Abstract: Apparatus and method provided herein are directed to a linear differential directional microphone array (LDDMA), which takes into account the directionality of the array elements. The LDDMA may be designed by generating a steering vector for a linear array (LA) having preselected parameters including parameters ?, p, ?, N, and M, generating a constraint matrix based on the steering vector, reformulating the constraint matrix based on a microphone response matrix and a steering matrix, obtaining a beamformer by applying a minimum norm solution in terms of the constraint matrix, verifying a desired characteristic of the LA by calculating the beamformer for a desired direction, and constructing the LA based on the preselected parameters and the beamformer.

Abstract: An example playback device is configured to play back a first media item and, while playing back the first media item, receive a voice command to enter a preview mode in which the playback device plays back one or more second media items for a predetermined duration that is shorter than a total duration of the one or more second media items. Responsive to receiving the voice command, the playback device enters the preview mode, thereby (i) suspending playback of the first media item and (ii) initiating playback of the one or more second media items for the predetermined duration. The playback device subsequently terminates the preview mode, thereby (i) terminating playback of the one or more second media items, and (ii) resuming playback of the first media item.

Abstract: A sound capture device is disclosed, including plural microphone capsules, distributed over portion P of sphere S circumscribed between two or three planes perpendicular to each other, the three planes intersecting at a point corresponding to the center of the sphere S, and the two planes intersecting at a straight line passing through the center of the sphere S, and the sphere portion P being such that P=n S/8, with n=1,2; and a processing unit connected to the capsules to receive the signals captured by the capsules. The processing unit is arranged to matrix the signals in an ambisonic representation which retains only the ambisonic components associated with spherical harmonics that are symmetrical in relation to at least two of the aforementioned planes, and process a matrix thus obtained to identify a sound source surrounding the sphere portion and interpret a sound signal from the source.

Abstract: A method of mixing audio segments from audio clips to generate a unique stream of non-repeating sound, by: (a) inputting a plurality of audio source clips into an audio system; (b) applying a transfer function system to the plurality of audio source clips to select audio segments of the plurality of audio source clips, or applying a scheduling function system to the plurality of audio source clips to select playback times for the plurality of audio source clips; (c) applying a timeline renderer system to arrange the order of the selected audio segments; (d) applying a track renderer system to generate a plurality of audio playback clip tracks; (e) cross-fading the selected audio segments, thereby generating an audio playback clip having a unique sound stream; and (f) playing the audio playback clip having the unique sound stream.

Abstract: In one implementation, a method of playing audio is performed at a device including one or more processors coupled to non-transitory memory. The method includes playing audio associated with an object at a frequency-dependent volume based on a distance between a user location and an object location, an object orientation with respect to the user location, and the frequency-dependent three-dimensional audio emission pattern. In various implementations, methods of visualizing the frequency-dependent three-dimensional audio emission pattern and changing properties thereof are disclosed.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reflection of the input audio signal in a surface of the virtual environment.

Abstract: Technologies are disclosed for improving the efficiency of real-time audio processing, and specifically for improving the efficiency of continuously modifying a real-time audio signal. Efficiency is improved by reducing memory bandwidth requirements and by reducing the amount of processing used to modify the real-time audio signal. In some configurations, memory bandwidth requirements are reduced by selectively transferring active samples in the frequency domain—e.g. avoiding the transfer samples with amplitudes of zero or near-zero. This has particular importance when the specialized hardware retrieves samples from main memory in real-time. In some configurations, the amount of processing needed to modify the audio signal is reduced by omitting operations that do not meaningfully affect the output audio signal. For example, a multiplication of samples may be avoided when at least one of the samples has an amplitude of zero or near-zero.

Abstract: Assigning spatial information to audio segments is disclosed. A method includes receiving a first audio segment that is non-spatialized and is associated with first video frames; identifying visual objects in the first video frames; identifying auditory events in the first audio segment; identifying a match between a visual object of the visual objects and an auditory event of the auditory events; and assigning a spatial location to the auditory event based on a location of the visual object.

Abstract: Embodiments are disclosed for user posture change detection for triggering re-centering of spatial audio. In an embodiment, a method comprises: obtaining source device motion data from a source device and headset motion data from a headset worn by a user; estimating a gravity vector from one of the source device or headset motion data; splitting the source device and headset motion data into vertical and horizontal planes, the vertical plane in the direction of the estimated gravity vector and the horizontal plane perpendicular to the estimated gravity vector; calculating similarity measures based on the source device motion data and headset motion data in the vertical and horizontal planes over a time window; detecting a posture change event based on the calculated similarity measures; and resetting a head tracker error after the detected user posture change event.

Abstract: Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

Abstract: In an embodiment, a method comprises: estimating a first gravity direction in a source device reference frame for a source device; estimating a second gravity direction in a headset reference frame for a headset; estimating a rotation transform from the headset frame into a face reference frame using the first and second estimated gravity directions, a rotation transform from a camera reference frame to the source device reference frame, and a rotation transform from the face reference frame to the camera reference frame; estimating a relative position and attitude using source device motion data, headset motion data and the rotation transform from the headset frame to the face reference frame; using the relative position and attitude to estimate a head pose; and using the estimated head pose to render spatial audio for playback on the headset.

Abstract: Disclosed herein are systems and methods for storing, organizing, and maintaining acoustic data for mixed reality systems. A system may include one or more sensors of a head-wearable device, a speaker of the head-wearable device, and one or more processors configured to execute a method. A method for execution by the one or more processors may include receiving a request to present an audio signal. An environment may be identified via the one or more sensors of the head-wearable device. One or more audio model components associated with the environment may be retrieved. A first audio model may be generated based on the audio model components. A second audio model may be generated based on the first audio model. A modified audio signal may be determined based on the second audio model and based on the request to present an audio signal. The modified audio signal may be presented via the speaker of the head-wearable device.

Abstract: A method provides binaural sound to a listener while the listener watches a movie so sounds from the movie localize to a location of a character in the movie. Sound is convolved with head related transfer functions (HRTFs) of the listener, and the convolved sound is provided to the listener who wears a wearable electronic device.

Abstract: A method for use in a biometric process, comprising: with a headset in a known acoustic environment, applying an acoustic stimulus at a first transducer of the headset; receiving a response signal at a second transducer of the headset, the response signal comprising a component of the acoustic stimulus reflected at an obstacle in the acoustic path of the first transducer; determining a condition of the headset based on the response signal; and performing the biometric process based on the determined condition.

Abstract: Disclosed are a method and apparatus for displaying music points, and an electronic device and a medium. One specific embodiment of the method includes: acquiring audio material; analyzing initial music points in the audio material, wherein the initial music points include beat points and/or note starting points in the audio material; and on an operation interface of video clipping, displaying, according to the position of the audio material on a clip timeline and the positions of target music points in the audio material, identifiers of the target music points on the clip timeline, wherein the target music points are some of or all of the initial music points. According to the embodiment, the time for a user to process audio material and to make music points is reduced, and the flexibility of tools is also guaranteed.

Abstract: Systems, devices, and methods for capturing audio which can be used in applications such as virtual reality, augmented reality, and mixed reality systems. Some systems can include a plurality of distributed monitoring devices. Each monitoring device can include a microphone and a location tracking unit. The monitoring devices can capture audio signals in an environment, as well as location tracking signals which respectively indicate the locations of the monitoring devices over time during capture of the audio signals. The system can also include a processor to receive the audio signals and the location tracking signals. The processor can determine one or more acoustic properties of the environment based on the audio signals and the location tracking signals.

Abstract: A system and method for providing a dynamic audio environment within a vehicle that includes receiving data associated with at least one audio stream that is being played through an audio system of the vehicle. The system and method also include determining a selected vehicle mode of the vehicle. The system and method additionally include selecting at least one audio effect that is associated with the selected vehicle mode of the vehicle to alter the at least one audio stream. The system and method further include controlling at least one audio source of the vehicle to provide the at least one audio stream according to the at least one audio effect.

Abstract: Systems, devices, and methods for capturing audio which can be used in applications such as virtual reality, augmented reality, and mixed reality systems. Some systems can include a plurality of distributed monitoring devices. Each monitoring device can include a microphone and a location tracking unit. The monitoring devices can capture audio signals in an environment, as well as location tracking signals which respectively indicate the locations of the monitoring devices over time during capture of the audio signals. The system can also include a processor to receive the audio signals and the location tracking signals. The processor can determine one or more acoustic properties of the environment based on the audio signals and the location tracking signals.

Abstract: A multiple sound source mixing method includes dividing a plurality of sound source data into segments each with a desired length; sequentially inputting sound source data of a corresponding segment for each segment through a desired number of nodes with respect to the plurality of sound source data and mixing the input sound source data into a single piece of sound source data; and concatenating the sound source data mixed for the respective segments.

Abstract: The present disclosure provides an information processing method, including: acquiring a voice test signal, wherein the voice test signal is used for testing an influence parameter of a specified environment on a voice signal; acquiring a voice feedback signal obtained after the voice test signal is reflected by at least one reflecting object, wherein the at least one reflecting object is located in the specified environment; and estimating the influence parameter of the specified environment on the voice signal based on the voice test signal and the voice feedback signal acquired. In addition, the present disclosure also provides an information processing system, a computer system and a computer readable medium.

Abstract: A vehicle sound synthesis system is provided with a controller and a loudspeaker. The controller is programmed to receive an input indicative of at least one of a gear selection, an engine speed, and a pedal position and to generate an audio signal indicative of synthesized engine noise (SEN). The controller is further programmed to attenuate the audio signal at a first rate in response to at least one of the gear selection, the engine speed, and the pedal position indicating first vehicle conditions. The loudspeaker is adapted to project sound within a passenger compartment of a vehicle in response to receiving the attenuated audio signal.

Abstract: Disclosed is a sound analysis method and apparatus which execute an installed artificial intelligence (AI) algorithm and/or a machine learning algorithm and are capable of communicating with other electronic devices and servers in a 5G communication environment. The sound analysis method and apparatus provide a sound learning model specialized for a sound environment of a target space.

Abstract: A guitar multi-effects pedalboard is provided. The pedalboard has footswitches and a memory storing guitar-effect presets for processing an inputted guitar signal when the processing is triggered by pressing a footswitch. The pedalboard has one or more processors coupled to the memory and configured to process a first portion of an inputted guitar signal on a first audio-engine thread with a first guitar-effect preset when processing the first portion is triggered by pressing a footswitch. The one or more processors also process a second portion of the inputted guitar signal on a second audio-engine thread with a second guitar-effect preset while simultaneously processing the first portion of the inputted guitar signal on the first audio-engine thread with the first guitar-effect preset when processing the second portion is triggered by pressing a footswitch. The one or more processors simultaneously output the processed first portion and the processed second portion.

Abstract: A method of determining a position of a sound source is provided which comprises generating a spatially encoded sound-field signal using a sound-field microphone system comprising at least two microphones, wherein the spatially encoded sound-field signal comprises a plurality of components, each component including sound from the sound source. The method further comprises generating a local microphone signal corresponding to sound from the sound source using a local microphone positioned close to the sound source, comparing the local microphone signal with each of the plurality of components to generate a plurality of comparison results and using the plurality of comparison results to determine the position of the sound source relative to the sound-field microphone system.

Abstract: A method for determining proper placement of a sensor pod on a patient comprising: performing a first quality control procedure on a detection device, wherein said detection device comprises a base unit, at least two sensor pods, a computer system implementing appropriate software, and a display; wherein the first quality control procedure generates a tone from a speaker embedded within said base unit and wherein each of said sensor pods measures and compares the measured sound to a predetermined measurement in real-time; wherein a sensor pod is determined to have met quality control if said sound is within 10% of the predicted measurements; performing a second quality control procedure on said sensor pods, wherein said sensor pods measure sounds on a patient; wherein the system, once engage, detects sounds from the sensor pods and compares the detected sounds in real-time to a predicted sound based on the fluid flow vessel; and wherein said method provides for an audio or visual alarm when said sensor pod is

Abstract: A vehicle sound synthesis system is provided with a loudspeaker and a controller. The loudspeaker projects sound indicative of synthesized engine noise (SEN) within a cabin of a vehicle in response to receiving a SEN signal. The controller is programmed to generate the SEN signal; and receive a first input indicative of an engine start command. The controller is further programmed to, modulate a characteristic of the SEN signal to align with a corresponding engine operating characteristic during starting conditions in response to the engine start command; and provide an adjusted SEN signal including the modulated characteristic to the loudspeaker.

