Patent Publication Number: US-8126708-B2

Title: Systems, methods, and apparatus for dynamic normalization to reduce loss in precision for low-level signals

Description:
RELATED APPLICATIONS 
     This patent application is a continuation-in-part of U.S. patent application Ser. No. 11/669,407 entitled “SYSTEMS AND METHODS FOR DYNAMIC NORMALIZATION TO REDUCE LOSS IN PRECISION FOR LOW-LEVEL SIGNALS”, filed on Jan. 31, 2007, which claims priority to U.S. Provisional Application No. 60/868,476 entitled “DYNAMIC NORMALIZATION TO REDUCE LOSS IN PRECISION FOR LOW-LEVEL SIGNALS” filed Dec. 4, 2006, which are both assigned to the assignee hereof and are hereby expressly incorporated by reference herein. 
    
    
     TECHNICAL FIELD 
     The present disclosure relates generally to signal processing technology. More specifically, the present disclosure relates to systems, methods, and apparatus for dynamic normalization to reduce loss in precision for low-level signals. 
     BACKGROUND 
     Various over-the-air interfaces have been developed for wireless communication systems including, e.g., Frequency Division Multiple Access (“FDMA”), Time Division Multiple Access (“TDMA”), Code Division Multiple Access (“CDMA”), and Orthogonal Frequency Division Multiple Access (“OFDMA”). In connection therewith, various domestic and international standards have been established including, e.g., Advanced Mobile Phone Service (“AMPS”), Global System for Mobile Communications (“GSM”), and Interim Standard 95 (“IS-95”). 
     An exemplary wireless telephony communication system is a Code Division Multiple Access (“CDMA”) system. The IS-95 standard and its derivatives, IS-95A, ANSI J-STD-008, IS-95B, and third generation standards IS-95C and IS-2000, etc. (referred to collectively herein as IS-95), are promulgated by the Telecommunication Industry Association (TIA). Other well known standards bodies, such as The 3rd Generation Partnership Project 2 (“3GPP2”), specify the use of a CDMA over-the-air interface for cellular or PCS telephony communication systems. Exemplary wireless communication systems configured substantially in accordance with the use of the IS-95 standard are described in U.S. Pat. Nos. 5,103,459 and 4,901,307, which are assigned to the assignee of the present invention and fully incorporated herein by reference. 
     Multimedia streams may include speech, and may be from one or more sources that communicate with or are otherwise associated with a broadcast system. The broadcast system can use, without limitation, CDMA principles, GSM principles, or other wireless principles including wideband CDMA (WCDMA), cdma2000 (such as cdma2000 1x or 3x air interface standards, for example), TDMA, or TD-SCDMA, and OFDM. The multimedia content, including speech, can alternatively be provided, for example, over a bidirectional point-to-point link if desired, such as, e.g., a Bluetooth link or a 802.11 link or a CDMA link or GSM link. Likewise, speech content may also be transmitted using a Voice Over Internet Protocol (“VoIP”). VoIP is a protocol optimized for the transmission of voice through the Internet or other packet switched networks, which may interface with and/or merge with CDMA and GSM based systems. 
     Transmission of voice by digital techniques has become widespread, particularly in long distance and digital air-interface radio telephone applications. This, in turn, has created interest in determining the least amount of information which can be sent over the channel while maintaining the perceived quality of the reconstructed speech. If speech is transmitted by simply sampling and digitizing, a data rate on the order of 64 kilobits per second (kbps) is required to achieve a quality, known as “toll quality,” of a conventional analog telephone. However, through the use of speech analysis, followed by the appropriate coding, transmission, and resynthesis at the receiver, a significant reduction in the data rate can be achieved. 
     Devices which employ techniques to compress voiced speech by extracting parameters that relate to a model of human speech generation are typically called vocoders. Such devices are composed of an encoder, which analyzes the incoming speech to extract the relevant parameters, and a decoder, which resynthesizes the speech using the parameters which it receives over the transmission channel. In order to enhance quality, the speech codec model adapts to the changing speech signal. Modern vocoders typically operate on a digitized input signal that has been divided into blocks of time called analysis frames. Parameters are then extracted corresponding to the analysis frames. 
     Of the various classes of coders the Code Excited Linear Predictive Coding (“CELP”), Stochastic Coding or Vector Excited Speech Coding are of one class. An example of a coding algorithm of this particular class is described in the paper “A 4.8 kbps Code Excited Linear Predictive Coder” by Thomas E. Tremain et al., Proceedings of the Mobile Satellite Conference, 1988. Modern vocoders typically operate at variable rates, and are defined by standards. While various types of vocoders exist, modern commercial telecommunications vocoders generally fall into two general classes, namely the CDMA type and the GSM type. 
     A modern CDMA type network speech codec is known as Enhanced Variable Rate CODEC (“EVRC”). A version of EVRC is defined by The Telecommunications Industry Association as IS-127-B, and is formally entitled “Enhanced Variable Rate Codec Speech Service Option 3 and YY for Wideband Spread Spectrum Digital Systems,” dated December 2006. A modern GSM type of network speech codec is known as Adaptive Multi-Rate (“AMR”). A version of AMR is defined by The 3rd Generation Partnership Project (“3GPP”) as 3G TS 26.090, version 3.1.0, release 1999, and is formally entitled “Universal Mobile Telecommunications System (“UMTS”); Mandatory Speech Codec speech processing functions AMR speech codec; Transcoding functions,” dated January 2000. 
     Modern Second Generation (“2G”) and Third Generation (“3G”) radio telephone communication systems have sought to produce voice quality commensurate with the conventional public switched telephone network (“PSTN”). The PSTN have traditionally been limited in bandwidth to the frequency range of 300-3400 kHz. New networks for voice communications, such as cellular telephony and Voice over IP (“VoIP”), are not necessarily constrained by the same bandwidth limits. Accordingly, it may be desirable to transmit and receive voice communications that include a wideband frequency range over such networks. For example, it may be desirable to support an audio frequency range that extends down to 50 Hz and/or up to 7 or 8 kHz. It may also be desirable to support other applications, such as high-quality audio or audio/video conferencing, that may have audio speech content in ranges outside the traditional PSTN limits. Codecs which seek to extend the audio frequency range as set forth above are commonly referred to as wideband codecs. 
     Extension of the range supported by a speech coder into higher frequencies may improve intelligibility. For example, the information that differentiates fricatives such as ‘s’ and ‘f’ is largely in the high frequencies. Highband extension may also improve other qualities of speech, such as presence. For example, even a voiced vowel may have spectral energy far above the PSTN limit. 
     The term signal processing may refer to the processing and interpretation of signals. Signals of interest may include sound, images, and many others. Processing of such signals may include storage and reconstruction, separation of information from noise, compression, and feature extraction. The term digital signal processing may refer to the study of signals in a digital representation and the processing methods of these signals. Digital signal processing is an element of many communications technologies such as mobile phones and the Internet. The algorithms that are utilized for digital signal processing may be performed using specialized computers, which may make use of specialized microprocessors called digital signal processors (sometimes abbreviated as DSPs). 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  illustrates a wireless communication system; 
         FIG. 2  illustrates a wideband encoder that may be utilized in a wireless communication system; 
         FIG. 3  illustrates a high band encoder from the wideband encoder of  FIG. 2 ; 
         FIG. 3A  illustrates another example of a high band encoder from the wideband encoder of  FIG. 2 ; 
         FIG. 4  illustrates a factor determination component from the high band encoder of  FIG. 3 ; 
         FIG. 5  illustrates a wideband decoder that may be utilized in a wireless communication system; 
         FIG. 6  illustrates a method for dynamic normalization to reduce loss in precision for low-level signals; 
         FIG. 7  illustrates a method for determining a normalization factor for a current frame of a low band excitation signal; and 
         FIG. 8  illustrates various components that may be utilized in a communications device. 
     
