An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

Adapting masking thresholds for encoding a low frequency transient signal in audio data

An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

Method and an apparatus for processing an audio signal

A device and method for estimating a tonality of a sound signal comprise: calculating a current residual spectrum of the sound signal; detecting peaks in the current residual spectrum; calculating a correlation map between the current residual spectrum and a previous residual spectrum for each detected peak; and calculating a long-term correlation map based on the calculated correlation map, the long-term correlation map being indicative of a tonality in the sound signal.

Method and device for sound activity detection and sound signal classification

An apparatus for decomposing an input signal having a number of at least three input channels comprises a downmixer (12) for downmixing the input signal to obtain a downmixed signal having a smaller number of channels. Furthermore, an analyzer (16) for analyzing the downmixed signal to derive an analysis result is provided, and the analysis result 18 is forwarded to a signal processor (20) for processing the input signal or a signal derived from the input signal to obtain the decomposed signal (26).

Apparatus and method for decomposing an input signal using a downmixer

An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoding section (202) encodes an input signal and generates first encoded information, a first layer decoding section (203) decodes the first encoded information and generates a first decoded signal, a characteristic judging section (206) analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoding section (207) changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information .

Encoder, decoder, and encoding method

A system and method for adaptive rate control in audio processing is provided. The process could include receiving uncompressed audio data from an input and generating MDCT spectrum for each frame of the uncompressed audio data using a filterbank. The process could also include estimating masking thresholds for current frame to be encoded based on the MDCT spectrum. The masking thresholds reflect a bit budget for the current frame. The process could also include performing quantization of the current frame based on the masking thresholds. After the quantization of the current frame, the bit budget for next frame is updated for estimating the masking thresholds of the next frame. The process could also include encoding the quantized audio data.

Adaptive rate control algorithm for low complexity AAC encoding

Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

George Sapna

Papers

Adaptive rate control algorithm for low complexity AAC encoding

Estimating rate controlling parameters in perceptual audio encoders