Abstract: A multi-rate speech coded supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The fixed codebook contains pulse subcodebooks and noise-like subcodebooks. To assist in selection of one of the subcodebooks, an adaptive weighting approach is applied in a searching procedure wherein residual classification and various parameters are used to generate a weighting function that is used to favor one subcodebook over another.

Abstract: The invention provides a speech coding system with a music classifier. An encoder is disposed to receive an input signal and provides a bitstream based upon a speech coding of a portion of the input signal. The encoder provides a classification of the input signal as one of noise, speech, and music. The music classifier analyzes or determines signal properties of the input signal. The music classifier compares the signal properties to thresholds to determine the classification of the input signal.

Abstract: There is provided a speech decoder comprising a means for generating an excitation signal and a means for performing harmonic analysis and synthesis on the excitation signal in order to generate a smooth, periodic speech signal. The speech decoder further comprises a mixing means for mixing the excitation signal with the smooth, periodic signal and a synthesizing means for synthesizing the modified excitation signal into a speech signal that can be played to a user through a listening means. There is also provided a receiver that incorporates a speech decoder such as the decoder described above as well as a method for speech decoding.

Abstract: There is provided a conference bridge or transcoder configured to intelligently handle multiple speech channels in the contest of a packet network, wherein various speech channels may adhere to variety of speech encoding standards. For example, the conference bridge establishes framing and alignment of multiple incoming speech channels associated with multiple participants, extracts parameters from the speech samples, mixes the parameters, and re-encodes the resulting speech samples for transmission to the participants. In one aspect, a speech processing method comprises decoding a first bitstream according to a first coding scheme to generate first speech samples and a first side information; generating second speech samples and a second side information using the first speech samples and the first side information, for use according to a second coding scheme; and creating a second bitstream, encoded based on the second coding scheme, using the second speech samples and the second side information.

Abstract: In a coding procedure, coding parameters are selected for coding the speech signal to achieve enhanced perceptual quality of reproduced speech. At least one coding parameter value or preferential coding parameter value is selected to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

Abstract: A method for preparing a speech signal for encoding comprises determining whether the spectral content of an input speech signal is representative of a defined spectral characteristic (e.g., a defined characteristic slope). A frequency specific filter component of a weighting filter is controlled based on the determination of the spectral content of the speech signal or/and its location in the encoder. A core weighting filter component of the weighting filter may be maintained regardless of the spectral content of the speech signal.

Abstract: A speech encoding comb codebook structure for providing good quality reproduced low bit-rate speech signals in a speech encoding system. The codebook structure requires minimal training, if any, and allows for reduced complexity and memory requirements. The codebook includes a first and at least one additional sub-codebooks, each having a plurality of code-vectors. The codebook may be randomly populated. All even elements may be set to zero in a first codebook, and all odd elements may be set to zero on a second codebook. The resulting comb codebook includes code-vector combination of the code-vectors from the sub-codebooks. In certain embodiments, the code-vectors of the sub-codebooks may contain zero valued elements. In other embodiments where the code-vectors of the sub-codebooks contain only non-zero elements, zero valued elements may be inserted in between the non-zero elements of the sub-codebooks during the forming of the resultant comb codebook.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors.

Abstract: In a coding procedure, a spectral content of a speech signal is estimated. A preferential coding algorithm or preferential value of at least one coding parameter is selected based on the estimated spectral content of the speech signal. The speech signal is coded in accordance with the selected coding algorithm or the selected coding parameter to control the operation of one or more of the following: a pre-processing filter, a post-processing filter, a coding control coefficient, a weighting filter, a synthesis filter, and a quantization table.

Abstract: A pitch lag coding device and method using interframe correlation inherent in pitch lag values to reduce coding bit requirements. A pitch lag value is extracted for a given speech frame, and then refined for each subframe. For every speech frame having N samples of speech, LPC analysis and vector quantization are performed for the whole coding frame. The LPC residual obtained for each frame is then processed such that pitch lag values for all subframes within the coding frame are analyzed concurrently. The remaining coding parameters, i.e., the codebook search, gain parameters, and excitation signal, are then analyzed sequentially according to their respective subframes.

