The Experts below are selected from a list of 297 Experts worldwide ranked by ideXlab platform
Stefan Bruhn - One of the best experts on this subject based on the ideXlab platform.
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A novel sinusoidal approach to audio signal frame loss concealment and its application in the new evs Codec standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Stefan Bruhn, Erik Norvell, Jonas Svedberg, Sigurdur SverrissonAbstract:The new 3GPP Codec for Enhanced Voice Services (EVS) comprises a collection of frame loss concealment techniques, each specifically designed for the different coding modes of that Codec. One of them, called “Phase Error Concealment Unit (Phase ECU)”, was developed for the High Quality (HQ) MDCT coding mode. Despite this target application, Phase ECU is a generic stand-alone tool operating on a buffer of the previously decoded and reconstructed time signal. Its framework is based on the sinusoidal analysis and synthesis paradigm. Besides a description of the basic technology we present optimizations and adaptations required for meeting the challenging 3GPP EVS Codec Performance requirements, and that make the method robust for a broad range of audio signals under various frame loss conditions from isolated frame erasures to severe burst loss. Test results are reported that show significant improvements over traditional techniques.
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ICASSP - A novel sinusoidal approach to audio signal frame loss concealment and its application in the new evs Codec standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Stefan Bruhn, Erik Norvell, Jonas Svedberg, Sigurdur SverrissonAbstract:The new 3GPP Codec for Enhanced Voice Services (EVS) comprises a collection of frame loss concealment techniques, each specifically designed for the different coding modes of that Codec. One of them, called “Phase Error Concealment Unit (Phase ECU)”, was developed for the High Quality (HQ) MDCT coding mode. Despite this target application, Phase ECU is a generic stand-alone tool operating on a buffer of the previously decoded and reconstructed time signal. Its framework is based on the sinusoidal analysis and synthesis paradigm. Besides a description of the basic technology we present optimizations and adaptations required for meeting the challenging 3GPP EVS Codec Performance requirements, and that make the method robust for a broad range of audio signals under various frame loss conditions from isolated frame erasures to severe burst loss. Test results are reported that show significant improvements over traditional techniques.
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EUSIPCO - ITU-T EV-VBR: A robust 8-32 kbit/s scalable coder for error prone telecommunications channels
2008Co-Authors: Tommy Vaillancourt, Lasse Laaksonen, Jon Gibbs, Milan Jelinek, Stefan Bruhn, Jacek Stachurski, Anssi Ramo, Udar Mittal, A.e. Ertan, Volodya GrancharovAbstract:This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) Codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The Codec provides a scalable solution for compression of 16 kHz sampled speech and audio signals at rates between 8 kbit/s and 32 kbit/s, robust to significant rates of frame erasures or packet losses. It comprises 5 layers where higher layer bitstreams can be discarded without affecting the lower layer decoding. The core layer takes advantage of signal-classification based CELP encoding. The second layer reduces the coding error from the first layer by means of additional pitch contribution and another algebraic codebook. The higher layers encode the weighted error signal from lower layers using MDCT transform coding. Sev-eral technologies are used to encode the MDCT coefficients for best Performance both for speech and music. The Codec Performance is demonstrated with selected results from ITU-T Characterization test.
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ICASSP - ITU-T G.EV-VBR baseline Codec
2008 IEEE International Conference on Acoustics Speech and Signal Processing, 2008Co-Authors: Milan Jelinek, Lasse Laaksonen, Jon Gibbs, Tommy Vaillancourt, Jacek Stachurski, Anssi Ramo, A.e. Ertan, Stefan BruhnAbstract:We present the Q.EV-VBR winning candidate Codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 Codec is an embedded Codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The Codec has been designed with the primary objective of a high-Performance wideband speech coding for error- prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The Codec Performance is demonstrated with selected test results.
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ITU-T G.EV-VBR baseline Codec
2008 IEEE International Conference on Acoustics Speech and Signal Processing, 2008Co-Authors: Milan Jelinek, Jon Gibbs, Tommy Vaillancourt, Erdem A. Ertan, Jacek Stachurski, Anssi Ramo, Lasse Laaksonen, Stefan BruhnAbstract:We present the Q.EV-VBR winning candidate Codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 Codec is an embedded Codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The Codec has been designed with the primary objective of a high-Performance wideband speech coding for error- prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The Codec Performance is demonstrated with selected test results.
Milan Jelinek - One of the best experts on this subject based on the ideXlab platform.
