An example sound output apparatus may include an input unit, an output unit, and a processor configured to perform first-filtering of a frequency band of an audio signal received through the input unit based on a sound pressure of the sound output apparatus, second-filtering of a frequency band of the first-filtered audio signal based on a frequency response characteristic of the sound output apparatus, adjust a loudness of the received audio signal based on a perception volume level of the second-filtered audio signal, and output the adjusted audio signal through the output unit.

The present disclosure relates to processing a plurality of audio signals. The method includes receiving the plurality of audio signals in the frequency domain and determining an overall attenuation multiplier based on the plurality of audio signals and an overall lookup table that relates decibel values to different overall attenuation multipliers. The method further includes determining an attenuation vector comprising a plurality of bin-specific attenuation multipliers, each bin-specific attenuation multiplier respectively corresponding to a different frequency bin of the plurality of frequency bins. The method further includes scaling each bin-specific attenuation value in the attenuation vector with the overall attenuation multiplier, and editing each of the audio signals based on the scaled bin-specific attenuation values in the attenuation vector.

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

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.

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.

A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

A method (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles. The method (600) comprises determining a first rendering mode from the plurality of different rendering modes; determining (609, 610) one or more DRC profiles from a subset of DRC profiles comprised within a current frame of the sequence of frames; determining (611) whether at least one of the one or more DRC profiles is applicable to the first rendering mode; selecting (604) a default DRC profile as a current DRC profile, if none of the one or more DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder (100) for decoding the encoded audio signal (102); and decoding the current frame using the current DRC profile.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

An audio equalizer circuit for controlling and modifying an audio signal includes a signal divider positioned to receive an audio signal from an audio source. The divided audio signal is then passed through multiple bandpass filters situated in parallel with one another. One or more of the bandpass filters includes a compressor in order to limit the dynamic range of the audio source. After the compressor, the audio circuit may include two summation circuits with the second compressor interposed between the two.