In this letter, we propose a shared adaptive packet loss concealment scheme for the high quality guaranteed Internet telephony service which connects multiple users. In order to recover packet loss efficiently in the all-IP based convergence environment, we provide a robust signal recovery scheme which is based on the shared adaptive both-side information utilization. This scheme is provided according to the average magnitude variation across the frames and the pitch period replication on the 1-port gateway (G/W) system. The simulated performance demonstrates that the proposed scheme has the advantages of low processing times and high recovery rates in the all-IP based ubiquitous environment.

We proposed and evaluated a speech packet loss concealment method which predicts lost segments from speech included in packets either before, or both before and after the lost packet. The lost segments are predicted recursively by using linear prediction both in the forward direction from the packet preceding the loss, and in the backward direction from the packet succeeding the lost segment. Predicted samples in each direction are smoothed by averaging using linear weights to obtain the final interpolated signal. The adjacent segments are also smoothed extensively to significantly reduce the speech quality discontinuity between the interpolated signal and the received speech signal. Subjective quality comparisons between the proposed method and the the packet loss concealment algorithm described in the ITU standard G.711 Appendix I showed similar scores up to about 10% packet loss. However, the proposed method showed higher scores above this loss rate, with Mean Opinion Score rating exceeding 2.4, even at an extremely high packet loss rate of 30%. Packet loss concealment of speech degraded with G.729 coding, and babble noise mixed speech showed similar trends, with the proposed method showing higher qualities at high loss rates. We plan to further improve the performance by using adaptive LPC prediction order depending on the estimated pitch, and adaptive LPC bandwidth expansion depending on the consecutive number of repetitive prediction, among many other improvements. We also plan to investigate complexity reduction using gradient LPC coefficient updates, and processing delay reduction using adaptive forward/bidirectional prediction modes depending on the measured packet loss ratio.

Speech quality of VoIP (Voice over Internet Protocol) may potentially be degraded by transmission errors such as packet loss and delay which are basically inevitable in best-effort communications. This study investigates an error concealment technique for such degradation by using a receiver-based technique called pitch waveform replication. For enhancing the conventional technique, this study proposes a waveform reconstruction technique that also takes account of the pitch variation between the backward and forward frames of gap frames. From experimental results of objective evaluation, it is indicated that the proposed technique may potentially be useful for improving the speech quality, compared with the conventional technique.

Speech quality of VoIP (Voice over Internet Protocol) may potentially be degraded by transmission errors such as packet loss and delay that are basically inevitable in best-effort communications. This study newly proposes an error concealment technique for such degradation by taking account of both sender-based and receiver-based techniques. In the proposed technique, sender-based side information, which is required by the receiver-based technique, is transmitted by using steganography, so that its datagram is completely compatible with the conventional format of VoIP. From experimental results of objective evaluation, it is indicated that the proposed technique may potentially be useful for improvement of speech quality, compared with the conventional technique.