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 codec 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. The overall quality of the system is strongly related to the excitation. In order to enhance the excitation, the system contains a fixed codebook comprising several subcodebooks. The invention reveals a way to apply a pitch enhancement efficiently and differently for different subcodebooks without using additional bits. The technique is particularly applicable to selectable mode vocoder (SMV) systems.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS  
       [0001]    This application claims priority to Provisional Application 60/232,938, filed Sep. 15, 2000. Other applications and patents listed below relate to and are useful in understanding various aspects of the embodiments disclosed in the present application. All are incorporated by reference in their entirety.  
         [0002]    U.S. patent application Ser. No. 09/663,242, “SELECTABLE MODE VOCODER SYSTEM,” Attorney Reference Number: 98RSS365CIP (10508.4), filed on Sep. 15, 2000, and now U.S. Pat. No. ______.  
         [0003]    U.S. Provisional Application Serial No. 60/233,043, “INJECTING HIGH FREQUENCY NOISE INTO PULSE EXCITATION FOR LOW BIT RATE CELP,” Attorney Reference Number: 00CXT0065D (10508.5).  
         [0004]    U.S. Provisional Application Serial No. 60,232,939, “SHORT TERM ENHANCEMENT IN CELP SPEECH CODING,” Attorney Reference Number: 00CXT0666N (10508.6), filed on Sep. 15, 2000.  
         [0005]    U.S. Provisional Application Serial No. 60/233,045, “SYSTEM OF DYNAMIC PULSE POSITION TRACKS FOR PULSE-LIKE EXCITATION IN SPEECH CODING,” Attorney Reference Number: 00CXT0573N (10508.7).  
         [0006]    U.S. Provisional Application Serial No. 60/232,958, “SPEECH CODING SYSTEM WITH TIME-DOMAIN NOISE ATTENUATION,” Attorney Reference Number: 00CXT0554N (10508.8), filed on Sep. 15, 2000.  
         [0007]    U.S. Provisional Application Serial No. 60/233,042, “SYSTEM FOR AN ADAPTIVE EXCITATION PATTERN FOR SPEECH CODING,” Attorney Reference Number: 98RSS366 (10508.9), filed on Sep. 15, 2000.  
         [0008]    U.S. Provisional Application Serial No. 60/233,046, “SYSTEM FOR ENCODING SPEECH INFORMATION USING AN ADAPTIVE CODEBOOK WITH DIFFERENT RESOLUTION LEVELS,” Attorney Reference Number: 00CXT0670N (10508.13), filed on Sep. 15, 2000.  
         [0009]    U.S. patent application Ser. No. 09/663,837, “CODEBOOK TABLES FOR ENCODING AND DECODING,” Attorney Reference Number: 00CXT0669N (10508.14), filed on Sep. 15, 2000, and now U.S. Pat. No. ______.  
         [0010]    U.S. patent application Ser. No. 09/662,828, “BIT STREAM PROTOCOL FOR TRANSMISSION OF ENCODED VOICE SIGNALS,” Attorney Reference Number: 00CXT0668N (10508.15), filed on Sep. 15, 2000, and now U.S. Pat. No. ______.  
         [0011]    U.S. Provisional Application Serial No. 60/233,044, “SYSTEM FOR FILTERING SPECTRAL CONTENT OF A SIGNAL FOR SPEECH ENCODING,” Attorney Reference Number: 00CXT0667N (10508.16), filed on Sep. 15, 2000.  
         [0012]    U.S. patent application Ser. No. 09/633,734, “SYSTEM FOR ENCODING AND DECODING SPEECH SIGNALS,” Attorney Reference Number: 00CXT0665N (10508.17), filed on Sep. 15, 2000, and now U.S. Pat. No. ______.  
         [0013]    U.S. patent application Ser. No. 09/663,002, “SYSTEM FOR SPEECH ENCODING HAVING AN ADAPTIVE FRAME ARRANGEMENT,” Attorney Reference Number: 98RSS384CIP (10508.18), filed on Sep. 15, 2000, and now U.S. Pat. No. ______.  
         [0014]    U.S. Provisional Application Serial No. 60/097,569 (Attorney Docket No. 98RSS325), entitled “ADAPTIVE RATE SPEECH CODEC,” filed Aug. 24, 1998.  
         [0015]    U.S. patent application Ser. No. 09/154,675 (Attorney Docket No. 97RSS383), entitled “SPEECH ENCODER USING CONTINUOUS WARPING IN LONG TERM PREPROCESSING,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0016]    U.S. patent application Ser. No. 09/156,649 (Attorney Docket No. 95EO20), entitled “COMB CODEBOOK STRUCTURE,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0017]    U.S. patent application Ser. No. 09/156,648 (Attorney Docket No. 98RSS228), entitled “LOW COMPLEXITY RANDOM CODEBOOK STRUCTURE,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0018]    U.S. patent application Ser. No. 09/156,650 (Attorney Docket No. 98RSS343), entitled “SPEECH ENCODER USING GAIN NORMALIZATION THAT COMBINES OPEN AND CLOSED LOOP GAINS,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0019]    U.S. patent application Ser. No. 09/156,832 (Attorney Docket No. 97RSS039), entitled “SPEECH ENCODER USING VOICE ACTIVITY DETECTION IN CODING NOISE,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0020]    U.S. patent application Ser. No. 09/154,654 (Attorney Docket No. 98RSS344), entitled “PITCH DETERMINATION USING SPEECH CLASSIFICATION AND PRIOR PITCH ESTIMATION,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0021]    U.S. patent application Ser. No. 09/154,657 (Attorney Docket No. 98RSS328), entitled “SPEECH ENCODER USING A CLASSIFIER FOR SMOOTHING NOISE CODING,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0022]    U.S. patent application Ser. No. 09/156,826 (Attorney Docket No. 98RSS382), entitled “ADAPTIVE TILT COMPENSATION FOR SYNTHESIZED SPEECH RESIDUAL,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0023]    U.S. patent application Ser. No. 09/154,662 (Attorney Docket No. 98RSS383), entitled “SPEECH CLASSIFICATION AND PARAMETER WEIGHTING USED IN CODEBOOK SEARCH,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0024]    U.S. patent application Ser. No. 09/154,653 (Attorney Docket No. 98RSS406), entitled “SYNCHRONIZED ENCODER-DECODER FRAME CONCEALMENT USING SPEECH CODING PARAMETERS,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0025]    U.S. patent application Ser. No. 09/154,663 (Attorney Docket No. 98RSS345), entitled “ADAPTIVE GAIN REDUCTION TO PRODUCE FIXED CODEBOOK TARGET SIGNAL,” filed Sep. 18, 1998, and now U.S. Pat. No. ______.  
         [0026]    U.S. patent application Ser. No. 09/154,660 (Attorney Docket No. 98RSS384), entitled “SPEECH ENCODER ADAPTIVELY APPLYING PITCH LONG-TERM PREDICTION AND PITCH PREPROCESSING WITH CONTINUOUS WARPING,” filed Sep. 18, 1998, and now U.S. Pat. No. ______. 
     
    
     
       BACKGROUND OF THE INVENTION  
         [0027]    1. Technical Field  
           [0028]    This invention relates to speech communication systems and, more particularly, to systems and methods for digital speech coding.  
           [0029]    2. Related Art  
           [0030]    One prevalent mode of human communication involves the use of communication systems. Communication systems include both wireline and wireless radio systems. Wireless communication systems electrically connect with the landline systems and communicate using radio frequency (RF) with mobile communication devices. Currently, the radio frequencies available for communication in cellular systems, for example, are in the frequency range centered around 900 MHz and in the personal communication services (PCS) frequency range centered around 1900 MHz. Due to increased traffic caused by the expanding popularity of wireless communication devices, such as cellular telephones, it is desirable to reduced bandwidth of transmissions within the wireless systems.  
           [0031]    Digital transmission in wireless radio communications is increasingly being applied to both voice and data due to noise immunity, reliability, compactness of equipment and the ability to implement sophisticated signal processing functions using digital techniques. Digital transmission of speech signals involves the steps of: sampling an analog speech waveform with an analog-to-digital converter, speech compression (encoding), transmission, speech decompression (decoding), digital-to-analog conversion, and playback into an earpiece or a loudspeaker. The sampling of the analog speech waveform with the analog-to-digital converter creates a digital signal. However, the number of bits used in the digital signal to represent the analog speech waveform creates a relatively large bandwidth. For example, a speech signal that is sampled at a rate of 8000 Hz (once every 0.125 ms), where each sample is represented by 16 bits, will result in a bit rate of 128,000 (16×8000) bits per second, or 128 Kbps (Kilo bits per second).  
           [0032]    Speech compression reduces the number of bits that represent the speech signal, thus reducing the bandwidth needed for transmission. However, speech compression may result in degradation of the quality of decompressed speech. In general, a higher bit rate will result in higher quality, while a lower bit rate will result in lower quality. However, speech compression techniques, such as coding techniques, can produce decompressed speech of relatively high quality at relatively low bit rates. In general, coding techniques attempt to represent the perceptually important features of the speech signal, with or without preserving the actual speech waveform.  
           [0033]    One coding technique used to lower the bit rate involves varying the degree of speech compression (i.e., varying the bit rate) depending on the part of the speech signal being compressed. Typically, parts of the speech signal for which adequate perceptual representation is more difficult or more important (such as voiced speech, plosives, or voiced onsets) are coded and transmitted using a higher number of bits, while parts of the speech signal for which adequate perceptual representation is less difficult or less important (such as unvoiced, or the silence between words) are coded with a lower number of bits. The resulting average bit rate for the speech signal may be relatively lower than would be the case for a fixed bit rate that provides decompressed speech of similar quality.  
           [0034]    These speech compression techniques have resulted in lowering the amount of bandwidth used to transmit a speech signal. However, further reduction in bandwidth is important in a communication system for a large number of users. Accordingly, there is a need for systems and methods of speech coding that are capable of minimizing the average bit rate needed for speech representation, while providing high quality decompressed speech.  
         SUMMARY  
         [0035]    A technique uses a pitch enhancement to improve the use of the fixed codebooks in cases where the fixed codebook comprises a plurality of subcodebooks. Code-excited linear prediction (CELP) coding utilizes several predictions to capture redundancy in voiced speech while minimizing data to encode the speech. A first short-term prediction results in an LPC residual, and a second long term prediction results in a pitch residual. The pitch residual may be coded using a fixed codebook that includes a plurality of fixed subcodebooks. The disclosed embodiments describe a system for pitch enhancements to improve the use of communication systems employing a plurality of fixed subcodebooks.  
           [0036]    A pitch enhancement is used in a predictable manner to add pulses to the output from the fixed subcodebooks but without requiring any additional bits to encode this additional information. The pitch lag is calculated in an adaptive codebook portion of the speech encoder/decoder. These additional pulses result in encoded speech that more closely approximates the voiced speech. In the improvement, an adaptive pitch gain and a modifying factor are used to enhance the pulses from the fixed subcodebooks differently for different subcodebooks. These techniques are used in such a manner that no extra bits of data are added to the bitstream that constitutes the output of an encoder or the input to a decoder.  
           [0037]    Accordingly, the speech coder is capable of selectively activating a series of encoders and decoders of different bitstream rates to maximize the overall quality of a reconstructed speech signal while maintaining the desired average bit rate.  
           [0038]    Other systems, methods, features and advantages of the invention will be or will become apparent to one with skill in the art upon examination of the following figures and detailed description. It is intended that all such additional systems, methods, features and advantages be included within this description, be within the scope of the invention, and be protected by the accompanying claims.  
       
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0039]    The components in the figures are not necessarily to scale, emphasis instead being placed upon illustrating the principles of the invention. Moreover, in the figures, like reference numerals designate corresponding parts throughout the different views.  
         [0040]    [0040]FIG. 1 is a graph representing time-domain speech patterns.  
         [0041]    [0041]FIG. 2 is a block diagram of a speech-coding system according to the invention.  
         [0042]    [0042]FIG. 3 is another block diagram of a speech coding system.  
         [0043]    [0043]FIG. 4 is an expanded block diagram of a speech encoding system.  
         [0044]    [0044]FIG. 5 is a block diagram of fixed codebooks.  
         [0045]    [0045]FIG. 6 is an expanded block diagram of the encoding system of FIG. 4.  
         [0046]    [0046]FIG. 7 is a flow chart for searching a fixed codebook.  
         [0047]    [0047]FIG. 8 is a flow chart for searching a fixed codebook.  
         [0048]    [0048]FIG. 9 is a schematic diagram illustrating pitch enhancements.  
         [0049]    [0049]FIG. 10 is a schematic diagram illustrating pitch enhancements.  
         [0050]    [0050]FIG. 11 is a schematic diagram illustrating pitch enhancements.  
         [0051]    [0051]FIG. 12 is a schematic diagram illustrating pitch enhancements.  
         [0052]    [0052]FIG. 13 is a schematic diagram illustrating pitch enhancements.  
         [0053]    [0053]FIG. 14 is a schematic diagram illustrating pitch enhancements.  
         [0054]    [0054]FIG. 15 is a schematic diagram illustrating pitch enhancements.  
         [0055]    [0055]FIG. 16 is a schematic diagram illustrating pitch enhancements.  
         [0056]    [0056]FIG. 17 is another expanded block diagram of the encoding system of FIG. 4.  
         [0057]    [0057]FIG. 18 is an expanded block diagram of the decoding system of FIG. 3. 
     
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS  
       [0058]    [0058]FIG. 1 depicts the waveforms in CELP speech coding. An input speech signal 2 has some measure of predictability or periodicity  4 . At least a pitch gain, a pitch lag and a fixed codebook index are calculated from the speech signal  2 . The code-excited linear prediction (CELP) coding approach uses two types of predictors, a short-term predictor and a long-term predictor. The short-term predictor is typically applied before the long-term predictor. The short-term predictor is also referred to as linear prediction coding (LPC) or spectral envelope representation, and typically may comprise ten prediction parameters.  
         [0059]    Using CELP coding, a first prediction error may be derived from the short-term predictor and is called a short-term or LPC residual  6 . The short-term LPC parameters, fixed-codebook indices and gain, as well as an adaptive codebook lag and its gain for the long-term predictor are quantized. The quantization indices, as well as the fixed codebook indices, are sent from the encoder to the decoder. The quality of the speech may be enhanced through a system that uses a plurality of fixed subcodebooks, rather than merely a single fixed subcodebook. Each lag parameter also may be called a pitch lag, and each long-term predictor gain parameter also may be called an adaptive codebook gain. The lag parameter defines an entry or a vector in the adaptive codebook.  
         [0060]    Following the LPC analysis, the long-term predictor parameters and the fixed codebook entries that best represent the prediction error of the long-term residual are determined. A second prediction error may be derived from the long-term predictor and is called a long-term or pitch residual  8 . The long-term residual may be coded using a fixed codebook that includes a plurality of fixed codebook entries or vectors. During coding, one of the entries is multiplied by a fixed codebook gain to represent the long-term residual. Analysis-by-synthesis (ABS), that is, feedback, is employed in the CELP coding. In the ABS approach, synthesizing with an inverse prediction filter and applying a perceptual weighting measure determine the best contribution from the fixed codebook and the best long-term predictor parameters.  
         [0061]    The CELP decoder uses the fixed codebook indices to extract a vector from the fixed codebook or subcodebooks. The vector is multiplied by the fixed-codebook gain to create a fixed codebook contribution. A long-term predictor contribution is added to the fixed codebook contribution to create a synthesized excitation that is referred to as an excitation. The long-term predictor contribution comprises the excitation from the past multiplied by the long-term predictor gain. The long-term predictor contribution alternatively comprises an adaptive codebook contribution or a long-term pitch-filtering characteristic. The synthesized excitation is passed through a short-term synthesis filter, which uses the short-term LPC prediction coefficients quantized by the encoder to generate synthesized speech. The synthesized speech may be passed through a post-filter that reduces the perceptual coding noise. Other codecs and associated coding algorithms may be used, such as a selectable mode locoer (SUM) system, extended code excited linear prediction (eX-CELP), and algebraic CELP (A-CELP).  
