Abstract: A method and apparatus to encoding or decoding an audio signal is provided. In the method and apparatus, a noise-floor level to use in encoding or decoding a high frequency signal is updated according to the degree of a voiced or unvoiced sound included in the signal.

Abstract: Test system and method for analog-to-digital converter (ADC) based on a loopback architecture are provided to test an M-bit ADC. In the invention, an N-bit digital-to-analog converter (DAC) converts a digital input to a basic test signal, a segmentation circuit scales the basic test signal and superposes it with segmentation DC levels for providing corresponding segmented test signals, such that the ADC converts the segmented test signals to reflect result of testing. With the invention, practical loopback architecture of low-cost can be adopted for testing.

Abstract: In order to achieve the best improvement of ITU G.711 related codec perceptual quality, perceptual weighting controlling parameter(s) should be at least adaptive to relative quantization error statistics or adaptive to signal level. When the relative quantization error statistics are larger or the signal level is lower, the perceptual weighting should be “stronger”, which means ? in (5) is smaller; when the relative quantization error statistics are smaller or the signal level is larger, the perceptual weighting should be “weaker”, which means ? in (5) is larger.

Abstract: The invention relates to a method and an apparatus in which samples of at least a part of an audio signal of a first channel and a part of an audio signal of a second channel are used to estimate a time delay between said part of the audio signal of said first channel and said part of the audio signal of said second channel. The method includes windowing the samples; performing a time-to-frequency domain transform; and determining an inter-channel time delay between said part of the audio signal of the first channel and said part of the audio signal of said second channel on the basis of the frequency domain representations. There is also disclosed a method and an apparatus for decoding the encoded samples.

Abstract: A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data.

Abstract: An apparatus for processing audio data comprising an interaural time delay correction factor unit for receiving a plurality of channels of audio data and generating an interaural time delay correction factor. An interaural time delay correction factor insertion unit for modifying the plurality of channels of audio data as a function of the interaural time delay correction factor.

Abstract: There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).

Abstract: According to one embodiment, 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: In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: Presented herein are systems and methods for processing sound signals for use with electronic speech systems. Sound signals are temporally parsed into frames, and the speech system includes a speech codebook having entries corresponding to frame sequences. The system identifies speech sounds in an audio signal using the speech codebook.

Abstract: A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

Abstract: A method for the selection of synthesis units of a piece of information that can be decomposed into synthesis units, comprises at least the following steps for a considered information segment: determining the mean fundamental frequency value F0 for the information segment considered; selecting a sub-set of synthesis units defined as being the sub-set whose mean pitch values are the closest to the pitch value F0; applying one or more proximity criteria to the selected synthesis units to determine a synthesis unit representing the information segment.

Abstract: A speech decoding device of the invention smoothes, in decoding speech signal in a voice-less period, RMS and filter coefficients which is discontinuously transmitted, and provides them to a synthesis filter. Thereby, it is capable of preventing discontinuous changing of the filter coefficient caused by the intermittent transmission of the filter coefficient. As a result, a quality of decoding can be improved. Also, to remove an effect, caused by the smoothing process, from the filter coefficients or the RMS which are transmitted in the past frames, a smoothing factor is adjusted not to perform smoothing while a certain time period (or a certain number of frames) from when a transition is made from a voice period from a voice-less period, or when a decoded feature parameter satisfies a predetermined condition.

Abstract: Playback by a decoder of a lossy compressed digital media file without quantization gaps is disclosed. The digital media file can be formed of a number of audio samples grouped into a corresponding number of audio frames. As a method, one embodiment can be carried out by identifying an encoder used to compress the digital media file; obtaining an encoder delay value for the identified encoder; obtaining a decoder delay value for the decoder; determining a audio sample count corresponding to a last valid audio sample; setting a re-synchronization after seek option marker N audio frames from the last valid audio sample; and decoding valid audio samples using the encoder delay value, the decoder delay value, and the sample count corresponding to the last valid audio sample.

Abstract: A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules.

Abstract: Provided are an encoding method and apparatus for efficiently encoding a sinusoidal signal whose magnitude is less than a masking value according to a psychoacoustic model, a decoding method and apparatus for decoding an encoded sinusoidal signal, and a computer-readable recording medium having recorded thereon a program for executing the encoding method/the decoding method. By using a particular code indicating that the magnitude of a first sinusoidal signal is less than a masking value according to a psychoacoustic model to encode the first sinusoidal signal, difference coding for a third sinusoidal signal of a next frame, which is connected to the first sinusoidal signal, is performed using a sinusoidal signal or sinusoidal signals selected according to a method to use the particular code, and a decoding apparatus obtains a sum with a transmitted difference using the selected sinusoidal signal(s).

Abstract: In a case of transient audio input signals, in a multi-channel audio reconstruction, uncorrelated output signals are generated from an audio input signal in that the audio input signal is mixed with a representation of the audio input signal delayed by a delay time such that, in a first time interval, a first output signal corresponds to the audio input signal, and a second output signal corresponds to the delayed representation of the audio input signal, wherein, in a second time interval, the first output signal corresponds to the delayed representation of the audio input signal, and the second output signal corresponds to the audio input signal.

Abstract: A method, device and system for signal encoding and decoding, the method comprising: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. In embodiments of the present invention, according to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract: Embodiments related to utilizing substantially optimal entries for a relatively low complexity dictionary for matching pursuits coding is disclosed. In various embodiments, methods are invoked for determining a substantially optimal entry from a bases dictionary comprising a plurality of entries; and utilizing the substantially optimal entry in a relatively low complexity matching pursuits data coding. In various embodiments, a system is provided comprising a bases dictionary comprising a plurality of entries each with a width of 15 or less; a signal to be coded; and a selection module configured to receive at least one of the plurality of entries from the bases dictionary, to calculate an inner product between the at least one of the plurality of entries and the signal to be coded, and to select the entry from the at least one of the plurality of entries that produces a maximum inner product for use in at least partially coding the signal to be coded.

Abstract: The disclosure relates to a communication terminal having a bandwidth expansion device for expanding the bandwidth of a narrowband voice signal, on a low-frequency and/or high-frequency side, by synthesizing at least one frequency band on the basis of the narrowband voice signal. A qualitatively satisfactory bandwidth expansion is thus performed using a plurality of net bit rates. The bandwidth expansion device is further connected to a memory containing a lookup table comprising at least one parameter value for the bandwidth expansion, for at least two net bit rates of the narrowband voice signal. A method for expanding a bandwidth of a narrowband voice signal having at least two net bit rates in a communication terminal is also disclosed herein.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: In one embodiment, a method of signal processing including includes encoding a low-frequency portion of a speech signal into at least an encoded narrowband excitation signal and a plurality of narrowband filter parameters; and generating a highband excitation signal based on a narrowband excitation signal. The encoded narrowband excitation signal includes a time warping, and the method includes applying a time shift to a high-frequency portion of the speech signal based on the information related to the time warping. The method also includes encoding the time-shifted high-frequency portion of the speech signal into at least one (A) a plurality of highband filter parameters and (B) a plurality of high band gain factors.

Abstract: A method of reducing computations utilizing a threshold to increase efficiency and speed of a video encoder is described. Computations of transform and scaling processes of a video encoder are able to be streamlined by utilizing one or more thresholds stored within one or more lookup tables. A selected threshold is compared with a value before scaling. If the value before scaling is less than the threshold, it is known that the coefficient will be zero and thus no further computations are required. Furthermore, the coefficient is set to zero. If the value before scaling is greater than the threshold, then further calculations are performed. The method is able to be extended to eliminate computations in forward transform as well. By skipping computations when the coefficient is zero, the method eliminates wasted computation power and time.

Abstract: A stereo encoding method and apparatus are provided, so as to reduce distortion caused by delay adjustment. The stereo encoding method includes: extracting a current interchannel delay of a stereo signal and a previous delay adjacent to the current interchannel delay; performing adjustment frame judgment according to characteristics of the current stereo signal when the current delay and the previous delay are different; and performing delay adjustment on the stereo signal by using the current interchannel delay if it is judged that a frame where the current delay occurs is an adjustment frame.

