Abstract:
A low complexity, robust speech/music classifier useful for performing frame loss concealment (FLC) in an audio decoder or for other applications are described. The classifier uses a feature set that is to a large extent common to audio decoders and frame loss concealment (FLC) systems. The classifier uses running averages to eliminate feature set buffering. Classification is performed using a simple parameter normalization and sum scheme that consumes an insignificant number of processor cycles and an insignificant amount of memory. An advanced energy tracker is integrated within the classifier to provide robust classification in the presence of noise.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
       [0001]    This application claims priority to provisional U.S. Patent Application No. 60/835,094, filed Aug. 3, 2006, the entirety of which is incorporated by reference herein. 
     
     BACKGROUND OF THE INVENTION 
       [0002]    1. Field of the Invention 
         [0003]    The present invention relates to digital communication systems. More particularly, the present invention relates to methods for classifying audio signals in a digital communications system as speech or non-speech. 
         [0004]    2. Background Art 
         [0005]    Humans can easily discriminate between speech and music signals. Many different solutions have been proposed for performing this discrimination automatically. An important part of any such solution is the selection of features used in the discrimination process. The choice is based on a priori knowledge of the characteristics of the signals for classification and a set of features that are capable of separating the signals. In one approach, features are chosen which focus on the choice between music/non-music as proposed by Scheirer and Slaney in “Construction and Evaluation of a Robust Multifeature Speech/Music Discriminator”, ICASSP&#39;97, Munich, pp. 1331-1334 in which the variation of the spectral magnitude is used to detect harmonic continuity. On the other hand, the discrimination can be the choice between speech/non-speech as by Gauvin et al. in “Audio partitioning and transcription for broadcast data indexation,” CBMI&#39;99 Toulouse, pp. 67-73 which focuses on cepstral features. Other features proposed include Line Spectral Frequencies (LSFs), zero-crossing rates, pitch, and energy. 
         [0006]    Several applications have been developed that can benefit from automatic discrimination of speech and music signals. In the application of automatic speech recognition for broadcast news, it is necessary to turn off the speech recognition during non-speech segments. In the emerging application of video search, one approach is to index it according to the audio content. Users are aided in their search by a speech/music classification. Yet another application is in the field of audio coding where upfront classification of the input into either speech/music can be used to switch between speech or music based modules of the algorithm. Indeed, this approach is used in the parametric coder of the MPEG4 standard. Just as in coding the original signal, the concealment of frame loss can also benefit from the knowledge of whether the signal is speech or music. In U.S. patent Ser. No. 11/285,311 to Chen, entitled “Classification-Based Frame Loss Concealment for Audio Signals”, different speech/music FLC methods are used based on a classification of the input signal. 
         [0007]    An important consideration in many applications is the complexity and cost of the solution. With respect to the feature set, it is desirable to use features that are potentially already available. For example, if the classification is used with an audio coder or an FLC module, the feature set can be extracted from there. Typically, the feature set is estimated over an interval of 0.5-5 s and can consume considerable data storage. The classification framework can also add significant complexity. Common frameworks include multidimensional Gaussian maximum a posteriori (MAP) estimators, Gaussian mixture model classification, spatial partitioning schemes based on k-d trees, and nearest-neighbor classification. These methods can be very complex in terms of cycles, data memory, table space, and program memory. 
         [0008]    One factor that can significantly degrade performance of the classifier is the presence of noise. In particular, noisy speech produces features that behave increasingly like music at lower SNRs. This can cause significant system performance degradation. Prior art solutions do not seem to address this. 
       SUMMARY OF THE INVENTION 
       [0009]    The present invention is a low complexity, robust speech/music classifier. The classifier uses a set of parameters that is to a large extent common to audio decoders and frame loss concealment (FLC) systems. In an embodiment, the complete feature set may be extracted with minimal additional processing from a system such as that described in U.S. patent application Ser. No. 11/234,291 to Chen, filed Sep. 26, 2005 and entitled “Packet Loss Concealment for Block-Independent Speech Codecs”, the entirety of which is incorporated by reference herein. In an embodiment, running averages are used to eliminate feature set buffering. In a further embodiment, classification is performed using a simple parameter normalization and sum scheme that consumes an insignificant number of processor cycles and an insignificant amount of memory. In a still further embodiment, an advanced energy tracker is integrated within the classifier to provide robust classification in the presence of noise. 
         [0010]    Further features and advantages of the invention, as well as the structure and operation of various embodiments of the invention, are described in detail below with reference to the accompanying drawings. It is noted that the invention is not limited to the specific embodiments described herein. Such embodiments are presented herein for illustrative purposes only. Additional embodiments will be apparent to persons skilled in the relevant art(s) based on the teachings contained herein. 
     
     
       BRIEF DESCRIPTION OF THE DRAWINGS/FIGURES 
         [0011]    The accompanying drawings, which are incorporated herein and form a part of the specification, illustrate one or more embodiments of the present invention and, together with the description, further serve to explain the purpose, advantages, and principles of the invention and to enable a person skilled in the art to make and use the invention. 
           [0012]      FIG. 1  illustrates an audio decoding system that performs classification-based frame loss concealment (FLC) system in accordance with an embodiment of the present invention. 
           [0013]      FIG. 2  illustrates a flowchart of a method for performing classification-based FLC in an audio decoding system in accordance with an embodiment of the present invention. 
           [0014]      FIG. 3  illustrates a flowchart of a method for determining which of a plurality of FLC methods to apply when a signal classifier has identified an input signal as speech in accordance with an embodiment of the present invention. 
           [0015]      FIG. 4  illustrates a flowchart of a method for determining which of a plurality of FLC methods to apply when a signal classifier has identified an input signal as music in accordance with an embodiment of the present invention. 
           [0016]      FIG. 5  illustrates a flowchart of a method for performing frame-repeat based FLC for music-like signals in accordance with an embodiment of the present invention. 
           [0017]      FIG. 6  illustrates a first portion of a flowchart of a method for performing FLC for speech signals in accordance with an embodiment of the present invention. 
           [0018]      FIG. 7  illustrates a second portion of a flowchart of a method for performing FLC for speech signals in accordance with an embodiment of the present invention. 
           [0019]      FIG. 8  is a block diagram of a speech/non-speech classifier in accordance with an embodiment of the present invention. 
           [0020]      FIG. 9  shows a flowchart providing example steps for tracking energy of an audio signal, according to embodiments of the present invention. 
           [0021]      FIG. 10  shows an example block diagram of an energy tracking module, in accordance with an embodiment of the present invention. 
           [0022]      FIG. 11  shows a flowchart providing example steps for analyzing features of an audio signal, according to embodiments of the present invention. 
           [0023]      FIG. 12  shows an example block diagram of an audio signal feature extraction module, in accordance with an embodiment of the present invention. 
           [0024]      FIG. 13  shows a flowchart providing example steps for normalizing audio signal features, according to embodiments of the present invention. 
           [0025]      FIG. 14  shows an example block diagram of a normalization module, in accordance with an embodiment of the present invention. 
           [0026]      FIG. 15  shows a flowchart providing example steps for classifying audio signals as speech or music, according to embodiments of the present invention. 
           [0027]      FIG. 16  shows a flowchart providing example steps for overlapping first and second decomposed signals, according to embodiments of the present invention. 
           [0028]      FIG. 17  shows a system configured to overlap first and second decomposed signals, according to an example embodiment of the present invention. 
           [0029]      FIG. 18  shows a flowchart providing example steps for overlapping a decomposed signal with a non-decomposed signal, according to embodiments of the present invention. 
           [0030]      FIG. 19  shows a system configured to overlap a decomposed signal with a non-decomposed signal, according to an example embodiment of the present invention. 
           [0031]      FIG. 20  shows a flowchart providing example steps for overlapping a mixed first signal with a mixed second signal, according to an embodiment of the present invention. 
           [0032]      FIG. 21  shows a system configured to overlap a mixed first signal with a mixed second signal, according to an example embodiment of the present invention. 
           [0033]      FIG. 22  shows a flowchart providing example steps for determining a pitch period of an audio signal, according to an example embodiment of the present invention. 
           [0034]      FIG. 23  shows block diagram of a pitch refinement system, in accordance with an example embodiment of the present invention. 
           [0035]      FIG. 24  shows a flowchart for performing a decimated bisectional search, according to an example embodiment of the present invention. 
           [0036]      FIGS. 25A-25D  show plots related to an example determination of a pitch period, in accordance with an embodiment of the present invention. 
           [0037]      FIG. 26  is a block diagram of a computer system in which embodiments of the present invention may be implemented. 
       
    
    
