Abstract:
A method for speech processing includes determining a first unvoicing parameter for a first subframe of a speech signal, and determining a smoothed unvoicing parameter for the first subframe according to a second unvoicing parameter of a second subframe prior to the first subframe of the speech signal. The first unvoicing parameter is determined according to a periodicity parameter and a spectral tilt parameter. The method further includes computing a difference between the first unvoicing parameter for the first subframe and the smoothed unvoicing parameter for the first subframe and determining a classification of the first subframe using the computed difference as a decision parameter. The classification indicates whether the first subframe is an unvoiced speech signal or not an unvoiced speech signal. Bandwidth extension is performed on the speech signal for the first subframe according to the classification of the first subframe.
Abstract:
A method for speech processing includes determining an unvoicing parameter for a first frame of a speech signal and determining a smoothed unvoicing parameter for the first frame by weighting the unvoicing parameter of the first frame and a smoothed unvoicing parameter of a second frame. The unvoicing parameter reflects a speech characteristic of the first frame. The smoothed unvoicing parameter of the second frame is weighted less heavily when the smoothed unvoicing parameter of the second frame is greater than the unvoicing parameter of the first frame. The method further includes computing a difference, by a processor, between the unvoicing parameter of the first frame and the smoothed unvoicing parameter of the first frame, and determining a classification of the first frame according to the computed difference. The classification includes unvoiced speech or voiced speech. The first frame is processed in accordance with the classification of the first frame.
Abstract:
A mobile device with the NFC function includes an NFC chip, multiple SIM card slots, a power supply unit, and an eSE integrated into the NFC chip. One SIM card slot is connected to a first power port on the NFC chip. The power supply unit is connected to a second power port on the NFC chip. When the mobile device performs near field communication, the second power port on the NFC chip is triggered to output a first level signal. Each of the rest SIM card slots is connected to the power supply unit. The eSE is connected to the power supply unit. The power supply unit is configured to supply power to the eSE and the SIM card slot that is connected to the power supply unit, when the first level signal is received.
Abstract:
A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.
Abstract:
A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.
Abstract:
In accordance with an embodiment of the present invention, a method of speech processing included receiving a coded audio signal having coding noise. The method further includes generating a decoded audio signal from the coded audio signal, and determining a pitch corresponding to the fundamental frequency of the audio signal. The method also includes determining the minimum allowable pitch and determining if the pitch of the audio signal is less than the minimum allowable pitch. If the pitch of the audio signal is less than the minimum allowable pitch, applying an adaptive high pass filter on the decoded audio signal to lower the coding noise at frequencies below the fundamental frequency.
Abstract:
A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.
Abstract:
In accordance with an embodiment of the present invention, a method of speech processing included receiving a coded audio signal having coding noise. The method further includes generating a decoded audio signal from the coded audio signal, and determining a pitch corresponding to the fundamental frequency of the audio signal. The method also includes determining the minimum allowable pitch and determining if the pitch of the audio signal is less than the minimum allowable pitch. If the pitch of the audio signal is less than the minimum allowable pitch, applying an adaptive high pass filter on the decoded audio signal to lower the coding noise at frequencies below the fundamental frequency.
Abstract:
In an embodiment, a method of receiving a digital audio signal, using a processor, includes generating a high band time domain signal; generating low band time domain signal; estimating an energy ratio between the high band and the low band from a last good frame; keeping the energy ratio for following frame-erased frames by applying an energy correction scaling gain to a high band signal segment by segment in the time domain; and combining the low band signal and the high band signal into a final output.
Abstract:
MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis.