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 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 encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.
摘要:
An encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.
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:
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.
摘要:
According to an aspect of an embodiment, a method for regenerating an audio signal including a low frequency component and a high frequency component by decoding a coded data including a first coded data and a second coded data, the method comprising the steps of: generating the low frequency component; generating the high frequency component; determining whether the low frequency component has transient characteristics or not; generating a low frequency correction component by removing a stationary component when the audio signal has the transient characteristics; generating a corrected high frequency component by correcting the high-frequency component on the basis of the duration of the low frequency correction component when the audio signal has the transient characteristics; and regenerating the audio signal by synthesizing the low frequency component with the corrected high-frequency component.
摘要:
A gain adjusting method and a gain adjusting device for adjusting gain of a processed voice signal that is obtained by signal processing an input voice signal are disclosed. According to the gain adjusting method,a masking property of the processed voice signal is computed, andgain is adjusted for every frequency if the frequency is masked according to the masking property, while canceling a difference between the processed voice signal and the input voice signal where the frequency is not masked.
摘要:
In a voice packet communication system, a voice packet loss concealment device compensates for the deterioration of voice quality due to voice packet loss. In the device, a detecting section detects a loss of a voice packet and outputting information; an estimating section estimates the voice characteristics of the lost segment using a pre-loss voice packet received before the lost segment or a post-loss voice packet received after the lost segment; a pitch signal generating section generates a pitch signal having the voice characteristics; and a lost packet generating section outputs the pitch signal generated by the pitch signal generating section, with the voice characteristics estimated by the estimating section, which allows abnormal noise and feeling of mute, subjective deterioration of naturalness and continuity to be improved, and the voice packet loss concealment to be further improved.
摘要:
When creating SBR data in a the low-resolution mode, an encoding device divides a high-frequency component of input audio data being encoded by SBR method into a high-frequency band and a low-frequency band, and calculates an average high-frequency power value that indicates the average value of the power in the high-frequency band of the audio data, as well as an average low-frequency power value that indicates the average value of the power in the low-frequency band of the audio data. The encoding device then compares the average high-frequency power value and the average low-frequency power value, selecting the smaller of the two. The encoding device then corrects the power of the high-frequency component of the signal being encoded by the SBR method so that it equals the selected average power value.