摘要:
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.
摘要:
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.
摘要:
A coding apparatus (500) includes: a fourth layer codebook that shows N number of (N is a natural number) codes indicating uniquely respective N integers that increment one by one; and first to third layer codebooks that show M number of (M is a natural number that satisfies M
摘要:
It possible not only to reduce a delay, but also to enhance the coding efficiency and reduce audio artifact upon coding.An audio signal coding method for coding an audio signal to be coded, the method comprising: judging for each of frames whether or not coding should be performed on each of two or more subframes into which the frame is divided, based on an audio signal contained in the frame into which the audio signal to be coded is divided for every set of samples; and when judged that the coding should be performed on each of the subframes, performing, for each subframe, subframe processing of (i) determining a value representing a characteristic of an audio signal of the subframe, and (ii) coding the audio signal using the determined value, wherein in the performing of the subframe processing, whether or not all the values determined for the subframes are the same is judged, and when all the values are the same, the audio signal is coded as exceptional processing, using a characteristic different from characteristics of audio signals indicated by the values.
摘要:
A coding apparatus includes a fourth layer codebook that shows N number of codes indicating uniquely respective N integers that increment one by one; and first to third layer codebooks that show M number of codes indicating uniquely respective M integers that are a subset of the N integers, and codes a digital signal using any one of the first to fourth layer codebooks. The coding apparatus does not need to do rescaling even when switching one of the first to fourth layer codebooks into another of them.
摘要:
A portable player (710) or a multi-channel home player (730) includes: a mixed signal decoding unit (711) that extracts, from a first inputted coded stream, a second coded stream representing a downmix signal into which multi-channel audio signals are mixed and supplementary information for reverting the downmix signal back to the multi-channel audio signals before being downmixed, and that decodes the second coded stream representing the downmix signal; a signal separation processing unit (731) that separates the downmix signal obtained by decoding based on the extracted supplementary information and that generates audio signals which are acoustically approximate to the multi-channel audio signals before being downmixed; and headphones or speakers (720) that reproduce the decoded downmix signal or speakers (740) that reproduce the multi-channel audio signals separated from the downmix singal.
摘要:
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.
摘要:
A beamforming method according to the present invention is a method of processing echo signals of a target region which are obtained from a probe including a plurality of receiving elements arrayed on a predetermined line. The beamforming method includes the following steps (S1 to S3). At S1, seed beams are formed from echo signals received by at least two receiving elements from among the plurality of receiving elements. At S2, a main beam and sub beams are formed by synthesizing at least one of the seed beams. At S3, a narrow beam for the target region is formed by multiplying the sub beams by respective predetermined coefficients and subtracting the multiplied sub beams from the main beam. Here, an signal intensity for the target region regarding the main beam is higher than a signal intensity for the target region regarding each of the sub beams.
摘要:
A beamforming method according to the present invention is a method of processing echo signals of a target region which are obtained from a probe including a plurality of receiving elements arrayed on a predetermined line. The beamforming method includes the following steps (S1 to S3). At S1, seed beams are formed from echo signals received by at least two receiving elements from among the plurality of receiving elements. At S2, a main beam and sub beams are formed by synthesizing at least one of the seed beams. At S3, a narrow beam for the target region is formed by multiplying the sub beams by respective predetermined coefficients and subtracting the multiplied sub beams from the main beam. Here, an signal intensity for the target region regarding the main beam is higher than a signal intensity for the target region regarding each of the sub beams
摘要:
It possible not only to reduce a delay, but also to enhance the coding efficiency and reduce audio artifact upon coding.An audio signal coding method for coding an audio signal to be coded, the method comprising: judging for each of frames whether or not coding should be performed on each of two or more subframes into which the frame is divided, based on an audio signal contained in the frame into which the audio signal to be coded is divided for every set of samples; and when judged that the coding should be performed on each of the subframes, performing, for each subframe, subframe processing of (i) determining a value representing a characteristic of an audio signal of the subframe, and (ii) coding the audio signal using the determined value, wherein in the performing of the subframe processing, whether or not all the values determined for the subframes are the same is judged, and when all the values are the same, the audio signal is coded as exceptional processing, using a characteristic different from characteristics of audio signals indicated by the values.