Abstract: An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Abstract: Techniques for speech processing using a deep neural network (DNN) based acoustic model front-end are described. A new modeling approach directly models multi-channel audio data received from a microphone array using a first model (e.g., multi-channel DNN) that takes in raw signals and produces a first feature vector that may be used similarly to beamformed features generated by an acoustic beamformer. A second model (e.g., feature extraction DNN) processes the first feature vector and transforms it to a second feature vector having a lower dimensional representation. A third model (e.g., classification DNN) processes the second feature vector to perform acoustic unit classification and generate text data. These three models may be jointly optimized for speech processing (as opposed to individually optimized for signal enhancement), enabling improved performance despite a reduction in microphones and a reduction in bandwidth consumption during real-time processing.

Abstract: An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Abstract: Techniques for speech processing using a deep neural network (DNN) based acoustic model front-end are described. A new modeling approach directly models multi-channel audio data received from a microphone array using a first model (e.g., multi-channel DNN) that takes in raw signals and produces a first feature vector that may be used similarly to beamformed features generated by an acoustic beamformer. A second model (e.g., feature extraction DNN) processes the first feature vector and transforms it to a second feature vector having a lower dimensional representation. A third model (e.g., classification DNN) processes the second feature vector to perform acoustic unit classification and generate text data. These three models may be jointly optimized for speech processing (as opposed to individually optimized for signal enhancement), enabling improved performance despite a reduction in microphones and a reduction in bandwidth consumption during real-time processing.

Abstract: Techniques for speech processing using a deep neural network (DNN) based acoustic model front-end are described. A new modeling approach directly models multi-channel audio data received from a microphone array using a first model (e.g., multi-channel DNN) that takes in raw signals and produces a first feature vector that may be used similarly to beamformed features generated by an acoustic beamformer. A second model (e.g., feature extraction DNN) processes the first feature vector and transforms it to a second feature vector having a lower dimensional representation. A third model (e.g., classification DNN) processes the second feature vector to perform acoustic unit classification and generate text data. These three models may be jointly optimized for speech processing (as opposed to individually optimized for signal enhancement), enabling improved performance despite a reduction in microphones and a reduction in bandwidth consumption during real-time processing.

Abstract: An apparatus for generating a representation of a bandwidth-extended signal on the basis of an input signal representation includes a phase vocoder configured to obtain values of a spectral domain representation of a first patch of the bandwidth-extended signal on the basis of the input signal representation. The apparatus also includes a value copier configured to copy a set of values of the spectral domain representation of the first patch, which values are provided by the phase vocoder, to obtain a set of values of a spectral domain representation of a second patch, wherein the second patch is associated with higher frequencies than the first patch. The apparatus is configured to obtain the representation of the bandwidth-extended signal using the values of the spectral domain representation of the first patch and the values of the spectral domain representation of the second patch.

Abstract: An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Abstract: An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Abstract: An apparatus for generating an encoded signal includes: a window sequence controller for generating a window sequence information for windowing an audio or image signal, the window sequence information indicating a first window for generating a first frame of spectral values, a second window function and at least one third window function for generating a second frame of spectral values, wherein the first window function, the second window function and the one or more third window functions overlap within a multi-overlap region; a preprocessor for windowing a second block of samples corresponding to the second window function and the at least one third window functions using an auxiliary window function to acquire a second block of windowed samples, a spectrum converter for applying an aliasing-introducing transform; and a processor for processing the first frame and the second frame to acquire encoded frames of the audio or image signal.

Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

Abstract: Disclosed are techniques for generating an error concealment signal, where such techniques may include an LPC representation generator for generating a replacement LPC representation; a gain calculator for calculating a gain information from the LPC representations; a compensator for compensating a gain influence of the replacement LPC representation using the gain information; and an LPC synthesizer for filtering codebook information using the replacement LPC representation to obtain the error concealment signal, where the compensator is configured for weighting the codebook information or an LPC synthesis output signal.

Abstract: A time scaler for providing a time scaled version of an input audio signal is configured to compute or estimate a quality of a time scaled version of the input audio signal obtainable by a time scaling of the input audio signal. The time scaler is configured to perform the time scaling of the input audio signal in dependence on the computation or estimation of the quality of the time scaled version of the input audio signal obtainable by the time scaling. An audio decoder has such a time scaler.

Abstract: The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.

Abstract: 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.

Abstract: 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.

Abstract: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame.

Abstract: A jitter buffer control for controlling a provision of a decoded audio content on the basis of an input audio content is configured to select a frame-based time scaling or a sample-based time scaling in a signal-adaptive manner. An audio decoder uses such a jitter buffer control.

Abstract: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal includes a receiving interface for receiving one or more frames comprising information on a plurality of audio signal samples of an audio signal spectrum of the encoded audio signal, and a processor for generating the reconstructed audio signal. The processor is configured to generate the reconstructed audio signal by fading a modified spectrum to a target spectrum, if a current frame is not received by the receiving interface or if the current frame is received by the receiving interface but is corrupted, wherein the modified spectrum includes a plurality of modified signal samples, wherein, for each of the modified signal samples of the modified spectrum, an absolute value of the modified signal sample is equal to an absolute value of one of the audio signal samples of the audio signal spectrum.

Abstract: 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.

Abstract: An apparatus for encoding an audio or image signal, includes: a controllable windower for windowing the audio or image signal to provide the sequence of blocks of windowed samples; a converter for converting the sequence of blocks of windowed samples into a spectral representation including a sequence of frames of spectral values; a transient location detector for identifying a location of a transient within a transient look-ahead region of a frame; and a controller for controlling the controllable windower to apply a specific window having a specified overlap length to the audio or image signal in response to an identified location of the transient, wherein the controller is configured to select the specific window from a group of at least three windows, wherein the specific window is selected based on the transient location.