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Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.


Papers
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Journal Article
TL;DR: An Analysis/Synthesis Audio Codec (ASAC) is presented, which allows the coding of audio signals at very low bit rates for applications like mobile communication or multimedia database access via modem and analog telephone lines.
Abstract: An Analysis/Synthesis Audio Codec (ASAC) is presented, which allows the coding of audio signals at very low bit rates for applications like mobile communication or multimedia database access via modem and analog telephone lines. Compression with bit rates between 6 kbit/s and 24 kbit/s is addressed. Furthermore the implementation of special effects like independent pitch change and speed change in the decoder is described.

70 citations

Patent
28 Feb 2003
TL;DR: In this paper, a video coding-decoding (CODEC) method in an error resilient mode, a computer readable medium having a computer program for the video CODEC method, and a video-coding apparatus.
Abstract: A video coding-decoding (CODEC) method in an error resilient mode, a computer readable medium having a computer program for the video CODEC method, and a video CODEC apparatus. The video CODEC method provides more resilience against channel error such that communications are less affected by error under conditions in which errors are a serious problem such as in a wireless communications channel. In the video CODEC method, a header data part (HDP) bit region, a motion vector data part (MVDP) bit region and a discrete cosine transform data part (DDP) bit regions are partitioned from each macro block of the video data in an error resilient mode, and then the partitioned bit regions are variable-length-coded. Then, the bit regions selected from the variable-length coded bit regions according to a predetermined priority for recovery are reversible-variable-length-coded, and markers are then inserted into the variable-length coded or reversible-variable-length-coded bit regions.

70 citations

Proceedings ArticleDOI
12 May 2008
TL;DR: The key element of the method is an alternative search strategy for the ACELP codebook which allows for joint data hiding and speech coding and it is pointed out that the method can also be exploited to reduce the codec bit rate.
Abstract: A new method for hiding digital data in the bitstream of an ACELP speech codec is proposed in this paper. The key element of our method is an alternative search strategy for the ACELP codebook which allows for joint data hiding and speech coding. The concept has been examplarily applied to the AMR speech codec (12.2 kbit/s mode) and it is shown that steganographic data can be reliably transmitted at a rate of up to 2 kbit/s both with a negligible effect on the subjective quality of the coded speech and with reasonable computational complexity. Apart from data hiding, it is further pointed out that our method can also be exploited to reduce the codec bit rate.

70 citations

Proceedings ArticleDOI
A.J. Accardi1, R.V. Cox
15 Mar 1999
TL;DR: In this paper, a modified version of Ephraim and van trees's (see IEEE Trans. Speech and Audio Proc., vol.3, p.251-66, 1995) spectral domain constrained signal subspace estimator is used in this manner, obtaining a system with greater flexibility and similar performance.
Abstract: Ephraim and Malah's (1984, 1985) MMSE-LSA speech enhancement algorithm, while robust and effective, is difficult to tune and adjust for the tradeoff between noise reduction and distortion. We suggest a means of generalizing this design, which allows for other estimators besides the MMSE-LSA to be used within the same supporting framework. When a modified version of Ephraim and Van Trees's (see IEEE Trans. Speech and Audio Proc., vol.3, p.251-66, 1995) spectral domain constrained signal subspace estimator is used in this manner, we obtain a system with greater flexibility and similar performance. We also explore the possibility of using different speech enhancement techniques as pre-processors for different parameter extraction modules of the IS-641 speech coder (a 7.4 kbit/s ACELP codec). We show that such a strategy can increase the quality of the coded speech and lead to a system that is more robust to differing noise types.

70 citations

Proceedings ArticleDOI
S.A. Ramprashad1
12 May 1998
TL;DR: A two stage hybrid embedded speech/audio coding structure uses a speech coder as a core to provide the minimal bitrate and an acceptable performance on speech inputs and a transform coder using a modified discrete cosine transform and perceptual coding principles is proposed.
Abstract: A two stage hybrid embedded speech/audio coding structure is proposed. The structure uses a speech coder as a core to provide the minimal bitrate and an acceptable performance on speech inputs. The second stage is a transform coder using a modified discrete cosine transform (MDCT) and perceptual coding principles. This stage is itself embedded both in complexity and bitrate, and provides various levels of enhancement of the core output, particularly for general audio signals like music. Informal A-B comparison tests show that the performance of the structure at 16 kb/s is between that of the GSM enhanced full rate coder at 12.2 kb/s, and the G.728 LD-CELP coder at 16 kb/s.

69 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202310
202214
20201
20193
20183
201721