Abstract: The present technology relates to an information processing device and method and a program that make it possible to reduce the total number of objects while the influence on the sound quality is suppressed. The information processing device includes a pass-through object selection unit configured to acquire data of L objects and select, from the L objects, M pass-through objects whose data is to be outputted as it is, and an object generation unit configured to generate, on the basis of the data of multiple non-pass-through objects that are not the pass-through objects among the L objects, the data of N new objects, N being smaller than (L?M). The present technology can be applied to an information processing device.

Abstract: An audio signal encoding method and device are provided. The method and device are used to encode an audio signal to obtain a bitstream representing the analog audio signal, in which a proper bit allocation for spectral coefficients can be performed.

Abstract: An audio signal encoding method is provided comprising: receiving first and second audio signal frames; processing a second portion of the first audio signal frame and a first portion of the second audio signal frame using an orthogonal transformation to determine in part a first intermediate encoding result; and processing the first intermediate encoding result using an orthogonal transformation to determine a set of spectral coefficients that corresponds to at least a portion of the first audio signal frame.

Abstract: A method of signal processing includes receiving samples of a signal and processing the samples using a time-windowed transform function to generate spectral data corresponding to each time window. The method includes generating first spectrogram data based on magnitudes of the spectral data and generating second spectrogram data based on phase differences of the spectral data. The method further includes combining the first spectrogram data and the second spectrogram data to generate a combined spectrogram and processing the combined spectrogram to generate output.

Abstract: A microcontroller for recording and storing physiological data includes an analog-to-digital converter for converting analog physiological sensor signals to digital signals, a sample buffer for holding a temporal sequence of the digital signals, a central processing unit (CPU), and a non-volatile memory. The non-volatile memory includes (i) a code storage encoding machine-readable data compression instructions that, when executed by the CPU, control the CPU to (a) transform the temporal sequence of the digital signals to produce transformed physiological data characterized by a set of transformation coefficients and (b) compress the set of transformation coefficients to generate compressed physiological data, and (ii) a data storage configured to contain several different instances of the compressed physiological data respectively associated with several different instances of the temporal sequence of the digital signals.

Abstract: The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

Abstract: A deep learning medical device implantable in a body is provided. The device includes a processing and communication unit and a sensing and actuation unit. The processing and communication unit includes a deep learning module including a neural network trained to process the input samples, received from the sensing and actuation unit, through a plurality of layers to classify physiological parameters and provide classification results. A communication interface in communication with the deep learning module receives the classification results for ultrasonic transmission through biological tissue. Methods of sensing and classifying physiological parameters of a body and methods of embedding deep learning into an implantable medical device are also provided.

Abstract: An audio signal encoding method is provided comprising: receiving first and second audio signal frames; processing a second portion of the first audio signal frame and a first portion of the second audio signal frame using an orthogonal transformation to determine in part a first intermediate encoding result; and processing the first intermediate encoding result using an orthogonal transformation to determine a set of spectral coefficients that corresponds to at least a portion of the first audio signal frame.

Abstract: A communication system uses multiple communications links, preferably links that use different communications media. The multiple communications links may include a high latency/high bandwidth link using a fiber-optic cable configured to carry large volumes of data but having a high latency. The communications links may also include a low latency/low bandwidth link implemented using skywave propagation of radio waves and configured to carry smaller volumes of triggering data with a lower latency across a substantial portion of the earth's surface. The triggering data may be sent in a data stream as data frames without headers, security information, or error checking codes. The two communications links may be used together to coordinate various activities such as the buying and selling of financial instruments.

Abstract: An encoder comprising a processor configured to obtain candidate motion vectors (MVs) corresponding to neighboring blocks of a current block, the neighboring blocks neighboring the current block; obtain precisions of the candidate MVs; round the precisions to a target precision based on a rounding scheme; round the candidate MVs based on the target precision; perform pruning of the candidate MVs; generate a candidate list based on the rounding of the candidate MVs and the pruning; select one of the candidate MVs from the candidate list for encoding the current block; and encode an MV candidate index corresponding to the one of the candidate MVs that was selected in a bitstream.

Abstract: An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events.

Abstract: 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.

