| 摘要 |
In audio processing for an audio or video conference, a terminal receives audio packets having transform coefficients for reconstructing an audio signal that has undergone transform coding. When receiving the packets, the terminal determines whether there are any missing packets and interpolates transform coefficients from the preceding and following good frames. To interpolate the missing coefficients, the terminal weights first coefficients from the preceding good frame with a first weighting, weights second coefficients from the following good frame with a second weighting, and sums these weighted coefficients together for insertion into the missing packets. The weightings can be based on the audio frequency and/or the number of missing packets involved. From this interpolation, the terminal produces an output audio signal by inverse transforming the coefficients. |
