The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Audio encoder and decoder

An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits.

Quality improvement techniques in an audio encoder

An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Multi-channel audio encoding and decoding

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

An apparatus for processing an audio signal and method thereof are disclosed. The preset invention includes a signal receiving unit a plurality of preset informations and preset attribute information; a display unit displaying a plurality of preset metadatas; a preset mode input unit being inputted a select signal selecting one preset metadata; a preset mode select unit selecting the preset metadata and a preset information based on the select signal; a static preset mode receiving unit receiving the preset information and the preset metadata corresponding all data regions of the downmix signal based on the preset attribute information; a dynamic preset mode receiving unit receiving the preset information and the preset metadata corresponding single data region of the downmix signal based on the preset attribute information; and a rendering unit controlling the object by applying the preset information to the all data regions or the single data region of the downmix signal.

A method and an apparatus for processing an audio signal

In a transform encoder for audio data, encoded data in the form of mantissas, exponents and coupling data is packed into fixed length frames in an output bitstream. The fields within the frame for carrying the different forms of data are variable in length, and apace within the frame must be allocated between them to fit all of the required information into the frame. The space required by the various data types depends on certain encoding parameters, which are calculated for a particular frame before the data is encoded, thus ensuring that the encoded data will fit into the frame before the computationally expensive encoding process is carried out. Information in relation to, for example, transform length, coupling parameters and exponent strategy are determined, which allows the space required for the coupling and exponent data to be calculated. The mantissa encoding parameters can then be iteratively determined so that the encoded mantissas will fit into the frame with the other encoded data. The determined encoding parameters are stored and the audio data is encoded according to those parameters after it has been determined that the encoded data will fit into the frame.

Fast frame optimization in an audio encoder

A method for coding digital audio data in which coded Fast Modified Discrete Cosine Transform (FMDCT) coefficients are computed utilising a Fast Fourier Transform (FFT) method. The described method allows a significant reduction in computations as compared to an ordinary DCT coding procedure. Also, pairs of audio channels can be combined to use a single FFT computation, where the selected transform length for the paired channels is the same. In such cases where pairing of identical transform length channels is not possible, a long transform length channel is combined with a short transform length channel and converted in two short transforms. A windowing function is also combined with a pre-processing stage to the transformation, further decreasing computational requireements.

A fast frequency transformation techique for transform audio coders

A method of parametrically encoding a transient audio signal, including the steps of: determining a set V of the N largest frequency components of the transient audio signal, where N is a predetermined number; determining an approximate envelope of the transient audio signal; and determining a predetermined number P of samples W of the approximate envelope for use in generating a spline approximation of the approximate envelope, whereby a parametric representation of the transient audio signal is given by parameters including V, N, P and W, such that a decoder receiving the parametric representation can reproduce a received approximation of the transient audio signal.

Method and system for parametric characterization of transient audio signals

AC-3 is a high quality audio compression format widely used in feature films and, more recently, on Digital Versatile Disks (DVD). For consumer applications the algorithm is usually coded into the firmware of a DSP Processor, which due to cost considerations may be capable of only fixed point arithmetic. Commercial AC-3 Encoders have been successfully implemented on 20-bit and 24-bit word-length processors. However, it is generally assumed that 16-bit processing is incapable of delivering the high fidelity audio, expected from the AC-3 technology. Double precision computation can be utilised on such processors to provide the high quality; but the computational burden of such implementation will be beyond the capacity of the processor to enable real-time operation. Through extensive simulation study of a high quality AC-3 Encoder implementation, a multi-precision technique for each processing block is presented whereby the quality of the encoder on a 16-bit processor matches the single precision 24-bit implementation very closely without excessive additional computational complexity.

Multi-precision technique for digital audio encoder

A method and apparatus for assigning an exponent coding strategy in a digital audio transform coder. Different coding strategies having different differential coding limits may be assigned to different set of transform exponents according to the frequency domain characteristics of the audio signal. A neural network processing system is utilised to perform an efficient mapping of each exponent set to an appropriate coding strategy.

Mohammed Javed Absar

Papers

Fast frame optimization in an audio encoder

A fast frequency transformation techique for transform audio coders

Method and system for parametric characterization of transient audio signals

Multi-precision technique for digital audio encoder

A neural network based method for exponent coding in a transform coder for high quality audio