Abstract: An apparatus and method for communications are disclosed. The apparatus may include an a quantizer having three levels, and a switching power amplifier configured to drive a load having first and second terminals, wherein the switching power amplifier is further configured to switch the first and second terminals between first and second power rails only if the output from the quantizer is at one of the three levels.

Abstract: An apparatus and method for communications is disclosed. The apparatus includes a receiver configured to scale an audio signal, and a transducer circuit configured to provide an ambient signal in response to an ambient condition, wherein the receiver is further configured to scale the ambient signal from the transducer circuit and combine the scaled ambient signal with the scaled audio signal, the receiver being further configured to adjust the scaling applied to at least one of the ambient and audio signals.

Abstract: An apparatus and method for processing signals are disclosed. The apparatus may include an oversampling circuit configured to receive a plurality of audio signal samples, the oversampling circuit being further configured to replicate each of the audio signal samples n times, wherein n is variable.

Abstract: An apparatus and method for processing signals are disclosed. The apparatus may include a receiver configured to receive an audio signal having a plurality of audio artifacts, and an audio circuit configured to reduce the audio artifacts during at least a portion of a time period as a function of an energy level of the audio signal during that time period.

Abstract: According to one aspect of the present invention, a method and apparatus is provided in which input data (e.g., input video data) is encoded in accordance with a first coding standard (e.g., MPEG-4) to generate encoded data. The input data is also encoded based on a reconstruction of the input data to generate encoded side information associated with the input data. The encoded data are transmitted to a destination (e.g., a decoding subsystem) over a first channel and the encoded side information are transmitted to the destination over a second channel. The encoded data and the encoded side information are decoded and combined at the destination to generate output data.

Abstract: An apparatus for audio processing including a first device (e.g., a multiplier, digital signal gain module, etc.) adapted to apply a gain to a first digital audio signal to generate a second digital audio signal; a second device (e.g., a digital-to-analog converter (DAC), etc.) adapted to generate an analog audio signal from the second digital audio signal; a third device (e.g., a detector, sensor, user interface, etc.) adapted to generate an audio characteristic signal related to a characteristic of the first or second digital audio signal, or the analog audio signal; and a fourth device (e.g., a controller, control module, etc.) adapted to control the gain of the first device based on a first function of the audio characteristic signal, and control a power supplied to the second device based on a second function of the audio characteristic signal.

Abstract: The rate at which data is provided by one device and the rate at which that data is processed by another device may differ. For example, a transmitting device may transmit data according to a transmit clock while a receiving device that receives the transmitted data may process the data according to a receive clock. If there is a timing mismatch between the transmit and receive clocks, the receiving device may receive data faster or slower than it processes the data. In such a case, there may be errors relating to the processing of the received data. To address timing mismatches such as this, the receiving device may delete data from or insert data into the received data. In conjunction with these operations, the receiving device may modify the received data at or near the insertion point or the deletion point in a manner that mitigates any adverse effect the insertion or deletion may have on a resulting output signal.

Abstract: Low latency and computationally efficient techniques may be employed to account for errors in data such as low bit-width, oversampled data. In some aspects these techniques may be employed to mitigate audio artifacts associated with sigma-delta modulated audio data. In some aspects an error may be detected in a set of encoded data based on an outcome of a channel decoding process. Upon determining that a set of data may contain at least one error, the set of data may be replaced with another set of data that is based on one or more neighboring data sets. For example, in some aspects a set of data including at least one bit in error may be replaced with data that is generated by applying a cross-fading operation to neighboring data sets. In some aspects a given data bit may be flipped as a result of a linear prediction operation that is applied to PCM equivalent data that is associated with the given data bit and its neighboring data bits.

Abstract: Low latency and computationally efficient techniques may be employed to account for errors in data such as low bit-width, oversampled data. In some aspects these techniques may be employed to mitigate audio artifacts associated with sigma-delta modulated audio data. In some aspects an error may be detected in a set of encoded data based on an outcome of a channel decoding process. Upon determining that a set of data may contain at least one error, the set of data may be replaced with another set of data that is based on one or more neighboring data sets. For example, in some aspects a set of data including at least one bit in error may be replaced with data that is generated by applying a cross-fading operation to neighboring data sets. In some aspects a given data bit may be flipped as a result of a linear prediction operation that is applied to PCM equivalent data that is associated with the given data bit and its neighboring data bits.

Abstract: The rate at which data is provided by one device and the rate at which that data is processed by another device may differ. For example, a transmitting device may transmit data according to a transmit clock while a receiving device that receives the transmitted data may process the data according to a receive clock. If there is a timing mismatch between the transmit and receive clocks, the receiving device may receive data faster or slower than it processes the data. In such a case, there may be errors relating to the processing of the received data. To address timing mismatches such as this, the receiving device may delete data from or insert data into the received data. In conjunction with these operations, the receiving device may modify the received data at or near the insertion point or the deletion point in a manner that mitigates any adverse effect the insertion or deletion may have on a resulting output signal.

