126results about "Analogue recording/reproducing" patented technology

A micro-actuator is comprised of a piezoelectric motor mounted on a flexure tongue with offsetting hinges, to perform a fine positioning of the magnetic read / write head. The substantial gain in the frequency response greatly improves the performance and accuracy of the track-follow control for fine positioning. The simplicity of the enhanced micro-actuator design results in a manufacturing efficiency that enables a high-volume, low-cost production. The micro-actuator is interposed between a flexure tongue and a slider to perform an active control of the fly height of the magnetic read / write head. The induced slider crown and camber are used to compensate for thermal expansion of the magnetic read / write head, which causes the slider to be displaced at an unintended fly height position relative to the surface of the magnetic recording disk. The enhanced micro-actuator design results in reduced altitude sensitivity, ABS tolerances, and reduced stiction. The controlled fly height of the magnetic read / write head prevents a possibility of a head crash, while improving the performance and data integrity.

Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

According to one embodiment, video image and audio signals are transmitted to a plurality of electronic devices each having at least one of a video image display function and an audio reproducing function. Information indicating the fact that at least one of a video image and audio has been set in a mute state is acquired from each of the electronic devices. Based on the information, a processing operation is applied to the video image and audio signals supplied to each of the electronic devices.

A playback apparatus used for an audio database includes a first acquisition unit configured to acquire first position information and second position information, an estimation unit configured to estimate a required period from a first position to a second position based on the first position information and the second position information, a second acquisition unit configured to acquire a data length of an audio signal to be played back from the audio database, a determination unit configured to determine a playback speed required to play back the audio signal based on the required period and the data length to complete playback of the audio signal within the required period, and a playback unit configured to play back the audio signal according to the playback speed.

A method of and an apparatus to restore high frequency of a moving picture experts group audio layer 3 (MP3) audio signal within a decoder. The method includes: setting modified discrete cosine transform (MDCT) coefficients of low bands and high bands of an audio signal, based on scale factor information of each band; extracting MDCT coefficients of low bands per band based on scale factors of each band after dequantizing inputted compressed audio bitstream; selecting the MDCT coefficients of the set low bands that corresponds to patterns of MDCT coefficients of low bands of the inputted compressed audio bitstream, and selecting the MDCT coefficients of the high bands that matches with the MDCT coefficients of the selected low bands; and performing an inverse MDCT by adding the MDCT coefficients of the selected high bands with the MDCT coefficients of the low bands.

A video signal copy guard system in which a special color burst signal (or "copy prohibition signal") is inserted into the video signal to be copy protected so that the protected signal includes both a special color burst signal and a normal color burst signal. The special color burst signal is placed adjacent to the normal color burst signal, and its phase, frequency, and / or amplitude is varied with respect to the normal color burst signal. The special color burst signal interferes with the normal color burst signal when an analog VCR attempts to process the protected signal. Thereby, preventing unauthorized dubbing of the protected signal.

The information reproduction device for reproducing contents information based on license information added to the contents information and required for reproducing the contents information, includes a determining section for determining whether encrypted contents information is the officially copied contents information or privately copied contents information based on the contents of the license information, and a reproduction section configured to reproduce the contents information determined as the privately copied contents information by the determining section under more severe restriction that the officially copied contents information. Thus, the contents copied by the proper route (officially copied contents) can be distributed in a more advantageous form than the contents copied without proper authorization.

According to one embodiment, a magnetic medium's readback signal samples are processed iteratively to provide a slip-resistant read channel by feeding the decoder output decisions back to the read channel front end where they are used to drive the decision-aided digital signal processing functions and control loops. Since data decisions provided by the decoder are typically more reliable than those provided by the detector, a significant performance improvement is obtained. A more reliable operation of the digital front-end signal processing functions in turn allows improvements to the reliability of the decoded data. Usage of Error Correcting Code (ECC) schemes that are soft decodable makes the read channel technique, described according to various embodiments herein, particularly efficient.

