Context-aware loudness control of audio content may include choosing from a plurality of loudness level models based on an audio reproduction device, measuring loudness level of the audio content based on the chosen loudness model, processing the real-time loudness measurement of the input audio signal to output real-time loudness level adjustment, processing a momentary loudness measurement of the input audio signal to output a momentary loudness level adjustment, processing a short-term loudness measurement of the input audio signal to output a short-term loudness level adjustment, adjusting the input audio signal based on the real-time, momentary, and short-term loudness level adjustments to output a post-processing input signal, measuring long-term loudness of the post-processing input signal to output a long-term loudness measurement, processing the long-term loudness measurement to output a post-processing level adjustment, and processing the real-time, momentary, short-term, and post-processing level adjustments to output an overall loudness level adjustment.

BACKGROUND

Programming such as television programs or theatrical feature films is, in many cases, produced with variable loudness and wide dynamic range to convey emotion or a level of excitement in a given scene. For example, a movie may include a scene with the subtle chirping of a cricket and another scene with the blasting sound of a shooting cannon. Interstitial material such as commercial advertisements, on the other hand, is very often intended to convey a coherent message, and is, thus, often produced at a constant loudness, narrow dynamic range, or both. Other type of content, such as news gathering, documentaries, children programming, modern music, classical music, live talk-shows, etc., may have inconsistent loudness levels or unpredictable loudness ranges.

Conventionally, annoying disturbances occurred at the point of transition between the various programming, and often between the programming and the interstitial material. This is commonly known as the “loudness inconsistency problem” or the “loud commercial problem.” In some cases, even when switching between programming and interstitial material that had matched average loudness and dynamic range, the loudness of the programming may decrease for artistic reasons for a period of time, possibly enough time to cause users to increase the volume of the audio. When this quieter-than-average section of the program switched to interstitial material that matched the original average loudness of the programming, the interstitial material may be too loud due to the increase in volume by the user.

This loudness inconsistency problem is experienced by TV viewers, radio listeners, and any other media user (such as web media, streaming, mobile, OTT, portable player, in-flight entertainment, etc.) when the reproduced content (or a sequence of different content) generates inconsistent, uncomfortable, or annoying sound pressure levels. Another example is a feature film being transmitted on TV or on a mobile device. Because of the way the film was initially produced for the theatrical representation, the modulation of its loudness levels would exceed the hearing comfort zone when reproduced in a home environment via a consumer device such as a TV set or a mobile device. The viewer/listener would have to repeatedly control the volume level of the device in order to make soft levels audible (like dialogs) and loud levels not annoying (like action scenes with loud music and sound effects).

Conventionally, processes addressing the loudness inconsistency problem modified the audio itself and at its source, thus making the processes irreversible. However, not all viewers may desire to have the programming audio changed in such a way. Furthermore, the user device could be used to retransmit the live stream to any other consumer device, rather than to reproduce the content itself. Consequently, reducing the dynamic range for fulfilling the audio characteristics of the receiver would generate a useless audio quality degradation in case the final reproduction device was capable of supporting larger dynamics or frequency range. Due to the variety of possible distribution platforms, predicting how a programming would be ultimately reproduced is no longer possible, and any audio processing applied beforehand could result inappropriate to the specific listening scenario.

Also conventionally, processes addressing the loudness inconsistency problem introduced sound artifacts or alterations to the spectral balance of the source content. These issues diminish the listener's experience.

SUMMARY

Context-aware loudness control of an audio stream may include processing the input audio signal in real-time based on a real-time loudness level adjustment, a momentary loudness level adjustment, a short-term loudness level adjustment, and a post-processing level adjustment to output an overall loudness level adjustment. The input audio signal may then be processed based on the overall loudness level adjustment to generate an output audio signal to be reproduced by the audio reproduction device.

Context-aware loudness control improves over prior art technology at least in that it controls based on psychoacoustic models and thus it emulates how human hearing works. It also improves over prior art technology in that it automatically and adaptively drives loudness level processing depending on real-time loudness analysis, on the specific audio reproduction device, and the listening scenario. The disclosed control does not apply dynamics processing to audio transients and does not rely on any compression or expansion technology that applies level gain to the source audio level. It only applies adaptation of its loudness component. Thus, the disclosed control preserves original sound quality and does not introduce spectral artifacts that degrade the audio.

