Abstract: An apparatus and method of transmission control for an audio device. The audio device uses sources other than the microphone to determine nuisance, and uses this to calculate a gain as well as to make the transmit decision. Using the gain results in a more nuanced nuisance mitigation than using the transmit decision on its own.

Abstract: The processing of audio signals during playback is provided, so that audio signals that fall below a specified threshold loudness level are processed to avoid making unwanted background noise audible. N-channel audio is received from a playback volume controller/leveler (101). The level of the audio is compared with a threshold level. If the level is greater than the threshold level, the audio is processed with a first amount of gain in accordance with a first dynamic range control (DRC) compression curve that is tuned for professionally produced audio. If the level is less than or equal to the threshold level, the audio is processed with a second amount of gain in accordance with a second DRC compression curve that is designed to avoid boosting unwanted background noise. After applying the gain to the audio, the audio is sent to a downstream device.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A method and an apparatus of encoding/decoding intra prediction mode using a plurality of candidate intra prediction modes are disclosed. The method includes deriving three candidate intra prediction modes about a current block and deriving an intra prediction mode of the current block.

Abstract: Theater designs including social media, and theater entrances configured to draw patrons into a main area of the theater including an entrance having a curved wall and curved walkway having a horizon drop-off for display of content related to a motion picture or other feature to be shown in the theater. The content may be interactive or otherwise linked to patrons passing a hallway including the curved wall. Spatialized sound corresponding to the video “moves” with objects in the video and with patrons as they traverse walkway.

Abstract: A novel projection system includes first and second light sources (e.g., sets of lasers), a spatial light modulator (SLM) that receives light from the first light source, and a beam steering device that receives light from the second light source and steers the light to highlight regions of the SLM. The SLM then modulates the light from both light sources to generate a highlighted imaging beam which can then be projected on a viewing surface. The highlighted imaging beam can represent a highlighted 2D image or a highlighted left- or right-eye view of a 3D image. The projection system thus improves peak brightness in the displayed highlighted images without incorporating a separate highlight projector or other expensive equipment. Methods for highlighting projected images are also described.

Abstract: An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: A speckle damping system for dampening speckle on a projection screen for a projection display system employing coherent or partially coherent light sources (e.g., lasers, LEDs) are disclosed. In one embodiment, a rotatably coupled system is disclosed, comprising: a set of actuators; a set of rotatably coupled mounts, each of said set of rotatably coupled mount capable of mounting at least one said actuator; and wherein said at least one actuator mounted on said rotatably coupled mount is in moveable mechanical communication with said projection screen. In another embodiment, a linearly coupled system is disclosed comprising: a set of actuators; a set of linearly coupled mounts, each of said set of linearly coupled mount capable of mounting at least one said actuator; and wherein said at least one actuator mounted on said linearly coupled mount is in moveable mechanical communication with said projection screen.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: A system and method of adjusting an audio output in one location so that its propagation into another location is reduced. As a first device in a first location generates sound, a second device in a second location detects the propagated sound. The first device then adjusts its output based on the detected sound.

Abstract: A projector controller includes an object detector and control electronics, and is configured to protect audience members from intense light imposing an exclusion zone in front of a projector. The object detector is configured to optically sense a presence of an object in a detection region beneath the exclusion zone and above the audience members. The control electronics is configured to control the projector when the object detector indicates the presence of the object in the detection region. A method for protecting audience members from intense light imposing an exclusion zone in front of an output of a projector includes: (i) optically sensing a presence of an object in a detection region between the exclusion zone and the audience members, and (ii) controlling the projector when the presence of the object is sensed in the detection region.

Abstract: Given a sequence of images in a first codeword representation, methods, processes, and systems are presented for image reshaping using rate distortion optimization, wherein reshaping allows the images to be coded in a second codeword representation which allows more efficient compression than using the first codeword representation. Syntax methods for signaling reshaping parameters are also presented.

Abstract: An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF.

