Class / Patent application number | Description | Number of patent applications / Date published |
704229000 | Adaptive bit allocation | 26 |
20080228475 | Method for Generating Encoded Audio Signal and Method for Processing Audio Signal - A method for generating an encoded audio signal, and a method for processing the same during the multi-channel audio coding are disclosed. The present invention provides the method for generating an encoded audio signal comprising: including basic configuration information requisite for a multi-channel audio coding process; and including extension configuration information, wherein the extension configuration information includes configuration information of extension environment which is identified by a type identifier (ID). | 09-18-2008 |
20090048828 | GAP INTERPOLATION IN ACOUSTIC SIGNALS USING COHERENT DEMODULATION - Information is estimated to fill-in even relatively long gaps (e.g., up to 250 ms) that occur in a signal due to physical errors in media or transmission, where the omitted information causes signal distortion. The signal is first divided into a plurality of subbands, since the gaps in each subband are individually easier to interpolate. Coherent demodulation is then employed on each subband signal to reduce the time-varying signals to a collection of pairs of frequency-modulated carriers multiplied by complex-valued envelopes, or modulators. Standard interpolation is then separately applied to the modulators and carriers of these pairs to fill-in the gaps in each of the subbands, and the interpolated pairs are remodulated. The resulting interpolated signals from each of the subbands are recombined to form the final interpolated output signal in which the gaps are filled in with estimated data. | 02-19-2009 |
20090070107 | SCALABLE ENCODING DEVICE AND SCALABLE ENCODING METHOD - Provided is a scalable encoding device capable of improving quality of a decoded signal without increasing an encoding amount and compensating data with a sufficient quality upon data loss. In the scalable encoding device, an extension layer bit distribution calculation unit ( | 03-12-2009 |
20090089052 | Signal Recording and Reproducing Apparatus and Method - A signal recording and reproducing apparatus includes an encoder encoding an input signal to produce a first group of encoded data, and a second group of encoded data used for reproducing a signal of higher quality than a signal resulting from decoding of the first group of encoded data, a recording unit recording record-data, including the first group and the second group of encoded data, into a recording medium, a reproducing unit reproducing the record-data from the recording medium, a decoder decoding at least the first group of encoded data out of the record-data from the reproducing unit, and a controller controlling an operation of each part of the recording and reproducing apparatus, and the controller performs control so as to cause the recording unit to erase the second group of encoded data according to a command to increase the amount of free storage capacity of the recording medium. | 04-02-2009 |
20090171658 | SELECTION OF SPEECH ENCODING SCHEME IN WIRELESS COMMUNICATION TERMINALS - A method for communication includes receiving modulated signals, which convey encoded speech. A measure of information entropy associated with the received signals is estimated. A speech encoding scheme is selected responsively to the estimated measure of the information entropy. A request to encode subsequent speech using the selected speech encoding scheme is sent to a transmitter. | 07-02-2009 |
20090319263 | CODING OF TRANSITIONAL SPEECH FRAMES FOR LOW-BIT-RATE APPLICATIONS - Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed. | 12-24-2009 |
20100153103 | METHOD AND SYSTEM FOR DECODING WCDMA AMR SPEECH DATA USING REDUNDANCY - WCDMA speech data is received over a plurality of channels each with at least one bit-sequence generated using a channel decoding such as a convolution decoding. At least one junction is selected in the generated at least one bit-sequence using a determined channel metric and/or physical constraint metric. Bits in the generated at least one bit-sequence are concatenated based on redundancy and the selected junctions to form at least one speech stream. A single speech stream is selected based on speech constraints for voice decoding. The at least one bit-sequence is selected, for example, using a maximum likelihood metric, by searching starting from a selected junction corresponding to a highest junction metric value. The selected at least one bit-sequence is verified using a selected redundancy verification parameter. The single speech stream is formed using the selected at least one bit-sequence over different channels for voice decoding. | 06-17-2010 |
20100161325 | Individual Codec Pathway Impairment Indicator for use in a Communication System - A system and method are described for controlling the setup of a connection in a communication network that includes a set of nodes, such as Mobile Switching Centers (MSCs) and Media Gateways (MGWs). In one example, a speech connection is established between MGWs subject to the control of the MSCs, which can selectively activate or deactivate codecs along the connection. The codecs are selected from a list of supported codecs, each potentially affecting connection quality by differing amounts. A Total Accumulated Impairment (TAI) element is forwarded between the MSCs and step by step updated by these MSCs that includes individual partially accumulated impairment values corresponding to each of the supported codec candidates. Each individual indicator value provides information representative of the expected accumulated impairment along a candidate connection path leading up to, and including, the corresponding codec. By providing information pertaining to the expected accumulated impairment along each candidate connection path, the MSCs can determine an optimum sequence of codecs so as to minimize the overall connection impairment. | 06-24-2010 |
