Abstract: For protecting a speaker, an input signal is received, and an excursion of the speaker that would be caused by the input signal is predicted. In response to the predicted excursion exceeding a threshold, a targeted excursion of the speaker is determined by compressing the predicted excursion. The targeted excursion is translated into an output signal, which is output to the speaker.

Abstract: An analog-to-digital converter (ADC) is provided. The ADC includes a plurality of pipelined ADCs and an adjustment circuit. Each pipelined ADC is adapted to receive an analog input signal, has an adjustable transfer function, and includes a compensator. The adjustment circuit is coupled to each pipelined ADC to be able to adjust the transfer function for each pipelined ADC so as to generally eliminate an estimation ambiguity. Additionally, the adjustment circuit estimates an inter-stage error that includes at least one of an inter-stage gain error and a DAC gain error and adjusts the compensator for each pipelined ADC to compensate for the inter-stage error.

Abstract: A method for converting a sampled analog signal into digital is provided. An input signal is sampled at a sampling instant to generate a sample voltage. A first current is then applied to a node to change a voltage on the node, and a first interval to change the voltage on the node to a reference voltage from the sample voltage using the first current is determined. A second current is then applied to the node to change a voltage on the node prior to a subsequent sampling instant, and a determination of a second interval to change the voltage on the node to the reference voltage from the sample voltage using the second current is made.

Abstract: Compressive sensing is an emerging field that attempts to prevent the losses associated with data compression and improve efficiency overall, and compressive sensing looks to perform the compression before or during capture, before energy is wasted. Here, several analog-to-digital converter (ADC) architectures are provided to perform compressive sensing. Each of these new architectures selects resolutions for each sample substantially at random and adjusts the sampling rate as a function of these selected resolutions.

Abstract: For protecting a speaker, an input signal is received and filtered into component signals. A sum of the component signals is approximately equal to the input signal. The component signals include at least first and second component signals. A perceived loudness to a human from the speaker is more sensitive to the first component signal than to the second component signal. A temperature of the speaker is estimated. In response to the estimated temperature, the second component signal is scaled. An output signal is output to the speaker in response to the first component signal and the scaled second component signal.

Abstract: A method for converting a sampled analog signal into digital is provided. An input signal is sampled at a sampling instant to generate a sample voltage. A first current is then applied to a node to change a voltage on the node, and a first interval to change the voltage on the node to a reference voltage from the sample voltage using the first current is determined. A second current is then applied to the node to change a voltage on the node prior to a subsequent sampling instant, and a determination of a second interval to change the voltage on the node to the reference voltage from the sample voltage using the second current is made.

Abstract: Compressive sensing is an emerging field that attempts to prevent the losses associated with data compression and improve efficiency overall, and compressive sensing looks to perform the compression before or during capture, before energy is wasted. Here, several analog-to-digital converter (ADC) architectures are provided to perform compressive sensing. Each of these new architectures selects resolutions for each sample substantially at random and adjusts the sampling rate as a function of these selected resolutions.

Abstract: For protecting a speaker, an input signal is received, and an excursion of the speaker that would be caused by the input signal is predicted. In response to the predicted excursion exceeding a threshold, a targeted excursion of the speaker is determined by compressing the predicted excursion. The targeted excursion is translated into an output signal, which is output to the speaker.

Abstract: An analog-to-digital converter (ADC) is provided. The ADC includes a plurality of pipelined ADCs and an adjustment circuit. Each pipelined ADC is adapted to receive an analog input signal, has an adjustable transfer function, and includes a compensator. The adjustment circuit is coupled to each pipelined ADC to be able to adjust the transfer function for each pipelined ADC so as to generally eliminate an estimation ambiguity. Additionally, the adjustment circuit estimates an inter-stage error that includes at least one of an inter-stage gain error and a DAC gain error and adjusts the compensator for each pipelined ADC to compensate for the inter-stage error.

Abstract: A receiving modem (20) including functionality for adjusting the synchronization of its timing relative to a transmitting modem (10), by recovering timing from a plurality of pilot tones transmitted by the transmitting modem (10) in a multicarrier modulation signal, is disclosed. For a subset of the pilot tones, the receiving modem (20) sums phase error differences, for example according to a weighted sum, to derive a phase error estimate relative to reference signals, in which noncoherent random noise cancels out, to correct for timing differences between the modems (10, 20). For another subset of the pilot tones not used in the phase error estimate, the receiving modem (20) analyzes the demodulated signal for those tones to adjust reference signals for each of those pilot tones. The adjusting is performed either by deriving an adjusted complex reference signal, or by adjusting coefficients in a frequency domain equalizer (28).

