Abstract:
Audio artifacts due to overrun or underrun in a playout buffer caused by the sampling rates at a sending and receiving side not being at the same rate are reduced. An LPC-residual is modified on a sample-by-sample basis. The LPC-residual block, which includes N samples, is converted to a block comprising N+1 or N−1 samples. A sample rate controller decides whether samples should be added to or removed from the LPC-residual. The exact position at which to add respective remove samples is either chosen arbitrarily or found by searching for low energy segments in the LPC-residual. A speech synthesiser module then reproduces the speech. By using the proposed sample rate conversion method the playout buffer can be continuously controlled. Furthermore, since the method works on a sample-by-sample basis the buffer can be kept to a minimum and hence no extra delay is introduced.

Description:
TECHNICAL FIELD OF THE INVENTION 
   The present invention relates generally to apparatuses and methods for improving speech quality in e.g. IP-telephony systems. More particularly, the present invention relates to a method and apparatus for reducing audio artifacts due to overrun or underrun in a playout buffer. 
   The invention also relates to an arrangement for carrying out the method. 
   DESCRIPTION OF RELATED ART 
   When sampling frequencies, in e.g. a speech coding system, are not controlled, underrun or overrun might occur in the playout buffer, which is a buffer storing speech samples for later playout. Underrun means that the playout buffer will run into starvation, i.e. it will no longer have any samples to play on the output. Overrun means that the playout buffer will be filled with samples and that following samples cannot be buffered and consequently will be lost. Underrun is probably more common than overrun since the size of the playout buffer can increase until there is no memory left, while it only can decrease until there are no samples left. 
   Currently, most systems do not deal with the problem that the sampling frequency might differ considerably between the sending and the receiving side. One possible solution proposed in, EP-0680033 A2, works on pitch periods. Adding or removing pitch periods in the speech signal achieves a different duration of a speech segment without affecting other speech characteristics other than speed. This proposed solution might be used as an indirect sample rate conversion method. 
   Another solution uses the beginning of talkspurts as an indication to reset the playout buffer to a specified level. The distance, in number of samples, between two consecutive talkspurts is increased if the receiving side is playing faster than the sending side and decreased if the receiving side is playing slower than the sending side. In IP-telephony solutions using the IP/UDP/RTP-protocols (Internet Protocol/User Datagram Protocol/Real Time Protocol), a marker flag in the RTP header is used to identify the beginning of a talkspurt. At the beginning of a talkspurt, the playout buffer is set to a suitable size. 
   The solution according to EP-0680033 A2, where pitch periods are removed or inserted, assumes a fixed conversion factor between the receiving and transmitting side. Therefore, it cannot be used in dynamic systems, i.e. where the sampling frequencies varies. Further, it does not solve the problem with underrun or overrun situations, but is instead focused on changing the playback rate of a speech signal stored in compressed form for playback later and at a different speed to that at which it was stored. 
   Using the method of resetting the playout buffer to a certain size causes problems if there are very long talkspurts, e.g. broadcast from one speaker to several listeners. Since the length of a talkspurt is not defined in the beginning of the talkspurt, the size to reset to might be either too small or too large. If it is too small, underrun will occur and if it is too large, unnecessary delay is introduced. Thus, the problem persists. 
   The general problem with the currently known approaches is that they are static and inflexible. Therefore, dynamic solutions are required. 
   SUMMARY OF THE INVENTION 
   The present invention deals with the problem of improving speech quality in systems where the sampling rate at a transmitting terminal differs from the playout rate of a receiving buffer at a receiving terminal. This is often the case in e.g. IP-telephony. 
   When sampling frequencies are not controlled, underrun or overrun might occur in the playout buffer at the receiving side, which causes audible artifacts in the speech signal. To avoid said overrun or underrun there is an need for dynamically keeping the playout buffer to an average size, i.e. controlling the fullness of the playout buffer. 
   One object of the present invention is thus to provide a method for reducing audio artifacts in a speech signal due to overrun or underrun in the playout buffer. 
   Another object of the invention is to dynamically control the fullness of the playout buffer so as not to introduce extra delay. 
