Abstract: This disclosure describes a scaler architecture for image and/or video processing. One aspect relates to an apparatus comprising an image processing unit, a memory, and a coder. The memory is configured to store processed image data from the image processing unit. The coder is configured to retrieve the stored, processed image data from the memory. The coder comprises a scaler configured to upscale the retrieved image data from the memory. The coder is configured to encode the scaled image data.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: A method for modifying a window with a frame associated with an audio signal is described. A signal is received. The signal is partitioned into a plurality of frames. A determination is made if a frame within the plurality of frames is associated with a non-speech signal. A modified discrete cosine transform (MDCT) window function is applied to the frame to generate a first zero pad region, where the region has a length of (M?L)/2, where L is an arbitrary value, and a second zero pad region if it was determined that the frame is associated with a non-speech signal. The frame is encoded. The decoder window is the same as the encoder window.

Abstract: A method for reconstructing an erased speech frame is described. A second speech frame is received from a buffer. The index position of the second speech frame is greater than the index position of the erased speech frame. The type of packet loss concealment (PLC) method to use is determined based on one or both of the second speech frame and a third speech frame. The index position of the third speech frame is less than the index position of the erased speech frame. The erased speech frame is reconstructed from one or both of the second speech frame and the third speech frame.

Abstract: Automatic white balance of captured images can be performed based on a gray world assumption. Initially, a flat field gray image is captured for one or more reference illuminations. The statistics of the captured gray image are determined and stored for each reference illumination during a calibration process. For each subsequent captured image, the image is filtered to determine a subset of gray pixels. The gray pixels are further divided into a one or more gray clusters. The average weight of the one or more gray clusters is determined and a distance from the average weights to the reference illuminants is determined. An estimate of the illuminant is determined depending on the distances. White balance gains are applied to the image based on the estimated illuminant.

Abstract: A method of processing an image includes selecting, for each of a plurality of picture element values in at least a portion of the image, one among a plurality of offset values. For each of the plurality of picture element values, an index value is obtained based on (A) the selected offset value and (B) a portion of the picture element value. For each of the plurality of picture element values, an entry is retrieved from a lookup table according to the corresponding index value.

Abstract: Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed.

Abstract: Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed.

Abstract: A color interpolation method uses a first interpolation function (F1) to obtain a first missing color sub-pixel value for a pixel of interest and uses a second interpolation function (F2) to obtain a second missing color sub-pixel value for the pixel of interest. First metric (V) indicative of an edge extending in a first direction (D1) is obtained. Second metric (H) indicative of an edge extending in a second direction (D2) is obtained. The two metrics are used to generate first and second weighting factors (k1, k2). A confidence factor value can be used to place more emphasis on one metric versus the other metric in the determination of the weighting factors. In one embodiment, the sub-pixel value being interpolated is the weighted sum of the first weighting factor multiplied by the first missing color sub-pixel value plus the second weighting factor multiplied by the second missing color sub-pixel value.

Abstract: Techniques for efficiently encoding an input signal are described. In one design, a generalized encoder encodes the input signal (e.g., an audio signal) based on at least one detector and multiple encoders. The at least one detector may include a signal activity detector, a noise-like signal detector, a sparseness detector, some other detector, or a combination thereof. The multiple encoders may include a silence encoder, a noise-like signal encoder, a time-domain encoder, a transform-domain encoder, some other encoder, or a combination thereof. The characteristics of the input signal may be determined based on the at least one detector. An encoder may be selected from among the multiple encoders based on the characteristics of the input signal. The input signal may be encoded based on the selected encoder. The input signal may include a sequence of frames, and detection and encoding may be performed for each frame.

Abstract: A method for dimming a first packet associated with a first bit rate to a second packet associated with a second bit rate is described. A first packet is received. The first packet is analyzed to determine a first bit rate associated with the first packet. Bits associated with at least one parameter are discarded from the first packet. Remaining bits associated with one or more parameters and a special identifier are packed into a second packet associated with a second bit rate. The second packet is transmitted.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: Disclosed configurations include systems, methods, and apparatus arranged to generate a sequence of spectral tilt values that is based on inactive frames of a speech signal. For each of a plurality of inactive frames of the speech signal, a transmit decision is made according to a change calculated among at least two corresponding values of the sequence. The outcome of the transmit decision determines whether a silence description is transmitted for the corresponding inactive frame.

Abstract: The range of disclosed configurations includes methods in which subbands of a speech signal are separately encoded, with the excitation of a first subband being derived from a second subband. Gain factors are calculated to indicate a time-varying relation between envelopes of the original first subband and of the synthesized first subband. The gain factors are quantized, and quantized values that exceed the pre-quantized values are re-coded.

Abstract: A method for including an identifier with a packet associated with a speech signal is described. A signal is received. The signal is partitioned into a plurality of frames. A frame of the signal is encoded into a packet. A determination is made if the packet is encoded as a wideband packet or a narrowband packet. An identifier is packed in the packet based on the determination. The packet is transmitted. At least two illegal values are provided from an N-bit parameter, wherein at least one bit from the N-bit parameter is used to carry information. A number of bits from the N-bit parameter that are used to carry information is equal to log2(X), wherein X is the number of illegal values provided from the N-bit parameter.

Abstract: Applications of dim-and-burst techniques to coding of wideband speech signals are described. Reconstruction of a highband portion of a frame of a wideband speech signal using information from a previous frame is also described.

Abstract: A method for modifying a window with a frame associated with an audio signal is described. A signal is received. The signal is partitioned into a plurality of frames. A determination is made if a frame within the plurality of frames is associated with a non-speech signal. A modified discrete cosine transform (MDCT) window function is applied to the frame to generate a first zero pad region and a second zero pad region if it was determined that the frame is associated with a non-speech signal. The frame is encoded. The decoder window is the same as the encoder window.

Abstract: Speech encoders and methods of speech encoding are disclosed that encode inactive frames at different rates. Apparatus and methods for processing an encoded speech signal are disclosed that calculate a decoded frame based on a description of a spectral envelope over a first frequency band and the description of a spectral envelope over a second frequency band, in which the description for the first frequency band is based on information from a corresponding encoded frame and the description for the second frequency band is based on information from at least one preceding encoded frame. Calculation of the decoded frame may also be based on a description of temporal information for the second frequency band that is based on information from at least one preceding encoded frame.

Abstract: A speech converter in a speech processing system modifies various aspects of input speech. The speech converter receives a formants signal representing an input speech signal. The speech converter may also receive a formant scaling command or a user selection of one of multiple control signals, each specifying a manner of modifying one or more of the received signals (i.e., formants, voicing, pitch, gain). The speech converter modifies at least one of the formants, voicing, pitch, and/or gain signals as specified by the selected voice font.