Abstract: To provide an acoustic signal processing apparatus which can reduce the amount of calculation in matrix arithmetic. An acoustic signal processing apparatus converts down-mixed acoustic signals of NI channels to acoustic signals of NO channels, where NO>NI.

Abstract: In one embodiment, the method includes receiving the audio signal including a block of audio data partitioned into N sub-blocks, and restoring a plurality of code parameters s(0), s(1), . . . , s(N?1), respectively. The restoring step includes detecting s(0) from the audio signal, where s(0) represents the code parameter of the first sub-block; detecting a difference s(i)?s(i?1) from the audio signal for i=1, . . . N?1, where s(i) representing the code parameter of each sub-block following the first sub-block. The difference s(i)?s(i?1) is encoded by using first entropy code. The restoring step further includes calculating s(i) for i=1, . . . , N?1 using s(0) and the detected differences, and the method further includes decoding the N sub-blocks using the restored code parameters.

Abstract: In one embodiment, the method includes receiving the audio signal including at least one block of audio data and configuration information, and reading coding type information and partitioning information from the configuration information. The coding type information indicates an entropy coding scheme used in encoding the audio signal, and the partitioning information indicates a sub-block partition scheme by which the block is divided into sub-blocks. Sub-block information is read from the block of audio data, and the sub-block information indicates a number of the sub-blocks into which the block is partitioned given the sub-block partitioning scheme. The number of the sub-blocks is determined based on the entropy coding scheme and the sub-block partition scheme. The partitioned sub-blocks are decoded based on the entropy coding scheme.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: This invention improves the perceived quality of frequency-domain time scale modification by selection of spectral bands used in phase locking based upon a Bark scale according to the variation in human hearing frequency response. A spectral peak is identified for each band. At these peaks the phases are rotated using the phase vocoder algorithm. For a few spectral lines near these peaks, the phase differences are copied from the non-rotated spectrum. The number selected is preferably 4. Remaining spectral lines within each spectral band located farther from the peak are phase rotated using the phase vocoder algorithm. The boundaries of the spectral bands may be adjusted based upon the digital audio data to maintain important frequency groups within the same spectral band.

Abstract: Techniques for generating a target digital media item based on a source digital media item are described. A digital media item may be a song, a video clip, an album, or any length of audio or video. When adjusting the bit count for a portion of the target digital media item, instead of using the same set of parameter values used in a perceptual model for each portion of the source media item, the set of parameter values may be modified to encode the portion of the source digital media item. In this way, how audio or video is perceived is taken into account when adjusting a proposed bit count for a given portion of the target digital media item. Thus, while maintaining the same statistical bitrate as before increased digital media quality is achieved.

Abstract: A speech band extension device (100), which generates an audio signal capable of realizing natural audibility after speech band extension, includes a band-extended audio generator which generates a band-extended audio signal from an original audio signal, the band-extended audio signal including components lying within a frequency band that is not included in a frequency band of the original audio signal, and an adjustment adder (20) which detects a timing shift between the original audio signal and the band-extended audio signal, adjusts timing of the original audio signal and timing of the band-extended audio signal in accordance with the detected timing shift, and combines the both signals after the adjusting of the timing, wherein the detection of the timing shift is performed, for example, using zero-crossing and cross-correlation.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: A parameter calculator calculates lower resolution parametric information and interpolation information. On a decoder-side, an upmixer is used for generating the output channels. The upmixer uses high resolution parametric information generated by a parameter interpolator using the low resolution parametric information and decoder-side derived interpolation information or encoder-generated interpolation information for selecting one of a plurality of different interpolation characteristics.

Abstract: A data hiding system is described for hiding data within an audio signal. The system can be used for watermarking, data communications, audience surveying etc. The system hides data in an audio signal by adding artificial echoes whose polarity varies with the data to be hidden. In one embodiment, each data value is represented by a positive and a negative echo having different delays. A receiver can then remove the effects of natural echoes and/or periodicities in the audio signal by differencing measurements obtained at the different delays.

