Class / Patent application number | Description | Number of patent applications / Date published |
704220000 | Analysis by synthesis | 15 |
20080235009 | Method and apparatus for reducing synchronization delay in packet switched voice terminals using speech decoder modification - A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal u | 09-25-2008 |
20090099844 | EFFICIENT IMPLEMENTATION OF ANALYSIS AND SYNTHESIS FILTERBANKS FOR MPEG AAC AND MPEG AAC ELD ENCODERS/DECODERS - An encoder may include a core MDCT filterbank that can be used to implement an advanced audio coding (AAC) algorithm, an AAC-enhanced low delay (ELD) algorithm or both algorithms. For the AAC algorithm, a sequence of input samples is sent directly to the MDCT filterbank to obtain a sequence of output samples. For the AAC-ELD algorithm, the signs of input samples of the sequence of input samples are inverted, the MDCT analysis filterbank is applied to the sign-inverted sequence of input samples to obtain a sequence of output samples, the order of the sequence of output samples is reversed, and the signs of alternating output samples of the sequence of output samples are inverted. Similarly, a decoder may include a core IMDCT synthesis filterbank that can be used to implement AAC-ELD or both AAC and AAC-ELD algorithms. The steps for the decoder are merely the reverse of the encoder. | 04-16-2009 |
20090326931 | HIERARCHICAL ENCODING/DECODING DEVICE - A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks. | 12-31-2009 |
20100076755 | DECODING APPARATUS AND AUDIO DECODING METHOD - A decoding apparatus that uses a less number of hierarchical layers and a less amount of calculation to obtain a decoded signal having a high quality in terms of audibility. In the decoding apparatus, a first layer decoding part ( | 03-25-2010 |
20100121633 | STEREO AUDIO ENCODING DEVICE AND STEREO AUDIO ENCODING METHOD - Provided is a stereo audio encoding device which can improve ICP accuracy of a stereo audio signal having a low inter-channel correlation while suppressing a bit rate. The device ( | 05-13-2010 |
20100153101 | Automated sound segment selection method and system - A computerized method and system is provided for automatically selecting from a digitized sound sample a segment of the sample that is optimal for the purpose of measuring clinical metrics for voice and speech assessment. A quality measure based on quality parameters of segments of the sound sample is applied to candidate segments to identify the highest quality segment within the sound sample. The invention can optionally provide feedback to the speaker to help the speaker increase the quality of the sound sample provided. The invention also can optionally perform sound pressure level calibration and noise calibration. The invention may optionally compute clinical metrics on the selected segment and may further include a normative database method or system for storing and analyzing clinical measurements. | 06-17-2010 |
20100241425 | Method and Device for Coding Transition Frames in Speech Signals - There is provided a transition mode device and method for use in a predictive-type sound signal codec for producing a transition mode excitation replacing an adaptive codebook excitation in a transition frame and/or a frame following the transition in the sound signal, comprising an input for receiving a codebook index and a transition mode codebook for generating a set of codevectors independent from past excitation. The transition mode codebook is responsive to the index for generating, in the transition frame and/or frame following the transition, one of the codevectors of the set corresponding to the transition mode excitation. There is also provided an encoding device and method and a decoding device and method using the above described transition mode device and method. | 09-23-2010 |
20110040557 | TRANSMITTER AND RECEIVER FOR SPEECH CODING AND DECODING BY USING ADDITIONAL BIT ALLOCATION METHOD - The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain. | 02-17-2011 |
20110178797 | VOICE DIALOG SYSTEM WITH REJECT AVOIDANCE PROCESS - The invention relates to a process for operating a voice dialog system and a voice dialog system which can be controlled over a telecommunications link by a communications terminal, a speech element transmitted by the communications terminal being received by a receiving unit of the voice dialog system and being analyzed for statement content in a processing unit, the speech element being filed in a memory assigned to the processing unit and after the telecommunications link is broken being analyzed by the processing unit. | 07-21-2011 |
