An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided, having: a receiving interface for receiving one or more frames, a coefficient generator, and a signal reconstructor. The coefficient generator is configured to determine one or more first audio signal coefficients, and one or more noise coefficients. Moreover, the coefficient generator is configured to generate one or more second audio signal coefficients, depending on the one or more first audio signal coefficients and depending on the one or more noise coefficients. The audio signal reconstructor is configured to reconstruct a first portion of the reconstructed audio signal depending on the one or more first audio signal coefficients and the audio signal reconstructor is configured to reconstruct a second portion of the reconstructed audio signal depending on the one or more second audio signal coefficients, if the current frame is not received by the receiving interface or if the current frame being received by the receiving interface is corrupted.

An apparatus for decoding an audio signal is provided, having a receiving interface, configured to receive a first frame having a first audio signal portion of the audio signal, and configured to receive a second frame having a second audio signal portion of the audio signal; a noise level tracing unit, wherein the noise level tracing unit is configured to determine noise level information depending on at least one of the first audio signal portion and the second audio signal portion; a first reconstruction unit for reconstructing, in a first reconstruction domain, a third audio signal portion of the audio signal depending on the noise level information; a transform unit for transforming the noise level information to a second reconstruction domain; and a second reconstruction unit for reconstructing, in the second reconstruction domain, a fourth audio signal portion of the audio signal depending on the noise level information.

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

Disclosed are some examples of systems, apparatus, methods and computer program products implementing techniques for extending the range of a set of decoded parameter values for a sequence of frequency bands in an identifiable time frame of an audio signal. In some implementations, the parameter values vary in relation to a sequence of time frames of the audio signal and in relation to a sequence of frequency bands in each time frame. In some implementations, it is determined that a decoded value corresponds to a minimum of a first range of values of a first coding protocol of a set of coding protocols. The determined value is modified to be below the minimum of the first range of values to produce an extended value. A modified set of decoded values including one or more extended values can thus be provided.

A method includes: receiving a first speech frame; identifying a first codec mode based at least in part on a Codec Mode Command (CMC) comprising the first speech frame; identifying a second codec mode based at least in part on a downlink (DL) Codec Mode Indication (DCMI) comprising the first speech frame; determining, based at least in part on a current uplink (UL) codec mode, to apply one of the first codec mode, the second codec mode, and a third codec mode having a higher bit rate than the first codec mode; and applying one of the first codec mode, the second codec mode, and the third codec mode. Apply the first codec mode when the RxLev and RXQual are determined not to exceed a predetermined threshold, and applying second or third codec modes otherwise.

A method performed by an encoder. The method comprises determining envelope representation residual coefficients as first compressed envelope representation coefficients subtracted from the input envelope representation coefficients. The method comprises transforming the envelope representation residual coefficients into a warped domain so as to obtain transformed envelope representation residual coefficients. The method comprises applying, at least one of a plurality of gain-shape coding schemes on the transformed envelope representation residual coefficients in order to achieve gain-shape coded envelope representation residual coefficients, where the plurality of gain-shape coding schemes have mutually different trade-offs in one or more of gain resolution and shape resolution for one or more of the transformed envelope representation residual coefficients. The method comprises transmitting, over a communication channel to a decoder, a representation of the first compressed envelope representation coefficients, the gain-shape coded envelope representation residual coefficients, and information on the at least one applied gain-shape coding scheme.

A method, computer program product, and computing system for obtaining machine vision encounter information using one or more machine vision systems. Audio encounter information may be obtained using a plurality of audio acquisition devices of an audio recording system. The audio encounter information may be encoded using an audio codec. The encoding of the audio encounter information by the audio codec may be adapted based upon, at least in part, the machine vision encounter information.

A method, computer program product, and computing system for generating a plurality of acoustic relative transfer functions associated with a plurality of audio acquisition devices of an audio recording system deployed in an acoustic environment. At least a pair of the plurality of acoustic relative transfer functions from time frames may be compared. A change in the acoustic environment may be detected based upon, at least in part, the comparison of the plurality of acoustic relative transfer functions from at least the pair of time frames.

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain.