A device includes an encoder and a transmitter. The encoder is configured to determine a mismatch value indicative of an amount of temporal mismatch between a reference channel and a target channel. The encoder is also configured to determine whether to perform a first temporal-shift operation on the target channel at least based on the mismatch value and a coding mode to generate an adjusted target channel. The encoder is further configured to perform a first transform operation on the reference channel to generate a frequency-domain reference channel and perform a second transform operation on the adjusted target channel to generate a frequency-domain adjusted target channel. The encoder is also configured to estimate one or more stereo cues based on the frequency-domain reference channel and the frequency-domain adjusted target channel. The transmitter is configured to transmit the one or more stereo cues to a receiver.

Encoding of multiple audio signals

An audio decoder (100; 300) for providing a decoded audio information (112; 312) on the basis of an encoded audio information (110; 310) comprises an error concealment (130; 380; 500) configured to provide an error concealment audio information (132; 382; 512) for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation (322) using a time domain excitation signal (532).

Audio decoder and method for providing a decoded audio information using an error concealment based on a time domain excitation signal

The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system (500) configured to generate a bitstream (564) indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system (500) comprises a downmix processing unit (510) configured to generate the downmix signal from a multi-channel input signal (561); wherein the downmix signal comprises m channels and wherein the multi-channel input signal (561) comprises n channels; n, m being integers with m<n. Furthermore, the system (500) comprises a parameter processing unit (520) configured to determine the spatial metadata from the multi-channel input signal (561). In addition, the system (500) comprises a configuration unit (540) configured to determine one or more control settings for the parameter processing unit (520) based on one or more external settings; wherein the one or more external settings comprise a target data-rate for the bitstream (564) and wherein the one or more control settings comprise a maximum data-rate for the spatial metadata.

Methods for parametric multi-channel encoding

The present invention concerns 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.

Encoder, decoder and method for encoding and decoding audio content using parameters for enhancing a concealment

A data augmentation method for improving the performance of spontaneous natural language speech recognition in a speech recognition system based on a deep neural network is disclosed The data augmentation method for improving the performance of spontaneous natural language speech recognition in a speech recognition system based on a deep neural network includes: a step of augmenting speech data for a speech speed variation among natural language speech variation characteristics, a step of augmenting voice data for incorrect pronunciation among the natural language speech variation characteristics; and a step of learning the speech recognition system based on the neural network by using the augmented speech data for the speech speed variation and incorrect pronunciation Therefore, the performance of the speech recognition system can be improved

Data augmentation method for spontaneous speech recognition

An approach is described that obtains spectrum coefficients for a replacement frame (m) of an audio signal. A tonal component of a spectrum of an audio signal is detected based on a peak that exists in the spectra of frames preceding a replacement frame (m). For the tonal component of the spectrum a spectrum coefficients for the peak (502) and its surrounding in the spectrum of the replacement frame (m) is predicted, and for the non-tonal component of the spectrum a non-predicted spectrum coefficient for the replacement frame (m) or a corresponding spectrum coefficient of a frame preceding the replacement frame (m) is used.

Method and apparatus for obtaining spectrum coefficients for a replacement frame of an audio signal, audio decoder, audio receiver and system for transmitting audio signals

An apparatus for coding a portion of an audio signal (10) to obtain an encoded audio signal (26) for the portion of the audio signal comprises a transient detector (12) for detecting whether a transient signal is located in the portion of the audio signal to obtain a transient detection result (14), an encoder stage (16) for performing a first encoding algorithm on the audio signal, the first encoding algorithm having a first characteristic, and for performing a second encoding algorithm on the audio signal, the second encoding algorithm having a second characteristic being different from the first characteristic, a processor (18) for determining which encoding algorithm results in an encoded audio signal being a better approximation to the portion of the audio signal with respect to the other encoding algorithm to obtain a quality result (20), and a controller (22) for determining whether the encoded audio signal for the portion of the audio signal is to be generated by either the first encoding algorithm or the second encoding algorithm based on the transient detection result (14) and the quality result (20).

Apparatus and method for coding a portion of an audio signal using a transient detection and a quality result

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.

Apparatus and method for improved signal fade out for switched audio coding systems during error concealment

An apparatus for reconstructing a frame including a speech signal as a reconstructed frame is provided, the apparatus including a determination unit and a frame reconstructor being configured to reconstruct the reconstructed frame, such that the reconstructed frame completely or partially includes the first reconstructed pitch cycle, such that the reconstructed frame completely or partially includes a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle.

Apparatus and method for improved concealment of the adaptive codebook in a celp-like concealment employing improved pulse resynchronization

An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, comprises a domain converter (100) for converting a first domain epresentation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter (104) for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function comprising a numerator and a denominator, wherein the numerator comprises a gain value indicated by the gain information, and wherein the denominator comprises an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Apparatus and method for processing an audio signal using a harmonic post-filter

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