503results about How to "Enhance listening experience" patented technology

CBR control strategies provide constant or relatively constant bitrate output with variable quality. The control strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder uses a trellis in two-pass or delayed-decision CBR encoding. The trellis nodes are states derived by quantizing buffer fullness values. The transitions between nodes of a previous stage and nodes of a current stage depend on encoding a current chunk of audio at different quality levels. When pruning the trellis, the encoder uses a cost function that considers smoothness in quality as well as quality in absolute terms. The encoder may store compressed data at different quality levels, then output the compressed data after simplification of the trellis to a suitable point. If the two-pass or delayed-decision CBR encoding fails, the encoder uses one-pass CBR encoding for the sequence or part of the sequence.

A method for processing audio signals optimizes the listening experience for hearing impaired individuals to feel stigmatized by requiring them to employ special hearing-impaired equipment. A user actuated controller controls a mixture of a preferred audio signal and a remaining audio signal across a range sufficiently wide enough to encompass all individuals. The preferred audio is recorded and maintained separate from all remaining audio and delivered to the listener in a manner that maintains the separateness of the preferred audio and the remaining audio. The user actuated controller includes the capability of automatically maintaining the listener established ratio in the face of changes in the audio signal. The user actuated controller enables the user to specify a range about the ratio in which the audio may vary, which permits the listener to expand the audio across a continuous range to whatever dynamic range his hearing can accommodate. The controller automatically adjusts to changes in incoming audio. The controller can react to relatively slowly moving changes or prevent short bursts of sound in the remaining audio from modifying the signal levels. The combination of the above aspects provides a heretofore not possible listening experience that can accommodate the listening desires of all listeners. The combination of the ability to control the ratio of preferred audio to remaining audio and to specify the dynamic range about the ratio in which the audio may vary, coupled with the ability of the controller to automatically adjust the signal levels in response to sudden changes in incoming audio, provides a powerful user capability that truly optimizes the listening experience for any listener.

An encoder uses multi-pass VBR control strategies to provide constant or relatively constant quality for VBR output while guaranteeing (within tolerance) either compressed file size or, equivalently, overall average bitrate. The control strategies include various techniques and tools, which can be used in combination or independently. For example, in a first pass, an audio encoder encodes a sequence of audio data partitioned into variable-size chunks. In a second pass, the encoder encodes the sequence according to control parameters to produce output of relatively constant quality. The encoder sets checkpoints in the second pass to adjust the control parameters and / or subsequent checkpoints. The encoder selectively considers a peak bitrate constraint to limit peak bitrate. The encoder stores auxiliary information from the first pass for use in the second pass, which increases the speed of the second pass. Finally, the encoder compares signatures for the input data to check consistency between passes.

An acoustic correction apparatus processes a pair of left and right input signals to compensate for spatial distortion as a function of frequency when said input signals are reproduced through loudspeakers in a sound system. The sound-energy of the left and right input signals is separated and corrected in a first low-frequency range and a second high-frequency range. The resultant signals are recombined to create image-corrected audio signals having a desired sound-pressure response when reproduced by the loudspeakers in the sound system. The desired sound-pressure response creates an apparent sound image location with respect to a listener. The image-corrected signals can also be spatially-enhanced to broaden the apparent sound image and improve the low frequency characteristics of the sound when played on small loudspeakers.

An external microphone array is configured to be used with a hearing aid and comprises a number of microphones configured to detect one or more sound signals from a sound source and means for wirelessly sending the detected sound signal to at least one hearing aid. The external microphone array comprises means for automatically determining the direction of the sound source either by: a) receiving a wireless signal transmitted from a remote control; b) by receiving acoustic signals picked up by the hearing aid(s) and by further comparing the signals with signals received by the external microphone array. The external microphone array may e.g. take the form of a sphere, a hemi-sphere or a neckband.

A passive acoustic booster attachment for a mobile media device is disclosed, the attachment including an attaching portion, and a passive acoustic boosting portion extending from the attaching portion. The attaching portion attaches the passive acoustic booster attachment to the mobile media device. The passive acoustic boosting portion includes an acoustic waveguide, the acoustic waveguide being a conduit capable of guiding sound waves from at least one speaker of the mobile media device towards a listener. In some embodiments, the attaching portion serves as a case that holds and secures the entire perimeter of the mobile media device. In some embodiments, the attaching portion is a clip that can attach to a mobile media device. By guiding the sound waves produced by the speakers of the mobile media device, the passive acoustic booster attachment provides an enhanced listening experience.

A method for processing audio signals optimizes the listening experience for hearing impaired listeners, as well as non-hearing impaired listeners, without forcing hearing impaired individuals to feel stigmatized by requiring them to employ special hearing-impaired equipment. A user actuated controller controls a mixture of a preferred audio signal and a remaining audio signal across a range sufficiently wide enough to encompass all individuals. The preferred audio is recorded and maintained separate from all remaining audio and delivered to the listener in a manner that maintains the separateness of the preferred audio and the remaining audio. The user actuated controller includes the capability of automatically maintaining the listener established ratio in the face of changes in the audio signal. The user actuated controller enables the user to specify a range about the ratio in which the audio may vary, which permits the listener to expand the audio across a continuous range to whatever dynamic range his hearing can accommodate. The controller automatically adjusts to changes in incoming audio. The controller can react to relatively slowly moving changes or prevent short bursts of sound in the remaining audio from modifying the signal levels. The combination of the above aspects provides a heretofore not possible listening experience that can accommodate the listening desires of all listeners. The combination of the ability to control the ratio of preferred audio to remaining audio and to specify the dynamic range about the ratio in which the audio may vary, coupled with the ability of the controller to automatically adjust the signal levels in response to sudden changes in incoming audio, provides a powerful user capability that truly optimizes the listening experience for any listener.

The invention relates to a sound effect compensation method, a sound effect compensation device, a readable storage medium and a terminal. The method comprises the following steps: determining an acoustic transmission function of an audio processing device in an auditory meatus of a user according to an audio signal currently output by the audio processing device; acquiring a current monitoring frequency response curve of the audio processing device according to the acoustic transmission function; and comparing the monitoring frequency response curve with a standard frequency response curve tocompensate the audio signal. Ear print information of the user and the monitoring frequency response curve corresponding to the ear print information can be acquired through the currently output audio signal of the audio processing device, the monitoring frequency response curve is adjusted automatically according to the standard frequency response curve, thus, timbre of the audio signal can be compensated, consequently, the method and the device can be adapted to the users with different ear print information, and better auditory experience can be provided for different users.

The invention discloses an audio playing method and audio playing equipment, and is used for solving the technical problem of low intelligence of the audio playing equipment. The method comprises the following steps of: obtaining the scene parameters of an environment where the audio playing equipment is positioned, and the user information of a user in the environment, wherein the scene parameters at least comprise a physical position where the audio playing equipment is positioned and a current moment when the scene parameter is obtained; according to the scene parameters and the user information, constructing a first playing scene; according to a corresponding relationship between the playing scene and a playing strategy, carrying out audio playing by the first playing strategy corresponding to the first playing scene, wherein the playing strategy comprises at least one selection strategy includssding an audio type selection strategy, a playing volume selection strategy and a playing sound effect selection strategy.

The invention discloses a playing control method and electronic equipment, relates to the field of electronic device applications, and is designed for automatic adjustment of a playing direction of a speaker according to the position of a user. The method comprises obtaining a first position parameter and a second position parameter in a preset reference system, wherein the first position parameter is the position parameter of the speaker and the second position parameter is the position parameter of the user; obtaining a first playing direction parameter according the first position parameter and the second position parameter; and transmitting the first playing direction parameter to the speaker so that the speaker can play audio according to the first playing direction parameter. The playing control method and the electronic equipment are mainly used in the playing process of smart televisions.

The present invention discloses an automatic adjusting method for output volume of media player. The method comprises the following steps: 1) the output volume control module of the media player is opened; 2) the default valve of the standard output volume is set and saved; 3) the output volume control module measures the actual primary volume output value of the multimedia file; adjusts the output value to the default value and then output the default value; 4) the output volume of the player is adjusted based on the actual output volume. With the technical proposal, the output volume is constant by utilizing the media player no matter the frequency of the primary media file is recorded in whatever volume; the output of the frequency is more humanization and more adapt to the habit of consumer. The output volume remains constant under the unstable frequency output condition of the primary media file to increase the sound experience, so as to protect the audition of the consumers and to prevent the earphone from being damaged in time.

A music visualization system and methods involving a central processing unit capable of converting waveform data to geometry data, a graphics processing unit capable of recognizing and accepting the geometry data and rendering a plurality of graphical images, a custom shader software program being operable on the graphics processing unit, an embeddable platform being in electronic communication with the graphics processing unit, and an audiovisual display device in electronic communication with the graphics processing unit and the embeddable platform.

The invention relates to a method and device for determining earphone noise reduction parameters and a computer readable medium. The earphone comprises a loudspeaker, an in-ear microphone and a filterassembly. The method comprises the following steps of: playing a low-frequency audio signal of which the frequency is out of the auditory range of human ears by a loudspeaker, determining a current transfer function of a transmission path from the loudspeaker to the in-ear microphone or a parameter of a current audio signal acquired by the in-ear microphone, determining a current noise reductionparameter based on the current transfer function or the parameter of the current audio signal by referring to N groups of corresponding relationships between a preset transfer function or a parameterof a preset audio signal and a preset noise reduction parameter, the noise reduction parameter comprising a filter coefficient of a filter enabled in the filter assembly, and configuring the filter assembly with the current noise reduction parameter to perform noise reduction. By playing the low-frequency audio signals beyond the auditory range of the human ears for many times, the dynamic adjustment of the noise reduction parameters of the earphone can be realized, the noise reduction effect in different use scenes is ensured, and meanwhile, the listening experience of a user is improved.

A method for generating an auditory environment for a user may include receiving a signal representing an ambient auditory environment of the user, processing the signal using a microprocessor to identify at least one of a plurality of types of sounds in the ambient auditory environment, receiving user preferences corresponding to each of the plurality of types of sounds, modifying the signal for each type of sound in the ambient auditory environment based on the corresponding user preference, and outputting the modified signal to at least one speaker to generate the auditory environment for the user. A system may include a wearable device having speakers, microphones, and various other sensors to detect a noise context. A microprocessor processes ambient sounds and generates modified audio signals using attenuation, amplification, cancellation, and / or equalization based on user preferences associated with particular types of sounds.

An apparatus for spatial sound reproduction comprises a receiver (101) for receiving a multi-channel audio signal. An analyzer (107) determines a spatial property of the multi-channel audio signal, such as a spatial complexity or organization. A selection processor (109) then selects a reproduction mode from a plurality of sound reproduction modes where the multi-channel sound reproduction modes employ different spatial rendering techniques. A reproduction circuit (103) then drives a set of loud-speakers (105) to reproduce the multi-channel audio signal using the selected reproduction mode. The switching between the reproduction modes may be fast (e.g. in the order of 100 ms to 10 secs) thereby allowing a short term adaptation of the reproduction mode to the signal characteristics. The approach may in particular provide an improved spatial experience to a listener.

The invention discloses an audio playing method and an audio player. In the embodiment of the invention, corresponding equalizer parameter is obtained according to the audio type information; the equalizer parameter is loaded to play the audio. The invention realizes automatic EQ setup of audio playing, and users can obtain better playing effect without setting EQ corresponding to the audio type,thus improving hearing experience of users.

An apparatus, such as a decoder is arranged to modify the sweet-spot of a spatial M-channel audio signal by modifying spatial parameters. Specifically, a receiver (201) receives an N-channel audio signal where N<M. The M-channel signal may specifically be an MPEG Surround sound signal and the N-channel signal may be a stereo signal. A parameter unit (203) determines spatial parameters relating the N-channel audio signal to the spatial M-channel audio signal and a modifying unit (207) modifies the sweet-spot of the spatial M-channel audio signal by modifying at least one of the spatial parameters. A generating unit (205) then generates the spatial M-channel audio signal by up-mixing the N-channel audio signal using the at least one modified spatial parameter. An efficient and / or low complexity sweet-spot manipulation is achieved by an integration of sweet-spot manipulation and multi-channel generation.

A sound enhancement system (SES) that can enhance reproduction of sound emitted by headphones and other sound systems. The SES improves sound reproduction by simulating a desired sound system without including unwanted artifacts typically associated with simulations of sound systems. The SES facilitates such improvements by transforming sound system outputs through a set of one or more sum and cross filters, where such filters have been derived from a database of known direct and indirect head-related transfer functions (HRTFs).

