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41.
公开(公告)号:CA2940382A1
公开(公告)日:2015-09-03
申请号:CA2940382
申请日:2015-02-26
Applicant: ERICSSON TELEFON AB L M (PUBL)
Inventor: SVEDBERG JONAS
IPC: H03M7/30 , G10L19/038 , H04N19/124
Abstract: A method for pyramid vector quantization indexing of audio/video signals comprises obtaining (402) of an integer input vector representing the audio/video signal samples. A leading sign is extracted(404) from the integer input vector. The leading sign is a sign of a terminal non-zero coefficient in the integer input vector. The terminal non-zero coefficient is one of a first non-zero coefficient and a last non-zero coefficient in the integer input vector. The integer input vector is indexed (406) with a pyramid vector quantization enumeration scheme into an output index representing the audio/video signal samples. The pyramid vector quantization enumeration scheme is designed for neglecting the sign of the terminal non-zero coefficient. The output index and the leading sign are outputted (408). A corresponding method for de-indexing, an encoder, a decoder, and computer programs therefore are also disclosed.
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公开(公告)号:CA2340160C
公开(公告)日:2010-11-30
申请号:CA2340160
申请日:1999-09-10
Applicant: ERICSSON TELEFON AB L M
Inventor: JOHANSSON INGEMAR , SVEDBERG JONAS , UVLIDEN ANDERS
Abstract: In producing an approximation of an original speech signal from encoded information about the original speech signal, current parameters (EnPar(i)) associated with a current segment of the original speech signal are determined from the encoded information. Reproduction of a noise component of the original speech signal is improved by using at least one of the current parameters and corresponding previous parameters respectively associated with previous segments of the original speech signal (31, 37, 39) to produce a modified parameter (EnPar(i)mod). The modified parameter is then used (25, 40) to produce an approximation of the current segment of the original speech signal.
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公开(公告)号:CA2343191C
公开(公告)日:2009-10-27
申请号:CA2343191
申请日:1999-08-24
Applicant: ERICSSON TELEFON AB L M
Inventor: UVLIDEN ANDERS , SVEDBERG JONAS
Abstract: A multi-codebook fixed bitrate CELP signal block encoder/decoder includes a codebook selector (22) for selecting, for each signal block, a corresponding codebook identification in accordance with a deterministic selection procedure that i s independent of signal type. Included are also means for encoding/decoding each signal block by using a codebook having the selected codebook identification.
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公开(公告)号:BRPI0520720A2
公开(公告)日:2009-06-13
申请号:BRPI0520720
申请日:2005-11-30
Applicant: ERICSSON TELEFON AB L M
Inventor: SANDGREN NICKLAS , SVEDBERG JONAS
IPC: G10L19/14 , G10L19/012 , G10L19/16
Abstract: Speech frames of a first speech coding scheme are utilized (210) as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified (216), preferably either by determining (214) an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted (218). The present invention also presents transcoders and communications systems providing such transcoding functionality.
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公开(公告)号:AT414315T
公开(公告)日:2008-11-15
申请号:AT01932448
申请日:2001-05-10
Applicant: ERICSSON TELEFON AB L M
Inventor: SVEDBERG JONAS , SUNDQVIST JIM , UVLIDEN ANDERS , NOHLGREN ANDERS , WESTERLUND MAGNUS
Abstract: An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such "interaction," for instance, may take the form of updating states in one model using the other model.
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公开(公告)号:AT363710T
公开(公告)日:2007-06-15
申请号:AT01963659
申请日:2001-09-05
Applicant: ERICSSON TELEFON AB L M
Inventor: MINDE TOR , STEINARSON ARNE , SVEDBERG JONAS , LUNDBERG TOMAS
Abstract: A multi-channel linear predictive analysis-by-synthesis signal encoding method detects (S 26 , S 27 ) inter-channel correlation and select one of several possible encoding modes (S 24, S 29, S 30 ) based on the detected correlation.
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47.
公开(公告)号:MY124630A
公开(公告)日:2006-06-30
申请号:MYPI9905074
申请日:1999-11-20
Applicant: ERICSSON TELEFON AB L M
Inventor: JOHANSSON INGEMAR , EKUDDEN ERIK , SVEDBERG JONAS , UVLIDEN ANDERS
Abstract: PERCEPTUALLY RELEVANT NON-SPEECH INFORMATION CAN BE PRESERVED DURING ENCODING OF AN AUDIO SIGNAL BY DETERMINING WHETER THE AUDIO SIGNAL INCLUDES SUCH INFORMATION (122, 124, 125). IF SO, A SPEECH/ NOISE CLASSIFICATION OF THE AUDIO SIGNAL IS OVERRIDDEN (43) TO PREVENT MISCLASSIFICATION OF THE AUDIO SIGNAL AS NOISE.(FIGURE 1)
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公开(公告)号:DE69929069D1
公开(公告)日:2006-01-26
申请号:DE69929069
申请日:1999-08-24
Applicant: ERICSSON TELEFON AB L M
Inventor: UVLIDEN ANDERS , SVEDBERG JONAS
Abstract: A multi-codebook fixed bitrate CELP signal block encoder/decoder includes a codebook selector ( 22 ) for selecting, for each signal block, a corresponding codebook identification in accordance with a deterministic selection procedure that is independent of signal type. Included are also means for encoding/decoding each signal block by using a codebook having the selected codebook identification.
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公开(公告)号:AU5897301A
公开(公告)日:2001-11-20
申请号:AU5897301
申请日:2001-05-10
Applicant: ERICSSON TELEFON AB L M
Inventor: SVEDBERG JONAS , SUNDQVIST JIM , UVLIDEN ANDERS , NOHLGREN ANDERS , WESTERLUND MAGNUS
Abstract: An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such "interaction," for instance, may take the form of updating states in one model using the other model.
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公开(公告)号:ZA200101222B
公开(公告)日:2001-08-16
申请号:ZA200101222
申请日:2001-02-13
Applicant: ERICSSON TELEFON AB L M
Inventor: JOHANSSON INGEMAR , SVEDBERG JONAS , UVLIDEN ANDERS
Abstract: In producing an approximation of an original speech signal from encoded information about the original speech signal, current parameters (EnPar(i)) associated with a current segment of the original speech signal are determined from the encoded information. Reproduction of a noise component of the original speech signal is improved by using at least one of the current parameters and corresponding previous parameters respectively associated with previous segments of the original speech signal (31, 37, 39) to produce a modified parameter (EnPar(i) mod ). The modified parameter is then used (25, 40) to produce an approximation of the current segment of the original speech signal.
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