Abstract: A non-acoustic sensor is used to measure a user's speech and then broadcasts an obscuring acoustic signal diminishing the user's vocal acoustic output intensity and/or distorting the voice sounds making them unintelligible to persons nearby.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

Abstract: Techniques for introducing information into a data stream first obtains the spectral values of the short-term spectrum of the audio signal. Separately, information to be introduced are combined with a spread sequence obtaining a spread information signal, whereupon a spectral representation of the spread information is generated, then weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal are then summed and afterwards processed again to obtain a processed data stream including audio information and information to be introduced.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for detecting and correcting abnormal stress patterns in unit-selection speech synthesis. A system practicing the method detects incorrect stress patterns in selected acoustic units representing speech to be synthesized, and corrects the incorrect stress patterns in the selected acoustic units to yield corrected stress patterns. The system can further synthesize speech based on the corrected stress patterns. In one aspect, the system also classifies the incorrect stress patterns using a machine learning algorithm such as a classification and regression tree, adaptive boosting, support vector machine, and maximum entropy. In this way a text-to-speech unit selection speech synthesizer can produce more natural sounding speech with suitable stress patterns regardless of the stress of units in a unit selection database.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Disclosed is a method capable of adaptively aligning windows to extract features according to the types and characteristics of voice signals. To this end, window lengths based on the window update points in a corresponding order are determined by employing the concept of a higher order peak, and windows are aligned according to window lengths. When the windows are aligned according to such a manner, the start and end points of each window is known, so that it becomes possible to easily extract and analyze peak feature information.

Abstract: In a masking sound generation apparatus, a CPU analyzes a speech utterance speed of a received sound signal. Then, the CPU copies the received sound signal into a plurality of sound signals and performs the following processing on each of the sound signals. Namely, the CPU divides each of the sound signals into frames on the basis of a frame length determined on the basis of the speech utterance speed. Reverse process is performed on each of the frames to replace a waveform of the frame with a reverse waveform, and a windowing process is performed to achieve a smooth connection between the frames. Then, the CPU randomly rearranges the order of the frames and mixes the plurality of sound signals to generate a masking sound signal.

Abstract: 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 (2308) adjusts, based on a control of a decoding control part (2301), the gain of a basic layer decoded signal outputted from a basic layer decoding part (2302). A gain adjusting part (2309) adjusts, based on a control of the decoding control part (2301), the gain of a first expansion layer decoded signal outputted from a first expansion layer decoding part (2303). A gain adjusting part (2310) adjusts, based on a control of the decoding control part (2301), the gain of a second expansion layer decoded signal outputted from a second expansion layer decoding part (2304).

Abstract: A system for identifying audio data is provided. The system includes a transform system receiving left channel audio data and right channel audio data and generating a plurality of frequency bins of left channel magnitude data, left channel phase data, right channel magnitude data and right channel phase data. A watermarking system receives watermarking data and modifies predetermined frequency bins of the left channel phase data and the right channel phase data to encode the watermarking data. A magnitude system receives the left channel magnitude data and the right channel magnitude data and increases the left channel magnitude data and the right channel magnitude data for one or more of the predetermined frequency bins to a threshold level if the left channel magnitude data and the right channel magnitude data for the corresponding frequency bin is less than the threshold level.

Abstract: A sound encoder having an improved quantization performance while suppressing an increase of the bit rate to a lowest level. In a second layer encoder, a standard deviation calculator calculates a standard deviation ?c of a first layer decoding spectrum after decoding a scale factor ratio multiplication and outputs the standard deviation ?c to a selector. The selector selects a linear transform function as a function for a nonlinear transform of a residual spectrum according to the standard deviation ?c A nonlinear transform function selects one of prepared nonlinear transform functions #1 to #N according to a result of the selection by the selector, and outputs the selected one to an inverse transformer. The inverse transformer subjects an inverse transform (expansion) to a residual spectrum candidate that is stored in a residual spectrum code book using the nonlinear transform function outputted from the nonlinear transform function and outputs the result to an adder.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: A method and apparatus for creating a signature of a sampled work in real-time is disclosed herein. Unique signatures of an unknown audio work are created by segmenting a file into segments having predetermined segment and hop sizes. The signature then may be compared against reference signatures. One aspect may be characterized in that the hop size of the sampled work signature is less than the hop size of reference signatures. A method for identifying an unknown audio work is also disclosed.

Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

Abstract: An audio file is divided into frames in the time domain and each frame is compressed, according to a psycho-acoustic algorithm, into file in the frequency domain. Each frame is divided into sub-bands and each sub-band is further divided into split sub-bands. The spectral energy over each split sub-band is averaged for all frames. The resulting quantity for each split sub-band provides a parameter. The set of parameters can be compared to a corresponding set of parameters generated from a different audio file to determine whether the audio files are similar. In order to provide for the higher sensitivity of the auditory response, the comparison of individual split sub-bands of the lower order sub-bands can be performed. Selected constants can be used in the comparison process to improve further the sensitivity of the comparison. In the side-information generated by the psycho-acoustic compression, data related to the rhythm, i.e., related percussive effects, is present.

Abstract: Quality of industrial products is evaluated by evaluating non-stationary operation sound, which is a kind of operation sound, from an aspect of tone, using closely simulated evaluation levels of evaluation of non-stationary sound by used of a human sense of hearing. An operation sound of a conforming product sample is converted into sound waveform data by a sound collecting unit, and the sound waveform data is input into a computer via an A-D converter, and then converted into psychoacoustic parameters. Data of pseudo conforming products is additionally obtained from the psychoacoustic parameters of a plurality of conforming product samples by making use of deviation in the data of the conforming product samples. Threshold data is obtained using thresholds and masking data for evaluation calculated from psychoacoustic parameters of data of the conforming product samples and the pseudo conforming products by a statistical technique.

Abstract: An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

Abstract: A method, medium, and apparatus encoding/decoding audio data in which audio data is hierarchically encoded, and at least one extension data of the audio data is encoded using at least one encoding method, and decoding is performed in the same manner, thereby ensuring fine grain scalability (FGS) and unlimited extendibility of the audio data.

Abstract: The system and method establish a communication with a first party. A first emotion is detected during a first step during the communication. Steps in the communication can include: establishing a call, waiting in a queue, talking with an agent, communicating with an IVR system, and the like. A second emotion is detected at a different step in the communication. Alternatively, the first and second emotions are detected in two different but related communications. The detection of the first and/or second emotion determines how to process the communication. Examples of processing a communication can include routing the communication differently, selecting a different script for an agent, selecting a different agent, and the like. In addition, the communication can also be processed differently based on other additional parameters.

Abstract: A system and associated methods provide an audio experience such that a user spatially perceives one or more audio events. One particular method set forth involves obtaining audio events and presenting the audio events so that an audio experience is provided. According to one embodiment of the method, upon obtaining audio events, the audio events are associated with one or more corresponding audio components. Thereafter, the audio experience is determined based on the audio events and associated audio components. The audio experience is then presented such that the user may spatially perceive the audio events.

Abstract: A system and method for phone detection. The system includes a microphone configured to receive a speech signal in an acoustic domain and convert the speech signal from the acoustic domain to an electrical domain, and a filter bank coupled to the microphone and configured to receive the converted speech signal and generate a plurality of channel speech signals corresponding to a plurality of channels respectively. Additionally, the system includes a plurality of onset enhancement devices configured to receive the plurality of channel speech signals and generate a plurality of onset enhanced signals. Each of the plurality of onset enhancement devices is configured to receive one of the plurality of channel speech signals, enhance one or more onsets of one or more signal pulses for the received one of the plurality of channel speech signals, and generate one of the plurality of onset enhanced signals.

Abstract: Method and apparatus for encoding/decoding audio data with scalability are provided. The method includes slicing audio data so that sliced audio data corresponds to a plurality of layers, obtaining scale band information and coding band information corresponding to each of the plurality of layers, coding additional information containing scale factor information and coding model information based on scale band information and coding band information corresponding to a first layer, obtaining quantized samples by quantizing audio data corresponding to the first layer with reference to the scale factor information, coding the obtained plurality of quantized samples in units of symbols in order from a symbol formed with most significant bits (MSB) down to a symbol formed with least significant bits (LSB) by referring to the coding model information, and repeatedly performing the steps with increasing the ordinal number of the layer one by one every time, until coding for the plurality of layers is finished.

