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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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Proceedings Article
01 Sep 1996
TL;DR: This paper further extends a multiple layer video codec using affine motion compensation by incorporating a new block level and designing a coding control strategy, which makes the codec perform efficiently at very low bit rate and for small size image sequences.
Abstract: The performance of a very low bit rate video codec largely depends on the efficient use of motion compensated prediction technique and on a good coding control strategy. In our previous approach [6], we proposed a multiple layer video codec using affine motion compensation. In this paper, we further extend our affine compensated multi-layer codec by incorporating a new block level and designing a coding control strategy. A measure of coherent motion is used in the decision process which makes the codec perform efficiently at very low bit rate and for small size image sequences (QCIF and sub-QCIF format). The experimental results conduced on 15 MPEG test sequences in QCIF format show improvement in PSNR of 0.2 dB and reduction in bit rate of 0.9 kbits/second.

1 citations

Proceedings ArticleDOI
04 Nov 1991
TL;DR: The authors present a real-time digital voice terminal for slow frequency-hopping (slow-FH) radio applications to ensure a reliable radio link and address aspects of the DVT architecture, the implementation of algorithms in the signal processors, and the system performance.
Abstract: The authors present a real-time digital voice terminal (DVT) for slow frequency-hopping (slow-FH) radio applications to ensure a reliable radio link. Slow-FH technologies have been introduced for ECCM communications as a near-term solution which will not disturb the usage of the present conventional radio spectrum. Digital secure algorithms with digital speech codec are known to achieve a high degree of security. The DVT is composed of a speech codec and a voice-band modem. The speech codec that is used for the DVT is a 4.8 kbit/s pitch predictive coder using adaptive transform coding (PP-ATC). The modem is a QAM voice-band modem. This modem has a specific signal format to allow waveform shaping by a hopping transmitter to keep RF compatibility with conventional radio. The authors address aspects of the DVT architecture, the implementation of algorithms in the signal processors, and the system performance. >

1 citations

Proceedings ArticleDOI
01 Oct 2015
TL;DR: A video codec evaluation system that can evaluate the performance of video codec by changing encoding parameters from different dimensions: video content, resolution, codec preset, and bitrate.
Abstract: This paper introduces a video codec evaluation system. The system can evaluate the performance of video codec by changing encoding parameters from different dimensions: video content, resolution, codec preset, and bitrate. For each encoder, the system can enumerate all kinds of parameters above to encode test video sequences and then record codec's output video quality and time-consumption. Through this system, we can make detailed analysis of the performance of video codec.

1 citations

Book ChapterDOI
09 Jan 2007
TL;DR: With the SBPC, the bit-plane coding for scalable audio can be implemented in a perceptually more efficient manner, and the perceptual quality of the audio under aforementioned scenario is much improved and stable.
Abstract: Scalable audio coding technique such as MPEG-4 Scalable Lossless coding (SLS) is a unified solution for demands in high-compression perceptual audio and high-quality lossless audio. It provides a fine-grain scalable extension of the well-known MPEG-4 Advanced Audio Coding (AAC) perceptual audio coder up to fully lossless reconstruction at word lengths and sampling rates typically used for high-resolution audio. Recently, the combination of SLS and AAC is renamed as “High Definition Advanced Audio Coding” (HD-AAC). It is observed that HD-AAC can be further improved at intermediate enhancement bitrate for many audio sequences when the core bitrate is low. Based on this observation, a Switchable Bit-Plane Coding (SBPC) is proposed in this paper. The SBPC consists of a normal BPC and a Prioritized BPC (PBPC). The corresponding optimal bit-plane coding method is switched into action according to the residual energy distribution in different frequency ranges. With the SBPC, the bit-plane coding for scalable audio can be implemented in a perceptually more efficient manner, and the perceptual quality of the audio under aforementioned scenario is much improved and stable.

1 citations

Patent
08 Nov 2006
TL;DR: In this paper, a device receives from a base station speech data encoded with a first speech codec, transcodes the speech data into a second speech codec and transmits the speech encoded with the second speech decoder toward a controller used for controlling the transmission of the transmitted speech data.
Abstract: A device receives from a base station speech data encoded with a first speech codec, transcodes the speech data into a second speech codec and transmits the speech data encoded with the second speech codec toward a base station controller used for controlling the transmission of the speech data.

1 citations


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