Abstract: Disclosed are methods and systems for decoding immersive audio content encoded by an adaptive number of scene elements for channels, audio objects, higher-order ambisonics (HOA), and/or other sound field representations. The decoded audio is rendered to the speaker configuration of a playback device. For bit streams that represent audio scenes with a different mixture of channels, objects, and/or HOA in consecutive frames, fade-in of the new frame and fade-out of the old frame may be performed. Crossfading between consecutive frames happen in the speaker layout after rendering, in the spatially decoded content type before rendering, or between the transport channels as the output of the baseline decoder but before spatial decoding and rendering. Crossfading may use an immediate fade-in and fade-out frame (IFFF) for the transition frame or may use an overlap-add synthesis technique such as time-domain aliasing cancellation (TDAC) of MDCT.

Abstract: A decoder-side method for outputting several audio channels of a sound program is described. An audio channel of the sound program, a residual signal, a gain parameter, and a delay parameter are received, for example within a bitstream. The audio channel is adjusted in accordance with the gain parameter and the delay parameter, to produce an adjusted audio signal, and is then combined with the residual signal to produce a combined signal. The audio channel is output as a first audio channel of the sound program for playback, while the combined signal is output as a second audio channel of the sound program. Other aspects are also described and claimed.

Abstract: A bitstream is obtained by a decoder side, that contains an encoded version of an audio signal and an instantaneous loudness sequence of the audio signal. The instantaneous loudness sequence has not been loudness normalized. A dynamic range control, DRC, gain sequence is produced by applying the instantaneous loudness sequence to a DRC characteristic, with loudness normalization. The DRC gain sequence is applied to the decoded audio signal. Other aspects are also described and claimed.

Abstract: A method that includes displaying a user interface (UI) of a three-dimensional (3D) acoustic environment that includes a sound source; receiving a user selection, via an input device, of a location within the UI, the location towards which the sound source is to be oriented within the 3D acoustic environment; determining a set of parameters that define an orientation of the sound source towards the location within the 3D acoustic environment; determining that there is motion of the location or the sound source within the 3D acoustic environment; updating, without user intervention and based on the motion, the set of parameters to maintain the orientation of the sound source towards the location.

Abstract: A bitstream is obtained by a decoder side, that contains an encoded version of an audio signal and an instantaneous loudness sequence of the audio signal. The instantaneous loudness sequence has not been loudness normalized. A dynamic range control, DRC, gain sequence is produced by applying the instantaneous loudness sequence to a DRC characteristic, with loudness normalization. The DRC gain sequence is applied to the decoded audio signal. Other aspects are also described and claimed.

Abstract: A method that includes receiving a bitstream that comprises: an encoded version of an audio signal that is associated with a sound source that is within a first 3D scene, a scene tree structure that includes an origin of the first scene relative to an origin of a second scene, and a position of the sound source within the first scene relative to the origin of the first scene, wherein the position references the origin of the first scene using an identifier, wherein the scene tree structure defines an initial configuration of the sound source with respect to the first and second scenes; determining a position of a listener; producing a set of spatially rendered audio signals by spatially rendering the audio signal according to the position of the sound source with respect to the position of the listener; and using the spatially rendered audio signals to drive speakers.

Abstract: A method that includes receiving a first bitstream that includes an encoded version of an audio signal for a three-dimensional (3D) scene and a first set of metadata that has 1) a position of a 3D sub-scene within the scene and 2) a position of a sound source associated with the audio signal within the sub-scene; determining a position of a listener; spatially rendering the scene to produce the sound source with the audio signal at the position of the sound source with respect to the position of the listener; receiving a second bitstream that includes a second set of metadata that has a different position of the sub-scene; and adjusting the spatial rendering of the scene such that the position of the sound source changes to correspond to movement of the sub-scene from the position of the sub-scene to the different position of the sub-scene.

Abstract: A method that includes receiving an audio component associated with an audio scene, the audio component including an audio signal, determining a loudness level of the audio component based on the audio signal, receiving a target loudness level for the audio component, producing a bitstream with the audio component by encoding the audio signal and including metadata that has the loudness level and the target loudness level, and transmitting the bitstream to an electronic device.

Abstract: A method that includes receiving a bitstream that includes: a first signal of a first audio component associated with an audio scene, a first target loudness, and a first source loudness determined by an encoder side based on the first signal, and a second signal of a second audio component associated with the scene, a second target loudness, and a second source loudness determined by the encoder side based on the second signal; determining a first gain based on the first source and target loudness; determining a second gain based on the second source and target loudness; producing a first gain-adjusted signal by applying the first gain to the first signal; producing a second gain-adjusted signal by applying the second gain to the second signal; and producing the scene that includes the first and second audio components by combining the gain-adjusted audio signals into a group of signals.