         [0062]    [0062]FIG. 2 is a block diagram of a speech coding system  100  with according to one embodiment that uses CELP coding. The speech coding system  100  includes a first communication device  105  operatively connected via a communication medium  110  to a second communication device  115 . The speech coding system  100  may be any cellular telephone, radio frequency, or other communication system capable of encoding a speech signal  145  and decoding the encoded signal to create synthesized speech  150 . The communications devices  105  and  115  may be cellular telephones, portable radio transceivers, and the like.  
         [0063]    The communications medium  110  may include systems using any transmission mechanism, including radio waves, infrared, landlines, fiber optics, any other medium capable of transmitting digital signals (wires or cables), or any combination thereof. The communications medium  110  may also include a storage mechanism including a memory device, a storage medium, or other device capable of storing and retrieving digital signals. In use, the communications medium  110  transmits a bitstream of digital between the first and second communications devices  105  and  115 .  
         [0064]    The first communication device  105  includes an analog-to-digital converter  120 , a preprocessor  125 , and an encoder  130  connected as shown. The first communication device  105  may have an antenna or other communication medium interface (not shown) for sending and receiving digital signals with the communication medium  110 . The first communication device  105  may also have other components known in the art for any communication device, such as a decoder or a digital-to-analog converter.  
         [0065]    The second communication device  115  includes a decoder  135  and digital-to-analog converter  140  connected as shown. Although not shown, the second communication device  115  may have one or more of a synthesis filter, a postprocessor, and other components. The second communication device  115  also may have an antenna or other communication medium interface (not shown) for sending and receiving digital signals with the communication medium. The preprocessor  125 , encoder  130 , and decoder  135  comprise processors, digital signal processors (DSP), application specific integrated circuits, or other digital devices for implementing the coding and algorithms discussed herein. The preprocessor  125  and encoder  130  may comprise separate components or the same component  
         [0066]    In use, the analog-to-digital converter 120 receives a speech signal  145  from a microphone (not shown) or other signal input device. The speech signal may be voiced speech, music, or another analog signal. The analog-to-digital converter  120  digitizes the speech signal, providing the digitized speech signal to the preprocessor  125 . The preprocessor  125  passes the digitized signal through a high-pass filter (not shown) preferably with a cutoff frequency of about 60-80 Hz. The preprocessor  125  may perform other processes to improve the digitized signal for encoding, such as noise suppression. The encoder  130  codes the speech using a pitch lag, a pitch gain, a fixed codebook, a fixed codebook gain, LPC parameters and other parameters. The code is transmitted in the communication medium  110 .  
         [0067]    The decoder  135  receives the bitstream from the communication medium  110 . The decoder operates to decode the bitstream and generate a synthesized speech signal  150  in the form of a digitized signal. The synthesized speech signal  150  has been converted to an analog signal by the digital-to-analog converter  140 . The encoder  130  and the decoder  135  use a speech compression system, commonly called a codec, to reduce the bit rate of the noise-suppressed digitized speech signal. For example, the code excited linear prediction (CELP) coding technique utilizes several prediction techniques to remove redundancy from the speech signal.  
         [0068]    The CELP coding approach is frame-based. Samples of input speech signals (e.g., preprocessed, digitized speech signals) are stored in blocks of samples called frames. To minimize bandwidth use, each frame may be characterized. The frames are processed to create a compressed speech signal in digitized form. The frame characterization is based on the portion of the speech signal  145  contained in the particular frame. For example, frames may be characterized as stationary voiced speech, non-stationary voiced speech, unvoiced speech, onset, background noise, and silence. As will be seen, these classifications may be used to help determine the resources used to encode and decode each particular frame.  
         [0069]    [0069]FIG. 3 shows an embodiment of a speech coding system  10  that may utilize adaptive and fixed codebooks, and in particular, may utilize fixed codebooks that comprise a plurality of fixed subcodebooks for encoding at different rates as a function of the characterization. The encoding system  12  receives a speech signal  18  from a signal input device such as a microphone (not shown). The speech coding system  10  includes four codecs, a full-rate codec  22 , a half-rate codec  24 , a quarter-rate codec  26  and an eighth-rate codec  28 . There may be more or fewer codecs. Each codec has an encoder portion and a decoder portion located within the encoding and decoding systems  12  and  16  respectively. Each codec  22 ,  24 ,  26 , and  28  may process a portion of the bitstream between the encoding system  12  and the decoding system  16 . Desirably, the decoded speech is also post-processed by modules shown in later figures. The post-processed speech may be received by a human ear or by a recording device, or other device capable of receiving or using such a signal. Each codec generates a bitstream of a different bandwidth. In one embodiment, the full rate codec generates about 170 bits, the half-rate codec generates about 80 bits, the quarter-rate about 40 bits, and the eighth-rate about 16 bits respectively, per frame.  
         [0070]    The speech processing circuitry is constantly changing the codec used to code and decode speech. By processing the frames of the speech signal  18  with the various codecs, an average bit rate is achieved. The average bit rate of the bitstream may be calculated as an average of the codecs used in any particular interval of time. A mode-line  21  carries a mode-input signal from a communications system. The mode-input signal controls the average rate of the encoding system  12 , dictating which of a plurality of codecs is used within the encoding system  12 .  
         [0071]    In one embodiment of the speech compression system  10 , the full- and half-rate codecs use an eX-CELP (extended CELP) algorithm. The eX-CELP algorithm categorizes frames into different categories using a rate selection and a type classification. The quarter- and eighth-rate codecs are based on a perceptual matching algorithm. Different encoding approaches may be used for different categories of frames with different perceptual matching, different waveform matching, and different bit assignments. In this embodiment, the perceptual matching algorithms of the quarter-rate and eighth-rate codecs do not use waveform matching.  
         [0072]    The frames may be divided into a plurality of subframes. The subframes may be different in size and number for each codec. With respect to the eX-CELP algorithm, the subframes may be different in size for each classification. The CELP approach is used in eX-CELP to choose the adaptive codebook, the fixed codebook, and other parameters used to code the speech. The ABS scheme uses inverse prediction filters and perceptual weighting measures for selecting the codebook entries.  
         [0073]    [0073]FIG. 4 is an expanded block diagram of the encoding system  12  shown in FIG. 3. One embodiment of the encoding system  12  includes a preprocessing module  34 , a full-rate encoder  36 , a half-rate encoder  38 , a quarter-rate encoder  40 , and an eighth-rate encoder  42 , connected as illustrated. The pre-processing module  34  may be used to process speech on a frame basis to provide filtering, signal enhancement, noise enhancement, and amplification to optimize the signal for subsequent processing.  
         [0074]    The rate encoders include an initial frame-processing module  44  and an excitation-processing module  54 . The initial frame-processing module  44  is divided into a plurality of initial frame processing modules, namely, modules for the full-rate  46 , half-rate  48 , quarter-rate  50 , and an initial eighth-rate frame processing module  52 .  
         [0075]    The full, half, quarter and eighth-rate encoders  36 ,  38 ,  40 , and  42  comprise the encoding portion of the respective codecs  22 ,  24 ,  26 , and  28 . The initial frame-processing module  44  performs initial frame processing, extracts speech parameters, and determines which rate encoder will encode a particular frame. Module  44  determines a rate selection that activates one of the encoders  36 ,  38 ,  40 , or  42 . The rate selection may be based on the categorization of the frame of the speech signal  18  and the mode of the speech compression system. Activation of one of the rate encoders  36 ,  38 ,  40 , or  42 , correspondingly activates one of the initial frame-processing modules  46 ,  48 ,  50 , or  52 .  
         [0076]    In addition to the rate selection, the initial frame-processing module  44  also determines a type classification for each frame that is processed by the full and half rate encoders  36  and  38 . In one embodiment, the speech signal  18  as represented by one frame is classified as “type 0” or “type 1,” depending on the nature and characteristics of the speech signal  18 . In an alternative embodiment, additional classifications and supporting processing are provided.  
         [0077]    Type 1 classification includes frames of the speech signal  18  having harmonic and formant structures that do not change rapidly. Type 0 classification includes all other frames. The type classification optimizes encoding by the initial full-rate frame-processing module  46  and the initial half-rate frame-processing module  48 . In addition, the classification type and rate selection are used to optimize the encoding by the excitation-processing module  54  for the full and half-rate encoders  36  and  38 .  
         [0078]    In one embodiment, the excitation-processing module  54  is sub-divided into a full-rate module  56 , a half-rate module  58 , a quarter-rate module  60 , and an eighth-rate module  62 . The rate modules  56 ,  58 ,  60 , and  62  correspond to the rate encoders  36 ,  38 ,  40 , and  42 . The full and half rate modules  56  and  58  in one embodiment both include a plurality of frame processing modules and a plurality of subframe processing modules, but provide substantially different encoding. The term “F” indicates full rate processing, “H” indicates half-rate processing, and “0” and “1” indicate type 0 and type 1, respectively.  
         [0079]    The initial frame-processing module  44  includes modules for full-rate frame processing  46  and half-rate frame processing  48 . These modules may calculate an open loop pitch  144   a  for a full-rate frame, or an open loop pitch  176   a  for a half-rate frame. These components may be used later.  
         [0080]    The full rate module  56  includes an F type selector module  68 , and an F 0  subframe-processing module  70 . Module  56  also includes modules for F 1  processing, including an F 1  first frame processing module  72 , an F 1  subframe processing module  74 , and an F 1  second frame-processing module  76 . In a similar manner, the half rate module  58  includes an H type selector module  78 , an H 0  sub-frame processing module  80 , an H 1  first frame processing module  82 , an H 1  sub-frame processing module  84 , and an H 1  second frame-processing module  86 .  
         [0081]    The selector modules  68  and  78  direct the processing of the speech signals  18  to further optimize the encoding process based on the type classification. When the frame being processed is classified as full rate, selector module  68  directs the speech signal to either the F 0  or F 1  processing to encode the speech and generate the bitstream. Type 0 classification for a frame activates the processing module to process the frame on a subframe basis. Type 1 processing proceeds on both a frame and subframe basis. In type 0 processing, a fixed codebook component  146   a  and a closed loop adaptive codebook component  144   b  are generated and are used to generate fixed and adaptive codebook gains  148   a  and  150   a . In type 1 processing, an adaptive gain  148   b  is derived from the first frame-processing module  72 , and a fixed codebook  146   b  is selected and used to encode the speech with the subframe-processing module  74 . A fixed codebook gain  150   b  is derived from the second frame-processing module  76 . Type signal  142  designates the type as either F 0  or F 1  in the bitstream.  
         [0082]    If the frame of the speech signal is classified as half-rate, selector module  78  directs the frame to either H 0  (type 0) or H 1  (type 1) processing. The same classifications are made with respect to type 0 or type 1 processing. In type 0 processing, H 0  subframe processing module  80  generates a fixed codebook component  178   a  and a closed loop adaptive codebook component  176   b , used to generate fixed and adaptive codebook gains  180   a  and  182   a . In type 1 processing, an H 1  first frame processing module  82 , an H 1  subframe processing module  84  and an H 1  second frame processing module  86  are used. An adaptive gain  180   b , a fixed codebook component  178   b , and a fixed codebook gain are calculated. Type signal  174  designates the type as either H 0  or H 1  in the bitstream.  
         [0083]    In a manner known to those skilled in the art, adaptive codebooks are then used to code the signal in the full rate and half rate codecs. An adaptive codebook search and selection for the full rate codec uses components  144   a  and  144   b . These components are used to search, test, select and designate the location of a pitch lag from an adaptive codebook. In a similar manner, half-rate components  176   a  and  176   b  search, test, select and designate the location of the best pitch lag for the half-rate codec. These pitch lags are subsequently used to improve the quality of the encoded and decoded speech through fixed codebooks employing a plurality of fixed subcodebooks.  
         [0084]    [0084]FIG. 5 is a block diagram depicting the structure of fixed codebooks and subcodebooks in one embodiment. The fixed codebook  160  for the F 0  codec comprises three (different) subcodebooks, each of them having 5 pulses. The fixed codebook for the F 1  codec is a single 8-pulse subcodebook  162 . For the half-rate codec, the fixed codebook  178  comprises three subcodebooks for the H 0 , a 2-pulse subcodebook  192 , a three-pulse subcodebook  194 , and a third subcodebook  196  with gaussian noise. In the H 1  codec, the fixed codebook comprises a 2-pulse subcodebook  193 , a 3-pulse subcodebook  195 , and a 5-pulse subcodebook  197 .  
         [0085]    Fixed Codebook Encoding for Type 0 Frames  
         [0086]    [0086]FIG. 6 comprises F 0  and H 0  subframe processing modules  70  and  80 , including an adaptive codebook section  362 , a fixed codebook section  364 , and a gain quantization section  366 . The adaptive codebook section  368  receives a pitch track  348  to calculate an area in the adaptive codebook to search for an adaptive codebook vector (v a )  382  (a pitch lag). The adaptive codebook section  368  also performs a search to determine and store the best lag vector v a  for each subframe. An adaptive gain, g a    384 .  
         [0087]    [0087]FIG. 6 depicts the fixed codebook section  364 , including a fixed codebook  390 , a multiplier  392 , a synthesis filter  394 , a perceptual weighting filter  396 , a subtractor  398 , and a minimization module  400 . The gain quantization section  366  may include a 2D VQ gain codebook  412 , a first multiplier  414 , a second multiplier  416 , an adder  418 , a synthesis filter  420 , a perceptual weighting filter  422 , a subtractor  424  and a minimization module  426 . The gain quantization section  366  makes use of the second resynthesized speech  406  generated in the fixed codebook section, and also generates a third resynthesized speech  438 .  
         [0088]    The fixed codebook  390  fixed codebook vector (v c )  402  representing the long-term residual for a subframe. The multiplier  392  multiplies the fixed codebook vector (v c )  402  by a gain (g c )  404 . The gain (g c )  404  is unquantized and is a representation of the initial value of the fixed codebook gain. The resulting signal is provided to the synthesis filter  394 . The synthesis filter  394  receives the quantized LPC coefficients A q (z)  342  and together with the perceptual weighting filter  396 , creates a resynthesized speech signal  406 . The subtractor  398  subtracts the resynthesized speech signal  406  from the long-term error signal  388  to generate the weighted mean square error (WMSE), a fixed codebook error signal  408 .  
         [0089]    The minimization module  400  receives the fixed codebook error signal  408 . The minimization module  400  uses the fixed codebook error signal  408  to control the selection of vectors for the fixed codebook vector (v c )  402  from the fixed codebook  292  in order to reduce the error. The minimization module  400  also receives the control information  356  that may include a final characterization for each frame.  
         [0090]    The final characterization class contained in the control information  356  controls how the minimization module  400  selects vectors for the fixed codebook vector (v c )  402  from the fixed codebook  390 . The process repeats until the search by the second minimization module  400  has selected the best vector for the fixed codebook vector (v c )  402  from the fixed codebook  390  for each subframe. The best vector for the fixed codebook vector (v c )  402  minimizes the error in the second resynthesized speech signal  406 . The indices identify the best vector for the fixed codebook vector (v c )  402  and, as previously discussed, may be used to form the fixed codebook components  146   a  and  178   a.    
         [0091]    Weighting Factors in Selecting a Fixed Subcodebook and a Codevector  
         [0092]    Low-bit rate coding uses the important concept of perceptual weighting to determine speech coding. We introduce here a special weighting factor different from the factor previously described for the perceptual weighting filter in the closed-loop analysis. This special weighting factor is generated by employing certain features of speech, and applied as a criterion value in favoring a specific subcodebook in a codebook featuring a plurality of subcodebooks. One subcodebook may be preferred over the other subcodebooks for some specific speech signal, such as noise-like unvoiced speech. The features used to estimate the weighting factor include, but are not limited to, the noise-to-signal ratio (NSR), sharpness of the speech, the pitch lag, the pitch correlation, as well as other features. The classification system for each frame of speech is also important in defining the features of the speech.  
