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.

Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).

Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).

A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.

A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.

An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder includes a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder includes an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.

It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder includes a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder includes an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.