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公开(公告)号：US20190333528A1
公开(公告)日：2019-10-31
申请号：US16419777
申请日：2019-05-22
摘要： Embodiments of the present application provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.
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公开(公告)号：US20190320263A1
公开(公告)日：2019-10-17
申请号：US16454250
申请日：2019-06-27
IPC分类号： H04R5/00 , H04S5/00 , G10L19/16 , G10L19/008 , G10L19/032 , G10L19/26 , H04S3/02 , G10L19/02
摘要： A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.
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公开(公告)号：US10438601B2
公开(公告)日：2019-10-08
申请号：US15817218
申请日：2017-11-19
发明人： Stefan Bruhn
IPC分类号： G10L19/087 , G10L19/26 , G10L21/02 , G10L19/06 , G10L21/0308 , G10L25/84 , G10L19/012 , G10L21/0216
摘要： In a method for coding of information for enhancing a background noise representation, voice activity of an input speech signal is determined. A noisiness parameter is determined for an inactive speech signal, wherein the noisiness parameter is based on a ratio of prediction gains of two Linear Predictive Coder (LPC) prediction filters with different orders. The noisiness parameter is quantized, and the quantized noisiness parameter is encoded for transmission.
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94.发明授权公开(公告)号：US10431232B2
公开(公告)日：2019-10-01
申请号：US14811386
申请日：2015-07-28
摘要： A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described.
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95.发明授权公开(公告)号：US10418045B2
公开(公告)日：2019-09-17
申请号：US15482328
申请日：2017-04-07
摘要： Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.
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96.发明申请公开(公告)号：US20190272839A1
公开(公告)日：2019-09-05
申请号：US16256937
申请日：2019-01-24
IPC分类号： G10L19/22 , G10L19/12 , G10L19/26 , G10L19/09 , G10L19/032
摘要： An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.
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公开(公告)号：US20190259405A1
公开(公告)日：2019-08-22
申请号：US16398669
申请日：2019-04-30
发明人： Markus Lohwasser , Manuel Jander , Max Neuendorf , Ralf Geiger , Markus Schnell , Matthias Hildenbrand , Tobias Chalupka
摘要： An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.
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公开(公告)号：US10347268B2
公开(公告)日：2019-07-09
申请号：US15603946
申请日：2017-05-24
摘要： A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals.
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公开(公告)号：US10339946B2
公开(公告)日：2019-07-02
申请号：US15138552
申请日：2016-04-26
发明人： Jérémie Lecomte
IPC分类号： G10L19/005 , G10L19/125 , G10L19/022 , G10L19/038 , G10L19/012 , G10L19/26 , G10L19/02 , G10L19/08 , G10L25/90 , G10L19/12
摘要： An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.
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公开(公告)号：US10311882B2
公开(公告)日：2019-06-04
申请号：US15988135
申请日：2018-05-24
IPC分类号： G06F17/00 , G10L19/02 , G10L21/038 , G10L19/26 , G10L19/00
摘要： The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications.
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