Hybrid linear predictive speech coding system with phase alignment predictive quantization zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

Hybrid speech coding and system

In a speech encoder/decoder (200/300) a pitch delay contour endpoint modifier (208) is employed to shift the endpoints of a pitch delay interpolation curve up or down. Particularly, the endpoints of the pitch delay interpolation curve are shifted based on a variation and/or a standard deviation in pitch delay.

Method and apparatus for coding an information signal using pitch delay contour adjustment

There is disclosed an audio decoding device capable of improving audio quality of a decoded signal by considering the energy change of a past signal in loss compensation processing. In this device, an energy change calculation unit (143) calculates an average energy of an audio source signal of one-pitch cycle from the end of the ACB vector outputted from an adaptive codebook (106). Moreover, the energy change calculation unit (143) calculates a ratio of the average energy of the current sub-frame and the sub-frame immediately before and outputs the ratio to an ACB gain generation unit (135). The ACB gain generation unit (135) outputs a conceal processing ACB gain defined by the ACB gain decoded in the past or information on the energy change ratio outputted from the energy change calculation unit (143), to a multiplier (132).

Audio decoding device and compensation frame generation method

There is provided a method of selecting a pitch lag value from a plurality of pitch lag candidates for coding a speech signal The method comprises identifying the plurality of pitch lag candidates from a frame of the speech signal using correlation; classifying the speech signal to obtain a voice classification; determining whether one or more of the plurality of pitch lag candidates are in a temporal neighborhood of one or more previous pitch lag values; favoring the one or more of the plurality of pitch lag candidates determined to be in the temporal neighborhood of the one or more previous pitch lag values, by adaptive weighting, over other ones of the plurality of pitch lag candidates; and selecting the pitch lag value based on the voice classification and the one or more of the plurality of pitch lag candidates favored by the adaptive weighting

Pitch determination based on weighting of pitch lag candidates

Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.

Audio signal processing method and device

A speech communication system and method that has an improved way of handling information lost during transmission from the encoder to the decoder. More specifically, the improved speech communication system more accurately recovers from losing information about a frame of speech such as line spectral frequencies (LSF's), pitch lag (or adaptive codebook excitation), fixed codebook excitation and/or gain information. After estimating lost parameters in a lost frame and synthesizing the speech, the improved system matches the energy of the synthesized speech to the energy of the previously received frame.

A speech communication system and method for handling lost frames

The invention relates to a method of reproducing decoded speech in a communication system comprising: receiving speech parameters including an adaptive codebook gain and a fixed codebook gain for each subframe on a frame-by-frame basis, making a periodical decision whether the speech is a periodic speech or a non-periodic speech using the received speech parameters, detecting whether a current frame of speech parameters is lost, making a decision (1000, 1030) whether the current lost frame is a first lost frame after a received frame or not a first lost frame after a received frame, setting (1004, 1008, 1010, 1020, 1022) a gain parameter for the current lost frame based on the periodical decision and on the decision whether the current lost frame is a first lost frame after a received frame or not a first lost frame after a received frame and using the gain parameter for the reproducing of the speech signal.

Eyal Shlomot

Papers

A speech communication system and method for handling lost frames

Voice communication system and method for processing drop-out fram

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