       [0038]    The features and advantages of the present invention will become more apparent from the detailed description set forth below when taken in conjunction with the drawings, in which like reference characters identify corresponding elements throughout. In the drawings, like reference numbers generally indicate identical, functionally similar, and/or structurally similar elements. The drawing in which an element first appears is indicated by the leftmost digit(s) in the corresponding reference number. 
       DETAILED DESCRIPTION OF INVENTION 
     A. IMPROVED CLASSIFICATION-BASED FLC SYSTEM AND METHOD IN  ACCORDANCE WITH AN EMBODIMENT OF THE PRESENT INVENTION 
       [0039]      FIG. 1  illustrates an audio decoding system  100  that performs classification-based frame loss concealment (FLC) in accordance with an embodiment of the present invention. As shown in  FIG. 1 , audio decoding system  100  includes an audio decoder  110 , a decoded signal buffer  120 , a signal classifier  130 , FLC decision/control logic  140 , first and second FLC method selection switches  150  and  170 , FLC processing blocks  161  and  162 , and an output signal selection switch  180 . As will be readily appreciated by persons skilled in the relevant art(s), each of the elements of system  100  may be implemented as software, as hardware, or as a combination of software and hardware. In one embodiment of the present invention, each of the elements of system  100  is implemented as a series of software instructions that, when executed by a digital signal processor (DSP), perform the functions of that element as described herein. 
         [0040]    In general, audio decoding system  100  operates to decode each of a series of frames of an input audio bit-stream into corresponding frames of an output audio signal. System  100  decodes the input audio bit-stream one frame at a time. As used herein, the term “current frame” refers to a frame of the input audio bit-stream that system  100  is currently decoding, whereas “previous frame” refers to a frame of the input audio bit-stream that system  100  has already decoded. As also used herein, the term “decoding” may include both normal decoding of a received frame of the input audio bit-stream into corresponding output audio signal samples as well as generating output audio signal samples for a lost frame of the input audio bit-stream using an FLC technique. The function of each of the components of system  100  will now be described in more detail. 
         [0041]    If a current frame of the input audio bit-stream is deemed received, audio decoder  110  decodes the current frame using any of a variety of known audio decoding techniques to generate output audio signal samples. Output signal selection switch  180  is controlled by a lost frame indicator, which indicates whether the current frame of the input audio bit-stream is deemed received or is lost. If the current frame is deemed received, switch  180  is placed in the upper position shown in  FIG. 1  (connected to the node labeled “Frame Received”) and the decoded audio signal at the output of audio decoder  110  is used as the output audio signal for the current frame. Additionally, if the current frame is deemed received, the decoded audio signal for the current frame is also stored in decoded signal buffer  120  in preparation for possible FLC operations for future frames. 
         [0042]    In contrast, if the current frame of the input audio bit-stream is deemed lost, then output signal selection switch  180  is placed in the lower position shown in  FIG. 1  (connected to the node labeled “Frame Lost”). In this case, signal classifier  130  and FLC decision/control logic  140  operate together to select one of two possible FLC methods to perform the necessary FLC operations. 
         [0043]    As shown in  FIG. 1 , there are two possible FLC methods that audio decoding system  100  can use. These two possible FLC methods are implemented in first and second processing blocks  161  and  162 , respectively, in  FIG. 1 . In one embodiment of the invention, processing block  161  (labeled “First FLC Method”) is designed or tuned to perform FLC for an audio signal that has been classified as speech, while processing block  162  (labeled “Second FLC Method”) is designed or tuned to perform FLC for an audio signal that has been classified as music. 
         [0044]    The function of signal classifier  130  is to analyze the previously-decoded audio signal stored in decoded signal buffer  120 , or a portion thereof, in order to determine whether the current frame should be classified as speech or music. There are several approaches discussed in the related art that are appropriate for performing this function. In one embodiment, a signal classifier  130  is used that shares a feature set with one or both of the incorporated FLC methods of processing blocks  161  and  162  to reduce complexity. 
         [0045]    FLC decision/control logic  140  selects the FLC method for the current frame based on a classification output from signal classifier  130  and other decision logic. FLC decision/control logic selects the FLC method by generating a signal (labeled “FLC Method Decision” in  FIG. 1 ) that controls the operation of first and second FLC method selection switches  150  and  170  to apply either the FLC method of processing block  161  or the FLC method of processing block  162 . In the particular example shown in  FIG. 1 , switches  150  and  170  are in the uppermost position so that the FLC method of processing block  161  is selected. Of course, this is just an example. For a different frame that is lost, FLC decision/control logic  140  may select the FLC method of processing block  162 . 
         [0046]    If signal classifier  130  classifies the input signal as speech, FLC decision/control logic  140  performs further logic and analysis to determine which FLC technique to use. In one example implementation, signal classifier  130  passes FLC decision/control logic  140  a feature set used in performing speech classification. FLC decision/control logic  140  then uses this information along with the knowledge of the FLC algorithms to determine which FLC method would perform best for the current frame. 
         [0047]    Once a particular FLC method is selected, this FLC method uses the previously-decoded audio signal, or some portion thereof, stored in decoded signal buffer  120  and performs the associated FLC operations. The resulting output signal is then routed through switches  170  and  180  and becomes the output audio signal for the audio decoding system  100 . Note that although it is not depicted in  FIG. 1  for the sake of simplicity, it is understood and generally advisable that the FLC audio signal picked up by switch  170  is also passed back to decoded signal buffer  120  so that the audio signal produced by the selected FLC method for the current lost frame is also stored as the newest portion of the “previously-decoded audio signal.” This is done to prepare decoded signal buffer  120  for the next frame in case the next frame is also lost. In other words, it is generally advantageous for decoded signal buffer  120  to store the audio signal corresponding to the last frame immediately processed before a lost frame, whether or not the audio signal was produced by audio decoder  110  or one of FLC processing blocks  161  or  162 . 
         [0048]    Persons skilled in the relevant art(s) will readily appreciate that the placing of switches  150 ,  170  and  180  in an upper or lower position as described herein is not necessarily meant to denote the operation of a mechanical switch, but rather to describe the selection of one of two logical processing paths within system  100 . 
         [0049]      FIG. 2  illustrates a flowchart  200  of a method for performing classification-based FLC in an audio decoding system in accordance with an embodiment of the present invention. The method of flowchart  200  will be described with continuing reference to audio decoding system  100  of  FIG. 1 , although persons skilled in the relevant art(s) will appreciate that the invention is not limited to that implementation. 
         [0050]    As shown in  FIG. 2 , the beginning of flowchart  200  is indicated at step  202  labeled “start”. Processing immediately proceeds to step  204 , in which a decision is made as to whether the next frame of the input audio bit-stream to be received by audio decoder  110  is received or lost. If the frame is deemed received, then audio decoder  110  performs normal decoding operations on the received frame to generate corresponding decoded audio signal samples, as shown at step  206 . Processing then proceeds to step  208  in which the decoded audio signal corresponding to the received frame is stored in decoded signal buffer  120 . 
         [0051]    At step  210 , a determination is made whether or not this is the first good frame after erasure or loss. If it is, then a portion of the frame and an extrapolated signal provided by one of FLC processing blocks  161  or  162  are overlap-added, as shown in step  212 . In an embodiment, a “ramp up” operation is also performed for the first good frame. The overlap-add and ramp up operations will be described in more detail below in reference to the operation of processing blocks  161  and  162 . 
         [0052]    The decoded audio signal is then provided as the output audio signal of audio decoding system  100 , as shown at step  214 . With reference to  FIG. 1 , this is achieved through the operation of output signal selection switch  180  (under the control of the lost frame indicator) to couple the output of audio decoder  110  to the ultimate output of system  100 . Processing then proceeds to step  216 , where it is determined whether or not there are more frames in the input audio bit-stream to be processed by audio decoding system  100 . If there are more frames, then processing returns to decision step  204 ; otherwise, processing ends as shown at step  236  labeled “end”. 
         [0053]    Returning to decision step  204 , if it is determined that the next frame in the input audio bit-stream is lost, then processing proceeds to step  220 , in which signal classifier  130  analyzes at least a portion of the previously decoded audio signal stored in decoded signal buffer  120 . Based on this analysis, signal classifier  130  classifies the input signal as either speech or music as shown at step  222 . Several approaches have been discussed in the related art that are appropriate for performing this function. In an embodiment of the invention, a classifier is used that shares a feature set with one or both of the incorporated FLC methods of processing blocks  161  and  162  to reduce complexity. 
         [0054]    If it is determined in step  222  that the input signal is speech, then FLC decision/control logic  140  performs further logic and analysis to determine which FLC method to apply. In one embodiment, signal classifier  130  passes FLC decision/control logic a feature set used in the speech classification. FLC decision/control logic  140  then uses this information along with knowledge of the FLC algorithms to determine which FLC method would perform best for the current frame. For example, the input signal might be speech with background music and although the predominant signal is speech, there still may be localized frames for which the FLC method designed for music is most suitable. If the FLC method designed for speech is deemed most suitable, the flow continues to step  226 , in which the FLC method designed for speech is applied. However, if the FLC method designed for music is selected, the flow crosses over to step  230  and that method is applied. Likewise, if it is determined in step  222  that the input signal is music, FLC decision/control logic  140  then decides which FLC method is most suitable for the current frame, as shown at step  228 , and then the selected method is applied. For example, the input signal may be music with vocals and, even though signal classifier  130  has classified the input signal as music, there may be a strong vocal element such that the FLC method designed for speech will provide the best results. 
         [0055]    With reference to  FIG. 1 , the selection of the FLC method by FLC decision/control logic  140  is performed via the generation of the signal labeled “FLC Method Decision”, which controls FLC method selection switches  150  and  170  to select one of the processing blocks  161  or  162 . 
         [0056]    In an embodiment, FLC decision/control logic  140  also uses logic/analysis to control or modify the FLC algorithms. In accordance with such an embodiment, if signal classifier  130  classifies the input signal as speech, and further analysis has a high confidence in the ability of the FLC method designed for speech to conceal the loss of the current frame, then the FLC method designed for speech is selected and left unmodified. However, if further analysis shows that the signal is not very periodic, or that there are indications of some background music, etc., the speech FLC may be selected, but some part of the algorithm may be modified. 
         [0057]    For example, if the speech FLC is Periodic Waveform Extrapolation (PWE) based, an effective modification is to use a pitch multiple (double, triple, etc.) for extrapolation. If the signal is speech, using a pitch multiple will still produce an in-phase extrapolation. If the signal is music, using the pitch multiple increases the repetition period and the method becomes more like a frame-repeat method, which has been shown to provide good FLC performance for music signals. 
         [0058]    Modifications can also be performed on the FLC method designed for music. For example, if signal classifier  130  classifies the input signal as speech, but FLC decision/control logic  140  selects the FLC method designed for music, the FLC method designed for music may be modified to be more appropriate for speech. For example, the signal can be analyzed for the degree of mix between periodic and noise-like components in a manner similar to that described in U.S. patent application Ser. No. 11/234,291 to Chen (explaining the calculation of a “voicing measure”), the entirety of which has been incorporated by reference herein. The output of the FLC method designed for music can then be mixed with a speech-like derived (LPC analysis) noise signal. 
         [0059]    After either the FLC method designed for speech has been applied at step  226  or the FLC method designed for music has been applied at step  230 , the audio signal generated by application of the selected FLC method is then provided as the output audio signal of audio decoding system  100 , as shown at step  232 . In the implementation shown in  FIG. 1 , this is achieved through the operation of output signal selection switch  180  (under the control of the lost frame indicator) to couple the output at switch  170  to the ultimate output of system  100 . The audio signal generated by application of the selected FLC method is also stored in decoded signal buffer  120  as shown in step  234 . Processing then proceeds to step  216 , where it is determined whether or not there are more frames in the input audio bit-stream to be processed by audio decoding system  100 . If there are more frames, then processing returns to decision step  204 ; otherwise, processing ends at step  236  labeled “end”. 
         [0060]      FIG. 3  illustrates a flowchart  300  of one method that may be used by FLC decision/control logic  140  for determining which FLC method to apply when signal classifier  130  has identified the input signal as speech. This method utilizes a feature set provided by signal classifier  130 , which includes a single speech likelihood measure for the current frame, denoted SLM, and a long-term running average of the speech likelihood measure, denoted LTSLM. The derivation of each of these values is described in Section B below. As discussed in that section, SLM is in the range {−4,+4}, wherein values close to the minimum or maximum indicate the likelihood of speech, while values close to zero indicate the likelihood of music or other non-speech signals. The method also uses values of SLM associated with previously-decoded frames, which may be stored and subsequently accessed in a local buffer. 
         [0061]    As shown in  FIG. 3 , the beginning of flowchart  300  is indicated by step  302  labeled “start”. Processing immediately proceeds to step  304 , in which a dynamic threshold for SLM is determined based on LTSLM. In one implementation, this step is carried out by setting the dynamic threshold to −4 if LTSLM is greater than 2.18, and otherwise setting the dynamic threshold to (1.8/LTSLM) 3  if LTSLM is less than or equal to 2.18. This has the effect of eliminating the dynamic threshold for signals that exhibit a strong long-term tendency for speech, while setting the dynamic threshold to a value that is inversely proportional to LTSLM for signals that do not. As will be made evident below, the higher the dynamic threshold is set, the less likely it is that the method of flowchart  300  will select the FLC method designed for speech. 
         [0062]    At step  306 , a first series of tests are performed to determine if the FLC method designed for speech should be applied. These tests may include determining if SLM, and/or the absolute value thereof, exceeds a certain threshold, if the sum total of one or more SLM values associated with prior frames exceeds certain thresholds, and/or if a pitch prediction gain associated with the last good frame is large. If true, this last condition would indicate that the frame is very periodic at the detected pitch period and that an FLC method designed for speech would work well. If the results of these tests indicate that the FLC method designed for speech should be applied, then processing proceeds via decision step  308  to step  310 , wherein the FLC method designed for speech is selected. 
         [0063]    In one implementation, the series of tests applied in step  306  include (1) determining if the absolute value of SLM is greater than 1.8; (2) determining if SLM is greater than the dynamic threshold set in step  304  AND if the one of the following is true: the sum of the SLM values associated with the two preceding frames is greater than 3.4 OR the sum of the SLM values associated with the three preceding frames is greater than 4.8 OR the sum of the SLM values associated with the four preceding frames is greater than 5.6 OR the sum of the SLM values associated with the five preceding frames is greater than 7; (3) determining if the sum of the SLM values associated with the two preceding frames is less than −3.4; (4) determining if the sum of the SLM values associated with the three preceding frames is less than −4.8; (5) determining if the sum of the SLM values associated with the four preceding frames is less than −5.6; (6) determining if the sum of the SLM values associated with the five preceding frames is less than −7; and (7) determining if the pitch prediction gain associated with the last good frame is greater than 6. If any one of tests (1)-(7) is passed (the condition is evaluated as true), then speech is indicated and the FLC method designed for speech is selected. 
         [0064]    After the FLC method designed for speech has been selected at step  310 , additional tests are performed to see if the pitch period should be doubled prior to application of the FLC method. First, a series of tests are applied to determine if the speech classification is a borderline one as shown at step  312 . This series of tests may include determining if SLM is less than a certain threshold and/or determining if LTSLM is less than a certain threshold. For example, in one implementation, these additional tests include determining if SLM is less than 1.4 and if LTSLM is less than 2.4. If either of these conditions is evaluated as true, then a borderline classification is indicated and processing proceeds via decision step  314  to decision step  316 . Otherwise, the pitch period is not doubled and processing ends at step  328  labeled “end.” 
         [0065]    At decision step  316 , the pitch prediction gain is compared to a threshold value to determine how periodic the current frame is. If the pitch prediction gain is low, this indicates that the frame has very little periodicity. In one implementation, this step includes determining if the pitch prediction gain is less than 0.3. If decision step  316  determines that the frame has very little periodicity, then processing proceeds to step  318 , in which the pitch period is doubled prior to application of the FLC method designed for speech, after which processing ends as shown at step  328 . Otherwise, the pitch period is not doubled and processing ends at step  328 . 
         [0066]    Returning now to decision step  308 , if the series of tests applied during step  306  do not indicate speech, then processing proceeds to decision step  320 . In decision step  320 , SLM is compared to a threshold value to determine if there is at least some indication that the current frame is voiced speech or periodic. If the comparison provides such an indication, then processing proceeds to step  322 , wherein the FLC method designed for speech is selected. In one implementation, decision step  308  includes determining if SLM is greater than 1.5. 
         [0067]    After the FLC method designed for speech has been selected at step  322 , a determination is made as to whether there are at least two pitch periods in the current frame. In one implementation, this is achieved by determining if the frame size divided by the pitch period is greater than two. If there are at least two pitch periods in the current frame, then the pitch period is doubled prior to application of the FLC method designed for speech as shown at step  318 , after which processing ends as shown at step  328 . Otherwise, the pitch period is not doubled and processing ends at step  328 . 
         [0068]    Returning now to decision step  320 , if the test applied in that step does not provide at least some indication that the current frame is voiced speech or periodic, then processing proceeds to step  326 , in which the FLC method designed for music is selected. After this, processing ends at step  328 . 
         [0069]      FIG. 4  illustrates a flowchart  400  of one method that may be used by FLC decision/control logic  140  for determining which FLC method to apply when signal classifier  130  has identified the input signal as music. Like the method described above in reference to flowchart  300  of  FIG. 3 , this method utilizes a feature set provided by signal classifier  130 , which includes a single speech likelihood measure for the current frame, denoted SLM, and a long-term running average of the speech likelihood measure, denoted LTSLM. The method also uses values of SLM associated with previously-decoded frames, which may be stored and subsequently accessed in a local buffer. 
         [0070]    As shown in  FIG. 4 , the beginning of flowchart  400  is indicated by step  402  labeled “start”. Processing immediately proceeds to step  404 , in which a dynamic scaling factor is determined based on LTSLM. In one implementation, the dynamic scaling factor is set to a value that is inversely proportional to LTSLM. For example, in one implementation, the dynamic scaling factor is set to  1 . 81 LTSLM. As will be made evident below, the higher the scaling factor, the less likely that the FLC method designed for speech will be selected. 
         [0071]    At step  404 , a series of tests are performed to detect speech in music and thereby determine if the FLC method designed for speech should be applied. These tests may include determining if SLM exceeds a certain threshold, if the sum total of one or more SLM values associated with prior frames exceeds certain thresholds, or a combination of both. If the results of these tests indicate speech in music, then processing proceeds via decision step  408  to step  410 , wherein the FLC method designed for speech is selected. Processing then ends as shown at step  422  denoted “end”. 
         [0072]    In one implementation, the series of tests performed in step  406  include (1) determining if SLM is greater than 1.8 times the scaling factor determined in step  404  and (2) determining if the sum of the SLM values associated with the three preceding frames is greater than 5.4 times the scaling factor determined in step  404  OR if the sum of the SLM values associated with the four preceding frames is greater than 7.2 times the scaling factor determined in step  404 . If both tests (1) and (2) are passed (the conditions are evaluated as true), then speech in music is indicated. 
         [0073]    Returning now to decision step  408 , if the series of tests applied during step  406  do not indicate speech in music, then processing proceeds to step  412 , in which a weaker test for speech in music is performed. This test may include determining if SLM exceeds a certain threshold and/or if the sum total of one or more SLM values associated with prior frames exceeds certain thresholds. For example, in one implementation, speech in music is indicated if SLM is greater than 1.8 and the sum of the SLM values associated with the two preceding frames is greater than 4.0. As shown at decision step  414 , if the test of step  412  indicates speech in music, then processing proceeds to step  416 , in which the FLC method for speech is selected. 
         [0074]    After the FLC method designed for speech has been selected at step  416 , the pitch period is set to the largest multiple of the pitch period that will fit within frame size. This is done because there is a weak indication of speech in the recent past but a long-term indication of music. Consequently, the FLC method designed for speech is used but with a larger pitch multiple, thereby making it act more like an FLC method designed for music (e.g., a frame repeat FLC method). After this, processing ends at step  422  labeled “end”. 
         [0075]    Returning now to decision step  414 , if the weaker test performed at step  412  does not indicate speech in music, then the FLC method designed for music is selected as shown at step  420 . After this processing ends at step  422 . 
       1. FLC METHODS DESIGNED FOR SPEECH AND MUSIC IN ACCORDANCE WITH AN EMBODIMENT OF THE PRESENT INVENTION 
       [0076]    As noted above, an embodiment of the present invention includes a processing block  161  that performs an FLC method designed for speech and a processing block  162  that performs an FLC method designed for music. In this section, further detail will be provided about each of these FLC methods and how they are implemented by processing blocks  161  and  162 . In addition, a ringing signal computation that is common to both approaches will be described. 
         [0077]    The present invention is for use with either audio codecs that employ overlap-add synthesis at the decoder or with codecs that do not, such as PCM. As used herein, AOLA denotes the number of samples in the window used for overlap-add synthesis at the decoder. Thus, for codecs that employ overlap-add synthesis at the decoder, AOLA&gt;0, while for codecs that do not, AOLA=0. 
         [0078]    a. Ringing Signal Computation 
         [0079]    For both FLC methods described in this section, a “ringing” signal, r, is obtained to maintain continuity between the previously-decoded frame and the lost frame. For the case where there is no audio overlap-add synthesis at the decoder (AOLA=0), this ringing signal is calculated as the zero-input response of a synthesis filter associated with the audio decoder  110 . As discussed in U.S. patent application Ser. No. 11/234,291 to Chen, filed Sep. 26, 2005, and entitled “Packet Loss Concealment for Block-Independent Speech Codecs” (the entirety of which is incorporated by reference herein), an effective approach is to use the ringing of the cascaded long-term and short-term synthesis filters of the decoder. 
         [0080]    The length of the ringing signal for overlap-add is denoted herein as ROLA. If the pitch period is less than the overlap length, the ringing is computed for one pitch period and then waveform repeated to obtain ROLA samples. The pitch used for ringing, ppr, may be a multiple of the original pitch period, pp, depending on the mode (SPEECH or MUSIC) as determined by signal classifier  130  and the decision logic applied by FLC decision/control logic  140 . In one implementation, ppr is determined as follows: if the selected mode is MUSIC and the frame size (FRSZ) is greater than or equal to two times the original pitch period (pp) then ppr is set to two times pp. Otherwise, ppr is set to ppm. As used herein, ppm refers to a modified pitch period that results when the pitch period is multiplied. As discussed above, such multiplication of the pitch period may occur as a result of the operation of FLC decision/control logic  140 . 
         [0081]    If an audio overlap-add signal is available, there is no zero-input response computation, and the ringing signal is set to the audio fade-out signal provided by the decoder, denoted herein as A out . 
         [0082]    b. Improved Frame Repeat Method 
         [0083]    In accordance with an embodiment of the present invention, the FLC method designed for music is an improved frame repeat method. As discussed in U.S. patent application Ser. No. 11/285,311 to Chen, filed Nov. 23, 2005, and entitled “Classification-Based Frame Loss Concealment for Audio Signals”, a frame repeat method combined with the overlapping windows of typical audio coders produces surprisingly sufficient quality for most music. 
         [0084]      FIG. 5  is a flowchart  500  illustrating an improved frame repeat method in accordance with an embodiment of the present invention. As shown in FIG.  5 , the beginning of flowchart  500  is indicated by a step  502  labeled “start”. Processing immediately proceeds to step  504 , in which it is determined whether the current frame is the first bad (i.e., erased) frame since a good (i.e., non-erased) frame was received. If so, step  506  is performed. In step  506 , the last good frame played out, denoted Lgf, is overlap-added with the ringing signal, r, to form the “correlated” repeat component fr cor : 
         [0000]                                if (AOLA &gt; 0)        fr cor (n) = Lgf(n) · wc in (n) + r(n) · wc out (n)  n = 0..AOLA − 1        fr cor (n) = Lgf(n)              n = AOLA..FS − 1       else        fr cor (n) = Lgf(n) · wc in (n) + r(n) · wc out (n)  n = 0..ROLA − 1        fr cor (n) = Lgf(n)              n = ROLA..FS − 1                    
where wc in  is a correlated fade-in window, wc out  is a correlated fade-out window, AOLA is the length in samples of the overlap-add window, ROLA is the length in samples of the ringing signal for overlap-add, and FS is the number of samples in a frame (i.e., the frame size).
 
