acoustic coupler

Waterfall Flow All-sen beautiful anime large collection- lazy load + waterfall stream Width fixed so as to do waterfall flow and then with absolute positioning, the first row of the top is equal to 0

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The AEC echo suppression algorithm is difficult. Currently, directsound can be used for processing, but it can only be used in xp. It is not supported by other systems!Currently, gips has excellent implementations of this algorithm. Skype is the

In speech recognition, in order to enhance the robustness of audio features, it is necessary to extract the characteristic vectors with strong distinguishing ability, and the common method is PCA and LDA algorithm. The PCA algorithm seeks to

divided into two signals through the splitter, which goes directly into the R receiver and the other way through the switch input to the corresponding test port, so the R receiver is tested to get the input signal information.When the excitation signal is input to the test piece, the reflected signal and the input excitation signal are transmitted on the same physical path, and the directional coupler is responsible for separating the signal transmit

cards, video compression cards, video decompression cards, and video transfer cards. 36. Video Capture card function: video capture card continuously converts video signals into digital video data stored by computers. 37. Image digitalization The spatial distribution and brightness (color) Distribution of image information are continuous, and the continuous information must be discretized. The process of discretization is the process of digital processing, it should include the discretization o

Speech Recognition overview Speech recognition refers to the process of converting speech signals into words. The current speech recognition system framework is as follows:The signal processing module will be based on the auditory perception of the human ear to extract the most important features of speech, the speech signal into a feature vector sequence. The common acoustic features in current speech recognition systems are linear predictive coding

of the touchscreen, and the current flowing through the four electrodes is proportional to the distance between the fingers and the four corners, and the controller obtains the position of the touch point by accurately calculating the four current ratios.2.2 Defects of Capacitive touch screenCapacitive touch screen transmittance and clarity is better than four-wire resistive screen, of course, and can not be compared with the surface acoustic wave sc

. The model of speech recognition system is usually composed of two parts: acoustic model and language model, which correspond to the calculation of phonetic to syllable probability and syllable-to-word probability. Sphinx is a large vocabulary, non-specific person, continuous English speech recognition system developed by Carnegie Mellon University in the United States. A continuous speech recognition system can be broadly divided into four parts:

the switching status of the waveguide oriented Coupler on the LiNbO substrate to select different delay paths, thus forming a 6-Bit 0 ~ 63. A total of 64 variable bidirectional delay units with different latencies. As shown in System 2, the delay unit is composed of two LiNbO3 substrates, each of which has two 4x4 oriented Coupler Optical Switches, each of which is controlled by one bias electrode and one

gateway one end of the PSTN connected to the other end of the Internet. 　　 Although the circuit echoes are generated in the PSTN, they also transmit the IP telephony gateway, which is one of the echo sources in the Internet voice transmission, and the second echo source in the Internet voice transmission is the so-called "acoustic echo". Acoustic echo is the sound of the Speaker's voice is picked up by the

the IP Phone gateway is connected to the PSTN and the other end is connected to the Internet. Although circuit ECHO is generated in the PSTN, it is also transmitted to the IP Phone gateway, which is one of the ECHO sources in Internet voice transmission, the second echo source in Internet voice transmission is the so-called "acoustic echo ". Acoustic Echo means that the sound played by the speaker is pick

telephone systems, there is a so-called "circuit counterattack ". The main original ECHO is a 2-4 Line conversion in the system. The mixer that completes the 2-4 conversion results in "leakage" due to impedance matching, resulting in "circuit echo ". From the connection method of the Internet IP Phone gateway, we can see that one end of the IP Phone gateway is connected to the PSTN and the other end is connected to the Internet.Although circuit ECHO is generated in the PSTN, it is also transmit

0 mean value based on the spoken statement of the unit variance. However, it is not recommended to use this method, but instead use the model-based mean and variance normalization, such as linear vtln (LVTLN). You can use a small, phoneme-based language model for fast normalization. The feature extraction Code compute-mfcc-feats.cc/compute-plp-feats.cc also provides the –substract-mean option to obtain a 0 mean feature. If you want to get a normalized feature based on speaker or sentence-based

the need to increase. Previously, the commonly used phoneme synthesis techniques should include: Co-vibration peak combination, LPC combination, psola combination and LMA acoustic pattern combination. They each have their own advantages and disadvantages. In the process of application, people will combine multiple techniques with machines, or use the advantages of one technique to another, to take advantage of the disadvantages of another technique.J

the mystery of echo cancellation technology. Second, the principle of echo cancellation from the source of communication echoes, can be divided into acoustic echo (acoustic echo) and line echo (lines echo), the corresponding echo cancellation technology is called acoustic echo cancellation (acoustic echo cancellation,

From the reason of communication echo, can be divided into acoustic echo (acoustic echo) and line echo (lines echo), the corresponding echo cancellation technology is called acoustic echo cancellation (acoustic echo cancellation, AEC) and line echo cancellation (lines Echo cancellation, LEC).

Address: http://www.eefocus.com/jjbearustc/blog/07-09/3460_8d0d0.html#articletop MP3 encoding mainly consists of three functional modules,It includes a hybrid filter group (sub-band filter and mdct), a psychological acoustic model, and a quantization code (bit and bit Factor Allocation and Harman encoding ). 1. hybrid filter group. This part includes the sub-band filter group and mdct. The sub-band filter group code maps the sample signal from the tim

network terminal ONT, WDM coupler and 1 × N filter ). The Optical Fiber Device Used for the optical fiber access system must be able to test all three wavelengths. Optical fiber network operators and testing equipment manufacturers are quite familiar with the 1310nm and 1550nm wavelengths, but the emergence of 1490nm wavelengths poses a new challenge to OTDR. The. ITU G.983.3 optical access system includes optical line terminals, optical network ter