Abstract: Systems, methods, and apparatus for the detection of signals having spectral peaks with narrow bandwidth are described herein. The range of described configurations includes implementations that perform such detection using parameters of a linear prediction coding (LPC) analysis scheme.

Abstract: A low-bit-rate coding technique for unvoiced segments of speech includes the steps of extracting high-time-resolution energy coefficients from a frame of speech, quantizing the energy coefficients, generating a high-time-resolution energy envelope from the quantized energy coefficients, and reconstituting a residue signal by shaping a randomly generated noise vector with quantized values of the energy envelope. The energy envelope may be generated with a linear interpolation technique. A post-processing measure may be obtained and compared with a predefined threshold to determine whether the coding algorithm is performing adequately.

Abstract: This disclosure describes deblock filtering techniques in which an in-loop deblock filter of a first codec is used as a post deblock filter of a second codec. A number of techniques are also described to facilitate input parameter adjustments and allow for the effective use of the filter with both codecs. The techniques can simplify the architecture of a device that includes multiple codecs operating according to different coding standards. Specifically, the different codecs can use the same deblocking filter regardless of whether the coding standard calls for in-loop filtering or whether post filtering is used. For example, a filter designed as an in-loop deblocking filter for a codec that complies with the ITU-T H.264 coding standard can be used as a post deblocking filter for MPEG-4 video.

Abstract: A method and apparatus for the variable rate coding of a speech signal. An input speech signal is classified and an appropriate coding mode is selected based on this classification. For each classification, the coding mode that achieves the lowest bit rate with an acceptable quality of speech reproduction is selected. Low average bit rates are achieved by only employing high fidelity modes (i.e., high bit rate, broadly applicable to different types of speech) during portions of the speech where this fidelity is required for acceptable output. Lower bit rate modes are used during portions of speech where these modes produce acceptable output. Input speech signal is classified into active and inactive regions. Active regions are further classified into voiced, unvoiced, and transient regions. Various coding modes are applied to active speech, depending upon the required level of fidelity. Coding modes may be utilized according to the strengths and weaknesses of each particular mode.

Abstract: Method and apparatus for subsampling phase spectrum information by analyzing and reconstructing a prototype of a frame. The prototype is analyzed by correlating phase parameters generated from the prototype with phase parameters generated from a reference prototype in multiple frequency bands. The prototype is reconstructed using linear phase shift values by producing a set of phase parameters of the reference prototype, generating a set of linear phase shift values associated with the prototype, and composing a phase vector from the set of phase parameters and the set of linear phase shift values across multiple frequency bands. The prototype is reconstructed using circular rotation values by producing a set of circular rotation values associated with the prototype, generating a set of bandpass waveforms associated with the phase parameters of the reference prototype in multiple frequency bands, and modifying the set of bandpass waveforms based upon the circular rotation values.

Abstract: This disclosure describes video encoding techniques and video encoding devices that implement such techniques. In one embodiment, this disclosure describes a video encoding device comprising a motion estimator that computes a motion vector predictor based on motion vectors previously calculated for video blocks in proximity to a current video block to be encoded, and uses the motion vector predictor in searching for a prediction video block used to encode the current video block, and a motion compensator that generates a difference block indicative of differences between the current video block to be encoded and the prediction video block.

Abstract: An adaptive Intra-refresh (IR) technique for digital video encoding adjusts IR rate based on video content, or a combination of video content and channel condition. The IR rate may be applied at the frame level or macroblock (MB) level. At the frame level, the IR rate specifies the percentage of MBs to be Intra-coded within the frame. At the MB level, the IR rate defines a statistical probability that a particular MB is to be Intra-coded. The IR rate is adjusted in proportion to a combined metric that weighs estimated channel loss probability, frame-to-frame variation, and texture information. The IR rate can be determined using a close-form solution that requires relatively low implementation complexity. For example, such a close-form does not require iteration or an exhaustive search. In addition, the IR rate can be determined from parameters that are available before motion estimation and compensation are performed.

Abstract: Method and apparatus for subsampling phase spectrum information by analyzing and reconstructing a prototype of a frame. The prototype is analyzed by correlating phase parameters generated from the prototype with phase parameters generated from a reference prototype in multiple frequency bands. The prototype is reconstructed using linear phase shift values by producing a set of phase parameters of the reference prototype, generating a set of linear phase shift values associated with the prototype, and composing a phase vector from the set of phase parameters and the set of linear phase shift values across multiple frequency bands. The prototype is reconstructed using circular rotation values by producing a set of circular rotation values associated with the prototype, generating a set of bandpass waveforms associated with the phase parameters of the reference prototype in multiple frequency bands, and modifying the set of bandpass waveforms based upon the circular rotation values.

Abstract: A low-bit-rate coding technique for unvoiced segments of speech includes the steps of extracting high-time-resolution energy coefficients from a frame of speech, quantizing the energy coefficients, generating a high-time-resolution energy envelope from the quantized energy coefficients, and reconstituting a residue signal by shaping a randomly generated noise vector with quantized values of the energy envelope. The energy envelope may be generated with a linear interpolation technique. A post-processing measure may be obtained and compared with a predefined threshold to determine whether the coding algorithm is performing adequately.

Abstract: A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.

