Patent classifications
H04S7/307
Method of adjusting volume of audio output by a mobile robot device
Implementations of the disclosed subject matter provide a method of transmitting, from a mobile robot device, sound and/or at least one image captured by a sensor to a remote user device. The mobile robot device may receive at least one first control operation for the mobile robot device to move within an area via a communications network from a remote user device. An audio signal may be transmitted based on sound received at a microphone of the mobile robot device in the area. The audio signal received from the remote user device may be output at a speaker of the mobile robot device. A volume of the audio signal output by the speaker may be adjusted based on a size of the area and on an average or a median of an amplitude of frequencies in the area based on the sound received by the microphone.
ATTENTION BASED AUDIO ADJUSTMENT IN VIRTUAL ENVIRONMENTS
An attention-based audio adjustment method includes identifying, at a processor and at a first time, a first estimated gaze direction of a first participant within a virtual environment. First audio data is received at the processor from a compute device of the first participant. A second estimated gaze direction of the first participant within the virtual environment is determined by the processor at a second time. Second audio data, different from the first audio data and associated with a virtual representation of a second participant and/or a virtual object, is automatically generated by the processor, based on the first audio data and the second estimated gaze direction. A signal is sent from the processor to the compute device of the first participant, at a third time, to cause an adjustment to an audio output of the compute device of the first participant based on the second audio data.
AUTOMATIC LEVEL-DEPENDENT PITCH CORRECTION OF DIGITAL AUDIO
In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.
Audio signal processing method and audio signal processing apparatus
An audio signal processing method performs signal processing on a first audio signal to be outputted to a first device that a performer uses, the first audio signal on which the signal processing has been performed being a second audio signal, receiving a setting that causes the first audio signal to send to a monitor bus which is for to output the second audio signal, and performing signal processing on the second audio signal, which is received via the monitor bus and is to be outputted to a second device different from the first device, such that a sound quality of a sound to be outputted by the second device is closer to sound quality of a sound to be outputted by the first device than in a case where the signal processing is not performed on the second audio signal.
DISPLAY APPARATUS AND OPERATING METHOD THEREOF
A method of operating a display apparatus includes: transmitting stereo data corresponding to first audio data included in content being reproduced, to an external audio apparatus using a first audio transmission profile; changing an audio transmission profile from the first audio transmission profile to a second audio transmission profile, based on an audio-related event occurring while the stereo data is transmitted using the first audio transmission profile; and obtaining first mono audio data by selecting any one of a plurality of pieces of sound data included in the stereo data, and transmitting the first mono audio data and second mono audio data generated based on second audio data corresponding to the audio-related event, to the external audio apparatus using the second audio transmission profile.
Psychoacoustic enhancement based on audio source directivity
A device includes a memory configured to store directivity data of one or more audio sources corresponding to one or more input audio signals. The device also includes one or more processors configured to determine one or more equalizer settings based at least in part on the directivity data. The one or more processors are also configured to generate, based on the equalizer settings, one or more output audio signals that correspond to a psychoacoustic enhanced version of the one or more input audio signals.
Method for Audio Processing
A method for audio processing, the method comprising: determining at least one input audio object that includes an input audio object signal and an input audio object location, wherein the input audio object location includes a distance and a direction relative to a listener location; depending on the distance, applying a delay, a gain, and/or a spectral modification to the input audio object signal to produce a first dry signal; depending on the direction, panning the first dry signal to the locations of a plurality of speakers around the listener location to produce a second dry signal; depending on one or more predetermined room characteristics, generating an artificial reverberation signal from the input audio object signal; mixing the second dry signal and the artificial reverberation signal to produce a multichannel audio signal; and outputting each channel of the multichannel audio signal by one of the plurality of speakers.
BINAURAL MULTI-CHANNEL DECODER IN THE CONTEXT OF NON-ENERGY-CONSERVING UPMIX RULES
A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.
APPARATUS AND METHOD FOR GENERATING A DIFFUSE REVERBERATION SIGNAL
An audio apparatus for generating a diffuse reverberation signal comprises a receiver (501) receiving audio signals representing sound sources and metadata comprising a diffuse reverberation signal to total source relationship indicative of a level of diffuse reverberation sound relative to total emitted sound in the environment. The metadata also for each audio signal comprises a signal level indication and a directivity data indicative of directivity of sound radiation from the sound source represented by the audio signal. A circuit (505, 507) determines a total emitted energy indication based on the signal level indication and the directivity data, and a downmix coefficient based on the total emitted energy and the diffuse reverberation signal to total signal relationship. A downmixer (509) generates a downmix signal by combining signal components for each audio signal generated by applying the downmix coefficient for each audio signal to the audio signal. A reverberator (407) generates the diffuse reverberation signal for the environment from thedownmix signal component.
NETWORKED AUDIO AURALIZATION AND FEEDBACK CANCXELLATION SYSTEM AND METHOD
The present embodiments generally relate to enabling participants in an online gathering with networked audio to use a cancelling auralizer at their respective locations to create a common acoustic space or set of acoustic spaces shared among subgroups of participants. For example, there are a set of network connected nodes, and the nodes can contain speakers and microphones, as well as participants and node mixing-processing blocks. The node mixing-processing blocks generate and manipulate signals for playback over the node loudspeakers and for distribution to and from the network. This processing can include cancellation of loudspeaker signals from the microphone signals and auralization of signals according to control parameters that are developed locally and from the network. A network block can contain network routing and processing functions, including auralization, synthesis, and cancellation of audio signals, synthesis and processing of control parameters, and audio signal and control parameter routing.