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:
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:
A decoding apparatus includes a unit decoding and inversely quantizing coded data to obtain frequency domain audio signal data, a unit computing from the coded data one of the number of scale bits composed of the number of bits corresponding to the scale value of the coded data and the number of spectrum bits composed of the number of bits corresponding to the spectrum value of the coded data, a unit estimating a quantization error of the frequency domain audio signal data based on one of the number of scale bits and the number of spectrum bits of the coded data, a unit computing a correction amount based on the estimated quantization error and correct the frequency domain audio signal data obtained by the frequency domain data obtaining unit based on the computed correction amount, and a unit converting the corrected frequency domain audio signal data into the audio signal.
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:
A decoding apparatus includes a unit decoding and inversely quantizing coded data to obtain frequency domain audio signal data, a unit computing from the coded data one of the number of scale bits composed of the number of bits corresponding to the scale value of the coded data and the number of spectrum bits composed of the number of bits corresponding to the spectrum value of the coded data, a unit estimating a quantization error of the frequency domain audio signal data based on one of the number of scale bits and the number of spectrum bits of the coded data, a unit computing a correction amount based on the estimated quantization error and correct the frequency domain audio signal data obtained by the frequency domain data obtaining unit based on the computed correction amount, and a unit converting the corrected frequency domain audio signal data into the audio signal.
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:
A code separation/decoding unit restores a vocal tract characteristic sp1 and a vocal source signal r1. A vocal tract characteristic modification unit modifies the vocal tract characteristic sp1 and outputs the modified vocal tract characteristic sp2. In this method, an emphasized vocal tract characteristic sp2 is generated to output by applying formant emphasis, using amplification ratios calculated based on estimated formants, directly to the vocal tract characteristic sp1 for instance. A signal synthesis unit synthesizes the modified vocal tract characteristic sp2 and the vocal source signal r1 to generate and output an output voice, s.
摘要:
It is so arranged that a voice code can be converted even between voice encoding schemes having different subframe lengths. A voice code conversion apparatus demultiplexes a plurality of code components (Lsp1, Lag1, Gain1, Cb1), which are necessary to reconstruct a voice signal, from voice code in a first voice encoding scheme, dequantizes the codes of each of the components and converts the dequantized values of code components other than an algebraic code component to code components (Lsp2, Lag2, Gp2) of a voice code in a second voice encoding scheme. Further, the voice code conversion apparatus reproduces voice from the dequantized values, dequantizes codes that have been converted to codes in the second voice encoding scheme, generates a target signal using the dequantized values and reproduced voice, inputs the target signal to an algebraic code converter and obtains an algebraic code (Cb2) in the second voice encoding scheme.
摘要:
A voice intensifier capable of reducing abrupt changes in the amplification factor between frames and realizing excellent sound quality with less noise feeling by dividing input voices into the sound source characteristic and the vocal tract characteristic, so as to individually intensify the sound source characteristic and the vocal tract characteristic and then synthesize them before being output. The voice intensifier comprises a signal separation unit for separating the input sound signal into the sound source characteristic and the vocal tract characteristic, a characteristic extraction unit for extracting characteristic information from the vocal tract characteristic, a corrective vocal tract characteristic calculation unit for obtaining vocal tract characteristic correction information from the vocal tract characteristic and the characteristic information, a vocal tract characteristic correction unit for correcting the vocal tract characteristic by using the vocal tract characteristic correction information, and a signal synthesizing means for synthesizing the corrective vocal tract characteristic from the vocal tract characteristic correction unit and the sound source characteristic, so that the sound synthesized by the signal synthesizing means is output.
摘要:
A data embedding device for embedding data in a speech code obtained by encoding a speech in accordance with a speech encoding method based on a voice generation process of a human being, includes an embedding judgment unit, every speech code, judging whether or not data should be embedded in the speech code, and an embedding unit embedding data in two or more parameter codes of a plurality of parameter codes constituting the speech code for which it is judged by the embedding judgment unit that the data should be embedded.