Abstract: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

Abstract: An apparatus and method of excursion protection of a loudspeaker. The method includes attenuating selected bands in a transform domain, controlled by a feedback signal resulting from an excursion transfer function that has been modified according to the real-time operational characteristics of the loudspeaker. In this manner, the system reduces the amount of wideband attenuation needed to address the predicted excursion, resulting in a better listening experience.

Abstract: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

Abstract: Microphone signals are received from microphones of a computer device. Each microphone signal of the microphone signals is acquired by a respective microphone of the microphones. A previously unselected microphone is selected from the microphones as a reference microphone, which generates a reference microphone signal. An adaptive filter is used to create, based on microphone signals of the microphones other than the reference microphone, predicted microphone signals for the reference microphone. Based on the predicted microphone signals for the reference microphone, an enhanced microphone signal is outputted for the reference microphone. The enhanced microphone signal may be used as microphone signal for the reference microphone in subsequent audio processing operations.

Abstract: Audio signals from microphones of a mobile device are received. Each audio signal is generated by a respective microphone of the microphones. First microphones are selected from among the microphones to generate a front audio signal. Second microphones are selected from among the microphones to generate a back audio signal. A first audio signal portion, which is determined based at least in part on the back audio signal, is removed from the front audio signal to generate a modified front audio signal. A second audio signal portion is removed from the modified front audio signal to generate a left-front audio signal. A third audio signal portion is removed from the modified front audio signal to generate aright-front audio signal.

Abstract: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

Abstract: The quantization parameter QP is well-known in digital video compression as an indication of picture quality. Digital symbols representing a moving image are quantized with a quantizing step that is a function QSN of the quantization parameter QP, which function QSN has been normalized to the most significant bit of the bit depth of the digital symbols. As a result, the effect of a given QP is essentially independent of bit depth a particular QP value has a standard effect on image quality, regardless of bit depth. The invention is useful, for example, in encoding and decoding at different bit depths, to generate compatible, bitstreams having different bit depths, and to allow different bit depths for different components of a video signal by compressing each with the same fidelity (i.e., the same QP).

Abstract: A method of calibrating a feedback-based noise cancellation system of an ear device may involve obtaining a measured plant response of the ear device, obtaining a reference plant response value and determining a plant response variation between the reference plant response value and a value corresponding to the measured plant response. The method may involve obtaining a measured a coupler response of the ear device, obtaining a reference coupler response value and determining a coupler response variation between the reference coupler response value and a value corresponding to the measured coupler response. The method may involve determining, based at least in part on the plant response variation and the coupler response variation, a microphone signal gain correction factor and applying the microphone signal gain correction factor to ear device microphone signals that are input to a feedback loop of the feedback-based noise cancellation system.

Abstract: 3D mapping statistics are generated for a first image of a first dynamic range and a second image of a second dynamic range different from the first dynamic range. Multivariate multiple regression (MMR) coefficients are generated by solving an optimization problem formulated using an MMR matrix built with the 3D mapping statistics without a letterbox constraint, and used to generate chroma mappings for predicting chroma codeword values of the second image. It is determined whether a letterbox exists in the images. If so, it is determined whether the chroma mappings accurately predict chroma codeword values in the second image. A reconstructed image generated by a recipient device by backward reshaping one of the images is rendered by a display device operating in conjunction with the recipient device.

Abstract: A pair of sets of audio blocks is determined from a multi-channel audio signal. The pair comprises a first set for a first time block over audio channels and a second set for a second time block over the audio channels. Audio features including inter-channel audio features are generated from the pair. The audio features are graphically presented with user interface components on a display page. A specific perceptible audio characteristic is visually conveyed to the user using the user interface components dynamically updated with sets of audio features computed from pairs of sets of audio blocks of the multi-channel audio signal.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (?e) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to ?e=?log2(?log2(?{square root over (KMAX)}·0)?+1)?.

