48 results about "Pre-echo" patented technology

A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.

A downlink activity and double talk probability detector and method for an echo canceler circuit (10) improves the stability of an echo canceler adaptive filter (300) and improves the attenuation of post-echo canceler uplink data (388). The echo canceler circuit (10) includes a downlink activity and double talk probability data generator (30) and an echo canceler stage (20). The downlink activity and double talk probability data generator (30) receives pre-echo canceler uplink data (40) and downlink data (50) and in response produces double talk probability data (60) and downlink activity data (70). The echo canceler stage (20) receives the downlink data (50), the pre-echo canceler uplink data (40), the double talk probability data (60) and the downlink activity data (70), and in response produces uplink data (80).

Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding / decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections. By applying a corresponding switching control, pre-echo effects are avoided and a high coding gain and a low coding delay are achieved.

Provided is a highband coding apparatus and method for a wideband coding system. The coding apparatus and method can reduce a pre-echo phenomenon by encoding the highband based on lowband encoding information and Temporal Noise Shaping technique. A highband encoding apparatus includes: a domain converter for converting the domain of an input highband signal into a frequency domain; a linear prediction order determiner for determining a linear prediction order based on the lowband encoding information; a linear prediction analyzer for analyzing a highband signal of the frequency domain based on the determined linear prediction order to thereby generate a linear prediction coefficient; a linear prediction coefficient quantizer for quantizing the linear prediction coefficient based on the lowband encoding information; and a residual signal quantizer for obtaining a residual signal by dequantizing the quantized linear prediction coefficient and quantizing the residual signal.

A double talk activity detector (30) and method for an echo canceler circuit (10) improves the probability of detecting a double talk condition based on at least pre-echo canceler uplink data (40). The echo canceler circuit (10) includes a double talk activity probability data generator (30) and an echo canceler stage (20). The double talk activity probability data generator (30) receives pre-echo canceler uplink data (40) and in response produces double talk activity probability data (50). The echo canceler stage (20) is coupled to the double talk activity probability data generator (30) and receives downlink data (60), pre-echo canceler uplink data (40) and the double talk activity probability data (50) and in response produces attenuated uplink data (70).

A double talk activity detector (30) and method for an echo canceler circuit (10) improves the probability of detecting a double talk condition based on at least pre-echo canceler uplink data (40). The echo canceler circuit (10) includes a double talk activity probability data generator (30) and an echo canceler stage (20). The double talk activity probability data generator (30) receives pre-echo canceler uplink data (40) and in response produces double talk activity probability data (50). The echo canceler stage (20) is coupled to the double talk activity probability data generator (30) and receives downlink data (60), pre-echo canceler uplink data (40) and the double talk activity probability data (50) and in response produces attenuated uplink data (70).

An echo canceler circuit (10) and method attenuates at least post-echo canceler uplink data (90) to produce attenuated uplink data (100) in response to uplink echo return loss based attenuation data (40). The echo canceler circuit (10) includes an echo return loss based attenuation data generator (20) and at least an uplink data attenuator (30). The echo return loss based attenuation data generator (20) produces the uplink echo return loss based attenuation data (40) in response to echo return loss data (70). The echo return loss data (70) is based on at least one of: attenuated downlink data (50), pre-echo canceler uplink data (60), and / or amplifier gain data (80). The uplink data attenuator (30) attenuates the post-echo canceler uplink data (90) to produce attenuated uplink data (100) based on the uplink echo return loss based attenuation data (40).

A method and an apparatus are provided in an OFDM receiver for detecting and compensating for long echo. The method comprises a first pilot tone interpolation mechanism and a first window placement to filter a received OFDM symbol, a long echo channel detection coupled with a second pilot tone interpolation mechanism, a pre-echo and post-echo detection wherein the pre-echo condition is associated with a second new window placement, and both pre-echo and post-echo conditions place two time windows around a first peak channel response and a second peak channel response for channel estimation. The long echo is estimated by obtaining power spectra of a subset of subcarriers in one OFDM symbol, performing an inverse Fourier transform on the power spectra and determining the long echo by measuring the time between two peaks in the power profile.