Abstract: A method and corresponding system for generating harmonics from an input signal having an input frequency uses a harmonics generator having an associated gain controller generating adjustable gain control factors, controlling the amplitude of harmonic components. The method includes generating a series of harmonic components in a process including adjustment of relative balances between at least three harmonic components of different orders, the balances being independently adjustable by controlling at least two gain control factors. Each gain control factor is adjusted based on a frequency dependent measure of the input signal or signal derived therefrom, of the frequency dependent measure of the input signal and/or derived signal unique for the corresponding gain control factor. An output signal is created including the harmonic components according to the balances as controlled by the at least two gain control factors, wherein the balances between the harmonic components depend on the input frequency.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for training acoustic models and using the trained acoustic models. A connectionist temporal classification (CTC) acoustic model is accessed, the CTC acoustic model having been trained using a context-dependent state inventory generated from approximate phonetic alignments determined by another CTC acoustic model trained without fixed alignment targets. Audio data for a portion of an utterance is received. Input data corresponding to the received audio data is provided to the accessed CTC acoustic model. Data indicating a transcription for the utterance is generated based on output that the accessed CTC acoustic model produced in response to the input data. The data indicating the transcription is provided as output of an automated speech recognition service.

Abstract: A local extremum calculator detects a local maximum sample and a local minimum sample of a digital audio signal. A number-of-sample detector detects a sample interval between the local maximum sample and the local minimum sample. A difference value calculator calculates difference values between adjacent samples. A correction value calculator calculates a first correction value by multiplying the difference value between the local maximum sample and a first adjacent sample by a coefficient and calculates a second correction value by multiplying the difference value between the local minimum sample and a second adjacent sample by the coefficient. When a periodic signal detector detects that the digital audio signal is a single sine wave, an adder/subtractor does not add the first correction value to the first adjacent sample, and does not subtract the second correction value from the second adjacent sample.

Abstract: Methods, systems, devices and apparatuses for a sound enhancement system. The sound enhancement system includes an airflow sensor. The airflow sensor is configured to measure an airflow into an engine of the vehicle. The sound enhancement system includes an electronic control unit. The electronic control unit is coupled to the airflow sensor. The electronic control unit is configured to determine a time of when to output an audio signal that mimics or enhances engine sound based on the airflow into the engine of the vehicle. The electronic control unit is configured to generate the audio signal based on the airflow into the engine of the vehicle. The sound enhancement system includes an audio device. The audio device is configured to output the audio signal based on the timing of when to output the audio signal.

Abstract: [Technical problem] Circulatory sound regulation that changes environmental sound into input sound, converted sound with arbitrarily regulatory frequency component, arbitrarily regulatory amplitude or both of them, output sound, another input sound synthesized with this output sound and an environmental sound, and another converted sound made from this input sound.

Abstract: A method and device for audio signal processing is provided. The method includes steps of: obtaining an inputted audio signal; parsing the audio signal to obtain at least one audio feature; determining at least one vibration feature corresponding to the at least one audio feature; and generating a vibration signal corresponding to the audio signal according to the at least one vibration feature. The inputted audio signal is automatically converted into a vibration signal by the vibration feature corresponding to the audio feature of the inputted audio signal, which can avoid errors caused by manual operation and make the vibration signal possess high versatility.

Abstract: A multifunctional Bluetooth microphone includes a handle having a microphone head formed at one end to receive a sound signal; a device mounting unit coupled to the handle to be adjacent to the microphone head, equipped with Bluetooth technology, and provided with a high-power speaker on one side; a digital signal processing unit provided in the device mounting unit to add an additional function of at least one type of echo mode or at least one type of voice modulation mode to a voice of a person by performing digital signal processing on the voice; a function input unit provided in the device mounting unit for operation of the digital signal processing unit; and a microphone controller for controlling operation of the digital signal processing unit on the basis of an input signal of the function input unit.