Abstract: A bitstream is obtained by a decoder side, that contains an encoded version of an audio signal and an instantaneous loudness sequence of the audio signal. The instantaneous loudness sequence has not been loudness normalized. A dynamic range control, DRC, gain sequence is produced by applying the instantaneous loudness sequence to a DRC characteristic, with loudness normalization. The DRC gain sequence is applied to the decoded audio signal. Other aspects are also described and claimed.

Abstract: The present invention relates to a method for continuously obtaining carbon dioxide from a carbon-dioxide containing atmosphere, in which a fibrous carrier material charged with polyethylene imine is guided alternately through at least one adsorption zone and at least one desorption zone. In addition, the present invention relates to a device by which the method according to the invention can be carried out.

Abstract: A device with microphones can generate microphone signals during an audio recording. The device can store, in an electronic audio data file, the microphone signals, and metadata that includes impulse responses of the microphones. Other aspects are described and claimed.

Abstract: A decoder-side method for outputting several audio channels of a sound program is described. An audio channel of the sound program, a residual signal, a gain parameter, and a delay parameter are received, for example within a bitstream. The audio channel is adjusted in accordance with the gain parameter and the delay parameter, to produce an adjusted audio signal, and is then combined with the residual signal to produce a combined signal. The audio channel is output as a first audio channel of the sound program for playback, while the combined signal is output as a second audio channel of the sound program. Other aspects are also described and claimed.

Abstract: Disclosed is an audio codec that represents an immersive signal by a two-channel stereo signal that is a stereo rendering of the immersive signal and directional parameters. The directional parameters may be based on a perceptual model describing the direction of virtual speaker pairs to recreate the perceived location of dominant sounds. Audio processing at the decoder may be performed on the stereo signal in the frequency domain for multiple channel pairs using time-frequency tiles. Spatial localization of the audio signals may use a panning approach by applying weightings to the time-frequency tiles of the stereo signal for each output channel pair. The weightings for the time-frequency tiles may be derived based on the directional parameters, an analysis of the stereo signal, and the output channel layout. The weightings may be used to adaptively process the time-frequency tiles using a decorrelator to reduce or minimize spectral distortions from spatial rendering.

Abstract: An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

Abstract: A system for producing an encoded digital audio recording has an audio encoder that encodes a digital audio recording having a number of audio channels or audio objects. An equalization (EQ) value generator produces a sequence of EQ values which define EQ filtering that is to be applied when decoding the encoded digital audio recording, wherein the EQ filtering is to be applied to a group of one or more of the audio channels or audio objects of the recording independent of any downmix. A bitstream multiplexer combines the encoded digital audio recording with the sequence of EQ values, the latter as metadata associated with the encoded digital audio recording. Other embodiments are also described including a system for decoding the encoded audio recording.

Abstract: A first layer of data having a first set of Ambisonic audio components can be decoded where the first set of Ambisonic audio components is generated based on ambience and one or more object-based audio signals. A second layer of data is decoded having at least one of the one or more object-based audio signals. One of the object-based audio signals is subtracted from the first set of Ambisonic audio components. The resulting Ambisonic audio components are rendered to generate a first set of audio channels. The one or more object-based audio signals are spatially rendered to generate a second set of audio channels. Other aspects are described and claimed.

Abstract: A first layer of data having a first set of Ambisonic audio components can be decoded where the first set of Ambisonic audio components is generated based on ambience and one or more object-based audio signals. A second layer of data is decoded having at least one of the one or more object-based audio signals. One of the object-based audio signals is subtracted from the first set of Ambisonic audio components. The resulting Ambisonic audio components are rendered to generate a first set of audio channels. The one or more object-based audio signals are spatially rendered to generate a second set of audio channels. Other aspects are described and claimed.

Abstract: A bitstream is obtained by a decoder side, that contains an encoded version of an audio signal and an instantaneous loudness sequence of the audio signal. The instantaneous loudness sequence has not been loudness normalized. A dynamic range control, DRC, gain sequence is produced by applying the instantaneous loudness sequence to a DRC characteristic, with loudness normalization. The DRC gain sequence is applied to the decoded audio signal. Other aspects are also described and claimed.

Abstract: Transfer functions can describe responses of microphones or ears to sounds at different locations on a sphere. The transfer functions can be compressed by determining, based on transfer functions, a) one or more basis transfer functions, and b) spherical harmonics coefficients that describe variations of the transfer functions with respect to spherical coordinates. Other aspects are described and claimed.