Abstract: Techniques are described for adaptive encoding different portions of media content based on content. Characteristics of GOPs of media content can be determined and used to set encoding parameters for the GOs. The GOPs can be encoded such that one GOP is encoded differently than another GOP if they have different characteristics.

Abstract: Systems, apparatuses, and methods are described to increase a signal-to-noise ratio difference between a main channel and reference channel. The increased signal-to-noise ratio difference is accomplished with an adaptive threshold for a desired voice activity detector (DVAD) and shaping filters. The DVAD includes averaging an output signal of a reference microphone channel to provide an estimated average background noise level. A threshold value is selected from a plurality of threshold values based on the estimated average background noise level. The threshold value is used to detect desired voice activity on a main microphone channel.

Abstract: Methods and systems are provided for implementing preprocessing operations and augmentation operations upon image datasets transformed to frequency domain representations, including decoding images of an image dataset to generate a frequency domain representation of the image dataset; performing a resizing operation based on resizing factors on the image dataset in a frequency domain representation; performing a reshaping operation based on reshaping factors on the image dataset in a frequency domain representation; and performing a cropping operation on the image dataset in a frequency domain representation. The methods and systems may further include performing an augmentation operation on the image dataset in a frequency domain representation. Methods and systems of the present disclosure may free learning models from computational overhead caused by transforming image datasets into frequency domain representations.

Abstract: The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

Abstract: 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.

Abstract: An inventive concept relates to an audio coding method to which CNN-based frequency spectrum recovery is applied. An inventive concept transmits a part of frequency spectral coefficients generated in transform coding to a decoder and the decoder recovers the frequency spectral coefficient not transmitted. Furthermore, the signs of frequency spectral coefficient are transmitted from an encoder to the decoder depending on a sign transmission rule.

Abstract: An encoding method includes: receiving configuration data related to encoding with a predetermined encoding mode; determining an encoding strategy based on the configuration data, wherein the encoding strategy includes parameters associated with encoding the data on an entity; and causing the data to be encoded on the entity based on the encoding strategy.

Abstract: Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled.

Abstract: Multi-channel audio signals are coded into a monaural audio signal and information allowing to recover the multi-channel audio signal from the monaural audio signal and the information. The information is generated by determining a first portion of the information for a first frequency region of the multi-channel audio signal, and by determining a second portion of the information for a second frequency region of the multi-channel audio signal. The second frequency region is a portion of the first frequency region and thus is a sub-range of the first frequency region. The information is multi-layered enabling a scaling of the decoding quality versus bit rate.

Abstract: An apparatus and method encode audio data, and an apparatus and method decode encoded audio data. An audio data encoding apparatus includes: a scalable encoding unit dividing audio data into a plurality of layers, representing the audio data in predetermined numbers of bits in each of the plurality of layers, and encoding a lower layer prior to encoding an upper layer and an upper bit of each layer prior to encoding a lower bit of each layer; an SBR encoding unit generating spectral band replication (SBR) data that has information with respect to audio data in a frequency band of frequencies equal to or greater than a predetermined frequency among the audio data to be encoded, and encoding the SBR data; and a bitstream production unit generating a bitstream using the encoded SBR data and the encoded audio data corresponding to a predetermined bitrate.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: The invention relates to the compression coding of digital signals such as multimedia signals (audio or video), and more particularly a method for multiple coding, wherein several encoders each comprising a series of functional blocks receive an input signal in parallel. Accordingly, a method is provided in which, a) the functional blocks forming each encoder are identified, along with one or several functions carried out of each block, b) functions which are common to various encoders are itemized and c) said common functions are carried out definitively for a part of at least all of the encoders within at least one same calculation module.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling.

Abstract: An encoding unit encodes data at a first rate during an initial predetermined section of uncompressed data (a), and encodes data at a second rate after the initial predetermined section (where the first rate<the second rate), and encoded data are stored in a storage. The storage is read out the stored data to a network at a transmission rate equal to the transmission rate of the network. A decoding unit decodes received data at the first rate during an initial predetermined section, and decodes received data at a second rate after the initial predetermined section. For finite contents, receiving completion and decoding completion of the received data are simultaneous. In accordance with the present invention, while tolerance with respect to data incoming fluctuations is ensured, playback of dynamic images and music and the like can be immediately started after receiving.

Abstract: Methods and apparatus for voice activity detection are disclosed.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.