System and method for adaptive audio signal generation, coding and rendering

Abstract

This embodiment illustrates an adaptive audio system for processing audio data comprising several independent mono audio streams. One or more streams have metadata associated with them, indicating whether the particular stream is channel-based or object-based. The channel-based stream has presentation information encoded by channel names; and the object-based stream has location information encoded by location representations in the associated metadata. The codec encapsulates the independent audio streams into a single serial bitstream containing all the audio data. This configuration allows for the presentation of sound based on referenced non-egocentric frames, where the sound's location is based on characteristics of the reproduction environment (e.g., room size, shape, etc.) to match the mixing engineer's intent. The object location metadata contains appropriate non-egocentric frames with reference information that require the correct speaker placement available in the room where the adaptive audio content is installed.

Description

Systems and methods for the generation, decoding, and presentation of adaptive audio signals For one or more implementations generally related to audio signal processing, and more particularly to object-based and channel-based audio processing for cinema, home, or other environments. The subject matter discussed in the background section should not be assumed to be prior art merely because of the results mentioned in the background section. Similarly, the problems mentioned in the background section or the subject matter related to the background section should not be assumed to have been previously recognized in the prior art. The subject matter in the background section only represents different methods, which may also be inventions in themselves or within them. Since the introduction of sound films, there has been a steady technological evolution to capture the artistic intent of the creator for the film sound track and reproduce it correctly in the film environment. The fundamental role of film sound is to support the story shown on the screen. A typical film sound track contains many different sound elements corresponding to elements and on-screen images, dialogues, noises, and sound effects originating from different on-screen elements, and combines background music and environmental sound effects to create an overall audience experience. The artistic intent of the creator and producer represents how they want these sounds to be reproduced in a way that closely matches the on-screen display in terms of the location, intensity, movement, and other similar parameters of the sound source. Current film creation, distribution, and reproduction are constrained by the limitations in creating truly immersive and realistic sound effects. Traditional channel-based audio systems transmit audio content in the form of speakers that are fed to individual speakers in the reproduction environment, such as stereo and 5.1 systems. The introduction of digital films has established new standards for sound on film, such as the combination of up to 16 audio channels to allow for greater creativity for content creators and a more encompassing and realistic auditory experience for the audience. The introduction of 7.1 surround systems has provided a new format that increases the number of surround channels by separating the existing left and right surround channels into four regions, thereby increasing the scope for sound designers and mixers to control the positioning of audio elements in the theater. To further improve the listener's experience, the reproduction of sound in a virtual three-dimensional environment has become an area of increasing research and development. The spatial representation of sound utilizes audio objects, which are audio signals that are parametric source descriptions related to the location of the apparent source (e.g., 3D coordinates), apparent source width, and other parameters. Object-based audio is increasingly being used in many current multimedia applications, such as digital films, video games, and 3D videos. Expanding beyond traditional speaker feeds and声道-based audio as a means for distributing spatial audio is crucial and has had considerable benefits in model-based audio descriptions, which hold the promise of allowing listeners / exhibitors to freely choose a reproduction configuration suitable for their individual needs or budgets, along with an audio rendering specifically tailored to their chosen configuration. At a high level, there are currently four main spatial audio description formats: speaker feed describes audio as signals intended for speakers at nominal speaker positions; microphone feed describes audio as signals captured by virtual or physical microphones in a predefined array; model-based description describes audio in terms of the sequence of audio events at described positions; and stereo describes audio as signals reaching the listener's ears. These four description formats are typically associated with one or more rendering techniques that transform the audio signal into a speaker feed. Today's rendering techniques include panning, where an audio stream is transformed into a speaker feed using a set of panning laws and known or assumed speaker positions (usually presented prior to distribution); high-fidelity stereophonic reproduction (Ambisonics), where microphone signals are transformed into feeds for a scalable array of speakers (usually presented after distribution); WFS (wave field synthesis), where sound events are transformed into appropriate signals to synthesize a sound field (usually presented after distribution); and binaural, where L / R (left / right) stereo signals are sent to the L / R ears, typically using headphones, but also by speakers and crosstalk cancellation (presented before or after distribution). Among these formats, the speaker-feed