Entries |
Document | Title | Date |
20080208575 | Split-band encoding and decoding of an audio signal - For enabling an improved reconstruction of a high frequency band of an audio signal in a split-band coding approach, a value representative of a background noise level in an audio signal that is to be encoded is determined. Further, a gain value for the higher frequency band is determined. Further, a correction factor for the determined gain value is determined based on the determined value representative of the background noise level. The correction factor may be used at an encoding end for correcting the gain value before a corresponding codebook index is provided to a decoding end. Alternatively, the correction factor may be provided together with a codebook index for the gain value to a decoding end, and the decoding end may use the correction factor to correct the gain value if appropriate. | 08-28-2008 |
20080228473 | METHOD AND APPARATUS FOR ADJUSTING HEARING INTELLIGIBILITY IN MOBILE PHONES - The present invention relates to a method and apparatus for speech intelligibility enhancement. First, the noise estimator determines the amount of ambient noise when the listener is placed in a high ambient noise environment. Second, the perceptual feature associated with speech intelligibility is enhanced with minimal processing artifacts. Third, the listener's volume gain is automatically and adaptively adjusted based on the psycho-acoustic model. The method and apparatus has ample capability to enhance speech intelligibility in any noisy environment conditions, and hence it can be a more desired solution than manual volume adjustment. Thus, the present invention can be implemented in mobile handsets, telephones, public address systems, and the like. | 09-18-2008 |
20080235011 | Automatic Level Control Of Speech Signals - Automatic level control of speech portions of an audio signal is provided. An audio signal is received in the form of a sequence of samples and may contain speech portion and non-speech portions. The sequence of samples is divided into a sequence of sub-frames. Multiple sub-frames adjacent to a present sub-frame are examined to determine a peak value of samples in the sub-frames. A gain factor is computed for the present sub-frame based on the peak value and a desired maximum value for said speech portion, and each sample in the present sub-frame is amplified by the gain factor. In an embodiment, variations in filtered energy values of multiple sub-frames enable determination of whether a sub-frame corresponds to a speech or non-speech/noise portion. | 09-25-2008 |
20080319740 | Adaptive gain reduction for encoding a speech signal - There is provided a method of encoding an input speech signal. The method comprises identifying a fixed codebook vector from a fixed codebook; identifying an adaptive codebook vector from a adaptive codebook; calculating an adaptive codebook gain; reducing the adaptive codebook gain by an amount; optimally selecting a fixed codebook gain based on the adaptive codebook gain while both the fixed codebook vector and the adaptive codebook vector remain fixed; and converting the input speech signal into an encoded speech using the fixed codebook gain, the adaptive codebook gain, the fixed codebook vector and the adaptive codebook vector. The amount of reducing the adaptive codebook gain may be varied. | 12-25-2008 |
20090006086 | Signal Decoding Apparatus - A signal decoding apparatus that can suppress any large unusual sounds to provide decoded signals of improved audibility even when the number of hierarchical layers to be used in the decoding process varies due to a packet loss or the like in communication utilizing a scalable encoding/decoding technique. In the signal decoding apparatus, a gain adjusting part ( | 01-01-2009 |
20090070105 | VOICE COMMUNICATION APPARATUS - A voice communication apparatus includes a communication portion that receives a plurality of frames including at least a first frame having first voice data and a second frame having second voice data subsequent to the first frame, the first voice data and the second voice data being encoded by a predetermined encoding system, a decoding portion that decodes the first voice data and the second voice data received by the communication portion, a buffer that retains the first voice data and the second voice data decoded by the decoding portion, a calculation portion that calculates an amplitude envelope based on the first voice data decoded by the decoding portion, and a controlling portion that judges whether or not the second voice data decoded by the decoding portion exceeds the amplitude envelope and corrects the second voice data that exceeds the amplitude envelope. | 03-12-2009 |
20090070106 | Method and system for reducing effects of noise producing artifacts in a speech signal - There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1−k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain. | 03-12-2009 |
20090076810 | Sound processing apparatus, apparatus and method for cotrolling gain, and computer program - A sound processing apparatus is provided for estimating the power of background noise using a directional sound receiving technology using a plurality of sound receiving units, computing a gain control value on the basis of the estimated power of background noise and a predetermined power target value, and outputting the gain control value, so that a delay time of starting gain control can be reduced, and a slow response of a speech recognition application program or degradation of the speech quality of a voice communication program can be prevented. | 03-19-2009 |
20090089051 | Vocal fry detecting apparatus - A VF detecting apparatus capable of highly accurate vocal fry (VF) detection includes: a very-short-term peak detection processing unit framing a speech signal with a first frame of a first frame length and first frame shift amount and detecting each power peak; a short-term periodicity detecting unit framing the speech signal with a second frame of a second frame length longer than the first frame length and a second frame shift amount larger than the first frame length and determining presence/absence of periodicity in each of the resulting frame; a periodicity checking unit for detecting power peaks in those frames determined to have no periodicity, from among the detected power peaks; and a similarity checking unit for detecting, for each of the selected power peaks, neighboring power peaks having high cross-correlation and detecting the section therebetween as the VF section. | 04-02-2009 |
20090112582 | ON-VEHICLE DEVICE, VOICE INFORMATION PROVIDING SYSTEM, AND SPEECH RATE ADJUSTING METHOD - In an on-vehicle device ( | 04-30-2009 |
20090177466 | DETECTION OF SPEECH SPECTRAL PEAKS AND SPEECH RECOGNITION METHOD AND SYSTEM - The present invention provides a method and apparatus for detecting speech spectral peaks and a speech recognition method and system. The method for detecting speech spectral peaks comprises detecting speech spectral peak candidates from power spectrum of the speech, and removing noise peaks from the speech spectral peak candidates according to peak duration and/or peak positions of adjacent frames, to detect speech spectral peaks. In the present invention, reliable speech spectral peaks can be obtained by removing noise peaks using the limitations of peak duration and adjacent frames in the detection of the speech spectral peaks. Further the energy values of the speech spectral peaks are used to extract the MFCC feature of speech instead of a sample sequence of the whole power spectrum in the conventional technique, the noise robustness of speech recognition can be enhanced while not increasing the speech feature dimensions. | 07-09-2009 |