         [0093]    The NSR is a traditional distortion criterion that may be calculated as the ratio between an estimate of the background noise energy and the frame energy of a frame. One embodiment of the NSR calculation ensures that only true background noise is included in the ratio by using a modified voice activity decision. In addition, previously calculated parameters representing, for example, the spectrum expressed by the reflection coefficients, the pitch correlation R p , the NSR, the energy of the frame, the energy of the previous frames, the residual sharpness and the sharpness may also be used. Sharpness is defined as the ratio of the average of the absolute values of the samples to the maximum of the absolute values of the samples of speech. It is typically applied to the amplitude of the signals.  
         [0094]    Pitch Correlation  
         [0095]    One embodiment of the target signal for time warping is a synthesis of the current segment derived from the modified weighted speech that is represented by s′ w (n) and the pitch track  348  represented by L p (n). According to the pitch track  348 , L p (n), each sample value of the target signal s t   w (n), n=0, . . . , N s −1 may be obtained by interpolation of the modified weighted speech using a 21 st  order Hamming weighted Sinc window,  
                   s   w   t          (   n   )       =       ∑     i   =     -   10       10              w   s          (       f        (       L   p          (   n   )       )       ,   i     )       ·       s   w   t          (     n   -     I        (       L   p          (   n   )       )       +   i     )             ,     
            for                 n     =   0     ,   …              ,       N   s     -   1             (     Equation                 1     )                               
 
         [0096]    where I(L p (n)) and f(L p (n)) are the integer and fractional parts of the pitch lag, respectively; w s (f, i) is the Hamming weighted Sinc window, and N, is the length of the segment. A weighted target, s w   wt (n), is given by s w   w (n)=w e (n)·s t   w (n). The weighting function, w e (n), may be a two-piece linear function, which emphasizes the pitch complex and de-emphasizes the “noise” in between pitch complexes. The weighting may be adapted according to a classification, by increasing the emphasis on the pitch complex for segments of higher periodicity.  
         [0097]    Signal Warping  
         [0098]    The modified weighted speech for the segment may be reconstructed according to the mapping given by  
         [ S   w ( n+τ   acc ),  S   w ( n+τ   acc +τ c +τ opt )]→[ s   t   w ( n+τ   c −1)],   (Equation 2) 
         [0099]    and  
         [ S   w ( n+τ   acc +τ c +τ opt ),  S   w ( n+τ   acc +τ opt   +N   S −1)]→[ s   t   w ( n+τ   c ),  s   t   w ( n+N   s −1)],   (Equation 3) 
         [0100]    where τ c  is a parameter defining the warping function. In general, τ c  specifies the beginning of the pitch complex. The mapping given by Equation 2 specifies a time warping, and the mapping given by Equation 3 specifies a time shift (no warping). Both may be carried out using a Hamming weighted Sinc window function.  
         [0101]    Pitch Gain and Pitch Correlation Estimation  
         [0102]    The pitch gain and pitch correlation may be estimated on a pitch cycle basis and are defined by Equations 2 and 3, respectively. The pitch gain is estimated in order to minimize the mean squared error between the target s t   w (n), defined by Equation 1, and the final modified signal s t   w (n), defined by Equations 2 and 3, and may be given by  
               g   a     =           ∑     n   =   0         N   s     -   1                s   w   ′          (   n   )       ·       s   w   t          (   n   )               ∑     n   =   0         N   s     -   1                s   w   t          (   n   )       2         .             (     Equation                 4     )                               
 
         [0103]    The pitch gain is provided to the excitation-processing module  54  as the unquantized pitch gains. The pitch correlation may be given by  
               R   a     =           ∑     n   =   0         N   s     -   1                s   w   ′          (   n   )       ·       s   w   t          (   n   )                 (       ∑     n   =   0         N   s     -   1                s   w   ′          (   n   )       2       )     ·     (       ∑     n   =   0         N   s     -   1                s   w   t          (   n   )       2       )           .             (     Equation                 5     )                               
 
         [0104]    Both parameters are available on a pitch cycle basis and may be linearly interpolated.  
         [0105]    Type 0 Fixed Codebook Search for the Full-Rate Codec  
         [0106]    The fixed codebook component  146   a  for frames of Type 0 classification may represent each of four subframes of the full-rate codec  22  using the three different 5-pulse subcodebooks  160 . When the search is initiated, vectors for the fixed codebook vector (v c )  402  within the fixed codebook  390  may be determined using the error signal  388 , represented by:  
                 t   ′          (   n   )       =       t        (   n   )       -       g   a     ·       (       e        (     n   -     L   p   opt       )       *     h        (   n   )         )     .                 (     Equation                 6     )                               
 
         [0107]    where t′(n) is a target for a fixed codebook search, t(n) is an original target signal, g a  is an adaptive gain, e(n) is a post excitation to generate an adaptive codebook contribution, L p   opt  is an optimized lag, and h(n) is an impulse response of a perceptually-weighted LPC synthesis filter.  
         [0108]    Pitch enhancement may be applied to the 5-pulse codebooks  160  within the fixed codebook  390  in the forward direction or the backward direction during the search. The search is an iterative, controlled complexity search for the best vector from the fixed codebook  160 . An initial value for the fixed codebook gain represented by the gain (g c )  404  may be found simultaneously with the search.  
         [0109]    [0109]FIGS. 7 and 8 illustrate the procedure used to search for the best indices in the fixed codebook. In one embodiment, a fixed codebook has k subcodebooks. More or fewer subcodebooks may be used in other embodiments. In order to simplify the description of the iterative search procedure, the following example first features a single subcodebook containing N pulses. The possible location of a pulse is defined by a plurality of positions on a track. In a first searching turn, the encoder processing circuitry searches the pulse positions sequentially from the first pulse  633  (P N=1 ) to the next pulse  635 , until the last pulse  637  (P N =N). For each pulse after the first, the searching of the current pulse position is conducted by considering the influence from previously-located pulses. The influence is the desirable minimizing of the energy of the fixed subcodebook error signal  408 . In a second searching turn, the encoder processing circuitry corrects each pulse position sequentially, again from the first pulse  639  to the last pulse  641 , by considering the influence of all the other pulses. In subsequent turns, the functionality of the second or subsequent searching turn is repeated, until the last turn is reached  643 . Further turns may be utilized if the added complexity is allowed. This procedure is followed until k turns are completed  645  and a value is calculated for the subcodebook.  
         [0110]    [0110]FIG. 8 is a flow chart for the method described in FIG. 7 to be used for searching a fixed codebook comprising a plurality of subcodebooks. A first turn is begun  651  by searching a first subcodebook  653 , and searching the other subcodebooks  655 , in the same manner described for FIG. 7, and keeping the best result  657 , until the last subcodebook is searched  659 . If desired, a second turn  661  or subsequent turn  663  may also be used, in an iterative fashion. In some embodiments, to minimize complexity and shorten the search, one of the subcodebooks in the fixed codebook is typically chosen after finishing the first searching turn. Further searching turns are done only with the chosen subcodebook. In other embodiments, one of the subcodebooks might be chosen only after the second searching turn or thereafter, should processing resources so permit. Computations of minimum complexity are desirable, especially since two or three times as many pulses are calculated, rather than one pulse before enhancements described herein are added.  
         [0111]    In an example embodiment, the search for the best vector for the fixed codebook vector (v c )  402  is completed in each of the three 5-pulse codebooks  160 . At the conclusion of the search process within each of the three 5-pulse codebooks  160 , candidate best vectors for the fixed codebook vector (v c )  402  have been identified. Selection of which of the candidate best vectors from which of the 5-pulse codebooks  160  will be used may be determined minimizing the corresponding fixed codebook error signal  408  for each of the three best vectors. For purposes of this discussion, the corresponding fixed codebook residual error  408  for each of the three candidate subcodebooks will be referred to as first, second, and third fixed codebook error signals.  
         [0112]    The minimization of the weighted mean square errors (WMSE) from the first, second and third fixed codebook error signals is mathematically equivalent to maximizing a criterion value which may be first modified by multiplying a weighting factor in order to favor selecting one specific subcodebook. Within the full-rate codec  22  for frames classified as Type Zero, the criterion value from the first, second and third fixed codebook error signals may be weighted by the subframe-based weighting measures. The weighting factor may be estimated by a using a sharpness measure of the residual signal, a voice-activity detection module, a noise-to-signal ratio (NSR), and a normalized pitch correlation. Other embodiments may use other weighting factor measures. Based on the weighting and on the maximal criterion value, one of the three 5-pulse fixed codebooks  160 , and the best candidate vector in that subcodebook, may be selected.  