The disclosed control is specifically designed to assess the loudness characteristic of an audio program and to hit the target loudness range and loudness level set for the specific use case (TV, Radio, Streaming, Inflight, Gaming, Car-Audio, etc.) It achieves the required target loudness range and target program loudness level while applying only the required amount of processing. This means that it does not reduce the loudness range if not necessary while it achieves loudness standard compliance. It automatically adapts its setting upon real-time assessment of the input signal and depending on the defined destination medium, reproduction device, or user's taste. Not affecting the source asset, but rather only processing the output signal, it allows parallel non-destructive operations, and thus concurrently accommodates multiple user scenarios. Furthermore, off-line processing is also supported.

DETAILED DESCRIPTION

Context-aware loudness control is based on psychoacoustic models to intelligently adapt loudness levels of audio content in real-time in order to hit a specific average target loudness level as well as to maintain the level range within a comfortable zone as perceived by the user. Where required, it consistently and adaptively modifies the loudness levels of source audio content in order to accommodate the technical requirement of the content provider/broadcaster/distributor or the listening preference of the final user. This may be accomplished by adapting loudness depending on the destination listening environment and/or the audio reproducing device. Context-aware loudness control may modify loudness levels according to pre-defined settings or according to custom adjustments operated by the user or system administrator, and depending on the reproducing device and/or its environment. This type of control is effective on real-time audio as well as on pre-stored audio. Similar technology as described herein in the context of real-time audio may be utilized offline to adapt pre-existing audio assets in a faster-than-real-time implementation. Since, for the pre-existing audio, we do not have to wait for the audio to produced (or reproduced), the pre-existing audio may be adapted all at once or at least significantly faster than if we were waiting for production (or reproduction) of the audio.

The term “real-time” or an operation occurring in “real-time” as used herein means that an operation is executed approximately within the time range of being perceived as immediate by a human being (i.e., the minimum sound duration necessary to human hearing to assess loudness). Studies claim that in regard to loudness perception this duration is approximately 200 ms.

The term “momentary measurement” as used herein means that the measurement is executed on the sliding window across the number of audio samples obtained as multiple of the number of audio samples defined for the computation of real-time measurement (Real-Time Measurement Window). Momentary measurement does not refer to similar parameters used in the audio literature where it is typically used to represent loudness level of very short program parts, from 50 to 500 ms (usually 400 ms). In fact, in the present disclosure it is also possible to use momentary measurement, and the consequent processing, to add a second pass of real-time processing, by means of setting the momentary measurement multiplier to 1. “Momentary processing” is the loudness adaptation applied after real-time processing based on the momentary measurement values.

The term “short-term measurement” as used herein means that the short-term measurement is executed on the sliding window across the number of audio samples obtained as multiple of the number of audio samples defined for the computation of real-time measurement (Real-Time Measurement Window). Short-term measurement does not refer to similar parameters used in the audio literature where it is typically used to represent loudness level of short program parts, from 500 ms to 10 seconds (usually 3 seconds). In fact, in the present disclosure it is also possible to use short-term measurement, and the consequent processing, to add a third pass of real-time processing, by means of setting the short-term measurement multiplier to 1. “Short-term processing” is the loudness adaptation applied after momentary processing based on the short-term measurement values.

The term “long-term measurement” as used herein means that the measurement is executed on the sliding window across the duration defined by the post-processing size. Long-term measurement is typically executed across a significant program part of which duration is sufficient to determine the average loudness level of the program. In the present disclosure, long-term measurement is used to compute the overall gain adaptation necessary to achieve loudness standard compliance. This duration is typically larger than 30 seconds. In the present disclosure, long-term measurement does not refer to similar parameters used in the audio literature.

Context-aware loudness control may be implemented in professional equipment such as transmitters, online distributors, plug-ins, audio-video hardware, audio mixers, digital-audio-workstations, video editing suites, radio workstations, music mixers, audio processors, servers and software, etc. as well as in commercial and home devices such as portable players, smart-phones, smart-TV sets, computers, tablets, hi-fi, home-cinema, car audio, gaming consoles, etc. Context-aware loudness control may be implemented for any type of programming including TV, radio, Public Address, live music, theatre, cinema, gaming, in-flight entertainment, web-streaming, internet cloud, Virtual Reality, etc., and any other production, distribution, transmission, or reproduction implementation where audio is involved. Loudness measurement and processes involved in context-aware loudness control are applicable to any sort of audio format (mono, stereo, multichannel, immersive, etc.)

