The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality. A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.

Distortion reducing multi-band compressor with timbre preservation is provided. Timbre preservation is achieved by determining a time-varying threshold in each of a plurality frequency bands as a function of a respective fixed threshold for the frequency band and, at least in part, an audio signal level and a fixed threshold outside such frequency band. If a particular frequency band receives significant gain reduction due to being above or approaching its fixed threshold, then a time-varying threshold of one or more other frequency bands are also decreased to receive some gain reduction. In a specific embodiment, time-varying thresholds can be computed from an average difference of the audio input signal in each frequency band and its respective fixed threshold.

Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Apparatus and methods for generating an encoded audio bitstream, including by including program loudness metadata and audio data in the bitstream, and optionally also program boundary metadata in at least one segment (e.g., frame) of the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, e.g., including by performing adaptive loudness processing of the audio data of an audio program indicated by the bitstream, or authentication and/or validation of metadata and/or audio data of such an audio program. Another aspect is an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method.

A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audiometric profile to transform audio content so that the listener hears the content similarly to how the content was originally heard by a creative producer of the content. An audio testing tool may be implemented as software application to identify and capture the listener's audiometric profile. A signal processor may operate an algorithm used for processing source audio content, obtaining an identity and an audiometric reference profile of the creative producer from metadata associated with the content. The signal processor may then provide audio output based on a difference between the listener's and creative producer's audiometric profiles.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

There is provided a method for processing an audio input signal and a corresponding audio filter system. The method includes applying non-linear time-domain processing to the input signal to generate a processed copy of the input signal, transforming the input signal and the processed copy of the input signal to the frequency domain, and performing a comparison in the frequency-domain based on the transformed input signal and the transformed processed copy of the input signal. The method also includes determining at least one control parameter based on the comparison, performing frequency-domain processing of the transformed input signal based on the control parameter, and generating an output signal by transforming the frequency-domain processed signal to the time domain.

Some embodiments of the invention are directed to enabling a user to modify the manner in which one or more settings specified by a predefined template for a particular sound source are applied, so as to provide the user with greater control over the settings which are applied to a track than conventional tools afford. Some embodiments are directed to automatically applying one or more settings for a track based at least in part upon an analysis of the spectral and/or dynamic content of the track, such as by automatically performing sound equalization by applying one or more digital filters to a track, defining the frequency range(s) in which one or more filter(s) are applied, applying dynamic range compression, defining the manner in which compression is applied in multiple sub-bands of the audible spectrum, and/or applying one or more other settings. Such settings may be designed to achieve any of numerous (e.g., artistic) goals, such as to bring to the forefront certain elements of the natural character of the sound in a track, or to enhance the track's overall balance and/or clarity.

Provided are an information processing device and an information processing method which are capable of obtaining sound volume correction effects more suitable for an auditory sensation. Target data which is a statistical value of metadata of each audio signal of an audio signal group is acquired, metadata of an audio signal to be reproduced is acquired, and either or both of a correction value of a sound volume of the audio signal to be reproduced and a correction value of a sound quality of the audio signal to be reproduced is calculated using the acquired target data and the acquired metadata.

A converter includes a time window cut-out block, a fast Fourier transform (FFT) block, an attenuation amount limitation block, a quantization noise attenuation block, an overtone generation block, an inverse fast Fourier transform (IFFT) block, and a time window resynthesis block. The attenuation amount limitation block determines the maximum attenuation amount of quantization noise to be attenuated in the quantization noise attenuation block based on the magnitude of a signal level of sound data supplied from the time window cut-out block. The quantization noise attenuation block adjusts amplitude in a frequency domain based on the maximum attenuation amount determined by the attenuation amount limitation block, to attenuate the quantization noise.