    
    
     DETAILED DESCRIPTION 
     An apparatus that is configured for dynamic normalization to reduce loss in precision for low-level signals is described. The apparatus includes a processor and memory in electronic communication with the processor. Instructions are stored in the memory. The instructions are executable to determine a normalization factor for a current frame of a signal. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The instructions are also executable to normalize the current frame of the signal based on the normalization factor that is determined, and adjust the states&#39; normalization factor based on the normalization factor that is determined. 
     A method for dynamic normalization to reduce loss in precision for low-level signals is also described. A normalization factor is determined for a current frame of a signal. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal is normalized based on the normalization factor that is determined. The states&#39; normalization factor is adjusted based on the normalization factor that is determined. 
     An apparatus that is configured for dynamic normalization to reduce loss in precision for low-level signals is also described. The apparatus includes means for determining a normalization factor for a current frame of a signal. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The apparatus also includes means for normalizing the current frame of the signal based on the normalization factor that is determined, and means for adjusting the states&#39; normalization factor based on the normalization factor that is determined. 
     A computer-readable medium configured to store a set of instructions is also described. The instructions are executable to determine a normalization factor for a current frame of a signal. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The instructions are also executable to normalize the current frame of the signal based on the normalization factor that is determined, and adjust the states&#39; normalization factor based on the normalization factor that is determined. 
     An apparatus that is configured for dynamic normalization to reduce loss in precision for low-level signals is described. The apparatus includes a processor and memory in electronic communication with the processor. Instructions are stored in the memory. The instructions are executable to determine a first gain of a first frame. The first frame is a current frame. The instructions are also executable to determine a second gain of a second frame. The second frame is a previous frame. The instructions are further executable to derive a number of bits corresponding to the first gain and the second gain, and subtract the number of bits corresponding to the first gain and the second gain from a normalization factor associated with the first frame. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. 
     A method for dynamic normalization to reduce loss in precision for low-level signals is also described. A first gain of a first frame is determined. The first frame is a current frame. A second gain of a second frame is determined. The second frame is a previous frame. A number of bits corresponding to the first gain and the second gain is derived. The number of bits corresponding to the first gain and the second gain is subtracted from a normalization factor associated with the first frame. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. 
     An apparatus that is configured for dynamic normalization to reduce loss in precision for low-level signals is described. The apparatus includes means for determining a first gain of a first frame. The first frame is a current frame. The apparatus also includes means for determining a second gain of a second frame. The second frame is a previous frame. The apparatus further includes means for deriving a number of bits corresponding to the first gain and the second gain, and means for subtracting the number of bits corresponding to the first gain and the second gain from a normalization factor associated with the first frame. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. 
     A computer-readable medium configured to store a set of instructions is also described. The instructions are executable to determine a first gain of a first frame. The first frame is a current frame. The instructions are also executable to determine a second gain of a second frame. The second frame is a previous frame. The instructions are further executable to derive a number of bits corresponding to the first gain and the second gain, subtract the number of bits corresponding to the first gain and the second gain from a normalization factor associated with the first frame. The normalization factor depends on an amplitude of the current frame of the signal. The normalization factor also depends on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. 
     As used herein, the term “determining” (and grammatical variants thereof) is used in an extremely broad sense. The term “determining” encompasses a wide variety of actions and, therefore, “determining” can include calculating, computing, processing, deriving, investigating, looking up (e.g., looking up in a table, a database or another data structure), ascertaining and the like. Also, “determining” can include receiving (e.g., receiving information), accessing (e.g., accessing data in a memory) and the like. Also, “determining” can include resolving, selecting, choosing, establishing and the like. 
     The phrase “based on” does not mean “based only on,” unless expressly specified otherwise. In other words, the phrase “based on” describes both “based only on” and “based at least on.” 
       FIG. 1  illustrates a wireless communication system  100  that may include a plurality of mobile stations  102 , a plurality of base stations  104 , a base station controller (BSC)  106  and a mobile switching center (MSC)  108 . The MSC  108  may be configured to interface with a public switched telephone network (PSTN)  110 . The MSC  108  may also be configured to interface with the BSC  106 . There may be more than one BSC  106  in the system  100 . The mobile stations  102  may include cellular or portable communication system (PCS) telephones. 
     Each base station  104  may include at least one sector (not shown), where each sector may have an omnidirectional antenna or an antenna pointed in a particular direction radially away from the base station  104 . Alternatively, each sector may include two antennas for diversity reception. Each base station  104  may be designed to support a plurality of frequency assignments. The wireless communication system  100  may be configured to implement code-division multiple access (CDMA) techniques. In a CDMA system  100 , the intersection of a sector and a frequency assignment may be referred to as a CDMA channel. 
     During operation of the wireless communication system  100 , the base stations  104  may receive sets of reverse link signals from sets of mobile stations  102 . The mobile stations  102  may be conducting telephone calls or other communications. Each reverse link signal received by a given base station  104  may be processed within that base station  104 . The resulting data may be forwarded to the BSC  106 . The BSC  106  may provide call resource allocation and mobility management functionality including the orchestration of soft handoffs between base stations  104 . The BSC  106  may also route the received data to the MSC  108 , which may provide additional routing services for interfacing with the PSTN  110 . Similarly, the PSTN  110  may interface with the MSC  108 , and the MSC  108  may interface with the BSC  106 , which in turn may control the base stations  104  to transmit sets of forward link signals to sets of mobile stations  102 . 
     For purposes of example, certain systems, methods and apparatus will be described in relation to speech signals that may be processed by a wideband vocoder. (The term “wideband vocoder” will be discussed in greater detail below.) However, the systems methods, and apparatus disclosed herein are applicable outside the context of speech signals. In fact, the systems, methods, and apparatus disclosed herein may be used in connection with the processing of any type of signal (e.g., music, video, etc.) in finite precision. 
     The discussion that follows includes references to filter states. However, the systems, methods and apparatus disclosed herein are applicable to other types of states. Also, the term “states” should be construed broadly to mean any configuration of information or memories in a program or machine. 
     Transmission of voice by digital techniques has become widespread, particularly in long distance and digital radio telephone applications. In the past, voice communications have been limited in bandwidth to the frequency range of 300-3400 kHz. New networks for voice communications, such as cellular telephony and voice over IP, may not have the same bandwidth limits, and it may be desirable to transmit and receive voice communications that include a wideband frequency range over such networks. 
     A voice coder, or “vocoder,” is a device that facilitates the transmission of compressed speech signals across a communication channel. A vocoder may comprise an encoder and a decoder. An incoming speech signal may be divided into blocks of time, or analysis frames. The encoder may analyze an incoming speech frame to extract certain relevant parameters, and then quantize the parameters into a binary representation. The binary representation may be packed into transmission frames and transmitted over a communication channel to a receiver with a decoder. The decoder may process the transmission frames, dequantize them to produce the parameters, and resynthesize the speech frames using the dequantized parameters. The encoding and decoding of speech signals may be performed by digital signal processors (DSPs) running a vocoder. Because of the nature of some voice communication applications, the encoding and decoding of speech signals may be performed in real time. 
     A device (e.g., a mobile station  102  or a base station  104 ) that is deployed in a wireless communication system  100  may include a wideband vocoder, i.e., a vocoder that is configured to support a wideband frequency range. A wideband vocoder may comprise a wideband encoder and a wideband decoder. 
       FIG. 2  illustrates a wideband encoder  212 . The wideband encoder  212  may be implemented in an apparatus that may be utilized within a wireless communication system  100 . The apparatus may be a mobile phone, a personal digital assistant (PDA), a laptop computer, a digital camera, a music player, a game device, or any other device with a processor. The apparatus may function as a mobile station  102  or a base station  104  within a wireless communication system  100 . 
     A wideband speech signal  214  may be provided to the wideband encoder  212 . The wideband encoder  212  may include an analysis filter bank  216 . The filter bank  216  may filter the wideband speech signal  214  to produce a low band signal  218  and a high band signal  220 . 
     The low band signal  218  may be provided to a low band encoder  222 . The low band encoder  222  may encode the low band signal  218 , thereby generating an encoded low band signal  224 . The low band encoder  222  may also output a low band excitation signal  226 . 
     The high band signal  220  may be provided to a high band encoder  228 . The low band excitation signal  226  that is output by the low band encoder  222  may also be provided to the high band encoder  228 . The high band encoder  228  may encode the high band signal  220  according to information in the low band excitation signal  226 , thereby generating an encoded high band signal  230 . 
       FIG. 3  illustrates the high band encoder  228 . As discussed above, the low band excitation signal  226  may be provided to the high band encoder  228 . The high band encoder  228  may include a high band excitation generator  332 . The high band excitation generator  332  may derive a high band excitation signal  334  from the low band excitation signal  226 . 
     A finite number of bits is available to represent the amplitude of the signals within the wideband encoder  212 , such as the incoming wideband speech signal  214  and the low band excitation signal  226 . The precision with which these signals may be represented may be directly proportional to the number of bits that are used to represent them. The term “amplitude,” as used herein, may refer to any amplitude value of an array of amplitude values. For example, the term “amplitude” may refer to the maximum of the absolute values of the elements of an array of amplitude values. 
     The high band excitation generator  332  may perform a number of arithmetic operations on the low band excitation signal  226  (or, as will be explained below, a normalized version  336  of the low band excitation signal  226 ) in order to generate the high band excitation signal  334 . In performing at least some of these arithmetic operations on the low band excitation signal  226 , the high band excitation generator  332  may utilize the N most significant bits (MSBs) within the low band excitation signal  226 . In other words, if M bits are used to represent the amplitude of the low band excitation signal  226 , the high band excitation generator  332  may discard the M-N least significant bits (LSBs) within the low band excitation signal  226  and may utilize the N MSBs of the low band excitation signal  226  for the arithmetic operations that are performed. 
     Human speech may be classified in many different ways. Some classifications of speech may include voiced speech, unvoiced sounds, transient speech, and silence intervals/background noise during pauses between words. Under certain circumstances (e.g., for unvoiced sounds, transient speech, and silence intervals/background noise), the amplitude of the wideband speech signal  214  may be relatively low. The term low-level signal may be used herein to refer to a wideband speech signal  214  that has a relatively low amplitude. Where the incoming wideband speech signal  214  is a low-level signal, the amplitude of the low band excitation signal  226  may be fully represented, or at least mostly represented, within the LSBs of the available bits. If the LSBs are discarded by the high band excitation generator  332 , then there may be a significant loss in the precision with which the low band excitation signal  226  is represented. In an extreme case, the low band excitation signal  226  may be approximated to zero by the high band excitation generator  332 . 
     To address this issue and potentially reduce the loss of precision, the high band encoder  228  may include a signal normalizer  338 . The signal normalizer  338  may normalize the low band excitation signal  226 , thereby obtaining the normalized low band excitation signal  336 . Additional details about the operation of the signal normalizer  338  in normalizing the low band excitation signal  226  will be discussed below. 
     The low band excitation signal  226  may be normalized based on a normalization factor  344 . The normalization factor  344  may alternatively be referred to as a Q factor  344 . The normalization factor  344  may be selected so as to prevent saturation, as will be discussed below. The component that determines the normalization factor  344  may be referred to as a factor determination component  346 . 
     The low band excitation signal  226  may be divided into a number of frames. The term “current frame” may refer to the frame that is presently being processed by the wideband encoder  212 . The term “previous frame” may refer to the frame of the low band excitation signal  226  that was processed immediately prior to the current frame. 
     Normalization may be performed on a frame-by-frame basis. Thus, different normalization factors  344  may be determined for different frames of the low band excitation signal  226 . Because the normalization factor  344  may change over time, the type of normalization that may be performed by the signal normalizer  338  and the filter states normalization factor adjuster  340  may be referred to as dynamic normalization. 
     Once the normalization factor  344  for the current frame of the low band excitation signal  226  has been determined, the signal normalizer  338  may normalize the current frame of the low band excitation signal  226  based on the normalization factor  344 . Normalizing the low band excitation signal  226  may comprise left-shifting the bits of the low band excitation signal  226  by an amount that corresponds to the normalization factor  344 . 
     In some implementations, the normalization factor  344  may be negative. For example, once the normalization factor  344  is initially determined, an amount (e.g., 1) may be subtracted from the initial value of the normalization factor  344  as a protection to prevent saturation. This may be referred to as providing “head room.” Where the normalization factor  344  is negative, left-shifting by a negative normalization factor  344  may be the same as right-shifting by the corresponding positive number. 
     Additionally, a filter states normalization factor adjuster  340  may be provided. The filter states normalization factor adjuster  340  may adjust the normalization factor of the filter states  342  based on the normalization factor  344  that is determined. Adjusting the normalization factor of the filter states  342  may comprise left-shifting the bits of the filter states  342  by an amount that corresponds to the difference between the normalization factor  344  that is determined for the current frame of the low band excitation signal  226  and the normalization factor  344  that was determined for the previous frame of the low band excitation signal  226 . This operation brings the filter states  342  into the same normalization factor  344  as the normalized low band excitation signal  336 , which may facilitate filtering operations being performed. 
     When the normalization factor  344  has been determined, the current frame of the low band excitation signal  226  has been normalized, and the normalization factor of the filter states  342  of the high band excitation generator  332  has been adjusted, the high band excitation generator  332  may derive the high band excitation signal  334  from the normalized low band excitation signal  336 . This may involve performing filtering operations on the normalized low band excitation signal  336  using the adjusted filter states  342 , both of which have a normalization factor  344 . 
     The normalization factor  344  for the current frame of the low band excitation signal  226  may be selected so that saturation does not occur. There may be several ways that saturation may occur. For example, saturation may occur by left-shifting the bits of the low band excitation signal  226  to an extent where the low band excitation signal falls out of range, the range given by the number of bits used to represent the low band excitation signal. In the example discussed above, it was assumed that M bits are used to represent the low band excitation signal  226 . In this case, the maximum value of the low band excitation signal  226  using 2&#39;s complement signed arithmetic may be 2 (M-1) −1 and the minimum value may be −2 M . If M=16 (i.e., if 16 bits are used to represent the low band excitation signal  226 ), the maximum value of the low band excitation signal  226  using 2&#39;s complement signed arithmetic may be 2 15 −1, or 32767 and the minimum value may be −2 15 , or −32768. In this situation, saturation may occur if the bits of the low band excitation signal  226  are left-shifted so that the value of the low band excitation signal  226  exceeds 32767 (for positive numbers) or becomes less than −32768 (for negative numbers). The normalization factor  344  may be determined so that this type of saturation does not occur. Thus, the normalization factor  344  may depend on the amplitude of the current frame of the low band excitation signal  226 . Accordingly, the current frame of the low band excitation signal  226  may be provided to the factor determination component  346  and used to determine the normalization factor  344 . 
     As another example, saturation may occur by left-shifting the bits of the filter states  342  of the high band excitation generator  332  to an extent where the filter states fall out of range. As discussed in the example above, if M=16, this range is given by the set of numbers which fall into the category of numbers no greater than +32767 and no less than −32768. The normalization factor  344  may be determined so that this does not occur. When the normalization factor of the filter states  342  is adjusted, the values of the filter states  342  may depend on the filtering operations that were performed on the previous frame of the normalized low band excitation signal  336 . Thus, the normalization factor  344  may depend on the values of the filter states  342  after the filtering operations were performed on the previous frame of the normalized low band excitation signal  336 . Accordingly, information  348  about the values of the filter states  342  after the filtering operations were performed on the previous frame of the normalized low band excitation signal  336  may be provided to the factor determination component  346  and used to determine the normalization factor  344 . 
     Each frame of the low band excitation signal  226  may be normalized in the manner described above. More specifically, for each frame of the low band excitation signal  226 , a normalization factor  344  may be determined. The current frame of the low band excitation signal  226  may be normalized based on the normalization factor  344  that is determined for that frame. Also, the normalization factor of the filter states  342  may be adjusted based on the normalization factor  344  that is determined for that frame. These steps (i.e., determining the normalization factor  344 , normalizing the current frame of the low band excitation signal  226 , and adjusting the normalization factor of the filter states  342 ) may be performed for each frame of the low band excitation signal  226 . 
       FIG. 4  illustrates the factor determination component  346 . As discussed above, the factor determination component  346  may determine the normalization factor  344   a  for the current frame of the low band excitation signal  226 . 
     As discussed above, the current frame of the low band excitation signal  226  may be provided to the factor determination component  346 . The current frame of the low band excitation signal  226  may be analyzed to determine an optimal value for the normalization factor  344   a  for the current frame of the low band excitation signal  226 . (The optimal value is labeled with reference number  450  in  FIG. 4 , and will be referred to as optimal value  450  hereinafter.) The component that implements this functionality may be referred to as an optimal value determination component  452 . 
     The optimal value  450  for the normalization factor  344  may be determined based on the amplitude of the current frame of the low band excitation signal  226 . Since the low band excitation signal  226  of the current frame comprises an array of numbers, the optimal value  450  of the normalization factor  344  may refer to the number of bits of the maximum of the absolute value of the array of numbers that can be left-shifted without causing saturation, also referred to as the block normalization factor. The optimal value  450  for the normalization factor  344  may indicate to what extent the bits of the current frame of the low band excitation signal  226  may be left-shifted without causing saturation. 
     As discussed above, information  348  about the values of the filter states  342  after the filtering operations were performed on the previous frame of the normalized low band excitation signal  336  may also be provided to the factor determination component  346 . This information  348  may be used to determine a scaling factor  454  for the filter states  342  of the high band excitation generator  332 . The component that implements this functionality may be referred to as a scaling factor determination component  456 . 
     The scaling factor  454  may be determined based on the filter states information  348  that is received. The scaling factor  454  may indicate to what extent the bits of the filter states  342  may be left-shifted without causing saturation. The procedure for obtaining this scaling factor  454  may be similar to the above-mentioned procedure of determining the optimal value  450  for the normalization factor  344 , the array of numbers in this case being the filter states, where the filter states may be states from different filters. 
     In some implementations, some filter states may be double precision (DP, 32 bits) and some filter states may be single precision (SP, 16 bits). In such implementations, the block normalization factor of the double precision filter states may be obtained. This block normalization factor may then be scaled down by a factor of two to bring it to the single precision domain. It may then be determined which is the lowest block normalization factor between this scaled down double precision block normalization factor and the block normalization factor of the single precision filter states. The lowest block normalization factor may then be outputted as the scaling factor  454 . In this specific example the terms current frame normalization factor  344   a  and previous frame normalization factor  344   b  refer to the normalization factor in the single precision domain. The filter states normalization factor adjuster  340  scales up by a factor of two the difference between the normalization factor  344  that is determined for the current frame of the low band excitation signal  226  and the normalization factor  344  that was determined for the previous frame of the low band excitation signal  226 , before left-shifting the bits of the double precision filter states  342 . 
     In additional implementations, some filters may operate on the squared values of the signal and some other filters may operate directly on the signal values (in the linear domain). Hence, some filter states may be in the squared domain and some filter states may be in the linear domain. In such implementations, the block normalization factor of the squared domain filter states and the linear domain states may be obtained separately. The squared domain filter states and the linear domain filter states may not be compared directly to obtain a block normalization factor, since in some implementations, the norm factor of the squared domain filter states may be twice that of the linear domain filter states. As shown in  FIG. 3A , a non-linear factor determination  360  may be implemented for non-linear filter states, such as squared values and cubic values of the signal. 
     The block normalization factor from the squared domain filter states may then be scaled down by a factor of two to bring it to the linear domain. It may then be determined which is the lowest block normalization factor between this scaled down squared domain block normalization factor and the block normalization factor of the linear domain filter states. The lowest block normalization factor may then be outputted as the scaling factor  454 . In this example, the terms “current frame normalization factor”  344   a  and “previous frame normalization factor”  344   b  refer to the normalization factor in the linear domain. The filter states normalization factor adjuster  340  scales up by a factor of two the difference between the normalization factor  344  that is determined for the current frame of the low band excitation signal  226  and the normalization factor  344  that was determined for the previous frame of the low band excitation signal  226 , before left-shifting the bits of the squared domain filter states  342 . 
     In some implementations, the block normalization factor from the linear domain filter states may be scaled up by a factor of two to bring it to the squared domain. It may then be determined which is the lowest block normalization factor between this scaled up linear domain block normalization factor and the block normalization factor of the squared domain filter states. The lowest block normalization factor may then be outputted as the scaling factor  454 . In this example, the terms “current frame normalization factor”  344   a  and “previous frame normalization factor”  344   b  refer to the normalization factor in the squared domain. The filter states normalization factor adjuster  340  scales down by a factor of two the difference between the normalization factor  344  that is determined for the current frame of the low band excitation signal  226  and the normalization factor  344  that was determined for the previous frame of the low band excitation signal  226 , before left-shifting the bits of the linear domain filter states  342 . In this example, if a squared domain norm factor needs to be applied to signal or filter state values in linear domain, the squared domain norm factor needs to be scaled down by a factor of two. 
     In an additional example, some filters may operate on the cubic values of the signal. For example, some of the filter states may be in a cube domain. A cubic spline normalization between frames may be implemented. 
     A saturation condition may be evaluated. The component that implements this functionality may be referred to as a condition evaluation component  458 . The saturation condition may depend on the optimal value  450  for the normalization factor  344   a  for the current frame of the low band excitation signal  226 . The saturation condition may also depend on the scaling factor  454  for the filter states  342  of the high band excitation generator  332 . 
     The saturation condition may also depend on the normalization factor  344   b  for the previous frame of the low band excitation signal  226 . The normalization factor  344   b  for the previous frame of the low band excitation signal  226  may indicate to what extent the bits of the previous frame of the low band excitation signal  226  were shifted prior to filtering operations being performed on the previous frame of the normalized low band excitation signal  336 . 
     The saturation condition that is evaluated may be expressed as:
 
