#Qube compensator full
#Qube compensator code
#Qube compensator professional

#Qube compensator full

Signals with an amplitude close to full scale are prone to clipping. Consequently, the signal at the receiver side is only regenerated with correct power but without waveform preservation. spectral band replication, SBR) parameterize the signal power in a rather course manner. Secondly, parametric coding methods (e.g. If amplitude or phase errors add up constructively, the resulting attitude in the time domain may temporarily be higher than the original waveform. Quantization errors in the frequency domain result in small deviations of the signal amplitude and phase with respect to the original waveform. First of all, the audio encoder applies quantization to the transmitted signal which is available in a frequency decomposition of the input waveform in order to reduce the transmission data rate. Assuming an audio signal at the encoder's input that is below the threshold of clipping, the reasons for clipping in a modern perceptual audio encoder are manifold. Although typically an effort will be made at the encoder side to prevent the occurrence of clipping at the decoder side, clipping may nevertheless occur at the decoder side for various reasons, such as different decoder implementations, rounding errors, transmission errors, etc.

This may lead to audible artifacts (clicks or short distortion). modulated according to PCM) for the output waveform. 16 bit) of a uniformly quantized fixed-point representation (e.g.

It appeared in rare cases that some decoded PCM samples are вЂњclippingвЂќ which means that two or more consecutive samples reached the maximum level that can be represented by the underlying bit resolution (e.g. The resulting AAC bitstream is then made available for purchase e.g.

#Qube compensator professional

by a professional AAC (Advanced Audio Coding) encoder at the content creation side.

#Qube compensator code

In the current audio content production and delivery chains the digitally available master content (PCM stream (pulse code modulated stream)) is encoded e.g. Challenges arise as modern perceptual audio codecs are necessitated to deliver satisfactory audio quality at increasingly low bit rates. Further embodiments of the present invention relate to clipping prevention for spectral decoded audio signals using time domain level adjustment in combination with side information.ДЄudio signal processing becomes more and more important. Some embodiments of the present invention relate to adjusting the level of the signal to be frequency-to-time converted (or time-to-frequency converted) to the dynamic range of a corresponding converter implemented in fixed-point or integer arithmetic. The present invention relates to audio signal encoding, decoding, and processing, and, in particular, to adjusting a level of a signal to be frequency-to-time converted (or time-to-frequency converted) to the dynamic range of a corresponding frequency-to-time converter (or time-to-frequency converter). The level shift compensator acts on the time-domain representation for at least partly compensating a corresponding level shift and for obtaining a substantially compensated time-domain representation. The frequency-to-time-domain converter converts the level shifted frequency band signals into a time-domain representation. The level shifter shifts levels of the frequency band signals according to the level shift factor. The clipping estimator analyzes the encoded audio signal representation and/or side information relative to a gain of the frequency band signals in order to determine a current level shift factor.

An audio signal decoder for providing a decoded audio signal representation on the basis of an encoded audio signal representation has a decoder preprocessing stage for obtaining a plurality of frequency band signals from the encoded audio signal representation, a clipping estimator, a level shifter, a frequency-to-time-domain converter, and a level shift compensator.