         [0085]    The overlap-add is performed with a window containing the following property: 
         [0000]        wc   in ( n )+ wc   out ( n )=1. 
         [0000]    Note that A out  likely has a portion or all of w out  already applied. Typically, the audio encoder applies √{square root over (wc out (n))} and the decoder does the same. It should be understood that whatever portion of the window has been applied is not reapplied to the ringing signal, r. 
         [0086]    At step  508 , locally-generated white or Gaussian noise is passed through an LPC filter in a manner similar to that described in U.S. patent application Ser. No. 11/234,291 to Chen (the entirety of which has been incorporated by reference herein), except that in the present embodiment, scaling is applied to the noise signal after it has been passed through the LPC filter rather than before, and the scaling factor is based on the average magnitude of the speech signal associated with the last frame rather than on the average magnitude of the LPC prediction residual signal of the last frame. This step produces a filtered noise signal n lpc . Enough samples (FS+OLAG) are produced for the current frame and for an overlap-add window for the first good frame. 
         [0087]    At step  510 , an appropriate mixture of the repeated signal fr cor  and the filtered noise signal n lpc  is determined. Many different methods can be used to perform this step. In one implementation, a “voicing measure” or figure of merit (fom) such as that described in U.S. patent application Ser. No. 11/234,291 to Chen is used to compute a scale factor, β, that ranges from 0 to 1. The scale is overwritten to 0 if the current classification from signal classifier  130  is MUSIC. 
         [0088]    At step  512 , a scaled overlap-add of the repeated signal fr cor  and the filtered noise signal n lpc  is performed. The scaled overlap-add is preferably performed in accordance with the method described in Section C below. Hence: 
         [0000]                                                sq(N+n) = fr cor (n)·(1−β)+(A out (n)·wu out (n)+   n = 0..AOLA−1           n lpc (n)·wu in (n))·β           sq(N+n) = fr cor (n)·(1−β)+n lpc (n)·β    n = AOLA..FS−1                        
where sq is the output signal buffer, N is the position of the first sample of the current frame in the output signal buffer, fr cor  is the correlated repeat component, β is the scale factor described in the preceding paragraph, n lpc  is the filtered noise signal, A out  is the audio fade-out signal, wu out  is the uncorrelated fade-out window, wu in  is the uncorrelated fade-in window, AOLA is the overlap add window length, and FS is the frame size. Where there is no overlap-add synthesis at the decoder, AOLA=0, and the foregoing simply becomes:
 
         [0000]        sq ( N+n )= fr   cor ( n )·(1−β)+ n   lpc ( n )·β  n= 0 ..FS −1. 
         [0089]    At step  514 , denoted “update speech-FLC”, any frame-to-frame memory is updated in order to maintain continuity (signal buffer, decimation filters, LPC filters, pitch buffers, etc.). 
         [0090]    If the frame erasure lasts for an extended period of time, the output of the FLC scheme is preferably ramped down to zero in a gradual manner in order to avoid buzzy sounds or other artifacts. At step  516 , a measure of the time in frame erasure is compared to a predetermined threshold, and if it exceeds the threshold, step  518  is performed which attenuates the signal in the output signal buffer denoted sq(N..FS−1). A linear ramp starting at 43 ms and ending at 63 ms is preferably used. Finally, at step  520 , the samples in sq(N..FS−1) are released to a playback buffer. After this, processing ends as indicated by step  522  labeled “end”. 
       i. Overlap-Add in First Good Frame 
       [0091]    As described above in reference to step  212  of  FIG. 2 , an overlap-add is performed on the first good frame after erasure for both FLC methods. The overlap window length for this step is denoted OLAG herein. If an audio codec that employs overlap-add synthesis at the decoder is being used, this overlap-add length will be the length of the built-in analysis overlap. Otherwise, it is a tuned parameter. The overlap-add is again performed in accordance with a method described below in Section C below. For the improved frame repeat method, the function is: 
         [0000]                                    sq(N+n) = (fr cor (n)·wc out (n)+sq(N+n)·wc in (n))·(1−β)+   n = 0..OLAG−1           (n lpc (n+FS)·wu out (n)+sq(N+n)·wu in (n))·β                    
where sq is the output signal buffer, N is the position of the first sample of the current frame in the output signal buffer, fr cor  is the correlated repeat component, β is the scale factor, n lpc  is the filtered noise signal, wc out  is the correlated fade-out window, wc in  is the correlated fade-in window, wu out  is the uncorrelated fade-out window, wu in  is the uncorrelated fade-in window, OLAG is the overlap-add window length, and FS is the frame size. It should be noted that sq(N+n) likely has a portion or all of wc in  already applied if the frame is from an audio decoder. Typically, the audio encoder applies √{square root over (wc in (n))} and the decoder does the same. It should be understood that whatever portion of the window has been applied is not reapplied.
 
       ii. Gain Attenuation 
       [0092]    In a manner similar to that described in U.S. patent application Ser. No. 11/234,291 to Chen, which has been incorporated by reference herein, if the frame erasure lasts too long, the output is attenuated to avoid buzzy artifacts. The gain attenuation duration is from 43 ms to 63 ms. 
       iii. Ramp Up in First Good Frame 
       [0093]    As described above in reference to step  212  of  FIG. 2 , a “ramp up” operation is performed on the first good frame after erasure for both FLC methods. In particular, in order to avoid an abrupt energy change from FLC frames to the first good frame, the output signal in the first good frame is ramped up from a scale factor associated with a last sample in the previously-described gain attenuation step, to 1, over a period of
       min(OLAG,0.02*SF)
 
where SF is the sampling frequency.
       
 
         [0095]    c. FLC Method Designed for Speech 
         [0096]    In an embodiment of the present invention, the FLC method applied by processing block  161  is a modified version of that described in U.S. patent application Ser. No. 11/234,291 to Chen, which is incorporated by reference herein. A flowchart of the modified approach is collectively depicted in  FIGS. 6 and 7  of the present application. Because the flowchart is large, it has been divided into two portions, one depicted in  FIG. 6  and one depicted in  FIG. 7 , with a node “A” as the connecting point between the two portions. 
         [0097]    The method begins at step  602 , which is located in the upper left corner of  FIG. 6  and is labeled “start”. Processing then immediately proceeds to decision step  604 , in which it is determined whether the current frame is erased. If the current frame is not erased, then processing proceeds to decision step  606 , in which it is determined whether the current frame is the first good frame after an erasure. If the current frame is not the first good frame after an erasure, then the decoded speech samples in the current frame are copied to a corresponding location in the output buffer as shown at step  608 . 
         [0098]    If it is determined at decision step  606  that the current frame is the first good frame after erasure, then the current frame is overlap added with an extrapolated frame loss signal as shown at step  610 . The overlap window length is designated OLAG. If an audio codec that employs overlap-add synthesis at the decoder is being used, this overlap-add length will be the length of the built-in analysis overlap. Otherwise, it is a tuned parameter. The overlap-add is performed in accordance with a method described in Section C below. The function is: 
         [0000]                                    sq(N+n) = (1−β)·(sq(N+n)·wc in (n)+sq(N+FS+n)·   n = 0..OLAG−1           wc out (n))+ β·(sq(N+n)·wu in (n)+n lpc (FS+n)·           wu out (n))                    
where sq is the output signal buffer, N is the position or the first sample or the current frame in the output signal buffer, β is a scale factor that will be described in more detail herein, wc out  is the correlated fade-out window, wc in  is the correlated fade-in window, wu out  is the uncorrelated fade-out window, wu in  is the uncorrelated fade-in window, OLAG is the overlap-add window length for the first good frame, and FS is the frame size.
 
         [0099]    After step  610 , control flows to step  612  in which a “ramp up” operation is performed on the current frame. In particular, in order to avoid an abrupt energy change from FLC frames to the first good frame, the output signal in the first good frame is ramped up from a scale factor associated with a last sample in a gain attenuation step (described herein in reference to step  648  of  FIG. 6 ) to 1, over a period of
       min(OLAG,0.02*SF)
 
where SF is the sampling frequency.
       
 
         [0101]    After step  608  or  612  is completed, processing proceeds to step  614 , which updates the coefficients of a short-term predictor by performing a so-called “LPC analysis”, a technique that is well-known by persons skilled in art. One method of performing this step is described in more detail in U.S. patent application Ser. No. 11/234,291. After step  614  is completed, control flows to node  650 , labeled “A”. This node is identical to node  702  in  FIG. 7 . 
         [0102]    Returning now to decision step  604 , if it is determined during this step that the current frame is erased, then processing proceeds to decision step  618 , in which it is determined whether the current frame is the first frame in this current stream of erasure. If the current frame is not the first frame in this stream of erasure, processing proceeds directly to decision step  624 . 
         [0103]    However, if the current frame is the first frame in this stream of erasure, then a determination is made at decision step  620  as to whether or not there is audio overlap-add synthesis at the decoder. If there is no audio overlap-add synthesis at the decoder (i.e., if AOLA=0), then the ringing signal of a cascaded long-term synthesis filter and short-term synthesis filter is calculated at step  622 . This calculation is discussed above in Section A.1.a, and described in detail in U.S. patent application Ser. No. 11/234,291 to Chen. 
         [0104]    If there is audio overlap-add synthesis at the decoder (i.e., if AOLA&gt;0), then an audio overlap-add signal is available and the ringing signal is not calculated at step  622 . Rather, the ringing signal is set to an audio fade-out signal provided by the decoder, denoted A out . In either case, control then flows to decision step  624 . 
         [0105]    At decision step  624 , it is determined whether a voicing measure (the calculation of which is described below in reference to step  718  of  FIG. 7 ) has a value greater than a first threshold value T 1 . If the answer is “No”, the waveform in the last frame is considered not periodic enough to warrant doing any periodic waveform extrapolation. As a result, steps  626 ,  628  and  630  are bypassed and control flows directly to decision step  632 . On the other hand, if the answer is “Yes”, the waveform in the last frame is considered to have at least some degree of periodicity. Consequently, control flows to decision step  626 . 
         [0106]    At decision step  626 , a determination is made as to whether or not there is audio overlap-add synthesis at the decoder. If there is no audio overlap-add synthesis at the decoder (i.e., if AOLA=0), then processing proceeds directly to step  630 . However, if there is audio overlap-add synthesis at the decoder (i.e., if AOLA&gt;0), then pitch refinement based on the audio fade-out signal is performed at step  628  prior to performance of step  630 . 
         [0107]    The pitch used for frame erasure is that estimated during the last good frame, denoted pp. Due to the local stationarity of speech, it is a good estimate for the pitch in the lost frame. However, due to the time separation between frames, it can be expected that the pitch has deviated from the last frame. As is described elsewhere herein, an embodiment of the invention utilizes an audio fade-out signal to overlap-add with the periodic extrapolated signal. If the pitch has deviated, this can result in the overlapping signals becoming out-of-phase, and to begin to cancel each other. This is especially problematic for small pitch periods. To alleviate the cancellation, step  628  uses the audio fade-out signal to refine the pitch. 
         [0108]    Many different methods can be used to refine the pitch. One such method is to maximize the normalized cross correlation between the two signals. In this approach, the signal buffer sq is extrapolated for each pitch candidate and the resulting signal is correlated with the audio fade-out signal. However, at high sampling rates, this approach quickly becomes very complex. A low complexity alternative described in Section D below is preferably used. The sq buffer is extrapolated for each pitch candidate in this reduced complexity method. The initial conditions used are: 
         [0000]      Δ 0 =min(127 ,┌pp *0.2┐) 
         [0000]      P 0 =ppm 
         [0000]    The final refined pitch will be denoted ppmr. If pitch refinement is not performed at step  628 , ppmr is set to equal ppm. 
         [0109]    Regardless of whether pitch refinement is performed at step  628 , control then flows to step  630 . At step  630 , the signal buffer sq is extrapolated and simultaneously overlap-added with the ringing signal on a sample-by-sample basis using the refined pitch ppmr. The extrapolation is computed as: 
         [0000]                                    sq(N+n) = sq(N+n−ppmr)·wc in (n)+ring(n)·wc out (n)   n = 0..ROLA−1       sq(N+n) = sq(N+n−ppmr)   n = ROLA..FS+           OLAG                    
where sq is the output signal buffer, N is the position of the first sample of the current frame in the output signal buffer, ppmr is the refined pitch, wc in  is the correlated fade-in window, wc out  is the correlated fade-out window, ring is the ringing signal, ROLA is the length in samples of the ringing signal for overlap-add, OLAG is the overlap-add length for the first good frame, and FS is the frame size. Note that A out  likely has a portion or all of wc out  already applied. Typically, the audio encoder applies √{square root over (wc out (n))} and the decoder does the same. It should be understood that whatever portion of the window has been applied is not reapplied.
 