 Qinp −prev —   Qinp&gt;Q _states  (1)
 
     In equation (1), the term Qinp may refer to the optimal value  450  for the normalization factor  344   a  for the current frame of the low band excitation signal  226 . The term prev_Qinp may refer to the normalization factor  344   b  for the previous frame of the low band excitation signal  226 . The term Q_states may refer to the scaling factor  454  for the filter states  342 . 
     If it is determined that the saturation condition is not satisfied, this may be interpreted to mean that setting the normalization factor  344   a  equal to the optimal value  450  that was determined is not going to cause saturation. In this case, determining the normalization factor  344   a  for the current frame of the low band excitation signal  226  may involve setting the normalization factor  344   a  equal to the optimal value  450  that was determined. 
     If it is determined that the saturation condition is satisfied, this may be interpreted to mean that setting the normalization factor  344   a  equal to the optimal value  450  that was determined is going to cause saturation. In this case, determining the normalization factor  344   a  for the current frame of the low band excitation signal  226  may involve setting the normalization factor  344   a  equal to prev_Qinp+Q_states. In this expression, the terms Qinp, prev_Qinp and Q_states may have the same meaning as was discussed above in connection with equation (1). Hence, the normalization factor  344   a  may be given by the expression MIN (Q_inp, prev_Qinp+Q_states). 
     The 3 rd  Generation Partnership Project (3GPP2) implements a standard titled “Enhanced Variable Rate Codec, Speech Service Options 3, 68, and 70 for Wideband Spread Spectrum Digital Systems.” This standard may be referred to hereafter as 3GPP2 C.S0014-C. Multiple sections of the above-mentioned standard address dynamic normalization. These sections are further described below. 
     3GPP2 C.S0014-C, Version 0.3 Published July 2006 Section 5.2.3.15 “Synthesis of the Decoder Output Signal” 
     Dynamic normalization may be implemented during the synthesis of a decoder output signal. In one example, a combined excitation signal, may be filtered through a synthesis filter using interpolated LPCs. A synthesized speech signal may be created. An example of source code to implement dynamic normalization applied to the synthesis of the decoder output signal may be as follows: 
     
       
         
           
               
             
               
                   
               
             
            
               
                 #ifdef IMPROVE_SYN_DEC_SPCH_PRECISION 
               
               
                  Word40 curr_lpc_gain_fx=0; 
               
               
                  Word40 prev_lpc_gain_fx=0; 
               
               
                  Word16 n_decspch=15,lpcgain_bits,max_s16,Q_curr_bck,Q_prev_bck; 
               
               
                 #endif 
               
               
                 #ifdef IMPROVE_SYN_DEC_SPCH_PRECISION 
               
               
                   curr_lpc_gain_fx = compute_lpc_gain_fx(Llsp_fx,pci_fx,ORDER); 
               
               
                   prev_lpc_gain_fx = compute_lpc_gain_fx(LOldlspD_fx,pci_fx,ORDER); 
               
               
                   for (i=0,max_s16=0;i&lt;FSIZE;i++) 
               
               
                    max_s16=MAX_FX(max_s16,abs_s(LPitchMemoryD_frame_fx[i])); 
               
               
                   n=norm_s(max_s16); 
               
               
                   Q_curr =(max_s16==0)?15:n; 
               
               
                   lpcgain_bits=31-23-norm32_140(MAX_FX(prev_lpc_gain_fx,curr_lpc_gain_fx)); 
               
               
                   if ((lpcgain_bits &amp; 1) == 1) lpcgain_bits=shr(lpcgain_bits,1)+1; 
               
               
                   else 
               
               
                    lpcgain_bits=shr(lpcgain_bits,1); 
               
               
                   Q_curr=sub(sub(Q_curr,lpcgain_bits),2);//2 bits space for saturation 
               
               
                 #if 1//added this patch to prevent saturation 
               
               
                   find_max_decspch(&amp;n_decspch); 
               
               
                   if (sub(Q_curr,Q_prev) &gt; n_decspch) Q_curr=add(Q_prev,n_decspch); // Q_curr 
               
               
                 lies in the interval [Q_curr,n_decspch+ Q_prev] 
               
               
                 #endif 
               
               
                   
               
            
           
         
       
     
     In one example, the above section of source code (and the corresponding functions it calls) may implement the factor determination  346  of  FIG. 3 . The following example applies the factor determination  346  to a specific configuration of when the output&#39;s dynamic range is greater than the input&#39;s dynamic range (i.e., output is amplified by a gain factor). The curr_lpc_gain_fx may be the current frame&#39;s LPC gain and prev_lpc_gain_fx may be the previous frame&#39;s LPC gain. The above code may also derive the number of bits corresponding to the LPC gain using the mapping that 1 bit equals 6 dB. In addition, the above code may subtract off the lpcgain_bits from the current frame normalization factor to provide the head room for the output and prevent saturation of the output. This may be performed in addition to the saturation prevention discussed above. A further example of the source code is as follows: 
     
       
         
           
               
               
             
               
                   
                   
               
             
            
               
                   
                 for (i=0;i&lt;FSIZE;i++) 
               
               
                   
                  LPitchMemoryD_frame_fx[i]= 
               
               
                   
                  shl(LPitchMemoryD_frame_fx[i],Q_curr); //Q_curr 
               
               
                   
                   
               
            
           
         
       
     
     In one configuration, this section of code (and the corresponding functions it calls) may implement the signal normalizer  338  of  FIG. 3 . A further example of the source code is as follows: 
     
       
         
           
               
               
             
               
                   
                   
               
             
            
               
                   
                 Q_factor_adjust_decspch(Q_curr−Q_prev); 
               
               
                   
                 #endif 
               
               
                   
                   
               
            
           
         
       
     
     The above-mentioned section of the code (and the corresponding functions) may implement the filter states norm factor adjuster  340  of  FIG. 3 . A further example of the code is as follows: 
     
       
         
           
               
               
             
               
                   
                   
               
             
            
               
                   
                 #ifdef IMPROVE_SYN_DEC_SPCH_PRECISION 
               
               
                   
