Abstract:
An approach to reduce the quality impact due to lost voiced frame data is presented. The decoder reconstructs the lost frame using the pitch track from a directly prior frame. When the decoder receives the next frame data, it makes a copy of the reconstructed frame data and continuously time warping it and the received frame data so that the peaks of their pitch cycles coincide. Subsequently, the decoder fades out the time-warped reconstructed frame data while fading in the time-warped received frame data. Meanwhile, the endpoint of the received frame data remains fixed to preclude discontinuity with the subsequent frame.

Description:
RELATED APPLICATIONS 
     The present application claims the benefit of U.S. provisional application Ser. No. 60/455,435, filed Mar. 15, 2003, which is hereby fully incorporated by reference in the present application. 
     U.S. patent application Ser. No. 10/799,533, “SIGNAL DECOMPOSITION OF VOICED SPEECH FOR CELP SPEECH CODING.” 
     U.S. patent application Ser. No. 10/799,503, “VOICING INDEX CONTROLS FOR CELP SPEECH CODING.” 
     U.S. patent application Ser. No. 10/799,505, “SIMPLE NOISE SUPPRESSION MODEL.” 
     U.S. patent application Ser. No. 10/799,460, “ADAPTIVE CORRELATION WINDOW FOR OPEN-LOOP PITCH.” 
    
    
     BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates generally to speech coding and, more particularly, to recovery of erased voice frames during speech decoding. 
     2. Related Art 
     From time immemorial, it has been desirable to communicate between a speaker at one point and a listener at another point. Hence, the invention of various telecommunication systems. The audible range (i.e. frequency) that can be transmitted and faithfully reproduced depends on the medium of transmission and other factors. Generally, a speech signal can be band-limited to about 10 kHz without affecting its perception. However, in telecommunications, the speech signal bandwidth is usually limited much more severely. For instance, the telephone network limits the bandwidth of the speech signal to between 300 Hz to 3400 Hz, which is known in the art as the “narrowband”. Such band-limitation results in the characteristic sound of telephone speech. Both the lower limit at 300 Hz and the upper limit at 3400 Hz affect the speech quality. 
     In most digital speech coders, the speech signal is sampled at 8 kHz, resulting in a maximum signal bandwidth of 4 kHz. In practice, however, the signal is usually band-limited to about 3600 Hz at the high-end. At the low-end, the cut-off frequency is usually between 50 Hz and 200 Hz. The narrowband speech signal, which requires a sampling frequency of 8 kb/s, provides a speech quality referred to as toll quality. Although this toll quality is sufficient for telephone communications, for emerging applications such as teleconferencing, multimedia services and high-definition television, an improved quality is necessary. 
     The communications quality can be improved for such applications by increasing the bandwidth. For example, by increasing the sampling frequency to 16 kHz, a wider bandwidth, ranging from 50 Hz to about 7000 Hz can be accommodated. This bandwidth range is referred to as the “wideband”. Extending the lower frequency range to 50 Hz increases naturalness, presence and comfort. At the other end of the spectrum, extending the higher frequency range to 7000 Hz increases intelligibility and makes it easier to differentiate between fricative sounds. 
     The frame may be lost because of communication channel problems that results in a bitstream or a bit package of the coded speech being lost or destroyed. When this happens, the decoder must try to recover the speech from available information in order to minimize the impact on the perceptual quality of speech being reproduced. 
     Pitch lag is one of the most important parameters for voiced speech, because the perceptual quality is very sensitive to pitch lag. To maintain good perceptual quality, it is important to properly recover the pitch track at the decoder. Thus, a traditional practice is that if the current voiced frame bitstream is lost, pitch lag is copied from the previous frame and the periodic signal is constructed in terms of the estimated pitch track. However, if the next frame is properly received, there is a potential for quality impact because of discontinuity introduced by the previously lost frame. 
     The present invention addresses the impact in perceptual quality due to discontinuities produced by lost frames. 
     SUMMARY OF THE INVENTION 
     In accordance with the purpose of the present invention as broadly described herein, there is provided systems and methods for recovering an erased voice frame to minimize degradation in perceptual quality of synthesized speech. 
     In one embodiment, the decoder reconstructs the lost frame using the pitch track from the directly prior frame. When the decoder receives the next frame data, it makes a copy of the reconstructed frame data and continuously time warping it and the next frame data so that the peaks of their pitch cycles coincide. Subsequently, the decoder fades out the time-warped reconstructed frame data while fading in the time-warped next frame data. Meanwhile, the endpoint of the next frame data remains fixed to preclude discontinuity with the subsequent frame. 
     These and other aspects of the present invention will become apparent with further reference to the drawings and specification, which follow. It is intended that all such additional systems, methods, features and advantages be included within this description, be within the scope of the present invention, and be protected by the accompanying claims. 
    
    
     
       BRIEF DESCRIPTION OF DRAWINGS 
         FIG. 1  is an illustration of the time domain representation of a coded voiced speech signal at the encoder. 
         FIG. 2  is an illustration of the time domain representation of the coded voiced speech signal of  FIG. 1 , as received at the decoder. 
         FIG. 3  is an illustration of the discontinuity in the time domain representation of the coded voiced speech signal after recovery of a lost frame. 
         FIG. 4  is an illustration of the time warping process in accordance with an embodiment of the present invention. 
         FIG. 5  illustrates real-time voiced frame recovery in accordance with an embodiment of the present invention. 
     
