摘要:
An apparatus for calculating bandwidth extension data of an audio signal in a bandwidth extension system, in which a first spectral band is encoded with a first number of bits and a second spectral band different from the first spectral band is encoded with a second number of bits, the second number of bits being smaller than the first number of bits, has a controllable bandwidth extension parameter calculator for calculating bandwidth extension parameters for the second frequency band in a frame-wise manner for a sequence of frames of the audio signal. Each frame has a controllable start time instant. The apparatus additionally includes a spectral tilt detector for detecting a spectral tilt in a time portion of the audio signal and for signaling the start time instant for the individual frames of the audio signal depending on spectral tilt.
摘要:
An audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation.
摘要:
A watermark signal provider provides a watermark signal suitable for being hidden in an audio signal when the watermark signal is added to the audio signal, such that the watermark signal represents watermark data. The watermark signal provider includes a psychoacoustical processor for determining a masking threshold of the audio signal; and a modulator for generating the watermark signal from a superposition of sample-shaping functions spaced apart from each other at a sample time interval of a time-discrete representation of the watermark data, each sample-shaping function being amplitude-weighted with a respective sample of the time-discrete representation, multiplied by a respective amplitude weight depending on the masking threshold, the modulator being configured such that the sample time interval is shorter than a time extension of the sample-shaping functions; and the respective amplitude weight also depends on samples of the time-discrete representation neighboring the respective sample in time.
摘要:
For classifying different segments of a signal which has segments of at least a first type and second type, e.g. audio and speech segments, the signal is short-term classified on the basis of the at least one short-term feature extracted from the signal and a short-term classification result is delivered. The signal is also long-term classified on the basis of the at least one short-term feature and at least one long-term feature extracted from the signal and a long-term classification result is delivered. The short-term classification result and the long-term classification result are combined to provide an output signal indicating whether a segment of the signal is of the first type or of the second type.
摘要:
An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.
摘要:
Quantizing an information signal of a sequence of information values includes frequency-selective filtering the sequence of information values to obtain a sequence of filtered information values and quantizing the filtered information values to obtain a sequence of quantized information values by means of a quantizing step function which maps the filtered information values to the quantized information values and the course of which is steeper below a threshold information value than above the threshold information value.
摘要:
Coding an audio signal of a sequence of audio values into a coded signal includes determining a first listening threshold for a first block of audio values of the sequence of audio values and a second listening threshold for a second block of audio values of the sequence of audio values; calculating a version of a first parameterization of a parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first listening threshold and a version of a second parameterization of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the second listening threshold; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which in a predetermined manner depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block; quantizing the filtered audio values to obtain a block of quantized filtered audio values; forming a combination of the version of the first parameterization and the version of the second parameterization including at least a difference between the version of the first parameterization and the version of the second parameterization; and integrating information from which the quantized filtered audio values and a version of the first parameterization may be derived and which includes the combination into the coded signal.
摘要:
A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.
摘要:
A watermark signal provider comprises a time-frequency-domain waveform provider to provide time-domain waveforms for a plurality of frequency subbands. The time-frequency-domain waveform provider is configured to map a given value of a time-frequency-domain representation onto a bit shaping function, a temporal extension of which is longer than a bit interval, such that there is a temporal overlap between bit shaped functions provided for temporally subsequent values of the time-frequency-domain representation of the same frequency subband. A time-domain waveform of a given frequency subband contains a plurality of bit shaped functions provided for temporally subsequent values of the time-frequency-domain representation. The water mark signal provider further has a time-domain waveform combiner.
摘要:
An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.