format is the most common for its simplicity and effectiveness. Since there is no processing between the content creator and the listener, the best sound results (the most accurate, reliable) are achieved directly by mixing / monitoring and distributing to the speaker feed. If the reproduction system is known in advance, the speaker-feed description generally provides the highest accuracy. However, in many practical applications, the reproduction system is unknown. The model-based description is considered the most suitable because it does not assume anything about the rendering technique and is thus the easiest to apply to any rendering technique. Although the model-based description efficiently captures spatial information, it becomes very inefficient as the number of audio sources increases. For many years, movie systems have had discrete screen channels characterized in the form of left, center, right, and sometimes "left inside" and "right inside" channels. These discrete sources typically have sufficient frequency response and power to operate to allow sound to be correctly placed in different areas of the screen and to allow timbre matching when the sound moves or pans between positions. Recent developments to improve the listener experience attempt to correctly reproduce the position of sound relative to the listener. In a 5.1 setup, the surround "zone" contains an array of speakers, all carrying the same audio information in each left surround or right surround area. Such an array with "ambient" or diffuse surround sound can be effective; however, many sound effects in everyday life come from randomly placed point sources. For example, in a restaurant, the ambient music is clearly audible all around, while subtle but discrete sounds originate from specific points: people talking from one point, the clatter of knives on plates coming from another. Discretely placing such sounds around the auditorium can add an increased sense of realism without being overly obvious. Sounds from above are also important surround-defining components. In the real world, sounds come from all directions and are not always from a single horizontal plane. If sounds can be heard from above (in other words, from the upper hemisphere), an added sense of realism can be achieved. However, current systems are unable to provide a true and correct reproduction of different audio types of sounds in the variations of different reproduction environments. Good countermeasures for processing, knowledge, and the actual reproduction environment configuration are needed to use existing systems to attempt to correctly characterize point-specific sounds, thus presenting a current system that is impractical for most applications. What is needed is a system that supports multiple screen channels, resulting in increased definition and improved audio-visual cohesion for on-screen sound or dialogue, and the ability to accurately position sources in any surround band to improve the audio-visual transition from the screen to the room. For example, if a character on the screen appears to be looking into a room towards a sound source, the sound engineer ("mixer") should have the ability to accurately position the sound so as to match the character's line of sight and for this sound effect to be continuously spread across the audience. However, in traditional 5.1 or 7.1 surround sound mixing, this sound effect highly depends on the listener's seating position, which is disadvantageous for most large-scale listening environments. Increased surround resolution creates new opportunities to use sound in a centered-in-the-room manner relative to traditional methods, where the content is assumed to be created for a single listener in the "sweet spot". In addition to spatial issues, the multichannel state of the current art system suffers from problems regarding timbre. For example, the timbre quality of some sounds, such as the hissing of water vapor from a ruptured pipeline, suffers from problems in reproduction by a loudspeaker array. The ability to direct a specific sound to a single loudspeaker gives the sound engineer the ability to eliminate the artifacts of array reproduction and deliver a more realistic experience to the audience. Traditionally, surround loudspeakers do not support the same global audio frequencies and levels as those supported by the large screen channels. Historically, this has created problems for sound engineers, reducing the freedom to move global sounds from the screen to the room. As a result, theater owners have not felt compelled to upgrade their surround sound configurations, hindering the widespread adoption of higher quality installations. Systems and methods are described for a cinema sound format and processing system that includes a new loudspeaker layout (channel configuration) and an associated spatial description format. Adaptive audio systems and formats are defined that support multiple presentation technologies. Audio streams are transmitted along with metadata that describes the "intent of the sound engineer" for the desired location of the audio stream. This location can be represented as a named channel (from within a predefined channel configuration) or as three-dimensional position information. This channel plus object format combines the best of channel-based and model-based audio scene description methods. The audio data for the adaptive audio stream system contains a number of independent mono audio streams. Each stream is associated with its metadata, which details whether the stream is a channel-based or object-based stream. Channel-based streams have presentation information encoded depending on the channel name; and object-based streams have fixed-point information encoded through a mathematical expression encoded in further associated metadata. The original independent audio streams are encapsulated in a bitstream of a single list containing all the audio data. This configuration allows sounds to be