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Speech Coding (SPE-CODI)

NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN


Nonnegative matrix factorization (NMF) has recently been applied to temporal decomposition (TD) of speech spectral envelopes represented by line spectral frequencies. A couple of inherent TD constraints, which are otherwise handled as ad hoc exceptions, has also been incorporated using NMF, including LSF ordering and monotonic event functions. Here, these constraints are analyzed and a third inherent constraint is incorporated into an NMF analysis.

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Authors:
Miguel Arjona Ramírez
Submitted On:
27 February 2017 - 10:13pm
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ICASSP2017marjonaramirezSP-P1.10

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[1] Miguel Arjona Ramírez, "NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN", IEEE SigPort, 2017. [Online]. Available: http://sigport.org/1468. Accessed: Sep. 21, 2017.
@article{1468-17,
url = {http://sigport.org/1468},
author = {Miguel Arjona Ramírez },
publisher = {IEEE SigPort},
title = {NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN},
year = {2017} }
TY - EJOUR
T1 - NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN
AU - Miguel Arjona Ramírez
PY - 2017
PB - IEEE SigPort
UR - http://sigport.org/1468
ER -
Miguel Arjona Ramírez. (2017). NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN. IEEE SigPort. http://sigport.org/1468
Miguel Arjona Ramírez, 2017. NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN. Available at: http://sigport.org/1468.
Miguel Arjona Ramírez. (2017). "NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN." Web.
1. Miguel Arjona Ramírez. NON-NEGATIVE TEMPORAL DECOMPOSITION REGULARIZATION WITH AN AUGMENTED LAGRANGIAN [Internet]. IEEE SigPort; 2017. Available from : http://sigport.org/1468

System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding


The ITU-T Recommendation G.722 about subband adaptive differential pulse code modulation (SB-ADPCM) is the mandatory wideband speech codec in the new generation digital enhanced cordless telephony (NG-DECT). Although in ADPCM the difference signal instead of the original signal is quantized and adaptive prediction is employed, redundancy is yet observed within the quantized samples. In this paper we apply a soft-decision speech decoding technique which exploits this redundancy in terms of a priori knowledge and the channel reliability information to NG-DECT.

poster.pdf

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Authors:
Sai Han, Angel M. Gomez, José Luis Pérez-Córdoba, Tim Fingscheidt
Submitted On:
19 March 2016 - 8:58am
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poster.pdf

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[1] Sai Han, Angel M. Gomez, José Luis Pérez-Córdoba, Tim Fingscheidt, "System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding", IEEE SigPort, 2016. [Online]. Available: http://sigport.org/808. Accessed: Sep. 21, 2017.
@article{808-16,
url = {http://sigport.org/808},
author = {Sai Han; Angel M. Gomez; José Luis Pérez-Córdoba; Tim Fingscheidt },
publisher = {IEEE SigPort},
title = {System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding},
year = {2016} }
TY - EJOUR
T1 - System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding
AU - Sai Han; Angel M. Gomez; José Luis Pérez-Córdoba; Tim Fingscheidt
PY - 2016
PB - IEEE SigPort
UR - http://sigport.org/808
ER -
Sai Han, Angel M. Gomez, José Luis Pérez-Córdoba, Tim Fingscheidt. (2016). System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding. IEEE SigPort. http://sigport.org/808
Sai Han, Angel M. Gomez, José Luis Pérez-Córdoba, Tim Fingscheidt, 2016. System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding. Available at: http://sigport.org/808.
Sai Han, Angel M. Gomez, José Luis Pérez-Córdoba, Tim Fingscheidt. (2016). "System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding." Web.
1. Sai Han, Angel M. Gomez, José Luis Pérez-Córdoba, Tim Fingscheidt. System-Compatible Robustness Improvement for New Generation DECT Decoders by G.722 Soft-Decision Decoding [Internet]. IEEE SigPort; 2016. Available from : http://sigport.org/808

Sparse Reconstruction of Quantized Speech Signals


We propose sparse reconstruction techniques to improve the quality and / or reduce the bit-rate of standard speech coders. To that end, we assume signal sparsity in some transform domain and formulate the problem of reconstructing the original signal in terms of constrained l1-norm minimization. We use modern primal-dual methods in order to solve the resulting non-smooth convex optimization problem. Experiments show that with the proposed sparse reconstruction method the instrumentally predicted speech quality can be largely improved.