20090192793 | METHOD FOR INSTANTANEOUS PEAK LEVEL MANAGEMENT AND SPEECH CLARITY ENHANCEMENT - A method for raising the soft and mid-level amplitude of sounds for greater clarity and perceptual benefit, while simultaneously removing the high level amplitude peaks without delay and providing protection for the auditory sense organ. The method does not require a feedback mechanism for the accomplishment of this treatment and exploits the psychoacoustic phenomenon of temporal integration which reduces the audibility of short duration signals, including distortions associated with peak clipping. The human auditory system requires greater time to integrate signal energy for audibility than provided by brief duration waveform peaks. | 07-30-2009 |
20090259461 | Gain Control System, Gain Control Method, and Gain Control Program - Disclosed is a gain control system in which speech model constituted from a sound pressure and a feature is stored in a speech model storage unit for each of a plurality of phonemes or for each of clusters into which a speech is divided. When an input signal is given, a feature conversion unit calculates a feature and a sound pressure of the input signal. A sound pressure comparison unit determines a sound pressure ratio between the input signal and each of speech models. A distance calculation unit calculates a distance between the feature of the input signal and the feature of each of the speech models. A gain calculation unit calculates a gain value from the sound pressure ratio and information on the distance. A sound pressure compensation unit thereby compensates for the sound pressure of the input signal. | 10-15-2009 |
20090271185 | AUDIO-PEAK LIMITING IN SLOW AND FAST STAGES - A method and apparatus for limiting the absolute magnitude of an audio signal. The method may include firstly variable-gain reducing the gain of an audio signal, and then secondly variable-gain reducing the gain of the audio signal faster than the first variable-gain reduction, thereby limiting the absolute magnitude of the audio signal to a threshold. The first variable-gain reduction may include variable-gain reducing the gain of the audio signal in a first stage, and the second variable-gain reduction may include variable-gain reducing the gain of the audio signal in a second stage that reduces the gain faster than the first stage. The second variable-gain reduction may include delaying the audio signal, finding a peak among the delayed audio signal, calculating a fast gain from a found peak, and modifying the delayed audio signal with the calculated fast gain. | 10-29-2009 |
20090281801 | COMPRESSION FOR SPEECH INTELLIGIBILITY ENHANCEMENT - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal. | 11-12-2009 |
20090287480 | Method and apparatus for low bit rate speech coding detection - To increase channel capacity, mobile phone carriers have deployed speech coders, such as Advanced MultiBand Excitation coding (AMBE), in networks to reduce the bit rate of each call. One undesired consequence of employing such speech coders is that the voice quality can be much worse as compared to higher bit-rate speech coders. A method or corresponding apparatus in an example embodiment of the present invention performs voice quality enhancement transparently within a network by detecting use of a coder applying rate reduction to a speech signal and known to have an adverse effect on a coded speech signal. Upon detection of the use of such coder, the coded speech signal is corrected based on components introduced into the coded speech signal due to the rate reduction. As a result of applying the voice quality enhancement, adverse effects of speech coders can be reduced, while maintaining high quality voice signals. | 11-19-2009 |
20090292535 | SYSTEM AND METHOD FOR SYNTHESIZING MUSIC AND VOICE, AND SERVICE SYSTEM AND METHOD THEREOF - The present invention relates to a system and a method for synthesizing music and voice, and a service system and a service method using the same. The system and method according to the present invention is capable of making a listener feel maximum synthesizing effects to mix the voice and the music. Also, the system and method according to the present invention is capable of synthesizing the voice and music with various effects without the professional synthesizer's volume control. | 11-26-2009 |
20090292536 | Speech enhancement with minimum gating - A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec. | 11-26-2009 |
20090299739 | SYSTEMS, METHODS, AND APPARATUS FOR MULTICHANNEL SIGNAL BALANCING - A method for processing a multichannel audio signal may be configured to control the amplitude of one channel of the signal relative to another based on the levels of the two channels. One such example uses a bias factor, which is based on a standard orientation of an audio sensing device relative to a directional acoustic information source, for amplitude control of information segments of the signal. | 12-03-2009 |
20100017205 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER PROGRAM PRODUCTS FOR ENHANCED INTELLIGIBILITY - Techniques described herein include the use of equalization techniques to improve intelligibility of a reproduced audio signal (e.g., a far-end speech signal). | 01-21-2010 |
20100023327 | METHOD FOR IMPROVING SPEECH SIGNAL NON-LINEAR OVERWEIGHTING GAIN IN WAVELET PACKET TRANSFORM DOMAIN - The present invention relates to speech enhancement accomplished by applying an overweighting gain of a nonlinear structure in a wavelet packet transform domain or a Fourier transform domain. The present invention relates to a method for improving quality of speech signals, which can be applied in a variety of noise-level conditions using noise estimation of the least-square line method and a modified spectral subtraction method having a nonlinear overweighting gain for each sub-band. According to the method for improving quality of speech of the present invention, it is effective in that quality of speech can be further effectively improved in a variety of noise-level conditions. Particularly, according to the present invention, generation of musical tones can be efficiently suppressed, and intelligibility of speech is reliably guaranteed in the improved speech. | 01-28-2010 |
20100057449 | APPARATUS AND METHOD OF ENHANCING QUALITY OF SPEECH CODEC - An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a core codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced. | 03-04-2010 |
20100106495 | VOICE RECOGNITION SYSTEM, METHOD, AND PROGRAM - A voice recognition system comprises: a voice input unit that receives an input signal from a voice input element and output it; a voice detection unit that detects an utterance segment in the input signal; a voice recognition unit that performs voice recognition for the utterance segment; and a control unit that outputs a control signal to at least one of the voice input unit and the voice detection unit and suppresses a detection frequency if the detection frequency satisfies a predetermined condition. | 04-29-2010 |