In one aspect, a method of providing directive transitions between audio signals comprises associating a first / second browsing direction (A1, A2) with a first / second transition effect template. In response to a browsing action in one of said browsing directions, a transition is played in which an exit segment (S0-out1, S0-out2) and an entry segment (S1-in, S2-in) are mixed in accordance with the associated transition effect template. A further aspect proposes a method of transitioning between audio signals decoded from audio data including time markers encoded as audio metadata and indicating at least one section of the respective audio signal. The method includes retrieving time marking information in the audio data; extracting an exit segment (S0-out1, S0-out2) and an entry segment (S1-in, S2-in), wherein an endpoint of at least one of the segments is synchronized with a time marker; and playing a transition in which the exit and entry segments are mixed in accordance with a transition effect template.

The embodiment of the invention discloses a Bluetooth-based data transmission method and device, and relates to the technical field of data processing. According to the Bluetooth-based data transmission method and device, the transmission continuity of audio data can be guaranteed, therefore, the condition that an audio play equipment end generates the play blocking and unsmoothness phenomena is avoided, and the listening experience of a user is improved. According to the main technical scheme, the method comprises the steps that to-be-transmitted audio data is acquired; the audio data is compressed according to a preset coding and decoding algorithm to obtain a to-be-transmitted data packet, wherein the preset coding and decoding algorithm is the algorithm which is used for compressing and determined according to a preset mapping relation between the sizes of to-be-transmitted files and the Bluetooth communication bandwidths; and by means of a Bluetooth technology, the to-be-transmitted data packet is sent to a receiving terminal, so that the receiving terminal decompresses the received audio data according to the preset coding and decoding algorithm and then plays the audio data.The Bluetooth-based data transmission method and device are mainly used for transmitting data between different terminals by means of the Bluetooth technology.

The invention relates to an active noise reduction method and system, and an earphone, and the method comprises the steps: responding to an audio signal played by a loudspeaker of the earphone, and determining a current transfer function of a transmission path from the loudspeaker to an in-ear microphone of the earphone through an echo filter of the earphone; determining a current filter coefficient of a feed-forward filter based on a preset filter coefficient of the feed-forward filter of the earphone, a preset transfer function of a transmission path from the loudspeaker to an in-ear microphone of the earphone and a current transfer function; and configuring a feed-forward filter according to the current filter coefficient so as to carry out feed-forward active noise reduction. Through active adaptive adjustment of the current filter coefficient of the feed-forward filter, the influence of different wearing modes and different ear canal structures on the noise reduction system can bereduced, the noise reduction effect of the earphone is improved, and meanwhile, the listening experience of a user is improved.

A personal listening device used in connection with the amplification of signals for enhancing the perceived quality and ambiance of sound heard by listener using stereo headphones or small speaker systems utilizes a difference signal that is cross-fed between stereo channels. The invention provides a perception of spatial presence in music sound when used with either stereo headphones or small speaker systems. The circuitry, wiring and packaging provide for multiple output jacks allowing for convenient sharing of the enhanced output signal from a personal listening device with other persons using headphones. The invention provides effects similar to stereo enhancement, surround sound, and psychoacoustic imagery in a small, inexpensive package for enhancing a user's audio experience.

The invention discloses a noise generation method which includes the following steps: according to a noise parameter obtained beforehand, an initial value of a reconstruction parameter is obtained; according to the initial value of the reconstruction parameter, random value-taking scope is obtained; in the random value-taking scope, a value is picked by random as a reconstructed noise parameter; according to the reconstructed noise parameter, noise is obtained. The invention also discloses a noise generation device which comprises: an initial value unit used for obtaining the initial value of the reconstruction parameter according to the noise parameter obtained beforehand; a scope unit used for obtaining random value-taking scope according to the initial value of the reconstruction parameter; a reconstruction unit used for picking value by random as a reconstructed noise parameter in the random value-taking scope; an integration unit used for combing noise according to the reconstructed noise parameter. The method can be applicable to various standard protocols and cause that more comfortable noise can be decoded for a user in a decoding end.

The embodiment of the invention provides a tone quality configuration method and a mobile terminal. The method comprises the steps of outputting a first audio signal via a loudspeaker, wherein the first audio signal carries an infrasonic wave signal; according to a second audio signal collected by the microphone, acquiring a target amplitude value corresponding to the infrasonic wave signal; according to the target amplitude value, judging whether the current wearing state of an earphone is a standard wearing state; if no, extracting a locally stored target compensation array set correspondingto the target amplitude value; and performing tone quality compensation on the first audio signal according to the target compensation array set. Therefore, the wearing state of the current earphonecan be monitored during a process that a user uses the earphone, and under the condition that the wearing state is nonstandard, the current tone quality is compensated via the pre-stored compensationarray set, and thus the negative effects on middle and low frequencies when the coupling of the earphone is loosen are effectively improved, and the audition experience of the earphone user is effectively enhanced.

The embodiment of the invention provides a simultaneous interpretation method and device. The method comprises the steps that source language voice data is collected; voice translation results which are represented by a target language and correspond to the recognition results of the source language voice data are obtained and output; the voice translation results are obtained through natural person voice synthesis, and the target voice and the source language belong to different languages. Accordingly, automatic voice recognition and translation can be achieved, the labor cost is lowered, theaccuracy and integrity of the translation results are improved, the voice translation results have the natural person voice characteristics, and the listening experience is effectively improved.

The invention discloses a method for playing vehicle-mounted music. The method comprises the following steps of: acquiring the current position information of a vehicle-mounted terminal; determining the current position type corresponding to the current position information of the vehicle-mounted terminal; and playing matched music based on the current position type of the vehicle-mounted terminal. The invention also provides a device for playing the vehicle-mounted music. The device comprises an acquisition unit, a determination unit and a control unit, wherein the acquisition unit is used for acquiring the current position information of the vehicle-mounted terminal; the determination unit is used for determining the current position type corresponding to the current position information of the vehicle-mounted terminal which is acquired by the acquisition unit; and the control unit is used for controlling a media player of the vehicle-mounted terminal to play the matched music based on the current position type of the vehicle-mounted terminal which is determined by the determination unit. By the device for playing the vehicle-mounted music, the method for playing the vehicle-mounted music is realized, so that the media player of the vehicle-mounted terminal can provide diversified music playing service, the problem of alternate inter-cut between a broadcast and the vehicle-mounted music is solved, and the listening experience of the vehicle-mounted music during driving of an automobile user is enhanced.

The application provides an equalizer adaptive adjusting system and method and an earphone possessing an adaptive adjusting function. The equalizer adaptive adjusting method is used to provide the adaptive adjusting function aiming at an audio signal outputted by an equalizer, and mainly comprises the steps of using an analyzer to send a test signal to the interior of an ear to analyze the internal structure of the ear, adjusting an equalizer control parameter according to an analysis result, and controlling the equalizer to adjust and output the received audio signal according to the adjusted equalizer control parameter. Therefore, according to the application, by analyzing the internal structures of the ears of different users, the most suitable equalizer control parameters are selected automatically, thereby adapting to the users having different internal structures of the ears, and providing the best listening effects for the different users. Moreover, according to the application, the users do not need to adjust the tone quality manually, for the users, any professional skill requirement is not required, and the equalizer adaptive adjusting system, the equalizer adaptive adjusting method and the earphone are suitable for the general customers to operate and use, and have the advantage of wide in application range.

The invention relates to the technical field of signal processing, and provides an audio signal playing system and an electronic device. The system comprises two or more than two loudspeakers which are used for respectively playing audio signals of different frequencies, a filtering unit which is used for filtering the input audio signals, and carrying out amplifying and outputting to the corresponding loudspeakers, and a feedback control unit which is used for collecting the audio signals played by two or more than two loudspeakers, identifying the power sizes of the audio signals in different frequencies, and then adjusting the magnification factor of the corresponding frequency of each audio signal filtered by the filtering unit according to the identifying results. The feedback control of the audio playing strength of each loudspeaker is realized, so that the loudness of each loudspeaker is consistent. The audio playing effect of the system is optimized. The audio signal playing system and the electronic device can enable an audience to acquire a good listening experience and are conducive to the promotion and the application of a product.