The invention provides a weather radar image processing method and a weather radar image processing system. The method includes receiving echo images of a weather radar, inputting the echo images intoa full-convolution neural network, classifying pixel points of the echo images, and acquiring predicated images of the pixel points marked with true echoes, wherein the full-convolution neural network comprises a deconvolution layer and a softmax layer. The method has the advantages that through the full-convolution neural network, the echo images of the weather radar are classified and marked, so that the true echoes and clutters in the echo images of the weather radar can be recognized effectively, and processing efficiency and recognition accuracy of weather radar images are improved.

An adaptive equalizer for amplitude-modulation (AM) signal received by over-the-air radio transmission comprises feed-forward finite-impulse-response (FIR) filtering and further comprises infinite-impulse-response (IIR) filtering for feeding back decisions. The echo content of baseband signal demodulated from the AM signal is measured for initially determining the reception channel impulse response (CIR) in the time-domain. Thereafter, the CIR is updated by decision feedback. Periodically, the CIR is normalized with respect to the strength of a cursor component thereof. The strengths of longer-delay post-echoes in the normalized CIR are used to determine the weighting coefficients of the IIR filtering, which suppresses the longer-delay post-echoes. The pre-echo and short-delay post-echo portion of the normalized CIR is convolved with the normalized CIR to generate a synthetic normalized CIR. The strengths of pre-echoes and short-delay post-echoes in the synthetic normalized CIR are used to determine the weighting coefficients of the feed-forward FIR filtering, which suppresses the pre-echoes and short-delay post-echoes.

A signal processing system is disclosed. The signal processing system comprises an AGC and pre-echo cancellation system for receiving an analog signal, boosting it up (over all frequencies) to a pre-determined range by AGC, and removing the immediate transmit pulse from this received signal by pre-echo canceller to provide a second analog signal. The signal processing system further comprises a summer for receiving the analog signal; a feed forward equalization (FFE) unit for receiving a signal from the summer; and a slicer for receiving a signal from the FFE unit and providing an output signal. The signal processing system also comprises an Echo and NEXT cancellation system for receiving the output signal and for providing a signal to the summer for canceling the echo and crosstalk in the signal processing system. Accordingly, in a system and method in accordance with the present invention the Echo and crosstalk components associated with a signal processing system can be subtracted prior to the FFE. This allows for a practical analog implementation which has a good dynamic range of the signal through the FFE and subsequent analog elements. This system can be utilized to great advantage in low power CMOS designs where supply voltages are limited.

An echo canceler circuit (200) and method performs cascaded echo cancellation and noise suppression in a non-interfering manner. The echo canceler circuit (200) includes pre-noise suppression logic (210), echo canceler coefficient logic (218), noise suppression logic (212) and an echo canceler filter (216). The pre-noise suppression logic (210) receives pre-echo canceler uplink data (64) and downlink data (52), and in response produces pre-noise suppression uplink data (224). The echo canceler coefficient logic(218) receives the pre-noise suppression uplink data (224) and the pre-echo canceler uplink data (64), and in response produces filter coefficient data (226). The noise suppression logic (212) receives the pre-noise suppression uplink data (224), and in response produces noise suppressed uplink data (228). The echo canceler filter (216) receives the noise suppressed uplink data (228) and the filter coefficient data (226) and in response produces final uplink data (230).

A method is provided for processing pre-echo attenuation in a digital audio signal generated from a transform coding, wherein, at the decoding point, the method includes: detection of a position of attack in the decoded signal; determination of a pre-echo region preceding the position of attack detected in the decoded signal; calculation of attenuation factors per sub-block of the pre-echo region, according to at least the frame wherein the attack has been detected and the preceding frame; and pre-echo attenuation in the sub-blocks of the pre-echo region by the corresponding damping factors. The method also includes application of a filter for the spectral shaping of the pre-echo region on the current frame up to the detected position of the attack. A device and a decoder including the device are also proved for implementing the method.

The invention discloses an objective assessment method for audio quality of a compressed domain based on distortion correction, and belongs to the communication field. The objective assessment method is characterized by comprising the following steps of: extracting characteristic parameters including code rate, scale factor, pre-echo distortion, frequency distortion and speech coding distortion from a compressed audio stream; performing basic assessment on the audio quality according to the code rate and the scale factor, and giving an initial objective assessment score; and judging whether distortions exist according to the characteristic parameters of the pre-echo distortion, the frequency distortion and the speech coding distortion, and correcting the objective assessment score when the distortions exist so as to obtain a final objective assessment score. According to the objective assessment method disclosed by the invention, factors which influence the audio quality are taken full consideration, so that the objective assessment method can be used for simulating a subjective assessment result in a better manner.