FIG. 1illustrates a high-level block diagram of an exemplary system1for providing context-aware loudness control of audio content. The system1may be organized in three macro modules: loudness analyzer10, loudness pre-processor20(including real-time processor22, momentary processor24, and short-term processor26) and post-processor30. Upon proper setting, its processing may provide full loudness standard compliance according to the most common audio recommendations (e.g., ITU BS.1770-4).

Unlike other technologies used to process audio levels, the system1does not rely on any traditional audio level dynamics processing tool such as an audio compressor, automatic gain control (AGC), or expander based on audio signal magnitude control. This approach prevents the system1from introducing audio artifacts and sound degradation to the input audio signal5, allows tailored control on loudness parameters, and significantly improves the listening experience.

The system1is based on psychoacoustic loudness models that analyze all main aspects involved in human hearing: frequency, intensity, duration, masking and sound source direction. It extracts the loudness components of the audio asset5and intelligently adapts them in order to produce a comfortable listening experience with consistent average levels, without negatively affecting the output audio signal35. Unlike prior solutions, the system1relies on optimized processing that is capable of properly defining the ideal amount of required adaptation depending on input loudness levels, destination listening environment, reproducing device, and user's preferences. Since it is based on real-time non-destructive operations, multiple reversible processing can be operated concurrently, by means of several parallel units or systems.

The system1may also include the user profile40or profile menu, which stores profile settings that define how other portions of the system1operate. It communicates with the loudness analyzer10, real-time processor22, momentary processor24, short-term processor26, and post-processor30. The user profile40may also communicate with the digital level processor36and reproduction device70ofFIG. 2as described below. The profile settings can be either defined/recalled by the user, automatically recalled by the reproduction device70or by metadata included in the audio stream. The profile menu40may store several data necessary to process the audio signal according to the following aspects:user's preferencereproduction devicecontent metadata

User profiles may include universal profile, genre-oriented profiles (e.g., movie, music, news, etc.), as well as dialog-centric or agnostic adaptation profiles. The user profile40may store as settings one or more psychoacoustic loudness models and communicate the settings to the loudness meter10, the pre-processor20, the real-time processor22, the momentary processor24, the short-term processor26, the long-term loudness meter32, the post-processor30, and the digital level processor36such that they may perform based on the selected loudness model.

The system1may be integrated in an automated workflow or may be accessed via a control panel interface of the user profile40that sets the amount of targeted adaptation. This user profile40may be made available as a user's device application, allowing the user to customize his/her listening experience according to his/her own personal taste.

In one embodiment, the user profile40selects the loudness model from various loudness models based on the kind of the audio reproduction device (e.g., TV, radio, mobile phone, etc.) or the environment of the audio reproduction device (e.g., home, theatre, vehicle, etc.) Therefore, depending on the destination medium/device used to reproduce the audio program, the system1may define how loudness levels are measured and applied in real-time to the level gain control with the goal of providing adequate adaptation and achieve standard compliance.

The profiles menu40may also communicate with the audio reproduction device70, and select the loudness model from the plurality of loudness models based on at least one of a kind of the audio reproduction device or a measurement of an environment of the audio reproduction device.

As a result, the system1allows the user (or the content supplier/broadcaster/distributor) to select what profile or setting to be used for loudness processing. Profile's settings may be customized, saved, and recalled as user's settings.

The loudness analyzer10may split the input audio signal5into small slices which have a duration defined by the Real-Time Measurement Window as described below. Loudness levels of these audio slices may be measured with various integration times (real-time, momentary, short-term) and the gathered loudness values may be used by the pre-processor20to define how much level gain or attenuation should be applied to the input audio signal5in order to match with the target level.

The pre-processor20receives the input audio signal5and processes the input audio signal5to output a post-processing input signal PPIS. The pre-processor's gains may be further controlled and weighted by several additional parameters that aim at providing a smooth processing and at achieving the expected overall loudness range. After the pre-processor20the average program loudness level is measured again and the resulting value is used by the post-processor30to achieve standard compliance.

The system1may detect the specific amount of loudness correction required by acting on the micro and macro dynamics modulations of the input audio signal5. This way the mixing balance between voice, sound effects and music may be improved, resulting in increased dialog intelligibility, especially on devices where that aspect might be critical (e.g., inflight, mobile phones).

Also, by using loudness metadata45including, for example, loudness status, program level, dialog level, loudness range, real-time loudness level, and program duration, the system1may keep full control of any audio transmission while complying with international loudness recommendations.

FIG. 2illustrates a detailed block diagram of the exemplary system1for providing context-aware loudness control of audio content. The system1may be implemented in a machine (e.g., home audio receiver, TV, mobile phone, car-audio, etc.) or distributed among a group of machines. The system1receives incoming original audio assets consisting of, for example, audio signal, secondary event messages, and metadata. The audio assets may be passed to the high-pass filter50and/or the adaptive equalizer60while the secondary event messages and metadata may be passed to the loudness meter10, the digital level processor36, and the profiles menu40. This data may be used to recall/modify profile settings, to restart the loudness measurement and the loudness processors, or to set the loudness metadata of the digital level processor36on the Content Loudness Metadata (CLM) value if present, as explained below.

The system1may include a high-pass filter50that receives the input audio signal5and filters out low frequencies according to the parameter Audio High Pass (AHP): Cutoff Frequency (e.g., Hz) of the high pass filter.

The system1may also include the adaptive equalizer60that passes the filtered audio to the loudness meter10for measurement purposes and to the digital level processor36for level adaptation. The adaptive equalizer60may receive an audio asset and sound pressure level information from the audio reproduction device70as generated by the electroacoustic transducing system and process the audio asset based on the sound pressure level information. The adaptive equalizer60is particularly relevant to consumer devices. In the commercial user's device (e.g., mobile phone, TV, home receiver, etc.) only the user can operate the volume control of the reproduction device70in order to set the volume level (SPL) of the reproduction. This information may be passed to the adaptive equalizer60. The adaptive equalizer60receives the audio asset from the audio stream and SPL Level information from the reproduction device70. It may then smoothly process the frequency balance of the incoming audio content in order to balance the variation of loudness perception occurring at different SPL levels. This provides the best sonic experience to the listener because it compensates for the loss of energy at high and low frequencies occurring at low SPL values. The information generated by the reproduction device70in regard to output volume setting are also passed to the real-time loudness meter10which adapts its measurement filters accordingly in order to take into account the actual SPL generated by the consumer device70.

The loudness meter10measures loudness of the input audio signal based on a loudness model and outputs loudness levels of the input audio signal. The loudness meter10may assess the loudness levels of the audio content in real-time. The loudness meter10may select a loudness level model from a plurality of loudness models based on context such as the kind or type of the audio reproduction device36. The loudness meter10may also select a loudness level model from a plurality of loudness models as selected in the profile menu40.

The loudness meter10may also receive loudness metadata from the input audio5and data from the user profile or profile menu40. The loudness meter10may provide its output to the real-time processor22and the post-processor30. The loudness meter10may also apply gating, voice detection, or other loudness measurement implementations.

In one embodiment, the loudness meter10applies 75% overlap in the measurement of the real-time loudness level and it may be reset in order to begin a new measurement.

The frequency weighting filters of the real-time Loudness Meter may be specific for different channel groups (e.g., up to 7.1+4 or more).

The system1may also include an input level display15that displays the loudness level as measured by the loudness meter10.

The system1may also include the real-time processor22, which receives and processes the real-time loudness level of the input audio signal and outputs a real-time loudness level adjustment. The real-time processor22gathers values of instantaneous loudness levels (RTL) as provided by the loudness meter10and, according to the settings received from the profile menu40, computes the real-time loudness level adjustments that the digital level processor36should apply to the output. Absolute gating (e.g. −80 LUFS) may be applied to the measurement in order to hold the gain adaptation in case the incoming signal falls below such a threshold.

The system1may also include the momentary processor24, which receives and processes a momentary loudness level of the input audio signal5as received from the loudness meter10or the real-time processor22. The momentary processor24processes the momentary loudness level and outputs a momentary loudness level adjustment. The momentary processor24may gather momentary loudness values as provided at the output of the real-time processor22and, according to the Momentary Measurement Window size set in the profile menu40, may compute the momentary loudness level adjustments that the digital level processor36should apply. Absolute gating (e.g. −80 LUFS) may be applied to the measurement in order to hold the gain adaptation in case the incoming signal falls below such a threshold.

The system1may also include the short-term processor26, which receives a short-term loudness level of the input audio signal5from the loudness meter10, the real-time processor22, or the momentary processor24, process the short-term loudness level, and output a short-term loudness level adjustment.