                 Q_prev=Q_curr; 
               
               
                   
                 #endif 
               
               
                   
                 #ifdef IMPROVE_SYN_DEC_SPCH_PRECISION 
               
               
                   
                    for (j=0;j&lt;FSIZE;j++) 
               
               
                   
                     { 
               
               
                   
                      LoutFbuf_fx[j]=shr(LoutFbuf_fx[j],Q_curr); 
               
               
                   
                    } 
               
               
                   
                 #endif 
               
               
                   
                   
               
            
           
         
       
     
     The code provided above that may implement dynamic normalization applied to the synthesis of the decoder output signal calls various functions. The functions called by the above code may be as follows: 
                                /* FUNCTION  : compute_lpc_gain_fx( )            */       /*-------------------------------------------------------------------*/       /* PURPOSE  : Computes the LPC gain of the input LSP vector  */       /*-------------------------------------------------------------------*/       /* INPUT ARGUMENTS :                          */       /*   —  (Word16 [ ]) tmplsp_fx : input signal, Q15     */       /*   —  (Word16)  order  : LPC order           */       /*-------------------------------------------------------------------*/       /* OUTPUT ARGUMENTS :                   */       /*             */       /*  —  (Word40) output : lpc gain (linear) Q23            */       /* this scheme can compute a max LPC gain of around 48 dB   */       /*-------------------------------------------------------------------*/       /* INPUT/OUTPUT ARGUMENTS :                       */       /*             */       /*      —  None      */       /*-------------------------------------------------------------------*/       /* RETURN ARGUMENTS :  —  None.                     */       /*=============================================================       ======*/       /* NOTE: Length of impulse response is fixed at 55     */       /*=============================================================       ======*/       Word40  compute_lpc_gain_fx(Word16  tmplsp_fx[ ],Word16  tmppci_fx[ ],Word16       order)       {        //Word16 tmppci_fx[order];        Word16 L=55;        Word16 j;        Word16 temp_in[L];        Word32 temp_mem[order];        Word40 lpc_gain1_fx=0        lsp2lpc_fx(tmplsp_fx, tmppci_fx, tmppci_fx,order);        temp_in[0]=0x0800; // one impulse 1.0 in Q11        for (j=1; j&lt;L; j++) temp_in[j]=0;        for (j=0;j&lt;order;j++) temp_mem[j]=0;        synthesis_filter_fx (tmppci_fx, temp_in, H_fx, temp_mem,order, L, 3);        /* Get energy of H */        for (j=0,lpc_gain1_fx=0; j&lt;L; j++)        lpc_gain1_fx = L_mac40(lpc_gain1_fx, H_fx[j], H_fx[j]); // Q23        return(lpc_gain1_fx);       }       void find_max_decspch(Word16 *norm)       {        Word16 i;        Word16 max=0;        Word32 max32=0,temp;        Word16 n,n1;         if ((data_packet.Celp_Mdct_Flag==1) &amp;&amp; (prev_celp_mdct_dec==1)) //curr and       prev are MDCT frames         {        max=abs_s(dec_preemphmem[0]);        for (i=0;i&lt;ORDER;i++) max=MAX_FX(abs_s(dec_FormantFilterMemory[i]),max);         }        for (i=0;i&lt;ORDER;i++) max=MAX_FX(abs_s(FIRmempf_fx[i]),max);        for (i=0;i&lt;ORDER;i++)         {         if (SynMemory_fx[i]&lt;0) temp=L_negate(SynMemory_fx[i]);         else          temp=SynMemory_fx[i];         max32=MAX_FX(temp,max32);         }        for (i=0;i&lt;ORDER;i++)         {         if (PF_mem_syn_pst_fx[i]&lt;0) temp=L_negate(PF_mem_syn_pst_fx[i]);         else          temp=PF_mem_syn_pst_fx[i];         max32=MAX_FX(temp,max32);         }        n=norm_s(max);        n =(max==0)?15:n;        n1=norm_1(max32);        n1 =(max32==0)?31:n1;        *norm = MIN_FX(n,n1);       }       void Q_factor_adjust_decspch(Word16 shl_fac)       {        Word16 i;        if ((data_packet.Celp_Mdct_Flag==1) &amp;&amp; (prev_celp_mdct_dec==1)) //curr and prev       are MDCT frames        {        dec_preemphmem[0]=shl(dec_preemphmem[0],shl_fac);        for(i=0;i&lt;ORDER;i++)       dec_FormantFilterMemory[i]=shl(dec_FormantFilterMemory[i],shl_fac);        }        for (i=0;i&lt;ORDER;i++) FIRmempf_fx[i]=shl(FIRmempf_fx[i],shl_fac);        for (i=0;i&lt;ORDER;i++)         SynMemory_fx[i]=L_shl(SynMemory_fx[i],shl_fac);        for (i=0;i&lt;ORDER;i++)         {         PF_mem_syn_pst_fx[i]=L_shl(PF_mem_syn_pst_fx[i],shl_fac);         }       }                    
3GPP2 C.S0014-C, Version 0.3 Published July 2006 Section 4.18.3 “High-Band Excitation Generation”
 
     The dynamic normalization scheme may also be applied during the generation of high-band excitation signal  334  where some of the filter states are non-linear. The high-band excitation signal  334  may be derived from low-band excitation in the form of the excitation signal  226 . An example of source code to implement dynamic normalization applied to the generation of high-band excitation signal  334  may be as follows: 
     
       
         
           
               
               
             
               
                   
                   
               
             
            
               
                   
                 for (i=0,max_s16=0;i&lt;LB_FRAMESIZE;i++) 
               
               
                   
                   max_s16=MAX_FX(max_s16,abs_s(LB_Excitation_fx[i])); 
               
               
                   
                 // max_s16 has the sample with least sign bits 
               
               
                   
                  n=norm_s(max_s16); 
               
               
                   
                  Q_lbexct =(max_s16==0)?15:n; 
               
               
                   
                  Q_lbexct=sub(Q_lbexct,1);//1 bit space for saturation 
               
               
                   
                 #if 1//added this patch to prevent saturation 
               
               
                   
                  find_max_struc(&amp;gen_shape_noise_dec_fx,&amp;n_struc); 
               
               
                   
                  #ifdef WMOPS_FX 
               
               
                   
                  test( ); 
               
               
                   
                  #endif 
               
               
                   
                  if (sub(Q_lbexct,prev_Q_lbexct) &gt; n_struc) 
               
               
                   
                 Q_lbexct=add(prev_Q_lbexct,n_struc); // Q_lbexct 
               
               
                   
                 lies in the interval [Q_lbexct,n_struc+prev_Q_lbexct] 
               
               
                   
                 #endif 
               
               
                   
                  for (i=0;i&lt;LB_FRAMESIZE;i++) 
               
               
                   
                   LB_Excitation_fx[i]= shl(LB_Excitation_fx[i],Q_lbexct); 
               
               
                   
                   //Q_lbexct 
               
               
                   
                  // func. which brings the Qfac of states inside 
               
               
                   
                  gen_shape_noise_dec_fx struc 
               
               
                   
                  // to line up with curr frame&#39;s Qfac 
               
               
                   
                  Q_factor_adjust(&amp;gen_shape_noise_dec_fx,Q_lbexct- 
               
               
                   
                  prev_Q_lbexct); 
               
               
                   
                 prev_Q_lbexct= Q_lbexct; 
               
               
                   
                   
               
            
           
         
       
     
     Portions of the code provided below address non-linear filter states. Specific examples of the code which address such non-linear filter states may be as follows: 
     
       
         
           
               
               
             
               
                   
                   
               
             
            
               
                   
                 f-&gt;noise_iir_fx[0] = L_shl(f-&gt;noise_iir_fx[0],shl(shl_fac,1)) and 
               
               
                   
                 n1=norm_1(f-&gt;noise_iir_fx[0]); 
               
               
                   
                  n1 =(f-&gt;noise_iir_fx[0]==0)?31:n1; 
               
               
                   
                  n1=shr(n1,1); 
               
               
                   
                   
               
            
           
         
       