20120239390 | APPARATUS AND METHOD FOR SUPPORTING READING OF DOCUMENT, AND COMPUTER READABLE MEDIUM - According to one embodiment, an apparatus for supporting reading of a document includes a model storage unit, a document acquisition unit, a feature information extraction, and an utterance style estimation unit. The model storage unit is configured to store a model which has trained a correspondence relationship between first feature information and an utterance style. The first feature information is extracted from a plurality of sentences in a training document. The document acquisition unit is configured to acquire a document to be read. The feature information extraction unit is configured to extract second feature information from each sentence in the document to be read. The utterance style estimation unit is configured to compare the second feature information of a plurality of sentences in the document to be read with the model, and to estimate an utterance style of the each sentence of the document to be read. | 09-20-2012 |
20130297300 | Automatic determination of stability and validity of Speech Transmission Index measurements - The Speech Transmission Index (STI) uses the modulation transfer function to characterize speech communication channels with a single index. This index accurately predicts speech intelligibility. In order to measure the STI, a device or software is needed that utilizes an artificial test signal to estimate the modulation transfer function, and to derive the STI from that function. In practice, many STI measurements are found to be invalid and inaccurate, for two reasons: (1) the measurements are carried out in test conditions for which by design the STI is not intended; (2) equipment is used that does not meet specifications and requirements. A procedure is described to automatically verify if conditions for obtaining valid STI results are met. If possible, corrections are applied to compensate for incorrect settings and poorly adjusted measuring equipment. If automatic correction of the problem is not possible, then a warning is generated. | 11-07-2013 |
20140207445 | System and Method for Correcting for Lost Data in a Digital Audio Signal - In an embodiment, a method of receiving a digital audio signal, using a processor, includes generating a high band time domain signal; generating low band time domain signal; estimating an energy ratio between the high band and the low band from a last good frame; keeping the energy ratio for following frame-erased frames by applying an energy correction scaling gain to a high band signal segment by segment in the time domain; and combining the low band signal and the high band signal into a final output. | 07-24-2014 |
20140257800 | ERROR CONCEALMENT FOR SPEECH DECODER - Provided is a system, method, and computer program product for improving the quality of speech reproduction in wireless applications where the received speech frames are subject to transmission and packet losses. The speech decoding process is dynamically delayed by at least one frame period in order to perform additional error correction and concealment techniques during times when the wireless link quality if below a predetermined threshold. The wireless link is monitored and if the link quality falls below a predetermined threshold, the decoding process is delayed by at least one frame period so that one or more error correcting techniques can be performed to increase the quality of the reconstructed speech. | 09-11-2014 |
20140379333 | WAVEFORM RESYNTHESIS - A wave resynthesis method and system comprises receiving input wave form, processing received data to create an enhanced wave form, identifying the enhanced wave form, transmitting the identified wave form to a receiving unit, identifying the received wave form, resynthesizing the received wave form and outputting the resynthesized wave form. Identifying the enhanced wave form includes sampling the waveform and measuring the angle of the samples at two or more points in the waveform. The enhancing of voice audio input includes the parallel processing the input audio by a module that is a low pass filter with dynamic offset, an envelope controlled band-pass filter, a high pass filter and adding an amount of dynamic synthesized sub bass to the audio. The four processed audio signals are combined in a summing mixer with the original audio. The receiving unit has a complete set of encrypted tables for accurate resynthesizing/reproduction. | 12-25-2014 |
20160379653 | METHOD AND APPARATUS FOR INCREASING THE STRENGTH OF PHASE-BASED WATERMARKING OF AN AUDIO SIGNAL - A challenge of audio watermarking systems in which an acoustic path is involved is the robustness against microphone pickup in case of surrounding noise. The strength of phase-based watermarking is increased by determining a masking threshold for a current frequency bin in a frequency/phase representation changing the phase based on that masking threshold and an allowed phase change value, calculating an allowed magnitude change value for the current frequency bin and calculating from an audio quality level value a magnitude change scaling factor for the magnitude change value, and increasing its magnitude accordingly. | 12-29-2016 |