20100114569 | DYNAMIC RANGE CONTROL MODULE, SPEECH PROCESSING APPARATUS, AND METHOD FOR AMPLITUDE ADJUSTMENT FOR A SPEECH SIGNAL - The invention provides a dynamic range control module installed in a speech processing apparatus. In one embodiment, the dynamic range control module comprises a buffer, a voice activity detector, a peak calculation module, and an amplitude adjusting module. The buffer buffers a speech signal to obtain a delayed speech signal. The voice activity detector determines a syllable from the delayed speech signal. The peak calculation module calculates peak amplitude of the syllable. The amplitude adjusting module determines an attenuation factor corresponding to the syllable according to the peak amplitude in the syllable, and adjusts amplitude of the whole syllable with the same gain according to the attenuation factor to obtain an adjusted speech signal. | 05-06-2010 |
20100121635 | Enhancing the Intelligibility of Received Speech in a Noisy Environment - A long term signal level of an audio signal is computed at a local device, wherein the audio signal was transmitted from a remote device and received by the local device. An automatic gain control (AGC) gain is computed at the local device based on the long term signal level. A noise factor indicative of a level of ambient noise at the local device is computed at the local device. A dynamic range compression (DRC) gain is computed at the local device based on the noise factor. An amplitude of the audio signal is adjusted at the local device based on the AGC gain and the DRC gain. | 05-13-2010 |
20100169087 | SELECTIVE SCALING MASK COMPUTATION BASED ON PEAK DETECTION - A set of peaks in a reconstructed audio vector Ŝ of a received audio signal is detected and a scaling mask ψ(Ŝ) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output. | 07-01-2010 |
20100179808 | Speech Enhancement - A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal. | 07-15-2010 |
20100280825 | Voice Input Device, Method of Producing the Same, and Information Processing System - A voice input device includes a first microphone ( | 11-04-2010 |
20100332223 | AUDIO DECODING DEVICE AND POWER ADJUSTING METHOD - Provided is an audio decoding device capable of obtaining a preferable synthesized sound with a stable sound volume. The audio decoding device includes: a post filter ( | 12-30-2010 |
20110035214 | ENCODING DEVICE AND ENCODING METHOD - Good sound quality as perceived by the ear is obtained even with few information bits. A shape quantizer ( | 02-10-2011 |
20110054887 | Method and Apparatus for Maintaining Speech Audibility in Multi-Channel Audio with Minimal Impact on Surround Experience - In one embodiment the present invention includes a method of improving audibility of speech in a multi-channel audio signal. The method includes comparing a first characteristic and a second characteristic of the multi-channel audio signal to generate an attenuation factor. The first characteristic corresponds to a first channel of the multi-channel audio signal that contains speech and non-speech audio, and the second characteristic corresponds to a second channel of the multi-channel audio signal that contains predominantly non-speech audio. The method further includes adjusting the attenuation factor according to a speech likelihood value to generate an adjusted attenuation factor. The method further includes attenuating the second channel using the adjusted attenuation factor. | 03-03-2011 |
20110066428 | SYSTEM FOR ADAPTIVE VOICE INTELLIGIBILITY PROCESSING - An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content. | 03-17-2011 |
20110071822 | Selective audio/sound aspects - Certain aspects relate to providing an at least one audio source to at least one user. Certain aspects relate to selectively modifying an at least one first sound source to be provided to the at least one user, wherein the at least one first sound source is combined with an at least one second sound source, and wherein the selectively modifying is performed relative to the at least one audio source based at least in part on at least some specific information of the at least one first sound source. Other aspects relate to selectively modifying the at least one first sound source to be provided to the at least one user relative to the at least one second sound source based at least in part on at least some specific information of the at least one first sound source. | 03-24-2011 |
20110082692 | METHOD AND APPARATUS FOR REMOVING SIGNAL NOISE - A method and apparatus for removing signal noise using multiple bands are provided. The noise removal apparatus may divide the entire frequency band into a plurality of sub-bands using a multiband filter that has characteristics similar to an auditory system of a human being and may effectively remove noise in each of the sub-bands according to a frequency subtraction scheme. | 04-07-2011 |
20110082693 | SYSTEMS, METHODS, AND APPARATUS FOR FRAME ERASURE RECOVERY - In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values. | 04-07-2011 |
20110172996 | VOICE INPUT DEVICE, METHOD FOR MANUFACTURING THE SAME, AND INFORMATION PROCESSING SYSTEM - A voice input device, a method for manufacturing the same, and an information processing system are provided. The voice input device has a function of removing a noise component and includes a first microphone | 07-14-2011 |
20110191103 | PORTABLE TERMINAL TO ADJUST SOUND OUTPUT OF WIRELESS HEADSET - A portable terminal and method for adjusting transmitted sound output to a wireless headset. A portable terminal includes a wireless connection setting unit to establish a connection with a wireless headset and a loopback control unit. The loopback control unit may transmit a test signal within a voice band to the connected wireless headset, receive a sound signal inputted through a microphone of the portable terminal in response to the transmitted test signal, analyze the received sound signal and adjust sound output to be transmitted to the wireless headset. A method for adjusting sound output includes establishing a connection with a wireless headset, transmitting a test signal within a voice band, receiving a sound signal inputted through a microphone of the portable terminal in response to the transmitted test signal, analyzing the sound signal, and adjusting sound output to be transmitted to the wireless headset according to the analysis result. | 08-04-2011 |
20110301948 | ECHO-RELATED DECISIONS ON AUTOMATIC GAIN CONTROL OF UPLINK SPEECH SIGNAL IN A COMMUNICATIONS DEVICE - A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed. | 12-08-2011 |
20120065967 | COMMUNICATION DEVICE AND SIGNAL PROCESSING METHOD - Provided is a communication device which can easily provide a function to enable signal cross-reference among a plurality of voice signals having different frequency ranges and to enhance the quality of voice communications, at a low cost. In the communication device, a band expansion unit ( | 03-15-2012 |
20120078619 | CONTROL APPARATUS AND CONTROL METHOD - An apparatus may include a control unit to selectively control volume of content sound and volume of speech sound according to a priority assigned to a user corresponding to speech sound and a priority assigned to content data. When volume control is to be performed on a priority basis, the control unit may selectively control the volume of the content sound and the volume of the speech sound based on the assigned priorities so that the volume of the sound having a higher priority becomes louder than the volume of the other sound. | 03-29-2012 |
20120095759 | SYSTEM FOR IMPROVING SPEECH INTELLIGIBILITY THROUGH HIGH FREQUENCY COMPRESSION - A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range. | 04-19-2012 |