The short-term processor26may gather short-term loudness levels as provided at the output of the momentary processor24and, according to a short-term measurement window set received from the profile menu40, compute the loudness level adjustments that the digital level processor36should apply. Absolute gating (e.g. −80 LUFS) may be applied to the measurement in order to hold the gain adaptation in case the incoming signal falls below such a threshold. Alternatively, the short-term processor26can be set with any size as a multiplier of the RTL window, and used to apply further serial real-time loudness adaptation.

The system1may also include a long-term loudness meter32that measures a long-term loudness of the post-processing input signal PPIS and outputs a long-term loudness level of the post-processing input signal. The system1may also include the post-processor30, which receives and processes at least the long-term loudness level of the post-processing input signal to output a post-processing level adjustment. In addition to processing by the real-time processor22, the momentary processor24, and the short-term processor26, the audio signal level may further be controlled via the post-processor30whose processing is based on a long-term loudness measurement (called LTL) performed on the post-processing input signal PPIS. PPIS is generated according to the adaptations computed by the whole pre-processor20(including the real-time processor22, the momentary processor24, and the short-term processor26). The newly generated audio signal PPIS is only used to compute the required gain adaptation applied by the post-processor30.

A brand new audio signal calculated at the output of the Pre-Processing module and is labeled PPIS (Post-Processing Input Signal). It is the result of the sum of the initial source audio signal (IS) and the gain adaptations RTG, MG, and STG computed by the individual Real-Time, Momentary, and Short-term sliding processors, respectively, used in the Pre-Processing module:
PPIS=IS+RTG+MG+STG

The LTL measurement may be performed on PPIS using a sliding window of the size defined by the Post-Processing Size (PPS), and in accordance of the selected measurement method (e.g., ITU-R.BS1770-4). It is used to compute the final Long-Term level processing necessary to smoothly align the Long-term Loudness Level to the Target Level. LTL measurement applies absolute gating at −70 LUFS. If LTL is below the absolute threshold Post-Processing gain may be put on hold. To assess the Long-term Loudness Level the post-processor30may optionally use relative gating (e.g., ITU-R.BS1770-2 onwards) or dialog detection, or other means of loudness assessment. To enable voice detection for computing LTL the parameter Voice Detection is set to 1.

The system1may also include an output level display34which displays the long-term loudness level as measured by the long-term loudness meter32.

The reproduction device70is the actual user equipment which includes the system to transform the digital audio signal into sound waves. It is typically a portable media player, a TV-set, a Home-Theatre system, a mobile phone, a car-audio receiver, etc. It is not included in the signal-flow of the professional chain such as broadcasting, distribution, offline broadcasting, streaming, etc. The reproduction device70can communicate:the Volume Control Level to the adaptive equalizer60, in order to balance the overall sonic characteristics of the audio content and to compensate the mismatch between the electroacoustic transducer and the human hearing perception, according to the generated sound pressure level.the Volume Control Level to the real-time loudness meter10, in order to adapt the loudness analysis of the input signal to the human hearing as actually perceived according to the actual generated sound pressure level.the acoustic finger-print of the electroacoustic transducer to the profile menu40in order to automatically recall the most appropriate setting to pursue the best sonic experience according to the selected reproduction device70.

The reproduction device70can further detect the noise level of the environment and accordingly automatically amend the profile settings in user profile40in order to provide the listener with consistent loudness levels and listening comfort.

The system1may also include a digital level processor36configured to receive and process the input audio signal5, the real-time loudness level adjustment, the momentary loudness level adjustment, the short-term loudness level adjustment, and the post-processing level adjustment to output an overall loudness level adjustment based on which the input audio signal5may be processed to output the output audio signal35to be reproduced by the audio reproduction device70. The overall loudness level adjustment is performed by the digital level processor36summing the individual computations performed by all processors and according to all the conditions previously defined. The digital level processor36processes the input audio signal5to output the output audio signal35to match a target loudness average level and a target loudness modulation within predetermined tolerance (e.g., 0.1 LU, 0.5 LU, 1 LU, 2 LU, 5 LU, etc.).

As stated above, the loudness analyzer10may split the input audio signal5into small slices which have a duration defined by the Real-Time Measurement Window. Loudness levels of these audio slices may be measured with various integration times and the gathered loudness values may be used by the pre-processor20to define how much level gain or attenuation should be applied to the input audio signal5in order to match with the target level.