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Glottal-Shape Codebook to Improve Robustness of CELP Codecs
IEEE Transactions on Audio Speech and Language Processing, 2010Co-Authors: Václav Eksler, Milan JelinekAbstract:This paper presents a new technique for the class of code-excited linear prediction speech Codecs designed to reduce error propagation after lost frames. Its principle consists in replacing the interframe long-term prediction with a glottal-shape codebook in the subframe containing the first glottal impulse in a given frame. This technique, independent of previous frames, is of particular interest in voiced speech frames following transitions as these frames are the most sensitive to frame erasures. It is a basis of a structured coding scheme called transition coding (TC). The TC greatly improves Codec Performance in noisy channels while maintaining clean channel Performance. It is a part of the new embedded speech and audio Codec recently standardized as Recommendation G.718 by ITU-T.
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EUSIPCO - ITU-T EV-VBR: A robust 8-32 kbit/s scalable coder for error prone telecommunications channels
2008Co-Authors: Tommy Vaillancourt, Lasse Laaksonen, Jon Gibbs, Milan Jelinek, Stefan Bruhn, Jacek Stachurski, Anssi Ramo, Udar Mittal, A.e. Ertan, Volodya GrancharovAbstract:This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) Codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The Codec provides a scalable solution for compression of 16 kHz sampled speech and audio signals at rates between 8 kbit/s and 32 kbit/s, robust to significant rates of frame erasures or packet losses. It comprises 5 layers where higher layer bitstreams can be discarded without affecting the lower layer decoding. The core layer takes advantage of signal-classification based CELP encoding. The second layer reduces the coding error from the first layer by means of additional pitch contribution and another algebraic codebook. The higher layers encode the weighted error signal from lower layers using MDCT transform coding. Sev-eral technologies are used to encode the MDCT coefficients for best Performance both for speech and music. The Codec Performance is demonstrated with selected results from ITU-T Characterization test.
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ICASSP - ITU-T G.EV-VBR baseline Codec
2008 IEEE International Conference on Acoustics Speech and Signal Processing, 2008Co-Authors: Milan Jelinek, Lasse Laaksonen, Jon Gibbs, Tommy Vaillancourt, Jacek Stachurski, Anssi Ramo, A.e. Ertan, Stefan BruhnAbstract:We present the Q.EV-VBR winning candidate Codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 Codec is an embedded Codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The Codec has been designed with the primary objective of a high-Performance wideband speech coding for error- prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The Codec Performance is demonstrated with selected test results.
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ITU-T G.EV-VBR baseline Codec
2008 IEEE International Conference on Acoustics Speech and Signal Processing, 2008Co-Authors: Milan Jelinek, Jon Gibbs, Tommy Vaillancourt, Erdem A. Ertan, Jacek Stachurski, Anssi Ramo, Lasse Laaksonen, Stefan BruhnAbstract:We present the Q.EV-VBR winning candidate Codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 Codec is an embedded Codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The Codec has been designed with the primary objective of a high-Performance wideband speech coding for error- prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The Codec Performance is demonstrated with selected test results.
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ITU-T EV-VBR: A robust 8-32 kbit/s scalable coder for error prone telecommunications channels
2008 16th European Signal Processing Conference, 2008Co-Authors: Tommy Vaillancourt, Jon Gibbs, Milan Jelinek, Stefan Bruhn, Erdem A. Ertan, Jacek Stachurski, Anssi Ramo, Lasse Laaksonen, Udar Mittal, Volodya GrancharovAbstract:This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) Codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The Codec provides a scalable solution for compression of 16 kHz sampled speech and audio signals at rates between 8 kbit/s and 32 kbit/s, robust to significant rates of frame erasures or packet losses. It comprises 5 layers where higher layer bitstreams can be discarded without affecting the lower layer decoding. The core layer takes advantage of signal-classification based CELP encoding. The second layer reduces the coding error from the first layer by means of additional pitch contribution and another algebraic codebook. The higher layers encode the weighted error signal from lower layers using MDCT transform coding. Several technologies are used to encode the MDCT coefficients for best Performance both for speech and music. The Codec Performance is demonstrated with selected results from ITU-T Characterization test.
Lasse Laaksonen - One of the best experts on this subject based on the ideXlab platform.
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INTERSPEECH - Superwideband extension of g.718 and g.729.1 speech Codecs.
2020Co-Authors: Lasse Laaksonen, Mikko Tammi, Vladimir Malenovsky, Tommy Vaillancourt, Tomofumi Yamanashi, Masahiro Oshikiri, Claude Lamblin, Balazs Kovesi, Lei MiaoAbstract:This communication presents the recently standardized superwideband (SWB) extensions of ITU-T G.718 and G.729.1. These extensions were standardized as G.718 annex B and G.729.1 annex E. The SWB functionality is implemented using embedded scalable layers on top of the wideband (WB) core Codecs, and it extends the bit rate of the Codecs to 48 and 64 kbit/s for the G.718 and G.729.1, respectively. The main technology is a two-mode SWB coding method of the high frequencies. In addition, the G.729.1 SWB extension enhances the lower frequency range. The Codec Performance is illustrated with some listening test results extracted from the ITU-T Characterization phase.