Abstract: An integrated circuit 18 is provided that includes a memory 32 and a memory modification component 33. The memory 32 maintains a bits count, a gain, and a tone order for each of a plurality of discrete multi-tone sub-channels. The memory modification component 33 operable to control an in-service modification of at least some of the bits count, the gain, and the tone order using a single bits, gains and tone order table.

Abstract: The present disclosure describes a method and system of tone ordering for discrete multi-tone (DMT) systems with tone grouping. Some illustrative embodiments may include a method, comprising, dividing a plurality of sub-channels within a discrete multi-tone symbol into a plurality of groups, selecting each sub-channel of the plurality of sub-channels in non-sequential sub-channel order from the plurality of groups, and assigning data bits in bit order to each sub-channel in the order in which each sub-channel is selected. Each group comprises an equal number of sub-channels, and each sub-channel comprises a frequency bin with a unique center frequency.

Abstract: A communication system 10 initialized using an iterative training portion is disclosed. The communication system 10 comprises a first transceiver 12 operable to communicate according to a digital subscriber line standard and a second transceiver 14 operable to communicate according to the digital subscriber line standard with the first transceiver 12. The first and the second transceivers 12, 14 are operable to perform an initialization 50, including a handshake portion 52, a channel analysis portion 56, and a training portion 54, the first and second transceivers 12, 14 negotiating a plurality of iterations of the training portion 54.

Abstract: Multicarrier modulated communications, involving a transmitting modem (30; 30?) and a receiving modem (40; 40?) that operate according to normal operating mode and a low-power, quiescent, mode, are disclosed. In the quiescent mode, transmitter power is saved by scaling down the amplitude of the transmitted symbols in each frame by a scaling factor (A), and transmitting multiple instances of the frames in sequence. On receipt, the repeated frames are summed, and the summed amplitudes scaled to account for the repetition factor (M) and the scaling factor (A), to recover the transmitted information. If the repetition factor (M) is greater than or equal to the square of the scaling factor (A), the signal-to-noise ratio in quiescent mode is at least as high as that in normal operation. The repeating, scaling, and summing may all be performed in the frequency domain or in the time domain, as desired.

Abstract: A method of transmit power adjustment in multitone communication systems is performed by changing a power spectral density for each subchannel k the power spectral density to the minimum of the power spectral density and a miximum of the power spectral density REFPSD(k)=min(NOMPSD(k), NOMPSD?PCB) where REFPSD(k) is the transmitted PSD at tone k, NOMPSD(k) is the maximum transmit PSD allowed at each tone k, NOMPSD is the maximum value of NOMPSD(k) over all k and PCB is a power cutback level.

Abstract: A device comprises a first component operable to produce a plurality of discrete multitone symbols based in part on a subchannel spacing and a cyclic extension length and a second component operable to determine the subchannel spacing and the cyclic extension length, the cyclic extension length selected based on the subchannel spacing to increase the number of discrete multitone symbols transmitted during a low noise interval of a cyclostationary noise environment.

Abstract: The present invention provides, in one embodiment, a method of simplifying filtering with respect to a band plan. The method includes selecting a reflection point relative to the band plan and designating subchannels as unusable within a filterable separation of the reflection point that would otherwise introduce an asymmetry within the filterable separation. The present invention also provides a filtering simplifier for use with a DSL system having a band plan. The filtering simplifier includes a reflection point selector configured to select a reflection point relative to the band plan. The filtering simplifier also includes a subchannel designator coupled to the reflection point selector and configured to designate subchannels as unusable within a filterable separation of the reflection point that would otherwise introduce an asymmetry within the filterable separation.

Abstract: A method and apparatus for providing extended upstream data transmission in a band having a lowest frequency f0 by an end user terminal unit in an asymmetric digital subscriber line communication between a central office terminal unit and the end user terminal unit, using a loop having a length. In the method, a target rate of upstream data transmission is provided. A plurality of sets of values are determined, of (1) an extension frequency f2 that is higher than a frequency f1 for upstream data transmission, f1 being a frequency established for non-extended upstream data transmission, the region bounded by f1 and f2 being an extension band for upstream data transmission, and (2) a maximum power level S2 for the extension band determined by the extension frequency in the set.

Abstract: A method of communicating across a channel includes receiving information having a known bandwidth and a known spectrum. The information is preferably in the form of a multicarrier modulated signal, e.g., a DMT signal. In one aspect, this information is received at a receiver having a reduced channel bandwidth. An aliasing spectrum can be calculated based on the known spectrum and the frequency difference between the known bandwidth and the reduced-channel bandwidth. The received information can then be modified based upon the aliasing function to compensate for alias distortion. For example, the received information can be modified by modifying the noise component or signal-to-noise ratio of the received information.

Abstract: Time domain equalizer determination for discrete multitone systems by tone error minimization from a starting point of a single-tone equalizer.