Abstract: In an acoustic signal encoding apparatus (100), a tonal noise verification unit (110) verifies whether the input acoustic time-domain signals are tonal or noisy. If the input acoustic time-domain signals are tonal, tonal component signals are extracted by a tonal component extraction unit (121), and tonal component parameters are normalized and quantized in a normalization/quantization unit (122). The residual time-domain signals, obtained on extracting the tonal component signals from the acoustic time-domain signals, are transformed by an orthogonal transforming unit (131) into the spectral information, which spectral information is normalized and quantized by a normalization/quantization unit (132). A code string generating unit (140) generates a code string from the quantized tonal component parameters and the quantized residual component spectral information.

Abstract: The present invention makes it possible to encode information into a code with a smaller number of bits. An encoding apparatus encodes an acoustic time series signal such that the acoustic time series signal is split into band signals in predetermined bands and gain adjustment is made on the band signals, at a gain control position by a gain control amount, for each of as many positions as indicated by a gain control number. Although the gain control number can take any one of values from 0 to 7, it has a high probability of taking a particular value (0, for example) as shown in FIG. 10. In the present invention, in view of the above, the gain control number GA is encoded such that a code with a small number of bits is assigned to the gain control number GA equal to a value having a high-occurrence probability. The present invention may be applied to a voice recording apparatus.

Abstract: A signal component coding circuit codes spectral components from a transform circuit for converting an audio signal to spectral components. A code string generation circuit generates a code string block of each unit time from the coded data from the signal component coding circuit. A compression rate change circuit changes the compression rate of the code string from the code string generation circuit, if necessary. For example, when the compression rate needs to be changed because of a change of the transmission capacity of a transmission line, the compression rate change circuit extracts codes of respective signal components from the code string, if necessary, and thus generates a code string having a changed compression rate.

Abstract: Coding is made possible with higher efficiency while the listener is prevented from feeling a sense of incongruity. An adaptive mixing section performs a mixing process on input signals on the basis of distortion factor information supplied from a distortion factor detection section, and controls the operation time of MS stereo coding or IS stereo coding. Furthermore, the adaptive mixing section creates power correction information in accordance with a mixing coefficient, and causes power correction to be performed during reproduction. A coding control section selects a coding method of a coding process performed in a coding section and supplies it to the coding section. The coding section selects dual coding, MS stereo coding, or IS stereo coding in accordance with the instructions from the coding control section, and codes a spectrum signal supplied from a domain conversion section.

Abstract: The present invention relates to an information extraction apparatus capable of analyzing an acoustic signal with accuracy and high efficiency.

Abstract: The present invention relates to encoding apparatuses which allow encoding to be performed such that the occurrence of pre-echo and post-echo is suppressed. A predetermined waveform analysis is applied to a low-frequency-component input time-sequential signal which includes a high-frequency component occurring at a specific time, and a low-frequency-component time-sequential signal like that shown in FIG. 9A is generated according to a result of the analysis. The low-frequency-component time-sequential signal is removed from the input time-sequential signal to generate a residual time-sequential signal like that shown in FIG. 9B. An amplitude control process is applied such that the amplitude of the residual time-sequential signal is made almost constant in a block which serves as a unit of encoding to generate a time-sequential signal to be quantized, like that shown in FIG. 9C. The time-sequential signal to be quantized is quantized and encoded.

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 is to provide a drug for improvement of brain function which is effective to prevent from or to cure dementia such as Alzheimer's disease. The drug for improvement of brain function of this invention comprises leptin of mammals as an effective component wherefore has a superior effect in improvement of learning and memory.

Abstract: The invention intends to decode and reproduce digital audio signals in real time. A transmission rate of a transmission line is detected, and a selection instructing unit instructs, to an encoding selecting circuit, a coding method which can provide coded data having a bit rate corresponding to the detected transmission rate. In response to the instruction from the selection instructing unit, the encoding selecting circuit controls a switch so as to select one of a plurality of encoders for encoding an audio signal with different coding methods. Each frame of the audio signal cut out by a frame cutting circuit is supplied to the encoder selected by the switch for encoding thereof. Resulting coded data is outputted after an ID representing the instructed coding method has been added to the coded data in a header inserting circuit.

Abstract: Coding is made possible with higher efficiency while the listener is prevented from feeling a sense of incongruity. An adaptive mixing section performs a mixing process on input signals on the basis of distortion factor information supplied from a distortion factor detection section, and controls the operation time of MS stereo coding or IS stereo coding. Furthermore, the adaptive mixing section creates power correction information in accordance with a mixing coefficient, and causes power correction to be performed during reproduction. A coding control section selects a coding method of a coding process performed in a coding section and supplies it to the coding section. The coding section selects dual coding, MS stereo coding, or IS stereo coding in accordance with the instructions from the coding control section, and codes a spectrum signal supplied from a domain conversion section.

Abstract: The MIDI reproducing block has such a sound source as a synthesizer and synthesizes and reproduces electronic sounds sequentially according to entered MIDI playing information, thereby generating MIDI sound signals of an accompaniment music. Generated MIDI sound signals are supplied to the mixer. The audio reproducing block reproduces audio sound signals according to entered audio information. Generated audio sound signals are supplied to the mixer. The tempo change time setting block sets a tempo change time in entered tempo change information. Tempo change information in which a tempo change time is set is supplied to both MIDI reproducing block and audio reproducing block. The mixer mixes supplied MIDI sound signals with audio sound signals, thereby generating reproduction signals, then outputs the signals after the sound volume is adjusted. Generated reproduction signals are then supplied to the speaker. The speaker then outputs supplied reproduction signals as sounds.

Abstract: In a tone color map generating section, a tone color parameter and the effective time for making the tone color map effective are extracted from an SMF stored in a MIDI file database, and the tone color parameter is stored in a tone color map database. When a demand processing section receives a playback start position for starting the playback of the SMF, a read processing section reads the tone color parameter stored in the tone color map database in accordance with the playback start position, and generates time information indicating time to set the tone color parameter in a MIDI device. A sending circuit sends the tone color parameter together with the corresponding time information, and then sends a MIDI signal from the playback start position thereafter, stored in the MIDI file database.