Abstract: A method for audio downmix apparatus with dynamic-range control includes steps as follows. The procedure first calculates the power of a surround audio channel and a central audio channel and then estimates whether a gain of the surround audio channel and the central audio channel saturate or not. When the gain is saturated, the procedure enters a step for adjusting a gain of dynamic range control; while the gain is not saturated, the procedure enters a saturation process. After the step of adjusting a gain of dynamic range control, then processes a gain calculation; and then changes the gain and estimates whether the gain reaches a default value or not. When the gain reaches the default value, then the gain will be outputted. When the gain does not reach the default value, continues to process the gain calculation and change the gain.

Abstract: The present invention provides an apparatus and a method for emulating audio. The apparatus comprises an echo device, a reverberation device, an audio source device and a first signal synthesizer. Wherein, the echo device receives an audio source signals and applies thereof for generating echo signal and feedback signal; the reverberation device receives the audio source signal and applies thereof in accordance with the feedback signal of the echo device for generating a reverberation signal; the audio source device receives the audio source signal and applies thereof for outputting a direct audio source signal; and the first signal synthesizer is used to receive and synthesize the echo signal, the reverberation signal and the direct audio source signal so as to output a first synthesized signal, which is an audio effect emulation signal.

Abstract: A multiple step adaptive method for time scaling. Synthesizing S3[n] signal from signal S1[n]signal and S2[n]signal. Comprising following steps: (a) calculating a first magnitude of a cross-correlation function of S1[n]signal and S2[n]signal according to a first index; (b) comparing the first magnitude with a threshold value; (c) if first magnitude is smaller than threshold value, calculating a first reference magnitude of cross-correlation function of S1[n]signal and S2[n]signal according to a first reference index behind the first index by a first determined number, or calculating a second reference magnitude of the cross-correlation function of the S1[n] signal and the S2[n] signal according to a second reference index behind the first index by a second number; (d) synthesizing the S3[n] signal by adding S1[n]signal to the S2[n] signal in accordance with a maximum index corresponding to a largest magnitude among all the magnitudes calculated in (c).

Abstract: A method of setting a transfer function of an adaptive filter. The adaptive filter is used for processing an audio signal. The method includes setting a first, a second, and a third pole, setting a first zero without resting upon the audio signal. A real part of the first zero is a negative value and an imaginary part of the first zero is a positive value. A second zero is set having a real part corresponding a positive value and an imaginary part corresponding to a negative value and a third zero is set according to a key shifting associated with the audio signal.

Abstract: A nonlinear overlap method for time scaling to synthesize an S1[n] and an S2[n] into an S3[n] is disclosed. The S1[n] and the S2[n] having N1 and N2 signals respectively. The nonlinear overlap method includes the following steps: (a) delaying the S2[n] by a predetermined number and forming an S5[n], (b) establishing a correlogram of a cross-correlation function of the S1[n] and S5[n], and (c) setting S3[n] as a number of S1[n] when 0<=n<; as a number formed by overlap-adding the S1[n] and an S4[n] in a weighting manner when (the predetermined number+the maximum index+the first threshold)<=n<(N1?a second threshold); and as a number of S4 wherein the first and second thresholds are not equal to zero at the same time, and the S4[n] is formed by delaying the S5[n] by the maximum index.

Abstract: The present invention is related to a method and an apparatus for processing audio signals, which uses two speakers to simulate audio sounds with wide and 3D effects. The method provides a structure for processing audio signals by a plurality of analog elements (filter, gain amplifier, delay processing unit). The present invention also improves the bass part and strengthens 3D effects of the audio sounds. Therefore, the audio sounds output from the audio devices are more natural and smooth.

Abstract: A method for audio downmix apparatus with dynamic-range control includes steps as follows. The procedure first calculates the power of a surround audio channel and a central audio channel and then estimates whether a gain of the surround audio channel and the central audio channel saturate or not. When the gain is saturated, the procedure enters a step for adjusting a gain of dynamic range control; while the gain is not saturated, the procedure enters a saturation process. After the step of adjusting a gain of dynamic range control, then processes a gain calculation; and then changes the gain and estimates whether the gain reaches a default value or not. When the gain reaches the default value, then the gain will be outputted. When the gain does not reach the default value, continues to process the gain calculation and change the gain.

