## 1 Introduction  
With rapid development of speech communication based on a packet switching network, the issue of packet loss is paid more attention since it may cause noncontinuous speech transmission and degrade speech quality. In the process of realizing speech communication, the speech signal is generally divided into multiple small segments for coding with some standard codec [1] or free code [2]. The encoded bit stream is stored in the data packets for transmission. These data packets are addressed and sent through the network and reorganized in the correct order at the destination. However, due to the problems of multipath fading, channel congestion, and buffer overflow, the data packets may be lost, which will seriously cause the distortion of the decoded speech at the receiver [1]. Therefore, the packet loss processing methods with error detection and correction are needed to reconstruct the gap caused by the missed speech signal. Packet loss processing methods are generally divided into packet loss recovery at the sender and packet loss concealment (PLC) at the receiver [3]. The packet loss processing at the sender is mainly embodied in processing redundant data during the transmission [4, 5]. Redundant data can be used in two cases. One is used for packet loss repair based on the result of error correction coding, that is, the forward error correction [6], and the other one is related to the information in the data packet, that is, low-bit redundancy information [7]. The PLC is only performed at the receiver for packet loss processing. It does not occupy additional bandwidth and introduce excessive delay, so it has been widely used in real-time speech communication. The core idea of the PLC is to fill in the speech subjectively through perceptual effect of the brain to achieve the purpose of concealing the loss of data packets. The PLC can be divided into three categories [3], that is, insertion, interpolation, and reconstruction. The insertion method can realize the purpose of concealing the packet loss by inserting a waveform in the lost packet. This waveform has nothing to do with the waveform in the lost packet, and is mainly used to conceal or transit the lost packet. The interpolation method also introduces a section of waveform in the lost packet as a replacement of original waveform. The interpolation method considers the correlation of the signal, that is, the waveform that is most similar to the lost signal is found through the correctly received packet near the lost packet as a replacement. The interpolation method is divided into three categories, namely, waveform substitution, pitch waveform duplication, and time domain correction. According to a certain matching principle, the core idea of waveform substitution [8] is to find a waveform most similar to one in the lost packet within the waveform around the lost packet for substituting the lost signal. The pitch waveform duplication [9] is better than the waveform replacement since it can use pitch detection algorithm to refine waveform replacement. If the lost signal belongs to the unvoiced speech signal, the waveform before the lost signal is directly used as a replacement, whereas for the voiced speech signal in the lost packet, the pitch period of the speech signal is first estimated, and a suitable waveform is intercepted before the lost packet as a substitution according to the estimated pitch period. Although the computational load of this method is a little high, its concealment of lost packet is much more effective. It has been adopted in ITU-T G.711 [10]. The core idea of time domain correction is to extend the waveform before and after the lost packet inward until it covers all the gaps in the lost packet. This method can achieve the purpose of changing the length of speech signal in the time domain and keeping pitch period unchanged. The representative methods of the time domain correction are overlap-add (OLA) [11], synchronous overlap-add (SOLA) [12], and waveform similarity overlap-add (WSOLA) [13; 14; 15].