Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: The present document describes a method (800) for allocating bits to a frame (301) of a sequence of frames (301) to yield a bitstream having a constant average bitrate, wherein the frame (301) comprises audio data and metadata. The method (800) comprises maintaining (801) an overall bit reservoir (100) and maintaining (802) a virtual bit reservoir (510) being a subset of the overall bit reservoir (100), such that bits for the metadata of the frame (301) are allocated from the virtual bit reservoir (510) and such that bits for the audio data of the frame (301) are allocated from the overall bit reservoir (100).

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value for a first frequency subband of an audio signal is described. The first banded tonality value is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value for a first frequency subband of an audio signal is described. The first banded tonality value is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal.

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: The present document relates to methods and systems for music information retrieval (MIR). In particular, the present document relates to methods and systems for extracting a chroma vector from an audio signal. A method (900) for determining a chroma vector (100) for a block of samples of an audio signal (301) is described. The method (900) comprises receiving (901) a corresponding block of frequency coefficients derived from the block of samples of the audio signal (301) from a core encoder (412) of a spectral band replication based audio encoder (410) adapted to generate an encoded bitstream (305) of the audio signal (301) from the block of frequency coefficients; and determining (904) the chroma vector (100) for the block of samples of the audio signal (301) based on the received block of frequency coefficients.

Abstract: The present document relates to methods and apparatus for audio coding. In particular, the present document relates to methods and apparatus for enhanced block switching and/or bit allocation in audio coding of transient-tonal signals. A method of encoding samples of an audio signal comprises determining a first measure indicative of transient characteristics of the audio signal, determining a second measure indicative of tonal characteristics of the audio signal, selecting a transform length for the audio signal on the basis of the first measure and the second measure, and applying a time-frequency transform to a block of samples of the audio signal in accordance with the selected transform length, to thereby obtain a block of frequency coefficients corresponding to the block of samples of the audio signal.

Abstract: The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value (311, 312) for a first frequency subband (205) of an audio signal is described. The first banded tonality value (311, 312) is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal.

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: The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit-stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity.

Abstract: The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for reducing the complexity of a bit allocation process used in the context of audio encoding/decoding. An audio encoder (300) configured to encode an audio signal according to a first audio codec system is described. The audio encoder (300) comprises a transform unit (302) configured to determine a set of spectral coefficients (312) based on the audio signal. Furthermore, the encoder (300) comprises a floating-point encoding unit (304) configured to determine a set of scale factors and a set of scaled values (314), based on the set of spectral coefficients (312); and to encode the set of scale factors to yield a set of encoded scale factors (313).

Abstract: In a class of embodiments, an audio encoding system (typically, a perceptual encoding system that is configured to generate a single (“unified”) bitstream that is compatible with (i.e., decodable by) a first decoder configured to decode audio data encoded in accordance with a first encoding protocol (e.g., the multichannel Dolby Digital Plus, or DD+, protocol) and a second decoder configured to decode audio data encoded in accordance with a second encoding protocol (e.g., the stereo AAC, HE AAC v1, or HE AAC v2 protocol). The unified bitstream can include both encoded data (e.g., bursts of data) decodable by the first decoder (and ignored by the second decoder) and encoded data (e.g., other bursts of data) decodable by the second decoder (and ignored by the first decoder).

Abstract: A method for determining mantissa bit allocation of audio data values of frequency domain audio data to be encoded. The allocation method includes a step of determining masking values for the audio data values, including by performing adaptive low frequency compensation on the audio data of each frequency band of a set of low frequency bands of the audio data. The adaptive low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set of low frequency bands has prominent tonal content; and performing low frequency compensation on the audio data in each frequency band in the set of low frequency bands having prominent tonal content as indicated by the compensation control data, but not performing low frequency compensation on the audio data in any other frequency band in the set of low frequency bands.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: An audio encoder configured to encode an audio signal to generate a bitstream having E-AC-3 format, including by determining a first control parameter indicative of an allocation of available mantissa bits for quantized audio content of the signal. The encoder is configured to perform transcoding simulation to determine a second control parameter in a manner based at least in part on statistical analysis of results of E-AC-3 bit allocation processing of audio data assuming a first target data rate, and of AC-3 bit allocation processing of the data assuming a second target data rate, and to include the second control parameter in the bitstream for use by a converter to convert the bitstream into a second bitstream having AC-3 format at the second target data rate. Other aspects are converters configured to perform transcoding on a bitstream using such a second control parameter, and methods performed by any embodiment of the inventive encoder or converter.

Abstract: The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.