Abstract: The present technology relates to a signal processing device, a signal processing method, and a program that enable implementation of more effective distance feeling control. The signal processing device includes a reverb processing unit that generates a signal of a reverb component on the basis of object audio data of an audio object and a reverb parameter for the audio object. The present technology can be applied to a signal processing device.

Abstract: Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

Abstract: A display device includes: a display panel; a first sound generating device disposed below the display panel, where the first sound generating device generates a sound by causing the display panel to vibrate; and a second sound generating device disposed below the display panel, where the second generating sound generating device generates a sound by causing the display panel to vibrate. The first sound generating device includes: a first vibrator which vibrates in a first direction, which is a thickness direction of the display panel; and a second vibrator which vibrates in the first direction, and the second sound generating device includes: a third vibrator which vibrates in the first direction; and a fourth vibrator which vibrates in a second direction, which is perpendicular to the first direction.

Abstract: A game-agnostic event detector can be used to automatically identify game events. Game-specific configuration data can be used to specify types of pre-processing to be performed on media for a game session, as well as types of detectors to be used to detect events for the game. Event data for detected events can be written to an event log in a form that is both human- and process-readable. The event data can be used for various purposes, such as to generate highlight videos or provide player performance feedback.

Abstract: An audio engine is provided for acoustically rendering a three-dimensional virtual environment. The audio engine uses geometric volumes to represent sound sources and any sound occluders. A volumetric response is generated based on sound projected from a volumetric sound source to a listener, taking into consideration any volumetric occluders in-between. The audio engine also provides for modification of a level of detail of sound over time based on distance between a listener and a sound source. Other aspects are also described and claimed.

Abstract: The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

Abstract: An audio system and method of spatially rendering audio signals that uses modified virtual speaker panning is disclosed. The audio system may include a fixed number F of virtual speakers, and the modified virtual speaker panning may dynamically select and use a subset P of the fixed virtual speakers. The subset P of virtual speakers may be selected using a low energy speaker detection and culling method, a source geometry-based culling method, or both. One or more processing blocks in the decoder/virtualizer may be bypassed based on the energy level of the associated audio signal or the location of the sound source relative to the user/listener, respectively. In some embodiments, a virtual speaker that is designated as an active virtual speaker at a first time, may also be designated as an active virtual speaker at a second time to ensure the processing completes.

Abstract: In general, techniques are described for modeling occlusions when rendering audio data. A device comprising a memory and one or more processors may perform the techniques. The memory may store audio data representative of a soundfield. The one or more processors may obtain occlusion metadata representative of an occlusion within the soundfield in terms of propagation of sound through the occlusion, the occlusion separating the soundfield into two or more sound spaces. The one or more processors may obtain a location of the device, and obtain, based on the occlusion metadata and the location, a renderer by which to render the audio data into one or more speaker feeds that account for propagation of the sound in one of the two or more sound spaces in which the device resides. The one or more processors may apply the renderer to the audio data to generate the speaker feeds.

Abstract: The present disclosure is directed to devices and systems for auto configuration of directional sound. The system comprises: a first and second portable bass loudspeaker, each arranged to radiate sound waves responsive to audio signals. Each portable bass loudspeaker has a sensor for determining whether the portable bass loudspeaker is in an aligned state or non-aligned state, wherein in the aligned state, the first portable bass loudspeaker is oriented in a generally parallel and opposite direction relative to the second portable bass loudspeaker. Each portable bass loudspeaker has an equalization device providing a first equalization setting for outputting audio in response to detection of the aligned state, and a second equalization setting for outputting audio in response to detection of the non-aligned state, wherein the first equalization setting is configured to generate a radiating output from the first and second portable bass loudspeaker in a cardioid pattern.

