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.

Method for dynamic regulating uplink voice rate of terminal

Abstract: The invention is a terminal uplink voice rate dynamic adjustment method, including: when receiving the UTRAN empty rate control information, RRC module configures TFCS subset, and according to CTFC value, the MAC module finds the transmission channel corresponding to AMR A sub-flow, and determines the corresponding RLCSIZE set, and according to RLCSIZE set it finds the uplink voice rate activation set required by AMR module; sending the uplink voice rate activation set to AMR module; AMR module selects a new uplink voice rate, and sends the uplink coding data to RLC/MAC module, through encryption, TFC selection, and other processes, sending to the physical layer, and ultimately, through the UTRAN, sending to the CN AMR codec The method of the invention makes terminal AMR rate control and MAC, TFC selection close matching; and link layer and AMR keeping synchronous

Method and system for output device protection in an audio codec

Abstract: Aspects of a method and system for output device protection in an audio CODEC are provided. A hardware audio CODEC comprising one or more equalizers may measure strength of one or more frequency bands of an audio signal. Gain settings for the equalizers may be determined utilizing, for example, one or more look-up tables based on the measured strength(s). The gain settings may be determined based on whether the audio signal comprises voice, music, and/or ringtone data. The gain setting may also be determined based on a sample rate of the audio signal; a frequency response of one or more filters within the CODEC; gain settings of one or more gain processing circuits within the CODEC; a subjective loudness curve; characteristics of one or more audio output devices communicatively coupled to the wireless device; and/or characteristics of one or more audio input devices communicatively coupled to the wireless device.

Communication node, terminal, and communication control method

Abstract: An IMS node 310 relates to a communication node that, upon handover of one of two terminals communicating in a first network to a second network different from the first network, determines a codec and a codec mode that are to be used by the two terminals. A determination unit 506 establishes, as the codec and codec mode to be used by the two terminals, the common parts among information indicating the codecs and codec modes used for the communications in the first network, information indicating the codecs and codec modes supported by one of the two terminals, and information indicating the codecs and codec modes supported by the second network. A signaling generation unit 510 generates a signaling for requesting the two terminals to change to the established codec and codec mode that are to be used by the two terminals.

Scelp: low delay audio coding with noise shaping based on

Abstract: In this contribution a new wide band audio coding concept is pre­ sented that provides good audio quality at bit rates below 3 bits per sample with an algorithmic delay of less than 10 ms. The new con­ cept is based on the principle of Linear Predictive Coding (LPC) in an analysis-by-s ynthesis framework, as known from speech coding. A spherical codebook is used for quantization at bit rates which are higher in comparison to low bit rate speech coding for improved performance for audio signals. For superior audio quality, noise shaping is employed to mask the coding noise. In order to re­ duce the computational complexity of the encoder, the analysis-by­ synthesis framework has been adapted for the spherical code book to enable a very efficient excitation vector search procedure. The codec principle can be adapted to a large variety of application sce­ narios. In terms of audio quality, the new codec outperforms ITU- T G.722 [4] at the same bit rate of 48 kbit/sec and a sample rate of 16 kHz.

A novel method of Post processing algorithms for image and VP8 video codec's

Abstract: This paper proposes post processing algorithm by employing modified adaptive (method or technique) in loop deblocking filter for VP8 and Image codec's, mainly because blocking artifact is one of the most annoying artifact especially in video and image compression coding. The method proposed improves the quality of the reconstructed image and as well as video. In this paper, we have introduced deblocking algorithm, classified them into several categories and implemented on Image and Video codec's. On the other side the PSNR is widely used for checking the quality of the compressed image and video. However, PSNR sometimes does not reveal the quality perceived by human visual system. In this paper, we will introduce another measurement to estimate the blockiness in the compressed image and video. So that VP8 video codec and image codec is successfully implemented based on this method to achieve better visual quality of video data and also achieve good performance.