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Authors:
Christoph Brauer, Timo Gerkmann, Dirk Lorenz
Submitted On:
17 March 2016 - 12:42pm
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icassp_poster_brauer.pdf

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[1] Christoph Brauer, Timo Gerkmann, Dirk Lorenz, "Sparse Reconstruction of Quantized Speech Signals", IEEE SigPort, 2016. [Online]. Available: http://sigport.org/744. Accessed: Sep. 21, 2017.
@article{744-16,
url = {http://sigport.org/744},
author = {Christoph Brauer; Timo Gerkmann; Dirk Lorenz },
publisher = {IEEE SigPort},
title = {Sparse Reconstruction of Quantized Speech Signals},
year = {2016} }
TY - EJOUR
T1 - Sparse Reconstruction of Quantized Speech Signals
AU - Christoph Brauer; Timo Gerkmann; Dirk Lorenz
PY - 2016
PB - IEEE SigPort
UR - http://sigport.org/744
ER -
Christoph Brauer, Timo Gerkmann, Dirk Lorenz. (2016). Sparse Reconstruction of Quantized Speech Signals. IEEE SigPort. http://sigport.org/744
Christoph Brauer, Timo Gerkmann, Dirk Lorenz, 2016. Sparse Reconstruction of Quantized Speech Signals. Available at: http://sigport.org/744.
Christoph Brauer, Timo Gerkmann, Dirk Lorenz. (2016). "Sparse Reconstruction of Quantized Speech Signals." Web.
1. Christoph Brauer, Timo Gerkmann, Dirk Lorenz. Sparse Reconstruction of Quantized Speech Signals [Internet]. IEEE SigPort; 2016. Available from : http://sigport.org/744

On the influence of quantization on the identifiability of emotions from voice coding parameters


Although emotions play a major role in voice communication, the quality of their reproduction by low bit rate voice coders has never been investigated so far. This paper shows that the emotional state of a speaker can be identified automatically, with reasonable precision and accuracy, using conventional voice coding parameters (pitch, voicing, energy and LPC coefficients). It also shows that the performance of this identification degrades when these parameters are quantized, especially at lower rates (1200 bits/s).

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Authors:
Patrick Robitaille, Samuel Trempe, Philippe Gournay, Roch Lefebre
Submitted On:
14 March 2016 - 2:24pm
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ICASSP2016_Robitaille (poster).pdf

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[1] Patrick Robitaille, Samuel Trempe, Philippe Gournay, Roch Lefebre, "On the influence of quantization on the identifiability of emotions from voice coding parameters", IEEE SigPort, 2016. [Online]. Available: http://sigport.org/679. Accessed: Sep. 21, 2017.
@article{679-16,
url = {http://sigport.org/679},
author = {Patrick Robitaille; Samuel Trempe; Philippe Gournay; Roch Lefebre },
publisher = {IEEE SigPort},
title = {On the influence of quantization on the identifiability of emotions from voice coding parameters},
year = {2016} }
TY - EJOUR
T1 - On the influence of quantization on the identifiability of emotions from voice coding parameters
AU - Patrick Robitaille; Samuel Trempe; Philippe Gournay; Roch Lefebre
PY - 2016
PB - IEEE SigPort
UR - http://sigport.org/679
ER -
Patrick Robitaille, Samuel Trempe, Philippe Gournay, Roch Lefebre. (2016). On the influence of quantization on the identifiability of emotions from voice coding parameters. IEEE SigPort. http://sigport.org/679
Patrick Robitaille, Samuel Trempe, Philippe Gournay, Roch Lefebre, 2016. On the influence of quantization on the identifiability of emotions from voice coding parameters. Available at: http://sigport.org/679.
Patrick Robitaille, Samuel Trempe, Philippe Gournay, Roch Lefebre. (2016). "On the influence of quantization on the identifiability of emotions from voice coding parameters." Web.
1. Patrick Robitaille, Samuel Trempe, Philippe Gournay, Roch Lefebre. On the influence of quantization on the identifiability of emotions from voice coding parameters [Internet]. IEEE SigPort; 2016. Available from : http://sigport.org/679

Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec

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23 February 2016 - 1:44pm
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kmhGlobalSIP2015ALW_1212a.pdf

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[1] , "Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec", IEEE SigPort, 2015. [Online]. Available: http://sigport.org/489. Accessed: Sep. 21, 2017.
@article{489-15,
url = {http://sigport.org/489},
author = { },
publisher = {IEEE SigPort},
title = {Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec},
year = {2015} }
TY - EJOUR
T1 - Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec
AU -
PY - 2015
PB - IEEE SigPort
UR - http://sigport.org/489
ER -
. (2015). Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec. IEEE SigPort. http://sigport.org/489
, 2015. Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec. Available at: http://sigport.org/489.
. (2015). "Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec." Web.
1. . Adaptive selection of lag-window shape for linear predictive analysis in the 3GPP EVS codec [Internet]. IEEE SigPort; 2015. Available from : http://sigport.org/489

Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression


The harmonic plus noise model (HNM) is widely used for the modeling of audio signals. In this paper, we introduce perceptual frequency masking to the 2-band HNM, developed by Stylianou et al., applied to speech signals. An auditory model is used to recognize inaudible sinusoids, which will be removed from the set of model’s parameters in order to reduce the data size for speech coding.

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Authors:
Sonia Djaziri-Larbi
Submitted On:
23 February 2016 - 1:44pm
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GlobalSIPFBA_SL.pdf

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[1] Sonia Djaziri-Larbi, "Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression ", IEEE SigPort, 2015. [Online]. Available: http://sigport.org/400. Accessed: Sep. 21, 2017.
@article{400-15,
url = {http://sigport.org/400},
author = {Sonia Djaziri-Larbi },
publisher = {IEEE SigPort},
title = {Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression },
year = {2015} }
TY - EJOUR
T1 - Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression
AU - Sonia Djaziri-Larbi
PY - 2015
PB - IEEE SigPort
UR - http://sigport.org/400
ER -
Sonia Djaziri-Larbi. (2015). Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression . IEEE SigPort. http://sigport.org/400
Sonia Djaziri-Larbi, 2015. Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression . Available at: http://sigport.org/400.
Sonia Djaziri-Larbi. (2015). "Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression ." Web.
1. Sonia Djaziri-Larbi. Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression [Internet]. IEEE SigPort; 2015. Available from : http://sigport.org/400

Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression


The harmonic plus noise model (HNM) is widely used for the modeling of audio signals. In this paper, we introduce perceptual frequency masking to the 2-band HNM, developed by Stylianou et al., applied to speech signals. An auditory model is used to recognize inaudible sinusoids, which will be removed from the set of model’s parameters in order to reduce the data size for speech coding.

