摘要:
In a method of coding discrete time signals (X1) sampled with a first sampling rate, second time signals (x2) are generated using the first time signals having a bandwidth corresponding to a second sampling rate, with the second sampling rate being lower than the first sampling rate. The second time signals are coded in accordance with a first coding algorithm. The coded second signals (X2c) are decoded again in order to obtain coded/decoded second time signals (X2cd) having a bandwidth corresponding to the second sampling frequency. The first time signals, by frequency domain transformation, become first spectral values (X1). Second spectral values (X2cd) are generated from the coded/decoded second time signals, the second spectral values being a representation of the coded/decoded time signals in the frequency domain. To obtain weighted spectral values, the first spectral values are weighted by means of the second spectral values, with the first and second spectral values having the same frequency and time resolution. The weighted spectral values (Xb) are coded in accordance with a second coding algorithm in consideration of a psychoacoustic model and written into a bit stream. Weighting the first spectral values and the second spectral values comprises the subtraction of the second spectral values from the first spectral values in to obtain differential spectral values.
摘要:
An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.
摘要:
An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.
摘要:
An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.
摘要:
An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.
摘要:
A cochlear implant for processing signal parameters which are adapted for controlling the cochlear implant, which are based on the audio signal and which enable generating a representation of the audio signal by the cochlear implant includes a receive interface which is implemented to receive the signal parameters and a nerve stimulator for processing the signal parameters to generate nerve cell stimulation signals based on the signal parameters. A device for generating a control signal for a cochlear implant on the basis of an audio signal includes a cochlear parameter extractor for analyzing the audio signal which is implemented to generate signal parameters as input information for the cochlear implant based on an analysis of the audio signal using a human hearing simulation model, and a transmit interface for transmitting the signal parameters to the cochlear implant.
摘要:
Standard video compression techniques apply motion-compensated prediction combined with transform coding of the prediction error. In the context of prediction with fractional-pel motion vector resolution it was shown, that aliasing components contained in an image signal are limiting the prediction efficiency obtained by motion compensation. In order to consider aliasing, quantization and motion estimation errors, camera noise, etc., we analytically developed a two dimensional (2D) non-separable interpolation filter, which is independently calculated for each frame by minimizing the prediction error energy. For every fractional-pel position to be interpolated, an individual set of 2D filter coefficients is determined. Since transmitting filter coefficients as side information results in an additional bit rate, which is almost constant for different image resolutions and total bit rates, the loss in coding gain increases when total bit rates sink. Therefore, we developed an algorithm, which regards the non-separable two-dimensional filter as a polyphase filter. For each frame, predicting the interpolation filter impulse response through evaluation of the polyphase filter, we only have to encode the prediction error of the filter coefficients.
摘要:
An audio signal decoder has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value, based on time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation, based on an encoded audio signal representation and using the rescaled version of the time warp contour.
摘要:
In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.
摘要:
A method of concealing a packet loss during video decoding is provided. An input stream having a plurality of network abstraction layer units NAL is received. A loss of a network abstraction layer unit in a group of pictures in the input stream is detected. A valid network abstraction layer unit order from the available network abstraction layer units is outputted. The network abstraction layer unit order is received by a video coding layer (VCL) and data is outputted.