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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TL;DR: This work proposes a codec that simultaneously addresses both high quality and low delay, with a delay of only 8.7 ms at 44.1 kHz, and uses gain-shape algebraic vector quantization in the frequency domain with time-domain pitch prediction.
Abstract: With increasing quality requirements for multimedia communications, audio codecs must maintain both high quality and low delay. Typically, audio codecs offer either low delay or high quality, but rarely both. We propose a codec that simultaneously addresses both these requirements, with a delay of only 8.7 ms at 44.1 kHz. It uses gain-shape algebraic vector quantization in the frequency domain with time-domain pitch prediction. We demonstrate that the proposed codec operating at 48 kb/s and 64 kb/s out-performs both G.722.1C and MP3 and has quality comparable to AAC-LD, despite having less than one fourth of the algorithmic delay of these codecs.
78 citations
17 Sep 2000
TL;DR: An algorithm to generate wideband speech from narrow band speech using as low as 500 bit/s of side information is presented, which has enhanced quality compared to narrowband speech.
Abstract: Wireless telephone speech is usually limited to the 300-3400 Hz band, which reduces its quality. There is thus a growing demand for wideband speech systems that transmit from 50 Hz to 8000 Hz. This paper presents an algorithm to generate wideband speech from narrowband speech using as low as 500 bit/s of side information. The 50-300 Hz band is predicted from the narrowband signal. A source-excitation model is used for the 3400-8000 Hz band, where the excitation is extrapolated at the receiver, and the spectral envelope is transmitted. Though some artifacts are present, the resulting wideband speech has enhanced quality compared to narrowband speech.
77 citations
20 Jun 1999
TL;DR: A hybrid ACELP/TCX algorithm for coding speech and music signals at 16, 24, and 32 kbit/s is presented, which switches between algebraic code excited linear prediction (ACELP) and transform coded excitation (TCX) modes on a 20-ms frame basis.
Abstract: A hybrid ACELP/TCX algorithm for coding speech and music signals at 16, 24, and 32 kbit/s is presented. The algorithm switches between algebraic code excited linear prediction (ACELP) and transform coded excitation (TCX) modes on a 20-ms frame basis. Applying TCX on 20 ms frames improved the quality for music signals. Special care was taken to alleviate the switching artifacts between the two modes resulting in a transparent switching process. Subjective test results showed that for speech signals, the performance at 16, 24, and 32 kbit/s, is equivalent to G.722 at 48, 56, and 64 kbit/s, respectively. For music signals, the quality at 24 kbit/s was found equivalent to G.722 at 56 kbit/s. However, at 16 kbit/s, the quality for music was slightly lower than G.722 at 48 kbit/s.
76 citations
Journal Article•
75 citations
TL;DR: The key factors deterring the use of visual telephony are identified, and an overview of a typical system architecture is given.
Abstract: The key factors deterring the use of visual telephony are identified, and an overview of a typical system architecture is given. The video signal formats and video and audio coding algorithms used are described. Video codec implementation is considered, and an implementation based on application-specific integrated circuits is presented. In particular, three key signal processing modules in the video codec are examined: a discrete cosine transform chip, a motion estimation chip, and a variable-length codec chip. Standardization activities in the video coding area are discussed. >
72 citations