         [0110]    Compared to simply extrapolating the signal, this technique is advantageous. It incorporates the original signal fading out into the extrapolation so the extrapolation is closer to the original signal. The successive periods of the extrapolated signal are slightly different due to the incorporated fade-out signal resulting in a significant reduction in buzzy artifacts (these occur when the simple extrapolation results in identical pitch periods which get repeated over and over and are too periodic). 
         [0111]    After decision step  624  or step  630  is complete, processing then proceeds to decision step  632 , in which it is determined whether the voicing measure (the calculation of which is described below in reference to step  718  of  FIG. 7 ) is less than a second threshold T 2 . If the answer is “No”, the waveform in the last frame is considered highly periodic and there is no need to mix in any random, noisy component in the output audio signal; hence, control flows directly to decision step  640  as shown in  FIG. 6 . 
         [0112]    If, on the other hand, the answer to decision  632  is “Yes”, then control flows to step  634 . At step  634 , a sequence of pseudo-random white noise is generated. Following step  634 , the sequence of pseudo-random white noise is passed through a short-term synthesis filter to generate a filtered noise signal, as shown at step  636 . The manner in which steps  634  and  636  are performed is described in detail in U.S. patent application Ser. No. 11/234,291 to Chen, except that in the present embodiment, scaling is applied to the noise signal after it has been passed through the short-term synthesis filter rather than before, and the scaling factor is based on the average magnitude of the speech signal associated with the last frame rather than on the average magnitude of the LPC prediction residual signal of the last frame. 
         [0113]    After step  636 , control flows to step  638  in which the voicing measure is used to compute a scale factor, β, which ranges from 0 to 1. One manner of computing such a scale factor is set forth in detail in U.S. patent application Ser. No. 11/234,291 to Chen. If it was determined at decision step  624  that the voicing measure does not exceed T 1 , then β will be set to one. 
         [0114]    Following decision step  632  or step  638 , decision step  640  determines if the current frame is the first erased frame in a stream of erasure. If the current frame is the first frame in the stream of erasure, the audio fade-out signal, A out , is combined with the extrapolated signal and the LPC generated noise from step  636  (denoted n lpc ), as shown at step  642 . The signal and the noise are combined in accordance with the scaled overlap-add technique described in Section C below. Hence: 
         [0000]                                    sq(N+n) = (1−β)·(sq(N+n)·wc in (n)+A out (n)·    n = 0..AOLA−1           wc out (n))+ β·(n lpc (n)wu in (n)+A out (n)·           wu out (n))       sq(N+n) = (1−β)·(sq(N+n))+β·n lpc (n)   n = AOLA..FS−1                    
where sq is the output signal buffer, N is the position of the first sample of the current frame in the output signal buffer, β is the scale factor, n lpc  is the noise signal, A out  is the audio fade-out signal, wc out  is the correlated fade-out window, wc in  is the correlated fade-in window, wu out  is the uncorrelated fade-out window, wu in  is the uncorrelated fade-in window, AOLA is the overlap-add window length, and FS is the frame size. Note that if β=0, then only the extrapolated signal and the audio fade-out signal are combined and if β=1, then only the LPC generated noise and the audio fade-out signal are combined.
 
         [0115]    If it is determined at decision step  640  that the current frame is not the first erased frame in a stream of erasure, then there is no audio fade-out signal, A out , for overlapping. Consequently, only the extrapolated signal and the LPC generated noise are combined at step  644  in accordance with: 
         [0000]        sq ( N+n )=(1−β)·( sq ( N+n ))+β· n   lpc ( n )  n =0 ..FS −1. 
         [0000]    In this instance, even though there is no audio fade-out signal for overlapping, a smooth signal transition will still occur at the frame boundary because the ringing signal was overlap-added with the extrapolated signal contained in the output signal buffer during step  630 . 
         [0116]    After step  642  or step  644  completes, processing proceeds to step  646 , which determines whether the current erasure is too long—that is, whether the current frame is too “deep” into erasure. If the length of the current erasure has not exceeded a predetermined threshold, then control flows to node  650  (labeled “A”) in  FIG. 6 , which is the same as node  702  in  FIG. 7 . However, if the length of the current erasure has exceeded this threshold, then step  648  is performed. Step  648  attenuates the signal in the output signal buffer denoted sq(N..FS−1) in a manner similar to that described in U.S. patent application Ser. No. 11/234,291 to Chen. This is done to avoid buzzy artifacts. A linear ramp starting at 43 ms and ending at 63 ms is preferably used. 
         [0117]    Turning now to  FIG. 7 , after the processing in  FIG. 6  is done, step  704  and step  708  are performed. Step  704  plays back the output signal samples in output signal buffer, while step  706  calculates the average magnitude of the speech signal associated with the last frame. This value is stored and is later used in step  634  to scale the filtered noise signal. 
         [0118]    After step  708 , processing proceeds to decision step  710 , in which it is determined whether the current frame is erased. If the answer is “Yes”, then steps  712 ,  714 ,  716  and  718  are skipped, and control flows directly to step  720 . If the answer is “No”, then the current frame is a good frame, and steps  712 ,  714 ,  716  and  718  are performed. 
         [0119]    Step  712  uses any one of a large number of possible pitch estimators to generate an estimated pitch period pp that may be used by processes  622 ,  628  and  630  during processing of the next frame. Step  714  calculates an extrapolation scaling factor that may optionally be used by step  630  in the next frame. In the present implementation, this extrapolation scaling factor has been set to one and thus does not appear in any of the equations associated with step  630 . Step  716  calculates a long-term filter memory scaling factor that may be used in step  622  in the next frame. Step  718  calculates a voicing measure on the current frame of decoded speech. The voicing measure is a single figure of merit whose value depends on how strongly voiced the underlying speech signal is. One method of performing each of steps  712 ,  714 ,  716  and  718  is described in more detail in U.S. patent application Ser. No. 11/234,291 to Chen. 
         [0120]    After decision step  710  or step  718  is done, control flows to step  720 . Step  720  updates a pitch period buffer. In one implementation of the present invention, the pitch period buffer is used by signal classifier  130  of  FIG. 1  to calculate a pitch period change parameter that is used by signal classifier  130  and FLC decision/control logic  140 , as discussed elsewhere herein. After step  720  is complete, step  722  updates a short-term synthesis filter memory that may be used in steps  622  and  636  during processing of the next frame. After step  722  is complete, step  724  performs shifting and updating of the output speech buffer. After step  724  is complete, step  726  stores extra samples of the extrapolated speech signal beyond the need of the current frame as the ringing signal for the next frame. One method of performing each of steps  720 ,  722 ,  724  and  726  is described in more detail in U.S. patent application Ser. No. 11/234,291 to Chen. 
         [0121]    After step  726 , control flows to step  728 , which is labeled “end”. Node  728  denotes the end of the frame processing loop. Then, the control flow goes back to node  602  labeled “start” to start the frame processing for the next frame. 
       B. ROBUST SPEECH/MUSIC CLASSIFICATION FOR AUDIO SIGNALS IN ACCORDANCE WITH AN EMBODIMENT OF THE PRESENT INVENTION 
       [0122]    Embodiments for classifying audio signals as speech or music are described in the present section. The example embodiments described herein are provided for illustrative purposes, and are not limiting. Further structural and operational embodiments, including modifications/alterations, will become apparent to persons skilled in the relevant art(s) from the teachings herein. 
         [0123]      FIG. 8  shows a block diagram of a speech/non-speech classifier  800  in accordance with an example embodiment of the present invention. Speech/non-speech classifier  800  may be used to implement signal classifier  130  described above in reference to  FIG. 1 , for example. However, speech/non-speech classifier  800  may also be used in a variety of other applications as will be readily understood by persons skilled in the relevant art(s). 
         [0124]    As shown in  FIG. 8 , speech/non-speech classifier  800  includes an energy tracker module  810 , a feature extraction module  820 , a normalization module  830 , a speech likelihood measure module  840 , a long term running average module  850 , and a classification module  860 . These modules may be implemented in hardware, software, firmware, or any combination thereof. For example, one or more of these modules may be implemented in logic, such as a programmable logic chip (PLC), in a programmable gate array (PGA), in a digital signal processor (DSP), as software instructions that execute in a processor, etc. 
         [0125]    These various functional components of speech/non-speech classifier  800  will now be described. 
       1. Energy Tracker Module Embodiments 
       [0126]    In embodiments, energy tracker module  810  tracks one or both of a maximum frame energy estimate and a minimum frame energy estimate of a signal frame received on an input signal  802 . Input signal  802  is characterized herein as x(n). In an example embodiment, which is further described below, energy tracker module  810  tracks frame energy using a combination of long term and short term minimum/maximum estimators. A final threshold for active signals may be derived from both the minimum and maximum estimators. 
         [0127]    One example energy tracking algorithm tracks a base-2 logarithmic signal gain, lg. Note that frame energy is discussed in terms of lg in the following description for illustrative purposes, but may alternatively be referred to in other terms, as would be understood to persons skilled in the relevant art(s). 
         [0128]    Signal activity detectors, such as energy tracker module  810 , may be used to distinguish a desired audio signal from noise on a signal channel. For instance, in one implementation, a signal activity detector may detect a level of noise on the signal channel, and use this detected noise level as a minimum energy estimate. A predetermined offset value is added to the detected noise level to create a threshold level. A signal level on the signal channel that is above the threshold level is considered to be the desired audio signal. In this manner, signals with large dynamic range (e.g., speech) can be relatively easily distinguished from a noise floor. 
         [0129]    However, for signals with a smaller dynamic range (certain music for example), a threshold based on a maximum energy estimate may have better performance. For a smaller dynamic range signal, a tracking system based on a minimum energy estimate may undesirably determine the minimum energy estimate to be roughly equal to lower level audio portions of the audio signal. Thus, portions of the audio signal may be mistaken for noise. In contrast, a signal activity detector based on a maximum energy estimate detects a maximum signal level on the signal channel, and subtracts a predetermined offset level from the detected maximum signal level to create a threshold level. The subtracted offset level can be selected to maintain the threshold level below the lower level audio portions of the audio signal. A signal level on the signal channel that is above the threshold level is considered to be the desired audio signal. 
         [0130]    In embodiments, energy tracking module  810  may be configured to track a signal according to these minimum and/or maximum energy estimate techniques. In embodiments where both the minimum and maximum energy estimates are used, energy tracking module  810  provides a meaningful active signal threshold for a wide range of signal types. Furthermore, the tracking of short term estimators and long term estimators (as further described below) enables classifier  800  to adapt quickly to sudden changes in the signal energy profile while at the same time maintaining some stability and smoothness. The determined final active signal threshold is used by long term running average module  850  to indicate when to update the long term running average of the speech likelihood measure. In order to provide accurate classification in the presence of background noise or interfering signals, updates to detected minimum and/or maximum estimates are performed during active signal detection. 
         [0131]      FIG. 9  shows a flowchart  900  providing example steps for tracking energy of an audio signal, according to example embodiments of the present invention. Flowchart  900  may be performed by energy tracking module  810 , for example. The steps of flowchart  900  need not necessarily occur in the order shown in  FIG. 9 . Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. Flowchart  900  is described as follows. 
         [0132]    Flowchart  900  begins with step  902 . In step  902 , a maximum frame energy estimate is determined. The maximum frame energy estimate for an input audio signal may be measured and/or determined according to conventional or other techniques, as would be known to persons skilled in the relevant art(s). 
         [0133]    In step  904 , a minimum frame energy estimate is determined. The minimum frame energy estimate for an input audio signal may be measure and/or determined according to conventional or other techniques, as would be known to persons skilled in the relevant art(s). 
         [0134]    In step  906 , a threshold for active signals is determined based on the maximum frame energy estimate and the minimum frame energy estimate. For example, as described above, a first offset may be added to the determined minimum frame energy estimate, and a second offset may be subtracted from the determined maximum frame energy estimate, to generate respective first and second thresholds. The first and/or second thresholds may be compared to an input signal to determine whether the input signal is active. 
         [0135]      FIG. 10  shows an example block diagram of energy tracking module  810 , in accordance with an embodiment of the present invention. Energy tracking module  810  shown in  FIG. 10  may be used to implement flowchart  900  shown in  FIG. 9 . However, energy tracking module  810  may also be used in a variety of other applications as will be readily understood by persons skilled in the relevant art(s). As shown in  FIG. 10 , energy tracking module  810  includes a maximum energy tracker module  1002 , a minimum energy tracker module  1004 , and an active signal detector module  1006 . Example embodiments for these portions of energy tracking module  810  will now be described. 
       a. Maximum Energy Tracker Module Embodiments 
       [0136]    In an embodiment, maximum energy tracker module  1002  generates and maintains a short term estimate (StMaxEst) and a long term estimate (LtMaxEst) of the maximum frame energy for input signal  802 . In alternative embodiments, just one of StMaxEst and LtMaxEst may be generated/maintained, and/or other types of estimates may be generated. StMaxEst and LtMaxEst are output by maximum energy tracker module  1002  on maximum energy tracking signal  1008  in a serial, parallel, or other fashion. 
         [0137]    In a conventional maximum (or peak) energy tracker, energy of a received signal frame is compared to a current maximum energy estimate. If the current maximum energy estimate is less than the frame energy, the (new) maximum energy estimate is set to the frame energy. If the current maximum energy estimate is greater than the frame energy, the current maximum energy estimate is decreased by a predetermined static amount to create a new maximum energy estimate. This conventional technique results in a maximum energy estimate that jumps to a maximum amount instantaneously and then decays (by the static amount). The static amount for decay is selected as a trade-off between stability (slow decay) and a desired degree of responsiveness, especially if input signal characteristics have changed (e.g., a switch from speech to music or vice versa has occurred; switching from loud, to quiet, to loud, etc., in different sections of a music piece has occurred; or a shift from singing, where there may be many peaks and valleys in the energy profile, to a more instrumental segment that has a more constant energy profile has occurred). 
         [0138]    To help overcome the problem of a long term maximum energy estimate that jumps quickly to track a peak energy value, in an embodiment (further described below), LtMaxEst is compared to StMaxEst (which is a relatively quickly decaying average of the frame energy, and thus is a slightly smoothed version of the frame energy), and is then updated, with the resulting LtMaxEst including a running average component and a component based on StMaxEst. 
         [0139]    To improve the problem related to decay, in an embodiment (further described below), the decay rate is increased further and further as long as the frame energy is less than StMaxEst. The concept is that longer periods are expected where the frame energy does not reach LtMaxEst, but the frame energy should often cross StMaxEst because StMaxEst decays quickly. If it does not, this is unexpected behavior that is most likely a local or longer term decrease in energy indicating changing characteristics in the signal input. As a result, LtMaxEst is more aggressively decreased. This prevents LtMaxEst from remaining too high for too long when the input signal changes. 
         [0140]    It may be desirable to track maximum frame energy in this manner while maintaining similar performance over different input dynamic ranges. For example, if StMaxEst is tracking a signal maximum, and then the signal suddenly goes to the noise floor for a relatively long time period, it is desirable for the decay of StMaxEst to reach the noise floor in approximately the same amount of time whether a relatively high (e.g., 60 dB) dynamic range or a relatively low (e.g., 10 dB) dynamic range was present. Thus, in an embodiment, the adaptation of StMaxEst is normalized to the dynamic range. In an embodiment described further below, StMaxEst is updated based on the current estimated dynamic range of the input signal. In this way, the system becomes adaptive to the dynamic range, where the long term and short term maximum energy estimates adapt slower when receiving small dynamic range signals and adapt faster when receiving wide dynamic range signals. 
         [0141]    These embodiments allow for a smooth but responsive long term maximum energy estimate that functions well over a large dynamic range of input signals, and can track changes in dynamic range quickly. 
         [0142]    For example, in an embodiment, if the currently measured frame energy, lg, exceeds the currently stored value for StMaxEst, StMaxEst is updated as follows: 
         [0000]        St MaxEst= St MaxEst· St MaxBeta+ lg ·(1 −St MaxBeta) 
         [0000]    where StMaxBeta is a variable set between 0 and 1 (e.g., tuned to 0.5 in one embodiment). StMaxEst may have an initialization value, as appropriate for the particular application. For example, in an embodiment, StMaxEst may have an initial value of 6. The long term maximum estimate, LtMaxEst, is updated as follows: 
         [0000]        Lt MaxEst= Lt MaxEst· Lt MaxBeta+ lg ·(1 −Lt MaxBeta) 
         [0000]    where LtMaxBeta is a variable generated to be between 0 and 1. LtMaxEst may have an initialization value, as appropriate for the particular application. For example, in an embodiment, LtMaxEst may have an initial value of 16. After updating LtMaxEst, LtMaxBeta is reset to an initial value (e.g., 0.99 in one embodiment). Furthermore, if StMaxEst is greater than LtMaxEst, LtMaxEst is adjusted as follows: 
         [0000]                                if (StMaxEst &gt; LtMaxEst)        LtMaxEst = LtMaxEst · LtMaxAlpha + StMaxEst · (1 − LtMaxAlpha)                    
where LtMaxAlpha is set between 0 and 1 (e.g., tuned to 0.5 in one embodiment). Thus, as described above, if StMaxEst is greater than LtMaxEst, LtMaxEst is adjusted with the sum of a long term running average component (LtMaxEst·LtMaxAlpha) and a component based on StMaxEst (StMaxEst·(1−LtMaxAlpha)). If the frame energy is less than the short term maximum estimate StMaxEst, the more likely the long term maximum estimate LtMaxEst is lagging, so LtMaxBeta may be decreased in order to increase a change in long term maximum estimate LtMaxEst when there is an update:
 
