摘要:
A method for processing an audio signal (100) includes receiving an encoded audio signal and generating a decoded audio signal by processing consecutive audio frames of the audio signal to avoid discontinuities. Processing consecutive audio frames of the audio signal to avoid discontinuities includes using linear predictive filtering for removing (S102, S104, S300-S308, S400-S402) a discontinuity (106a, 106b) between a filtered past frame and a filtered current frame of the audio signal. The method includes filtering the current frame of the audio signal and removing the discontinuity by modifying a beginning portion of the filtered current frame by a signal obtained by linear predictive filtering a predefined signal with initial states of the linear predictive filter defined on the basis of a last part of the unfiltered past frame filtered using the set of filter parameters for filtering the current frame.
摘要:
Audio decoder device for decoding a bitstream, the audio decoder device comprising: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder comprises a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder comprises a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device comprising one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, for one or more of said memories by resampling a preceding memory state for synthesizing one or more audio parameters for a preceding decoded audio frame, which has a preceding sampling rate being different from the sampling rate of the decoded audio frame, for one or more of said memories and to store the memory state for synthesizing of the one or more audio parameters for the decoded audio frame for one or more of said memories into the respective memory.
摘要:
An audio decoder (100; 300) for providing a decoded audio information (112;312) on the basis of an encoded audio information (110; 310) comprises an error concealment (130; 380; 500) configured to provide an error concealment audio information (132;382;512) for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation (322) using a time domain excitation signal (532). This time domain excitation signal is modified by time-scaling it in dependence on a prediction of a pitch for the time of the one or more lost audio frames.
摘要:
Embodiments of the present invention provide an encoder comprising a quantization stage, an entropy encoder, a residual quantization stage and a coded signal former. The quantization stage is configured to quantize an input signal using a dead zone in order to obtain a plurality of quantized values. The entropy encoder is configured to encode the plurality of quantized values using an entropy encoding scheme in order to obtain a plurality of entropy encoded values. The residual quantization stage is configured to quantize a residual signal caused by the quantization stage, wherein the residual quantization stage is configured to determine at least one quantized residual value in dependence on the dead zone of the quantization stage. The coded signal former is configured to form a coded signal from the plurality of entropy encoded values and the at least one quantized residual value.
摘要:
A method for processing an audio signal (100) includes receiving an encoded audio signal and generating a decoded audio signal by processing consecutive audio frames of the audio signal to avoid discontinuities. Processing consecutive audio frames of the audio signal to avoid discontinuities includes using linear predictive filtering for removing (S102, S104, S300-S308, S400-S402) a discontinuity (106a, 106b) between a filtered past frame and a filtered current frame of the audio signal. The method includes filtering the current frame of the audio signal and removing the discontinuity by modifying a beginning portion of the filtered current frame by a signal obtained by linear predictive filtering a predefined signal with initial states of the linear predictive filter defined on the basis of a last part of the unfiltered past frame filtered using the set of filter parameters for filtering the current frame.
摘要:
An audio decoder (100; 300) for providing a decoded audio information (112;312) on the basis of an encoded audio information (110; 310) comprises an error concealment (130; 380; 500) configured to provide an error concealment audio information (132;382;512) for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation (322) using a time domain excitation signal (532).
摘要:
An audio decoder (100;200;300) for providing a decoded audio information (112;212;312) on the basis of an encoded audio information (110;210;310), the audio decoder comprises a linear-prediction-domain decoder (120;220;320) configured to provide a first decoded audio information (122;222;322; S C (n)) on the basis of an audio frame encoded in a linear prediction domain, a frequency domain decoder (130;230;330) configured to provide a second decoded audio information (132;232;332; S M (n)) on the basis of an audio frame encoded in a frequency domain, and a transition processor (140;240;340).The transition processor is configured to obtain a zero-input-response (150; 256;348) of a linear predictive filtering(148; 254; 346), wherein an initial state (146;252;344) of the linear predictive filtering is defined in dependence on the first decoded audio information and the second decoded audio information. The transition processor is also configured to modify the second decoded audio information (132; 232;332;S M (n)), which is provided on the basis of an audio frame encoded in the frequency domain following an audio frame encoded in the linear prediction domain, in dependence on the zero-input-response, to obtain a smooth transition between the first decoded audio information (S C (n)) and the modified second decoded audio information ( S M ^ n ).