The invention relates to compositions and methods for the manufacture and optimization of modified mRNA molecules via optimization of their terminal architecture.

Terminally modified rna

The invention relates to compositions and methods for the 5 preparation, manufacture and therapeutic use of signal sensor polynucleotides, primary transcripts and mmRNA molecules. 9385800_1 (GHMatters) P100390.AU.1 15/08/17 ug

Signal-sensor polynucleotides for the alteration of cellular phenotypes

New universal rotation-based fast computational structures for an efficient implementation of the DCT-IV/DST-IV and analysis/synthesis MDCT/MDST filter banks

For postprocessing spectral values which are based on a first transformation algorithm for Converting the audio Signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided. Hereupon, a weighted addition of spectral values of the sequence of blocks of spectral values is performed in order to obtain a sequence of blocks of postprocessed spectral values, wherein the combination is performed such that for calculating a postprocessed spectral value for a frequency band and a time duration a spectral value of the sequence of blocks for the frequency band and the time duration and a spectral value for another frequency band or another time duration are used, wherein the combination is further performed such that such weighting factors are used that the postprocessed spectral values are an approximation to the spectral values as they are obtained by converting the audio signal into a spectral representation using a second transformation algorithm which is different from the first transformation algorithm. The postprocessed spectral values are in particular used for a difference formation within a scalable encoder or for an addition within a scalable decoder, respectively.

Device and method for postprocessing spectral values and encoder and decoder for audio signals

In this paper, we consider the problem of characterizing the quality of audio that is impaired by time-varying distortions. Specifically, the quality of such audio must be characterized by a time-dependent function. We first study issues associated with the human subjective testing which must be addressed in order to create such a function, and we conclude that the best approach for achieving both high temporal resolution and high accuracy is most likely a combination of two testing methodologies. Using the collected subjective data, we then design an objective metric that attempts to replicate the subjective responses. The proposed metric uses as its core a subset of the model output variables that are part of the ITU perceptual evaluation of audio quality (PEAQ) recommendation as well as the structural similarity measure and the segmental signal to noise ratio. Depending on the dataset used in its design, we find that the estimated quality has an average correlation coefficient relative to the ground truth curves of between 0.887 and 0.933. Furthermore, we also find that it is possible to reduce the complexity of this metric considerably while only degrading its performance by between 0.3% and 1.6%. In addition, the reduced complexity metric appears to be more robust than the original one.

Assessing the Quality of Audio Containing Temporally Varying Distortions

The MPEG-4 Scalable to Lossless (SLS) audio coding is recently being developed to provide a unified solution for high - compression perceptual audio coding and high-quality lossless audio coding. SLS provides efficient Fine Granular Scalable (FGS) coding from AAC core layer to lossless, and achieves reasonable perceptual quality at its scalable coding range using a sequential bit-plane scanning method, which minimizes the audio distortion according to the spectral shape of the core layer quantization errors. In this paper, it is shown that the perceptual quality performanc of SLS at intermediate rates can be further improved by incorporating psychoacoustic model into the bit-plane coding process. In addition, it is also found that such an improvement can be achieved by slightly tweaking the original bit-plane coding process of SLS and hence preserving its nice features such as compatibility to lossless coding and low complexity.

/pdf/perceptually-enhanced-bit-plane-coding-for-scalable-audio-14uuie9jfd.pdf

Perceptually Enhanced Bit-Plane Coding for Scalable Audio

A perceptually enhanced prioritized bit-plane audio coding algorithm is presented in this paper. According to the energy distribution in different frequency regions, the bit-planes are prioritized with optimized parameters. Based on the statistical modeling of the frequency spectrum, a much more simplified implementation of prioritized bit-plane coding is integrated with the recent release of MPEG-4 scalable lossless (SLS) audio coding structure by replacing the sequential bit-plane coding in the enhancement layer. With zero extra side information, trivial added complexity, and modification to the original SLS structure, extensive experimental results show that the perceptual quality of SLS with noncore and very low core bit-rate is improved significantly in a wide range of bit-rate combinations. Fully scalable audio coding up to lossless with much enhanced perceptual quality is thus achieved.

Frequency Region-Based Prioritized Bit-Plane Coding for Scalable Audio

In MPEG-4 scalable lossless coding (SLS) which was recently published as an ISO standard in June 2006, the integer modified discrete cosine transform (IntMDCT) was adopted to enable efficient lossless reconstruction. In addition, there is an MDCT filterbank which is inherent to the advanced audio coding (AAC) core that is present in the SLS codec. The presence of two filterbanks have undoubtedly increased the complexity of the implementation, and it is for this reason that the MDCT is disabled and the IntMDCT is then the only type of filterbank that is employed in SLS for both lossy and lossless operations. Because of the rounding operations in the IntMDCT, there is a concern if the use of IntMDCT for perceptual audio coding will eventually degrade the fidelity of the audio codec. This paper addresses this concern by analyzing the performance of the IntMDCT in a lossy coding scenario. It is found that noise introduced by the IntMDCT does not affect the perceptual quality of the coded audio under standard playback circumstances. As such, it concludes that the MDCT and IntMDCT filterbanks are interchangeable at lossy bitrate, and the way of using only the IntMDCT filterbank in scalable audio coding is also justified.

On Integer MDCT for Perceptual Audio Coding

Embodiments of the invention provides a method and device for assigning bitrates to a plurality of channels in a scalable audio encoding/truncation process. Different bitrates are assigned to different channels in the scalable audio encoding/truncation process.

Method and device of bitrate distribution/truncation for scalable audio coding

The paper addresses a bitstream scalable coder based on the MPEG-4 scalable lossless (SLS) coding system where, in contrast to SLS, the bitrate of the enhancement layer is not fixed but instead an attempt is made to create a quality-fixed enhancement layer. With a PCM audio input, the proposed structure is able to produce an audio version with near-transparent quality on top of the existing low-quality version. In particular, the proposed fixed quality enhancing process with checking procedures is able to provide the minimum amount of enhancement for the low-quality version to obtain a near-transparent quality that is almost indistinguishable from the CD quality. In addition, a bitrate estimation model is proposed. The model enables the direct estimation of the enhancing bitrate from two parameters extracted from the encoding process of the low-quality version. Evaluation results indicate that a better defined quality level is guaranteed compared to a fixed bitrate setting and that in the mean a lower (approximately 20%) bitrate is attained. It is also shown that the estimation model proposed is able to accurately predict the necessary enhancing bitrate and at the same time, reduce the complexity by around 17%.

Fixed Quality Layered Audio Based on Scalable Lossless Coding

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