receive sound waves emanating from the surrounding environment and to generate an audio signal corresponding thereto. The DSP is coupled to the array of microphones. The DSP is configured to receive the audio signals from the array of microphones, to detect a direction of arrival (DOA) of a sound wave emanating from the mouth of a user based on the audio signals and to adaptively combine the audio signals based on the DOA to produce a first audio output signal.

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 system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.

Abstract: Typical communication systems operate with a single channel decoder, and hence would have to settle for the performance from the single channel decoder regardless of the conditions of the communications channel. The present invention uses a hybrid channel decoder comprising multiple channel decoders, each configured to optimize the quality of the re-constructed signal for different channel conditions. Therefore, the desired decoder can be selected as conditions of the communications channel, or the data signal, change over time, so as to optimize the re-constructed data signal. In embodiments, the data signal is a speech signal.

Abstract: A method of searching a plurality of Vector Quantization (VQ) codevectors for a preferred one of the VQ codevectors to be used as an output of a vector quantizer for encoding a speech signal, includes determining a quantized prediction residual vector, and calculating a corresponding unquantized prediction residual vector and the energy of the difference between these two vectors (that is, a VQ error vector).

Abstract: A system and method for high-quality variable speed playback of audio-visual (A/V) media is provided. The system receives an encoded visual signal and an encoded audio signal. The encoded visual signal is decoded to generate a decoded visual signal and the encoded audio signal is decoded to generate a decoded audio signal. The decoded audio signal is time scale modified to generate a time scale modified audio signal. The decoded visual signal and the time scale modified audio signal are then synchronized for playback at a predefined playback speed. Only partial decoding of the encoded audio signal may be performed to conserve processing power.

Abstract: A method of concealing bit errors in a signal is provided. The method comprises encoding a signal parameter according to a set of constraints placed on a signal parameter quantizer. The encoded signal parameter is decoded and compared against the set of c-onstraints. Finally, the method includes declaring the decoded signal parameter invalid when the set of constraints is violated.

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 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.