Abstract: The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state (715) located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation (745) of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: According to the invention a receiving end terminal (RET) enters a delay mode based on the detecting of the quality of the link being lower than a threshold. In this delay mode, the receiving end terminal provides a reception delay indicator (RDI) for a sending end terminal (SET). The sending end terminal (SET) receives the reception delay indicator (RDI) and provides an end of speech indicator (ESI) for the receiving end terminal (RET) at an end of a speech coding interval (SC). The receiving end terminal (RET) uses the reception delay indicator (RDI) and end of speech indicator (ESI) to define a first time interval (AL1) during which a speech decoder is disabled. The speech decoder is again activated after the first time interval (AL1).

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Abstract: An audio encoding scheme or a stream that encodes audio and video data is disclosed. The scheme has particular application in mezzanine-level coding in digital television broadcasting. The scheme has a mean effective audio frame length F that equals the video frame length 1/fV over an integral number M video frames, by provision of audio frames variable in length F in a defined sequence where length=F(j) at encoding. The length of the audio frames may be varied by altering the length of overlap between adjacent frames in accordance with an algorithm that repeats after a sequence of M frames. An encoder and a decoder for such a scheme are also disclosed.

Abstract: In one embodiment, at least one audio data frame having at least one channel is generated. Each channel is divided into a plurality of blocks. A sub-block partitioning scheme is selected, and a number of sub-blocks into which the block is to be partitioned is selected. The selected number of sub-blocks is chosen from numbers of sub-blocks available for the selected sub-block partitioning scheme. The block of audio data is partitioned into sub-blocks according to the selected number of sub-blocks, and the partitioned sub-blocks are coded according to a selected entropy coding scheme.

Abstract: An apparatus for decoding audio data that is capable of reducing the amount of calculations that are performed during the arithmetic decoding of an audio signal coded by bit sliced arithmetic coding (BSAC) to improve the performance of a decoder and a method thereof are provided.

Abstract: Loudspeakers in domestic or automotive environments are rarely placed ideally with respect to the sources supplying them, and the stereo and surround images are seldom satisfying. According to the invention there is provided a method and apparatus for combining a precise knowledge about the relative positions of the loudspeakers that were intended (the virtual loudspeakers) and a precise knowledge about the actual placement of listening loudspeakers into a vector space that enables calculation of running corrections to the signals used in order to simulate the presence of the virtual loudspeakers. Specifically the corrections may comprise gain/attenuations determined based on the distances in vector space between the virtual and actual loudspeakers and delays determined from these distances.

Abstract: There is provided an audio encoding device capable of generating an appropriate monaural signal from a stereo signal while suppressing the lowering of encoding efficiency of the monaural signal. In a monaural signal generation unit (101) of this device, an inter-channel prediction/analysis unit (201) obtains a prediction parameter based on a delay difference and an amplitude ratio between a first channel audio signal and a second channel audio signal; an intermediate prediction parameter generation unit (202) obtains an intermediate parameter of the prediction parameter (called intermediate prediction parameter) so that the monaural signal generated finally is an intermediate signal of the first channel audio signal and the second channel audio signal; and a monaural signal calculation unit (203) calculates a monaural signal by using the intermediate prediction parameter.

Abstract: Provided is a method and apparatus for encoding/decoding a multi-channel audio signal. The apparatus for encoding a multi-channel audio signal includes a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed multi-channel audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.

Abstract: Embodiments of the present invention are directed to a binaural cue coding (BCC) scheme in which an externally provided audio signal (e.g., a studio engineering audio signal) is transmitted, along with derived cue codes, to a receiver instead of an automatically downmixed audio signal. The cue codes are (adaptively) synchronized with the externally provided audio signal to compensate for time lags (and changes in those time lags) between the externally downmixed audio signal and the multi-channel signal used to generate the cue codes. If the receiver is a legacy receiver, then the studio engineered audio signal will typically provide a higher-quality playback than would be provided by the automatically downmixed audio signal. If the receiver is a BCC-capable receiver, then the synchronization of the cue codes with the externally provided audio signal will typically improve the quality of the synthesized playback.