Tencent Technology (shenzhen) Company Limited

The Chinese University of Hong Kong

Xiaoping Wu

Liang Wu

Shiyuan Zheng

Qiang Dai

Yifu Li

Yihui Wang

A blocking signal cancellation low noise amplifier system includes a first low noise amplifier, a second low noise amplifier, a blocking signal extraction and bias generation circuit, a bias switching circuit, and a bias switching signal generating circuit. The first low noise amplifier is used for dynamic input matching, and the first low noise amplifier receives an input signal and outputs it after amplifying. The blocking signal extraction and bias generation circuit is used to extract a blocking signal from the output signal of the first low noise amplifier, and output a DC voltage signal. The bias switching circuit is used to switch the first low noise amplifier between a blocking mode and a small signal mode. The bias switching signal generating circuit is used to compare the DC bias voltage signal Vwith a preset reference voltage signal V.

Blocking signal cancellation low noise amplifier system

A push-push frequency doubler based on complementary transistors is provided. The first differential amplifier circuit receives a differential input signal having an initial frequency, and amplifies the amplitude of the second harmonic of the differential input signal to obtain a first signal. The second differential amplifier circuit receives the differential input signal with the initial frequency and amplifies the amplitude of the second harmonic of the differential input signal to obtain the second signal. Where, the first signal and the second signal are a set of differential signals with the same amplitude and a phase difference of 180&#xb0;. The output load circuit extracts the second harmonic signal in the first and second signal respectively to obtain and output a pair of differential output signal with first output frequency whose value is twice of the initial frequency. As a result, the frequency doubler with differential output signal is realized.

Push-push frequency doubling scheme and circuit based on complementary transistors

Method for phonetically annotating text is performed at a computing device. The method includes: identifying a first polyphonic word segment in a text input, the first polyphonic word segment having at least a first pronunciation and a second pronunciation; determining at least a first probability for the first pronunciation and a second probability for the second pronunciation; determining a predetermined threshold difference based on: a comparison of the first and second probabilities with a preset threshold probability value, respectively, and a magnitude of a difference between the first and second probabilities; comparing the difference between the first probability and the second probability with the predetermined threshold difference; and selecting the first pronunciation as a current pronunciation for the first polyphonic word segment in accordance with a determination that the difference between the first probability and the second probability exceeds the predetermined threshold difference.

Method and apparatus for phonetically annotating text

Method and device of selecting training parameters for training a model are disclosed. The method includes: (1) setting a precision requirement for the model, and a first parameter value interval defined by an upper limit and a lower limit; (2) obtaining a first value point and a second value point within the first parameter value interval; (3) obtaining and comparing respective first and second error rates by respectively setting the training parameter at the first and second value points for the model; (4) updating three values out of the upper limit, the lower limit, the first value point and the second value point; (5) repeating steps (3) and (4), until the precision requirement is net by the respective first and second value points; and (6) obtaining the optimal value of the training parameter.

Method and apparatus for acquiring training parameters to calibrate a model

Method and device of processing audio signals are disclosed. The method includes: obtaining a set of data, the set of data comprising LSP parameters for an audio signal; determining a set of sampling data points from the set of LSP parameters using a predetermined sampling rule, the set of sampling data points including spectrum amplitude values for a plurality of sampled frequency values; identifying one or more local maxima among the set of sampling data points, and a respective preceding local minimum and a respective succeeding local minimum for each of the identified local maxima; for each of the identified local maxima, shifting one or more of the set of data comprising LSP parameters located between the respective preceding local minimum and the respective succeeding local minimum of an identified local maximum towards the identified local maximum; and adjusting the set of data comprising LSP parameters using an energy coefficient.

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Shenzhen, China

Xiaoping Wu