Abstract: The conventional error conceal processing generates a greatly fluctuating irregular sound which is unpleasant to ears and causes a remarkable echo effect and click noise. A notification signal detection unit (301) judges processing for an input frame. In case of an error frame, a sound detection unit (303) makes judgment whether a preceding non-error data frame is a sound signal. If it is a sound frame, a sound copying unit (304) generates a replacing frame. If it is a non-sound frame, a transient signal detection unit (305) judges whether it is an attack signal by the transient signal detection and selects an appropriate area from the preceding non-error frame.

Abstract: In the conventional art inventions for coding multi-channel audio signals, three of the major processes involved are: generation of a reverberation signal using an all-pass filter; segmentation of a signal in the time and frequency domains for the purpose of level adjustment; and mixing of a coded binaural signal with an original signal coded up to a fixed crossover frequency. These processes pose the problems mentioned in the present invention. The present invention proposes the following three embodiments: to control the extent of reverberations by dynamically adjusting all-pass filter coefficients with the inter-channel coherence cues; to segment a signal in the time domain finely in the lower frequency region and coarsely in the higher frequency region; and to control a crossover frequency used for mixing based on a bit rate, and if the original signal is coarsely quantized, to mix a downmix signal with an original signal in proportions determined by an inter-channel coherence cue.

Abstract: According to the present invention, it is possible to calculate appropriate chirp factor and noise component amount with a little processing amount. Input subband signal is segmented into a plurality of ranges by a range segmentation unit 101. The range segmentation is performed for energy value calculation, chirp factor calculation, noise component calculation, and tone component calculation, respectively, and determined range segmentation information ei, bi, qi, and hi are outputted. Respective processing for the energy calculation, the chirp factor calculation, the tone component calculation, and the noise component calculation are performed sequentially for the respective corresponding ranges. By using linear prediction processing, it is possible to obtain an parameter having higher accuracy with a little operation amount.

Abstract: Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit (103) performs LPC analysis on an L channel signal and an R channel signal so as to obtain an L channel LPC coefficient and an R channel LPC coefficient. An LPC coefficient adaptive filter (105) obtains an LPC coefficient adaptive filter parameter to minimize the mean square error between the L channel LPC coefficient and the R channel LPC coefficient. An LPC coefficient reconfiguration unit (106) reconfigures the R channel LPC coefficient by using the L channel LPC coefficient and the LPC coefficient adaptive filter parameter. A route calculation unit (107) calculates a polynomial route indicating the safety of the R channel reconfigured LPC coefficient. A selection unit (108) selects and outputs the LPC coefficient adaptive filter parameter or the R channel LPC coefficient according to the safety of the R channel reconfigured LPC coefficient.

Abstract: Multichannel signal coding equipment is provided for presenting a high quality sound at a low bit rate. In the multichannel signal coding equipment (2), a down mix part (10) generates monaural reference channel signals for N number of channel signals. A coding part (11) codes the generated reference channel signal. A signal analyzing part (12) extracts parameters indicating characteristics of each of the N number of channel signals. An MUX part (13) multiplexes the coded reference channel signal with the extracted parameters.

Abstract: This method and apparatus extract symbolic high-level musical structure resembling that of a music score. Humming or the like is converted with this invention into a sequence of notes that represent the melody that the user (usually human, but potentially animal) is trying to express. These retrieved notes each contain information such as a pitch, the start time and duration and the series contains the relative order of each note. A possible application of the invention is a music retrieval system whereby humming forms the query to some search engine.

Abstract: There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: A method (100) and apparatus (200) are disclosed for transcribing a humming signal into a sequence of musical notes. The method begins by grouping (305) the signal into frames of data samples. Each frame is then processed to derive (320) a frequency distribution for each frames. The frequency distributions are processed to derive (410) a Harmonic Product Energy (HPE) distribution over the frames. The MPE distribution is then segmented (115, 120) to obtain boundaries of musical notes. The frequency distributions of the frames are also processed to derive (412) a fundamental frequency distribution. A pitch for each note is determined (125) from the fundamental frequency distribution.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: A method for use in identifying an audio input, comprising the steps of: deriving a signature code from the audio input; subjecting the signature code to Correlation Matrix Memory (CMM) processing; and identifying the audio input based on an output of the CMM processing.