    
    
     DETAILED DESCRIPTION 
     The present application may be described herein in terms of functional block components and various processing steps. It should be appreciated that such functional blocks may be realized by any number of hardware components and/or software components configured to perform the specified functions. For example, the present application may employ various integrated circuit components, e.g., memory elements, digital signal processing elements, transmitters, receivers, tone detectors, tone generators, logic elements, and the like, which may carry out a variety of functions under the control of one or more microprocessors or other control devices. Further, it should be noted that the present application may employ any number of conventional techniques for data transmission, signaling, signal processing and conditioning, tone generation and detection and the like. Such general techniques that may be known to those skilled in the art are not described in detail herein. 
       FIG. 1  is an illustration of the time domain representation of a coded voiced speech signal at the encoder. As illustrated, the voiced speech signal is separated into frames (e.g. frames  101 ,  102 ,  103 ,  104 , and  105 ) before coding. Each frame may contain any number of pitch cycles (i.e. illustrated as big mounds). Each frame is transmitted from the encoder to the receiver as a bitstream after coding. Thus, for example, frame  101  is transmitted to the receiver at t n−1 , frame  102  at t n , frame  103  at t n+1 , frame  104  at t n+2 , frame  105  at t n+3 , and so on. 
       FIG. 2  is an illustration of the time domain representation of the coded voiced speech signal of  FIG. 1 , as received at the decoder. As illustrated, frame  101  arrives properly at the decoder as frame  201 ; Frame  103  arrives properly at the decoder as frame  203 ; Frame  104  arrives properly at the decoder as frame  204 ; and Frame  105  arrives properly at the decoder as frame  205 . However, frame  102  does not arrive at the decoder because it was lost in transmission. Thus, frame  202  is blank. 
     To maintain perceptual quality, frame  202  must be reproduced at the decoder in real-time. Thus frame  201  is copied into frame  202  slot as frame  201 A. However, as shown in  FIG. 3 , a discontinuity may exist at the intersection of frames  201 A and  203  (i.e. point  301 ) because the previous pitch track (i.e. frame  201 A) is likely not accurate . This is because frame  203  was properly received thus its pitch track is correct. But since frame  201 A is a reproduced frame  201 , its endpoint may not coincide with the beginning point of correct frame  203  thus creating a discontinuity that may affect perceptual quality. 
     Thus, although frame  201 A is likely incorrect, it may no longer be modified since it has already been synthesized (i.e. its time has passed and the frame has been sent out). The discontinuity at  301  created by the lost frame may produce an audible reproduction at the beginning of the next frame that is annoying. 
     Embodiments of the present invention use continuous time warping to minimize impact on perceptual quality. Time warping involves mainly modifying or shifting the signals to minimize the discontinuity at the beginning of the frame and also improve the perceptual quality of the frame. The process is illustrated using  FIG. 4  and  FIG. 5 . As illustrated in  FIG. 4 , time history  420  is the actual received data (see  FIG. 2 ) showing the lost frame  202 . Time history  410  is a pseudo received data constructed from the received data. Time history  410  is constructed in real-time by placing a copy of received frame  201  into frame slot  202  as frame  201 A and into frame slot  203  as frame  201 B. Note that frame  203 , frame  204 , and frame  205  arrive properly in real-time and are correctly received in this illustration. 
     The process involves continuously time warping frames  201 B of  410  and frame  203  of  420  so that their peaks,  411  and  421 , coincide in time while maintaining the intersection point (e.g. endpoint  422 ) between frames  203  and  204  fixed. For instance, peak  411  may be stretched forward (as illustrated by arrow  414 ) in time by some delta while peak  421  is stretched backward (as illustrated by arrow  424 ) in time. The intersection point  422  must be maintained because the next frame (e.g.  204 ) may be a correct frame and it is desired to keep continuity between the current frame and the correct next frame, as in this illustration. After time-warping, an overlap-add of the two signals of the warped frames may be used to create the new frame. Line  413  fades out the reconstructed previous frame while line  423  fades in the current frame. The sum of curves  413  and  423  has a magnitude of one at all points in time.  FIG. 5  illustrates real-time voiced frame recovery in accordance with an embodiment of the present invention. 
     As illustrated in  FIG. 5 , a current frame of voiced data is received in block  502 . A determination is made in block  504  whether the frame is properly received. If not, the previous frame data is used to reconstruct the current frame data in block  506  and processing returns back to block  502  to receive the next frame data. If, on the other hand, the current frame data is properly received (as determined in block  504 ), further determination is made in block  508  whether the previous frame was lost, i.e., reconstructed. If the previous frame was not lost, the decoder proceeds to use the current frame data in block  510  and then returns back to block  502  to receive the next frame data. 
     If, on the other hand, the previous frame data was lost received (as determined in block  508 ) and the current frame data is properly received, then time warping is necessary. In block  512 , the pitch of the current frame and that of the reconstructed frame is time-warped so that they will coincide. During time-warping, the end-point of the current frame is maintained because the next frame may be a correct frame. 
     After the frames are time warped in block  512 , the time-warped current frame is faded in while the time-warped reconstructed frame is faded out in block  514 . The combined fade-in and fade-out process (over-lap-add process) may take on the form of the following equation:
 
NewFrame( n )=ReconstFrame( n ).[1− a ( n )]+CurrentFrame( n ). a ( n ), n=0, 1, 2 . . . , L−1;
 
     where 0&lt;=a(n)&lt;=1, usually a(0)=0 and a(L−1)=1. 
     After the fade process is completed in block  514 , processing returns to block  502  where the decoder awaits receipt of the next frame data. Processing continues for each received frame and the perceptual quality is maintained. 
     The methods and systems presented above may reside in software, hardware, or firmware on the device, which can be implemented on a microprocessor, digital signal processor, application specific IC, or field programmable gate array (“FPGA”), or any combination thereof, without departing from the spirit of the invention. Furthermore, the present invention may be embodied in other specific forms without departing from its spirit or essential characteristics. The described embodiments are to be considered in all respects only as illustrative and not restrictive.