Abstract: A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: A scalable encoding apparatus capable of reducing the bit rates of encoded parameters and also capable of efficiently encoding audio signals in which a plurality of harmonic structures are coexistent. In the apparatus, an MDCT analyzer MDCT analyzes an audio signal for converting/encoding processes. A pitch frequency converter determines an inverse of a pitch period to calculate a pitch frequency. A selector selects spectra located at frequencies that are integral multiples of the pitch frequency, and a second layer encoder encodes the selected spectra.

Abstract: A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.

Abstract: The invention provides a softscaled frequency compensation function that allows the evaluation of a first quality measure indicating a global impact of all distortions in an audio transmission system, including linear frequency response distortions and second quality measure that only lakes into account the impact of linear frequency response distortions. The softscaled frequency compensation function is derived from a softscaled ratio between a time integrated output and a time integrated input power density functions. The first quality measure is derived from the difference loudness density function as function of time and frequency, using the frequency compensated input loudness density function and the gain compensated output loudness density function both as a function of time and frequency, in the same manner as carried out in ITU standard P.862.

Abstract: A scalable encoder enabling improvement of the encoding efficiency in the second layer and improvement of the quality of the original signal decoded using the encoding signal in the second layer. A predictive coefficient encoder of the scalable encoder has a predictive coefficient codebook where candidates of the predictive coefficient are recorded. After searching the predictive coefficient codebook, the scale factor of the first layer decoded signal inputted from a scale factor calculator is multiplied, and a predictive coefficient which most approximates the multiplication result to the scale factor of the original signal inputted from the scale factor calculator is determined and encoded, and the coded code is inputted to a multiplexer.

Abstract: 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.

Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

Abstract: A downsampler 101 converts input data having a sampling rate 2*FH to a sampling rate 2*FL which is lower than the sampling rate 2*FH. A base layer coder 102 encodes the input data having the sampling rate 2*FL in predetermined base frame units. A local decoder 103 decodes a first coded code. An upsampler 104 increases the sampling rate of the decoded signal to 2*FH. A subtractor 106 subtracts the decoded signal from the input signal and regards the subtraction result as a residual signal. A frame divider 107 divides the residual signal into enhancement frames having a shorter time length than that of the base frame. An enhancement layer coder 108 encodes the residual signal divided into the enhancement frames and outputs a second coded code obtained by this coding to a multiplexer 109.

Abstract: A contour for a syllable (or other speech segment) in a voice undergoing conversion is transformed. The transform of that contour is then used to identify one or more source syllable transforms in a codebook. Information regarding the context and/or linguistic features of the contour being converted can also be compared to similar information in the codebook when identifying an appropriate source transform. Once a codebook source transform is selected, an inverse transformation is performed on a corresponding codebook target transform to yield an output contour. The corresponding codebook target transform represents a target voice version of the same syllable represented by the selected codebook source transform. The output contour may be further processed to improve conversion quality.

Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.

Abstract: Methods are provided for encoding watermark information into media data containing a series of digital samples in a sample domain. The method involves: dividing the series of digital samples into a plurality of sections in the sample domain, each section comprising a corresponding plurality of samples; processing the corresponding plurality of samples in each section to obtain a single energy value associated with each section; grouping the sections into groups, each group containing three or more sections; assigning a nominal bit value to each group according to a bit assignment rule, the bit assignment rule based on the energy values of the sections in the group; and assigning a watermark bit value to each group.

Abstract: Methods and devices are used for switching between sound signal coding modes and for producing from a decoded target signal, an overlap-add target signal in a current frame coded according to a first mode. On a coder side, switching is at the junction between a previous frame coded according to a first coding mode and a current frame coded according to a second coding mode, a sound signal is filtered through a weighting filter to produce a weighted signal in the current frame, and a windowed zero-input response of the weighting filter is removed from the weighted signal. On a decoder side, a current frame of the target signal is first windowed, a left portion of a resulting window is skipped, and then a windowed zero-input response of the weighting filter is added to the decoded target signal to reconstruct the overlap-add target signal.

Abstract: A method of and an apparatus for encoding/decoding an MPEG-4 bit sliced arithmetic coding (BSAC) audio bitstream having ancillary information. A time domain audio signal is converted to a frequency domain audio signal and quantized. A number of data bits is counted and a number of available bits per layer is obtained. The number of available bits per layer is modified considering the size of ancillary information. Actual audio data is encoded in units of layers and ancillary information is embedded in the encoded bitstream. A header is decoded and a layer structure of an audio bitstream is calculated to determine the size of the ancillary information as a difference between a size of data up to a top layer and a size of a frame. The ancillary information is extracted to improve meta data and sound quality of audio contents.