The system1may include a channel split21that splits the audio signal into channels such as Left (L), Right (R), Center (C), Low Frequency Effect (LFE), Left Surround (Ls), Right Surround, (Rs), etc. so that each channel may be treated independently.FIG. 2illustrates only the Center channel. Similar architecture is used for the other channels.

The system1may operate in Multiband Mode and Wideband Mode.

In Multiband Mode, the system1includes multiband crossovers23that split the channel source audio signal into several frequency bands (in the following description, as well as inFIG. 2, a 3-band model is depicted. Less or more bands could similarly apply to the described technology). Wideband Mode Enable (WME): enables/disables Wideband Mode.

Example methods may be better appreciated with reference to the flow diagrams ofFIG. 3, which illustrates a flow diagram for an example method300for providing context-aware loudness control of audio content. At310, the method300loads a loudness level model chosen from a plurality of loudness level models. The loudness level model may be chosen from the plurality of loudness models based on an audio reproduction device70to reproduce an output audio signal derived from the input audio signal5. At320, the method300includes measuring a loudness level of the audio content using a loudness meter10based on the loudness model from the plurality of loudness model to output at least real-time loudness level of the input audio signal.

At330, the method300includes receiving and processing the real-time loudness measurement of the input audio signal to output a real-time loudness level adjustment. At340, the method300receives and processes the momentary loudness measurement of the input audio signal to output a momentary loudness level adjustment. At350, the method300includes receiving and processing a short-term loudness measurement of the input audio signal to output a short-term loudness level adjustment. At360, the method300includes adjusting the input audio signal based on the real-time loudness level adjustment, the momentary loudness level adjustment, and the short-term loudness level adjustment to output a post-processing input signal.

At370, the method300includes measuring a long-term loudness of the post-processing input signal to output a long-term loudness level of the post-processing input signal. At380, the method300receives and processes—the long-term loudness measurement of the post-processing input signal to output a post-processing level adjustment. At390, the method300receives and processes the input audio signal, the real-time loudness level adjustment, the momentary loudness level adjustment, the short-term loudness level adjustment, and the post-processing level adjustment to output an overall loudness level adjustment based on which the input audio signal is to be processed to output the output audio signal to be reproduced by the audio reproduction device70.

WhileFIG. 3illustrates various actions occurring in serial, it is to be appreciated that various actions illustrated could occur substantially in parallel, and while actions may be shown occurring in parallel, it is to be appreciated that these actions could occur substantially in series. While a number of processes are described in relation to example method300, it is to be appreciated that a greater and/or lesser number of processes could be employed and that lightweight processes, regular processes, threads, and other approaches could be employed. It is to be appreciated that other example methods may, in some cases, also include actions that occur substantially in parallel. Example method300and other embodiments may operate in real-time, faster than real-time in a software or hardware or hybrid software/hardware implementation, or slower than real time in a software or hardware or hybrid software/hardware implementation.

While for purposes of simplicity of explanation, the illustrated methodologies are shown and described as a series of blocks, it is to be appreciated that the methodologies are not limited by the order of the blocks, as some blocks can occur in different orders and/or concurrently with other blocks from that shown and described. Moreover, less than all the illustrated blocks may be required to implement an example methodology. Furthermore, additional methodologies, alternative methodologies, or both can employ additional, not illustrated blocks.

In the flow diagrams, blocks denote “processing blocks” that may be implemented with logic. The processing blocks may represent a method step and/or an apparatus element for performing the method step. The flow diagrams do not depict syntax for any particular programming language, methodology, or style (e.g., procedural, object-oriented). Rather, the flow diagrams illustrate functional information one skilled in the art may employ to develop logic to perform the illustrated processing. It will be appreciated that in some examples, program elements like temporary variables, routine loops, and so on, are not shown. It will be further appreciated that electronic and software applications may involve dynamic and flexible processes so that the illustrated blocks can be performed in other sequences that are different from those shown and/or that blocks may be combined or separated into multiple components. It will be appreciated that the processes may be implemented using various programming approaches like machine language, procedural, object oriented and/or artificial intelligence techniques.

FIG. 4illustrates a block diagram of an exemplary machine400for providing context-aware loudness control of audio content. The machine400includes a processor402, a memory404, file system430, and I/O Ports410operably connected by a bus408.

