Features are disclosed for measuring and correcting clock drift and propagation delay in an audio system through one or more waveforms embedded in an audio signal. A first device in communication with a speaker may be configured to obtain an audio signal and insert one or more waveforms into the audio signal. For example, the waveforms may be inserted during an interval of time. A second device in communication with a microphone may be configured to detect sound as an audio input signal. The second device may obtain a spectral representation of the audio input signal and determine a rotation based on the spectral representation at the frequency of at least one of the inserted waveforms. Clock drift may be determined based on the rotation.

Aspects of the present disclosure relate to techniques for processing a source audio signal in order to localize sounds. In particular, aspects of the present disclosure relate to sound localization techniques which externalize sounds for headphone audio, such as a virtual surround sound headphone system. In various implementations, room reverberations and other acoustic effects of the environment may be more accurately modeled using improved room reverberation models. For example, in some implementations, the underlying source signal may be filtered with a filter representing a room impulse response that is a combination of a stereo room impulse response and a mono room impulse response. By way of further example, in some implementations the source signal may be filtered with a combined impulse response filter that is derived from binaural recordings of simulated impulses recorded in a desired reverberant environment.

Methods are provided for equalizing the group delay of a sound reproduction system, in particular a system comprising acoustic transducers with at least one crossover between a lower-frequency and a higher-frequency range. A correction is applied to a signal in the lower-frequency range, including the crossover region, to substantially equalize the group delay for the lower-frequency range, and a signal delay is applied to a signal in the higher-frequency range to bring it into closer alignment with the equalized lower-frequency range signal. The methods may be implemented in the design of an acoustic transducer system and also via a computer program product, which can be implemented as an update or enhancement to an existing digital signal processor loudspeaker system.

A system and method for driving a loudspeaker array across directivities and frequencies to maintain timbre constancy in a listening area is described. In one embodiment, a frequency independent room constant describing the listening area is determined using the directivity index of a first beam pattern, the direct-to-reverberant ratio DR at the listener's location in the listening area, and an estimated reverberation time T.sub.60 for the listening area at a designated frequency. On the basis of this room constant, an offset may be generated for a second beam pattern. The offset describes the decibel difference between first and second beam patterns to achieve constant timbre and may be used to adjust the second beam pattern at multiple frequencies. Maintaining constant timbre improves audio quality regardless of the characteristics of the listening area and the beam patterns used to represent sound program content. Other embodiments are also described.

A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal.

A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal.

Apparatus and methods for processing an audio signal. The apparatus may include a digital signal processor (“DSP”) configured to receive an audio signal from a vehicular audio signal line. The audio signal may include a flat component and a compensatory component. The flat component may correspond to the audio signal in a state before combination with the compensatory component. The compensatory component may include an OEM EQ component. The apparatus may include a microprocessor in electronic communication with the DSP. The microprocessor may be configured to retrieve from the memory a restorative signal component. The microprocessor may be configured to instruct the DSP to apply the restorative signal component to the audio signal to reduce the audio signal to the flat component.

A vibrator is fitted to a human body for use. A variable filter extracts a signal of a predefined frequency range from an input music signal. A driving controller drives the vibrator based on the signal of the frequency range extracted by the variable filter. It is possible to adjust the frequency range that is predefined by using a user operation interface. The frequency range that is predefined may be determined by a frequency determination interface.

A method for changing an image file based on a toning coefficient includes receiving an image file including an image having a plurality of pixels. Toning characteristic function is determined for each pixel of the received image. Distribution of the plurality of pixels of the received image is analyzed using the toning characteristic function. The toning coefficient of the received image is determined based on the analyzed distribution of the plurality of pixels. A degree of toning of the received image file is changed using a user specified criteria based on the determined toning coefficient.

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.