Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. A data structure type indicator can be inserted in the bitstream to enable a decoder to determine the data structure type and to invoke an appropriate decoding process. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type as indicated by the data structure type indicator. For variable data structure types, the slot position information can be encoded with a variable number of bits based on the position of the slot in the ordered set of slots.

Apparatus for encoding and decoding audio signal and method thereof

An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a group reference value corresponding to a plurality of data included in one group through data grouping and internal grouping for the data grouping and a difference value corresponding to the group reference value and obtaining the data using the group reference value and the difference value.

Method and apparatus for signal processing and encoding and decoding method, and apparatus therefor

A method and apparatus for encoding and decoding an audio signal are provided. The present invention includes receiving an audio signal including a downmix signal and a spatial information signal, if a header is included in the spatial information signal, extracting configuration information from the header, extracting spatial information included in the spatial information signal, and converting the downmix signal to a multi-channel signal using the configuration information and the spatial information. Accordingly, the header can be selectively included in the spatial information signal, thereby if the header is plurally included in the spatial information signal, it is able to decode spatial information in case of reproducing the audio signal from a random point.

Method and apparatus for encoding and decoding an audio signal

The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Removing time delays in signal paths

Provided are an audio signal processing apparatus and method. The audio signal processing apparatus includes: a sensing unit for sensing an impedance value of a headphone and adjusting a gain to be applied to an audio signal based on the sensed impedance value; and an amplifier for generating an output audio signal by receiving at least one bias voltage corresponding to the adjusted gain and amplifying the input audio signal according to the adjusted gain. The audio signal processing apparatus is capable of appropriately adjusting the magnitude of an acoustic signal output from a headphone, according to the product specifications of the headphone.

Method and apparatus for processing audio signal

This paper presents a scalable audio format, called "multi-layer scalable LPC audio format", that addresses similar functionalities of MPEG-4. The format offers different levels of data rate and audio quality, and answers to the most important requirements of transmission and storage purposes, such as channel error robustness, cell loss robustness, low delay, and playback control. It operates in four modes. The first mode is based on a modified version of the LD-CELP algorithm, in which each 6 samples are represented by one single byte. In order to improve the signal-to-noise ratio (SNR), additional enhancement layers are embedded in the bit stream to allow higher quality at higher bit rates. The resultant bit rates are integer-multiple of 10.67 kbps. The other three modes use QMF splitting to two, four and eight subbands. These modes allow efficient representation of wideband audio and speech signals, and offer extension layers of 5.33 and 2.66 kbps. A simple and efficient header structure is embedded in the bitstream to allow the decoding process even in channel error conditions and even when the bitstream has been down-scaled somewhere during the transmission but has not been acknowledged to the decoder. Comparison results are conducted with respect to MPEG and ITU standards.

A. Jbira

Papers

Multi-layer scalable LPC audio format