摘要:
An audio encoder for encoding an audio signal comprises a first coding branch (400), the first coding branch comprising a first converter (410) for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder comprises a second coding branch (500) comprising a second time/frequency converter (523). Additionally, a signal analyzer (300/525) for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters (410, 523) when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.
摘要:
An encoder for encoding an audio signal. The encoder is configured to encode the audio signal in a transform domain or filter-bank domain, wherein the encoder is configured to determine spectral coefficients of the audio signal for a current frame and at least one previous frame, wherein the encoder is configured to selectively apply predictive encoding to a plurality of individual spectral coefficients or groups of spectral coefficients which are separated by at least one spectral coefficient.
摘要:
An audio encoder (10) adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame comprises a number of time domain audio samples. The audio encoder (10) comprises a predictive coding analysis stage (12) for determining information on coefficients of a synthesis filter and an excitation frame based on a frame of audio samples, the excitation frame comprising samples of an excitation signal for the synthesis filter. The audio encoder (10) further comprises a time-aliasing introducing transformer (14) for transforming overlapping excitation frames to the frequency domain to obtain excitation frame spectra, wherein the time-aliasing introducing transformer (14) is adapted for transforming the overlapping excitation frames in a critically-sampled way. Moreover, the audio encoder (10) comprises a redundancy reducing encoder (16) for encoding the excitation frame spectra to obtain the encoded frames based on the coefficients and the encoded excitation frame spectra.
摘要:
An audio encoder comprises a window function controller (504), a windower (502), a time warper (506) with a final quality check functionality, a time/frequency converter (508), a TNS stage (510) or a quantizer encoder (512), the window function controller (504), the time warper (506), the TNS stage (510) or an additional noise filling analyzer (524) are controlled by signal analysis results obtained by a time warp analyzer (516) or a signal classifier (520). Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.
摘要:
An audio encoder comprises a window function controller (504), a windower (502), a time warper (506) with a final quality check functionality, a time/frequency converter (508), a TNS stage (510) or a quantizer encoder (512), the window function controller (504), the time warper (506), the TNS stage (510) or an additional noise filling analyzer (524) are controlled by signal analysis results obtained by a time warp analyzer (516) or a signal classifier (520). Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.
摘要:
An audio signal processor (100) comprising an analysis means (104), a manipulation factor unit (106), and a time-stretching and compression device (108). The analysis means (104) is implemented to determine a first measure of information content (M 1 ) of a first time section of an audio signal and a second measure of information content (M 2 ) of a second time section. The manipulation factor unit (106) is implemented to determine a time manipulation factor (ΔD 1 ) for the first time section in dependence on the first measure of information content (M 1 ) and the second measure of information content (M 2 ). The time-stretching and compression device (108) is implemented to time-stretch or compress the first time section according to the manipulation factor (ΔD 1 ) and to treat the second time section differently from the first time section. A corresponding method for adjusting time information content variations of an audio signal (s) is also disclosed. The determination of the first measure of information content and of the second measure of information content may be based on externally provided control information such as meta-data provided along with the audio signal.
摘要:
An apparatus for obtaining a parameter describing a variation of a signal characteristic of a signal on the basis of actual transform-domain parameters describing the audio signal in transform-domain comprises a parameter determinator. The parameter determinator is configured to determine one or more model parameters of a transform-domain variation model describing an evolution of the transform-domain parameters in dependence on one or more model parameters representing a signal characteristic.