摘要:
In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information. Since the channel side information only occupy a low number of bits, and since the decoder does not use dematrixing, an efficient and high quality multi-channel extension for stereo players and enhanced multi-channel players is obtained.
摘要:
In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information. Since the channel side information only occupy a low number of bits, and since the decoder does not use dematrixing, an efficient and high quality multi-channel extension for stereo players and enhanced multi-channel players is obtained.
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.
摘要:
For flexibly signaling a synchronous mode or an asynchronous mode in the multi-channel parameter reconstruction, a parameter configuration cue is inserted in the data stream, which is used by a configurator on the side of a multi-channel decoder to configure a multi-channel reconstructor. If the parameter configuration cue has a first meaning, the configurator will look for further configuration information in its input data, while, when the parameter configuration cue has another meaning, the configurator performs a configuration setting of the multi-channel reconstructor based on information on a coding algorithm with which transmission channel data have been coded, so that it is ensured efficiently on the one hand and flexibly on the other hand that there will always be obtained a correct association between parameter data and decoded transmission channel data.
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.
摘要:
For flexibly signaling a synchronous mode or an asynchronous mode in the multi-channel parameter reconstruction, a parameter configuration cue is inserted in the data stream, which is used by a configurator on the side of a multi-channel decoder to configure a multi-channel reconstructor. If the parameter configuration cue has a first meaning, the configurator will look for further configuration information in its input data, while, when the parameter configuration cue has another meaning, the configurator performs a configuration setting of the multi-channel reconstructor based on information on a coding algorithm with which transmission channel data have been coded, so that it is ensured efficiently on the one hand and flexibly on the other hand that there will always be obtained a correct association between parameter data and decoded transmission channel data.
摘要:
A transient detector is provided for generating an ambience signal suitable for being emitted via loudspeakers for which there is no special loudspeaker signal to detect a transient period. A synthesis signal generator produces a synthesis signal which fulfills the transient condition on the one hand and the continuity condition for the synthesis signal on the other hand. A signal substituter will then substitute a portion of the examination signal by the synthesis signal to obtain an ambience signal for the surround channels.
摘要:
An apparatus for extracting an ambient signal from an input audio signal comprises a gain-value determinator configured to determine a sequence of time-varying ambient signal gain values for a given frequency band of the time-frequency distribution of the input audio signal in dependence on the input audio signal. The apparatus comprises a weighter configured to weight one of the sub-band signals representing the given frequency band of the time-frequency-domain representation with the time-varying gain values, to obtain a weighted sub-band signal. The gain-value determinator is configured to obtain one or more quantitative feature-values describing one or more features of the input audio signal and to provide the gain-value as a function of the one or more quantitative feature values such that the gain values are quantitatively dependent on the quantitative values. The gain value determinator is configured to determine the gain values such that ambience components are emphasized over non-ambience components in the weighted sub-band signal.
摘要:
A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter. Accordingly, an encoded audio signal representation representing the audio signal can be obtained.
摘要:
An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.