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.

Rate control of a region-based H.263 video codec under time-varying channels

Abstract: A region-based video codec based on the H.263+ standard is examined and its associated novel rate control schemes are proposed for visual communication through time-varying low-bit-rate channels. The region-based coding scheme is a hybrid method that consists of the traditional block DCT coding and the object-based coding. Basically, we adopt H.263+ as the platform, and develop a fast macroblock-based segmentation method to implement the region-based video codec. The proposed rate control solution includes rate control in three levels: encoding frame selection, frame-layer rate control and macroblock-layer rate control. The goal is to enhance the visual quality of decoded frames without obvious motion unsmoothness under time-varying channels. The efficiency of the proposed rate control schemes applied to the region-based video codec is demonstrated via several typical test sequences.

Performance of ITU-T G.729 8 kb/s CS-ACELP speech codec with nonvoice narrowband signals

Abstract: A report is given on the results of a series of objective measurements conducted by COMSAT in a laboratory environment aimed at characterizing the narrowband performance of the ITU-T G.729 8 kb/s conjugate-structure algebraic code-excited linear prediction (CS-ACELP) speech coder. The test procedures followed ITU-T Recommendation G.720, "Characterization of Low-Rate Voice Coder Performance with Non-Voice Signals". It was concluded that the G.729 algorithm has excellent performance with narrowband signals in general (e.g., single tones and DTMF signals). As for Signaling System No. 5 (SS5) interregister signals, the G.729 CS-ACELP frequently failed to correctly identify SS5 digit 6 in a number of occurrences, using worst-case analysis equipment. This indicates that the SS5 performance of G.729 codecs in trunks where SS5 is used should be carefully measured before the network planner decides on its deployment. Great care should also be taken for tandem connections, since no test has been performed for these configurations.

Wideband Speech and Audio Coding in the Perceptual Domain

Abstract: A new critical band auditory filterbank with superior auditory masking properties is proposed and is applied to wideband speech and audio coding. The analysis and synthesis are performed in the perceptual domain using this filterbank. The outputs of the analysis filters are processed to obtain a series of pulse trains that represent neural firing. Simultaneous and temporal masking models are applied to reduce the number of pulses in order to achieve a compact time-frequency parameterization. The pulse amplitudes and positions are then coded using a run-length coding algorithm. The new speech and audio coder produces high quality coded speech and audio, with both temporal and spectral fidelity.

Method for coding audio

Abstract: The invention relates to a method for encoding audio signals Belonging to a sound field sound sources are first compressed individually with a conventional compression method If the maximum allowable data rate is exceeded, a summary is made of sound sources and subsequent new compression or new compression with stronger compression factor Each of these sound sources is assigned to a position information and information about the source Information on the properties of the space to be emulated as well as information about the current horizontal and vertical viewing angles are inserted into the generated data stream In the playback device the size and position of the image projection is evaluated and based on that, and performed on the other parameters, an image of the sound source to the speakers you have

Process for the cascade coding and decoding of audio data

Abstract: In a process for the cascade coding and decoding of audio data, the spectral components of the relevant short-time spectrum are formed for a data block with a given number of time input data, the coded signal is formed on the basis of the spectral components of said data block using a psycho-acoustic model of the bit distribution for the spectral components by quantifying and coding, whereupon time output data are obtained by decoding at the end of each codec stage. To prevent a deterioration in the sound quality in codec cascades with a plurality of stages, an identification signal is added to the coded signal at an initial stage to mark the start of the data block, whereby the subsequent codec stages undertake the classification of the data blocks to be coded on the basis of said identification signal.