A gain adjustment apparatus for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter configured to estimate an accuracy measure of the shape representation, and to determine a gain correction based on the estimated accuracy measure. An envelope adjuster further included in the apparatus is configured to adjust the gain representation based on the determined gain correction.

An audio encoder for encoding an audio signal includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal including a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion.

A frame loss compensation processing method and apparatus is presented, where the method includes, when a i.sup.th frame is a lost frame, estimating a spectrum frequency parameter, a pitch period, and a gain of the i.sup.th frame according to at least one of an inter-frame relationship between first N frames of the i.sup.th frame or an intra-frame relationship between first N frames of the i.sup.th frame. A parameter of the i.sup.th frame is determined using the signal correlation between the first N frames, the signal energy stability between the first N frames, intra-frame signal correlation of each frame, and intra-frame signal energy stability of each frame.

A frame loss compensation processing method and apparatus is presented, where the method includes, when a i.sup.th frame is a lost frame, estimating a spectrum frequency parameter, a pitch period, and a gain of the i.sup.th frame according to at least one of an inter-frame relationship between first N frames of the i.sup.th frame or an intra-frame relationship between first N frames of the i.sup.th frame. A parameter of the i.sup.th frame is determined using the signal correlation between the first N frames, the signal energy stability between the first N frames, intra-frame signal correlation of each frame, and intra-frame signal energy stability of each frame.

A device includes an encoder and a transmitter. The encoder is configured to generate a first high-band portion of a first signal based on a left signal and a right signal. The encoder is also configured to generate a set of adjustment gain parameters based on a high-band non-reference signal. The high-band non-reference signal corresponds to one of a left high-band portion of the left signal or a right high-band portion of the right signal as a high-band non-reference signal. The transmitter is configured to transmit information corresponding to the first high-band portion of the first signal. The transmitter is also configured to transmit the set of adjustment gain parameters corresponding to the high-band non-reference signal.

A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.

A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.

Example signal reconstructing method and apparatus are described. One example method includes obtaining a reference sound channel and a target sound channel. An adaptive length of a transition segment is obtained based on an inter-channel time difference in the current frame and an initial length of the transition segment. A transition window in the current frame is obtained based on the adaptive length of the transition segment. A gain modification factor of a reconstructed signal is obtained. A transition segment signal on the target sound channel is obtained based on the inter-channel time difference, the adaptive length of the transition segment, the transition window, the gain modification factor, and a reference sound channel signal and a target sound channel signal.

Example signal reconstructing method and apparatus are described. One example method includes obtaining a reference sound channel and a target sound channel. An adaptive length of a transition segment is obtained based on an inter-channel time difference in the current frame and an initial length of the transition segment. A transition window in the current frame is obtained based on the adaptive length of the transition segment. A gain modification factor of a reconstructed signal is obtained. A transition segment signal on the target sound channel is obtained based on the inter-channel time difference, the adaptive length of the transition segment, the transition window, the gain modification factor, and a reference sound channel signal and a target sound channel signal.

Described are an encoder for coding speech-like content and/or general audio content, wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters enhance a concealment in case an original frame is lost, corrupted or delayed, and a decoder for decoding speech-like content and/or general audio content, wherein the decoder is configured to use parameters which are sent later in time to enhance a concealment in case an original frame is lost, corrupted or delayed, as well as a method for encoding and a method for decoding.