         [0113]    The selected 5-pulse codebook  161 ,  163  or  165  may then be fine searched for a final decision of the best vector for the fixed codebook vector (v c )  402 . The fine search is performed on the vectors in the selected 5-pulse codebook  160  that are in the vicinity of the best candidate vector chosen. The indices that identify the best vector (maximal criterion value) from the fixed codebook vector are in the bitstream to be transmitted to the decoder.  
         [0114]    Encoding the pitch lag generates an adaptive codebook vector  382  (lag) and an adaptive codebook gain g a    384 , for each subframe of type 1 processing. The lag is incorporated into the fixed codebook in one embodiment, by using the pitch enhancement differently for different subcodebooks, to increase excitation density. The use of the pitch enhancement should be incorporated during the searches in the encoder and the same pitch enhancement should be applied to the codevector from the fixed codebook in the decoder. For every vector found in the fixed codebook, the density of the codevector may be increased by convoluting with an impulsive response of pitch enhancement. This impulsive response always has a unit pulse at time  0  and includes an addition pulse at +1 pitch lag, −1 pitch lag, +2 pitch lags, −2 pitch lags, and so on. The magnitudes of these additional pitch pulses are determined by a pitch enhancement coefficient, which may be different for different subcodebooks. For type 0 processing, the pitch enhancement coefficient is calculated according the pitch gain, g a—m  from the previous subframe of the adaptive codebook section, multiplied by a factor that depends on the fixed subcodebook.  
         [0115]    Examples of typical pitch enhancement coefficients are listed in Table 1. This table is typically used for the half-rate codec, although it could also be employed for the full-rate. The benefit from a more flexible pitch enhancement for the full-rate codec is less significant, because the full rate excitation from a large fixed codebook with a short subframe size is already very rich. The coefficients for Type 1 will be explained below.  
                                           TABLE 1                           Pitch Enhancement Coefficients                Type 0   Type 1                        Subcodebook #1   0.5 ≦ 0.75 · g a     —     m  1.0   0.5 ≦ 0.75 · g a  1.0       Subcodebook #2   0.0 ≦ 0.25 · g a     —     m  0.5   0.0 ≦ 0.50 · g a  0.5       Subcodebook #3   0   0.0 ≦ 0.50 · g a  0.5                  
 
         [0116]    In one embodiment for F 0  processing, the pitch enhancement coefficient for the whole fixed codebook could be the previous pitch gain g a—m  multiplied by a factor of 0.75. The result may be limited to a value between 0.0 and 1.0. The above Table may also be used to determine the pitch enhancement coefficients for different subcodebooks. The pitch enhancement coefficient for the first subcodebook may be the pitch gain of the previous subframe, g a—m , multiplied by 0.75. The result may be limited to values between 0.5 and 1.0. Similarly, for F 0  processing with a second subcodebook, the pitch enhancement coefficients could be limited to values between 0.0≦0.25·g a—m ≦0.5; the pitch enhancement coefficient could be zero for the third subcodebook.  
         [0117]    In the example of FIG. 9, speech is processed in frames of 160 samples with four subframes of 40 samples for F 0 . A pitch lag of 16 samples may be calculated and forwarded by an adaptive codebook contribution. The use of 16 samples is merely a convenience, and pitch lags are usually larger than 16. A fixed codebook in the same speech coder/decoder may be searched and a close match of one of the pulses from the fixed codebook found at sample  6 . In this example, the fixed codebook generates a pulse at sample  6  and the pitch enhancement generates additional pulses at sample  22  and at sample  38 . Because the pitch enhancement coefficient has been calculated according to available information, no additional bits need to be transmitted to capture the extra pulse density.  
         [0118]    [0118]FIG. 9 illustrates a single pulse  902  at about location  6  (samples) generated by a fixed codebook. In one embodiment, shown in FIG. 10, a pitch enhancement adds pulses  904  and  906  additional to the original pulse  902  from the fixed codebook. The additional pulses correspond to at intervals  910  of 16 samples, as shown in FIG. 11. This illustrates a pitch enhancement applied in a “forward” direction.  
         [0119]    In another embodiment, the pitch enhancement may be applied in a “backward” direction. FIG. 12 illustrates a pulse  912  from a fixed codebook at 24 (samples). Using the previous example of a pitch lag of 16 samples, a pulse  916  is added in a forward direction at 40 (samples), as seen in FIG. 13. A pulse  914  is added in a backward direction at 8 (samples), calculated by subtracting 16 from 24. It has been found that speech coded with these enhancements sounds more natural and more similar to an original spoken voice. The fixed codebook pulses in this embodiment are processed as described and shown in the previous examples. In this example, a pitch enhancement coefficient is applied to the pitch pulses that are +1 or −1 pitch lag away from the main pulse.  
         [0120]    Type 0 Fixed Codebook Search for the Half-Rate Codec  
         [0121]    The fixed codebook component  178   a  for frames of Type 0 classification represents the fixed codebook contribution for each of the two subframes of the half-rate codec  24 . The representation may be based on the pulse codebooks  192  and  194  and the gaussian subcodebook  196 . The initial target for the fixed codebook gain represented by the gain (g c )  404  may be determined similarly to the full-rate codec  22 . In addition, during the search for the fixed codebook vector (v c )  402  within the fixed codebook  390 , the criterion value may be weighted similarly to the full-rate codec  22 , from a perceptual point of view. In the half-rate codec  24 , the weighting may be applied to favor selecting the best vector from the gaussian subcodebook  196  when the input reference signal is noise-like. The weighting helps determine the most suitable fixed subcodebook vector (v c )  402 .  
         [0122]    The pitch enhancement discussed in the F 0  processing applies also to the half rate H 0 , which in one embodiment is processed in subframes of 80 samples. The pitch lags are derived in the same manner from the adaptive codebook, as is the pitch gain, g a    384 . In H 0  processing, as in F 0  processing, a pitch gain from the previous subframe, g a—m , is used. In one embodiment, the pitch enhancement coefficient for the first subcodebook  192  is estimate by multiplying the pitch gain of the previous subframe by a factor of 0.75, where resulting 0.75·g a—m  is limited to values between 0.5 and 1.0. Similarly, for H 0  processing with a second subcodebook, the pitch enhancement coefficient is multiplied by 0.25, with the resulting 0.25·g a—m  is limited to values between 0.0 and 0.25.  
         [0123]    An example is depicted in FIGS.  14 - 16 . For the H 0  codec, 2-subframe processing is used, and in this example, an initial pulse from a subcodebook for the H 0  codec is at about  44 . This is shown in FIG. 14 as  922 . Additional pulses introduced by the pitch enhancement are located at ±1 and ±2 pitch lags away from the initial pulse, or in this example, at  12 ,  28 ,  60  and  76 , for a pitch lag of 16. This is depicted in FIG.  15 , with pulses at ±1 pitch lag at  28  and  60 ,  926  and  928  respectively, and ±2 pitch lags, at  12  and  76 ,  924  and  930  respectively. FIG. 16 depicts a pitch enhancement coefficient of 0.5 applied once to the pulses  936  and  938 . The coefficient is applied twice (0.5 to the second power, or  0 . 25 ) to the pulses  934  and  940 .  
         [0124]    The search for the best vector for the fixed codebook vector (v c )  402  is based on minimizing the energy of the fixed codebook error signal  408  as previously discussed. The search may first be performed on the 2-pulse subcodebook  192 . The 3-pulse codebook  194  may be searched next, in several steps. The current step may determine a starting point for the next step. Backward and forward pitch enhancement may be applied during the search and after the search in both pulse subcodebooks  192  and  194 . The gaussian subcodebook  196  may be searched last, using a fast search routine based on two orthogonal basis vectors.  