     
     The code provided above that may implement dynamic normalization applied to the generation of high-band excitation signal  334  calls various functions. The functions called by the above code may be as follows: 
                                void Q_factor_adjust(struct GEN_SHP_NOISE_fx *f,Word16 shl_fac)       {        Word16 i;        for (i=0;i&lt;18;i++)         f-&gt;rapf_filt_mem_fx[i] = shl(f-&gt;rapf_filt_mem_fx[i],shl_fac);        for (i=0;i&lt;6;i++)         f-&gt;gen_pulses_dec_fx-&gt;mem1_fx[i]   =   shl(f-&gt;gen_pulses_dec_fx-       &gt;mem1_fx[i],shl_fac);        for (i=0;i&lt;2;i++)         f-&gt;gen_pulses_dec_fx-&gt;rhpf_filt_mem_fx[i]   =   shl(f-&gt;gen_pulses_dec_fx-       &gt;rhpf_filt_mem_fx[i],shl_fac);        for (i=0;i&lt;4;i++)         f-&gt;res_down_dec_fx-&gt;memup_fx[i]      =   shl(f-&gt;res_down_dec_fx-       &gt;memup_fx[i],shl_fac);        for (i=0;i&lt;LEN_DN_HB − 2;i++)         f-&gt;res_down_dec_fx-&gt;state_fx[i] = shl(f-&gt;res_down_dec_fx-&gt;state_fx[i],shl_fac);        for (i=0;i&lt;7;i++)         f-&gt;res_down_dec_fx-&gt;memdown_fx[i]   =   shl(f-&gt;res_down_dec_fx-       &gt;memdown_fx[i],shl_fac);        for (i=0;i&lt;7;i++)         f-&gt;memdown1_fx[i] = shl(f-&gt;memdown1_fx[i],shl_fac);        for (i = 0; i &lt; 4; i++)         f-&gt;stateExc_fx[i] = shl(f-&gt;stateExc_fx[i],shl_fac);        f-&gt;noise_iir_fx[0] = L_shl(f-&gt;noise_iir_fx[0],shl(shl_fac,1)); //in the squared domain        for (i = 0; i &lt; 54; i++)         f-&gt;stateExcW_fx[i] = shl(f-&gt;stateExcW_fx[i],shl_fac);;       }       void find_max_struc(struct GEN_SHP_NOISE_fx *f,Word16 *struc_norm)       {        Word16 i;        Word16 max=0;        Word16 n,n1;        for (i=0;i&lt;18;i++) {         max=MAX_FX(abs_s(f-&gt;rapf_filt_mem_fx[i]),max);        }        for (i=0;i&lt;6;i++) {         max=MAX_FX(abs_s(f-&gt;gen_pulses_dec_fx-&gt;mem1_fx[i]),max);        }        for (i=0;i&lt;2;i++) {         max=MAX_FX(abs_s(f-&gt;gen_pulses_dec_fx-&gt;rhpf_filt_mem_fx[i]),max);        }        for (i=0;i&lt;4;i++) {         max=MAX_FX(abs_s(f-&gt;res_down_dec_fx-&gt;memup_fx[i]),max);        }        for (i=0;i&lt;LEN_DN_HB − 2;i++) {         max=MAX_FX(abs_s(f-&gt;res_down_dec_fx-&gt;state_fx[i]),max);        }       #ifdef DEC_ALLPASS_STEEP_DP        for (i=0;i&lt;7;i++)         f-&gt;res_down_dec_fx-&gt;memdown_fx[i]   =   L_shl(f-&gt;res_down_dec_fx-       &gt;memdown_fx[i],shl_fac);        for (i=0;i&lt;7;i++)         f-&gt;memdown1_fx[i] = L_shl(f-&gt;memdown1_fx[i],shl_fac);       #else        for (i=0;i&lt;7;i++) {         max=MAX_FX(abs_s(f-&gt;res_down_dec_fx-&gt;memdown_fx[i]),max);        }        for (i=0;i&lt;7;i++) {         max=MAX_FX(abs_s(f-&gt;memdown1_fx[i]),max);        }       #endif        for (i = 0; i &lt; 4; i++) {         max=MAX_FX(abs_s(f-&gt;stateExc_fx[i]),max);        }        n1=norm_1(f-&gt;noise_iir_fx[0]);        n1 =(f-&gt;noise_iir_fx[0]==0)?31:n1;        n1=shr(n1,1); //bring from squared domain to linear domain        for (i=0; i &lt; 54; i++) {         max=MAX_FX(abs_s(f-&gt;stateExcW_fx[i]),max);        }        n=norm_s(max);        n =(max==0)?15:n;        *struc_norm = sub(MIN_FX(n,n1),1); //1 bit head room       }                    
3GPP2 C.S0014-C, Version 0.3 Published July 2006 Section 4.5 “Analysis Filterbank”
 
     Dynamic normalization may also be implemented during the high band analysis filterbank. An example of source code to implement dynamic normalization applied to the high band analysis filterbank in accordance with section 4.5 of the 3GPP2 C.S0014-C standard may be as follows: 
     
       
         
           
               
               
             
               
                   
                   
               
             
            
               
                   
                 #ifdef USE_NORM_FOR_ANA_HB 
               
               
                   
                     Word16 Q_hbana; 
               
               
                   
                     static Word16 prev_Q_hbana=0; 
               
               
                   
                     Q_hbana = normalize_hb_ana 
               
               
                   
                 (wb_in_ptr, S_ana_hb_fx, prev_Q_hbana, 
               
               
                   
                 2*ibuf_len+extra_samples); 
               
               
                   
                 #endif 
               
               
                   
                 #ifdef USE_NORM_FOR_ANA_HB 
               
               
                   
                     scale_hb_ana  (hb_out_ptr,  wb_14k_ptr, 
               
               
                   
                 Q_hbana,  ibuf_len*7/8  + extra_samples*7/16); 
               
               
                   
                 # ifdef WMOPS_FX 
               
               
                   
                 move16( ); 
               
               
                   
                 # endif 
               
               
                   
                     prev_Q_hbana = Q_hbana; 
               
               
                   
                 #endif 
               
               
                   
                   
               
            
           
         
       
     
     The code provided above that may implement dynamic normalization applied to the high band analysis filterbank calls various functions. The functions called by the above code may be as follows: 
                                Word16  normalize_hb_ana  (Word16  *wb_in_ptr,  STATE_ANA_HB_fx       &amp;S_ana_hb_fx, Word16 prev_Q_hbana, Word16 len)       {           Word16 max_a16=0;           Word16 n_struc=15;           Word16 n, k;           Word16 Q_hbana=0;           for(k=0,max_a16=0;k&lt;len;k++)            max_a16=MAX_FX(max_a16,abs_s(wb_in_ptr[k])); // max_s16 has the       sample with least sign bits           n=norm_s(max_a16);         #ifdef WMOPS_FX         move16( );         test( );         move16( );         #endif           Q_hbana =(max_a16==0)?15:n;           Q_hbana=sub(Q_hbana,3);//3 bits space for saturation       # if 0//added this patch to prevent saturation           find_max_anahb_struc(&amp;S_ana_hb_fx,&amp;n_struc);           if     (sub(Q_hbana,prev_Q_hbana)     &gt;     n_struc)       Q_hbana=add(prev_Q_hbana,n_struc);  //  Q_hbana  lies  in  the  interval       [Q_hbana,n_struc+prev_Q_hbana]       # endif       # if 1//added this patch to prevent saturation           find_max_anahb_struc(&amp;S_ana_hb_fx,&amp;n_struc);           if     (sub(Q_hbana,prev_Q_hbana)     &gt;     n_struc)       Q_hbana=add(prev_Q_hbana,n_struc);  //  Q_hbana lies in the interval       [Q_hbana,n_struc+prev_Q_hbana]       # endif           for(k=0;k&lt;len;k++)            wb_in_ptr[k]= shl(wb_in_ptr[k],Q_hbana);           // func. which brings the Qfac of states inside syn_hb_fx struct struc           // to line up with curr frame&#39;s Qfac           Q_factor_adjust_anahb(&amp;S_ana_hb_fx,Q_hbana−prev_Q_hbana);           return (Q_hbana);       }       void scale_hb_ana (Word16 *hb_out_ptr, Word16 *wb_14k_ptr, Word16 Q_hbana,       Word16 len)       {        Word16 k;       #define BUG_FIX 0       #if BUG_FIX        for(k=0;k&lt;len;k++){        hb_out_ptr[k]= shr(hb_out_ptr[k],Q_hbana);        }        for(k=0;k&lt;2*len;k++){        wb_14k_ptr[k]= shr(wb_14k_ptr[k],Q_hbana);        }       #else        for(k=0;k&lt;len;k++){        hb_out_ptr[k]= shr(hb_out_ptr[k],Q_hbana);        wb_14k_ptr[k]= shr(wb_14k_ptr[k],Q_hbana);        }       #endif       }       void Q_factor_adjust_anahb(struct STATE_ANA_HB_fx *f,Word16 shl_fac)       {        Word16 i;        for(i=0;i&lt;2*ALLPASSSECTIONS;i++)        f-&gt;state_ana_filt_hb_1_fx[i] = shl(f-&gt;state_ana_filt_hb_1_fx[i],shl_fac);        for(i=0;i&lt;LEN_DN_HB−2;i++)        f-&gt;state_ana_filt_hb_2_fx[i] = shl(f-&gt;state_ana_filt_hb_2_fx[i],shl_fac);        for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP+1;i++)        f-&gt;state_ana_filt_hb_3_fx[i] = shl(f-&gt;state_ana_filt_hb_3_fx[i],shl_fac);        for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP+1; i++)        f-&gt;state_ana_filt_hb_4_fx[i] = shl(f-&gt;state_ana_filt_hb_4_fx[i],shl_fac);         f-&gt;mem_ana_filt_hb_iir_fx[0] = L_shl(f-&gt;mem_ana_filt_hb_iir_fx[0],shl_fac);       }       void find_max_anahb_struc(struct STATE_ANA_HB_fx *f,Word16 *struc_norm)       {        Word16 i;        Word16 max=0;        Word16 n,n1;        for(i=0;i&lt;2*ALLPASSSECTIONS;i++)        max=MAX_FX(abs_s(f-&gt;state_ana_filt_hb_1_fx[i]),max);        for(i=0;i&lt;LEN_DN_HB−2;i++)        max=MAX_FX(abs_s(f-&gt;state_ana_filt_hb_2_fx[i]),max);        for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP+1;i++)        max=MAX_FX(abs_s(f-&gt;state_ana_filt_hb_3_fx[i]),max);        for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP+1; i++)        max=MAX_FX(abs_s(f-&gt;state_ana_filt_hb_4_fx[i]),max);        n=norm_s(max);        n =(max==0)?15:n;        n1=norm_1(f-&gt;mem_ana_filt_hb_iir_fx[0]);        n1 =(f-&gt;mem_ana_filt_hb_iir_fx[0]==0)?31:n1;        *struc_norm = sub(MIN_FX(n,n1),3); //3 bit head room       }                    
3GPP2 C.S0014-C, Version 0.3 Published July 2006 Section 5.13 “Decoding of Synthesis Filterbank for 16 KHz Decoding for SO 70”
 