   The above mentioned and other objects are achieved by means of dynamic sample rate and conversion of speech frames, i.e. converting speech frames comprising N samples to instead comprise either N+1 or N−1 samples. More specifically, the invention works on an LPC-residual of the speech frame. By adding or removing a sample in the LPC-residual, a sample rate conversion will be achieved. The LPC-residual is the output from an LPC-filter, which removes the short-term correlation from the speech signal. The LPC-filter is a linear predictive coding filter where each sample is predicted as a linear combination of previous samples. 
   By using the proposed sample rate conversion method, the playout buffer, of e.g. an IP-telephony terminal, can be continuously controlled with only small audio artifacts. Since the method works on a sample-by-sample basis, the playout buffer can be kept to a minimum and hence no extra delay is introduced. The solution also has very low complexity, especially when the LPC-residual already is available, as in the case in e.g. a speech decoder. 
   The term “comprises/comprising” when used in this specification is taken to specify the presence of stated features, integers, steps or components but does not preclude the presence or addition of one or more other features, integers, steps, components or groups thereof. 
   Although aspects of the invention have been summarised above, the method and apparatus according to the appended claims define the scope of the invention. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  shows a transmitter and a receiver to which the method of the invention can be applied. 
       FIG. 2  shows a speech signal in the time domain. 
       FIG. 3  shows an LPC-residual of a speech signal in the time domain. 
       FIG. 4  illustrates four modules of the sample rate conversion method according to the invention. 
       FIG. 5A  shows an analysis-by-synthesis speech encoder with LTP-filter. 
       FIG. 5B  shows an analysis-by-synthesis speech encoder with adaptive codebook. 
       FIGS. 5C-5F  show different implementations of the LPC-residual extraction depending on the realisation of the speech encoder. 
       FIGS. 5G-5J  show four ways of placing the sample rate conversion within the feed back loop of the speech decoder. 
       FIG. 6  illustrates how to use information about pitch pulses to find samples with low energy. 
       FIG. 7  illustrates LPC-history extension. 
       FIG. 8  illustrates copying of the history of the LPC residuals. 
   

   DETAILED DESCRIPTION 
   Referring to  FIG. 1 , a method for improving speech quality in a communication system includes a first terminal unit TRX 1  transmitting speech signals having a first sample frequency F 1  and a second terminal unit TRX 2  receiving said speech signals, buffering them in a playout buffer  100  with said first frequency F 1  and playing out from said playout buffer with a second frequency F 2 . When the buffering frequency F 1  is larger than the playout frequency F 2 , the playout buffer  100  will eventually be filled with samples and subsequent samples will have to be discarded. When the buffering frequency F 1  is lower than the playout frequency F 2  the playout buffer will run into starvation, i.e. it will no longer have any samples to play on the output. These two problems are called overrun and underrun, respectively, and cause audible artifacts like popping and clicking sounds in the speech signal. 
   The above and other problems with underrun and overrun are solved by using dynamic sample rate conversion based on modifying the LPC-residual of the speech signal and will be further described with reference to  FIGS. 2-8 . 
     FIG. 2  shows a typical segment of a speech signal in the time domain. This speech signal shows a short-term correlation, which corresponds to the vocal tract and a long-term correlation, which corresponds to the vocal cords. The short-term correlation can be predicted by using the LPC-filter and the long-term correlation can be predicted by using an LTP-filter. LPC means linear predictive coding and LTP means long term prediction. Linear in this case implies that the prediction is a linear combination of previous samples of the speech signal. 
   The LPC-filter is usually denoted: 
         H   ⁡     (   z   )       =       1     A   ⁡     (   z   )         =     1     1   -       ∑     i   =   1     n     ⁢           ⁢       a   i     ⁢     z     -   i                     
 
   By feeding a speech frame through the LPC-filter, H(z), the LPC-residual is found. The LPC-residual, shown in  FIG. 3 , contains pitch pulses P generated by the vocal cords. The distance L between two pitch pulses P is called lag. The pitch pulses P are also predictable, and since they represent the long-term correlation of the speech signal they are predicted through an LTP-filter given by the distance L between the pitch pulses P and the gain b of a pitch pulse P. The LTP-filter is usually detected.
 
 F ( z )= b·z   −L  
 
   When the LPC-residual is fed through the inverse of the LTP-filter F(z), and LTP-residual is created. In the LTP-residual, the long-term correlation in the LPC-residual is removed, giving the LTP-residual a noise-like appearance. 