         [0000]                                                                          if (lg ≦ StMaxEst)                LtMaxBeta = LtMaxBeta · LtMaxBetaDecay                where                                    LtMaxBetaDecay   =     0.9998   ·     FS   344     ·     16   SF                                    
and FS is the frame size, and SF is the sampling frequency in kHz.
 
         [0143]    Finally, the short-term maximum estimate StMaxEst is updated by reducing it slightly, by a factor that depends on the input dynamic range, as mentioned above. As shown in  FIG. 10 , maximum energy tracker module  1002  receives a minimum energy tracking signal  1010  from minimum energy tracker module  1004 . Minimum energy tracking signal  1010  includes a long term minimum energy estimate, LtMinEst, generated by minimum energy tracker module  1004 , which is used as an indication of the input dynamic range: 
         [0000]    
       
         
               
               
             
               
               
             
               
               
             
               
               
             
               
               
             
               
               
             
           
               
                   
                   
               
             
             
               
                   
                 if (StMaxEst &gt; LtMinEst) 
               
             
          
           
               
                   
                 StMaxEst = StMaxEst − (StMaxEst − LtMinEst) · 
               
               
                   
                 StMaxStepSize 
               
             
          
           
               
                   
                 else 
               
             
          
           
               
                   
                 StMaxEst = LtMinEst 
               
             
          
           
               
                   
                 where 
               
               
                   
                   
               
             
          
           
               
                   
                 
                   
                     
                       
                         
                           StMaxStepSize 
                           = 
                           
                             0.0005 
                             · 
                             
                               FS 
                               344 
                             
                             · 
                             
                               16 
                               SF 
                             
                           
                         
                         , 
                       
                     
                   
                 
               
               
                   
                   
               
             
          
         
       
     
       In this way, the short-term estimate adaptation rate increases with the input dynamic range. 
     b. Minimum Energy Tracker Module Embodiments 
       [0144]    In an embodiment, minimum energy tracker module  1004  generates and maintains a short term estimate (StMinEst) and a long term estimate (LtMinEst) of the minimum frame energy for input signal  802 . In alternative embodiments, just one of StMinEst and LtMinEst is generated/maintained, and/or other types of estimates may be generated. StMinEst and LtMinEst are output by minimum energy tracker module  1004  on minimum energy tracking signal  1010  in a serial, parallel, or other fashion. 
         [0145]    Similarly to conventional maximum energy trackers described above, conventional minimum energy trackers compare energy of a received signal frame to a current minimum energy estimate. If the current minimum energy estimate is greater than the frame energy, the minimum energy estimate is set to the frame energy. If the current minimum energy estimate is less than the frame energy, the current minimum energy estimate is increased by a predetermined static amount. Again, this conventional technique results in a minimum energy estimate that jumps to a minimum amount instantaneously and then decays upward (by the static amount). To help overcome the problem of a long term minimum energy estimate dropping quickly to track a minimum energy value, in an embodiment (further described below), LtMinEst is compared to StMinEst and is then updated, with the resulting LtMinEst including a running average component and a component based on StMinEst. 
         [0146]    Similarly to above, to improve the problem related to decay, in an embodiment (further described below), the decay rate is increased further and further as long as the frame energy is greater than StMinEst. The concept is that longer periods are expected where the frame energy does not reach LtMinEst, but the frame energy should often cross StMinEst because StMinEst decays upward quickly. If it does not, this is unexpected behavior that is most likely a local or longer term increase in energy indicating changing characteristics in the signal input. As a result, LtMinEst is more aggressively increased. This prevents LtMinEst from remaining too low for too long when the input signal changes. 
         [0147]    Furthermore, as described above for maximum energy trackers, it may be desirable to track minimum frame energy with similar performance provided over different input dynamic ranges. In an embodiment, the adaptation of StMinEst is normalized to the dynamic range. As described further below, StMinEst is updated based on the current estimated dynamic range of the input signal. In this way, the system becomes adaptive to the dynamic range, where long term and short term minimum energy estimates adapt slower when receiving small dynamic range signals and adapt faster when receiving wide dynamic range signals. 
         [0148]    These embodiments allow for a smooth but responsive long term minimum energy estimate that functions well over a large dynamic range of input signals, and can track changes in dynamic range quickly. 
         [0149]    For example, in an embodiment, if lg is less than the short term minimum estimate, StMinEst, StMinEst and LtMinEst are updated as follows: 
         [0000]        St MinEst= St MinEst· St MinBeta+ lg ·(1 −St MinBeta) 
         [0000]    where StMinBeta is set between 0 and 1 (e.g., tuned to 0.5 in one embodiment). StMinEst may have an initialization value, as appropriate for the particular application. For example, in an embodiment, StMinEst may have an initial value of 21. LtMinEst is updated according to: 
         [0000]        Lt MinEst= Lt MinEst· Lt MinBeta+ lg ·(1 −Lt MinBeta) 
         [0000]    After updating LtMinEst, LtMinBeta is reset to an initial value (e.g., tuned to 0.99 in one embodiment). LtMinEst may have an initialization value, as appropriate for the particular application. For example, in an embodiment, LtMinEst may have an initial value of 6. If the short term min estimate StMinEst is less than the long term estimate LtMinEst, the long term estimate LtMinEst may be adjusted more aggressively, as follows: 
         [0000]                                if (StMinEst &lt; LtMinEst)        LtMinEst = LtMinEst · LtMinAlpha + StMinEst · (1 − LtMinAlpha)                    
where LtMinAlpha is set between 0 and 1 (e.g., tuned to 0.5 in one embodiment). Thus, as described above, if StMinEst is less than LtMinEst, LtMinEst is adjusted with the sum of a long term running average component (LtMinEst·LtMinAlpha) and a component based on StMinEst (StMinEst·(1−LtMinAlpha)).
 
         [0150]    However, if the frame energy is not less than the short term minimum estimate StMinEst, the more likely that the long term min estimate LtMinEst is lagging. In this case, LtMinBeta is decreased in order to increase a change to LtMinEst when there is an update: 
         [0000]        Lt MinBeta= Lt MinBeta· Lt MinBetaDecay 
         [0000]    where 
         [0000]    
       
         
           
             LtMinBetaDecay 
             = 
             
               0.9998 
               · 
               
                 FS 
                 344 
               
               · 
               
                 16 
                 SF 
               
             
           
         
       
     
         [0000]    As described above, the short term minimum estimate StMinEst is then updated by increasing it slightly by a factor that depends on the dynamic range of input signal  802 . As shown in  FIG. 10 , minimum energy tracker module  1004  receives maximum energy tracking signal  1008  from maximum energy tracker module  1002 . Maximum energy tracking signal  1008  includes long term maximum energy estimate, LtMaxEst, generated by maximum energy tracker module  1002 , which is used as an indication of the input dynamic range: 
         [0000]    
       
         
               
               
             
               
               
             
               
               
             
               
               
             
               
               
             
               
               
             
           
               
                   
                   
               
             
             
               
                   
                 if (StMinEst &lt; LtMaxEst) 
               
             
          
           
               
                   
                 StMinEst = StMinEst + (LtMaxEst − StMinEst) · 
               
               
                   
                 StMinStepSize 
               
             
          
           
               
                   
                 else 
               
             
          
           
               
                   
                 StMinEst = LtMaxEst 
               
             
          
           
               
                   
                 where 
               
               
                   
                   
               
             
          
           
               
                   
                 
                   
                     
                       
                         StMinStepSize 
                         = 
                         
                           0.0005 
                           · 
                           
                             FS 
                             344 
                           
                           · 
                           
                             16 
                             SF 
                           
                         
                       
                     
                   
                 
               
               
                   
                   
               
             
          
         
       
     
         [0151]    Finally, if either the short term minimum estimate StMinEst or long term minimum estimate LtMinEst is below a minimum threshold (e.g., set to −1 in one embodiment), they are set to that threshold. 
       c. Active Signal Detector Module Embodiments 
       [0152]    As shown in  FIG. 10 , active signal detector module  1006  receives input signal  802 , maximum energy tracking signal  1008  and minimum energy tracking signal  1010 . Active signal detector module  1006  generates a threshold, ThActive, which may be used to indicate an active signal for input signal  802 . ThActive may be generated according to: 
         [0000]        Th Max= Lt MaxEst−4.5 
         [0000]        Th Min= Lt MinEst+5.5 
         [0000]        Th Active=max(min( Th Max, Th Min),11.0) 
         [0000]    In alternative embodiments, values other than 4.5, 5.5, and/or 11.0 may be used to generate ThActive, depending on the particular application. Active signal detector module  1006  may further perform a comparison of energy of the current frame, lg, to ThActive, to determine whether input signal  802  is currently active: 
         [0000]                                            if (lg &gt; ThActive)             ActiveSignal = TRUE           else             ActiveSignal = FALSE                        
If ActiveSignal is TRUE, then input signal  802  is currently active. If ActiveSignal is FALSE, then input signal  802  is not active. Active signal detector module  1006  outputs ActiveSignal on active signal indicator signal  1012 . Energy tracker module  810  outputs maximum energy tracking signal  1008 , minimum energy tracking signal  1010 , and active signal indicator signal  1008  in a serial, parallel, or other fashion on energy tracking signal  804 .
 
       2. Feature Extraction Module Embodiments 
       [0153]    As shown in  FIG. 8 , feature extraction module  820  receives input audio signal  802 . Feature extraction module  820  analyzes one or more features of the input audio signal  802 . The analyzed features may be used by classifier  800  to determine whether the audio signal is a speech or non-speech (e.g., music, general audio, noise) signal. Thus, the features typically discriminate in some manner between speech and non-speech, and/or between unvoiced speech and voiced speech. In embodiments, any number and type of suitable features of input signal  802  may be analyzed by feature extraction module  820 . It is noted that feature extraction module  820  may alternatively be used in other applications as will be readily understood by persons skilled in the relevant art(s). 
         [0154]      FIG. 11  shows a flowchart  1100  providing example steps for analyzing features of an audio signal, according to example embodiments of the present invention. Flowchart  1100  may be performed by feature extraction module  820 . The steps of flowchart  1100  need not necessarily occur in the order shown in  FIG. 11 . Furthermore, in embodiments, not all steps of flowchart  1100  are necessarily performed. For example, flowchart  1100  relates to the analysis of four features of an audio signal. In alternative embodiments, fewer, additional, and/or alternative features of the audio signal may be analyzed. Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. 
         [0155]    Flowchart  1100  is described as follows with respect to  FIG. 12 .  FIG. 12  shows an example block diagram of feature extraction module  820 , in accordance with an example embodiment of the present invention. As shown in  FIG. 12 , feature extraction module  820  includes a pitch period change determiner module  1202 , a pitch prediction gain determiner module  1204 , a normalized autocorrelation coefficient determiner module  1206 , and a logarithmic signal gain determiner module  1208 . These modules of feature extraction module  820  are further described below along with a corresponding step of flowchart  1100 . 
         [0156]    In step  1102  of flowchart  1100 , a change in a pitch period between the frame and a previous frame of the audio signal is determined. Pitch period change determiner module  1202  may perform step  1102 . Pitch period change determiner module  1202  analyzes a first signal feature, which is a fractional change in pitch period, pp Δ , from one signal frame to the next. In an embodiment, the change in pitch period is calculated by pitch period change determiner module  1202  according to: 
         [0000]    
       
         
           
             
               pp 
               Δ 
             
             = 
             
               
                  
                 
                   
                     pp 
                     i 
                   
                   - 
                   
                     pp 
                     
                       i 
                       - 
                       1 
                     
                   
                 
                  
               
               
                 pp 
                 i 
               
             
           
         
       
     
         [0000]    where: 
         [0157]    pp i =a pitch period of a current input signal frame; and 
         [0158]    pp i−1 =a pitch period of a previous input signal frame. 
         [0159]    In step  1104 , a pitch prediction gain is determined. For example, pitch prediction gain determiner module  1204  may perform step  1104 . Pitch prediction gain determiner module  1204  analyzes a second signal feature, which is pitch prediction gain, ppg. In an embodiment, pitch prediction gain is calculated by pitch prediction gain determiner module  1204  according to: 
         [0000]    
       
         
           
             
               ppg 
               = 
               
                 10 
                 · 
                 
                   
                     log 
                     10 
                   
                    
                   
                     ( 
                     
                       E 
                       R 
                     
                     ) 
                   
                 
               
             
             , 
           
         
       
     
         [0000]    where: 
         [0160]    E=the signal energy in the pitch analysis window; and 
         [0161]    R=the pitch prediction residual energy. 
         [0000]    E may be calculated by: 
         [0000]    
       
         
           
             
               E 
               = 
               
                 
                   ∑ 
                   
                     n 
                     = 
                     
                       N 
                       - 
                       K 
                       + 
                       1 
                     
                   
                   N 
                 
                  
                 
                   
                     x 
                     2 
                   
                    
                   
                     ( 
                     n 
                     ) 
                   
                 
               
             
             , 
           
         
       
     
         [0000]    where: 
         [0162]    K=the analysis window size. 
         [0000]    R may be calculated by: 
         [0000]    
       
         
           
             
               R 
               = 
               
                 E 
                 - 
                 
                   
                     
                       c 
                       2 
                     
                      
                     
                       ( 
                       
                         pp 
                         i 
                       
                       ) 
                     
                   
                   
                     
                       ∑ 
                       
                         n 
                         = 
                         
                           N 
                           - 
                           K 
                           + 
                           1 
                         
                       
                       N 
                     
                      
                     
                       
                         x 
                         2 
                       
                        
                       
                         ( 
                         
                           n 
                           - 
                           
                             pp 
                             i 
                           
                         
                         ) 
                       
                     
                   
                 
               
             
             , 
           
         
       
     
         [0000]    where: 
         [0163]    c(·)=the signal correlation, which may be calculated by: 
         [0000]    
       
         
           
             
               c 
                
               
                 ( 
                 j 
                 ) 
               
             
             = 
             
               
                 ∑ 
                 
                   n 
                   = 
                   
                     N 
                     - 
                     K 
                     + 
                     1 
                   
                 
                 N 
               
                
               
                 
                   x 
                    
                   
                     ( 
                     n 
                     ) 
                   
                 
                 · 
                 
                   
                     x 
                      
                     
                       ( 
                       
                         n 
                         - 
                         j 
                       
                       ) 
                     
                   
                   . 
                 