Abstract: The present disclosure relates to methods, apparatus and systems for encoding an audio signal into a bitstream, in particular at an encoder, comprising: encoding or including audio signal data associated with 3DoF audio rendering into one or more first bitstream parts of the bitstream, and encoding or including metadata associated with 6DoF audio rendering into one or more second bitstream parts of the bitstream. The present disclosure further relates to methods, apparatus and systems for decoding an audio signal and audio rendering based on the bitstream.

Abstract: The present document describes a method (700) for encoding a multi-channel input signal (201). The method (700) comprises determining (701) a plurality of downmix channel signals (203) from the multi-channel input signal (201) and performing (702) energy compaction of the plurality of downmix channel signals (203) to provide a plurality of compacted channel signals (404). Furthermore, the method (700) comprises determining (703) joint coding metadata (205) based on the plurality of compacted channel signals (404) and based on the multi-channel input signal (201), wherein the joint coding metadata (205) is such that it allows upmixing of the plurality of compacted channel signals (404) to an approximation of the multi-channel input signal (201). In addition, the method (700) comprises encoding (704) the plurality of compacted channel signals (404) and the joint coding metadata (205).

Abstract: The present document relates to time-alignment of encoded data of an audio encoder with associated metadata, such as spectral band replication (SBR) metadata. An audio decoder configured to determine a reconstructed frame of an audio signal from an access unit of a received data stream is described. The access unit comprises waveform data and metadata, wherein the waveform data and the metadata are associated with the same reconstructed frame of the audio signal. The audio decoder comprises a waveform processing path configured to generate a plurality of waveform subband signals from the waveform data, and a metadata processing path configured to generate decoded metadata from the metadata.

Abstract: A method for creating an output soundfield signal from an input soundfield signal, the method including the steps of: (a) forming at least one delayed signals from the input soundfield signal, (b) for each of the delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and (c) combining together the acoustically transformed delayed signals and the input soundfield signal to produce the output soundfield signal.

Abstract: A standard dynamic range (SDR) image is received. Composer metadata is generated for mapping the SDR image to an enhanced dynamic range (EDR) image. The composer metadata specifies a backward reshaping mapping that is generated from SDR-EDR image pairs in a training database. The SDR-EDR image pairs comprise SDR images that do not include the SDR image and EDR images that corresponds to the SDR images. The SDR image and the composer metadata are encoded in an output SDR video signal. An EDR display operating with a receiver of the output SDR video signal is caused to render an EDR display image. The EDR display image is derived from a composed EDR image composed from the SDR image based on the composer metadata.

Abstract: Methods and systems for mapping images from a first dynamic range to a second dynamic range using a set of reference color-graded images and neural networks are described. Given a first and a second image representing the same scene but at a different dynamic range, a neural network (NN) model is selected from a variety of NN models to determine an output image which approximates the second image based on the first image and the second image. The parameters of the selected NN model are derived according to an optimizing criterion, the first image and the second image, wherein the parameters include node weights and/or node biases for nodes in the layers of the selected NN model. Example HDR to SDR mappings using global-mapping and local-mapping representations are provided.

Abstract: Embodiments are described for a high-frequency waveguide that improves the performance of large-scale surround sound and immersive audio environments. A horn waveguide is configured to be asymmetric about one of a vertical axis and horizontal axis of the waveguide to form an asymmetric horn waveguide. A spherical enclosure surrounds the asymmetric horn waveguide to form a horn speaker, and a three-axis mounting system is configured to fix the horn speaker to one of a wall or ceiling surface of the venue, wherein the mounting system facilitates rotating the horn speaker to a location that provides maximum coverage of the venue within the passband of the asymmetric horn waveguide.

Abstract: Smaller halftone tiles are implemented on a first modulator of a dual modulation projection system. This technique uses multiple halftones per frame in the pre-modulator synchronized with a modified bit sequence in the primary modulator to effectively increase the number of levels provided by a given tile size in the halftone modulator. It addresses the issue of reduced contrast ratio at low light levels for small tile sizes and allows the use of smaller PSFs which reduce halo artifacts in the projected image and may be utilized in 3D projecting and viewing.