   The solution according to the invention modifies the LPC-residual, shown in  FIG. 3 , on a sample-by-sample basis. That is, an LPC-residual block comprising N samples is converted to an LPC-residual block comprising either N+1 or N−1 samples. The LPC-residual contains less information and less energy compared to the speech signal, but the pitch pulses P are still easy to locate. When modifying the LPC-residual, samples that are close to a pitch pulse P should be avoided, because these samples contain more information and thus have a large influence on the speech synthesis. The LTP-residual is not as suitable as the LPC-residual to use for modification since the pitch pulse positions P are no longer available. Thus, the LPC-residual is better suited for modifications both compared to the speech signal and to the LTP-residual, since the pitch pulses P are easily located in the LPC-residual. 
   A sample rate conversion consists of four modules, shown in FIG.  4 . 
   1) A Sample Rate Controller (SRC) module  400  that calculates whether a sample should be added or removed; 
   2) LPC-Residual Extraction (LRE) modules  410  that are used to obtain the LPC-residual r LPC ; 
   3) Sample Rate Conversion Methods (RCM) modules  420  that find the position at which to add or remove samples and determine how to perform the insertion and deletion, i.e. converting the LPC residual block r LPC  comprising N samples to a modified LPC-residual block r LPC  comprising N+1 or N−1 samples; and 
   4) A Speech Synthesiser Module (SSM)  430  to reproduce the speech. 
   An idea behind embodiments of the invention is that it is possible to change the playout rate of the playout buffer  440  by removing or adding samples in the LPC-residual r LPC . 
   The SRC module  400  decides whether samples should be added or removed in the LPC residual r LPC . This is done on the basis of at least one of the four following parameters: the sampling frequencies of the sending TRX 1  and receiving terminal units TRX 2 , information about the speech signal e.g. a voice activity detector signal, status of the playout buffer, an indicator of the beginning of a talkspurt. The four parameters are designated SRC Inputs in FIG.  4 . On the basis of a function of one or several of these parameters the SRC  400  decides when to insert or remove a sample in the LPC residual r LPC  and optionally which RCM  420  to use. Since digital processing of speech signals usually is made on a frame-by-frame basis, the decision of when to remove or add samples basically is to decide within which LPC-residual r LPC  frame the ROM  420  is to insert or remove a sample. 
   There are basically three methods of obtaining the LPC-residual r LPC  that is needed as input to the RCM&#39;s  420 . The methods depend on the implementation of the speech encoder and will be described with reference to  FIGS. 5A-5F . The LRE solution also directly influences the SSM solution, which will become apparent below. 
   In  FIG. 5A  an analysis-by-synthesis speech encoder  500  with LTP-filter  540  is shown. This is a hybrid encoder where the vocal tract is described with an LPC-filter  550  and the vocal cords is described with a LTP-filter  540 , while the LTP-residual {circumflex over (r)} LPC   (n)  is waveform-compared with a set of more or less stochastic codebook vectors from a fixed codebook  530 . The input signal S is divided into frames  510  with a typical length of 10-30 ms. For each frame the LPC-filter  550  is calculated through an LPC-analysis  520  and the LPC-filter  550  is included in a closed loop to find the parameters of the LTP-filter  540 . The speech decoder  580  is included in the encoder and consists of the fixed codebook  530 , whose output {circumflex over (r)} LPC   (n)  is connected to the LTP-filter  540 , whose output {circumflex over (r)} LPC   (n)  is connected to the LPC-filter  550 , which generates an estimate ŝ(n) of the original speech signal s(n). Each estimate signal ŝ(n) is compared with the original speech signal s(n) and a difference signal e(n) is calculated. The difference signal e(n) is then weighted by an error-weighting block  560  to calculate a perceptual weighted error measure e w (n). The set of parameters that gives the least perceptual weighted error measure e w (n) is transmitted to a receiving side  570 . 
   As can be seen in  FIG. 5C , the LPC-residual {circumflex over (r)} LPC   (n)  is the output from the LTP-filter  540 . SRC/RCM modules  545  can be connected directly to the output of the LTP-filter  540  and integrated into the speech encoder. An LRE consists of the fixed codebook  530  and the long-term predictor  540  and the SSM consists of an LPC-filter  550 , thus the LRE-module and the SSM-module are natural parts of the speech decoder. 
