Topic
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.
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01 Dec 2011
TL;DR: It is showed that the quality of intra frame improves by using the proposed codec in comparison with two other methods, including scalable MPEG-2 codec and scalable DWT2 method for HDTV video signals.
Abstract: In this paper, a scalable hybrid video codec has been developed for HDTV video signals. The intra frame pictures are coded based on wavelet transform and inter frame pictures are coded with a standard MPEG-2 codec. By scalable coding of HDTV, channels transmit with various speeds and capacities; In addition, receivers can work at different resolutions. Base layer and enhancement layer of the frame picture are coded with same codec. This paper compares scalable hybrid coding with scalable MPEG-2 codec and scalable DWT2 method for HDTV video signals. It is showed that the quality of intra frame improves by using the proposed codec in comparison with two other methods. By this improvement, the quality of sequence increases.
12 May 2008
TL;DR: A new structure for a scalable codec that works with 10 ms input frame for wideband speech and audio signals at bit rates ranging from 8 to 32 kbit/s and is assessed to be equivalent to the recently standardized embedded codec ITU-T G.729.
Abstract: This paper proposes a new structure for a scalable codec. Our proposed codec works with 10 ms input frame for wideband speech and audio signals at bit rates ranging from 8 to 32 kbit/s. The core layer is the ITU-T G.729 at 8 kbit/s producing a narrowband output. The first enhancement layer is a band-width extension providing a wideband output with 2 kbit/s. The second enhancement layer is based on algebraic quantization of wavelet packet coefficients and improves gradually the synthesized signal as the bitrate increases. For speech signals, at bitrates of 24 and 32 kbit/s, the codec is shown to be equivalent to the ITU-T G.722 codec at 56 and 64 kbit/s, respectively. Moreover, the codec at 32 kbit/s is assessed to be equivalent to the recently standardized embedded codec ITU-T G.729.1 at the same bitrate with a lower algorithmic delay.
06 Dec 2010
TL;DR: Experiments prove that the running speed of optimized code have been significantly increased in ARM core smart phones.
Abstract: This paper introduces AMR (Adaptive Multi Rate) codec algorithm and the basic characteristics of ARM architecture, and focus on the optimization methods of AMR algorithm based on ARM platform, making use of ARMv5E core hardware features. Firstly the CPU usage rate of AMR key functions are analyzed, then two methods including inline optimization and assembly optimization are discussed, finally optimization result is verified through experiment in smart phones. Experiments prove that the running speed of optimized code have been significantly increased in ARM core smart phones.
TL;DR: An ultra low delay audio coder by very short block processing and embedded coding implemented in fixed-point DSP and the embedded coding offers the error resilience feature so that joint source-channel coding scheme for unequal error protection can be easily designed by varying both source coding bit rate and channel coding redundancy.
Abstract: Digital audio coding delays have become increasingly critical in real-time wireless applications. In live productions, a codec with ultra low delay is required within the constraints of the available channel bandwidth. However, such a threshold can hardly be reached by means of standard audio coding schemes. To achieve low delay as well as to satisfy cost and power consumption constraints, this paper presents an ultra low delay audio coder by very short block processing and embedded coding implemented in fixed-point DSP. The short block two dimensional (2D) spatial-frequency processing of audio input signal fully exploits the correlation for better compression performance. Lifting wavelet transform with boundary effects minimized by changing wavelet shape is developed using bit shifts and additions to replace multiplications in a fixed-point specification under accuracy constraint. The embedded coding offers the error resilience feature so that joint source-channel coding scheme for unequal error protection can be easily designed by varying both source coding bit rate and channel coding redundancy. Experimental results demonstrate that the proposed coder is efficient and requires less memory in fixed-point computation which guarantees no overflow.
23 Oct 2009
TL;DR: This paper enhances the G.722.2 codec by removing further redundancies in the multiple frames encapsulated in piggybacking and by having only one set of LP coefficients for all the subframes encapsulated.
Abstract: In this paper, we present the design of a new piggybacking algorithm for VoIP implemented using the G.722.2 codec. In piggybacking, multiple speech frames that include those transmitted in the past are encapsulated in a single packet. Because redundant copies of each frame are transmitted to the receiver, the receiver can recover those lost frames when one or more packets are lost or arrive late in their transmission. In this paper, we have enhanced the G.722.2 codec by removing further redundancies in the multiple frames encapsulated in piggybacking and by having only one set of LP coefficients for all the subframes encapsulated. We create multiple versions of the codec, each using a different frame size. Our new codec can encode the multiple frames with little degradation in PESQ, while having substantial bit savings. Its performance is evaluated against the original method of piggybacking over random losses, as well as that using packet traces collected in the PlanetLab.