20120101816 | ENHANCING THE INTELLIGIBILITY OF RECEIVED SPEECH IN A NOISY ENVIRONMENT - A level of ambient noise at a local device is determined. A dynamic range compression (DRC) gain is computed based on the level of ambient noise at the local device. An additional gain factor is computed. A total gain is computed based on an adding of the DRC gain and the additional gain factor. An amplitude of an audio signal is adjusted based on the total gain, wherein the audio signal was transmitted from a remote device and received by the local device. | 04-26-2012 |
20120123769 | GAIN CONTROL APPARATUS AND GAIN CONTROL METHOD, AND VOICE OUTPUT APPARATUS - Provided is a technology which adjusts an input signal such that the volume of a conversation or speech contained in a content is substantially constant, thereby alleviating the audience from a burden of making a volume control operation. An acoustic signal processor comprises an acoustic signal storage unit which buffers an acoustic input signal for a predetermined period of time; a voice detection unit which detects a voice section from the buffered acoustic signal; an acoustic signal-to-loudness level transformation which calculates a loudness level from the buffered acoustic signal; a threshold/level comparator which compares the calculated loudness level with a predetermined target level; a voice amplification calculation unit which calculates a gain control amount for the buffered acoustic signal on the basis of the detection and comparison results; and an acoustic signal amplifier which amplifies or dampens the buffered acoustic signal in accordance with the calculated gain control amount. | 05-17-2012 |
20120143603 | SPEECH PROCESSING APPARATUS AND METHOD - A speech processing apparatus and method. The speech processing apparatus includes a microphone to receive a speech signal, an analog/digital converter to convert the speech signal generated by the microphone into a digital speech signal, and an automatic gain controller to calculate an average value of the magnitude of the digital speech signal generated by the analog/digital converter in a plurality of frames, to determine in which region of a speech signal band the average value is located, the speech signal band being divided into a plurality of regions according to the strength of speech, and to adjust gain according to a location of the average value on the speech signal band so that the strength of speech has a level of an optimal region capable of processing the speech signal. Accordingly, speech recognition may be maximized without being constrained by the distance of a speech source. | 06-07-2012 |
20120173231 | SYSTEM FOR COMFORT NOISE INJECTION - A noise injection system adds comfort noise to an audio signal. The system includes a background noise estimator that determines a spectral content of a background noise associated with the audio signal. A comfort noise generator generates a comfort noise signal having a random phase. A gain circuit adjusts the comfort noise signal based on the spectral content of the background noise. A combining circuit combines a gain-adjusted comfort noise signal and the audio signal to generate an output signal. | 07-05-2012 |
20120185245 | SOUND OUTPUT SETTING APPARATUS, METHOD, AND COMPUTER PROGRAM PRODUCT - An apparatus is provided with a device storing machine readable code and a processor executing the machine readable code. The machine readable code includes sound setting code and audio processing code. The sound setting code detects use of a microphone and sets sound characteristics that are suitable for conversation in response to detecting the use of the microphone. The audio processing code processes sound on the basis of the sound characteristics set by the sound setting code. | 07-19-2012 |
20120215530 | METHOD AND SYSTEM FOR SPEECH ENHANCEMENT IN A ROOM - A method of speech enhancement in a room ( | 08-23-2012 |
20120221328 | Enhancement of Multichannel Audio - The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods. | 08-30-2012 |
20120221329 | SPEECH ENHANCEMENT METHOD AND SYSTEM - A method of speech enhancement in a room ( | 08-30-2012 |
20120239391 | AUTOMATIC EQUALIZATION OF COLORATION IN SPEECH RECORDINGS - Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal. | 09-20-2012 |
20120259625 | SYSTEM FOR PROCESSING AN AUDIO SIGNAL TO ENHANCE SPEECH INTELLIGIBILITY - An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content. | 10-11-2012 |
20120271630 | SPEECH SIGNAL PROCESSING SYSTEM, SPEECH SIGNAL PROCESSING METHOD AND SPEECH SIGNAL PROCESSING METHOD PROGRAM - A speech signal processing system that includes a speech input unit for inputting a speech signal; input speech storage unit for storing an input speech signal that is the speech signal inputted through the speech input unit; characteristic estimation unit for referring to the input speech signal stored in the input speech storage unit, and estimating characteristics of an input speech indicated by the input speech signal, the characteristics including an environmental sound included in the input speech signal; reference speech output unit for causing a predetermined speech signal that becomes a reference speech, to output; and characteristic adding unit for adding the characteristics of the input speech estimated by the characteristic estimation unit, in a reference speech signal that is the speech signal caused to output by the reference speech output unit. | 10-25-2012 |
20120284021 | CONCEALING AUDIO INTERRUPTIONS - A method of processing an audio signal in a communications network, the method comprising: receiving, at a speech buffer, a first portion of the audio signal over the network from a base station of the network, the speech buffer being configured to store and subsequently output the first portion of the audio signal; determining the presence of an interruption to the received audio signal, the interruption being such that a subsequent portion of the audio signal which is intended to be output from the speech buffer immediately following the output of the first portion is not stored in the speech buffer at the time that the subsequent portion is intended to be output from the speech buffer; in the event that the presence of the interruption has been determined, appending a second portion of the audio signal to the first portion in such a way as to form an output audio signal having no signal discontinuities in the time domain, the second portion having a predetermined duration and having a pitch matching that of the first portion over the predetermined duration; applying a fade out envelope to the second portion to gradually reduce the amplitude of the second portion over the predetermined duration; and outputting the output audio signal. | 11-08-2012 |
20120303363 | Processing Audio Signals - A method, user device and computer program product for processing audio signals during a communication session between a user device and a remote node. The method comprising: receiving a plurality of audio signals at audio input means at the user device including at least one primary audio signal and unwanted signals; receiving direction of arrival information of the audio signals at a gain control means; providing to the gain control means known direction of arrival information representative of at least some of said unwanted signals; processing the audio signals at the gain control means by applying a level of gain to generate a gain controlled signal for transmission to the remote node, wherein the level of gain applied is dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information. | 11-29-2012 |