   If the speech encoder, on the other hand, is an analysis-by-synthesis speech encoder where the LTP-filter  540  is exchanged to an adaptive codebook  590  as shown in  FIG. 5B , the LPCresidual LPC(n) is the output from the sum of the adaptive and the fixed codebooks  590  and  530 . All other elements have the same function as in  FIG. 5A  which shows an analysis-by-synthesis speech encoder with LTP-filter  500 . As can be seen in  FIG. 5D  the LPC residual {circumflex over (r)} LPC   (n)  is the sum of the output from the adaptive and fixed codebook  590  and  530 . The SRC/RCM modules  545  can thus again be connected to the output and integrated into the speech encoder as shown in FIG.  5 D. The LRE consists of the adaptive and the fixed codebook  590  and  530  and the SSM consists of an LPC-filter  550 , thus the LRE module and the SSM module are again natural parts of the speech decoder. 
   When the speech encoder has some sort of backward adaptation, it is not feasible to make alterations in the LPC-residual since this would affect the adaptation process in a detrimental way. In  FIG. 5E  is shown how in these cases the parameters ŝ(n) from the LPC-filter  550  can be fed to an inverse LPC-filter  525  placed after the speech decoder. After the sample rate conversion has been made in the SRC/RCM modules  545  an LPC-filtering  550  is performed to reproduce the speech signal. The LRE module consists of the inverse LPC-filter  525  and the SSM module consists of the LPC-filter  550 . 
   In  FIG. 5F  it is shown how it is possible to produce an LPC residual {circumflex over (r)} LPC   (n)  through a full LPC analysis. The output ŝ(n) from the speech decoder is fed to both an LPC analysis block  520  and an LPC-inverse filter  525 . After the sample rate conversion has been made in the SRC/RCM modules  545 , and LPC filtering  550  is performed to reproduce the speech signal. The LRE consists in this case of the LPC analysis  520  respective the LPC inverse filter  525  and the SSM module consists of the LPC filter  550 . Performing an LPC analysis is considered to be well known to a person skilled in the art and is therefore not discussed any further. 
   Referring again to  FIG. 4 , assume that the SRC-module  400  has decided that a sample should be added or removed in the LPC residual r LPC  and that the LRE module  410  has produced an LPC residual r LPC . The RCM-module  420  then has to find the exact position in the LPC-residual r LPC  where to add or remove a sample and performing the adding respective removing. There are four different methods for the RCM-module  420  to find the insertion or deletion point. 
   The first and most primitive method arbitrarily removes or adds a sample whenever this becomes necessary. If the sample rate difference between the terminals is small this will only lead to mirror artifacts since the adding or removing is performed very seldom. 
   By inserting or removing samples at positions where the energy in the LPC-residual is low the synthesis will be less affected. This is due to the fact that segments close to pitch pulses will then be avoided. To find these segments of low energy either a sliding window method or a simplier block energy analysis can be used. 
   The second method, called the sliding window energy method, calculates a weighted energy value for each sample in the LPC-residual. This is done by multiplying k samples surrounding a sample with a window function of size k (k&lt;&lt;N), where N equals the number of samples in the LPC-residual. Each sample is then squared and the sum of the resulting k values is calculated. The window is shifted one position and the procedure is repeated. The position where to insert or remove samples is given by the sample with the lowest weighted energy value. 
   The third method, block energy analysis, is a simpler solution for finding the insertion or deletion point. The LPC-residual is simply divided into blocks of equal length and an arbitrary sample is removed or added in the block with the lowest energy. 
   The fourth method, illustrated in  FIG. 6 , uses knowledge about the position P of a pitch pulse, and the lag L between two pitch pulses. With this knowledge, it is possible to calculate a position P′ having low energy at which it is therefore appropriate to add or remove a sample. The new position P′ can be expressed as P′=P+k·L, wherein the constant k is selected so that P′ is selected to be somewhere in the middle between two pitch pulses, thus avoiding positions with high energy. A typical value of k is in the range of 0.5 to 0.8. 