Abstract: A data reproduction device is provided for achieving seamless reproduction of a stream where a validity of a bandwidth extension function is switched in the stream. The data reproduction device includes an input frequency obtainment unit analyzing header information Hdr and obtaining an input frequency FSin, which is the frequency of basic data, an output frequency determination unit performing predetermined processing based on the input frequency FSin and determining an output frequency FSout, which is the sampling frequency of a decoded frame Fdata, and a decoding unit (2003) which, if the SBR function is valid in a frame to be decoded, decodes sample data at the input frequency FSin and extends the bandwidth of the sampling frequency up to the output frequency FSout, while if the SBR function is not valid in the frame, upsamples the decoding result obtained at the input frequency FSin to the output frequency FSout.

Abstract: A method and apparatus for performing speech quality assessment in a speech communication system (such as, for example, a VoIP communication system) which detects and measures the presence of impulsive noise is provided. Specifically, in one illustrative embodiment, an autoregressive (AR) model of speech (and, in particular, of the excitation of the vocal tract) is advantageously employed to estimate a short-term variance of the speech excitation, and the standard deviation of the speech excitation (i.e., the square root of the variance) is then advantageously compared to a predetermined threshold to identify whether impulsive noise is present. Then, based on a statistic analysis of any such identified impulsive noise, a speech quality assessment is generated.

Abstract: A method for dynamically adjusting the spectral content of an audio signal, which increases the harmonic content of said audio signal, said method comprising translating an encoded digital signal into data bands, creating a psychoacoustic model to identify sections of said data bands that are deficient in harmonic quality, analyzing the fundamental frequency and amplitude of said harmonically deficient data bands, creating additional higher order harmonics for said harmonically deficient data bands, adding said higher order harmonics back to said encoded digital signal to form a newly enhanced signal, inverse filtering said newly enhanced signal, and converting said inverse filtered signal to an analog waveform for consumption by the listener.

Abstract: An intermediate channel representation of a multi-channel signal can be reconstructed highly efficient and with high fidelity, when upmix parameters for upmixing a transmitted downmix signal to the intermediate channel representation are derived that allow for an upmix using the same upmixing algorithms as within the multi-channel reconstruction. This can be achieved when a parameter re-calculator is used to derive the upmix parameters that takes into account also parameters having information on channels that are not included in the intermediate channel representation.

Abstract: An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder. A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.

Abstract: Disclosed is an acoustic communication method that includes filtering an audio signal to attenuate a high frequency section of the audio signal; generating a residual signal which corresponds to a difference between the audio signal and the filtered signal; generating a psychoacoustic mask for the audio signal based on a predetermined psychoacoustic model; generating a psychoacoustic spectrum mask by combining the residual signal with the psychoacoustic mask; generating an acoustic communication signal by modulating digital data according to the acoustic signal spectrum mask; and combining the acoustic communication signal with the filtered signal.

Abstract: An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.

Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.

Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. At least two transforms are applied on an audio signal, each with different transform periods for better resolutions at both low and high frequencies. The transform coefficients are selected and combined such that the data rate remains similar as a single transform. The transform coefficients may be coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: A speech coding apparatus comprises a repetition period pre-selecting unit for generating a plurality of candidates for the repetition period of a driving excitation source by multiplying the repetition period of an adaptive excitation source by a plurality of constant numbers, respectively, and for pre-selecting a predetermined number of candidates from all the candidates generated. A driving excitation source coding unit provides both excitation source location information and excitation source polarity information that minimize a coding distortion, for each of the predetermined number of candidates, and provides an evaluation value associated with the minimum coding distortion for each of the predetermined number of candidates.

Abstract: A speech coding apparatus comprises a repetition period pre-selecting unit for generating a plurality of candidates for the repetition period of a driving excitation source by multiplying the repetition period of an adaptive excitation source by a plurality of constant numbers, respectively, and for pre-selecting a predetermined number of candidates from all the candidates generated. A driving excitation source coding unit provides both excitation source location information and excitation source polarity information that minimize a coding distortion, for each of the predetermined number of candidates, and provides an evaluation value associated with the minimum coding distortion for each of the predetermined number of candidates.