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.

Improved performance speech codec for mobile communications.

Abstract: This paper presents the application of a Voice Gender Normalization algorithm to the GSM Speech Codec and describes the refinements that can be made to the Codec as a result. By reducing the dynamic range of the speech signals entering the Codec gender specific adaptations can be made to the Codec to improve its performance in terms of subjective sound quality or its transmitted bit rate.

New directions in low bit rate speech coding

Abstract: It is pointed out that impressive progress has been made during recent years in coding speech with high quality at low bit rates and at low cost. The new digital cellular system in North America is using coded speech at 8 kb/s, but much lower bit rates are needed to support the increasing demand for cellular telephones. It is noted that incremental changes in the present technology are unlikely to produce high-quality speech at very low bit rates. Current research at Bell Laboratories is aimed at creating new representations of speech, both in frequency and time, to allow for efficient coding of speech information. The author reviews the key ideas that support the present speech coding technology and discusses some of the promising new directions. >

A space-time coded OFDM based MPEG-4 audio transceiver

Abstract: An MPEG-4 based space-time (ST) coded orthogonal frequency division multiplexing (OFDM) audio transceiver is proposed. The high-quality MPEG-4 audio codec is operated at bit rates between 16 and 64 kbit/s per channel and combined with turbo channel codes as well as space-time codes. As expected, the space-time coding scheme, using two transmitters and one receiver, outperformed the conventional one transmitter, one receiver arrangement by about 4 dB in channel SNR terms, when maintaining an error-free audio quality.

Method for selecting a codec as a function of the network capacity

Abstract: The invention relates to a method for selecting a codec for transmission of audio and/or video data between terminals ( 1, 3 ) through a packet switching connection ( 2 ), the terminals ( 1, 3 ) selecting the codec from a codec list, characterized in that the codec list for a new connection is determined by means of a table in which for at least one destination address range, the associated transmission capacity is indicated and the table is automatically updated through measurements, as well as a terminal for carrying out the method.