     Dynamic normalization may be implemented to the synthesis filterbank at the decoder. An example of source code to implement dynamic normalization applied to the synthesis filterbank at the decoder in accordance with section 5.13 of the 3GPP2 C.S0014-C standard may be as follows: 
     
       
         
           
               
             
               
                   
               
             
            
               
                 #ifdef USE_NORM_FOR_SYN_HB 
               
               
                  Word16 max_s16=0; 
               
               
                  Word16 n; 
               
               
                  Word16 n_struc=15; 
               
               
                  Word16 Q_hbsyn=0; 
               
               
                  static Word16 prev_Q_hbsyn=0; 
               
               
                  for(k=0,max_s16=0;k&lt;140;k++) 
               
               
                  max_s16=MAX_FX(max_s16,abs_s(buf_HB_out_fx[k])); 
               
               
                 //  max_s16 has the sample with least sign bits 
               
               
                  n=norm_s(max_s16); 
               
               
                  Q_hbsyn =(max_s16==0)?15:n; 
               
               
                  Q_hbsyn=sub(Q_hbsyn,3);//3 bits space for saturation 
               
               
                 #if 1//added this patch to prevent saturation 
               
               
                  find_max_synhb_struc(&amp;S_syn_hb_fx,&amp;n_struc); 
               
               
                  if (sub(Q_hbsyn,prev_Q_hbsyn) &gt; n_struc) 
               
               
                 Q_hbsyn=add(prev_Q_hbsyn,n_struc); // Q_hbsyn 
               
               
                 lies in the interval [Q_hbsyn,n_struc+prev_Q_hbsyn] 
               
               
                 #endif 
               
               
                  for(k=0;k&lt;UB_FRAMESIZE;k++) 
               
               
                  buf_HB_out_fx[k]= shl(buf_HB_out_fx[k],Q_hbsyn); //Q_lbexct 
               
               
                  // func. which brings the Qfac of states inside syn_hb_fx struct struc 
               
               
                  // to line up with curr frame&#39;s Qfac 
               
               
                  Q_factor_adjust_hb(&amp;S_syn_hb_fx,Q_hbsyn−prev_Q_hbsyn); 
               
               
                 #endif 
               
               
                 #ifdef USE_NORM_FOR_SYN_HB 
               
               
                    //do shr and wrap prevQ to Q 
               
               
                 #ifdef VOIP_DORA 
               
               
                    for(k=0;k&lt;2*160;k++) 
               
               
                 #else 
               
               
                    for(k=0;k&lt;2*obuf_len;k++) 
               
               
                 #endif 
               
               
                  buf_16fx[k]= shr(buf_16fx[k],Q_hbsyn); //Q_lbexct 
               
               
                  prev_Q_hbsyn= Q_hbsyn; 
               
               
                 #endif 
               
               
                   
               
            
           
         
       
     
     The code provided above that may implement dynamic normalization applied to the synthesis filterbank at the decoder calls various functions. The functions called by the above code may be as follows: 
     
       
         
           
               
             
               
                   
               
             
            
               
                 void Q_factor_adjust_hb(struct STATE_SYN_HB_fx *f,Word16 shl_fac) 
               
               
                 { 
               
               
                  Word16 i; 
               
               
                  for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP;i++) 
               
               
                  f-&gt;state_syn_filt_hb_0_fx[i] = shl(f-&gt;state_syn_filt_hb_0_fx[i],shl_fac); 
               
               
                  for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP;i++) 
               
               
                  f-&gt;state_syn_filt_hb_1_fx[i] = shl(f-&gt;state_syn_filt_hb_1_fx[i],shl_fac); 
               
               
                  for(i=0;i&lt;LEN_UP_HB;i++) 
               
               
                  f-&gt;state_syn_filt_hb_2_fx[i] = shl(f-&gt;state_syn_filt_hb_2_fx[i],shl_fac); 
               
               
                  for(i=0;i&lt;2*ALLPASSSECTIONS+1;i++) 
               
               
                  f-&gt;state_syn_filt_hb_3_fx[i] = shl(f-&gt;state_syn_filt_hb_3_fx[i],shl_fac); 
               
               
                  for(i=0;i&lt;2;i++) 
               
               
                  f-&gt;mem_syn_filt_hb_iir1_fx[i] = L_shl(f-&gt;mem_syn_filt_hb_iir1_fx[i],shl_fac); 
               
               
                  for(i=0;i&lt;2;i++) 
               
               
                  f-&gt;mem_syn_filt_hb_iir2_fx[i] = L_shl(f-&gt;mem_syn_filt_hb_iir2_fx[i],shl_fac); 
               
               
                 } 
               
               
                 void find_max_synhb_struc(struct STATE_SYN_HB_fx *f,Word16 *struc_norm) 
               
               
                 { 
               
               
                  Word16 i; 
               
               
                  Word16 max=0; 
               
               
                  Word32 max32=0,temp=0; 
               
               
                  Word16 n,n1; 
               
               
                  for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP;i++) 
               
               
                  max=MAX_FX(abs_s(f-&gt;state_syn_filt_hb_0_fx[i]),max); 
               
               
                  for(i=0;i&lt;2*ALLPASSSECTIONS_STEEP;i++) 
               
               
                  max=MAX_FX(abs_s(f-&gt;state_syn_filt_hb_1_fx[i]),max); 
               
               
                  for(i=0;i&lt;LEN_UP_HB;i++) 
               
               
                  max=MAX_FX(abs_s(f-&gt;state_syn_filt_hb_2_fx[i]),max); 
               
               
                  for(i=0;i&lt;2*ALLPASSSECTIONS+1;i++) 
               
               
                  max=MAX_FX(abs_s(f-&gt;state_syn_filt_hb_3_fx[i]),max); 
               
               
                  for(i=0;i&lt;2;i++) 
               
               
                  { 
               
               
                   if (f-&gt;mem_syn_filt_hb_iir1_fx[i] &lt; 0) 
               
               
                    temp = L_negate(f-&gt;mem_syn_filt_hb_iir1_fx[i]); 
               
               
                   else 
               
               
                    temp = f-&gt;mem_syn_filt_hb_iir1_fx[i]; 
               
               
                   max32=MAX_FX(temp,max32); 
               
               
                  } 
               
               
                 for(i=0;i&lt;2;i++) 
               
               
                  { 
               
               
                   if (f-&gt;mem_syn_filt_hb_iir2_fx[i] &lt; 0) 
               
               
                    temp = L_negate(f-&gt;mem_syn_filt_hb_iir2_fx[i]); 
               
               
                   else 
               
               
                    temp = f-&gt;mem_syn_filt_hb_iir2_fx[i]; 
               
               
                   max32=MAX_FX(temp,max32); 
               
               
                  } 
               
               
                  n=norm_s(max); 
               
               
                  n =(max==0)?15:n; 
               
               
                  n1=norm_1(max32); 
               
               
                  n1 =(max32==0)?31:n1; 
               
               
                  *struc_norm = sub(MIN_FX(n,n1),3); //1 bit head room 
               
               
                 } 
               
               
                   
               
            
           
         
       
     