Abstract: The present invention provides an apparatus and a method for emulating audio. The apparatus comprises an echo device, a reverberation device, an audio source device and a first signal synthesizer. Wherein, the echo device receives an audio source signals and applies thereof for generating echo signal and feedback signal; the reverberation device receives the audio source signal and applies thereof in accordance with the feedback signal of the echo device for generating a reverberation signal; the audio source device receives the audio source signal and applies thereof for outputting a direct audio source signal; and the first signal synthesizer is used to receive and synthesize the echo signal, the reverberation signal and the direct audio source signal so as to output a first synthesized signal, which is an audio effect emulation signal.

Abstract: A method of setting a transfer function of an adaptive filter. The adaptive filter is used for processing an audio signal. The method includes setting a first, a second, and a third pole, setting a first zero without resting upon the audio signal. A real part of the first zero is a negative value and an imaginary part of the first zero is a positive value. A second zero is set having a real part corresponding a positive value and an imaginary part corresponding to a negative value and a third zero is set according to a key shifting associated with the audio signal.

Abstract: A multiple step adaptive method for time scaling. Synthesizing S3[n] signal from signal S1[n]signal and S2[n]signal. Comprising following steps: (a) calculating a first magnitude of a cross-correlation function of S1[n]signal and S2[n]signal according to a first index; (b) comparing the first magnitude with a threshold value; (c) if first magnitude is smaller than threshold value, calculating a first reference magnitude of cross-correlation function of S1[n]signal and S2[n]signal according to a first reference index behind the first index by a first determined number, or calculating a second reference magnitude of the cross-correlation function of the S1[n] signal and the S2[n] signal according to a second reference index behind the first index by a second number; (d) synthesizing the S3[n] signal by adding S1[n]signal to the S2[n] signal in accordance with a maximum index corresponding to a largest magnitude among all the magnitudes calculated in (c).

Abstract: A nonlinear overlap method for time scaling to synthesize an S1[n] and an S2[n] into an S3[n] is disclosed. The S1[n] and the S2[n] having N1 and N2 signals respectively. The nonlinear overlap method includes the following steps: (a) delaying the S2[n] by a predetermined number and forming an S5[n], (b) establishing a correlogram of a cross-correlation function of the S1[n] and S5[n], and (c) setting S3[n] as a number of S1[n] when 0<=n<; as a number formed by overlap-adding the S1[n] and an S4[n] in a weighting manner when (the predetermined number+the maximum index+the first threshold)<=n<(N1?a second threshold); and as a number of S4 wherein the first and second thresholds are not equal to zero at the same time, and the S4[n] is formed by delaying the S5[n] by the maximum index.

Abstract: A method for generating output sound data in a predetermined time period includes mixing input sound data in the predetermined time period with input sound data in the time period previous to the predetermined time period, and with output sound data in the time period previous to the predetermined time period in order to generate the output sound data in the predetermined time period.

Abstract: A method for calculating the pitch estimation of speech signals. The method includes the following steps: (a) Providing an initial value to a lag parameter, (b) Calculating the autocorrelation values according to the lag parameters corresponding to the autocorrelation values, (c) Storing the lag parameter and the autocorrelation values corresponding to the lag parameters in a memory, (d) Determining a first increment value and a second increment value, (e) Comparing the autocorrelation values and the first threshold value in the step (b), (f) Repeat the steps (b), (c), (d) and (e), (g) Comparing the plurality of the autocorrelation values stored in the memory and finding out the maximum autocorrelation values, and calculating the pitch estimation with the lag parameter corresponding to the maximum autocorrelation value.

Abstract: A method and related apparatus for determined whether a voice signal is mixed with a vocal signal. When applying to a multi-channel system, the method includes: counting number of zero-crossings of a sound signal of each channel within a given period; if the zero-crossing number of the sound signal of a first channel are lower than those of the sound signal of a second channel by a predetermined threshold, determining that the sound signal of the first channel is mixed with a vocal signal.