53 results about "Audio signal" patented technology

Methods and apparatus for executing a lighting program to control a plurality of light emitting diodes (LEDs) in response to at least one characteristic of an audio input. In one embodiment, the audio input is digitally processed to determine the at least one characteristic. In other embodiments, control signals for the LEDs are generated in response to a timer and/or input from a user interface, as well as in response to the at least one characteristic of the audio input. In another embodiment, the control signals for the LEDs are generated by a same computer that processes the audio input to transmit signals to speakers to audibly play the audio input. In a further embodiment, a GUI is provided to assist in authoring the lighting program. In another embodiment, the audio signal is processed before being played back. In a further embodiment, the lighting program anticipates changes in the audio input.

A portable assistive listening system for enhancing sound for hearing impaired individuals includes a fully functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the present invention is directed to the handheld DSP device. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and/or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different processing algorithms for processing sound signals received from the hearing aid or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sources and selective processing of sound. In the context of being user programmable, the digital signal processing device includes a software platform that provides for the ability of the user to select or “plug-in” desired processing algorithms for application to selected incoming audio channels and a communication port for the user to connect to a PC or other device to download preferred processing algorithms. The communication port provides the user with the ability to retrieve desirable processing algorithms from a database of available algorithms and download those algorithms directly into the device for use.

There are provided methods and an apparatus for processing audio signals. According to one aspect of the present invention there is included a method for processing audio signals having the steps of receiving at least one audio signal having at least a center channel signal, a right side channel signal, and a left side channel signal; processing the right and left side channel signals with a first virtualizer processor, thereby creating a right virtualized channel signal and a left virtualized channel signal; processing the center channel signal with a spatial extensor to produce distinct right and left outputs, thereby expanding the center channel with a pseudo-stereo effect; and summing the right and left outputs with the right and left virtualized channel signals to produce at least one modified side channel output.

A network telephone system is provided with a distributed network, a network call processor, with the call processor connected to the network. A telephone line network interface is connected to a telephone line and is connected to the network for receiving packets from the network and sending packets to the network including packets with telephone voice data. A plurality of network telephones are part of the network telephone system with each network telephone connected to the network. Each network telephone has a display for displaying information and each network telephone is capable of engaging in a concurrent telephonic communication. Each network telephone has an I/O device in electrical communication with the network for receiving and sending packets to other devices connected to the network, an input device for producing audio signals from an input local to the device and a packet controller in electrical communication with the I/O device and the input device. The packet controller generates packets from the audio signals received by the input device, forwards the generated packets to the I/O device for transmission to the network and combines packets received by the I/O device to produce an audio signal with the combined packets and the audio signals from the local input device. Each network telephone displays information on the display corresponding to an identity of a source of packets combined whereby the identity may be selected for dropping a source form a concurrent telephonic communication.

A music application guides a user with some musical experience through the steps of creating and editing a musical enhancement file that enhances and plays in synchronicity with an audio signal of an original artist's recorded performance. This enables others, perhaps with lesser musical ability than the original artist, to play-along with the original artist by following melodic, chordal, rhythmic, and verbal prompts. The music application accounts for differences in the timing of the performance from a standard tempo by guiding the user through the process of creating a tempo map for the performance and by associating the tempo map with MIDI information of the enhancement file. Enhancements may contain MIDI information, audio signal information, and/or video signal information which may be played back in synchronicity with the recorded performance to provide an aural and visual aid to others playing-along who may have less musical experience.

A system and a method for the scalable coding of a multi-channel audio signal comprising a principal component analysis (PCA) transformation of at least two channels (L, R) of the audio signal into a principal component (CP) and at least one residual sub-component (r) by rotation defined by a transformation parameter (θ), comprising the following steps: formation of a frequency subband-based residual structure (Sf_r) on the basis of the at least one residual sub-component (r), and definition of a coded audio signal (SC) comprising the principal component (CP), at least one residual structure (Sf_r) of a frequency subband and the transformation parameter (θ).

To perform volume control matching the actual auditory sensation level of a human so as to improve the control accuracy. As shown in FIG. 1, an audio gain adjustment device 10 includes: an audio signal input section 12 that acquires an audio signal to be adjusted; a long-term gain reflection section 14 that reflects the adjustment of long-term gain in the acquired audio signal; a frequency division section 20 that divides the audio signal from the long-term gain reflection section 14 into three frequency bands; a short-term gain reflection section 30 that reflects the adjustment of short-term gain in each frequency band; a synthesis section 40 that synthesizes audio signals output from the short-term gain reflection section 30; and an audio signal output section 42 that outputs the synthesized audio signal.

This invention relates to retrieval for multimedia content, and provides a program endpoint time detection apparatus for detecting an endpoint time of a program by performing processing on audio signals of said program, comprising an audio classification unit for classifying said audio signals into a speech signal portion and a non-speech signal portion; a keyword retrieval unit for retrieving, as a candidate endpoint keyword, an endpoint keyword indicating start or end of the program from said speech signal portion; a content analysis unit for performing content analysis on context of the candidate endpoint keyword retrieved by the keyword retrieval unit to determine whether the candidate endpoint keyword is a valid endpoint keyword; and a program endpoint time determination unit for performing statistics analysis based on the retrieval result of said keyword retrieval unit and the determination result of said content analysis unit, and determining the endpoint time of the program. In addition, this invention also provides a program information retrieval system. With present invention, program information regarding a program attended by user can be rapidly obtained.

An apparatus for processing an audio signal and method thereof are disclosed. The method comprises receiving a downmix signal, object information indicating attribute of the object and including object number information, preset information to render the downmix signal, external preset information being inputted from external, and applied object number information indicating the number of object being applied the external preset information; determining whether the applied object number information is identical to the object number information; and rendering the downmix signal by using the external preset information, if the applied object number information is identical to the object number information, wherein the external preset rendering parameter renders the object being included in the downmix signal and the external preset metadata indicates attribute of the external preset rendering parameter.

Accordingly, an audio signal can be efficiently reconstructed by individually selecting and applying external preset information by a data region unit or by selecting and applying the same external preset information to a whole downmix signal.

Methods and apparatus are provided for coding and decoding a digital audio signal. Decoding includes: decoding according to an inverse transform decoding of a previous frame of samples of the digital signal, which is received and coded according to a transform coding; and decoding according to a predictive decoding of a current frame of samples of the digital signal, which is received and coded according to a predictive coding. The predictive decoding of the current frame is a transition predictive decoding which does not use any adaptive dictionary arising from the previous frame. At least one state of the predictive decoding is reinitialized to a predetermined default value, and an add-overlap step combines a signal segment synthesized by predictive decoding of the current frame and a signal segment synthesized by inverse transform decoding, corresponding to a stored segment of the decoding of the previous frame.

A device for attenuating sound on the human ear, especially for preventing sleep disturbances caused by noise and other sounds. The device for attenuating sound has good damping properties and also ensures that certain alarm sounds are heard in spite of the sound-attenuating measure. To this end, a radio receiver with means for converting radio signals received from a radio station into audio signals is integrated into at least one of two sound-attenuating ear plugs. The radio station is connected to at least one alarm-sounding means and/or at least one sound-receiving means in order to convert a specific alarm and/or at least one specific sound into a radio signal that can be received by a radio receiver as it occurs.

Methods and apparatus of managing a communication system, wherein a decision regarding a level of activity at a first end is made based at least in part on the level of activity at the second end. In one embodiment, the energy level of a first-end audio signal is measured. The first end is declared voice-active if the first-end energy level is greater than or equal to a first threshold value. The first end is declared voice-inactive if the first-end energy level is less than the first threshold value. To determine the value of the first threshold value, the energy level of a second-end audio signal is measured. If the second-end energy level is greater than or equal to a second threshold value, the second end is declared voice-active, in which case the first threshold is maintained at a relatively high level. If the second-end energy level is less than the second threshold value, the second end is declared voice-inactive, in which case the first threshold is maintained at a relatively lower level.

An information transmission system is broken down into a sound recorder, a transmission medium and a sound reproducer; a digital audio signal is produced from MIDI music data words through any kind of modulation technique such as, for example, Y-modulation technique, P-modulation technique or Q-modulation technique, and the digital audio signal is transferred from the sound recorder through the transmission medium such as a magneto-optic disc, a wire communication network or the free space to the sound reproducer, when the digital audio signal reaches the sound reproducer, the sound reproducer analyzes the digital audio signal to see what modulation technique was employed to produce it, and demodulates the digital audio signal through an appropriate demodulation technique, thereby permitting user to employ any sound recorder to produce the digital audio signal.

The invention discloses a method used for eliminating sound effect switching noise when playing an audio signal, which consists of sampling points. N sampling points constitute a sliding step length, and N is a natural number. When sound effect is switched, smoothing processing is carried out according to the last recorded audio pulse code modulation (PCM) value of audio data blocks before switching and after first sound effect processing and the audio PCM value of second sound effect processing after switching, thus obtaining a new audio PCM value. The invention also discloses an audio playing device. The proposal of the invention can lead the PCM waveform of audio signals before and after sound effect switching to keep smooth transition, thus avoiding sound effect switching noise. The invention needs extremely small calculation amount and storage space and is very applicable to be applied to portable multi-media players.