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Authors:
Sonia Djaziri-Larbi
Submitted On:
23 February 2016 - 1:44pm
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GlobalSIPFBA_SL.pdf

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[1] Sonia Djaziri-Larbi, "Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression ", IEEE SigPort, 2015. [Online]. Available: http://sigport.org/399. Accessed: Sep. 21, 2017.
@article{399-15,
url = {http://sigport.org/399},
author = {Sonia Djaziri-Larbi },
publisher = {IEEE SigPort},
title = {Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression },
year = {2015} }
TY - EJOUR
T1 - Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression
AU - Sonia Djaziri-Larbi
PY - 2015
PB - IEEE SigPort
UR - http://sigport.org/399
ER -
Sonia Djaziri-Larbi. (2015). Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression . IEEE SigPort. http://sigport.org/399
Sonia Djaziri-Larbi, 2015. Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression . Available at: http://sigport.org/399.
Sonia Djaziri-Larbi. (2015). "Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression ." Web.
1. Sonia Djaziri-Larbi. Perceptual Long-Term Harmonic plus Noise Modeling for Speech Data Compression [Internet]. IEEE SigPort; 2015. Available from : http://sigport.org/399

Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec


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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Authors:
Hiroyuki Ehara, Takuya Kawashima, Lasse Laaksonen, Hosang Sung, Kihyun Choo
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23 February 2016 - 1:44pm
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GlobalSIP2015_r1.6.pdf

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GlobalSIP2015_r1.6.pdf

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[1] Hiroyuki Ehara, Takuya Kawashima, Lasse Laaksonen, Hosang Sung, Kihyun Choo, "Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec", IEEE SigPort, 2015. [Online]. Available: http://sigport.org/358. Accessed: Sep. 21, 2017.
@article{358-15,
url = {http://sigport.org/358},
author = {Hiroyuki Ehara; Takuya Kawashima; Lasse Laaksonen; Hosang Sung; Kihyun Choo },
publisher = {IEEE SigPort},
title = {Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec},
year = {2015} }
TY - EJOUR
T1 - Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec
AU - Hiroyuki Ehara; Takuya Kawashima; Lasse Laaksonen; Hosang Sung; Kihyun Choo
PY - 2015
PB - IEEE SigPort
UR - http://sigport.org/358
ER -
Hiroyuki Ehara, Takuya Kawashima, Lasse Laaksonen, Hosang Sung, Kihyun Choo. (2015). Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec. IEEE SigPort. http://sigport.org/358
Hiroyuki Ehara, Takuya Kawashima, Lasse Laaksonen, Hosang Sung, Kihyun Choo, 2015. Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec. Available at: http://sigport.org/358.
Hiroyuki Ehara, Takuya Kawashima, Lasse Laaksonen, Hosang Sung, Kihyun Choo. (2015). "Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec." Web.
1. Hiroyuki Ehara, Takuya Kawashima, Lasse Laaksonen, Hosang Sung, Kihyun Choo. Super-Wideband Fine Spectrum Quantization for Low-rate High-Quality MDCT Coding Mode of The 3GPP EVS Codec [Internet]. IEEE SigPort; 2015. Available from : http://sigport.org/358

Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec


This presentation describes the bandwidth extension (BWE) method developed for the AMR-WB interoperable (AMR-WB IO) modes of the 3GPP EVS codec. The low-band signal (0-6.4 kHz) is coded using an enhanced version of ACELP as in AMR-WB and post-processed; the high-band (above 6.4 kHz) in contrast to AMR-WB is represented with a new BWE method. The decoded low-band excitation is adaptively extended to high frequencies and filtered in the DCT domain. The extended excitation is scaled by subframe gains and shaped by a weighted LPC synthesis filter.

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Authors:
Magdalena Kaniewska, Stéphane Ragot, Zexin Liu, Lei Miao, Xingtao Zhang, Jon Gibbs
Submitted On:
23 February 2016 - 1:44pm
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EVS_AMR_WB_IO_BWE_GlobalSIP2015.ppt