presented according to a reference non-ego-centric frame, where the fixed points of the presented sounds are based on the characteristics of the reproduction environment (such as the size, shape, etc. of the space) to conform to the intent of the sound engineer. The object position metadata contains a non-ego-centric frame of suitable reference information that requires the use of the loudspeaker positions available in the space (where the adaptive audio content is set to play) to correctly play the sound. It enables sounds to be optimally mixed for a particular reproduction environment, which may be different from the mixing environment experienced by the sound engineer. The adaptive audio system improves the audio quality in different rooms through benefits such as improved room equalization and bass management, so that loudspeakers (whether on-screen or off-screen) can be freely positioned by the sound engineer without considering timbre matching. The adaptive audio system adds the flexibility and power of dynamic audio objects to the traditional channel-based workflow. These audio objects allow the creator to control discrete sound elements, including overhead loudspeakers, regardless of any specific reproduction loudspeaker configuration. This system also introduces new efficiencies to the post-production workflow, allowing sound engineers to efficiently capture all of their intent and then, in real-time monitoring, or automatically generate 7.1 and 5.1 versions of the surround sound. An adaptive audio system in a digital movie processor simplifies distribution by encapsulating audio essence and artistic intent in a single-track file. The system provides optimal reproduction of artistic intent when mixing and presentation use the same channel configuration and have a single, downward-compatible list for presentation configuration (i.e., downmixing). These and other benefits are provided by way of an example, which relates to a movie sound platform, describes current system limitations, and delivers an audio experience that goes beyond currently available systems. 100: End-to-end environment 102: Content capture component 104: Audio preprocessing 106: Creation tools and processing 108: Codec 110: Presentation processing 112: Audio experience 200: Processing 202: Channel base data 204: Audio object data 206: Channel data + metadata 208: Adaptive audio mixing 300: Workflow 302: Input mixing control 304: Mixing console / audio workstation 306: Presentation and master unit 308: Other presentation deliverables 310: Adaptive audio mixing 312: Digital movie processor 314: Audio signal 316: Audio signal 318: Theater 320: Adaptive audio-enabled theater 400: System 402: Adaptive audio signal 404: Display 406: Level manager 408: Array correction component 410: B-chain processing component 500: Table 600: System 604: Re-recording mixer 606: Final mix 608: Presentation and master unit 610: Print host 612: Surround sound theater mix 614: Adaptive Audio Packaging 700: Workflow 702: Audio File 704: Packaging / Encryption Block 706: Block 708: Digital Movie Processor 710: Key Transfer Message Block 712: Compression 714: Subtitle 716: Video 800: Top View 801: Screen Speaker 802: Surround Speaker 804: Additional Speaker 806: Side Surround Speaker 808: Top Surround Speaker 810: Subwoofer 816: Additional Subwoofer 900: Front View 1000: Side View In the following figures, like reference numerals refer to like elements. Although the figures illustrate various examples, one or more implementations are not limited to the examples illustrated in the figures. FIG. 1 is a top-level overview of an audio creation and playback environment using an adaptive audio system in one embodiment. FIG. 2 illustrates the combination of channels and object-based data to create an adaptive audio mix in one embodiment. FIG. 3 is a block diagram illustrating a workflow for generating, packaging, and presenting adaptive audio content in one embodiment. FIG. 4 is a block diagram of the presentation stage of an adaptive audio system in one embodiment. FIG. 5 is a table listing metadata types and associated metadata elements of an adaptive audio system in one embodiment. FIG. 6 is a diagram illustrating post-production and control of an adaptive audio system in one embodiment. FIG. 7 is an example workflow diagram for a digital movie packaging process using an adaptive audio file in one embodiment. FIG. 8 is a top view of a demonstration layout for speaker placement recommended for use with an adaptive audio system in a typical auditorium. FIG. 9 is a front view of a demonstration placement for speaker placement recommended on a screen in a typical auditorium. FIG. 10 is a side view of a demonstration layout for speaker positioning recommended for use with an adaptive audio system in a typical auditorium. FIG. 11 is an exemplary positioning of a top surround speaker and a side surround speaker with respect to a reference point in one embodiment. Systems and methods for describing audio signals and data formats related to adapting audio systems and supporting multiple rendering techniques. Aspects of one or more embodiments described herein may be implemented in an audio or audiovisual system that processes source audio information in a mixing, rendering, and reproduction system that includes one or more computers or processing devices that execute software instructions. Any of the embodiments described may be used alone or in any easy combination with another. Although various embodiments may be motivated by various deficiencies of the prior art (discussed or implied at one or more places in the specification), the embodiments need not address any of these deficiencies. In other words, different embodiments may address different deficiencies discussed in the specification. Some embodiments may only partially address some deficiencies or one deficiency discussed in the specification, and some embodiments may not address any of these deficiencies. For the purposes