Clicks in voice and music signals are removed by detecting and interpolating the click. The use of a permanently set threshold value above the peak value of the audio signal is known for detecting the clicks, and interpolation carried out on the basis of the sample values before and after a click is known for filling them in. Disadvantages of these methods are, on the one hand, that many clicks do not have a larger amplitude than the peak signal and therefore remain undetected, and, on the other hand, interpolation is a highly complex procedure and is limited to a maximum click density. The invention provides for an error signal (en) to be determined by adaptive filtering (AF). A sample value is designated as distorted in the input signal (xn) if the absolute instantaneous value in the error signal (en) exceeds a dynamically adapted threshold value. The output signal of the adaptive filtering (yn) is used as a substitute value for the sample value designated as distorted, specifically both in the case of the signal which is output and in the input values of the adaptive filtering.

A four-channel high-capacity waveform storage system comprises an FPGA core chip, an FPGA core chip minimum peripheral circuit, ADC sampling chips, a Flash array, an internet port circuit and a power supply chip. 9A, 9B, 9C and 9D in the power supply chip convert a 5V power voltage into a 3.3V voltage, a 1.8V voltage, a 1.0V voltage and a 2.5V voltage respectively for other modules to use; after being normally powered on, the FPGA core chip controls the four paths of ADC sampling chips to collect waveform data from an external input interface and transmits the data to the Flash array; after storage, the FPGA core chip controls the internet port circuit to transmit the data in the Flash array to an upper computer through an internet port; all control instructions are transmitted to the FPGA core chip by the upper computer through a network interface. A construction method of the four-channel high-capacity waveform storage system includes seven steps. According to the four-channel high-capacity waveform storage system and the construction method, multiple paths of waveform data can be integrally and simultaneously collected, and implementation is simple; the logical functions are all written in a VHDL and easy to modify; the four-channel high-capacity waveform storage system is small in size and can be conveniently applied to multiple systems.

An apparatus for providing dynamic equalizer optimization is disclosed. The present invention solves the above-described problems by providing equalizer coefficient updates that converge towards the same solution as the direct method without having to first write a known pattern to the disk or requiring any prior knowledge of the data already written on the disk. The adaptive cosine function may be used to modify only a DFIR tap set, only the j and k parameters of a cosine equalizer or to modify both the tap set for a DFIR filter and the j and k parameters of the cosine equalizer. Another algorithm, such as the LMS algorithm, may be used to modify parameters not modified by the cosine algorithm.

Disclosed is an equalization apparatus and method based on audiogram. The equalization method includes the steps of a) outputting a reference sound having reference frequency and intensity for approximate assumption of auditory threshold, b) determining a start sound corresponding to an assumed value of the auditory threshold while calling a sound having intensity different from a prior sound at a sound source memory depending on whether a user has received an answering signal in response to the reference sound, c) outputting a first test sound whose intensity is controlled at a value set prior to the start sound, having the reference frequency, d) calling a test sound having intensity and frequency different from the prior sound at the sound source memory depending on whether the user has received the answering signal, e) storing data on reception of the answering signal of the user in response to the test sound, f) repeating the steps d) to e), g) determining the auditory threshold per previously set fiequency band using the reception data of the answering signal, and h) performing equalization depending on the auditory threshold per the determined frequency.

A method of setting an equalizer so as to enlarge a sound field in reproducing an audio file and a method of reproducing an audio file thereby, includes: dividing an input audio file into segments with a predetermined time length; extracting an audio feature value for each segment; determining equalizer information for reproducing each segment by the use of the extracted feature value; and determining an equalizer sequence for the audio file by the use of the determined equalizer information of each segment. Since the equalizer setting information can be automatically changed without user manipulation, the user can listen to an audio file of which the sound field is dynamically enlarged.

Dividing devices are provided for dividing an input information signal into a plurality of frequency bands, and a level detector is provided for detecting a level of a noise component of the information signal divided by the dividing device. A plurality of thresholds corresponding to the divided information signals are stored in a memory. One or more thresholds are selected from the threshold stored in the memory based on a detected result by the level detector. Attenuators are provided for comparing the information signals with a selected threshold and for attenuating an information signal the level of which is lower than the selected threshold.

