Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: An earpiece for real-time audio processing of ambient sound includes an ear bud that provides passive noise attenuation to the earpiece such that exterior ambient sound is substantially reduced within an ear of a wearer, an exterior microphone that receives ambient sound and converts the received ambient sound into analog electrical signals, and an analog-to-digital converter that converts the analog electrical signals into digital signals representative of the ambient sounds. The earpiece further includes a digital signal processor that performs a transformation operation on the digital signals according to instructions received from a mobile device, the transformation operation transforms the digital signals into modified digital signals, a digital-to-analog converter that converts the modified digital signals into modified analog electrical signals, and a speaker that outputs the modified analog electrical signals as audio waves.

Abstract: Disclosed are a method and apparatus for producing sounds in a virtual world, as well as a sound card. The method comprises the steps of determining a spring mass model of an object in the virtual world based on a 3D model of the object; analyzing force components produced on the 3D model by a collision in the virtual world; and generating sounds produced by the collision according to the spring mass model and the force components. By considering sound material of the object in the virtual world, the method, apparatus, and sound card may produce sounds in the virtual world more vividly and in real time.

Abstract: An audio object coder for generating an encoded object signal using a plurality of audio objects includes a downmix information generator for generating downmix information indicating a distribution of the plurality of audio objects into at least two downmix channels, an audio object parameter generator for generating object parameters for the audio objects, and an output interface for generating the imported audio output signal using the downmix information and the object parameters. An audio synthesizer uses the downmix information for generating output data usable for creating a plurality of output channels of the predefined audio output configuration.

Abstract: A multivoice radio system allows users to speak and listen, at the same time, to others using multivoice radios without using a base station. Several embodiments of the present invention are directed toward combining a multivoice radio with push-to-talk (PTT), long-range radio. Other embodiments combine two or more multivoice systems and/or use an extender to increase a distance that multivoice radios can communicate with each other. Combining multivoice systems, in some embodiments, creates a voice and/or data mesh network. Switch-to-talk (STT) functionality can be added to systems to increase a number of users and promote radio discipline. Additionally, a combined in-ear microphone and speaker is disclosed. Further embodiments are directed to combining duplex radios, such as mobile phones, to multivoice systems, and/or PTT systems.

Abstract: An earphone line control device includes a power supply management unit, a filter unit, a control unit, a first resistance, a first diode, a first switch, a second resistance and a second switch.

Abstract: A method and system for providing information to a user positioned in a room with two independent listening areas where, in one area, an audio signal is output compensating for the sound from the other area, and where a position between the two areas is indicated to a user. When the user moves about, the position and size of one listening area may change, and the user may be made aware of any conflict with the other listening area. The areas may be defined by boundaries where a compensation quality is at a threshold quality.

Abstract: An audio driver with reduced rub and buzz distortion that includes a digital processing module. A digital to audio converter (DAC) operable to receive a digital audio signal supplied by the digital processing module. One or more analog driver stages operable to receive an analog audio signal supplied by the DAC. A peak amplitude compressor.

Abstract: A sound signal processing apparatus includes: a sound collector configured to collect a sound from an environment and configured to generate a sound collection signal; a signal controller configured to generate a sound signal based on the sound collection signal; a housing; a detector configured to detect a posture of the housing; and a selector configured to output one of the sound signal and a soundless signal based on the detected posture.

Abstract: An apparatus for generating a merged audio data stream is provided. The apparatus includes a demultiplexer for obtaining a plurality of single-layer audio data streams, wherein each input audio data stream includes one or more layers, wherein the demultiplexer is adapted to demultiplex each one of one or more input audio data streams having one or more layers into two or more demultiplexed audio data streams having exactly one layer. Furthermore, the apparatus includes a merging module for generating the merged audio data stream based on the plurality of single-layer audio data streams. Each layer of the input data audio streams, of the demultiplexed audio data streams, of the single-layer data streams and of the merged audio data stream includes a pressure value of a pressure signal, a position value and a diffuseness value as audio data.

Abstract: A method for automatically switching an audio source includes the steps of receiving an audio signal from a video display device, and receiving a digital audio signal from one of a plurality of audio/video source devices which each can supply audio and video information to the video display device. The digital audio signal is compared with the audio signal from the video display device. The digital audio signal is output to a supplemental audio system of the video display device when the comparing step indicates that the digital audio signal and the audio signal from the video display device contain substantially similar audio programs.

Abstract: A sound detector communicates with a portable terminal by manipulation of the portable terminal by a user and may reproduce data communicated between the sound detector and the portable terminal.

