公开(公告)号：EP2625688A1

公开(公告)日：2013-08-14

申请号：EP11764739.6

申请日：2011-10-04

申请人： Fraunhofer Gesellschaft zur Förderung der angewandten Wissenschaft E.V. ,  VoiceAge Corporation

发明人： MULTRUS, Markus ,  GRILL, Bernhard ,  NEUENDORF, Max ,  RETTELBACH, Nikolaus ,  FUCHS, Guillaume ,  GOURNAY, Philippe ,  LEFEBVRE, Roch ,  BESSETTE, Bruno ,  WILDE, Stephan

IPC分类号： G10L19/00 ,  G10L19/02 ,  G10L21/04

CPC分类号： G10L19/12 ,  G10L19/0204 ,  G10L21/00 ,  G10L2019/0012

摘要： An apparatus for processing an audio signal is provided. The apparatus comprises a signal processor (110; 205; 405) and a configurator (120; 208; 408). The signal processor (110; 205; 405) is adapted to receive a first audio signal frame having a first configurable number of samples of the audio signal, Moreover, the signal processor (110; 205; 405) is adapted to upsample the audio signal by a configurable upsampling factor to obtain a processed audio signal. Furthermore, the signal processor (110; 205; 405) is adapted to output a second audio signal frame having a second configurable number of samples of the processed audio signal. The configurator 120; 208; 408) is adapted to configure the signal processor (110; 205; 405) based on configuration information such that the configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples has a first ratio value. Moreover, the configurator ( 120; 208; 408) is adapted to configure the signal processor (110; 205; 405) such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value. The first or the second ratio value is not an integer value.

公开(公告)号：EP2524374A1

公开(公告)日：2012-11-21

申请号：EP11732606.6

申请日：2011-01-13

申请人： VoiceAge Corporation

发明人： BESSETTE, Bruno

IPC分类号： G10L19/00 ,  G10L19/12

CPC分类号： G10L19/022 ,  G10L19/008 ,  G10L19/0212 ,  G10L19/06 ,  G10L19/12 ,  G10L19/18 ,  G10L21/04

摘要： In a coder, a method for producing forward aliasing cancellation (FAC) parameters for cancelling time-domain aliasing caused to a coded audio signal in a first transform-coded frame by a transition between the first transform-coded frame using a first coding mode with overlapping window and a second frame using a second coding mode with non-overlapping window, comprising: calculating a FAC target representative of a difference between the audio signal of the first frame prior to coding and a sum of synthesis of the coded audio signal of the first transform-coded frame and a time reversed version of last synthesis samples of the second frame as well as a zero point response of a synthesis filter used in the second frame; and weighting the FAC target to produce the FAC parameters. In a decoder, weighted forward aliasing cancellation (FAC) parameters are received and inverse weighted to produce a FAC synthesis. Upon synthesis of the coded audio signal in the first frame, the time-domain aliasing is cancelled from the audio signal synthesis using the FAC synthesis.

公开(公告)号：EP2301021A1

公开(公告)日：2011-03-30

申请号：EP09793768.4

申请日：2009-07-10

申请人： VoiceAge Corporation

发明人： GOURNAY, Philippe ,  BESSETTE, Bruno ,  SALAMI, Redwan

IPC分类号： G10L19/12

CPC分类号： G10L19/06 ,  G10L19/18 ,  H03M7/3082

摘要： A device and a method for quantizing, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantizing device and method comprises: an absolute quantizer for first quantizing one of the LPC filters using absolute quantization; and at least one quantizer of the other LPC filters using a quantization mode selected from the group consisting of absolute quantization and differential quantization relative to at least one previously quantized filter amongst the LPC filters. For inverse quantizing, at least the first quantized LPC filter is received and an inverse quantizer inverse quantizes the first quantized LPC filter using absolute inverse quantization. If any quantized LPC filter other than the first quantized LPC filter is received, an inverse quantizer inverse quantizes this quantized LPC filter using one of absolute inverse quantization and differential inverse quantization relative to at least one previously received quantized LPC filter.

公开(公告)号：EP1719116A1

公开(公告)日：2006-11-08

申请号：EP05706494.1

申请日：2005-02-18

申请人： Voiceage Corporation

发明人： BESSETTE, Bruno

IPC分类号： G10L19/04 ,  G10L19/12

CPC分类号： G10L19/0208 ,  G10L19/005 ,  G10L19/24 ,  G10L19/265 ,  G10L21/0232

摘要： An aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy, and for each block a gain is determined from the factor and is applied to the transform coefficients of the block.