20110077940 | Speech encoding - A method system and program for encoding and decoding a speech signal including error correction data. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, analysing the speech signal to determine side information and a residual signal, encoding the residual signal at a first bit rate, and generating an output bitstream based on the residual signal encoded at the first bit rate, and for at least one of the plurality of frames of the speech signal, encoding the residual signal at a second bit rate that is lower than the first bit rate; and generating error correction data based on the residual signal encoded at the second bit rate. | 03-31-2011 |
20110106532 | APPARATUS AND METHOD FOR ENCODING AND DECODING ENHANCEMENT LAYER - Provided is a method and apparatus for encoding and decoding an enhancement layer to reduce quantization error in a G.711 codec. Exponent indices of additional mantissa information of each sample are calculated based upon exponent information of each sample in a frame. A process of allocating 1 bit to each sample with a current exponent index is repeated, the exponent index starting from the maximum value while decreasing by 1 at every repetition until the total number of bits allocated to the samples is equal to the total number of available bits in the frame. And the most significant bits, as many as the number of bits allocated to each sample, are extracted from the additional mantissa information of each sample in the frame. | 05-05-2011 |
20110172998 | METHOD AND ARRANGEMENT FOR ENHANCING SPEECH QUALITY - The present invention relates to a method and arrangement for improving quality of a voice transmission by extracting filter coefficient parameters with respect to a voice signal in a first speech transmission rate, and using the extracted filter coefficient parameters in a second transmission rate that is equal or lower than the first transmission rate. | 07-14-2011 |
20110320196 | METHOD FOR ENCODING AND DECODING AN AUDIO SIGNAL AND APPARATUS FOR SAME - A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided. | 12-29-2011 |
20120022861 | PARALLEL ENTROPY ENCODER AND PARALLEL ENTROPY DECODER - An entropy encoder block for use in a context adaptive encoder and an entropy decoder block for use in a context adaptive decoder is presented. The encoder block includes a plurality of encoding elements, for processing encoding search tree look tables corresponding to encoding probabilities used by the context adaptive encoder, at least two of the encoding elements servicing the same probability. In an embodiment, at least one of the encoding search tree lookup tables comprises a set of shared encoding search tree lookup tables, accessible by at least two of the encoding elements. The decoder block includes a plurality of decoding elements, for processing decoding search tree lookup tables corresponding to the decoding probabilities used by the context adaptive decoder, at least two of the decoding elements servicing the same probability. In an embodiment, at least one of the decoding search tree lookup tables comprises a set of shared decoding search tree lookup tables, accessible by at least two of the decoding elements. | 01-26-2012 |
20120109646 | SPEAKER ADAPTATION METHOD AND APPARATUS - A speaker adaptation method and apparatus are provided including extracting adapted data from speech recognition data stored in a database, and modifying a sound model by using a speaker adaptation method selected based on a type of the extracted adapted data. | 05-03-2012 |
20120136657 | AUDIO CODING DEVICE, METHOD, AND COMPUTER-READABLE RECORDING MEDIUM STORING PROGRAM - An audio coding device includes a time-to-frequency converter that performs time-to-frequency conversion on each frame of a signal in at least one channel included in an audio signal in a predetermined length of time in order to convert the signal in the at least one channel to a frequency signal; a complexity calculator that calculates complexity of the frequency signal for each of the at least one channel. The audio further includes a bit allocation controller that determines a number of bits to be allocated to each of at least one channel so that more bits are allocated to the each of the at least one channel as the complexity of the each of at least one channel increases, and increases the number of bits to be allocated as an estimation error in the number; and a coder that codes the frequency signal. | 05-31-2012 |
20130191121 | DEVICES FOR REDUNDANT FRAME CODING AND DECODING - A method for redundant frame coding by an electronic device is described. The method includes determining an adaptive codebook energy and a fixed codebook energy based on a frame. The method also includes coding a redundant version of the frame based on the adaptive codebook energy and the fixed codebook energy. The method further includes sending a subsequent frame. | 07-25-2013 |
20140039884 | QUALITY IMPROVEMENT TECHNIQUES IN AN AUDIO ENCODER - An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits. | 02-06-2014 |