Abstract: A pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals (ST21 to ST25) of this pulse allocating method, for individual subframes, the stereophonic voice signals are compared (ST21) to judge similarity between channels, and are judged (ST22) on their characteristics. On the basis of the similarity between the channels and the characteristics of the stereophonic signals, the pulse numbers to be allocated to the individual channels are determined (ST23). Pulse retrievals are executed (ST24) to determine the pulse positions for the individual channels, so that the pulses determined at ST24 are coded (ST25).

Abstract: A frame type for a current SBR frame is determined according to a type of end border of a previous frame, as well as presence of a transient in the current SBR frame. A start border is determined according to the end border of the previous SBR frame. For a FIXFIX frame, a low time-resolution setting is used. For a FIXVAR or a VARVAR frame, a search for intermediate borders is conducted in the region between the transient and maximum allowed end border location. The end border is also determined at this stage. If there is excess capacity for more borders, another search is conducted in the region between the transient and the start border. For a VARFIX frame, only one search needs to be conducted, in the whole region partitioned by a variable start border and a fixed end border. All of the above are accomplished with two Forward Search operations and one Backward Search operation.

Abstract: A spectrum modifying method and the like wherein the efficiencies of the signal estimation and prediction can be improved and the spectrum can be more efficiently encoded. According to this method, the pitch period is calculated from an original signal, which serves as a reference signal, and then a basic pitch frequency (f0) is calculated. Thereafter, the spectrum of a target signal, which is a target of spectrum modification, is divided into a plurality of partitions. It is specified here that the width of each partition be the basic pitch frequency. Then, the spectra of bands are interleaved such that a plurality of peaks having similar amplitudes are unified into a group. The basic pitch frequency is used as an interleave pitch.

Abstract: A stereo signal generating apparatus capable of obtaining stereo signals that exhibit a low bit rate and an excellent reproducibility. In this stereo signal generating apparatus (90), an FT part (901) converts a monaural signal (M?t) of time domain to a monaural signal (M?) of frequency domain. A power spectrum calculating part (902) determines a power spectrum (PM?). A scaling ratio calculating part (904a) determines a scaling ratio (SL) for a left channel, while a scaling ratio calculating part (904b) determines a scaling ratio (SR) for a right channel. A multiplying part (905a) multiplies the monaural signal (M?) of frequency domain by the scaling ratio (SL) to produce a left channel signal (L?) of a stereo signal, while a multiplying part (905b) multiplies themonaural signal (M?)of frequency domain by the scaling ratio (SR) to produce a right channel signal (R?) of the stereo signal.

Abstract: According to the present invention, it is possible to calculate appropriate chirp factor and noise component amount with a little processing amount. Input subband signal is segmented into a plurality of ranges by a range segmentation unit 101. The range segmentation is performed for energy value calculation, chirp factor calculation, noise component calculation, and tone component calculation, respectively, and determined range segmentation information ei, bi, qi, and hi are outputted. Respective processing for the energy calculation, the chirp factor calculation, the tone component calculation, and the noise component calculation are performed sequentially for the respective corresponding ranges. By using linear prediction processing, it is possible to obtain an parameter having higher accuracy with a little operation amount.

Abstract: An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

Abstract: A wideband, high quality audio signal is decoded with few calculations at a low bitrate. Unwanted spectrum components accompanying sinusoidal signal injection by a synthesis subband filter built with real-value operations are suppressed by inserting a suppression signal to subbands adjacent to the subband to which the sine wave is injected. This makes it possible to inject a desired sinusoid with few calculations.

Abstract: An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

Abstract: A method for identifying and categorizing an audio signal into subclasses to determine a subframe block size of a transform coder. A number of block sizes available for the transform coder is determined. An input audio signal is then sampled at predetermined time intervals to produce a plurality of samples, which are grouped into frames, in which each frame has an equal number of samples. The frames are analyzed in a time domain to produce at least one comparison index, after which an appropriate block size is selected for the transform coder.