Abstract: A method of audio processing includes generating harmonics in a hybrid complex quadrature mirror filter domain. Generating the harmonics may include multiplication, using a feedback delay loop, and dynamic compression. The harmonics may be generated based on one or more hybrid sub-bands of the complex transform domain signal.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: Embodiments are disclosed for channel-based audio (CBA) (e.g., 22.2-ch audio) to object-based audio (OBA) conversion. The conversion includes converting CBA metadata to object audio metadata (OAMD) and reordering the CBA channels based on channel shuffle information derived in accordance with channel ordering constraints of the OAMD. The OBA with reordered channels is rendered in a playback device using the OAMD or in a source device, such as a set-top box or audio/video recorder. In an embodiment, the CBA metadata includes signaling that indicates a specific OAMD representation to be used in the conversion of the metadata. In an embodiment, pre-computed OAMD is transmitted in a native audio bitstream (e.g., AAC) for transmission (e.g., over HDMI) or for rendering in a source device. In an embodiment, pre-computed OAMD is transmitted in a transport layer bitstream (e.g., ISO BMFF, MPEG4 audio bitstream) to a playback device or source device.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: A method for multi-band ducking of audio signals is provided. In some implementations, the method involves receiving, at a decoder, an input audio signal, wherein the input audio signal is a downmixed audio signal. In some implementations, the method involves separating the input audio signal into a first set of frequency bands. In some implementations, the method involves determining a set of ducking gains, a ducking gain corresponding to a frequency band of the first set of frequency bands. In some implementations, the method involves generating a broadband decorrelated audio signal, wherein ducking gains of the set of ducking gains are applied to at least one of: 1) a second set of frequency bands prior to generating the at least one broadband decorrelated audio signal: or 2) a third set of frequency bands that separates the at least one broadband decorrelated audio signal.

Abstract: A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.

Abstract: The present document relates to methods and apparatus for rendering input audio for playback in a playback environment. The input audio includes at least one audio object and associated metadata, and the associated metadata indicates at least a location of the audio object. A method for rendering input audio including divergence metadata for playback in a playback environment comprises creating two additional audio objects associated with the audio object such that respective locations of the two additional audio objects are evenly spaced from the location of the audio object, on opposite sides of the location of the audio object when seen from an intended listener's position in the playback environment, determining respective weight factors for application to the audio object and the two additional audio objects, and rendering the audio object and the two additional audio objects to one or more speaker feeds in accordance with the determined weight factors.

Abstract: Described herein is a method (30) of rendering an audio signal (17) for playback in an audio environment (27) defined by a target loudspeaker system (23), the audio signal (17) including audio data relating to an audio object and associated position data indicative of an object position. Method (30) includes the initial step (31) of receiving the audio signal (17). At step (32) loudspeaker layout data for the target loudspeaker system (23) is received. At step (33) control data is received that is indicative of a position modification to be applied to the audio object in the audio environment (27). At step (38) in response to the position data, loudspeaker layout data and control data, rendering modification data is generated. Finally, at step (39) the audio signal (17) is rendered with the rendering modification data to output the audio signal (17) with the audio object at a modified object position that is between loudspeakers within the audio environment (27).

Abstract: Dialogue enhancement of an audio signal, comprising obtaining a set of time-varying parameters configured to estimate a dialogue component present in said audio signal, estimating the dialogue component from the audio signal, applying a compressor only to the estimated dialogue component, to generate a processed dialogue component, applying a user-determined gain to the processed dialogue component, to provide an enhanced dialogue component. The processing of the estimated dialogue may be performed on the decoder side or encoder side. The invention enables an improved dialogue enhancement.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

Abstract: Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA.

Abstract: Methods for transmitting and receiving an audio stream are provided. For transmission, the method involves obtaining a frame of an audio signal, determining a number of source blocks to divide the frame of the audio signal into and a number of parity blocks to generate for forward error correction, wherein the number of source blocks and the number of parity blocks are determined based on characteristics of a wireless communication protocol to be used. The wireless communication protocol may be BLUETOOTH. The parity blocks are usable by a decoder to reconstruct one or more corrupted or missing source blocks and may be obtained by means of Reed-Solomon encoding.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: Methods and systems for improving coding efficiency of video.

Abstract: A method for tracking a reference picture on an electronic device is described. The method includes receiving a bitstream. The method also includes decoding a portion of the bitstream to produce a decoded reference picture. The method further includes tracking the decoded reference picture in a decoded picture buffer (DPB) with reduced overhead referencing. The method additionally includes decoding a picture based on the decoded reference picture.

Abstract: A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.