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[1] Magdalena Kaniewska, Stéphane Ragot, Zexin Liu, Lei Miao, Xingtao Zhang, Jon Gibbs, "Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec", IEEE SigPort, 2015. [Online]. Available: http://sigport.org/317. Accessed: Sep. 21, 2017.
@article{317-15,
url = {http://sigport.org/317},
author = {Magdalena Kaniewska; Stéphane Ragot; Zexin Liu; Lei Miao; Xingtao Zhang; Jon Gibbs },
publisher = {IEEE SigPort},
title = {Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec},
year = {2015} }
TY - EJOUR
T1 - Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec
AU - Magdalena Kaniewska; Stéphane Ragot; Zexin Liu; Lei Miao; Xingtao Zhang; Jon Gibbs
PY - 2015
PB - IEEE SigPort
UR - http://sigport.org/317
ER -
Magdalena Kaniewska, Stéphane Ragot, Zexin Liu, Lei Miao, Xingtao Zhang, Jon Gibbs. (2015). Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec. IEEE SigPort. http://sigport.org/317
Magdalena Kaniewska, Stéphane Ragot, Zexin Liu, Lei Miao, Xingtao Zhang, Jon Gibbs, 2015. Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec. Available at: http://sigport.org/317.
Magdalena Kaniewska, Stéphane Ragot, Zexin Liu, Lei Miao, Xingtao Zhang, Jon Gibbs. (2015). "Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec." Web.
1. Magdalena Kaniewska, Stéphane Ragot, Zexin Liu, Lei Miao, Xingtao Zhang, Jon Gibbs. Enhanced AMR-WB Bandwidth Extension in 3GPP EVS Codec [Internet]. IEEE SigPort; 2015. Available from : http://sigport.org/317

A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching


This presentation describes a novel frequency domain bandwidth extension (BWE) scheme with relaxed synchronization, optimized for coding inactive and music/mixed content signals. The algorithm achieves high subjective quality at low and medium bitrates and it has a low algorithmic delay. The algorithm is part of the 3GPP Enhanced Voice Services (EVS) codec. In addition to the presented algorithm, the EVS codec employs also a time domain BWE scheme optimized for active speech coding.

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Authors:
Lei Miao, Zexin Liu, Xingtao Zhang, Chen Hu, Jon Gibbs, Kihyun Choo, Eunmi Oh
Submitted On:
23 February 2016 - 1:44pm
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EVS_FD_BWE_GlobalSIP2015.pptx

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[1] Lei Miao, Zexin Liu, Xingtao Zhang, Chen Hu, Jon Gibbs, Kihyun Choo, Eunmi Oh, "A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching", IEEE SigPort, 2015. [Online]. Available: http://sigport.org/316. Accessed: Sep. 21, 2017.
@article{316-15,
url = {http://sigport.org/316},
author = {Lei Miao; Zexin Liu; Xingtao Zhang; Chen Hu; Jon Gibbs; Kihyun Choo; Eunmi Oh },
publisher = {IEEE SigPort},
title = {A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching},
year = {2015} }
TY - EJOUR
T1 - A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching
AU - Lei Miao; Zexin Liu; Xingtao Zhang; Chen Hu; Jon Gibbs; Kihyun Choo; Eunmi Oh
PY - 2015
PB - IEEE SigPort
UR - http://sigport.org/316
ER -
Lei Miao, Zexin Liu, Xingtao Zhang, Chen Hu, Jon Gibbs, Kihyun Choo, Eunmi Oh. (2015). A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching. IEEE SigPort. http://sigport.org/316
Lei Miao, Zexin Liu, Xingtao Zhang, Chen Hu, Jon Gibbs, Kihyun Choo, Eunmi Oh, 2015. A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching. Available at: http://sigport.org/316.
Lei Miao, Zexin Liu, Xingtao Zhang, Chen Hu, Jon Gibbs, Kihyun Choo, Eunmi Oh. (2015). "A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching." Web.
1. Lei Miao, Zexin Liu, Xingtao Zhang, Chen Hu, Jon Gibbs, Kihyun Choo, Eunmi Oh. A Novel Frequency Domain BWE with Relaxed Synchronization and Associated BWE Switching [Internet]. IEEE SigPort; 2015. Available from : http://sigport.org/316

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