Fujitsu Corporation

Takashi Makiuchi

Masanao Suzuki

Yoshiteru Tsuchinaga

Miyuki Shirakawa

Masakiyo Tanaka

Takeshi Otani

Yasuji Ota

Keiichi Kojima

A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit. A decoding unit that decodes the low-frequency component decoded from the first encoded data, and the high-frequency components compensated by the high-frequency component compensating unit, into the audio signal.

Decoding method and apparatus for an audio signal through high frequency compensation

An SBR encoder includes a filter bank that receives an input signal, a time/frequency grid generator that controls a number of bits of various parameters, a parameter calculator that calculates various parameters, a parameter coding unit that encodes the parameters, an upper-limit number-of-bit storage unit that stores an upper limit of the number of bit of encoded data of high-frequency component finally generated in a high-pass encoding process, and a number-of-bit controller. The number-of-bit controller controls the high-pass encoding process by preferentially encoding a parameter having a large influence to sound quality and not encoding a parameter having a small influence to the sound quality relative to a plurality of parameters, so that the number of bits of the encoded data of high-frequency component finally generated in the high-pass encoding process becomes equal to or less than the upper limit to be stored in the upper-limit number-of-bit storage unit.

SBR encoder with high frequency parameter bit estimating and limiting

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.

Decoding apparatus and decoding method

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.

SBR encoder with spectrum power correction

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.

Jitter buffer controller

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.

Audio regeneration method

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.

Encoding device and encoding method

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.

Audio signal interpolation method and device

To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.

Apparatus and method for controlling audio-frame division

In a gain adjusting method and a gain adjusting device for adjusting gain of a processed voice signal that is obtained by signal processing of an input voice signal,

Gain adjusting method and a gain adjusting device

In a packet receiving method and device which convert a voice packet received into a voice, a receiving packet buffer temporarily stores a voice packet received; a plurality of parameter information monitors respectively determine different buffer adjustment values for determining a buffering amount of the receiving packet buffer based on one or more pieces of parameter information obtained from the voice packet temporarily stored; a buffer adjustment value determiner determines a receiving buffer adjustment value from the plural buffer adjustment values; and a buffer controller controls the buffering amount based on the receiving buffer adjustment value.

Packet receiving method and device

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.

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Fukuoka, Japan

Takashi Makiuchi