摘要:
There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.
摘要:
A pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals of this pulse allocating method, for individual subframes, the stereophonic voice signals are compared to judge similarity between channels, and are judged on their characteristics. On the basis of the similarity between the channels and the characteristics of the stereophonic signals, the pulse numbers to be allocated to the individual channels are determined. Pulse retrievals are executed to determine the pulse positions for the individual channels, so that the pulses determined are coded.
摘要:
There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.
摘要:
A method (100) and apparatus (200) are disclosed for transcribing a humming signal into a sequence of musical notes. The method begins by grouping (305) the signal into frames of data samples. Each frame is then processed to derive (320) a frequency distribution for each frames. The frequency distributions are processed to derive (410) a Harmonic Product Energy (HPE) distribution over the frames. The MPE distribution is then segmented (115, 120) to obtain boundaries of musical notes. The frequency distributions of the frames are also processed to derive (412) a fundamental frequency distribution. A pitch for each note is determined (125) from the fundamental frequency distribution.
摘要:
A spectrum modifying method and the like wherein the efficiencies of the signal estimation and prediction can be improved and the spectrum can be more efficiently encoded. According to this method, the pitch period is calculated from an original signal, which serves as a reference signal, and then a basic pitch frequency (f0) is calculated. Thereafter, the spectrum of a target signal, which is a target of spectrum modification, is divided into a plurality of partitions. It is specified here that the width of each partition be the basic pitch frequency. Then, the spectra of bands are interleaved such that a plurality of peaks having similar amplitudes are unified into a group. The basic pitch frequency is used as an interleave pitch.
摘要:
A stereo signal generating apparatus capable of obtaining stereo signals that exhibit a low bit rate and an excellent reproducibility. In this stereo signal generating apparatus (90), an FT part (901) converts a monaural signal (M′t) of time domain to a monaural signal (M′) of frequency domain. A power spectrum calculating part (902) determines a power spectrum (PM′). A scaling ratio calculating part (904a) determines a scaling ratio (SL) for a left channel, while a scaling ratio calculating part (904b) determines a scaling ratio (SR) for a right channel. A multiplying part (905a) multiplies the monaural signal (M′) of frequency domain by the scaling ratio (SL) to produce a left channel signal (L″) of a stereo signal, while a multiplying part (905b) multiplies the monaural signal (M′) of frequency domain by the scaling ratio (SR) to produce a right channel signal (R″) of the stereo signal.
摘要:
Multichannel signal coding equipment is provided for presenting a high quality sound at a low bit rate. In the multichannel signal coding equipment (2), a down mix part (10) generates monaural reference channel signals for N number of channel signals. A coding part (11) codes the generated reference channel signal. A signal analyzing part (12) extracts parameters indicating characteristics of each of the N number of channel signals. An MUX part (13) multiplexes the coded reference channel signal with the extracted parameters.
摘要:
A stereo signal generating apparatus capable of obtaining stereo signals that exhibit a low bit rate and an excellent reproducibility. In this stereo signal generating apparatus (90), an FT part (901) converts a monaural signal (M′t) of time domain to a monaural signal (M′) of frequency domain. A power spectrum calculating part (902) determines a power spectrum (PM′). A scaling ratio calculating part (904a) determines a scaling ratio (SL) for a left channel, while a scaling ratio calculating part (904b) determines a scaling ratio (SR) for a right channel. A multiplying part (905a) multiplies the monaural signal (M′) of frequency domain by the scaling ratio (SL) to produce a left channel signal (L″) of a stereo signal, while a multiplying part (905b) multiplies themonaural signal (M′)of frequency domain by the scaling ratio (SR) to produce a right channel signal (R″) of the stereo signal.
摘要翻译:一种立体声信号发生装置,其能够获得表现出低比特率和优异的再现性的立体声信号。 在该立体声信号生成装置(90)中,FT部(901)将时域的单声道信号(M'SUB T< SUB)转换为频域的单声道信号(M')。 功率谱计算部(902)确定功率谱(P SUB M')。 缩放比例计算部分(904a)确定左通道的缩放比(S SUB> L),而缩放比例计算部分(904b)确定缩放比(S SUB> SUB>)。 乘法部分(905a)将频域的单声道信号(M')乘以缩放比(S L L L)以产生立体声信号的左声道信号(L“),而 乘法部分(905b)将频域的声场信号(M')乘以缩放比(S SUB R),以产生立体声信号的右声道信号(R“)。
摘要:
A method (100) and apparatus (200) are disclosed for transcribing a humming signal into a sequence of musical notes. The method begins by grouping (305) the signal into frames of data samples. Each frame is then processed to derive (320) a frequency distribution for each frames. The frequency distributions are processed to derive (410) a Harmonic Product Energy (HPE) distribution over the frames. The HPE distribution is then segmented (115, 120) to obtain boundaries of musical notes. The frequency distributions of the frames are also processed to derive (412) a fundamental frequency distribution. A pitch for each note is determined (125) from the fundamental frequency distribution.
摘要:
The conventional error conceal processing generates a greatly fluctuating irregular sound which is unpleasant to ears and causes a remarkable echo effect and click noise. A notification signal detection unit (301) judges processing for an input frame. In case of an error frame, a sound detection unit (303) makes judgment whether a preceding non-error data frame is a sound signal. If it is a sound frame, a sound copying unit (304) generates a replacing frame. If it is a non-sound frame, a transient signal detection unit (305) judges whether it is an attack signal by the transient signal detection and selects an appropriate area from the preceding non-error frame.