公开(公告)号：EP3764356A1

公开(公告)日：2021-01-13

申请号：EP20182973.6

申请日：2010-06-23

申请人： VoiceAge Corporation

发明人： BESSETTE, Bruno

IPC分类号： G10L19/26 ,  G10L19/022 ,  G10L19/18

摘要： The present invention relates to methods and devices for forward time-domain aliasing cancellation in a coded signal transmitted from a coder to a decoder. Information related to correction of the time-domain aliasing in the coded signal is calculated at the coder and added in a bitstream sent from the coder to the decoder. The decoder receives the bitstream and cancels the time-domain aliasing in the coded signal in response to the information comprised in the bitstream. The information may be representative of a difference between a frame of audio signal to be encoded in a first coding mode and a decoded signal from the frame including time-domain aliasing effects.

公开(公告)号：EP3352168A1

公开(公告)日：2018-07-25

申请号：EP18160922.3

申请日：2010-06-23

申请人： VoiceAge Corporation

发明人： BESSETTE, Bruno

IPC分类号： G10L19/26 ,  G10L19/022 ,  G10L19/18

摘要： The present invention relates to methods and devices for forward time-domain aliasing cancellation in a coded signal transmitted from a coder to a decoder. Information related to correction of the time-domain aliasing in the coded signal is calculated at the coder and added in a bitstream sent from the coder to the decoder. The decoder receives the bitstream and cancels the time-domain aliasing in the coded signal in response to the information comprised in the bitstream. The information may be representative of a difference between a frame of audio signal to be encoded in a first coding mode and a decoded signal from the frame including time-domain aliasing effects.

公开(公告)号：EP2313887B1

公开(公告)日：2017-09-13

申请号：EP09793769.2

申请日：2009-07-10

申请人： VoiceAge Corporation

发明人： BESSETTE, Bruno ,  GOURNAY, Philippe ,  SALAMI, Redwan

IPC分类号： G10L19/06 ,  G10L19/18 ,  H03M7/30

CPC分类号： G10L19/06 ,  G10L19/18 ,  H03M7/3082

公开(公告)号：EP1514355B1

公开(公告)日：2009-08-19

申请号：EP03727093.1

申请日：2003-05-30

申请人： Voiceage Corporation

发明人： BESSETTE, Bruno ,  RAGOT, Stéphane ,  ADOUL, Jean-Pierre

IPC分类号： H03M7/30 ,  G10L19/10 ,  G10L19/06

CPC分类号： G10L19/10 ,  G10L19/06 ,  G10L2019/0008 ,  H03M7/3082

摘要： The present invention relates to a method and system for multi-rate lattice vector quantization of a source vector x representing a frame from a source signal to be used, for example, in digital transmission and storage systems. The multi-rate lattice quantization encoding method comprises the steps of associating to x a lattice point y in a unbounded lattice Lambda; verifying if y is included in a base codebook C derived from the lattice Lambda; if it is the case then indexing y in C so as to yield quantization indices if not then extending the base codebook using, for example a Voronoi based extension method, yielding an extended codebook; associating to y a codevector c from the extended codebook, and indexing y in the extended codebook C. The extension technique allows to obtain higher bit rate codebooks from the base codebooks compared to quantization method and system from the prior art.

公开(公告)号：EP1232494B1

公开(公告)日：2006-08-09

申请号：EP00978928.0

申请日：2000-11-17

申请人： Voiceage Corporation

发明人： BESSETTE, Bruno ,  SALAMI, Redwan ,  LEFEBVRE, Roch

IPC分类号： G10L19/14

CPC分类号： G10L19/083 ,  G10L2019/0012

摘要： The gain smoothing method and device modify the amplitude of an innovative codevector in relation to background noise present in a previously sampled wideband signal. The gain smoothing device comprises a gain smoothing calculator for calculating a smoothing gain in response to a factor representative of voicing in the sampled wideband signal, a factor representative of the stability of a set of linear prediction filter coefficients, and an innovative codebook gain. The gain smoothing device also comprises an amplifier for amplifying the innovative codevector with the smoothing gain to thereby produce a gain-smoothed innovative codevector. The function of the gain-smoothing device improves the perceived synthesized signal when background noise is present in the sampled wideband signal.

公开(公告)号：EP1125276B1

公开(公告)日：2003-08-06

申请号：EP99952199.0

申请日：1999-10-27

申请人： Voiceage Corporation

发明人： BESSETTE, Bruno ,  SALAMI, Redwan ,  LEFEBVRE, Roch

IPC分类号： G10L11/04

CPC分类号： G10L19/26 ,  G10L2019/0011

摘要： A pitch search method and device for digitally encoding a wideband signal, in particular but not exclusively a speech signal, in view of transmitting, or storing, and synthesizing this wideband sound signal. The new method and device which achieve efficient modeling of the harmonic structure of the speech spectrum uses several forms of low pass filters applied to a pitch codevector, the one yielding higher prediction gain (i.e. the lowest pitch prediction error) is selected and the associated pitch codebook parameters are forwarded.