20140172422 | SECURED AUDIO CHANNEL FOR VOICE COMMUNICATION - A security device for hindering data theft and data leaks via audio channel of a computer system is based on passing the audio signals through a coding vocoder that receives input audio signal from a computer and compressing the signal to a low bit-rate digital data indicative of human speech; and a decoding vocoder that decompress the digital data back to a secure audio signal. The data transfer of the protected audio channel is intentionally limited not to exceed the bit-rate needed to carry vocoder-compressed human speech which is well below the capabilities of unprotected audio channel. Both analog and digital audio ports may be protected. Hardware bit-rate limiter protect the system from software hacking. | 06-19-2014 |
20150127333 | EFFICIENT DIGITAL MICROPHONE RECEIVER PROCESS AND SYSTEM - A method for processing a bitstream starts by shifting a bitstream of a first sample of a signal into a buffer. The buffer also holds bits of one or more additional bitstreams for one or more additional samples of the signal. Bits of a first half of the buffer are incrementally compared to corresponding bits of a second half of the buffer. Each bit of the first half of the buffer is compared to a corresponding bit of the second half of the buffer. A computation is performed on each bit of the first half of the buffer that is equal to a corresponding bit of the second half of the buffer. The results of the computations are summed to determine an output value for the first sample of the signal. | 05-07-2015 |
20150325243 | AUDIO ENCODER AND DECODER WITH PROGRAM LOUDNESS AND BOUNDARY METADATA - Apparatus and methods for generating an encoded audio bitstream, including by including program loudness metadata and audio data in the bitstream, and optionally also program boundary metadata in at least one segment (e.g., frame) of the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, e.g., including by performing adaptive loudness processing of the audio data of an audio program indicated by the bitstream, or authentication and/or validation of metadata and/or audio data of such an audio program. Another aspect is an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method. | 11-12-2015 |
20150332677 | AUDIO CODEC MODE SELECTOR - There is inter alia a method comprising: receiving a request to change the coding rate of a multimode audio codec; determining that the request corresponds to a coding rate of another mode of operation of the multimode audio codec; determining a frame of an input audio signal of the multimode audio codec to be an active region of the audio signal; maintaining a current operating mode of the multimode audio codec; and reducing the coding rate of the multimode audio codec to a coding rate lower than the requested coding rate. | 11-19-2015 |
20160093306 | OPTIMIZING FREQUENT IN-BAND SIGNALING IN DUAL SIM DUAL ACTIVE DEVICES - A method includes: receiving a first speech frame; identifying a first codec mode based at least in part on a Codec Mode Command (CMC) comprising the first speech frame; identifying a second codec mode based at least in part on a downlink (DL) Codec Mode Indication (DCMI) comprising the first speech frame; determining, based at least in part on a current uplink (UL) codec mode, to apply one of the first codec mode, the second codec mode, and a third codec mode having a higher bit rate than the first codec mode; and applying one of the first codec mode, the second codec mode, and the third codec mode. | 03-31-2016 |
20160093307 | Latency Reduction - Provided are systems and methods for reducing end-to-end latency. An example method includes configuring an interface, between a codec and a baseband or application processor, to operate in a burst mode. Using the burst mode, a transfer of real-time data is performed between the codec and the baseband or application processor at a high rate. The high rate is defined as rate faster than a real-time rate. The exemplary method includes padding data in a time period remaining after the transfer, at the high rate, of a sample of the real-time data samples. The padded of the data may be configured such that data can be ignored by the receiving component. The interface can include a Serial Low-power Inter-chip Media Bus (SLIMBus). Power consumption may be reduced for the SLIMBus by utilizing the gear shifting or clock stopping SLIMbus features. | 03-31-2016 |
20160111104 | SIGNAL ENCODING AND DECODING METHODS AND DEVICES - Embodiments of the present disclosure provide signal encoding and decoding methods and devices. The method includes: determining, a quantity k of subbands to be encoded, where i is a positive number, and k is a positive integer; selecting, according to quantized envelopes of all subbands, k subbands from all the subbands, or selecting k subbands from all subbands according to a psychoacoustic model; and performing a first-time encoding operation on spectral coefficients of the k subbands. In the embodiments of the present disclosure, the quantity k of subbands to be encoded is determined according to the quantity of available bits and the first saturation threshold, and encoding is performed on the k subbands that are selected from all the subbands, instead of on an entire frequency band, which can reduce spectrum holes of a signal obtained through decoding, and therefore, can improve auditory quality of an output signal. | 04-21-2016 |
20160196827 | FRAME ERASURE CONCEALMENT FOR A MULTI-RATE SPEECH AND AUDIO CODEC | 07-07-2016 |
20190147897 | SPEECH AUDIO ENCODING DEVICE, SPEECH AUDIO DECODING DEVICE, SPEECH AUDIO ENCODING METHOD, AND SPEECH AUDIO DECODING METHOD | 05-16-2019 |