摘要:
The invention relates to an embedder for embedding a watermark in a representation of input information, comprising a unit for determining embedding parameters that is designed to apply a derivation function to an initial value once or repeatedly in order to obtain an embedding parameter for embedding the watermark in the representation of input information. The embedder further comprises a watermark inserter that is designed to provide the representation of input information with the watermark using the embedding parameter. The embedder is designed to select the number of times the derivation function is to be applied to the initial value.
摘要:
An embodiment of an analysis filterbank for filtering a plurality of time domain input frames, wherein an input frame comprises a number of ordered input samples, comprises a windower configured to generating a plurality of windowed frames, wherein a windowed frame comprises a plurality of windowed samples, wherein the windower is configured to process the plurality of input frames in an overlapping manner using a sample advance value, wherein the sample advance value is less than the number of ordered input samples of an input frame divided by two, and a time/frequency converter configured to providing an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.
摘要:
When an audio signal is coded, coded signals of inferior quality and bit-rate as well as coded signals of high quality and bit-rate are transmitted to a decoder. The low-bit-rate audio signal is coded and transmitted to the decoder first, before a further coded signal is transmitted to the decoder and either alone or together with the first coded signal produces a high-quality decoded signal within the decoder during decoding. This results first in a low-quality decoded signal in the decoder before decoding of the high-quality signal becomes possible.
摘要:
A process for transmitting and/or storing digital signals from several channels is particularly suitable for transmitting the five channels of a 3/2 stereophonic system, as well as for transmitting two stereo channels and three additional commentary channels. This process allows for example television programmes to be broadcast together with multilingual audio signals. This process is characterised in that only a bit rate of 384 kbit/s is required for transmission, thanks to the reduction of the data to be transmitted. In order to reduce the data, K input channels are reproduced segment by segment on N « K virtual spectral data channels, the spectral data channels are quantified, coded and transmitted, taking into account the laws of psychoacoustics, and K output channels are reproduced from the transmitted bit flow by means of a list transmitted therewith from the N « K spectral data channels.
摘要:
An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal comprises a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also comprises an audio stream provider configured to provide the audio stream such that the audio stream comprises an information describing an audio content of the frequency bands and an information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal comprises a noise filler configured to introduce noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.
摘要:
An audio encoder (100) for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder (100) comprising a processor (110) for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder (100) further comprises an entropy encoder (120) for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.
摘要:
An apparatus for generating a plurality of audio channels for a first speaker setup is characterized by an imaginary speaker determiner, an energy distribution calculator, a processor and a renderer. The imaginary speaker determiner is configured to determine a position of an imaginary speaker not contained in the first speaker setup to obtain a second speaker setup containing the imaginary speaker. The energy distribution calculator is configured to calculate an energy distribution from the imaginary speaker to the other speakers in the second speaker setup. The processor is configured to repeat the energy distribution to obtain a downmix information for a downmix from the second speaker setup to the first speaker setup. The renderer is configured to generate the plurality of audio channels using the downmix information.
摘要:
An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal comprises a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also comprises an audio stream provider configured to provide the audio stream such that the audio stream comprises an information describing an audio content of the frequency bands and an information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal comprises a noise filler configured to introduce noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.
摘要:
An audio encoder (100) for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder (100) comprising a processor (110) for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder (100) further comprises an entropy encoder (120) for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.
摘要:
An audio signal decoder (200) for providing a decoded representation (212) of an audio content on the basis of an encoded representation (310) of the audio content comprises a transform domain path (230, 240, 242, 250, 260) configured to obtain a time-domain representation (212) of a portion of the audio content encoded in a transform-domain mode on the basis of a first set (220) of spectral coefficients, a representation (224) of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters (222). The transform domain path comprises a spectrum processor (230) configured to apply a spectrum shaping to the first set of spectral coefficients in dependence on at least a subset of the linear-prediction-domain parameters, to obtain a spectrally-shaped version (232) of the first set of spectral coefficients. The transform domain path comprises a first frequency-domain-to-time-domain converter (240) configured to obtain a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path comprises an aliasing-cancellation stimulus filter configured to filter (250) the aliasing-cancellation stimulus signal (324) in dependence on at least a subset of the linear-prediction-domain parameters (222), to derive an aliasing-cancellation synthesis signal (252) from the aliasing-cancellation stimulus signal. The transform domain path also comprises a combiner (260) configured to combine the time-domain representation (242) of the audio content with the aliasing-cancellation synthesis signal (252), or a post-processed version thereof, to obtain an aliasing reduced time-domain signal.