According to the embodiment, the invention provides a voice signal endpoint detection method, a detection device, detection equipment and a storage medium. The method comprises the following steps: acquiring a first audio signal, wherein the first audio signal comprises a voice signal; detecting a starting point of the voice signal from the first audio signal via either a first voice activity detection (VAD) algorithm based on energy and zero-crossing rate or a second VAD algorithm based on deep learning; and after detecting the starting point of the voice signal, detecting an ending point ofthe voice signal via the other one of the first VAD algorithm and the second VAD algorithm. Therefore, the accuracy of detecting voice signal endpoints is guaranteed, and meanwhile, a computing amountis reduced.

An audio processing apparatus comprising: channel determination section for determining which one of a combination of the first left audio signal and the first right audio signal and a combination of the second left audio signal and the second right audio signal is included in multichannel audio data; and switching section for causing the first amplification section to amplify the first left audio signal and supply the amplified first left audio signal to the first speaker terminal and causing the second amplification section to amplify the first right audio signal and supply the amplified first right audio signal to the second speaker terminal when the combination of the first left audio signal and the first right audio signal is determined to be included; and causing the first amplification section to amplify the second left audio signal and supply the amplified second left audio signal to the third speaker terminal and causing the second amplification section to amplify the second right audio signal and supply the amplified second right audio signal to the fourth speaker terminal when the combination of the second left audio signal and the second right audio signal is determined to be included.

The invention discloses an anti-tonal modification interference audio fingerprint extraction method. The method comprises the following steps of framing and windowing an audio signal, performing Fourier transform, and collecting Fourier coefficients corresponding to all frames of signals; calculating energy segment indexes corresponding to all the frames of signals by using an energy segment indexcalculation method based on a peak point in the collected Fourier coefficients; and calculating energy segment values by using the energy segment indexes and filtering energy segments by a two-dimensional filtering core to extract fingerprints. By means of the method, the defect that an existing Philips fingerprint cannot resist tonal modification interference can be overcome; and on the premiseof keeping the sub-fingerprint continuity of the original Philips fingerprint and the robustness of various noise interferences, the resistance to the tonal modification interference is improved.

A method is described that estimates noise in an audio signal. An energy value for the audio signal is estimated and converted into the logarithmic domain. A noise level for the audio signal is estimated based on the converted energy value.

[Problem] To provide a simple and economical device capable of providing an excellent sound field in accordance with the listener's preferences by adjusting the listener's interaural correlation coefficient. [Solution] A sound field adjustment filter (5) is provided with a pair of filter circuit groups (5L, 5R) corresponding to a left and right channel. This sound field adjustment filter divides a desired frequency band of the audio signal into multiple frequency bands and assigns prescribed transmission characteristics so as to generate a sound field adjustment signal for each frequency band such that a prescribed impulse response is configured cooperatively for the entire desired frequency band of the audio signal. The transmission characteristics adjust the audio signal for each split frequency band by the addition of time delay and gain. By means of operation output units (4L, 4R), the sound field adjustment filter (5) adds the sound field adjustment signals generated by each of the filter circuit groups (5L, 5R) to the audio signal of the other of the channels, and outputs the result to stereo speakers (2). The opening angle of virtual speakers with respect to the listener is changed by adjustment of the sound field adjustment signals so as to approximate the interaural correlation coefficient of a specific diffuse sound field.

The embodiment of the invention discloses a method and device for detecting the wearing state of an earphone, electronic equipment and a computer storage medium. The method comprises of acquiring steps that environment noise signals around the earphone; controlling a first sounding unit of the earphone to output a first audio signal; and determining the wearing state of the earphone according tothe first audio signal and the environment noise signal. Thus, under the condition of considering the environment noise signal, the interference of the environment noise on the earphone wearing statedetection is reduced, and the accuracy of detecting the earphone wearing state in a noisy environment can be improved.

A signal processing device comprises: a band detecting means for detecting a frequency band which satisfies a predetermined condition from an audio signal; a reference signal generating means for generating a reference signal in accordance with a detection band by the band detecting means; a reference signal correcting means for correcting the generated reference signal on the basis of a frequency characteristic thereof; a frequency band extending means for extending the corrected reference signal up to a frequency band higher than the detection band; an interpolation signal generating means for generating an interpolation signal by weighting each frequency component within the extended frequency band in accordance with a frequency characteristic of the audio signal; and a signal synthesizing means for synthesizing the generated interpolation signal with the audio signal.