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51. 发明授权

US11727944B2 Apparatus and method for stereo filling in multichannel coding 有权
公开(公告)号：US11727944B2
公开(公告)日：2023-08-15
申请号：US16918812
申请日：2020-07-01
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Sascha Dick , Christian Helmrich , Nikolaus Rettelbach , Florian Schuh , Richard Fueg , Frederik Nagel
IPC分类号： G10L19/008 , G10L19/028 , G10L19/035
CPC分类号： G10L19/008 , G10L19/028 , G10L19/035
摘要： An apparatus for decoding an encoded multichannel signal of a current frame to obtain three or more current audio output channels is provided. A multichannel processor is adapted to select two decoded channels from three or more decoded channels depending on first multichannel parameters. Moreover, the multichannel processor is adapted to generate a first group of two or more processed channels based on the selected channels. A noise filling module is adapted to identify for at least one of the selected channels, one or more frequency bands, within which all spectral lines are quantized to zero, and to generate a mixing channel using, depending on side information, a proper subset of three or more previous audio output channels that have been decoded, and to fill the spectral lines of frequency bands, within which all spectral lines are quantized to zero, with noise generated using spectral lines of the mixing channel.

52. 发明申请

US20230133513A1 AUDIO DECODER, AUDIO ENCODER, AND RELATED METHODS USING JOINT CODING OF SCALE PARAMETERS FOR CHANNELS OF A MULTI-CHANNEL AUDIO SIGNAL 有权
公开(公告)号：US20230133513A1
公开(公告)日：2023-05-04
申请号：US18086110
申请日：2022-12-21
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： Emmanuel RAVELLI , Goran MARKOVIC , Jan Frederik KIENE , Franz REUTELHUBER , Stefan DÖHLA , Eleni FOTOPOULOU
IPC分类号： G10L19/008 , G10L19/035 , G10L19/22
摘要： Audio decoder for decoding an encoded audio signal having multi-channel audio data having data for two or more audio channels, and information on jointly encoded scale parameters, having: a scale parameter decoder for decoding the information on the jointly encoded scale parameters to obtain a first and a second set of scale parameters for a first channel and a second channel, respectively, of a decoded audio signal; and a signal processor for applying the first and second sets of scale parameters to a first and second channel representation, respectively, derived from the multi-channel audio data to obtain the first and second channels of the decoded audio signal, wherein the jointly encoded scale parameters have information on a first group and on a second group of jointly encoded scale parameters, and wherein the scale parameter decoder is configured to combine a jointly encoded scale parameter of the first group and one of the second group using a first and a second combination rule, respectively, to obtain a scale parameter of the first and second sets of scale parameters.

53. 发明申请

US20230085596A1 Self-Supervised Audio Representation Learning for Mobile Devices 有权
公开(公告)号：US20230085596A1
公开(公告)日：2023-03-16
申请号：US17986477
申请日：2022-11-14
申请人： Google LLC
发明人： Beat Gfeller , Dominik Roblek , Félix de Chaumont Quitry , Marco Tagliasacchi
IPC分类号： G10L19/035 , G06N20/00 , G10L19/038 , G10L25/18
摘要： Systems and methods for training a machine-learned model are provided. A method can include can include obtaining an unlabeled audio signal, sampling the unlabeled audio signal to select one or more sampled slices, inputting the one or more sampled slices into a machine-learned model, receiving, as an output of the machine-learned model, one or more determined characteristics associated with the audio signal, determining a loss function for the machine-learned model based at least in part on a difference between the one or more determined characteristics and one or more corresponding ground truth characteristics of the audio signal, and training the machine-learned model from end to end based at least in part on the loss function. The one or more determined characteristics can include one or more reconstructed portions of the audio signal temporally adjacent to the one or more sampled slices or an estimated distance between two sampled slices.

54. 发明申请

US20230025447A1 QUANTIZATION SCALE FACTOR DETERMINATION DEVICE AND QUANTIZATION SCALE FACTOR DETERMINATION METHOD 有权
公开(公告)号：US20230025447A1
公开(公告)日：2023-01-26
申请号：US17768801
申请日：2020-09-04
申请人： Panasonic Intellectual Property Corporation of America
发明人： Akira HARADA , Hiroyuki EHARA
IPC分类号： G10L19/035
摘要： This quantization scale factor determination device is provided with a correction circuit which corrects an initial value of a quantization scale factor on the basis of whether or not an audio signal spectrum is sparse, and a search circuit which searches for a quantization scale factor on the basis of the initial value.

