Abstract: An audio signal (X) is represented by a bitstream (B) segmented into frames. An audio processing system (500) comprises a buffer (510) and a decoding section (520). The buffer joins sets of audio data (D1; D2, . . . , DN) carried by N respective frames (F1, F2, . . . , FN) into one decodable set of audio data (D) corresponding to a first frame rate and to a first number of samples of the audio signal per frame. The frames have a second frame rate corresponding to a second number of samples of the audio signal per frame. The first number of samples is N times the second number of samples. The decoding section decodes the decodable set of audio data into a segment of the audio signal by at least employing signal synthesis, based on the decodable set of audio data, with a stride corresponding to the first number of samples of the audio signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: A method (600) for generating a bitstream (500) indicative of an object based audio program is described. The bitstream (500) comprises a sequence of containers (501). A first container (501) of the sequence of containers (501) comprises a plurality of substream entities (520) for a plurality of substreams (411, 412, 413, 414) of the object based audio program and a presentation section (510). The method (600) comprises determining (601) a set of object channels (424). The method (600) further comprises providing (602) a set of object related metadata (434, 444) for the set of object channels (424). In addition, the method (600) comprises inserting (603) a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities (520) of the first container (501). Furthermore, the method (600) comprises inserting (604) presentation data into the presentation section (510).

Abstract: The present document relates to the determination of a 200 bitrate related to an encoded bitstream, and describes a method for determining an estimate of a bitrate of a bitstream comprising a sequence of frames comprising a varying number of bitsand corresponding to excerpts of an audio and/or video signal. At least two frames of the sequence of frames comprise a parameter indicative of a processing delay for the corresponding frame. The method comprises determining: a total number of bits for a subsequence of frames from the bitstream; a corrected number 201 of frames based on a number of frames comprised within the subsequence and the parameters of at least two frames of the subsequence; and a lower bitrate bound and an upper bitrate bound of the bitrate based on the total number of bits, the corrected number of frames and a frame rate of the bitstream.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: The present document relates to the design of anti-aliasing and/or anti-imaging filters for resamplers using rational resampling factors. In particular, the present document relates to a method for implementing such anti-aliasing and/or anti-imaging filters with reduced computational complexity. In addition, the present document relates to further aspects of an audio encoding and decoding system, such as the phase relation between the channels of a multi-channel audio signal and/or the structure of the bitstream of an encoded audio signal.

Abstract: An audio processing system (100) accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (110), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage (120) and a sample rate converter (130), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: An audio signal (X) is represented by a bitstream (B) segmented into frames. An audio processing system (500) comprises a buffer (510) and a decoding section (520). The buffer joins sets of audio data (D1; D2, . . . , DN) carried by N respective frames (F1, F2, . . . , FN) into one decodable set of audio data (D) corresponding to a first frame rate and to a first number of samples of the audio signal per frame. The frames have a second frame rate corresponding to a second number of samples of the audio signal per frame. The first number of samples is N times the second number of samples. The decoding section decodes the decodable set of audio data into a segment of the audio signal by at least employing signal synthesis, based on the decodable set of audio data, with a stride corresponding to the first number of samples of the audio signal.

Abstract: A method (600) for generating a bitstream (500) indicative of an object based audio program is described. The bitstream (500) comprises a sequence of containers (501). A first container (501) of the sequence of containers (501) comprises a plurality of substream entities (520) for a plurality of substreams (411, 412, 413, 414) of the object based audio program and a presenta tion section (510). The method (600) comprises determining (601) a set of object channels (424). The method (600) further comprises providing (602) a set of object related metadata (434, 444) for the set of object channels (424). In addition, the method (600) comprises inserting (603) a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities (520) of the first container (501). Furthermore, the method (600) comprises inserting (604) presentation data into the presentation section (510).

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Disclosed is a non-transitory computer readable storage medium which receives, by an audio decoder (operating in a specific playback environment different from a reference channel configuration), an audio signal for the reference channel configuration. The audio signal includes audio sample data and encoder-generated loudness metadata which includes a plurality of portions of loudness metadata for a plurality of playback environments. The plurality of portions of loudness metadata includes one or more respective portions of loudness metadata for each playback environment in the plurality of playback environments. The medium also selects one or more portions of specific loudness metadata (based on the specific playback environment), from among the plurality of portions of loudness metadata for the plurality of playback environments.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: The present document relates to the determination of a 200 bitrate related to an encoded bitstream, and describes a method for determining an estimate of a bitrate of a bitstream comprising a sequence of frames comprising a varying number of bitsand corresponding to excerpts of an audio and/or video signal. At least two frames of the sequence of frames comprise a parameter indicative of a processing delay for the corresponding frame. The method comprises determining: a total number of bits for a subsequence of frames from the bitstream; a corrected number 201 of frames based on a number of frames comprised within the subsequence and the parameters of at least two frames of the subsequence; and a lower bitrate bound and an upper bitrate bound of the bitrate based on the total number of bits, the corrected number of frames and a frame rate of the bitstream.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Disclosed is a non-transitory computer readable storage medium which receives, by an audio decoder (operating in a specific playback environment different from a reference channel configuration), an audio signal for the reference channel configuration. The audio signal includes audio sample data and encoder-generated loudness metadata which includes a plurality of portions of loudness metadata for a plurality of playback environments. The plurality of portions of loudness metadata includes one or more respective portions of loudness metadata for each playback environment in the plurality of playback environments. The medium also selects one or more portions of specific loudness metadata (based on the specific playback environment), from among the plurality of portions of loudness metadata for the plurality of playback environments.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.