20120310635 | Enhancement of Multichannel Audio - The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods. | 12-06-2012 |
20120323571 | METHOD AND APPARATUS FOR INCREASING SPEECH INTELLIGIBILITY IN NOISY ENVIRONMENTS | 12-20-2012 |
20120330651 | VOICE DATA TRANSFERRING DEVICE, TERMINAL DEVICE, VOICE DATA TRANSFERRING METHOD, AND VOICE RECOGNITION SYSTEM - A voice data transferring device intermediates between an in-vehicle terminal and a voice recognition server. In order to check a change in voice recognition performance of the voice recognition server, the voice data transferring device performs a noise suppression processing on a voice data for evaluation in a noise suppression module; transmits the voice data for evaluation to the voice recognition server; and receives a recognition result thereof. The voice data transferring device sets a value of a noise suppression parameter used for a noise suppression processing or a value of a result integration parameter used for a processing of integrating a plurality of recognition results acquired from the voice recognition server, at an optimum value, based on the recognition result of the voice recognition server. This makes it possible to set a suitable parameter even if the voice recognition performance of the voice recognition server changes. | 12-27-2012 |
20130006619 | Method And System For Scaling Ducking Of Speech-Relevant Channels In Multi-Channel Audio - A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention. | 01-03-2013 |
20130013302 | AUDIO INPUT DEVICE - An audio input device is provided which can include a number of features. In some embodiments, the audio input device includes a housing, a microphone carried by the housing, and a processor carried by the housing and configured to modify an input sound signal so as to amplify frequencies corresponding to a target human voice and diminish frequencies not corresponding to the target human voice. In another embodiment, an audio input device is configured to treat an auditory gap condition of a user by extending gaps in continuous speech and outputting the modified speech to the user. In another embodiment, the audio input device is configured to treat a dichotic hearing condition of a user. Methods of use are also described. | 01-10-2013 |
20130024193 | APPARATUS AND METHOD FOR AUTOMATIC GAIN CONTROL - A speech signal is received at an input. At least one electrical value associated with the received speech signal is tracked. A dynamic adjustment of the speech signal is determined. The dynamic adjustment is selected at least in part so as to minimize a distortion and minimize an over-amplification of the speech signal based at least in part upon an analysis of the at least one electrical value. The dynamic adjustment is further selected to obtain a desired output signal characteristic for the speech signal presented at an output. The dynamic adjustment value is applied to the speech signal and the adjusted speech signal is presented at the output. The gain of the signal can also be limited to prevent over-amplification. | 01-24-2013 |
20130066627 | APPARATUS AND METHOD OF ENHANCING QUALITY OF SPEECH CODEC - An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced. | 03-14-2013 |
20130073282 | APPARATUS AND METHOD OF ENHANCING QUALITY OF SPEECH CODEC - An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced. | 03-21-2013 |
20130080157 | CODING APPARATUS AND METHOD USING RESIDUAL BITS - Disclosed is a coding apparatus and method using residual bits. Accordingly, performance (voice quality) is enhanced by quantizing a full-band gain of frequency coefficients existing in sub-bands to which bits are not assigned in an algebraic vector quantization (AVQ). Further, the performance (voice quality) is enhanced by sequentially quantizing a sub-band gain of sub-bands to which bits are not assigned until residual bits are removed. Furthermore, the performance (voice quality) is enhanced by demodulating AVQ coefficients, and correcting quantization noises starting with a coefficient having the greatest absolute coefficient among the AVQ coefficients, when residual bits additionally remain. | 03-28-2013 |
20130085752 | DECODER, ENCODER, AND METHODS THEREOF - Disclosed is a decoder capable of improving the sound quality of a decoded sound signal in an encoding method which combines speech encoding and music encoding in a hierarchical structure. A transform-encoding decoding unit ( | 04-04-2013 |
20130090922 | VOICE QUALITY OPTIMIZATION SYSTEM AND METHOD - The voice quality optimization system includes a controller that controls voice quality by adjusting parameters that control voice quality characteristics of the communication device; and a measuring unit that measures voice quality of the communication device and transmits the measured voice quality as a feedback to the controller. The controller controls voice quality by calibrating the parameters of the communication device, including a receiving sensitivity/frequency response characteristic curve, receiving loudness rating and idle channel noise-receiving. A method for setting voice optimization in a communication device includes measuring parameters of the communication device, determining whether the parameters of the communication device are within a target range, and calibrating a first parameter to be within the target range if the first parameter is outside the target range. | 04-11-2013 |
20130103396 | POST-FILTER COMMON-GAIN DETERMINATION - An apparatus for processing an input sound signal, the apparatus including: gain circuitry configured to control a gain based on a plurality of respective sub-signals of the input sound signal; and an amplification apparatus configured to adjust the amplification of all the plurality of amplitudes based on the gain. | 04-25-2013 |
20130117016 | METHOD AND AN APPARATUS FOR GENERATING A NOISE REDUCED AUDIO SIGNAL - A method and apparatus are provided for generating a noise reduced output signal from sound received by a first microphone. The method includes transforming the sound received by the first microphone into a first input signal and transforming sound received by a second microphone into a second input signal. The method includes calculating, for each of a plurality of frequency components, an energy transfer function value as a real-valued quotient by dividing a temporally averaged product of an amplitude of the first input signal and the second input signal by a temporally averaged absolute square of the second input signal, calculating a gain value as a function of the calculated energy transfer function value, and generating the noise reduced output signal based on the product of the first input signal and the calculated gain value at each of the plurality of frequency components. | 05-09-2013 |
20130124201 | DECODING DEVICE, ENCODING DEVICE, AND METHODS FOR SAME - Disclosed is a decoding device which can efficiently encode/decode spectral data in a high pass section of a broadband signal. In the disclosed device: a sample group extraction unit ( | 05-16-2013 |
20130144615 | METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL BASED ON AN ESTIMATED LOUDNESS - An apparatus comprising at least one processor and at least one memory including computer program code. The at least one memory and the computer program code is configured to, with the at least one processor, cause the apparatus at least to determine a loudness estimate of a first audio signal, generate a parameter dependent on the loudness estimate; and control the first audio signal dependent on the parameter. | 06-06-2013 |