               
             
           
         
       
     
         [0164]    In step  1106 , a first normalized autocorrelation coefficient is determined. For example, normalized autocorrelation coefficient determiner module  1206  may perform step  1106 . Normalized autocorrelation coefficient determiner module  1206  analyzes a third signal feature, which is the first normalized autocorrelation coefficient, ρ 1 . In an embodiment, the first normalized autocorrelation coefficient is calculated by normalized autocorrelation coefficient determiner module  1206  according to: 
         [0000]    
       
         
           
             
               ρ 
               1 
             
             = 
             
               
                 
                   ∑ 
                   
                     n 
                     = 
                     
                       N 
                       - 
                       K 
                       + 
                       2 
                     
                   
                   N 
                 
                  
                 
                   
                     x 
                      
                     
                       ( 
                       n 
                       ) 
                     
                   
                   · 
                   
                     x 
                      
                     
                       ( 
                       
                         n 
                         - 
                         1 
                       
                       ) 
                     
                   
                 
               
               E 
             
           
         
       
     
       Note that ρ 1  works well for narrowband signals (up to 16 kHz). Beyond the narrowband signal range, ρ [SF/16]  may instead be desirable to use, where SF is the sampling frequency in kHz. 
       [0165]    In step  1108 , a logarithmic signal gain is determined. For example, logarithmic signal gain determiner module  1208  may perform step  1108 . Logarithmic signal gain determiner module  1208  analyzes a fourth signal feature, which is the logarithmic signal gain, lg. In an embodiment, the logarithmic signal gain is calculated by logarithmic signal gain determiner module  1208  according to: 
         [0000]        lg =log 2 ( E/K ). 
         [0166]    As shown in  FIG. 12 , feature extraction module  820  outputs an extracted feature signal  806 , which includes the results of the analysis of the one or more analyzed signal features, such as change in pitch period, PP Δ  (from module  1202 ), pitch prediction gain, ppg (from module  1204 ), first normalized autocorrelation coefficient, ρ 1  (from module  1206 ), and logarithmic signal gain, lg (from module  1208 ). 
       3. Normalization Module Embodiments 
       [0167]    As shown in  FIG. 8 , normalization module  830  receives energy tracking signal  804  and extracted feature signal  806 . Normalization module  830  normalizes the analyzed signal feature results received on extracted feature signal  806 . In embodiments, normalization module  830  may normalize results for any number and type of received features, as desired for the particular application. In an embodiment, normalization module  830  is configured to normalize the feature results such that the normalized feature results tend in a first direction (e.g., toward −1) for unvoiced or noise-like characteristics and in a second direction (e.g., toward +1) for voiced speech or a signal that is periodic. 
         [0168]    In embodiments, signal features are normalized by normalization module  830  to be between a lower bound value and a higher bound value. For example, in an embodiment, each signal feature is normalized between −1 and +1, where a value near −1 is an indication that input signal  802  has unvoiced or noise-like characteristics, and a value near +1 indicates that input signal  802  likely includes voiced speech or a signal that is periodic. 
         [0169]    It should be noted that the normalization techniques provided below are just example ways of performing normalization. They are all basically clipped linear functions. Other normalization techniques may be used in alternative embodiments. For example, one could derive more complicated smooth higher order functions that would approach −1, +1. 
         [0170]      FIG. 13  shows a flowchart  1300  providing example steps for normalizing signal features, according to example embodiments of the present invention. Flowchart  1300  may be performed by normalization module  830 . The steps of flowchart  1300  need not necessarily occur in the order shown in  FIG. 13 . Furthermore, in embodiments, not all steps of flowchart  1300  are necessarily performed. For example, flowchart  1300  relates to the normalization of four features of an audio signal. In alternative embodiments, fewer, additional, and/or alternative features of the audio signal may be normalized. Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. 
         [0171]    Flowchart  1300  is described as follows with respect to  FIG. 14 .  FIG. 14  shows an example block diagram of normalization module  830 , in accordance with an example embodiment of the present invention. As shown in  FIG. 14 , normalization module  830  includes a pitch period change normalization module  1402 , a pitch prediction gain normalization module  1404 , a normalized autocorrelation coefficient normalization module  1406 , and a logarithmic signal gain normalization module  1408 . These modules of normalization module  830  are further described below along with a corresponding step of flowchart  1300 . 
         [0172]    a. Delta Pitch 
         [0173]    In step  1302  of flowchart  1300 , the change in a pitch period is normalized. Pitch period change normalization module  1402  may perform step  1302 . Pitch period change normalization module  1402  receives change in pitch period, pp Δ , on extracted feature signal  806 , and outputs a normalized pitch period change, N_pp Δ , on a normalized feature signal  808 . 
         [0174]    During voiced speech, the pitch changes very slowly from one frame (approx 20 ms frames) to the next, and so pp Δ  should tend to be small. During unvoiced speech, the detected pitch is essentially random, and so pp Δ  should tend to be large. An example pitch period change normalization that may be performed by module  1402  in an embodiment is given by: 
         [0000]        N   —   pp   Δ =(1−min(3 ·pp   Δ ,1))·2−1 
       In other embodiments, other equations for normalizing pitch period change may alternatively be used. 
       [0175]    b. Pitch Prediction Gain 
         [0176]    In step  1304 , the pitch prediction gain is normalized. For example, pitch prediction gain normalization module  1404  may perform step  1304 . Pitch prediction gain normalization module  1404  receives pitch prediction gain, ppg, on extracted feature signal  806 , and outputs a normalized pitch prediction gain, N_ppg , on normalized feature signal  808 . 
         [0177]    During voiced speech, the pitch prediction gain, ppg, will tend to be high, indicating periodicity at the pitch lag. However, during unvoiced speech, there is no periodicity at the pitch lag, and ppg will tend to be low. An example pitch prediction gain normalization that may be performed by module  1404  in an embodiment is given by: 
         [0000]    
       
         
           
             N_ppg 
             = 
             
               
                 
                   max 
                    
                   
                     ( 
                     
                       
                         min 
                          
                         
                           ( 
                           
                             ppg 
                             , 
                             10 
                           
                           ) 
                         
                       
                       , 
                       0 
                     
                     ) 
                   
                 
                 5 
               
               - 
               1 
             
           
         
       
     
       In other embodiments, other equations for normalizing pitch prediction gain may alternatively be used. 
       [0178]    c. First Normalized Autocorrelation Coefficient 
         [0179]    In step  1306 , the first normalized autocorrelation coefficient is normalized. For example, normalized autocorrelation coefficient normalization module  1406  may perform step  1306 . Normalized autocorrelation coefficient normalization module  1406  receives first normalized autocorrelation coefficient, ρ 1 , on extracted feature signal  806 , and outputs a normalized first normalized autocorrelation coefficient, N_ρ 1  on normalized feature signal  808 . 
         [0180]    During voiced speech, the first normalized autocorrelation coefficient, ρ 1 , will tend to be close to +1, whereas for unvoiced speech, ρ 1  will tend to be much less than 1. An example first normalized autocorrelation coefficient normalization that may be performed by module  1406  in an embodiment is given by: 
         [0000]        N _ρ 1 =max(ρ 1 ,0)·2−1 
       In other embodiments, other equations for normalizing the first normalized autocorrelation coefficient may alternatively be used. 
       [0181]    d. Logarithmic Signal Gain 
         [0182]    In step  1308 , the logarithmic signal gain is normalized. For example, logarithmic signal gain normalization module  1408  may perform step  1308 . Logarithmic signal gain coefficient normalization module  1408  receives logarithmic signal gain, lg, on extracted feature signal  806 , and outputs a normalized logarithmic signal gain, N_lg, on normalized feature signal  808 . 
         [0183]    During voiced speech, the logarithmic signal gain, lg, will tend to be high, while during unvoiced speech it will tend to be low. As shown in  FIG. 14 , in an embodiment, logarithmic signal gain normalization module  1408  receives energy tracking signal  804 . LtMaxEst, LtMinEst, and ThActive provided on energy tracking signal  804  are used to normalize the logarithmic signal gain. An example logarithmic signal gain normalization that may be performed by module  1408  in an embodiment is given by: 
         [0000]    
       
         
               
               
             
               
               
             
               
               
             
               
               
             
           
               
                   
                   
               
             
             
               
                   
                 if ((LtMaxEst − LtMinEst) &gt; 6) &amp; (lg &gt; ThActive) 
               
               
                   
                   
               
             
          
           
               
                   
                 
                   
                     
                       
                         N_lg 
                         = 
                         
                           max 
                            
                           
                             ( 
                             
                               
                                 min 
                                  
                                 
                                   ( 
                                   
                                     
                                       
                                         
                                           lg 
                                           - 
                                           
                                             ( 
                                             
                                               LtMaxEst 
                                               - 
                                               10 
                                             
                                             ) 
                                           
                                         
                                         5 
                                       
                                       - 
                                       1 
                                     
                                     , 
                                     1 
                                   
                                   ) 
                                 
                               
                               , 
                               
                                 - 
                                 1 
                               
                             
                             ) 
                           
                         
                       
                     
                   
                 
               
               
                   
                   
               
             
          
           
               
                   
                 else 
               
             
          
           
               
                   
                 N_lg = 0 
               
               
                   
                   
               
             
          
         
       
     
       In other embodiments, other equations for normalizing logarithmic signal gain may alternatively be used. 
     4. Speech Likelihood Measure Module Embodiments 
       [0184]    As shown in  FIG. 8 , speech likelihood measure module  840  receives normalized feature signal  808 . Speech likelihood measure module  840  makes a determination whether speech is likely to have been received on input signal  802 , by calculating one or more speech likelihood measures. 
         [0185]    In an embodiment, a single speech likelihood measure, SLM, is calculated by module  840  by combining the normalized features received on normalized feature signal  808 , as follows: 
         [0000]        SLM=N   —   pp   Δ   +N   —   ppg+N _ρ 1   +N   —   lg.    
         [0000]    In an embodiment, where each normalized feature is in a range (−1 to +1), SLM is in the range {−4 to +4}. Values close to the minimum or maximum values of the range indicate a likelihood that speech is present in input signal  802 , while values close to zero indicate the likelihood of the presence of music or other non-speech signals. 
         [0186]    Note that in alternative embodiments, SLM may have a range other than {−4 to +4}. For example, one or more normalized features in the equation for SLM above may have ranges other than (−1 to +1). Additionally, or alternatively, one or more normalized features in the equation for SLM may be multiplied, divided, or otherwise scaled by a weighting factor, to provide the one or more normalized features with a weight in SLM that is different from one or more of the other normalized features. Such variation in ranges and/or weighting may be used to increase or decrease the importance of one or more of the normalized features in the speech likelihood determination, for example. 
         [0187]    In an embodiment, a number and type of the features are selected to have little or no correlation between normalized features in tending toward the first value or the second value for a typical music audio signal. Enough features are selected such that this random direction tends to cancel the sum SLM when adding the normalized results to generally yield a sum near zero. The normalized features themselves may also generally be close to zero for certain music. For example, in multiple instrument music, a single pitch will give a pitch prediction gain that is low since the single pitch can only track one instrument and the prediction does not necessarily capture the energy in the other instrument (assuming the other instruments are at a different pitch). 
         [0188]    As shown in  FIG. 8 , speech likelihood measure module  840  outputs speech likelihood indicator signal  812 , which includes SLM. 
       5. Long Term Running Average Module Embodiments 
       [0189]    As shown in  FIG. 8 , long term running average module  850  receives speech likelihood indicator signal  812  and energy tracking signal  804 . Long term running average module  850  generates a running average of speech likelihood indicator signal  812 . 
         [0190]    In an embodiment, a long term speech likelihood running average, LTSLM, is generated by module  850  according to the equation: 
         [0000]                                            if (lg &gt; ThActive)             LTSLM = LTSLM * LtslAlpha + |SLM| * (1 − LtslAlpha)                        
where LtslAlpha is a variable that may be set between 0 and 1 (e.g., tuned to 0.99 in one embodiment). As indicated above, in an embodiment, the long term average is updated by module  850  only when an active signal is indicated by ThActive on energy tracking signal  804 . This provides classification robustness during background noise.
 
         [0191]    As shown in  FIG. 8 , long term running average module  850  outputs long term running average signal  814 , which includes LTSLM. 
       6. Classification Module Embodiments 
       [0192]    As shown in  FIG. 8 , classification module  860  receives long term running average signal  814 . Classification module  860  classifies the current frame of input signal  802  as speech or non-speech. 
         [0193]    For example, in an embodiment, the classification, Class(i), for the ith frame is calculated by module  860  according to the equation: 
         [0000]                                            if (Class(i − 1) == SPEECH)             if (LTSLM &gt; 1.75)               Class(i) = SPEECH             else               Class(i) = NONSPEECH           else             if (LTSLM &gt; 1.85)               Class(i) = SPEECH             else               Class(i) = NONSPEECH                        
where Class(i−1) is the classification of the prior (i−1) classified frame of input signal  802 . Threshold values other than 1.75 and 1.85 may alternatively be used by module  860 , in other embodiments.
 
