Dolby Laboratories Licensing Corporation

Dolby International Ab

Aaron Steven Master

Lie Lu

Alan Jeffrey Seefeldt

Jeffrey Charles Riedmiller

Harald Mundt

Arijit Biswas

Heidi-Maria Lehtonen

Marvin Pribadi

Jia Dai

Heiko Purnhagen

Yu Cecilia Hou

Computer-implemented methods and devices for combined audio separation and classification are provided. An estimated separated signal is time gated based on a determination of an audio classifier of, at least in part, the original mix of signals before separation. Combined separation, classification, and time gating of both the estimated signal and a residual signal are also provided.

Method and apparatus combining separation and classification of audio signals

A method comprises: obtaining softmask values for frequency bins of time-frequency tiles representing an audio signal; reducing, or expanding and limiting, the softmask values; and applying the reduced, or expanded and limited, softmask values to the frequency bins to create a time-frequency representation of an estimated target source. An alternative method comprises, for each time-frequency tile: obtaining softmask values; applying the softmask values to the frequency bins to create a time-frequency domain representation of an estimated target source; obtaining a panning parameter and a source concentration estimates for the target source; determining, using the panning parameter estimate and the softmask values, a magnitude for the time-frequency representation of the estimated target source; determining, using the panning parameter estimate and the source phase concentration estimate, a phase for the time-frequency representation of the estimated target source; and combining the magnitude and the phase.

Perceptual optimization of magnitude and phase for time-frequency and softmask source separation systems

In an embodiment, a spatio-level filter (SLF) is created by obtaining a first set of samples from a plurality of target source level and spatial distributions in frequency subbands in a frequency domain, obtaining a second set of samples from a plurality of background level and spatial distributions in frequency subbands in a frequency domain, adding the first and second sets of samples to create a combined set of samples, detecting level and spatial parameters for each sample in the combined set of samples for each subband, within subbands, weighting the detected level and spatial parameters by their respective level and spatial distributions for the target source and backgrounds; storing the weighted level, spatial parameters and signal-to-noise ratio (SNR) within subbands for each sample in the combined set of samples in a table; and re-indexing the table by the weighted level and spatial parameters and subband.

Separation of panned sources from generalized stereo backgrounds using minimal training

In an embodiment, a method comprises: transforming one or more frames of a two-channel time domain audio signal into a time-frequency domain representation including a plurality of time-frequency tiles, wherein the frequency domain of the time-frequency domain representation includes a plurality of frequency bins grouped into subbands. For each time-frequency tile, the method comprises: calculating spatial parameters and a level for the time-frequency tile; modifying the spatial parameters using shift and squeeze parameters; obtaining a softmask value for each frequency bin using the modified spatial parameters, the level and subband information; and applying the softmask values to the time-frequency tile to generate a modified time-frequency tile of an estimated audio source. In an embodiment, a plurality of frames of the time-frequency tiles are assembled into a plurality of chunks, wherein each chunk includes a plurality of subbands, and the method described above is performed on each subband of each chunk.

Methods, apparatus, and systems for detection and extraction of spatially-identifiable subband audio sources

Described herein is a method of low-bitrate coding of audio data and generating enhancement metadata for controlling audio enhancement of the low-bitrate coded audio data at a decoder side, including the steps of: (a) core encoding original audio data at a low bitrate to obtain encoded audio data; (b) generating enhancement metadata to be used for controlling a type and/or amount of audio enhancement at the decoder side after core decoding the encoded audio data; and (c) outputting the encoded audio data and the enhancement metadata. Described is further an encoder configured to perform said method. Described is moreover a method for generating enhanced audio data from low-bitrate coded audio data based on enhancement metadata and a decoder configured to perform said method.

Method and apparatus for controlling enhancement of low-bitrate coded audio

An input media signal that carries input media content is received. The input media content is used to generate output media content in an output media signal. It is determined whether identification-and-timing (IAT) data is to be authored for the output media content. In response to determining that the output IAT data is to be authored for the output media content, output IAT data is authored for the output media content. At least a part of the output IAT data for at least a part of the output media content is encoded, along with the part of the output media content, into the output media signal. In some example scenarios, this output media signal then contains the IAT data and other related data for synchronization of additional media content with the output media content in content rendering/presentation operations.

Identification and timing data for media content

Embodiments are described for calibrating a speaker in a device having a microphone by inputting an original audio signal to a processing component and an output stage of the device for playback through the speakers, receiving playback sound output from the speakers through the microphone to generate a microphone signal, and inputting the microphone signal into the processing component to calibrate the speakers for optimal playback of the original audio signal, wherein the processing component is configured to compare the original audio signal to the microphone signal and correct the microphone signal by one or more audio processing functions in accordance with a refresh schedule.

System and method to enhance speakers connected to devices with microphones

Essence of a media program can be encoded such that it carries essence identification information and a set of time points. Auxiliary content such as companion content and other auxiliary data can be associated with the essence of the media program by way of the set of time points. The set of time points and/or fingerprints used to retrieve/recover the set of time points can be delivered to one or more of a media device, a companion device/application, etc., in any of a variety of different methods. Through the set of time points and/or fingerprints, and user preferences, the auxiliary content can be retrieved by the one or more of the media device, the companion device/application, etc., presented and/or interacted with a user. In some embodiments, the auxiliary content as described herein can be customized for a specific sender and/or a specific recipient.

Growing community of inventors

San Francisco, CA, United States of America

Aaron Steven Master