An audio or other program signal is modified to provide protection against copying on a magnetic tape recorder. The signal from a source (10) is amplified (16) and then subjected to Fourier analysis (11) to determine the signal amplitude in a plurality of subbands across the audible range. The subband containing the component of greatest magnitude is selected on an adaptive basis and applied to a stored psycho-sensorial model (12), which determines the maximum amplitude of interference which can be masked by the selected subband. An anti-copy signal is generated (14) by complex division of the desired interference by a stored representation of the response of a tape recorder (13).

A lossless encoding apparatus encodes audio data and a lossless decoding apparatus restores the losslessly compression encoded audio data on a real-time basis, and a method therefor. The lossless encoding apparatus includes a lossless compression unit which losslessly compression encodes the audio data stored in an input buffer in units of predetermined data and outputs the encoded data in sequence, and an output buffer which stores the encoded audio data output from the lossless compression unit. A bitrate controller divides a plurality of the encoded audio data stored in the output buffer into first data having a data amount exceeding the maximum bitrate and second data having a data amount less than the maximum bitrate, divides the first data into third data being the encoded audio data having a data amount of the maximum bitrate and fourth data being the encoded data of the portion exceeding the maximum bitrate, and controls the output buffer so that the fourth data is output together with the second data.

An analog signal processor, accessing to a setting register through serial communication, for achieving high speed of selective bit setting therein, comprising: an address register 3, a data register 4, a mask register 5, and a AND-OR logic circuit 6, wherein address data “a”, setting data “d” and mask data “m” are transmitted through serial communication. Reading out a register designated by the address data “a” and conducting AND operation on the mask data “m” for each bit thereof, and further conducting OR operation upon the setting data “d” for each bit thereof, a result obtained thereby is written back into a register 8 designated by the address data. When no mask data portion is transmitted, the same process is conducted, assuming that it is the mask data of all bits being zero.

At a reserved recording time, if a television receiver is turned off, then the television receiver is turned on, and a video of a reservation setting target channel is forcibly displayed. A content of a reserved program can be thereby checked. At this time, a mute control signal is output to a voice processing circuit to mute a voice. If a user does not transmit a volume change instruction for a predetermined time, then a TV circuit is turned off, and a reservation recording is continued. This can suppress unnecessary power consumption during user's absence.

A system, device, and method for recording audio with the character and sonic benefits of a genuine analog recording is disclosed. More specifically, an electro-mechanical-software controlled closed loop analog signal processor (“CLASP”) system which is comprised of a CLASP unit containing firmware, a latency detection module, and CLASP hardware display and controls. The CLASP system further comprises CLASP software operably running on a digital audio workstation (“DAW”) which is also in operable communication with the CLASP unit. The CLASP unit is also in operable communication with an analog recordable medium. An analog audio signal is recorded on the analog recordable medium, which may consist of a coated tape, cup, cylinder, drum, or disk, and then immediately played back and routed to the DAW via an analog to digital converter, thus providing for digitally recorded analog audio. The CLASP system may also include converters and a mixing console.

An apparatus for providing dynamic equalizer optimization is disclosed. The present invention solves this by providing an equalizer coefficient update that converges towards the same solution as the direct method without having to first write a known pattern to the disc or requiring any prior knowledge of the data already written on the disc above question. Adaptive cosine functions can be used to modify only the DFIR tap settings, only the j and k parameters of the cosine equalizer, or both the tap settings of the DFIR filter and the j and k parameters of the cosine equalizer. Other algorithms, such as the LMS algorithm, can be used to modify parameters not modified by the cosine algorithm.

The present invention is a turntable which includes an internal analog to digital converter and a controller for a standard digital format. This allows a jack for a standard digital frat to be incorporated into the turntable, so that digital signals are generated to be received by the soundcard of an external computer or similar piece of digital equipment.