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 speech communication system is provided that uses pitch information, pitch lags, pitch gains, energy and/or other speech characteristics about the outgoing speech and the unknown signal on a frame basis to determine if the unknown signal is an echo signal of the outgoing speech or if the unknown signal also contains speech from a second talker (double talk). Additionally, a plurality of frames of these characteristics of the outgoing speech signal and the unknown incoming signal may be buffered so that the analysis and comparison can be made more efficiently and quickly in the frame domain as opposed to a time domain. Multiple characteristics may be optionally weighted and then analyzed. The system and method may further determine a level of confidence, based on any criterion, in the determination that may then be used to adjust the gain of a filter.

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: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

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: In an exemplary conversion scheme, a frame of a first speech signal comprising a plurality of frames encoded at a plurality of first rates, including a first non-speech rate, is received. The rate of the received frame is determined, and if the received frame is encoded at the first non-speech rate, then the received frame is re-encoded at either a second or third non-speech rate to generate a frame of a second speech signal. Moreover, a system for converting a speech signal comprises a receiver for receiving a frame of a first speech signal and a processor capable of determining the encoding rate of the received frame and re-encoding the received frame at either a second or third non-speech rate if the received frame was originally encoded at a first non-speech rate.

Abstract: A digitized signal detection system where the bit rate encoding is changed dynamically to provide encoding for different type signals and formats at bit rates optimized to properly reconstruct the input signal whether speech or non-speech and therefore can transfer signals of different character on a frame by frame basis. A change of encoding format can make the system a speech or music recognizer dependent what is to be listened for. Three basic components a recognizer which categorizes the type of input signal, an evaluator which evaluates the category of quality of the reconstructed signal and a recommender which make as recommendation based on the quality to change standards to encode the signals received pursuant to a standard which provides for improved quality. The dynamic signal detector receives the input signal directly and extracts the parameters for evaluation. These parameters are tested and a determination made if a switch of standards are required. To improve the reconstructed signal.

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: A text-prompted speaker verification system that can be configured by users based on a desired level of security. A user is prompted for a multiple-digit (or multiple-word) password. The number of digits or words used for each password is defined by the system in accordance with a user set preferred level of security. The level of training required by the system is defined by the user in accordance with a preferred level of security. The set of words used to generate passwords can also be user configurable based upon the desired level of security. The level of security associated with the frequency of false accept errors verses false reject errors is user configurable for each particular application.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: An exemplary decoder comprises a receiver that receives parameters of a speech signal on a frame-by-frame basis, a control logic for decoding parameters and for resynthesizing the speech signal, the control logic including a minimum spacing indicative of a minimum difference required between LSFs of consecutive frames, a frame recovery logic that, when a lost frame detector detects a lost frame, sets the minimum spacing for the lost frame to a first value which is greater than the minimum spacing for the previously received frame, and/or uses pitch lag parameters of a plurality of previously received frames to extrapolate a pitch lag parameter for the lost frame, and/or sets gain parameter of a subframe of the lost frame in a first manner if the lost gain parameter is an adaptive codebook gain parameter and in a second manner if the lost gain parameter is a fixed codebook gain parameter.

Abstract: In a cellular telephone system where a digital cellular telephone is connected to a regular telephone through the public switched telephone network (PSTN), a speech encoder/decoder is used with an A/&mgr;-Law encoder/decoder causing annoying audible noise at very low levels because of the quantization characteristics of the A/&mgr;-Law encoder/decoder. This noise is eliminated by adding a digital constant to the output of the speech coder, shifting the low level signal away from zero. The resulting DC level added to the speech signal is inaudible to the PSTN telephone user and does not degrade speech quality. Alternatively, the constant added to the output of the speech coder is confined to a small value added to the speech coder output to move the entire speech coder output during the silence period, between speech periods, above zero or below zero.

Abstract: A speech coding system that employs hybrid linear prediction coding during extraction of linear prediction coefficients within ITU-Recommendation speech coding standards. The present invention is operable within linear prediction speech coding systems including code-excited linear prediction speech coding systems, and it provides for a substantially improved perceptual quality of reproduced speech signals when compared to conventional speech coding methods that employ the commonly known auto-correlation method that is based on minimizing the linear prediction coding (LPC) prediction error energy. The invention is operable to provide for high perceptual quality of reproduced speech signals having substantial differences of energy in various frequency bands.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A speech coding system that employs target signal reference shifting in code-excited linear prediction speech coding. The speech coding system performs modification of a target signal that is used to perform speech coding of a speech signal. The modified target signal that is generated from a preliminary target signal is then used to calculate an adaptive codebook gain that is used to perform speech coding of the speech signal. The speech coding performed in accordance with the present invention provides for a substantially reduced bit-rate of operation when compared to conventional speech coding methods that inherently require a significant amount of bandwidth to encode a fractional pitch lag delay during pitch prediction that is performed within conventional code-excited linear prediction speech coding systems.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A zero delay continuous long term (LT) pre-processing method operable in a speech codec that introduces no delay. The present invention provides an elegant solution to perform long term (LT) pre-processing of the pitch lag of a speech signal to save a large number of bits required in various speech coding methods, including the code-excited linear prediction method. The present invention is ideal for speech coding standards and methods that any undesirable delay at the end of a speech frame of the speech signal. The present invention overcomes a significant limitation in the art of speech coding, in that, a speech coding system that performs the invention is operable while providing real time operation and introducing no delay whatsoever.

Abstract: A fully backward compatible intelligent discontinued transmission (DTX) and comfort noise generation (CNG) scheme that is operable in pulse code modulation (PCM) speech coding systems. The scheme, for example, provides a speech encoder comprising a speech signal analysis circuitry configured to calculates a predetermined plurality of parameters from the speech signal, a voice activity detector configured to determine voice activity in the speech signal, where the speech encoder enters a discontinued transmission mode of the voice activity detector does not detect voice activity, and a transmitter configured to transmit one or more speech samples of the speech signal after the speech encoder enters the discontinued transmission mode, where the one or more speech samples are capable of use by a remote speech decoder to extract a parameter from the one or more speech samples in order generate a background noise base on the parameter.