In one example, the machine400may transmit input and output signals including the audio signals (e.g., audio signal5, L, R, etc.) described above via, for example, I/O Ports410or I/O Interfaces418. The machine400may also include the loudness meter10, the pre-processor20, the split21, the real-time processor22, the momentary processor24, the short-term processor26, the pre-process low25, the pre-process mid27, the pre-process high28, the band mixer29, the long-term loudness meter32, the post-processor30, the digital level processor36, the profiles menu40, the high-pass filter50, the adaptive equalizer60, and all of their components. Thus, the loudness meter10, the pre-processor20, the split21, the real-time processor22, the momentary processor24, the short-term processor26, the pre-process low25, the pre-process mid27, the pre-process high28, the band mixer29, the long-term loudness meter32, the post-processor30, the digital level processor36, the profiles menu40, the high-pass filter50, the adaptive equalizer60, may be implemented in machine400as hardware, firmware, software, or combinations thereof and, thus, the machine400and its components may provide means for performing functions described herein as performed by the loudness meter10, the pre-processor20, the split21, the real-time processor22, the momentary processor24, the short-term processor26, the pre-process low25, the pre-process mid27, the pre-process high28, the band mixer29, the long-term loudness meter32, the post-processor30, the digital level processor36, the profiles menu40, the high-pass filter50, and the adaptive equalizer60.

The processor402can be a variety of various processors including dual microprocessor and other multi-processor architectures. The memory404can include volatile memory or non-volatile memory. The non-volatile memory can include, but is not limited to, ROM, PROM, EPROM, EEPROM, and the like. Volatile memory can include, for example, RAM, synchronous RAM (SRAM), dynamic RAM (DRAM), synchronous DRAM (SDRAM), double data rate SDRAM (DDR SDRAM), and direct RAM bus RAM (DRRAM).

A disk406may be operably connected to the machine400via, for example, an I/O Interfaces (e.g., card, device)418and an I/O Ports410. The disk406can include, but is not limited to, devices like a magnetic disk drive, a solid state disk drive, a floppy disk drive, a tape drive, a Zip drive, a flash memory card, or a memory stick. Furthermore, the disk406can include optical drives like a CD-ROM, a CD recordable drive (CD-R drive), a CD rewriteable drive (CD-RW drive), or a digital video ROM drive (DVD ROM). The memory404can store processes414or data416, for example. The disk406or memory404can store an operating system that controls and allocates resources of the machine400.

The bus408can be a single internal bus interconnect architecture or other bus or mesh architectures. While a single bus is illustrated, it is to be appreciated that machine400may communicate with various devices, logics, and peripherals using other busses that are not illustrated (e.g., PCIE, SATA, Infiniband, 1394, USB, Ethernet). The bus408can be of a variety of types including, but not limited to, a memory bus or memory controller, a peripheral bus or external bus, a crossbar switch, or a local bus. The local bus can be of varieties including, but not limited to, an industrial standard architecture (ISA) bus, a microchannel architecture (MCA) bus, an extended ISA (EISA) bus, a peripheral component interconnect (PCI) bus, a universal serial (USB) bus, and a small computer systems interface (SCSI) bus.

The machine400may interact with input/output devices via I/O Interfaces418and I/O Ports410. Input/output devices can include, but are not limited to, a keyboard, a microphone, a pointing and selection device, cameras, video cards, displays15and34, disk406, network devices420, and the like. The I/O Ports410can include but are not limited to, serial ports, parallel ports, and USB ports.

The machine400can operate in a network environment and thus may be connected to network devices420via the I/O Interfaces418, or the I/O Ports410. Through the network devices420, the machine400may interact with a network. Through the network, the machine400may be logically connected to remote devices. The networks with which the machine400may interact include, but are not limited to, a local area network (LAN), a wide area network (WAN), and other networks. The network devices420can connect to LAN technologies including, but not limited to, fiber distributed data interface (FDDI), copper distributed data interface (CDDI), Ethernet (IEEE 802.3), token ring (IEEE 802.5), wireless computer communication (IEEE 802.11), Bluetooth (IEEE 802.15.1), Zigbee (IEEE 802.15.4) and the like. Similarly, the network devices420can connect to WAN technologies including, but not limited to, point to point links, circuit switching networks like integrated services digital networks (ISDN), packet switching networks, and digital subscriber lines (DSL). While individual network types are described, it is to be appreciated that communications via, over, or through a network may include combinations and mixtures of communications.