55. 发明授权

US11545164B2 Audio signal encoding and decoding 有权
公开(公告)号：US11545164B2
公开(公告)日：2023-01-03
申请号：US16619039
申请日：2018-06-15
申请人： RTX A/S
发明人： Jens Toftgaard Petersen
IPC分类号： G10L19/005 , G10L19/035 , G10L19/08
摘要： An audio codec suitable for robust wireless transmission of high quality audio with low latency, still at a moderate bit rate. The encoding and decoding methods are based on ADPCM and in addition to the encoded output bits APM, additional data QB are included in output data blocks, namely data QB representing an internal value of the adaptive quantization ADQ of the ADPCM encoding algorithm, especially a scaling factor encoded and truncated to such as 8 bits. Further, output data blocks preferably include data CFB representing an internal value of the predictor PR of the ADPCM encoding algorithm, especially data CFB representing coefficients of a lattice prediction FIR filter which, truncated to such as 8 bits, can be sequentially included in output data blocks. These additional data QB, CFB regarding internal values of the ADPCM encoding algorithm can be utilized at the encoder side to increase robustness against loss of data blocks in wireless transmission. Especially, the decoding algorithm may comprise comparing its current internal ADPCM decoding values corresponding to the received internal values QB, CFB from the encoder, and in case there is a difference, the decoder can adapt or overwrite its internal values to the ones received QB, CFB. This helps to ensure fast recovery after lost data blocks, thereby ensuring robustness against artefacts in the reconstructed signal, e.g. clicks in case of audio.

56. 发明申请

US20210398547A1 METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO SIGNAL TO REDUCE QUANTIZATION NOISE 有权
公开(公告)号：US20210398547A1
公开(公告)日：2021-12-23
申请号：US17331416
申请日：2021-05-26
申请人： Electronics and Telecommunications Research Institute
发明人： Seung Kwon BEACK , Jongmo SUNG , Mi Suk LEE , Tae Jin LEE , Woo-taek LIM , Inseon JANG
IPC分类号： G10L19/035 , G10L19/022 , G10L19/06 , G10L19/16
摘要： An audio signal encoding method performed by an encoder includes identifying an audio signal of a time domain in units of a block, generating a combined block by combining i) a current original block of the audio signal and ii) a previous original block chronologically adjacent to the current original block, extracting a first residual signal of a frequency domain from the combined block using linear predictive coding of a time domain, overlapping chronologically adjacent first residual signals among first residual signals converted into a time domain, and quantizing a second residual signal of a time domain extracted from the overlapped first residual signal by converting the second residual signal of the time domain into a frequency domain using linear predictive coding of a frequency domain.

57. 发明授权

US11094332B2 Low-complexity tonality-adaptive audio signal quantization 有权
公开(公告)号：US11094332B2
公开(公告)日：2021-08-17
申请号：US16583119
申请日：2019-09-25
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Martin Dietz , Guillaume Fuchs , Christian Helmrich , Goran Markovic
IPC分类号： G10L19/00 , G10L21/00 , G10L19/035 , G10L19/02 , G10H1/06 , G10L25/18 , G10L25/21 , G10L25/45 , G10L25/03
摘要： The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.

58. 发明授权

US10827175B2 Signal encoding method and apparatus and signal decoding method and apparatus 有权
公开(公告)号：US10827175B2
公开(公告)日：2020-11-03
申请号：US16259341
申请日：2019-01-28
申请人： SAMSUNG ELECTRONICS CO., LTD.
发明人： Ho-sang Sung , Ki-hyun Choo , Eun-mi Oh
IPC分类号： G10L19/16 , H04N19/124 , H04N19/196 , H04N19/40 , G10L19/032 , G10L19/24 , H03M13/05 , H03M13/15 , H03M13/31 , G10L19/035 , G10L19/038 , G10L19/02
摘要： A spectrum coding method includes quantizing spectral data of a current band based on a first quantization scheme, generating a lower bit of the current band using the spectral data and the quantized spectral data, quantizing a sequence of lower bits including the lower bit of the current band based on a second quantization scheme, and generating a bitstream based on a upper bit excluding N bits, where N is 1 or greater, from the quantized spectral data and the quantized sequence of lower bits.

59. 发明授权

US10735734B2 Source coding scheme using entropy coding to code a quantized signal 有权
公开(公告)号：US10735734B2
公开(公告)日：2020-08-04
申请号：US16448511
申请日：2019-06-21
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Guillaume Fuchs , Benjamin Schubert , Grzegorz Pietrzyk , Markus Multrus , Bernhard Grill
IPC分类号： H04N7/18 , H04N19/126 , G10L19/035 , H04N19/13 , H04N19/184 , H04N19/44 , H04N19/70 , H04N19/91
摘要： Embodiments of the present invention provide an encoder including a quantization stage, an entropy encoder, a residual quantization stage and a coded signal former. The quantization stage is configured to quantize an input signal using a dead zone in order to obtain a plurality of quantized values. The entropy encoder is configured to encode the plurality of quantized values using an entropy encoding scheme in order to obtain a plurality of entropy encoded values. The residual quantization stage is configured to quantize a residual signal caused by the quantization stage, wherein the residual quantization stage is configured to determine at least one quantized residual value in dependence on the dead zone of the quantization stage. The coded signal former is configured to form a coded signal from the plurality of entropy encoded values and the at least one quantized residual value.

60. 发明授权

US10734010B2 Decoding audio bitstreams with enhanced spectral band replication metadata in at least one fill element 有权
公开(公告)号：US10734010B2
公开(公告)日：2020-08-04
申请号：US16568802
申请日：2019-09-12
申请人： DOLBY INTERNATIONAL AB
发明人： Lars Villemoes , Heiko Purnhagen , Per Ekstrand
IPC分类号： G10L19/16 , G10L19/035 , G10L19/24 , G10L21/038
摘要： Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

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