20130173262 | VOICE CLARIFICATION APPARATUS - The voice clarification apparatus includes a plurality of band-pass filters that respectively extract a plurality of band components, which are included in a voice band, from an input audio signal; a gain determination unit that determines a gain according to the level of a signal of a band component which is extracted by at least one band-pass filter of the plurality of band-pass filters; a level adjustment unit that adjusts the levels of signals of the plurality of band components which are extracted by the plurality of band-pass filters using the gain; and a first addition unit that adds a signal which is based on the audio signal to a signal in which the gain is adjusted by the level adjustment unit, and outputs a signal obtained through the addition. | 07-04-2013 |
20130253923 | MULTICHANNEL ENHANCEMENT SYSTEM FOR PRESERVING SPATIAL CUES - A method is disclosed for maintaining spatial queues in digital sound signals. Sound signals are received from each of a plurality of transducers. The sound signals are transformed using a common real-valued spectral gain, G, to maintain spatial cues within the sound signals, the common spectral gain, G, determined by: calculating G as a function of a derivative of a known cost function and as a function of at least one multichannel frequency-domain Bayesian short-time estimator. | 09-26-2013 |
20130311174 | AUDIO CONTROL DEVICE AND IMAGING DEVICE - An audio control device includes: a situation determination unit that determines a situation of environment when a sound is obtained; and a control unit that applies any one of automatic level control and a gain fixation control to a sound obtained by a sound obtaining unit, the control unit controlling gain automatically based on a result of determination by the situation determination unit such that a sound level is at a constant level in the automatic level control and fixing gain to a predetermined value in the gain fixation control. | 11-21-2013 |
20130332154 | ENCODING APPARATUS, DECODING APPARATUS, ENCODING METHOD AND DECODING METHOD - An encoding apparatus includes a first layer encoder that encodes an input signal, a first layer decoder that decodes the first layer encoded data, a weighting filter that filters a first layer error signal to acquire a weighted first layer error signal, a first layer error transform coefficient calculator that transforms the weighted first layer error signal into a frequency domain, and a second layer encoder that encodes the first layer error transform coefficient. The second layer encoder includes a first shape vector encoder that refers the first layer error transform coefficient to generate a first shape vector and first shape encoded information. A target gain calculator calculates a target gain using the first layer error transform coefficient and the first shape vector, a gain vector generator generates a gain vector, and a gain vector encoder encodes the gain vector to acquire gain encoded information. | 12-12-2013 |
20140025375 | Adaptive Gain-Shape Rate Sharing - An object of embodiments of the present invention is to provide an improved gain-shape VQ. This is achieved by determining a number of bits to be allocated to a gain adjustment- and shape-quantizer for a plurality of combinations of a current bit rate and a first signal property. The determined allocated number of bits to the gain adjustment- and shape quantizer should provide a better result for the given bitrate and signal property than using a single fixed allocation scheme. That can be achieved by deriving the bit allocation by using an average of optimal bit allocations for a training data set. Thus by pre-calculating a number of bits to the gain adjustment and the shape quantizers for a plurality of combinations of the bit rate and a first signal property and creating a table indicating the number of bits to be allocated to the gain adjustment- and the shape-quantizers for a plurality of combinations of the bit rate and a first signal property. In this way, the table can be used for achieving an improved bit allocation. | 01-23-2014 |
20140074462 | METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS - The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank. | 03-13-2014 |
20140074463 | SYSTEMS AND METHODS FOR IMPROVING REPRESENTATION BY AN AUDITORY PROSTHESIS SYSTEM OF AUDIO SIGNALS HAVING INTERMEDIATE SOUND LEVELS - An exemplary system includes a detection facility configured to detect an input sound level of an audio signal presented to an auditory prosthesis patient; and an adaptive gain control (AGC) facility configured to 1) determine whether the detected input sound level is in a quiet region, an intermediate region, or a loud region, and 2) apply a gain to the audio signal in accordance with an AGC gain function that specifies the gain to be substantially equal to or less than a first gain threshold if the detected input sound level is in the quiet region, substantially equal to or less than a second gain threshold if the detected input sound level is in the loud region, and greater than the first and second gain thresholds if the detected input sound level is in the intermediate region. Corresponding systems and methods are also disclosed. | 03-13-2014 |
20140081630 | METHOD AND APPARATUS FOR CONTROLLING VOLUME OF VOICE SIGNAL - A method for amplifying a voice signal in an electronic device. In the method, a voice signal received via a microphone is detected. When a volume of the detected voice signal is less than a predetermined average volume, the volume of the detected voice signal is amplified. The volume-amplified voice signal is transmitted. | 03-20-2014 |
20140088959 | BAND EXTENSION APPARATUS AND BAND EXTENSION METHOD - A band extension apparatus is provided. The band extension apparatus extends a narrow-band speech signal whose frequency band has been restricted to an arbitrary input band, such that the extension band includes signal components in an arbitrary extension band. The arbitrary extension band is a frequency band outside the input band. | 03-27-2014 |
20140095155 | METHOD AND APPARATUS FOR CONTROLLING SPEECH QUALITY AND LOUDNESS - A method and an apparatus for controlling speech quality and loudness are disclosed in the present invention, which belong to the field of communications technologies. The method includes: when a terminal starts hands-free calling, obtaining information of a scenario where the terminal is located; obtaining a speech quality preset value and a loudness gain value of the terminal based on the scenario information; and adjusting speech quality and loudness of the terminal respectively based on the obtained speech quality preset value and the loudness gain value. In the present invention, through obtaining a speech quality preset value and a loudness gain value based on scenario information to control speech quality and loudness of a terminal, a user can enjoy better speech quality, thereby improving user experience of hands-free calling. | 04-03-2014 |
