Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

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.

Abstract: A method of generating a substitution frame for a lost audio frame of an audio signal is presented. The method may comprise determining an audio filter based on samples of a valid audio frame preceding the lost audio frame. The method may comprise generating the substitution frame based on the audio filter and the samples of the valid audio frame preceding the lost audio frame. The method may be advantageously applied to a low frequency effects (LFE) channel of a multi-channel audio signal.

Abstract: Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Abstract: A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Abstract: A method comprising receiving a first input bit stream for a first parametrically coded input audio signal, the first input bit stream including data representing a first input core audio signal and a first set including at least one spatial parameter relating to the first parametrically coded input audio signal. A first covariance matrix of the first parametrically coded audio signal is determined based on the spatial parameter(s) of the first set. A modified set including at least one spatial parameter is determined based on the determined first covariance matrix, wherein the modified set is different from the first set. An output core audio signal is determined, which is based on, or constituted by, the first input core audio signal. An output bit stream for a parametrically coded output audio signal is generated, the output bit stream including data representing the output core audio signal and the modified set.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: A system utilizing a high throughput coding mode for CABAC in HEVC is described. The system may include an electronic device configured to obtain a block of data to be encoded using an arithmetic based encoder; to generate a sequence of syntax elements using the obtained block; to compare an Absolute-3 value of the sequence or a parameter associated with the Absolute-3 value to a preset value; and to convert the Absolute-3 value to a codeword using a first code or a second code that is different than the first code, according to a result of the comparison.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: Methods for generating an AV bitstream (e.g., an MPEG-2 transport stream or bitstream segment having adaptive streaming format) such that the AV bitstream includes at least one video I-frame synchronized with at least one audio I-frame, e.g., including by re-authoring at least one video or audio frame (as a re-authored I-frame or a re-authored P-frame). Typically, a segment of content of the AV bitstream which includes the re-authored frame starts with an I-frame and includes at least one subsequent P-frame. Other aspects are methods for adapting such an AV bitstream, audio/video processing units configured to perform any embodiment of the inventive method, and audio/video processing units which include a buffer memory which stores at least one segment of an AV bitstream generated in accordance with any embodiment of the inventive method.

Abstract: An encoding system encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: There is provided encoding and decoding methods for representing spatial audio that is a combination of directional sound and diffuse sound. An exemplary encoding method includes inter alia creating a single- or multi-channel downmix audio signal by downmixing input audio signals from a plurality of microphones in an audio capture unit capturing the spatial audio; determining first metadata parameters associated with the downmix audio signal, wherein the first metadata parameters are indicative of one or more of: a relative time delay value, a gain value, and a phase value associated with each input audio signal; and combining the created downmix audio signal and the first metadata parameters into a representation of the spatial audio.

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: Described are methods of processing audio data for hum noise detection and/or removal. The audio data comprises a plurality of frames. One method incudes: classifying frames of the audio data as either content frames or noise frames, using one or more content activity detectors; determining a noise spectrum from one or more frames of the audio data that are classified as noise frames; determining one or more hum noise frequencies based on the determined noise spectrum; generating an estimated hum noise signal based on the one or more hum noise frequencies; and removing hum noise from at least one frame of the audio data based on the estimated hum noise signal. Also described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

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.