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GlobalSIP - Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS Codec
2015 IEEE Global Conference on Signal and Information Processing (GlobalSIP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Hosang Sung, Kihyun ChooAbstract:This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the Performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.
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Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Xuan Zhou, Bin Wang, Lei Miao, Jon Gibbs, Venkatraman Atti, Vivek RajendranAbstract:This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the Performance of the coder is illustrated with some listening test results.
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ICASSP - Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Xuan Zhou, Bin Wang, Lei Miao, Jon Gibbs, Lasse LaaksonenAbstract:This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the Performance of the coder is illustrated with some listening test results.
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Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS Codec
2015 IEEE Global Conference on Signal and Information Processing (GlobalSIP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Hosang Sung, Kihyun ChooAbstract:This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the Performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.
Srikanth Nagisetty - One of the best experts on this subject based on the ideXlab platform.
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GlobalSIP - Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS Codec
2015 IEEE Global Conference on Signal and Information Processing (GlobalSIP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Hosang Sung, Kihyun ChooAbstract:This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the Performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.
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Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Xuan Zhou, Bin Wang, Lei Miao, Jon Gibbs, Venkatraman Atti, Vivek RajendranAbstract:This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the Performance of the coder is illustrated with some listening test results.
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ICASSP - Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Xuan Zhou, Bin Wang, Lei Miao, Jon Gibbs, Lasse LaaksonenAbstract:This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the Performance of the coder is illustrated with some listening test results.
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Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS Codec
2015 IEEE Global Conference on Signal and Information Processing (GlobalSIP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Hosang Sung, Kihyun ChooAbstract:This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the Performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.
Jon Gibbs - One of the best experts on this subject based on the ideXlab platform.
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Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Lasse Laaksonen, Xuan Zhou, Bin Wang, Lei Miao, Jon Gibbs, Venkatraman Atti, Vivek RajendranAbstract:This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the Performance of the coder is illustrated with some listening test results.
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ICASSP - Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard
2015 IEEE International Conference on Acoustics Speech and Signal Processing (ICASSP), 2015Co-Authors: Srikanth Nagisetty, Takuya Kawashima, Hiroyuki Ehara, Xuan Zhou, Bin Wang, Lei Miao, Jon Gibbs, Lasse LaaksonenAbstract:This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized Codec for Enhanced Voice Services. To maximize Codec Performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the Performance of the coder is illustrated with some listening test results.
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EUSIPCO - ITU-T EV-VBR: A robust 8-32 kbit/s scalable coder for error prone telecommunications channels
2008Co-Authors: Tommy Vaillancourt, Lasse Laaksonen, Jon Gibbs, Milan Jelinek, Stefan Bruhn, Jacek Stachurski, Anssi Ramo, Udar Mittal, A.e. Ertan, Volodya GrancharovAbstract:This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) Codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The Codec provides a scalable solution for compression of 16 kHz sampled speech and audio signals at rates between 8 kbit/s and 32 kbit/s, robust to significant rates of frame erasures or packet losses. It comprises 5 layers where higher layer bitstreams can be discarded without affecting the lower layer decoding. The core layer takes advantage of signal-classification based CELP encoding. The second layer reduces the coding error from the first layer by means of additional pitch contribution and another algebraic codebook. The higher layers encode the weighted error signal from lower layers using MDCT transform coding. Sev-eral technologies are used to encode the MDCT coefficients for best Performance both for speech and music. The Codec Performance is demonstrated with selected results from ITU-T Characterization test.
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ICASSP - ITU-T G.EV-VBR baseline Codec
2008 IEEE International Conference on Acoustics Speech and Signal Processing, 2008Co-Authors: Milan Jelinek, Lasse Laaksonen, Jon Gibbs, Tommy Vaillancourt, Jacek Stachurski, Anssi Ramo, A.e. Ertan, Stefan BruhnAbstract:We present the Q.EV-VBR winning candidate Codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 Codec is an embedded Codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The Codec has been designed with the primary objective of a high-Performance wideband speech coding for error- prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The Codec Performance is demonstrated with selected test results.
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ITU-T G.EV-VBR baseline Codec
2008 IEEE International Conference on Acoustics Speech and Signal Processing, 2008Co-Authors: Milan Jelinek, Jon Gibbs, Tommy Vaillancourt, Erdem A. Ertan, Jacek Stachurski, Anssi Ramo, Lasse Laaksonen, Stefan BruhnAbstract:We present the Q.EV-VBR winning candidate Codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 Codec is an embedded Codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The Codec has been designed with the primary objective of a high-Performance wideband speech coding for error- prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The Codec Performance is demonstrated with selected test results.