   When the RCM-module  420  has calculated the position at which to add or remove a sample it must be determined how to perform the insertion or deletion. There are three methods of performing such insertions or deletion depending on the type of LRE-module used. 
   In the first method, either zeros are added or samples with small amplitudes are removed. This method can be used for all LRE solutions described above. (See  FIGS. 5C-5F .) Notice that in  FIGS. 5C and 5D  the SRC/RCM-modules are placed before the synthesis filler SSM, but after the feed back of the LPC residual to the LTP-filter  540  respective the adaptive codebook  590 . 
   In the second method, insertion is carried out by adding zeros and interpolating surrounding samples. Deletion is performed by removing samples and preferably smoothing surrounding samples. This method can also be used for all of the LRE solutions described above. (See FIGS.  5 C- 5 F). Notice that in  FIGS. 5C and 5D  the SRC/RCM-modules are placed before the synthesis filter SSM, but after the feed back of the LPC residual to the LTP-filter  540  respective the adaptive codebook  590 . 
   In the third method, the SCR/RCM-modules  545  are placed within the feedback loop of the speech decoder instead of after the feedback loop as in the previous methods. (See  FIGS. 5G-5J .) Placing the SRC/RCM-modules within the feedback loop uses real LPC residual samples for the sample rate conversion, by changing the number of components in the LPC-residual. The implementation differs depending on whether it is an analysis-by-synthesis speech encoder with LTP filter shown in  FIG. 5A  or an analysis-by-synthesis speech encoder with adaptive codebook shown in  FIG. 5B  that is used. 
   For the speech decoder with LTP filter (see  FIG. 5A ) the SRC/RCM-modules  545  can be placed within the feedback loop in two different ways, either within the LTP feedback loop as shown in  FIG. 5G  or in the output from the fixed codebook  530  as shown in FIG.  5 H. For the speech decoder with adaptive codebook (see  FIG. 5B ) the SRC/RCM can also be placed in two different ways, i.e. either before ( FIG. 5J ) or after,  FIG. 51 , the summation of the outputs from the adaptive and the fixed codebook. 
   The alterations on the LPC residual consists of removing or adding samples just before, but since the SRC/RCM-modules  545  are placed within the LTP feedback loop, some modifications must be done. The extending or shortening of a segment can be done in three ways either at the respective ends of the segment or somewhere in the middle of the segment.  FIG. 7  shows the case where the LPC residual is extended by copying two overlapping segments, segment  1  and segment  2 , from the history of the LPC residual to create the longer LPC residual. The normal case when no insertion or deletion is needed would be to copy N samples. Shortening the LPC residual is achieved by copying two segments that has a gap between them instead of being overlapped. As before, it is important that a pitch pulse is not doubled or removed since this would introduce perceptual artifacts. Hence, an analysis should be performed in order to evaluate where to add or remove segments. The analysis is preferably made by using the same methods as discussed above regarding how to find the position where to add or remove a sample in the RCM-module. 
   For all implementations except when the SRC/RCM-modules  545  are placed between the fixed codebook  530  and the LTP filter  540  the history of the LPC residual also has to be modified. The lag L will be increased or decreased for the specific part of the history where a sample is inserted or deleted. Thus the starting position of the segment that will be copied from the history of the LPC residual, Pointer  1  or Pointer  2  in  FIG. 8 , needs modification. If the segment to copy is newer, i.e. the case of Pointer  1 , there is need to modify the starting position. If, however, the segment to copy is older, i.e. the case of Pointer  2 , then the pointer should be increased or decreased depending on if a sample is inserted or deleted. This has to be managed for subsequent sub-frames and frames as long as the modification is within the history of the LPC residual. 
   When the SRC/RCM-modules are placed before the summation of the outputs from the adaptive and the fixed codebook as shown in  FIG. 5J  the length of the fixed codebook also needs to be changed. This is done by adding a sample, preferably a zero sample, in the output from the fixed codebook or removing one of the components. The insertion and deletion in the fixed codebook should be synchronised with the insertion and deletion in the adaptive codebook. 
   Embodiments of the invention being thus described, it will be obvious that the same may be varied in many ways. Such variations are not to be regarded as a departure from the scope of the invention, and all such modifications as would be obvious to a person skilled in the art are intended to be included within the scope of the following claims.