       FIG. 5  illustrates a wideband decoder  560 . The wideband decoder  560  may be implemented in an apparatus that may be utilized within a wireless communication system  100 . The apparatus may be a mobile phone, a personal digital assistant (PDA), a laptop computer, a digital camera, a music player, a game device, or any other device with a processor. The apparatus may function as a mobile station  102  or a base station  104  within a wireless communication system  100 . 
     An encoded low band signal  524  (or  224 ) may be provided to the wideband decoder  560 . The wideband decoder  560  may include a low band decoder  562 . The low band decoder  562  may decode the encoded low band signal  524 , thereby obtaining a decoded low band signal  518 . The low band decoder  562  may also output a low band excitation signal  526 . 
     An encoded high band signal  530  (or  230 ) may also be provided to the wideband decoder  560 . The wideband decoder  560  may include a high band decoder  564 . The encoded high band signal  530  may be provided to the high band decoder  564 . The low band excitation signal  526  that is output by the low band decoder  562  may also be provided to the high band decoder  564 . The high band decoder  564  may decode the encoded high band signal  530  according to information in the low band excitation signal  526 , thereby obtaining a decoded high band signal  520 . 
     The wideband decoder  560  may also include a synthesis filter bank  516 . The decoded low band signal  518  that is output by the low band decoder  562  and the decoded high band signal  520  that is output by the high band decoder  564  may be provided to the synthesis filter bank  516 . The synthesis filter bank  516  may combine the decoded low band signal  518  and the decoded high band signal  520  to produce a wideband speech signal  514 . 
     The high band decoder  564  may include some of the identical components that were described above in connection with the high band encoder  228 . For example, the high band decoder  564  may include the high band excitation generator  332 , the signal normalizer  338 , the filter states normalization factor adjuster  340 , and the factor determination component  346 . (These components are not shown in  FIG. 5 .) The operation of these components may be similar or identical to the operation of the corresponding components that were described above in relation to the high band encoder  228 . Thus, the techniques described above for dynamic normalization of the low band excitation signal  226  in the context of a wideband encoder  212  may also be applied to the low band excitation signal  526  that is shown in  FIG. 5  in the context of a wideband decoder  560 . 
       FIG. 6  illustrates a method  600  for dynamic normalization to reduce loss in precision for low-level signals. The method  600  may be implemented by a wideband encoder  212  within a mobile station  102  or a base station  104  within a wireless communication system  100 . Alternatively, the method  600  may be implemented by a wideband decoder  560  within a mobile station  102  or a base station  104  within a wireless communication system  100 . 
     In accordance with the method  600 , a current frame of a low band excitation signal  226  may be received  602 . A normalization factor  344  for the current frame of the low band excitation signal  226  may be determined  604 . The normalization factor  344  may depend on the amplitude of the current frame of the low band excitation signal  226 . The normalization factor  344  may also depend on the values of filter states  342  of a high band excitation generator  332  after filtering operations were performed on a previous frame of a normalized low band excitation signal  336 . 
     The current frame of the low band excitation signal  226  may be normalized  606  based on the normalization factor  344  that is determined  604 . In addition, the normalization factor of the filter states of the high band excitation generator  332  may be adjusted  608  based on the normalization factor  344  that is determined  604 . 
       FIG. 7  illustrates a method  700  for determining a normalization factor  344   a  for the current frame of the low band excitation signal  226 . (The reference number  344   a  refers to the normalization factor  344   a  for the current frame, and the reference number  344   b  refers to the normalization factor  344   b  for the previous frame.) The method  700  may be implemented by a wideband encoder  212  within a mobile station  102  or a base station  104  within a wireless communication system  100 . Alternatively, the method  700  may be implemented by a wideband decoder  560  within a mobile station  102  or a base station  104  within a wireless communication system  100 . 
     In accordance with the method  700 , an optimal value  450  for the normalization factor  344   a  for the current frame of the low band excitation signal  226  may be determined  702 . The optimal value  450  for the normalization factor  344   a  may indicate to what extent the bits of the current frame of the low band excitation signal  226  may be left-shifted without causing saturation. 
     A scaling factor  454  for the filter states  342  of the high band excitation generator  332  may be determined  704 . The scaling factor  454  may indicate to what extent the bits of the filter states  342  may be left-shifted without causing saturation. 
     A saturation condition may be evaluated  706 . The saturation condition may depend on the optimal value  450  for the normalization factor  344   a  for the current frame of the low band excitation signal  226 . The saturation condition may also depend on the scaling factor  454  for the filter states  342  of the high band excitation generator  332 . The saturation condition may also depend on the normalization factor  344   b  for the previous frame of the low band excitation signal  226 . 
     If it is determined  706  that the saturation condition is not satisfied, this may be interpreted to mean that setting the normalization factor  344  equal to the optimal value  450  that was determined  702  is not going to cause saturation. Accordingly, the normalization factor  344  for the current frame of the low band excitation signal  226  may be set  708  equal to the optimal value  450  that was determined  702 . 
     If it is determined  706  that the saturation condition is satisfied, this may be interpreted to mean that setting the normalization factor  344  equal to the optimal value  450  that was determined  702  is going to cause saturation. Accordingly, the normalization factor  344   a  for the current frame of the low band excitation signal  226  may be set  710  equal to prev_Qinp+Q_states. As discussed above, the term prev_Qinp may refer to the normalization factor  344   b  for the previous frame of the low band excitation signal  226 . The term Q_states may refer to the scaling factor for the filter states  342 . 
       FIG. 8  illustrates various components that may be utilized in a communications device  801 . The communications device  801  may include a processor  803  which controls operation of the device  801 . The processor  803  may also be referred to as a CPU. Memory  805 , which may include both read-only memory (ROM) and random access memory (RAM), provides instructions and data to the processor  803 . A portion of the memory  805  may also include non-volatile random access memory (NVRAM). 
     The communications device  801  may also include a housing  809  that may include a transmitter  811  and a receiver  813  to allow transmission and reception of data between the communications device  801  and a remote location. The transmitter  811  and receiver  813  may be combined into a transceiver  815 . An antenna  817  may be attached to the housing  809  and electrically coupled to the transceiver  815 . 
     The communications device  801  may also include a signal detector  807  that may be used to detect and quantify the level of signals received by the transceiver  815 . The signal detector  807  may detect such signals as total energy, pilot energy per pseudonoise (PN) chips, power spectral density, and other signals. 
     A state changer  819  of the communications device  801  may control the state of the communications device  801  based on a current state and additional signals received by the transceiver  815  and detected by the signal detector  807 . The device  801  may be capable of operating in any one of a number of states. The communications device  801  may also include a system determinator  821  that may be used to control the device  801  and to determine which service provider system the device  801  should transfer to when it determines the current service provider system is inadequate. 
     The various components of the communications device  801  may be coupled together by a bus system  823  which may include a power bus, a control signal bus, and a status signal bus in addition to a data bus. However, for the sake of clarity, the various busses are illustrated in  FIG. 8  as the bus system  823 . The communications device  801  may also include a digital signal processor (DSP)  825  for use in processing signals. 
     Information and signals may be represented using any of a variety of different technologies and techniques. For example, data, instructions, commands, information, signals and the like that may be referenced throughout the above description may be represented by voltages, currents, electromagnetic waves, magnetic fields or particles, optical fields or particles or any combination thereof. 
     The various illustrative logical blocks, modules, circuits, methods, and algorithm steps disclosed herein may be implemented in hardware, software, or both. To clearly illustrate this interchangeability of hardware and software, various illustrative components, blocks, modules, circuits and steps have been described above generally in terms of their functionality. Whether such functionality is implemented as hardware or software depends upon the particular application and design constraints imposed on the overall system. Skilled artisans may implement the described functionality in varying ways for each particular application, but such implementation decisions should not be interpreted as limiting the scope of the claims. 
     The various illustrative logical blocks, modules and circuits described above may be implemented or performed with a general purpose processor, a digital signal processor (DSP), an application specific integrated circuit (ASIC), a field programmable gate array signal (FPGA) or other programmable logic device, discrete gate or transistor logic, discrete hardware components or any combination thereof designed to perform the functions described herein. A general purpose processor may be a microprocessor, but in the alternative, the processor may be a controller, microcontroller or state machine. A processor may also be implemented as a combination of computing devices, e.g., a combination of a DSP and a microprocessor, a plurality of microprocessors, one or more microprocessors in conjunction with a DSP core or any other such configuration. 
     The methods disclosed herein may be implemented in hardware, in software, or both. Software may reside in any form of storage medium that is known in the art. Some examples of storage media that may be used include RAM memory, flash memory, ROM memory, EPROM memory, EEPROM memory, registers, a hard disk, a removable disk, an optical disk, and so forth. Software may comprise a single instruction, or many instructions, and may be distributed over several different code segments, among different programs and across multiple storage media. A storage medium may be coupled to a processor such that the processor can read information from, and write information to, the storage medium. In the alternative, the storage medium may be integral to the processor. 
     The methods disclosed herein may comprise one or more steps or actions for achieving the described method. The method steps and/or actions may be interchanged with one another without departing from the scope of the claims. In other words, unless a specific order of steps or actions is specified, the order and/or use of specific steps and/or actions may be modified without departing from the scope of the claims. 
     While specific features, aspects, and configurations have been illustrated and described, it is to be understood that the claims are not limited to the precise configuration and components illustrated above. Various modifications, changes, and variations may be made in the arrangement, operation and details of the features, aspects, and configurations described above without departing from the scope of the claims.