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.

Multiple descriptions and path diversity using the AMR-WB speech codec for voice communication over MANETs

Abstract: We compare different source diversity methods for converstional voice communication over multiple routes in a mobile ad-hoc network (MANET). A new multiple description (MD) codec based on the AMR-WB codec, with two balanced side descriptions (6.9 kbps each) is presented. We compare the performance of the MD codec against two other diversity methods, 1) duplicating speech encoded with AMR-WB at 6.6 kbps and 2) duplicating speech encoded with AMR-WB at 12.65 kbps. We show that because of the large packet headers added to each packet by typical MANET protocols, the overhead of sending the simple path diversity methods is not much larger than the overhead for sending MD streams over different paths, and the gain in speech quality we get from duplicating AMR-WB at 12.65 kbps over sending MD codec streams is significant. We compare the speech quality delivered by each of the methods under random and bursty packet loss conditions. The quality of decoded speech is evaluated using WPESQ, a wideband extension to the PESQ algorithm.

Automatic response service method using voice and text

Abstract: PURPOSE: An ARS(Automatic Response Service) method using a voice and a character is provided to remove a predetermined frame from frames generated by an AMR(Advanced Multi Rate) CODEC, and to insert a desired character frame instead of the predetermined frame, thereby simultaneously servicing a voice and a character. CONSTITUTION: A system removes a predetermined voice frame among frames generated by an AMR(Advanced Multi Rate) CODEC. The system inserts a desired character frame instead of the removed voice frame, and transmits the character frame. A system receives an AMR packet, and checks each frame type. If the checked frame type is a character, the system obtains the character through frame processing, and outputs the character by using a graphic library adjusted to a corresponding display. If the checked frame type is a voice, the system decodes the voice through the AMR CODEC, and outputs the voice through an MIDI.

An Audio Codec for Multiple Generations Compression Without Loss of Perceptual Quality

Abstract: We describe a generic audio codec allowing for multiple, i.e., cascaded, lossy compression without loss of perceptual quality as compared to the first generation of compressed audio. For this sake we transfer encoding information to all subsequent codecs in a cascade. The supplemental information is embedded in the decoded audio signal without causing degradations. The new method is applicable to a wide range of current audio codecs as documented by our MPEG-1 implementation.

Low-complexity and robust coding mode decision in the EVS coder

Abstract: Several state-of-the-art switched audio codecs employ the closed-loop mode decision to select the best coding mode at every frame. The closed-loop mode selection is known to have good performance but also high complexity. The new approach we propose in this paper is a low-complexity version of the closed-loop approach, based on similar decisions which compute the coding distortion of each mode and select the one with the lowest distortion. Our approach differs mainly in the way the coding distortions are calculated. We are able to notably reduce the complexity by only estimating the distortions without encoding and decoding the input for each mode. The new approach was implemented in the EVS codec standard and evaluated both objectively and subjectively. Compared to the closed-loop approach, it yields similar performance and lower complexity.