Abstract: A low-bit-rate coding technique for unvoiced segments of speech includes the steps of extracting high-time-resolution energy coefficients from a frame of speech, quantizing the energy coefficients, generating a high-time-resolution energy envelope from the quantized energy coefficients, and reconstituting a residue signal by shaping a randomly generated noise vector with quantized values of the energy envelope. The energy envelope may be generated with a linear interpolation technique. A post-processing measure may be obtained and compared with a predefined threshold to determine whether the coding algorithm is performing adequately.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. To reduce the number of computations required to choose the optimal codebook vector, a subset of codevectors are selected based upon optimal pulse locations, wherein the subset of codevectors form a subcodebook. Rather than searching the entire codebook, only the entries of the subcodebook are searched.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. In encoding schemes that use forward and backward pitch enhancement, storage and processor load is reduced by approximating a two-dimensional autocorrelation matrix with a one-dimensional autocorrelation vector. The approximation is possible when a cross-correlation element is configured to determine the autocorrelation matrix of an impulse response and a pulse energy determination element is configured to determine the energy of a pulse code vector that incorporates secondary pulse positions.

Abstract: Methods and apparatus are presented for reducing the number of bits needed to represent an excitation waveform. An acoustic signal in an analysis frame is analyzed to determine whether it is a band-limited signal. A sub-sampled sparse codebook is used to generate the excitation waveform if the acoustic signal is a band-limited signal. The sub-sampled sparse codebook is generated by decimating permissible pulse locations from the codebook track in accordance with the frequency characteristic of the acoustic signal.

Abstract: Techniques are presented herein to provide tandem-free operation between two wireless terminals through two otherwise incompatible wireless networks. Specifically, embodiments provide tandem-free operation between a wireless terminal communicating through a continuous transmission (CTX) wireless channel to a wireless terminal communicating through a discontinuous transmission (DTX) wireless channel. In a first aspect, inactive speech frames are translated between DTX and CTX formats. In a second aspect, each wireless terminal includes an active speech decoder that is compatible with the active speech encoder on the opposite end of the mobile-to-mobile connection.

Abstract: A method and apparatus for the variable rate coding of a speech signal. An input speech signal is classified and an appropriate coding mode is selected based on this classification. For each classification, the coding mode that achieves the lowest bit rate with an acceptable quality of speech reproduction is selected. Low average bit rates are achieved by only employing high fidelity modes (i.e., high bit rate, broadly applicable to different types of speech) during portions of the speech where this fidelity is required for acceptable output. Lower bit rate modes are used during portions of speech where these modes produce acceptable output. Input speech signal is classified into active and inactive regions. Active regions are further classified into voiced, unvoiced, and transient regions. Various coding modes are applied to active speech, depending upon the required level of fidelity. Coding modes may be utilized according to the strengths and weaknesses of each particular mode.

Abstract: Methods and apparatus are presented for determining the type of acoustic signal and the type of frequency spectrum exhibited by the acoustic signal in order to selectively delete parameter information before vector quantization. The bits that would otherwise be allocated to the deleted parameters can then be re-allocated to the quantization of the remaining parameters, which results in an improvement of the perceptual quality of the synthesized acoustic signal. Alternatively, the bits that would have been allocated to the deleted parameters are dropped, resulting in an overall bit-rate reduction.

Abstract: A method and apparatus for the variable rate coding of a speech signal. An input speech signal is classified and an appropriate coding mode is selected based on this classification. For each classification, the coding mode that achieves the lowest bit rate with an acceptable quality of speech reproduction is selected. Low average bit rates are achieved by only employing high fidelity modes (i.e., high bit rate, broadly applicable to different types of speech) during portions of the speech where this fidelity is required for acceptable output. Lower bit rate modes are used during portions of speech where these modes produce acceptable output. Input speech signal is classified into active and inactive regions. Active regions are further classified into voiced, unvoiced, and transient regions. Various coding modes are applied to active speech, depending upon the required level of fidelity. Coding modes may be utilized according to the strengths and weaknesses of each particular mode.

Abstract: Method and apparatus for subsampling phase spectrum information by analyzing and reconstructing a prototype of a frame. The prototype is analyzed by correlating phase parameters generated from the prototype with phase parameters generated from a reference prototype in multiple frequency bands. The prototype is reconstructed using linear phase shift values by producing a set of phase parameters of the reference prototype, generating a set of linear phase shift values associated with the prototype, and composing a phase vector from the set of phase parameters and the set of linear phase shift values across multiple frequency bands. The prototype is reconstructed using circular rotation values by producing a set of circular rotation values associated with the prototype, generating a set of bandpass waveforms associated with the phase parameters of the reference prototype in multiple frequency bands, and modifying the set of bandpass waveforms based upon the circular rotation values.

Abstract: A frame erasure compensation method in a variable-rate speech coder includes quantizing, with a first encoder, a pitch lag value for a current frame and a first delta pitch lag value equal to the difference between the pitch lag value for the current frame and the pitch lag value for the previous frame. A second, predictive encoder quantizes only a second delta pitch lag value for the previous frame (equal to the difference between the pitch lag value for the previous frame and the pitch lag value for the frame prior to that frame). If the frame prior to the previous frame is processed as a frame erasure, the pitch lag value for the previous frame is obtained by subtracting the first delta pitch lag value from the pitch lag value for the current frame. The pitch lag value for the erasure frame is then obtained by subtracting the second delta pitch lag value from the pitch lag value for the previous frame.