         [0125]    The selection of one of the subcodebooks  192 ,  194  or  196  and the best vector (v c )  402  from the selected subcodebook may be performed in a manner similar to that used for the full-rate codec  22 . The indices that identify the best fixed codebook vector (v c )  402  within the selected subcodebook are the fixed codebook component  178   a  in the bitstream. The unquantized initial values of the gains (g a )  384  and (g c )  404  may now be finalized based on the vectors for the adaptive codebook vector (v a )  382  (lag) and the fixed codebook vector (v c )  402  previously determined. They are jointly quantized within the gain quantization section  366 . Determination and quantization of the gains occurs within the gain quantization section  366 .  
         [0126]    Fixed Codebook Encoding for Type 1 Frames  
         [0127]    Referring now to FIG. 17, the F 1  and H 1  first frame processing modules  72  and  82  include a 3D/4D open loop VQ module  454 . The F 1  and H 1  sub-frame processing modules  74  and  84  include the adaptive codebook  368 , the fixed codebook  390 , a first multiplier  456 , a second multiplier  458 , a first synthesis filter  460  and a second synthesis filter  462 . In addition, the F 1  and H 1  sub-frame processing modules  74  and  84  include a first perceptual weighting filter  464 , a second perceptual weighting filter  466 , a first subtractor  468 , a second subtractor  470 , a first minimization module  472  and an energy adjustment module  474 . The F 1  and H 1  second frame processing modules  76  and  86  include a third multiplier  476 , a fourth multiplier  478 , an adder  480 , a third synthesis filter  482 , a third perceptual weighting filter  484 , a third subtractor  486 , a buffering module  488 , a second minimization module  490  and a 3D/4D VQ gain codebook  492 .  
         [0128]    The processing of frames classified as Type 1 within the excitation-processing module  54  provides processing on both a frame basis and a sub-frame basis. For purposes of brevity, the following discussion refers to the modules within the full rate codec  22 . The modules in the half rate codec  24  function similarly unless otherwise noted. Quantization of the adaptive codebook gain by the F 1  first frame-processing module  72  generates the adaptive gain component  148   b . The F 1  subframe processing module  74  and the F 1  second frame processing module  76  operate to determine the fixed codebook vector and the corresponding fixed codebook gain, respectively as previously set forth. The F 1  subframe-processing module  74  uses the track tables to generate the fixed codebook component  146   b  as illustrated in FIG. 4.  
         [0129]    The F 1  second frame processing module  76  quantizes the fixed codebook gain to generate the fixed gain component  150   b . In one embodiment, the full-rate codec  22  uses 10 bits for the quantization of 4 fixed codebook gains, and the half-rate codec  24  uses 8 bits for the quantization of the 3 fixed codebook gains. The quantization may be performed using moving average prediction.  
         [0130]    First Frame Processing Module  
         [0131]    In FIG. 12, the 3D/4D open loop VQ module  454  receives the unquantized pitch gains  352  from a pitch pre-processing module (not shown). The 3D/4D open loop VQ module  454  quantizes the unquantized pitch gains  352  to generate a quantized pitch gain (g k   a )  496  representing quantized pitch gains for each subframe where k is the number of subframes. In one embodiment, there are four subframes for the full-rate codec  22  and three subframes for the half-rate codec  24  which correspond to four quantized gains (g 1   a , g 2   a , g 3   a , and g 4   a ) and three quantized gains (g 1   a , g 2   a , and g 3   a ) of each subframe, respectively. The index location of the quantized pitch gain (g k   a )  496  within the pre-gain quantization table represents the adaptive gain component  148   b  for the full-rate codec  22  or the adaptive gain component  180   b  for the half-rate codec  24 . The quantized pitch gain (g k   a )  496  is provided to the F 1  subframe-processing module  74  or the H 1  second subframe-processing module  84 .  
         [0132]    In one embodiment, for a first subcodebook and for type 1 processing, the quantized pitch gain for the subframe is multiplied by 0.75, and the resulting pitch enhancement coefficient is constrained to lie between 0.5 and 1.0, inclusive. In another embodiment, for a second or a third subcodebook, the quantized pitch gain may be multiplied by 0.5, and the resulting pitch enhancement factor constrained to lie between 0 and 0.5, inclusive. While this technique may be used for both the full rate and half-rate type 1 codecs, a greater advantage will inure to the use in the half-rate codec.  
         [0133]    Sub-Frame Processing Module  
         [0134]    The F 1  or H 1  subframe-processing module  74  or  84  uses the pitch track  348  to identify an adaptive codebook vector (v k   a )  498 , representing the adaptive codebook contribution for each subframe, where k =the subframe number. In one embodiment, there are four subframes for the full-rate codec  22  and three subframes for the half-rate codec  24  which correspond to four vectors (v 1   a , v 2   a , v 3   a , and v 4   a ) and three vectors (v 1   a , v 2   a , and v 3   a ) for the adaptive codebook contribution for each subframe, respectively.  
         [0135]    The adaptive codebook vector (v k   a )  498  selected and the quantized pitch gain (g k   a )  496  are multiplied by the first multiplier  456 . The first multiplier  456  generates a signal that is processed by the first synthesis filter  460  and the first perceptual weighting filter module  464  to provide a first resynthesized speech signal  500 . The first synthesis filter  460  receives the quantized LPC coefficients A q (z)  342  from an LSF quantization module (not shown) as part of the processing. The first subtractor  468  subtracts the first resynthesized speech signal  500  from the modified weighted speech  350  provided by a pitch pre-processing module (not shown) to generate a long-term residual signal  502 .  
         [0136]    The F 1  or H 1  subframe-processing module  74  or  84  also performs a search for the fixed codebook contribution that is similar to that performed by the F 0  and H 0  subframe-processing modules  70  and  80 . Vectors for a fixed codebook vector (v k   c )  504  that represents the long-term residual for a subframe are selected from the fixed codebook  390 . The second multiplier  458  multiplies the fixed codebook vector (v k   c )  504  by a gain (g k   c )  506  where k equals the subframe number as previously discussed. The gain (g k   c )  506  is unquantized and represents the fixed codebook gain for each subframe. The resulting signal is processed by the second synthesis filter  462  and the second perceptual weighting filter  466  to generate a second component of resynthesized speech signal  508 . The second resynthesized speech signal  508  is subtracted from the long-term error signal  502  by the second subtractor  470  to produce a fixed codebook error  510 .  
         [0137]    The fixed codebook error signal  510  is received by the first minimization module  472  along with control information  356 . The first minimization module  472  operates in the same manner as the previously discussed second minimization module  400  illustrated in FIG. 6. The search process repeats until the first minimization module  472  has selected a fixed codebook vector (v k   c )  504  from the fixed codebook  390  for each subframe. The best vector for the fixed codebook vector (v k   c )  504  minimizes the energy of the fixed codebook error signal  510 . The indices identify the best fixed codebook vector (v k   c )  504 , and form the fixed codebook components  146   b  and  178   b.    
         [0138]    Type 1 Fixed Codebook Search for Full-Rate Codec  
         [0139]    In one embodiment, the 8-pulse codebook  162 , illustrated in FIG. 5, is used for each of the four subframes for frames of type 1 by the full-rate codec  22 . The target for the fixed codebook vector (v k   c )  504  is the long-term error signal  502 . The long-term error signal  502 , represented by t′(n), is determined based on the modified weighted speech  350 , represented by t(n), with the adaptive codebook contribution from the initial frame processing module  44  removed according to:  
                     t   ′          (   n   )       =       t        (   n   )       -       g   a     ·     (         v   a          (   n   )       *     h        (   n   )         )           ,   where                        v   a          (   n   )       =       ∑     i   =     -   10       10              w   s          (       f        (       L   p          (   n   )       )       ,   i     )       ·     e        (     n   -     I        (       L   p          (   n   )       )       +   i     )                     (     Equation                 7     )                               
 
         [0140]    and where t′(n) is a target for a fixed codebook search, g a  is a pitch gain, h(n) is an impulse response of a perceptually weighted synthesis filter, e(n) is past excitation, I(L p (n)) is an integer part of a pitch lag and f(L p (n)) is a fractional part of a pitch lag, and w S  (f, i) is a Hamming weighted Sinc window.  
