摘要:
In a signal processing method and apparatus, a predetermined correcting signal having a same frame length as a second frame signal in which predetermined processing is performed to a frequency spectrum of a first frame signal of a frame length to which a predetermined window function is performed and is converted into a time domain is adjusted so that amplitudes of both ends of the correcting signal become equal to amplitudes of both or one of frame ends of the second frame signal, and a corrected frame signal is obtained by subtracting an adjusted correcting signal from the second frame signal.
摘要:
It is an object of this invention to improve speech quality in voice communications. Provided is a jitter buffer controller for controlling a jitter buffer in which arrived packets are accumulated, including: a jitter measuring portion for measuring jitters in the arrived packets; a judging portion for judging whether or not the jitters of the packets can be absorbed with an accumulation capacity of the jitter buffer; a determining portion for determining levels of importance of the packets; and a control portion for performing reproduction processing or discarding processing on a packet, among the packets accumulated in the jitter buffer, having jitter that cannot be absorbed with the accumulation capacity of the buffer, depending on a level of importance of the packet.
摘要:
A sleep apnea syndrome testing apparatus includes: an analyzing unit that analyzes every unit time a sound signal resulting from a subject during sleep; a determining unit that determines whether a unit time of the sound signal includes a breath sound or a characteristic sound produced when a patient with sleep apnea syndrome recovers from an apneic state into a breathing state, and the determining unit that determines that the sleep state of the subject in the unit time is any one of “breathing restored state,” “state with breathing,” and “state without breathing;” a storage unit in which a sleep state of the subject in each unit time is stored; and a detecting unit that detects an apneic state of the subject if a history of the sleep states of the subject indicates at least a transition from the “state without breathing” to the “breathing restored state.”
摘要:
An audio coding device includes a time-to-frequency converter that performs time-to-frequency conversion on each frame of a signal in at least one channel included in an audio signal in a predetermined length of time in order to convert the signal in the at least one channel to a frequency signal; a complexity calculator that calculates complexity of the frequency signal for each of the at least one channel. The audio further includes a bit allocation controller that determines a number of bits to be allocated to each of at least one channel so that more bits are allocated to the each of the at least one channel as the complexity of the each of at least one channel increases, and increases the number of bits to be allocated as an estimation error in the number; and a coder that codes the frequency signal.
摘要:
A decoding device to decode a main signal code obtained by encoding low-frequency components of an original signal and to output a lowband main signal for output of a main signal, includes: a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, decoding auxiliary information code obtained by encoding auxiliary information, the auxiliary information being for generating, from the lowband main signal, a highband main signal corresponding to high-frequency components of the original signal; decoding residual code obtained by encoding low-frequency components of a residual signal indicating error components produced by encoding of the original signal, and thereby output a lowband residual signal; generating a highband residual signal indicating high-frequency components of the residual signal, based on the lowband residual signal output by the residual decoder and the output auxiliary information; generating an output signal.
摘要:
An audio encoding device includes, a time-frequency transform unit that transforms signals of channels included in an audio signal having a first number of channels into frequency signals respectively, a down-mix unit that generates an audio frequency signal having a second number of channels, a low channel encoding unit that generates a low channel audio code by encoding the audio frequency signal, a space information extraction unit that extracts space information representing spatial information of a sound, an importance calculation unit that calculates importance on the basis of the space information, a space information correction unit that corrects the space information, a space information encoding unit that generates a space information code, and a multiplexing unit that generates an encoded audio signal by multiplexing the low channel audio code and the space information code.
摘要:
An audio decoding apparatus and method are provided. The audio decoding apparatus includes a spectrum converting part configured to divide the first frequency spectrum in each channel of the first audio signal in a time direction or in a frequency direction to calculate a first signal sequence having the same time resolution and the same frequency resolution in all the channels of the first audio signal, a down-mixing part configured to perform weighted addition on the signals at the same time and within the same frequency band included in the first signal sequence in all the channels to calculate a second signal sequence having channels of a second number different from the first number of channels.
摘要:
An encoding apparatus including an SBR (Spectral Band Replication) encoder creates high-frequency-component encoded data with reduced bits. The encoding apparatus converts an input signal into a frequency-domain spectrum signal, divides the converted spectrum signal into an arbitrary number of segments with respect to a time axis and a frequency axis, calculates a spectrum power of each segment and a feature parameter that represents a feature of the corresponding spectrum power, calculates a masking threshold using the calculated spectrum power of each segment, detects a segment having a spectrum power equal to or less than the calculated masking threshold, corrects the spectrum power of the detected segment, and encodes both the corrected spectrum power and the calculated feature parameter. The correction reduces a difference between quantization values, reducing the number of encoded bits.
摘要:
An audio signal interpolation device comprises a spectral movement calculation unit which determines a spectral movement which is indicative of a difference in each of spectral components between a frequency spectrum of a current frame of an input audio signal and a frequency spectrum of a previous frame of the input audio signal stored in a spectrum storing unit. An interpolation band determination unit determines a frequency band to be interpolated by using the frequency spectrum of the current frame and the spectral movement. A spectrum interpolation unit performs interpolation of spectral components in the frequency band for the current frame by using either the frequency spectrum of the current frame or the frequency spectrum of the previous frame.
摘要:
An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.