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. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: The present document describes a method (900) for establishing control information for a control policy of a client (102) for streaming data (103) from at least one server (101, 701). The method (900) comprises performing (901) a message passing process between a server agent of the server (101, 701) and a client agent of the client (102), in order to iteratively establish control information. Furthermore, the method (900) comprises generating (902) a convergence event for the message passing process to indicate that the control information has been established.

Abstract: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

Abstract: An optical filter to increase contrast of an image generated with a spatial light modulator includes a lens for spatially Fourier transforming modulated light from the spatial light modulator, and an optical filter mask positioned at a Fourier plane of the lens to filter the modulated light. The modulated light has a plurality of diffraction orders, and the optical filter mask transmits at least one of the diffraction orders of the modulated light and block a remaining portion of the modulated light. A method that improves contrast of an image generated with a spatial light modulator includes spatially Fourier transforming modulated light from the spatial light modulator onto a Fourier plane, and filtering the modulated light by transmitting at least one diffraction order of the modulated light at the Fourier plane and blocking a remaining portion of the modulated light at the Fourier plane.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: The disclosed teleconferencing methods involve detecting a howl state during a teleconference which involves two or more teleconference client locations and a teleconference server. The teleconference server is configured for providing full-duplex audio connectivity between the teleconference client locations. The howl state is a state of acoustic feedback involving two or more teleconference devices in a teleconference client location. Detecting the howl state involves an analysis of both spectral and temporal characteristics of teleconference audio data. The disclosed teleconferencing methods involve determining which client location is causing the howl state and involve mitigating the howl state or sending a howl state detection message.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: Methods, systems, and computer program products for synchronizing audio signals captured by multiple independent devices during an audio event are described. Multiple recording devices, e.g. several smartphones, record the audio event. A computer system receives audio signals from the devices. The system determines a first delay between two audio signals based on cross-correlation of waveforms of the two audio signals. Subsequently, the system detects attacks that are present in each audio signal by computing the derivative of a respective envelope for each audio signal. The system determines a second delay between the two audio signals based on cross-correlation of attacks of the two audio signals. The system synchronizes the audio signals using the second delay upon determining that using the second delay improves sound quality over using the first delay.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: The invention proposes a method and a device for arithmetic encoding of a current spectral coefficient using preceding spectral coefficients. Said preceding spectral coefficients are already encoded and both, said preceding and current spectral coefficients, are comprised in one or more quantized spectra resulting from quantizing time-frequency-transform of video, audio or speech signal sample values.

Abstract: Shaped glasses have curved surface lenses with spectrally complementary filters disposed thereon. The filters curved surface lenses are configured to compensate for wavelength shifts occurring due to viewing angles and other sources. Complementary images are projected for viewing through projection filters having passbands that pre-shift to compensate for subsequent wavelength shifts. At least one filter may have more than 3 primary passbands. For example, two filters include a first filter having passbands of low blue, high blue, low green, high green, and red, and a second filter having passbands of blue, green, and red. The additional passbands may be utilized to more closely match a color space and white point of a projector in which the filters are used. The shaped glasses and projection filters together may be utilized as a system for projecting and viewing 3D images.

Abstract: The present disclosure relates to a method of decoding audio scene content from a bitstream by a decoder that includes an audio renderer with one or more rendering tools.

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: Audio data in a first format may be processed to produce audio data in a second format, which may be a reduced or simplified version of the first format. A loudness correction process may produce loudness-corrected audio data in the second format. A first power of the audio data in the second format and a second power of the loudness-corrected audio data in the second format may be determined. A second-format loudness correction factor for the audio data in the second format may be based, at least in part, on a power ratio between the first power and the second power. A first-format loudness correction factor for the audio data in the first format may be based, at least in part, on the power ratio and a power relationship between the audio data in the first format and the audio data in the second format.