The invention discloses a method for inhibiting pre-echoes of audio transient signals by utilizing frequency domain filtering post-processing, which belongs to the field of audio signal processing and in particular relates to a post-processing method for carrying out noise shaping on the decoded transient signals in audio coding. The method comprises the following steps: acquiring frequency domain linear prediction coefficients through discrete cosine transform coefficients of the input audio transient signals, and acquiring a short-time post-filter according to the frequency domain linear prediction coefficients; sequentially carrying out short-time post-filtering and spectral tilt compensation filtering on the discrete cosine transform coefficients of the transient signals; carrying out inverse discrete cosine transform on the frequency domain transform coefficients after filtering to restore and acquire time domain signals; and then, carrying out gain adjustment to acquire the transient signals after post-processing. By carrying out filtering processing on the frequency domain of the audio transient signals, the method achieves the noise shaping effect on the time domain, effectively inhibits the pre-echo distortion caused by transient signal coding, enhances the peak energy of the transient signals, and can improve the audio quality of the decoded audio transient signals without consuming additional coding bits.

To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.

A method is provided for processing attenuation of pre-echo in a digital audio signal decoded by transform decoding. The method includes the following acts: decomposition of the decoded signal into at least two sub-signals according to a pre-determined decomposition criterion; calculation of attenuation factors per sub-signal and per sample of a previously determined pre-echo zone; attenuation of pre-echo in the pre-echo zone of each of the sub-signals by applying attenuation factors to the sub-signals; and production of the attenuated signal by addition of the attenuated sub-signals. Also provided are a processing device implementing the acts of the described method, and a decoder including such a device.

In one aspect, a noise suppression process for a decoded signal comprising a first decoded signal portion and a second decoded signal portion is provided. A first energy envelope generating curve and a second energy envelope generating curve of the first signal portion and of the second decoded signal portion are determined. An identification number depending on a comparison of the first and second energy envelope generating curves is formed. An amplification factor which depends on the identification number is derived. Multiplying the second decoded signal portion by the amplification factor, reduces pre-echo and post-echo interference noises.

The invention discloses a self-adaptive window switching method based on the TDA domain, belongs to the field of audio encoding, and particularly relates to a processing method used after performing time domain aliasing (TDA) on audio transient signals. The method is characterized in that detected time domain transient signals are mapped to the TDA domain after time domain aliasing, time-frequency analysis of variable window length can be executed on TDA domain signals by aid of a self-adaptive window switching method, different window switching modes can be selected according to TDA domain transient position and time-frequency analysis effect so as to realize switching and transiting of sub-frames with different lengths, and then a window switching mode of the TDA domain is selected by adopting a strategy of combining an open loop and a closed loop. By aid of the method, self-adaptive window switching aiming at TDA domain signals can be achieved to select the optimum time resolution, the pre-echoes distortion of the transient signals is avoided, and simultaneously fine time-frequency analysis results can be kept, thereby being favorable for increasing audio encoding quality on complex conditions such as mixing of transient and steady and the like.

There is provided a coding method which can effectively prevent a pre-echo and a post-echo from being generated and can perform effective coding to which an psycho-acoustic model is applied. A coding apparatus according to the coding method of the present invention detects the attack and release portions of a waveform signal, and performs gain control to a waveform signal before the attack portion and the waveform signal of the release portion by using a gain control amount adaptively calculated according to the characteristics of the waveform signal. A quantization precision determination circuit determines a quantization precision by using the masking level. An window circuit and a transform circuit transform the waveform signal into a plurality of frequency components. The quantization circuit quantizes the plurality of frequency components by using the quantization precision.

A method for discriminating and attenuating pre-echo in a digital audio signal and generated from transform coding. The method includes the following acts in which, for a current frame broken down into sub-blocks, the low-energy sub-blocks precede a sub-block in which a transition or attack is detected, and determine a pre-echo area in which a pre-echo attenuation process is carried out. In the event that an attack is detected from the sub-block of the current frame, the method includes: calculating an energy leading coefficient for at least two sub-blocks of the current frame preceding the sub-block in which an attack is detected; comparing the leading coefficient to a predefined threshold; and inhibiting the pre-echo attenuation process in the pre-echo area in the event that the calculated leading coefficient is lower than the predefined threshold. Also provided are a discrimination and attenuation device implementing the acts of the method described and a decoder including such a device.