20140114654 | METHOD AND SYSTEM FOR PEAK LIMITING OF SPEECH SIGNALS FOR DELAY SENSITIVE VOICE COMMUNICATION - A method and system for peak limiting of speech signals for delay sensitive voice communication is disclosed. In an embodiment, a position of a sample with highest magnitude within a current block of samples is determined. Further, a peak gain to be applied for the current block of samples to bring down the highest magnitude to a predetermined threshold value is determined. Furthermore, a gain delta by which an old gain is updated to the peak gain is computed. Then, a gain factor is computed for the current block of samples based on the position of the sample with highest magnitude and the gain delta. Subsequently, the gain factor is set to a predetermined minimum gain factor when the computed gain factor is less than the predetermined minimum gain factor. In addition, gain is applied to the current block of samples using the gain factor. | 04-24-2014 |
20140142933 | DEVICE AND METHOD FOR PROCESSING VOCAL SIGNAL - A method processes vocal sounds captured by a sound capture device of an electronic device. The captured vocal sounds are divided into a plurality of sound segments, and a zero-crossing rate (ZCR) and amplitude of each of the sound segments are obtained. If one or more breathing sound segments are detected to be included in the captured vocal sounds according to the ZCR and the amplitude of each of the sound segments, the captured vocal sounds are processed to decrease the amplitude of the one or more breathing sound segments. | 05-22-2014 |
20140297273 | SPEECH ENHANCEMENT APPARATUS AND METHOD FOR EMPHASIZING CONSONANT PORTION TO IMPROVE ARTICULATION OF AUDIO SIGNAL - In a speech enhancement apparatus, a generator part generates a value representing likelihood of a consonant from an input audio signal, and a calculator part generates a consonant/vowel discriminating signal for discriminating a consonant portion and a vowel portion based on the generated value, detects a first signal level of the vowel portion and a second signal level of the consonant portion based on the audio signal and the consonant/vowel discriminating signal, and outputs a level-related signal. A determining part determines a gain coefficient that exceeds one when the second signal level is smaller than the first signal level based on the level-related signal so that the gain coefficient increases as the second signal level becomes smaller than the first signal level. A multiplier part multiplies the audio signal by the gain coefficient to output an audio signal having an emphasized consonant portion. | 10-02-2014 |
20140324418 | VOICE INPUT/OUTPUT DEVICE, METHOD AND PROGRAMME FOR PREVENTING HOWLING - A voice separation means | 10-30-2014 |
20140324419 | METHOD OF AND APPARATUS FOR EVALUATING INTELLIGIBILITY OF A DEGRADED SPEECH SIGNAL - The present invention relates to a method of evaluating intelligibility of a degraded speech signal received from an audio transmission system conveying a reference speech signal. The method comprises sampling said reference and degraded signals into reference and degraded signal frames, and forming frame pairs by associating reference and degraded signal frames with each other. For each frame pair a difference function representing disturbance is provided, which is then compensated for specific disturbance types for providing a disturbance density function. Based on the density function of a plurality of frame pairs, an overall quality parameter is determined. The method provides for weighing disturbances in silent periods dependent on the loudness of the reference signal. | 10-30-2014 |
20140330557 | DEVICES THAT TRAIN VOICE PATTERNS AND METHODS THEREOF - A voice enhancement device including an earpiece configured to be positioned in an ear canal of a user. A microcontroller is operatively coupled to the earpiece. The microcontroller is configured to selectively provide at least multitalker babble. An accelerometer is located within the earpiece and operatively coupled to the microcontroller. The accelerometer is configured to detect speech by the user and communicate with the microcontroller to provide the multitalker babble to the earpiece during the detected speech by the user. A method of making the voice enhancement device, and a method for increasing vocal loudness in a patient using the voice enhancement device are also disclosed. | 11-06-2014 |
20140372109 | SMART VOLUME CONTROL OF DEVICE AUDIO OUTPUT BASED ON RECEIVED AUDIO INPUT - A method implemented by processing and other audio components of an electronic device provides a smart audio output volume control, which correlates a volume level of an audio output to that of an audio input that triggered generation of the audio output. According to one aspect, the method includes: receiving an audio input that triggers generation of an audio output response from the user device; determining an input volume level corresponding to the received audio input; and outputting the audio output response at an output volume level correlated to the input volume level. The media output volume control level of the device is changed from a preset normal level, including from a mute setting, to the determined output level for outputting the audio output. Following, the media output volume control level is automatically reset to a pre-set volume level for normal media output. | 12-18-2014 |
20150019212 | MEASURING AND IMPROVING SPEECH INTELLIGIBILITY IN AN ENCLOSURE - A method for accurately estimating and improving the speech intelligibility from a loudspeaker (LS) is disclosed. A microphone is placed in a desired position and using an adaptive filter, an estimate of the clean speech signal at the microphone is generated. By using the adaptive-filter estimate of the clean speech signal and measuring the background noise in the enclosure an accurate Speech Intelligibility Index (SII) or Articulation Index (AI) measurement at the microphone position is obtained. On the basis of the estimated speech intelligibility measurement, a decision can be made if the LS signal needs to be modified to improve the intelligibility. | 01-15-2015 |
20150019213 | MEASURING AND IMPROVING SPEECH INTELLIGIBILITY IN AN ENCLOSURE - A method for accurately estimating and improving the speech intelligibility from a loudspeaker (LS) is disclosed. A microphone is placed in a desired position and using an adaptive filter, an estimate of the clean speech signal at the microphone is generated. By using the adaptive-filter estimate of the clean speech signal and measuring the background noise in the enclosure an accurate Speech Intelligibility Index (SII) or Articulation Index (AI) measurement at the microphone position is obtained. On the basis of the estimated speech intelligibility measurement, a decision can be made if the LS signal needs to be modified to improve the intelligibility. | 01-15-2015 |
20150142424 | Enhancement of Multichannel Audio - The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods. | 05-21-2015 |
20150310875 | APPARATUS AND METHOD FOR IMPROVING SPEECH INTELLIGIBILITY IN BACKGROUND NOISE BY AMPLIFICATION AND COMPRESSION - An apparatus for generating a modified speech signal from a speech input signal which has a plurality of speech subband signals, the modified speech signal having a plurality of modified subband signals is provided, having: a weighting information generator for generating weighting information for each speech subband signal depending on a signal power of said speech subband signal, and a signal modifier for modifying each speech subband signal by applying the weighting information on said speech subband signal to obtain a modified subband signal. The weighting information generator is configured to generate the weighting information for each of the plurality of speech subband signals, wherein the signal modifier is configured to modify each of the speech subband signals so that a first speech subband signal having a first signal power is amplified with a first degree, and so that a second speech subband signal having a second signal power is amplified with a second degree, the first signal power being greater than the second signal power, and the first degree being lower than the second degree. | 10-29-2015 |