Abstract: A speech encoding comb codebook structure for providing good quality reproduced low bit-rate speech signals in a speech encoding system. The codebook structure requires minimal training, if any, and allows for reduced complexity and memory requirements. The codebook includes a first and at least one additional sub-codebooks, each having a plurality of code-vectors. The codebook may be randomly populated. All even elements may be set to zero in a first codebook, and all odd elements may be set to zero on a second codebook. The resulting comb codebook includes code-vector combination of the code-vectors from the sub-codebooks. In certain embodiments, the code-vectors of the sub-codebooks may contain zero valued elements. In other embodiments where the code-vectors of the sub-codebooks contain only non-zero elements, zero valued elements may be inserted in between the non-zero elements of the sub-codebooks during the forming of the resultant comb codebook.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. A speech encoder employing various encoding schemes based upon parameters including an available transmission bit rate. In addition, the speech encoder is operable to identify and apply an optimal encoding scheme for a given speech signal. The speech encoder may be applied code-excited linear prediction when the available bit rate is above a predetermined upper threshold. Pitch preprocessing, including continuous warping, may be applied when it is below a predetermined lower threshold.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. A speech encoder employing various encoding schemes based upon parameters including an available transmission bit rate. In addition, the speech encoder is operable to identify and apply an optimal encoding scheme for a given speech signal. The speech encoder may be applied code-excited linear prediction when the available bit rate is above a predetermined upper threshold. Pitch preprocessing, including continuous warping, may be applied when it is below a predetermined lower threshold.

Abstract: Silence description coding for multi-rate speech coding systems that employ discontinued transmission. Speech coding systems include multi-rate speech codecs having an encoder and a decoder. The silence description coding 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: Silence description coding for multi-rate speech coding systems that employ discontinued transmission. Speech coding systems include multi-rate speech codecs having an encoder and a decoder. The silence description coding 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: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech coder distinguishes various voice signals as a function of their voice content. For example, a Voice Activity Detection (VAD) algorithm selects an appropriate coding scheme depending on whether the speech signal comprises active or inactive speech. The encoder may consider varying characteristics of the speech signal including sharpness, a delay correlation, a zero-crossing rate, and a residual energy.

Abstract: A method of coding speech under background noise conditions or during noise-like speech periods wherein during active voice speech segments an analysis-by-synthesis method is used. However, when a background noise segment or noise-like speech segment is detected, an adaptive code book (pitch prediction) contribution is used as a source of a pseudo-random sequence in order to provide a better representation of the background noise or the noise-like speech. An improved gain quantization scheme is also employed when a background noise segment is detected, wherein energy of the total excitation with quantized gains is matched to the energy of total excitation with unquantized gains.

Abstract: A system and method to improve the quality of coded speech coexisting with background noise. For instance, the present invention receives a coded speech signal via a communication network and then decodes and synthesizes the different parameters contained within it to produce a synthesized speech signal. The present invention determines the non-speech periods that are represented within the synthesized speech signal. The determined non-speech periods are then utilized to determine and code LPC parameters needed for background noise synthesis. Because medium or low bit rate LPC-coded speech during voice activity periods has the coexisting background noise attenuated, the decoded signal has audible abrupt changes in the level of the background noise. To improve decoded speech quality, the present invention adds simulated background noise to decoded noisy speech when synthesizing the noisy speech signal during voice activity periods.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. The encoder applies adaptive gain reduction to optimize selection of appropriate gain contributions from the adaptive and fixed codebooks. Specifically, the encoder uses a first target signal to identify a contribution (a best code vector and a gain) from the adaptive codebook. Thereafter, a contribution from the fixed codebook is selected. The gain associated with the adaptive codebook contribution is then reduced by a factor, and the gain contribution from the fixed codebook is searched a second time, permitting fine tuning of the overall contribution.

Abstract: A method of coding speech under background noise conditions wherein during active voice speech segments an analysis-by-synthesis method is used. However, when a background noise segment is detected, an adaptive code book (pitch prediction) contribution is used as a source of a pseudo-random sequence in order to provide a better representation of the background noise. An improved gain quantization scheme is also employed when a background noise segment is detected, wherein an energy of the total excitation with quantized gains is matched to an energy of total excitation with unquantized gains.