Abstract: A method for conveying to a listener a directionality-preserving pseudo low frequency psycho-acoustic sensation of a multichannel sound signal, including: deriving from the sound signal, by a processing unit, a high frequency multichannel signal and a low frequency multichannel signal, generating a multichannel harmonic signal, the loudness of at least one channel signal of the multichannel harmonic signal substantially matching the loudness of a corresponding channel in the low frequency multichannel signal; and at least one interaural level difference (ILD) of at least one frequency of the at least one channel pair of the multichannel harmonic signal substantially matching an ILD of a corresponding fundamental frequency in a corresponding channel pair in the low frequency multichannel signal; and summing the harmonic multichannel signal and the high frequency multichannel signal thereby giving rise to a psychoacoustic alternative signal.

Abstract: Example speech enhancement and noise suppression systems and methods are described. In one implementation, a method receives an audio file comprising a combination of voice data and noise data, and divides the audio file into multiple frames. The method performs a discrete Fourier transform on each frame of a first subset of the multiple frames to provide a plurality of frequency-domain outputs, which are input to a neural network. A ratio mask is obtained as an output from the neural network and clean voice coefficients are computed using the ratio mask. The method outputs an audio file having enhanced speech based on the computed clean voice coefficients.

Abstract: Disclosed herein are a number of example embodiments for an improved audio sample playback unit. For example, multi-dimensional mapping of triggers to audio samples is disclosed. Also disclosed is low latency retrieval and playback of audio samples via pre-loading of sample heads into high speed memory. Furthermore, disclosed herein is a multi-threaded control operation for generating audio frames in response to trigger inputs, as well as the use of multiple pipes from which audio data can be generated. Further still, an example embodiment provides for multi-level control of audio properties, including voice-specific controls, pipe-specific controls, and global controls.

Abstract: A method of avoiding speech collisions among participants during teleconferences includes, during a teleconference, performing, using computer hardware, audio analysis of voices of a plurality of participants of the teleconference, determining, using the computer hardware, an emotive state for each of the plurality of participants during the teleconference, generating, using the computer hardware, a user frequency model for each of the plurality of participants based on the audio analysis and the emotive state, and adjusting, using the computer hardware, frequency of a voice of at least one of the plurality of participants during the teleconference based on the user frequency models of the plurality of participants to avoid a speech collision.

Abstract: Synthesizing engine sound in a cabin of a vehicle includes generating a synthesized engine noise audio signal including a plurality of engine order outputs; mixing the engine order outputs in accordance with a first set of gain levels to provide a first channel output; mixing the engine order outputs in accordance with a second set of gain levels to provide a second channel output; using a first localization filter set to provide a first localized output to localize the first channel output as a first sound image at a first location within the cabin, and a second localization filter set to provide a second localized output to localize the second channel output as a second sound image at a second location within the cabin; and mixing the first and second localized outputs into loudspeaker outputs to be applied to the loudspeakers in the cabin.

Abstract: A video game processing apparatus stores music information indicating music, the music being constructed by a plurality of tracks, the music being reproduced with plural kinds of reproducing styles by reproducing an arbitrary combination of one or two or more tracks; stores music reproducing style information, the music reproducing style information indicating a reproducing style of music according to a game point value; starts reproduction with a reproducing style according to the game point value; reproduces the music indicated by the music information; updates the game point value in accordance with a progress status of the video game; refers to the updated game point value and the music reproducing style information to determine whether the reproducing style of the reproduced music is to be updated or not; and updates the reproducing style of the reproduced music to a reproducing style according to the updated game point value.

Abstract: An apparatus comprising: a directional analyser configured to determine a directional component of at least two audio signals; an estimator configured to determine at least one virtual position or direction relative to the actual position of the apparatus; and a signal generator configured to generate at least one further audio signal dependent on the at least one virtual position or direction relative to the actual position of the apparatus and the directional component of at least two audio signals.

Abstract: A method of processing for sound effect of recording and a mobile terminal are disclosed in embodiments of the present disclosure. The method includes recording N audio signals generated from N sound sources, N being an integer greater than or equal to 2; obtaining location relationships of the N sound sources, determining a scene where the N sound sources are located in, and determining a reverberation algorithm according to the location relationships of the N sound sources and the scene where the N sound sources are located in; and performing a sound effect processing for the N audio signals according to the reverberation algorithm to obtain a reverbed sound effect.