20150317980 | ENERGY POST QUALIFICATION FOR PHRASE SPOTTING - In one embodiment, a computing device can detect an utterance of a target phrase within an acoustic input signal. The computing device can further determine a first estimate of cumulative signal and noise energy for the detected utterance in the acoustic input signal with respect to a first time period spanning the duration of the detected utterance, and a second estimate of noise energy in the acoustic input signal with respect to a second time period preceding (or following) the first time period. The computing device can then calculate a signal-to-noise ratio (SNR) for the detected utterance based on the first and second estimates and can reject the detected utterance if the SNR is below an SNR threshold. | 11-05-2015 |
20150325253 | SPEECH ENHANCEMENT DEVICE AND SPEECH ENHANCEMENT METHOD - A speech enhancement device which includes: a speech production section detection unit configured to detect a speech production section in which a speaker produces speech, from an input signal generated by a speech input unit; a timer unit configured to measure an elapsed time from a starting point of the speech production section; a gain determination unit configured to determine a gain, which represents a level of enhancement of the input signal, according to the elapsed time; and an enhancement unit configured to enhance the input signal or a spectrum signal of the input signal in the speech production section according to the gain, whereby the input signal is enhanced only at necessary portions thereof. | 11-12-2015 |
20150332585 | METHOD OF NOISE SUPPRESSION FOR VOICE BASED INTERACTIVE DEVICES - An apparatus including a security system protecting a secured area, a processor of the security system providing a voice connection between a control panel of the security system located within the secured area and a remotely located central monitoring station, the processor automatically providing notification of activation of the provided voice connection, an audio device providing audio entertainment within the secured area, a wireless interface providing a communication channel between the security system and audio entertainment system and a processor of the audio device receiving the automatic notification of activation of the voice connection from the processor of the security system through the wireless interface and automatically reducing a volume of the audio entertainment provided by the audio device within the secured area. | 11-19-2015 |
20150340048 | VOICE PROCESSING DEVICE AND VOICE PROCESSSING METHOD - A voice processing device, includes, a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: receiving, through a communication network, a first voice of a first user and a second voice of a second user inputted to a first microphone positioned nearer to the first user than the second user, and a third voice of the first user and a fourth voice of the second user inputted to a second microphone positioned nearer to the second user than the first user; calculating a first phase difference between the received first voice and the received second voice and a second phase difference between the received third voice and the received fourth voice; and performing at least one of: controlling transmission of the received second voice or the received fourth voice to a first speaker. | 11-26-2015 |
20160055859 | Smart Mute for a Communication Device - Methods, systems, and devices enable recovery of words spoken while a communication device is on mute during a voice call. A processor of the communication device or a network server may buffer audio segment in memory when the mute function is turned on. If the mute function is turned off soon after the input audio segment begins, or the processor recognizes from the spoken words that the speaker does not intend to be on mute, the processor may transmit to the third party participant a playback of at least one portion of the buffer in conjunction with turning off the mute function. Playback of the buffered audio segment may be sped up so that the playback catches up to current speech of the speaker. Buffering and playback of an input audio segment may be accomplished at the speaker's communication device or in a server within the communication network. | 02-25-2016 |
20160071527 | Method and System for Scaling Ducking of Speech-Relevant Channels in Multi-Channel Audio - A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention. | 03-10-2016 |
20160093314 | AUDIO COMMUNICATION SYSTEM, AUDIO COMMUNICATION METHOD, AUDIO COMMUNICATION PURPOSE PROGRAM, AUDIO TRANSMISSION TERMINAL, AND AUDIO TRANSMISSION TERMINAL PURPOSE PROGRAM - An audio communication system includes a generation unit that superimposes an addition sound having a volume level determined on the basis of a voice acquired by a voice acquisition unit on an input voice acquired by the voice acquisition unit of a transmission terminal and generates a synthesis sound and a transmission unit that transmits a signal of the synthesis sound generated by the generation unit to a reception terminal. | 03-31-2016 |
20160099007 | AUTOMATIC GAIN CONTROL FOR SPEECH RECOGNITION - This specification describes, among other things, a computer-implemented method. The method can include receiving a stream of audio data at a computing device. The stream of audio data can be segmented into a plurality of audio segments. Respective intensity levels are determined for each of the plurality of audio segments. For each of the plurality of audio segments and based on the respective intensity levels, a determination can be made as to whether the audio segment includes a speech signal. Selective gain control can be performed on the stream of audio data by automatically adjusting a gain of particular ones of the plurality of audio segments that are determined to include a speech signal. | 04-07-2016 |
20160118062 | Robust Voice Activity Detector System for Use with an Earphone - An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed. | 04-28-2016 |
20160133270 | METHOD FOR REDUCING NOISE AND COMPUTER PROGRAM THEREOF AND ELECTRONIC DEVICE - A method for reducing noise is used to divide a received voice into plural voice segments and set a predetermined energy value. The energy of voice segment which is higher than the predetermined energy value is determined as normal voice and outputs directly, and the energy of voice segment which is lower than the predetermined energy value is determined as noise and will be processed. | 05-12-2016 |
20160180861 | ELECTRONIC APPARATUS, CONTROL METHOD, AND COMPUTER PROGRAM | 06-23-2016 |
20160196835 | AUTOMATIC SOUND LEVEL CONTROL | 07-07-2016 |
20160379661 | NOISE REDUCTION FOR ELECTRONIC DEVICES - In one example a controller comprises logic, at least partially including hardware logic, configured to detect speech activity in an audio signal received in a non-aerial microphone and in response to the voice activity, to apply a noise cancellation algorithm to a speech input received in a aerial microphone. Other examples may be described. | 12-29-2016 |