Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the sound correctly using the available speaker positions in a room that is set up to play the adaptive audio content.

System and method for adaptive audio signal generation, coding and rendering

There is provided a method for optimizing reproduction of audio signals from an apparatus for audio reproduction with the apparatus for audio reproduction having a variable number of speakers. The method includes determining performance characteristics of each of the variable number of speakers; comparing performance characteristics of each of the variable number of speakers with each other; and designating a master speaker from the variable number of speakers either with or without manual intervention.

A method for optimizing reproduction of audio signals from an apparatus for audio reproduction

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

Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

System for rendering and playback of object based audio in various listening environments

The purpose of the present invention is to enable correct stream reception to take place at the receiving end, in a manner accurately adapted to dynamic changes in delivered content. In a first transmission mode, a predetermined number of extended video streams constituting scalable-coded image data, and respectively including a basic video stream that includes lowest-layer image data, and image data of layers other than the lowest layer. In a second transmission mode, only the basic video stream that includes basic image data is transmitted. Identifying information for identifying the first mode and the second mode is inserted into the video stream. At the receiving end, the transmission mode can be easily ascertained, making it possible for correct stream reception to take place in a manner accurately adapted to changes in the stream composition, that is, dynamic changes in delivered content.

Transmission device, transmission method, reception device, and reception method

An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Apparatus and method for generating audio output signals using object based metadata

The present invention relates to a data CODEC system for computer. The data CODEC system for computer comprises a system control software, a multichannel audio/speech and multimedia data signal processor, and a multichannel audio/speech and multimedia data input-output unit. The system control software communicates multichannel audio/speech and multimedia data with the multichannel audio/speech and multimedia data signal processor according to control of various application programs. The multichannel audio/speech and multimedia data signal processor processes multichannel audio/speech and multimedia data. The multichannel audio/speech and multimedia data input-output means inputs/outputs multichannel audio/speech and multimedia data from/to an external system.

Data CODEC system for computer

For synchronizing multichannel extension data with an audio signal, wherein the audio signal includes block division information and the multichannel extension data include reference audio signal fingerprint information, the block division information in the audio signal is detected by means of a block detector. Thereupon, block division of the audio signal is performed by a fingerprint calculator according to the block division information in order to obtain a sequence of test audio signal fingerprints. In addition to that, a sequence of reference audio signal fingerprints is extracted from the reference audio signal fingerprint information of the multichannel extension data. Both sequences of fingerprints are correlated in order to obtain a correlation result, by which a compensator is controlled in order to reduce or eliminate a time offset between the multichannel extension data and the audio signal.

Apparatus and method for synchronizing multichannel extension data with an audio signal and for processing the audio signal

MPEG-2 AAC Multichannel Realtime Implementation on Floating Point DSPs

Contrary to the MPEG-1 Audio compression schemes, Advanced Audio Coding (AAC) in its MPEG-2 and MPEG-4 flavours has no inherent upper limit for the sampling frequency of the input signal. Furthermore, the bitstream format allows to cover a dynamic range far beyond the range provided by 24 bit linear PCM coding. This makes the AAC coder an ideal candidate for representing audio signals with parameters that are usually associated with high resolution audio systems. This paper discusses the application of this highly efficient compression scheme to digital programme material represented with 24 bit and 96 kHz.

/pdf/beyond-cd-quality-advanced-audio-coding-aac-for-high-3kfoov77gk.pdf

Wolfgang Fiesel

Papers

Apparatus and method for generating audio output signals using object based metadata

Data CODEC system for computer

Apparatus and method for synchronizing multichannel extension data with an audio signal and for processing the audio signal

MPEG-2 AAC Multichannel Realtime Implementation on Floating Point DSPs

Beyond CD-Quality: Advanced Audio Coding (AAC) for High Resolution Audio with 24 Bit Resolution and 96 kHz Sampling Frequency