Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.

Sub-band voice codec with multi-stage codebooks and redundant coding

Techniques and tools are described for processing reconstructed audio signals. For example, a reconstructed audio signal is filtered in the time domain using filter coefficients that are calculated, at least in part, in the frequency domain. As another example, producing a set of filter coefficients for filtering a reconstructed audio signal includes clipping one or more peaks of a set of coefficient values. As yet another example, for a sub-band codec, in a frequency region near an intersection between two sub-bands, a reconstructed composite signal is enhanced.

Audio codec post-filter

Disclosed is a scalable encoding device capable of reducing an encoding rate thereby to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. In this device, an extension layer is coarsely divided into a system for processing a first channel and a system for processing a second channel. A sound source prediction unit (112) for processing the first channel predicts the drive sound source signal of the first channel from the drive sound source signal of a monaural signal, and outputs the predicted drive sound source signal through a multiplier (113) to a CELP encoding unit (114). A sound source prediction unit (115) for processing the second channel predicts the drive sound source signal of the second channel from the drive sound source signal of the monaural signal and the output from the CELP encoding unit (114), and outputs the predicted drive sound source signal through a multiplier (116) to a CELP encoding unit (117). The CELP encoding units (114, 117) perform the CELP encoding operations of the individual channels by using the individual predicted drive sound source signals.

Scalable encoding device and scalable encoding method

An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Joint enhancement of multi-channel audio

Various strategies for rate/quality control and loss resiliency in an audio codec are described. The various strategies can be used in combination or independently. For example, a real-time speech codec uses intra frame coding/decoding, adaptive multi-mode forward error correction [“FEC”], and rate/quality control techniques. Intra frames help a decoder recover quickly from packet losses, while compression efficiency is still emphasized with predicted frames. Various strategies for inserting intra frames and signaling intra/predicted frames are described. With the adaptive multi-mode FEC, an encoder adaptively selects between multiple modes to efficiently and quickly provide a level of FEC that takes into account the bandwidth currently available for FEC. The FEC information itself may be predictively encoded and decoded relative to primary encoded information. Various rate/quality and FEC control strategies allow additional adaptation to available bandwidth and network conditions.

Audio encoding and decoding with intra frames and adaptive forward error correction

An example of a bandwidth efficient adaptive multi rate (AMR) system for Voice over IP (VoIP) is presented. In VoIP, packet losses cause degradation of the synthesized speech. The distortions may propagate over several consecutive frames, since predictors in the codec exploit inter-frame correlations to gain coding efficiency. To reduce the effects of packet loss, forward error correction (FEC) that adds redundant information to voice packets can be used. However, while FEC can reduce the effects of packet loss, it will increase the amount of bandwidth used by the voice stream, which is not desirable. In this paper we propose FEC methods like partial redundancy, selective redundancy for the most sensitive frames and parameter interpolation in conjunction with AMR codec mode adaptation, which secure the speech quality when using AMR for VoIP without increasing the bandwidth substantially.

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P. Synnergren

Papers

Bandwidth efficient AMR operation for VoIP