         [0141]    During the search for the fixed codebook vector (v k   c )  504 , pitch enhancement may be applied in the forward, or forward and backward directions. In addition, the search procedure minimizes the fixed codebook error  508  using an iterative search procedure with controlled complexity to determine the best fixed codebook vector v k   c    504 . An initial fixed codebook gain represented by the gain (g k   c )  506  is determined during the search. The indices identify the best fixed codebook vector (v k   c )  504  and form the fixed codebook component  146   b  as previously discussed.  
         [0142]    Fixed Codebook Search for Half-Rate Codec  
         [0143]    In one embodiment, the long-term residual is represented by an excitation from a fixed codebook with  13  bits for each of the three subframes for frames classified as Type 1 for the half-rate codec  24 . The long-term residual error  502  may be used as a target in a similar manner to the fixed codebook search in the full-rate codec  22 . Similar to the fixed-codebook search for the half-rate codec  24  for frames of Type 0, high-frequency noise injection, additional pulses that are determined by correlation in the previous subframe, and a weak short-term filter may be added to enhance the fixed codebook contribution connected to the second synthesis filter  462 . In addition, forward, or forward and backward pitch enhancement may be also.  
         [0144]    For Type 1 processing, the adaptive codebook gain  496  calculated above is also used to estimate the pitch enhancement coefficients for the fixed subcodebook. However, in one embodiment of type 1 processing, the adaptive codebook gain of the current subframe, ga, rather than that of the previous subframe is used. In one embodiment, a full search is performed for a 2-pulse subcodebook  193 , a 3-pulse subcodebook  195 , and a 5-pulse subcodebook  197 , as illustrated in FIG. 5. The best fixed codebook vector (v k   c )  504  that minimizes the fixed codebook error signal  510  is selected for the representation of the long term residual for each subframe. In addition, an initial fixed codebook gain represented by the gain (g k   c )  506  may be determined during the search similar to the full-rate codec  22 . The indices identify the vector for the fixed codebook vector (v k   c )  504  and form the fixed codebook component  178   b.    
         [0145]    In one embodiment for H 1  processing, the pitch enhancement coefficients for different subcodebooks are also determined using Table 1. The pitch enhancement coefficient for the first subcodebook could be the pitch gain of the current subframe, g a , limited to a value between 0.5 and 1.0. Similarly, for H 1  processing with second and third subcodebooks, the pitch enhancement coefficient could be 0.0≦0.5 g a ≦0.5.  
         [0146]    As previously discussed, the F 1  or H 1  subframe-processing modules  74  or  84  operate on a subframe basis. However, the F 1  or H 1  second frame-processing modules  76  or  86  operate on a frame basis. Accordingly, parameters determined by the F 1  or H 1  subframe-processing module  74  or  84  are stored in the buffering module  488  for later use on a frame basis. In one embodiment, the parameters stored are the adaptive codebook vector (v k   a )  498  and the fixed codebook vector (v k   c )  504 , a modified target signal  512  and the gains  496  (g k   a ) and  506  (g k   c ) representing the initial adaptive and fixed codebook gains.  
         [0147]    Using the vectors and pitch gains, the fixed codebook gains (g k   c )  506  are determined by vector quantization (VQ). The fixed codebook gains (g k   c )  506  replace the unquantized initial fixed codebook gains determined previously. To determine the fixed codebook gains, a joint delayed quantization (VQ) of the fixed-codebook gains for each subframe is performed by the second frame-processing modules  76  and  86 .  
         [0148]    [0148]FIG. 17 comprises F 1  and H 1  subframe processing modules  74  and  84 , respectively. Each uses a pitch track provided to identify a pitch vector (v k   a )  498 . The pitch vector with the pitch gain represents a long-term prediction contribution for each subframe where k =the number of subframes. In one embodiment, there are four subframes for the F 1  codec  22  and three subframes for the H 1  codec  24 .  
         [0149]    Decoding System  
         [0150]    Referring now to FIG. 18, a functional block diagram represents the full and half rate decoders  90  and  92  of FIG. 4. One embodiment of the decoding system  16  includes a full-rate decoder  90 , a half-rate decoder  92 , a quarter-rate decoder  94 , and an eighth-rate decoder  96 , a synthesis filter module  98 , and a post-processing module  100 . The decoders are the decoding portion of the full, half, quarter and eighth rate codecs  22 ,  24 ,  26 , and  28  shown in FIG. 2.  
         [0151]    The decoders  90 ,  92 ,  94 , and  96  receive the bitstream as shown in FIG. 2, and transform the bitstream back to different parameters of the speech signal  18 . The decoders decode each frame as a function of the rate selection and classification. The rate selection is provided from the encoding system  12  to the decoding system  16  by an external signal in a control channel in a wireless communications system. The synthesis filter  98  assembles the parameters of the speech signal  18  that are decoded by the decoders, thus generating reconstructed speech. The reconstructed speech is passed thorough the post-processing module  100  to create post-processed synthesized speech  20 . Post-processing module  100  can include filtering, signal enhancement, noise modification, amplification, tilt correction, and other similar techniques capable of improving the perceptual quality of the synthesized speech.  
         [0152]    The decoders  90  and  92  perform inverse mapping of the components of the bit-stream to algorithm parameters. The inverse mapping may be followed by a type classification dependent synthesis within the full and half-rate codecs  22  and  24 .  
         [0153]    The decoding for the quarter-rate codec  26  and the eighth rate coded  28  are similar to those of the full and half rate codecs. However, the quarter-rate and eighth-rate codecs use vectors of similar yet random numbers and an energy gain, rather than the adaptive codebooks  368  and fixed codebooks  390 . The random numbers and an energy gain may be used to reconstruct an excitation energy that represents the excitation of a frame. Excitation modules  120  and  124  may be used respectively to generate portions of the quarter-rate and eighth-rate reconstructed speech. LSFs encoded during the encoding process may be used by LPC reconstruction modules  122  and  126  respectively for the quarter-rate and eighth-rate reconstructed speech.  
         [0154]    Within the full and half rate decoders  90  and  92 , operation of the excitation modules  104 ,  106 ,  114 , and  116  depends on the type classification provided by the type component  142  and  174 , just as did the encoding. The adaptive codebook  368  receives information reconstructed by the decoding system  16  from the adaptive codebook components  144  and  176  provided in the bitstream by the encoding system  12 . Depending on the type classification system provided, the synthesis filter assembles the parameters of the speech signal  18  that are decoded by the decoders,  90 ,  92 ,  94 , and  96 .  
         [0155]    One embodiment of the full rate decoder  90  includes an F-type selector  102  and a plurality of excitation reconstruction modules. The excitation reconstruction modules comprise an F 0  excitation reconstruction module  104  and an F 1  excitation reconstruction module  106 . In addition, the full rate decoder  90  includes an LPC reconstruction module  107 . The LPC reconstruction module  107  comprises an F 0  LPC reconstruction module  108  and an F 1  LPC reconstruction module  110 . The other speech parameters encoded by full rate encoder  36  are reconstructed by the decoder  90  to reconstruct speech.  
         [0156]    Similarly, an embodiment of the half-rate decoder  92  includes an H-type selector  1   12  and a plurality of excitation reconstruction modules. The excitation reconstruction modules comprise an H 0  excitation reconstruction module  114  and an H 1  excitation reconstruction module  116 . In addition, the half-rate decoder  92  comprises an H LPC reconstruction module  118 . In a manner similar to that of the full rate encoder, the other speech parameters encoded by the half rate encoder  38  are reconstructed by the half rate decoder to reconstruct speech.  
         [0157]    The F and H type selectors  102  and  112  selectively activate appropriate respective portions of the full and half rate decoders  90  and  92  respectively. A type 0 classification activates the F 0  reconstruction module  104  or H 0   114 . The respective F 0  or F 1  LPC reconstruction modules are used to reconstruct the speech from the bitstream. The same process used to encode the speech is used in reverse to decode the signals, including the pitch lags, pitch gains, and any additional factors used, such as the coefficients described above.  
         [0158]    While various embodiments of the invention have been described, it will be apparent to those of ordinary skill in the art that many more embodiments and implementations are possible that are within the scope of this invention.