Abstract: Apparatus and method for processing signals. A sigma-delta modulator is used. An adaptive dynamic range controller is configured to adaptively adjust the dynamic range of a signal output from the sigma-delta modulator.

Abstract: An audio coding technique based on modeling spectral dynamics is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. Each sub-band is then frequency transformed and linear prediction is applied. This results in a Hilbert envelope and a Hilbert Carrier for each of the sub-bands. Because of application of linear prediction to frequency components, the technique is called Frequency Domain Linear Prediction (FDLP). The Hilbert envelope and the Hilbert Carrier are analogous to spectral envelope and excitation signals in the Time Domain Linear Prediction (TDLP) techniques. Temporal masking is applied to the FDLP sub-bands to improve the compression efficiency. Specifically, forward masking of the sub-band FDLP carrier signal can be employed to improve compression efficiency of an encoded signal.

Abstract: This disclosure describes techniques that can facilitate multimedia telephony. In one example, a method for communication of multimedia data comprises determining a first level of throughput associated with multimedia data communication from a first access terminal to a network, determining a second level of throughput associated with multimedia data communication from the network to a second access terminal based on feedback from the second access terminal to the first access terminal via the network, determining a budget associated with communication of a video unit of the multimedia data, and coding the video unit of the multimedia data based on the budget and the first and second levels of throughput.

Abstract: Apparatus and method for processing signals. A sigma-delta modulator is used. An adaptive dynamic range controller is configured to adaptively adjust the dynamic range of a signal output from the sigma-delta modulator.

Abstract: A method, apparatus, and system for providing distributed source coding techniques that improve data coding performance, such as video data coding, when channel errors or losses occur. Errors in the reconstruction of the data is eliminated or reduced by sending extra information. Correlation between a predicted sequence and an original sequence can be used to design codebooks and find the cosets required to represent the original image. This information may be sent over another channel, or a secondary channel.

Abstract: A method and system for resynchronizing an embedded multimedia system using bytes consumed in an audio decoder. The bytes consumed provides a mechanism to compensate for bit error handling and correction in a system that does not require re-transmission. The audio decoder keeps track of the bytes consumed and periodically reports the bytes consumed. A host microprocessor indexes the actual bytes consumed since bit errors may have been handled or corrected to a predetermined byte count to determine whether resynchronization is necessary.

Abstract: In an apparatus and method, time-varying signals are processed and encoded via a frequency domain linear prediction (FDLP) scheme to arrive at an all-pole model. Residual signals resulted from the scheme are estimated. Quantized values of the all-pole model and the residual signals are packetized as encoded signals suitable for transmission or storage. To reconstruct the time-varying signals, the encoded signals are decoded. The decoding process is basically the reverse of the encoding process.

Abstract: Techniques are described to reduce rounding errors during computation of discrete cosine transform using fixed-point calculations. According to these techniques, a discrete cosine transform a matrix of scaled coefficients is calculated by multiplying coefficients in a matrix of coefficients by scale factors. Next, a midpoint bias value and a supplemental bias value are added to a DC coefficient of the matrix of scaled coefficients. Next, an inverse discrete cosine transform is applied to the resulting matrix of scaled coefficients. Values in the resulting matrix are then right-shifted in order to derive a matrix of pixel component values. As described herein, the addition of the supplemental bias value to the DC coefficient reduces rounding errors attributable to this right-shifting. As a result, a final version of a digital media file decompressed using these techniques may more closely resemble an original version of a digital media file.

Abstract: In an apparatus and method, time-varying signals are processed and encoded via a frequency domain linear prediction (FDLP) scheme to arrive at an all-pole model. Residual signals resulted from the scheme are estimated and transformed into a time domain signal. Through the process of heterodyning, the time domain signal is frequency shifted toward the baseband level as a downshifted carrier signal. Quantized values of the all-pole model and the frequency transform of the downshifted carrier signal are packetized as encoded signals suitable for transmission or storage. To reconstruct the time-varying signals, the encoded signals are decoded. The decoding process is basically the reverse of the encoding process.

Abstract: Techniques for efficiently performing full and scaled transforms on data received via full and scaled interfaces, respectively, are described. A full transform is a transform that implements the complete mathematical description of the transform. A full transform operates on or provides full transform coefficients. A scaled transform is a transform that operates on or provides scaled transform coefficients, which are scaled versions of the full transform coefficients. The scaled transform may have lower computational complexity whereas the full transform may be simpler to use by applications. The full and scaled transforms may be for a 2D IDCT, which may be implemented in a separable manner with 1D IDCTs. The full and scaled transforms may also be for a 2D DCT, which may be implemented in a separable manner with ID DCTs. The 1D IDCTs and 1D DCTs may be implemented in a computationally efficient manner.