         [0194]    As shown in  FIG. 8 , classification module  860  outputs classification signal  818 , which includes Class(i). Classification signal  818  is received by FLC/decision control logic  140 , shown in  FIG. 1 . 
       7. Example Classifier Process Embodiments 
       [0195]      FIG. 15  shows a flowchart  1500  providing example steps for classifying audio signals as speech or music, according to example embodiments of the present invention. Flowchart  1500  may be performed by signal classifier  130  described above with regard to  FIG. 1 , for example. The steps of flowchart  1500  need not necessarily occur in the order shown in  FIG. 15 . Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. Flowchart  1500  is described as follows. 
         [0196]    Flowchart  1500  begins with step  1502 . In step  1502 , an energy of the audio signal is tracked to determine if the frame of the audio signal comprises an active signal. For example, in an embodiment, energy tracker module  810  performs step  1502 . Furthermore, the steps of flowchart  900  shown in  FIG. 9  may be performed during step  1502 . 
         [0197]    In step  1504 , one or more signal features associated with a frame of the audio signal are extracted. For example, in an embodiment, feature extraction module  820  performs step  1504 . Furthermore, the steps of flowchart  1100  shown in  FIG. 11  may be performed during step  1504 . 
         [0198]    In step  1506 , each feature of the extracted signal features is normalized. For example, in an embodiment, normalization module  830  performs step  1506 . Furthermore, the steps of flowchart  1300  shown in  FIG. 13  may be performed during step  1506 . 
         [0199]    In step  1508 , the normalized features are combined to generate a first measure. For example, in an embodiment, speech likelihood measure module  840  performs step  1508 . In an embodiment, the first measure is the speech likelihood measure, SLM. 
         [0200]    In step  1510 , a second measure is updated based on the first measure. In an embodiment, the second measure comprises a long-term running average of the first measure. For example, in an embodiment, long term running average module  850  performs step  1510 . In an embodiment, the second measure is the long term speech likelihood running average, LTSLM. In an embodiment, step  1510  is performed only if the frame of the audio signal comprises an active signal, as determined by step  1502 . 
         [0201]    In step  1512 , the frame of the audio signal is classified as speech or non-speech based at least in part on the second measure. For example, in an embodiment, classification module  860  performs step  1512 . 
       C. SCALED WINDOW OVERLAP ADD FOR MIXED SIGNALS IN ACCORDANCE WITH AN EMBODIMENT OF THE PRESENT INVENTION 
       [0202]    An embodiment of the present invention uses a dynamic mix of windows to overlap two signals whose normalized cross-correlation may vary from zero to one. If the overlapping signals are decomposed into a correlated component and an uncorrelated component, they are overlap-added separately using the appropriate window, and then added together. If the overlapping signals are not decomposed, a weighted mix of windows is used. The mix is determined by a measure estimating the amount of cross-correlation between overlapping signals, or the relative amount of correlated to uncorrelated signals. 
         [0203]    The following methods are used to perform certain overlap-add operations as described above in Section A in the context of frame loss concealment. For example, in embodiments, the following techniques may be used in step  212  of flowchart  200  in  FIG. 2  and step  512  of flowchart  500  in  FIG. 5 . However, embodiments are not limited to those applications. The example embodiments described herein are provided for illustrative purposes, and are not limiting. Further structural and operational embodiments, including modifications/alterations, will become apparent to persons skilled in the relevant art(s) from the teachings herein. 
         [0204]    Two signals to be overlapped added may be defined as a first signal segment that is to be faded out, and a second signal segment that is to be faded in. For example, the first signal segment may be a first received segment of an audio signal, and the second signal segment may be a second received segment of the audio signal. 
         [0205]    A general overlap-add of the two signals can be defined by: 
         [0000]        s ( n )= s   out ( n )· w   out ( n )+ s   in ( n )· w   in ( n )  n =0 ..N− 1 
         [0000]    where s out  is the signal to be faded out, s in  is the signal to be faded in, w out  is a fade-out window, w in  is the fade-in window, and N is the overlap-add window length. 
         [0206]    Let the overlap-add window for correlated signals be denoted wc and have the property: 
         [0000]        wc   out ( n )+ wc   in ( n )=1  n =0 ..N −1 
         [0207]    Let the overlap-add window for uncorrelated signals be denoted wu and have the property: 
         [0000]        wu   out   2 ( n )+ wu   in   2 ( n )=1  n =0 ..N −1 
       1. First Embodiment: Overlapping Decomposed Signals with Decomposed Signals 
       [0208]    In this embodiment, the signals for overlapping are decomposed into a correlated component, sc out  and sc in , and an uncorrelated component, su out  and su in . The overlapped signal s(n) is then given by the following equation (Equation C.1): 
         [0000]    
       
         
               
               
               
             
           
               
                   
                   
               
             
             
               
                   
                 s(n) = [sc out (n)·wc out (n)+sc in (n)·wc in (n)]+ 
                 n = 0..N−1 
               
               
                   
                   [su out (n)·wu out (n)+su in (n)·wu in (n)] 
               
               
                   
                   
               
             
          
         
       
     
         [0209]      FIG. 16  shows a flowchart  1600  providing example steps for overlapping a first decomposed signal with a second decomposed signal according to the above Equation C.1. The steps of flowchart  1600  need not necessarily occur in the order shown in  FIG. 16 . Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. For example,  FIG. 17  shows a system  1700  configured to implement Equation C.1, according to an embodiment of the present invention. Flowchart  1600  is described as follows with respect to  FIG. 17 , for illustrative purposes. 
         [0210]    Flowchart  1600  begins with step  1602 . In step  1602 , a correlated component of the first segment is added to a correlated component of the second segment to generate a combined correlated component. For example, as shown in  FIG. 17 , the correlated component of the first segment, sc out , is multiplied with a correlated fade-out window, wc out , by a first multiplier  1702 , to generate a first product. The correlated component of the second segment, sc in , is multiplied with a correlated fade-in window, wc in , by a second multiplier  1704 , to generate a second product. The first product is added to the second product by a first adder  1710  to generate the combined correlated component, sc out (n)·wc out (n)+sc in (n)·wc in (n). 
         [0211]    In step  1604 , an uncorrelated component of the first segment is added to an uncorrelated component of the second segment to generate a combined uncorrelated component. For example, as shown in  FIG. 17 , the uncorrelated component of the first segment, su out , is multiplied with an uncorrelated fade-out window, wu out , by third multiplier  1706 , to generate a first product. The uncorrelated component of the second segment, su in , is multiplied with an uncorrelated fade-in window, wu in , by fourth multiplier  1708 , to generate a second product. The first product is added to the second product by a second adder  1712  to generate the combined uncorrelated component su out (n)·wu out (n)+su in (n)·wu in (n). 
         [0212]    In step  1606 , the combined correlated component is added to the combined uncorrelated component to generate an overlapped signal. For example, as shown in  FIG. 17 , the combined correlated component is added to the combined uncorrelated component by third adder  1714 , to generate the overlapped signal, shown as signal  1716 . 
         [0213]    Note that first through fourth multipliers  1702 ,  1704 ,  1706 , and  1708 , and first through third adders  1710 ,  1712 , and  1714 , and further multipliers and adders described in Section C., may be implemented in hardware, software, firmware, or any combination thereof, including respectively as sequence multipliers and adders that are well known to persons skilled in the relevant art(s). For example, such multipliers and adders may be implemented in logic, such as a programmable logic chip (PLC), in a programmable gate array (PGA), in a digital signal processor (DSP), as software instructions that execute in a processor, etc. 
       2. Second Embodiment: Overlapping a Mixed Signal with a Decomposed Signal 
       [0214]    In this embodiment, one of the overlapping signals (in or out) is decomposed while the other signal has the correlated and uncorrelated components mixed together. Ideally, the mixed signal is first decomposed and the first embodiment described above is used. However, signal decomposition is very complex and overkill for most applications. Instead, the optimal overlapped signal may be approximated by the following equation (Equation C.2.a): 
         [0000]                                                s(n) = [S out (n)·wc out (n)]·β+sc in (n)·wc in (n)+   n= 0..N−1             [(s out (n)·wc out (n)]·(1−β)+su in (n)·wu in (n)                        
here β is the desired fraction of correlated signal in the final overlapped signal s(n), or an estimate of the cross-correlation between s out  and sc in +su in . The above formulation is given for a mixed s out  signal and decomposed s in  signal. A similar formulation for the opposite case, where s out  is decomposed and s in  is mixed, is provided by the following equation (Equation C.2.b):
 
         [0000]    
       
         
               
               
               
             
           
               
                   
                   
               
             
             
               
                   
                 s(n) = sc out (n)·wc out (n)+[s in (n)·wc in (n)]·β+ 
                 n = 0..N−1 
               
               
                   
                   sc out (n)·wu out (n)+[s in (n)·wu in (n)]·(1−β) 
               
               
                   
                   
               
             
          
         
       
     
         [0215]    Notice that for both formulations, if the signals are completely correlated (β=1) or completely uncorrelated (β=0), each solution is optimal. 
         [0216]      FIG. 18  shows a flowchart  1800  providing example steps for overlapping a first signal with a second signal according to the above equation. The steps of flowchart  1800  need not necessarily occur in the order shown in  FIG. 18 . Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. For example,  FIG. 19  shows a system  1900  configured to implement the above Equation C.2.a, according to an embodiment of the present invention. It is noted that it will be apparent to persons skilled in the relevant art(s) how to reconfigure system  1900  to implement Equation C.2.b provided above. Flowchart  1800  is described as follows with respect to  FIG. 19 , for illustrative purposes. 
         [0217]    Flowchart  1800  begins with step  1802 . In step  1802 , the first segment is multiplied by an estimate β of the correlation between the first segment and the second segment to generate a first product. For example, as shown in  FIG. 19 , the first segment, s out , is multiplied with a correlated fade-out window, wc out , by a first multiplier  1902 , to generate a third product, s out (n)·wc out (n). The third product is multiplied with β by a second multiplier  1904  to generate the first product. 
         [0218]    In step  1804 , the first product is added to a correlated component of the second segment to generate a combined correlated component. For example, as shown in  FIG. 19 , the correlated component of the second segment, sc in (n), is multiplied with a correlated fade-in window, wc in (n), by a third multiplier  1906 , to generate a fourth product, sc in (n)·wc in (n). The first product is added to the fourth product by a first adder  1914  to generate the combined correlated component. 
         [0219]    In step  1806 , the first segment is multiplied by (1−β) to generate a second product. For example, the first segment, s out , is multiplied with an uncorrelated fade-out window, wu out (n), by a fourth multiplier  1908 , to generate a fifth product, s out (n)·wu out (n). The fifth product is multiplied with (1−β) by a fifth multiplier  1910  to generate the second product. 
         [0220]    In step  1808 , the second product is added to an uncorrelated component of the second segment to generate a combined uncorrelated component. For example, the uncorrelated component of the second segment, su in (n), is multiplied with an uncorrelated fade-in window, wu in (n), by a sixth multiplier  1912 , to generate a sixth product, su in (n)·wu in (n). The second product is added to the sixth product by a second adder  1916  to generate the combined uncorrelated component. 
         [0221]    In step  1810 , the combined correlated component is added to the combined uncorrelated component to generate an overlapped signal. For example, as shown in  FIG. 19 , the combined correlated component is added to the combined uncorrelated component by a third adder  1918 , to generate the overlapped signal, shown as signal  1920 . 
       3. Third Embodiment: Overlapping a Mixed Signal with a Mixed Signal 
       [0222]    In this embodiment, both overlapping signals are not decomposed. Once again, a desired solution is to decompose both signals and use the first embodiment of subsection C.1 above. However, for most applications, this is not required. In an embodiment, an adequate compromise solution is given by the following equation (Equation C.3): 
         [0000]                                                s(n) = [s out (n)·wc out (n)+s in (n)·wc in (n)]·β+   n = 0..N−1             [s out (n)·wu out (n)+s in (n)·wu in (n)]·(1−β)                        
where β is an estimate of the cross-correlation between s out  and s in . Again, notice that if the signals are completely correlated (β=1) or completely uncorrelated (β=0), the solution is optimal.
 
         [0223]      FIG. 20  shows a flowchart  2000  providing example steps for overlapping a mixed first signal with a mixed second signal according to the above Equation C.3. The steps of flowchart  2000  need not necessarily occur in the order shown in  FIG. 20 . Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. For example,  FIG. 21  shows a system  2100  configured to implement Equation C.3, according to an embodiment of the present invention. Flowchart  2000  is described as follows with respect to  FIG. 21 , for illustrative purposes. 
         [0224]    Flowchart  2000  begins with step  2002 . In step  2002 , the first segment is added to the second segment to generate a first combined component. For example, as shown in  FIG. 21 , the first segment, s out (n), is multiplied with a correlated fade-out window, wc out (n), by a first multiplier  2102 , to generate a third product, s out (n)·wc out (n). The second segment, s in (n), is multiplied with a correlated fade-in window, wc in (n), by a second multiplier  2104 , to generate a fourth product, s in (n)·wc in (n). The third product is added to the fourth product by a first adder  2110  to generate the first combined component. 
         [0225]    In step  2004 , the first combined component is multiplied by an estimate β of the correlation between the first segment and the second segment to generate a first product. For example, as shown in  FIG. 21 , the first combined component is multiplied with β by a third multiplier  2114  to generate the first product. 
         [0226]    In step  2006 , the first segment is added to the second segment to generate a second combined component. For example, as shown in  FIG. 21 , the first segment, s out (n), is multiplied with an uncorrelated fade-out window, wu out (n), by a fourth multiplier  2106 , to generate a fifth product. The second segment, s in (n), is multiplied with an uncorrelated fade-in window, wu in (n), by a fifth multiplier  2108 , to generate a sixth product, s in (n)·wu in (n). The fifth product is added to the sixth product by a second adder  2112  to generate the second combined component. 
         [0227]    In step  2008 , the second combined component is multiplied by (1−β) to generate a second product. For example, as shown in  FIG. 21 , the second combined component is multiplied with (1−β) by a sixth multiplier  2116  to generate the second product. 
         [0228]    In step  2010 , the first product is added to the second product to generate an overlapped signal. For example, as shown in  FIG. 21 , the first product is added to the second product by third adder  2118 , to generate the overlapped signal, shown as signal  2120 . 
       D. DECIMATED BISECTIONAL PITCH REFINEMENT IN ACCORDANCE WITH AN EMBODIMENT OF THE PRESENT INVENTION 
       [0229]    Embodiments for determining pitch period are described below. Such embodiments may be used by processing block  161  shown in  FIG. 1 , and described above in Section A. However, embodiments are not limited to that application. The example embodiments described herein are provided for illustrative purposes, and are not limiting. Further structural and operational embodiments, including modifications/alterations, will become apparent to persons skilled in the relevant art(s) from the teachings herein. 
         [0230]    An embodiment of the present invention uses the following procedure to refine a pitch period estimate based on a coarse pitch. The normalized correlation at the coarse pitch lag is calculated and used as a current best candidate. The normalized correlation is then evaluated at the midpoint of the refinement pitch range on either side of the current best candidate. If the normalized correlation at either midpoint is greater than the current best lag, the midpoint with the maximum correlation is selected as the current best lag. After each iteration, the refinement range is decreased by a factor of two and centered on the current best lag. This bisectional search continues until the pitch has been refined to an acceptable tolerance or until the refinement range has been exhausted. During each step of the bisectional pitch refinement, the signal is decimated to reduce the complexity of computing the normalized correlation. The decimation factor is chosen such that enough time resolution is still available to select the correct lag at each step. Hence, the decimated signal contains increasing time resolution as the bisectional search refines the pitch and reduces the search range. 
         [0231]      FIG. 22  shows a flowchart  2200  providing example steps for determining a pitch period of an audio signal, according to an example embodiment of the present invention. Flowchart  2200  may be performed by processing block  161 , for example. Other structural and operational embodiments will be apparent to persons skilled in the relevant art(s) based on the discussion provided herein. Flowchart  2200  is described as follows with respect to  FIG. 23 .  FIG. 23  shows block diagram of a pitch refinement system  2300  in accordance with an example embodiment of the present invention. As shown in  FIG. 23 , pitch refinement system  2300  includes a search range calculator module  2310 , a decimation factor calculator module  2320 , and a decimated bisectional search module  2330 . Note that modules  2310 ,  2320 , and  2330  may be implemented in hardware, software, firmware, or any combination thereof. For example, modules  2310 ,  2320 , and  2330  may be implemented in logic, such as a programmable logic chip (PLC), in a programmable gate array (PGA), in a digital signal processor (DSP), as software instructions that execute in a processor, etc. 
         [0232]    Flowchart  2200  begins with step  2202 . In step  2202 , a coarse pitch lag associated with the audio signal is set as a best pitch lag. The initial pitch estimate, also referred to as a “coarse pitch,” is denoted P 0 . The coarse pitch may be a pitch value from a prior received signal frame used as a best pitch lag estimate, or the coarse pitch may be obtained by other ways. 
         [0233]    In step  2204 , a normalized correlation associated with the coarse pitch lag is set as a best normalized correlation. In an embodiment, the normalized correlation at P 0  is denoted by c(P 0 ), and is calculated according to: 
         [0000]    
       