Abstract: The present research relates to controlling rendering of multi-object or multi-channel audio signals. The present research provides a method and apparatus for controlling rendering of multi-object or multi-channel audio signals based on spatial cues in a process of decoding the multi-object or multi-channel audio signals. To achieve the purpose, the method suggested in the research controls rendering in a spatial cue domain in the process of decoding the multi-object or multi-channel audio signals.

Abstract: An audio processing device including a factorization unit which factorizes frequency information obtained by performing time-frequency transformation on an audio signal of a plurality of channels into a channel matrix representing characteristics of a channel direction, a frequency matrix representing characteristics of a frequency direction, and a time matrix representing characteristics of a time direction; and an extraction unit which extracts the frequency information of audio from an arbitrary designated direction based on the channel matrix, the frequency matrix, and the time matrix.

Abstract: A method and portable terminal for automatically controlling an output volume of audio output of a head phone when the head phone is connected to a portable terminal are provided. The method of controlling an audio output in a portable terminal, includes outputting a test audio signal to a head phone for a preset playback time when the head phone is connected to the portable terminal; collecting neighboring noise interval information and impedance interval information of the head phone while the test audio signal is output to the head phone; determining impedance of the head phone based on the neighboring noise interval information and the impedance interval information; determining a gain of an audio signal based on the determined impedance; and amplifying the audio signal in proportion to the determined gain, and outputting the amplified audio signal to the head phone.

Abstract: An apparatus and method for improving the performance of a determination of whether an earphone is mounted to a portable terminal, such as a portable terminal supporting a 3.5 mm earphone plug, are provided. An apparatus for recognizing whether the earphone is mounted to the portable terminal includes a common ground terminal and a switching terminal of an earphone plug receptacle for electrically connecting to each other and the same pole of an earphone plug when the earphone plug is mounted in the earphone plug receptacle, and an external output managing unit for generating an earphone mounting notification signal through a detector.

Abstract: A speaker apparatus includes: a speaker operable to emit a sound; a receiver operable to receive an operation signal for operating an external apparatus; a relay section; a storage operable to store correspondence between operation information and operation signals that includes a first operation signal and a second operation signal; and a controller: when the operation signal received by the receiver is the first operation signal, operable to cause the relay section to transmit the operation signal received by the receiver to the external apparatus, and operate the speaker apparatus in accordance with operation information that corresponds to the first operation signal; and when the operation signal received by the receiver is the second operation signal, operable to prevent the relay section from transmitting the operation signal received by the receiver to the external apparatus.

Abstract: An apparatus for generating an enhanced downmix signal on the basis of a multi-channel microphone signal has a spatial analyzer configured to compute a set of spatial cue parameters having a direction information describing a direction-of-arrival of a direct sound, a direct sound power information and a diffuse sound power information on the basis of the multi-channel microphone signal. The apparatus also has a filter calculator for calculating enhancement filter parameters in dependence on the direction information describing the direction-of-arrival of the direct sound, in dependence on the direct sound power information and in dependence on the diffuse sound power information. The apparatus also has a filter for filtering the microphone signal, or a signal derived therefrom, using the enhancement filter parameters, to obtain the enhanced downmix signal.

Abstract: The present technology provides adaptive noise and echo reduction of an acoustic signal which can overcome or substantially alleviate problems associated with mistaken adaptation of speech and noise models to acoustic echo. The present technology carries out a multi-faceted analysis to identify echo within the near-end acoustic signal to derive an echo model. Echo classification information regarding the derived echo model is then utilized to build near-end speech and noise models. These echo, speech, and noise models are then used to generate one or more signal modifications applied to the acoustic signal to preserve the desired near-end speech signal and reduce the echo and near-end noise signals. By building near-end speech and noise models utilizing echo classification information, the present technology can prevent adaptation of the speech and noise model to the acoustic echo.

Abstract: There is provided a sound signal processing device, in which an observation signal analysis unit receives multi-channels of sound-signals acquired by a sound-signal input unit and estimates a sound direction and a sound segment of a target sound to be extracted and a sound source extraction unit receives the sound direction and the sound segment of the target sound and extracts a sound-signal of the target sound. By applying short-time Fourier transform to the incoming multi-channel sound-signals this device generates an observation signal in the time-frequency domain and detects the sound direction and the sound segment of the target sound. Further, based on the sound direction and the sound segment of the target sound, this device generates a reference signal corresponding to a time envelope indicating changes of the target's sound volume in the time direction, and extracts the signal of the target sound, utilizing the reference signal.