Class / Patent application number | Description | Number of patent applications / Date published |
704262000 | Linear prediction | 21 |
20090106027 | VOICE EDITION DEVICE, VOICE EDITION METHOD, AND VOICE EDITION PROGRAM - An object of the invention is to conveniently increase standard patterns registered in a voice recognition device to efficiently extend the amount of words that can be voice-recognized. | 04-23-2009 |
20090187409 | METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO SIGNALS - Techniques for efficiently encoding an input signal are described. In one design, a generalized encoder encodes the input signal (e.g., an audio signal) based on at least one detector and multiple encoders. The at least one detector may include a signal activity detector, a noise-like signal detector, a sparseness detector, some other detector, or a combination thereof. The multiple encoders may include a silence encoder, a noise-like signal encoder, a time-domain encoder, a transform-domain encoder, some other encoder, or a combination thereof. The characteristics of the input signal may be determined based on the at least one detector. An encoder may be selected from among the multiple encoders based on the characteristics of the input signal. The input signal may be encoded based on the selected encoder. The input signal may include a sequence of frames, and detection and encoding may be performed for each frame. | 07-23-2009 |
20090292542 | SIGNAL PROCESSING METHOD, PROCESSING APPARTUS AND VOICE DECODER - The present invention discloses a signal processing method adapted to process a synthesized signal in packet loss concealment. The method includes the following steps: receiving a good frame following a lost frame, obtaining an energy ratio of energy of a signal in the signal of the good frame signal to energy of a synthesized signal corresponding to the same time of the good frame; and adjusting the synthesized signal in accordance with the energy ratio. The present invention also discloses a signal processing apparatus and a voice decoder. Through using the method provided by the present invention, the synthesized signal is adjusted in accordance with the energy ratio of the energy of the first good frame following the lost frame to the energy of the synthesized signal to ensure that there be not a waveform sudden change or an energy sudden change at the place where the lost frame and the first good frame following the lost frame are jointed in the synthesized signal, to realize the waveform's smooth transition and to avoid music noises. | 11-26-2009 |
20100131276 | AUDIO SIGNAL SYNTHESIS | 05-27-2010 |
20100191534 | METHOD AND APPARATUS FOR COMPRESSION OR DECOMPRESSION OF DIGITAL SIGNALS - The subject matter disclosed herein relates generally to a system and method for linear prediction of sample values. | 07-29-2010 |
20100332232 | METHOD AND DEVICE FOR UPDATING STATUS OF SYNTHESIS FILTERS - A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal. | 12-30-2010 |
20110010179 | VOICE SYNTHESIS AND PROCESSING - A method and an apparatus for voice synthesis and processing have been presented. In one exemplary method, a first audio recording of a human speech in a natural language is received. Then speech analysis synthesis algorithm is applied to the first audio recording to synthesize a second audio recording from the first audio recording such that the second audio recording sounds humanistic and consistent, but unintelligible. | 01-13-2011 |
20110077945 | FLEXIBLE PARAMETER UPDATE IN AUDIO/SPEECH CODED SIGNALS - This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set. | 03-31-2011 |
20110099014 | SPEECH CONTENT BASED PACKET LOSS CONCEALMENT - Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal. | 04-28-2011 |
20110270614 | Method and Apparatus for Switching Speech or Audio Signals - A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user. | 11-03-2011 |
20120065980 | CODING AND DECODING A TRANSIENT FRAME - An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined. | 03-15-2012 |
20120116769 | SYSTEM FOR BANDWIDTH EXTENSION OF NARROW-BAND SPEECH - A method applies a parametric approach to bandwidth extension but does not require training. The method computes narrowband linear predictive coefficients from a received narrowband speech signal, computes narrowband partial correlation coefficients using recursion, computes M | 05-10-2012 |
20120150544 | METHOD AND SYSTEM FOR RECONSTRUCTING SPEECH FROM AN INPUT SIGNAL COMPRISING WHISPERS - A system for reconstructing speech from an input signal comprising whispers is disclosed. The system comprises an analysis unit configured to analyse the input signal to form a representation of the input signal; an enhancement unit configured to modify the representation of the input signal to adjust a spectrum of the input signal, wherein the adjusting of the spectrum of the input signal comprises modifying a bandwidth of at least one formant in the spectrum to achieve a predetermined spectral energy distribution and amplitude for the at least one formant; and a synthesis unit configured to reconstruct speech from the modified representation of the input signal. | 06-14-2012 |
20130024198 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 01-24-2013 |
20130090929 | HYBRID AUDIO ENCODER AND HYBRID AUDIO DECODER - Provided are a new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals. Currently, very low bitrate audio coding methods for speech and audio signal are proposed. These audio coding methods cause very long delay. Generally, in coding an audio signal, algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for these two kinds of input signals with very low bitrate, this invention provides a combination of a low delay filter bank like AAC-ELD and a CELP coding method. | 04-11-2013 |
20130185075 | Audio Signal Encoding Method, Audio Signal Decoding Method, Encoding Device, Decoding Device, Audio Signal Processing System, Audio Signal Encoding Program, and Audio Signal Decoding Program - When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized. | 07-18-2013 |
20130211839 | FEATURE SEQUENCE GENERATING DEVICE, FEATURE SEQUENCE GENERATING METHOD, AND FEATURE SEQUENCE GENERATING PROGRAM - Spread level parameter correcting means | 08-15-2013 |
20140180696 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 06-26-2014 |
20150106102 | GAIN SHAPE ESTIMATION FOR IMPROVED TRACKING OF HIGH-BAND TEMPORAL CHARACTERISTICS - A method includes determining, at a speech encoder, first gain shape parameters based on a harmonically extended signal and/or based on a high-band residual signal associated with a high-band portion of an audio signal. The method also includes determining second gain shape parameters based on a synthesized high-band signal and based on the high-band portion of the audio signal. The method further includes inserting the first gain parameters and the second gain shape parameters into an encoded version of the audio signal to enable gain adjustment during reproduction of the audio signal from the encoded version of the audio signal. | 04-16-2015 |
20160027430 | METHOD FOR FORMING THE EXCITATION SIGNAL FOR A GLOTTAL PULSE MODEL BASED PARAMETRIC SPEECH SYNTHESIS SYSTEM - A system and method are presented for forming the excitation signal for a glottal pulse model based parametric speech synthesis system. The excitation signal may be formed by using a plurality of sub-band templates instead of a single one. The plurality of sub-band templates may be combined to form the excitation signal wherein the proportion in which the templates are added is dynamically based on determined energy coefficients. These coefficients vary from frame to frame and are learned, along with the spectral parameters, during feature training. The coefficients are appended to the feature vector, which comprises spectral parameters and is modeled using HMMs, and the excitation signal is determined. | 01-28-2016 |
20160055858 | SYSTEM AND METHOD FOR REDUCING TANDEMING EFFECTS IN A COMMUNICATION SYSTEM - The present disclosure is directed towards a system and method for reducing tandeming effects in a communications system. The method may include receiving, at a speech decoder, an input bitstream associated with an incoming initial speech signal from a speech encoder. The method may further include determining whether or not coding is required and if coding is required, modifying an excitation signal associated with the bitstream. The method may also include providing the modified excitation signal to an adaptive encoder. | 02-25-2016 |