         
           
             
               c 
                
               
                 ( 
                 k 
                 ) 
               
             
             = 
             
               
                 
                   ∑ 
                   
                     n 
                     = 
                     1 
                   
                   M 
                 
                  
                 
                   
                     x 
                      
                     
                       ( 
                       n 
                       ) 
                     
                   
                    
                   
                     x 
                      
                     
                       ( 
                       
                         n 
                         - 
                         k 
                       
                       ) 
                     
                   
                 
               
               
                 
                   
                     
                       ∑ 
                       
                         n 
                         = 
                         1 
                       
                       M 
                     
                      
                     
                       
                         x 
                         2 
                       
                        
                       
                         ( 
                         n 
                         ) 
                       
                     
                   
                 
                  
                 
                   
                     
                       ∑ 
                       
                         n 
                         = 
                         1 
                       
                       M 
                     
                      
                     
                       
                         x 
                         2 
                       
                        
                       
                         ( 
                         
                           n 
                           - 
                           k 
                         
                         ) 
                       
                     
                   
                 
               
             
           
         
       
     
         [0000]    where M is the pitch analysis window length. The parameters P 0  and c(P 0 ) are assumed to be available before the pitch refinement is performed in subsequent steps. The normalized correlation may be calculated by one of modules  2310 ,  2320 ,  2330  or other module not shown in  FIG. 23  (e.g., a normalized correlation calculator module). 
         [0234]    In step  2206 , a refinement pitch range is calculated. For example, search range calculator module  2310  shown in  FIG. 23  calculates the search range for the current iteration. As shown in  FIG. 23 , search range calculator  2310  receives P 0  and c(P 0 ). The initial search range is selected while considering the accuracy of the initial pitch estimate. In an embodiment, the initial range Δ 0  is chosen as follows: 
         [0000]      Δ 0 =└(1+|( P   ideal   −P   0 )|/2)┘ 
         [0000]    where P ideal  is the ideal pitch. Then for each iteration, in an embodiment, a range for the iteration (i) is calculated based on the previous iteration (i−1) according to: 
         [0000]      Δ i =└Δ i−1 /2┘. 
       In other embodiments, Δ i−1  may be divided by factors other than 2 to determine Δ i . As shown in FIG. 23, search range calculator module  2310  outputs Δ i . 
       [0235]    In step  2208 , a normalized correlation is calculated at a first midpoint of the refinement pitch range preceding the best pitch lag and at a second midpoint of the refinement pitch range following the best pitch lag. In an embodiment, a decimated bisectional search is conducted to hone in a best pitch lag. As shown in  FIG. 23 , decimation factor calculator module  2320  receives Δ i . Decimation factor calculator module  2320  calculates a decimation factor, D, according to: 
         [0000]      D i ≦Δ i . 
         [0000]    If D i &gt;Δ i  then the time resolution of decimated signal is not sufficient to guarantee convergence of the bisectional search. As shown in  FIG. 23 , decimation factor calculator module  2320  outputs decimation factor D. 
         [0236]    As shown in  FIG. 23 , decimated bisectional search module  2330  receives decimation factor D, P i−1 , and c(P i−1 ). Decimated bisectional search module  2330  performs the decimated bisectional search. In an embodiment, decimated bisectional search module  2330  performs the steps of flowchart  2400  shown in  FIG. 24  to perform step  2208  of  FIG. 22 . 
         [0237]    In step  2402 , set P i =P i−1  and c(P i )=c(P i−1 ). 
         [0238]    In step  2404 , decimate the signal x(n). Let D(·) represent a decimator with decimation factor D. Then 
         [0000]        xd ( m )= D ( x ( n )). 
         [0239]    In step  2406 , decimate the signal x(n−k) for k=Δ i : 
         [0000]        xd   k ( m )= D ( x ( n−k )). 
         [0240]    In step  2408 , calculate the normalized correlation for the decimated signals. For example, the normalized correlation may be calculated according to: 
         [0000]    
       
         
           
             
               
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         [0241]    In step  2410 , repeat steps  2406  and  2408  for k=−Δ i . 
         [0242]    In step  2210  shown in  FIG. 22 , the normalized correlation at each of the first and second midpoints is compared to the best normalized correlation. In step  2212 , responsive to a determination that the normalized correlation at either of the first and second midpoints is greater than the best normalized correlation, the greatest normalized correlation associated with each of the first and second midpoints is set to the best normalized correlation and the midpoint associated with the greatest normalized correlation is set to the best pitch lag. 
         [0243]    In an embodiment, decimated bisectional search module  2330  performs steps  2210  and  2212  as follows. Separately for both of k=Δ i  and k=−Δ i , the correlation results of step  2408  are compared as follows, and an update to best normalized correlation and midpoint is made if necessary, as follows: 
         [0000]      If  c   d ( k )&gt; c ( P   i ) then  c ( P   i )= c   d ( k ) and  P   i   =P   i−1   +k    
         [0244]    In step  2214 , for one or more additional iterations, a new refinement pitch range is calculated and steps  2208 ,  2210 , and  2212  are repeated. Step  2214  may perform as many additional iterations as necessary, until no further decimation is practical, until an acceptable pitch value is determined, etc. As shown in  FIG. 23 , decimated bisectional search module  2330  outputs pitch estimate P i . 
         [0245]    In steps  2404  and  2406  of flowchart  2400 , the input signal and a shifted version of the input signal are decimated. In a traditional decimator, the signal is first lowpass filtered in order to avoid aliasing in the decimated domain. To reduce complexity, the lowpass filtering step may be omitted and still achieve near equivalent results, especially in voiced speech where the signal is generally lowpass. The aliasing rarely alters the normalized correlation enough to affect the result of the search. In this case, the decimated signal is given by: 
         [0000]        xd ( m )= x ( m·D ) 
         [0000]    and 
         [0000]    
       
         
           
             
               
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         [0246]    An example of the iterative process of flowchart  2200  is illustrated in  FIGS. 25A-25D .  FIGS. 25A-25D  show plots of normalized correlation values (c d (k)) versus values of k. For the initial conditions of the search, P 0 =Δ 0 =16, and c d (P 0 ) is calculated. 
         [0247]    In the first iteration shown in  FIG. 25A , Δ i =D i =8, and c d (P 0 ±8) is evaluated on the decimated signal. The time resolution of the decimated correlation is noted by the darkened sample points. The candidate that maximizes c d (k) is P 0 −8 and is selected as P 1 . 
         [0248]    In the second iteration, shown in  FIG. 25B , Δ i =D i =4, and the search is centered around P 1 . This time, neither candidate at c d (P 1 ±4) is greater than c d (P 1 ), and so P 2 =P 1 . 
         [0249]    In the third iteration, shown in  FIG. 25C , Δ i =D i =2, and the search is centered around P 2  (P 1 ). The candidate that maximizes c d (k) is P 2 +2, and is selected as P 3 . 
         [0250]    In the fourth iteration, shown in  FIG. 25D , Δ i =D i =1 (hence no decimation) and the search is centered around P 3 . The candidate at P 0 −7 (P 3 −1) maximizes c d (k), and is selected as the final pitch value. 
         [0251]    Note that the process of flowchart  2200  shown in  FIG. 22  may be adapted to determining/refining parameters other than just a pitch period parameter. For example, in a process for refining a parameter (e.g., a generic parameter “Q”) of a signal, an adapted step  2202  may include setting a coarse value for the parameter associated with the signal to a best parameter value. An adapted step  2204  may include setting a value of a function f(Q) associated with the coarse parameter value as a best function value. An adapted step  2206  may include calculating a refinement parameter range. An adapted step  2208  may include calculating a value of the function f(Q) at a first midpoint of the refinement parameter range preceding the best parameter value and at a second midpoint of the refinement parameter range following the best parameter value. An adapted step  2210  may include comparing the calculated function value at each of the first and second midpoints to the best function value. An adapted step  2212  may include, responsive to a determination that the calculated function value at either of the first and second midpoints is better than the best function value, setting the better function value associated with each of the first and second midpoints to the best function value and setting the midpoint associated with the better function value to the best parameter value. 
         [0252]    Flowchart  2200  may be adapted in this manner just described, or in other ways, to determine/refine a variety of signal parameters, as would be known to persons skilled in the relevant art(s) from the teachings herein. For example, the bisectional decimation techniques described further above may be applied to the just described process of determining/refining parameters other than just a pitch period parameter. For example, the adapted step  2208  may include decimating the signal prior to computing a value of the function f(Q) at the midpoint of the refinement parameter range to either side of the best parameter value. This process of decimation may include calculating a decimation factor, where the decimation factor is less than or equal to the refinement parameter range. The techniques of bisectional decimation described herein may be further adapted to the present example of determining/refining parameters, as would be apparent to persons skilled in the relevant art(s) from the teachings herein. 
       E. HARDWARE AND SOFTWARE IMPLEMENTATIONS 
       [0253]    The following description of a general purpose computer system is provided for the sake of completeness. The present invention can be implemented in hardware, or as a combination of software and hardware. Consequently, the invention may be implemented in the environment of a computer system or other processing system. An example of such a computer system  2600  is shown in  FIG. 26 . In the present invention, all of the processing blocks or steps of  FIGS. 1-24 , for example, can execute on one or more distinct computer systems  2600 , to implement the various methods of the present invention. The computer system  2600  includes one or more processors, such as processor  2604 . Processor  2604  can be a special purpose or a general purpose digital signal processor. The processor  2604  is connected to a communication infrastructure  2602  (for example, a bus or network). Various software implementations are described in terms of this exemplary computer system. After reading this description, it will become apparent to a person skilled in the relevant art(s) how to implement the invention using other computer systems and/or computer architectures. 
         [0254]    Computer system  2600  also includes a main memory  2606 , preferably random access memory (RAM), and may also include a secondary memory  2620 . The secondary memory  2620  may include, for example, a hard disk drive  2622  and/or a removable storage drive  2624 , representing a floppy disk drive, a magnetic tape drive, an optical disk drive, or the like. The removable storage drive  2624  reads from and/or writes to a removable storage unit  2628  in a well known manner. Removable storage unit  2628  represents a floppy disk, magnetic tape, optical disk, or the like, which is read by and written to by removable storage drive  2624 . As will be appreciated, the removable storage unit  2628  includes a computer usable storage medium having stored therein computer software and/or data. 
         [0255]    In alternative implementations, secondary memory  2620  may include other similar means for allowing computer programs or other instructions to be loaded into computer system  2600 . Such means may include, for example, a removable storage unit  2630  and an interface  2626 . Examples of such means may include a program cartridge and cartridge interface (such as that found in video game devices), a removable memory chip (such as an EPROM, or PROM) and associated socket, and other removable storage units  2630  and interfaces  2626  which allow software and data to be transferred from the removable storage unit  2630  to computer system  2600 . 
         [0256]    Computer system  2600  may also include a communications interface  2640 . Communications interface  2640  allows software and data to be transferred between computer system  2600  and external devices. Examples of communications interface  2640  may include a modem, a network interface (such as an Ethernet card), a communications port, a PCMCIA slot and card, etc. Software and data transferred via communications interface  2640  are in the form of signals which may be electronic, electromagnetic, optical, or other signals capable of being received by communications interface  2640 . These signals are provided to communications interface  2640  via a communications path  2642 . Communications path  2642  carries signals and may be implemented using wire or cable, fiber optics, a phone line, a cellular phone link, an RF link and other communications channels. 
         [0257]    As used herein, the terms “computer program medium” and “computer usable medium” are used to generally refer to media such as removable storage units  2628  and  2630 , a hard disk installed in hard disk drive  2622 , and signals received by communications interface  2640 . These computer program products are means for providing software to computer system  2600 . 
         [0258]    Computer programs (also called computer control logic) are stored in main memory  2606  and/or secondary memory  2620 . Computer programs may also be received via communications interface  2640 . Such computer programs, when executed, enable the computer system  2600  to implement the present invention as discussed herein. In particular, the computer programs, when executed, enable the processor  2600  to implement the processes of the present invention, such as any of the methods described herein. Accordingly, such computer programs represent controllers of the computer system  2600 . Where the invention is implemented using software, the software may be stored in a computer program product and loaded into computer system  2600  using removable storage drive  2624 , interface  2626 , or communications interface  2640 . 
         [0259]    In another embodiment, features of the invention are implemented primarily in hardware using, for example, hardware components such as Application Specific Integrated Circuits (ASICs) and gate arrays. Implementation of a hardware state machine so as to perform the functions described herein will also be apparent to persons skilled in the relevant art(s). 
       F. CONCLUSION 
       [0260]    While various embodiments of the present invention have been described above, it should be understood that they have been presented by way of example, and not limitation. It will be apparent to persons skilled in the relevant art that various changes in form and detail can be made therein without departing from the spirit and scope of the invention. 
         [0261]    The present invention has been described above with the aid of functional building blocks and method steps illustrating the performance of specified functions and relationships thereof. The boundaries of these functional building blocks and method steps have been arbitrarily defined herein for the convenience of the description. Alternate boundaries can be defined so long as the specified functions and relationships thereof are appropriately performed. Any such alternate boundaries are thus within the scope and spirit of the claimed invention. One skilled in the art will recognize that these functional building blocks can be implemented by discrete components, application specific integrated circuits, processors executing appropriate software and the like or any combination thereof. 
         [0262]    Furthermore, the description of the present invention provided herein references various numerical values, such as various minimum values, maximum values, threshold values, ranges, and the like. It is to be understood that such values are provided herein by way of example only and that other values may be used within the scope and spirit of the present invention. 
         [0263]    In accordance with the foregoing, the breadth and scope of the present invention should not be limited by any of the above-described exemplary embodiments, but should be defined only in accordance with the following claims and their equivalents.