of this description, the following terms have the associated meanings: Channel or audio channel: A mono audio signal or audio stream plus metadata at a position decoded by a channel ID (e.g., front left or right surround). A channel object may drive multiple speakers, e.g., the left surround channel (Ls) will feed all speakers in the Ls array. Channel configuration: A predefined set of speaker zones associated with a nominal fixed point, e.g., 5.1, 7.1, etc.; 5.1 refers to a six-channel surround audio system that has front left and right channels, a center channel, two surround channels, and a subwoofer channel; 7.1 refers to an eight-channel surround system that adds two additional surround channels to a 5.1 system. Examples of 5.1 and 7.1 configurations include Dolby surround systems. Speaker: An audio transducer or a group of transducers that presents an audio signal. Speaker zone: One or more speaker arrays that can be uniquely referenced and that receive a single audio signal, e.g., left surround as seen generally in a cinema, and particularly for object rendering excluding or including. Speaker channel or speaker feed channel: An audio channel associated with a named speaker or a speaker zone within a defined speaker configuration. A speaker channel nominally presents using the associated speaker zone. Speaker channel group: A group of one or more speaker channels corresponding to a channel configuration (e.g., stereo track, mono track, etc.). Object or object channel: One or more audio channels with parameter source descriptions, such as source location (e.g., 3D coordinates), source width, etc. An audio stream plus metadata at a position decoded in a 3D spatial position. Audio program: A complete set of speaker channels and / or object channels and associated metadata that describes the desired spatial audio performance. Non-egocentric reference: Defining spatial reference in an audio object related to the spatial reference of features within a presentation environment such as indoor walls and corners, standard speaker locations, and screen locations (e.g., the front left corner of a room). Egocentric reference: Defining spatial reference in an audio object related to the (listener's) listener's perspective and typically specified as related to an angle relative to the listener (e.g., 30 degrees to the right of the listener). Frame: A frame is a section into which a short, independently encodable whole audio program is divided. The audio frame rate and boundaries typically coincide with those of the video frame. Adaptive audio: Channel-based and / or object-based audio signals plus metadata for presenting the audio signal based on the reproduction environment. The movie sound formats and processing systems described herein (also referred to as "adaptive audio systems") utilize new spatial audio description and rendering techniques to allow for enhanced listener immersion, more artistic control, system flexibility and scalability, and ease of installation and maintenance. Examples of movie audio platforms include a number of distributed components, including mixing tools, packager / encoders, disassembler / decoders, theater final mix and rendering components, new speaker designs, and network amplifiers. The system includes recommendations for new channel configurations to be used by content creators and presenters. The system utilizes model-based descriptions that support features such as: a single inventory with downward and upward adaptive presentation configurations, i.e., deferred presentation and enabling optimal use of available speakers; improved sound development, including optimized downmixing to avoid channel correlations; increased spatial resolution through steer-thru arrays (e.g., dynamic assignment of audio objects to speakers within one or more surround arrays); and support for alternative presentation methods. Figure 1 is a top-level overview of an audio creation and reproduction environment using an adaptive audio system in one embodiment. As shown in Figure 1, the comprehensive, end-to-end environment 100 includes content creation, packaging, distribution, and reproduction / presentation components across a multitude of endpoint devices and usage scenarios. The overall system 100 originates from content captured from and for a number of different usage scenarios that include different user experiences 112. For example, the content capture component 102 includes cinema, TV, live broadcast, user-generated content, recorded content, games, music, etc., and may include audio / visual or pure audio content. As the content progresses through the system 100 from the capture stage 102 to the end-user experience 112, the content traverses some key processing steps through the distributed system components. These processing steps include pre-processing 104 of the audio, authoring tools and processing 106, via an audio codec 108 (which captures, for example, audio data, additional metadata, and reproduction information), and object channel coding. Various processing effects, such as compression (lossy or lossless), encryption, etc. may be applied to the object channels for efficient and secure distribution through various media. Suitable endpoint-specific decoding and presentation processing 110 is then applied to reproduce and deliver a particular adaptive audio user experience 112. The audio experience 112 represents the reproduction of audio or audio / visual content through suitable speakers and reproduction devices, and may represent any environment in which the listener is experiencing the reproduced captured content, such as a cinema, concert hall, outdoor theater, home or indoor, listening stand, car, gaming console, headphones or headphone system, public address (PA) system, or any other reproduction environment. Embodiments of the system 100 include an audio codec 108 and storage of multi-channel audio programs that are capable of efficient distribution, and may thus be referred to as a "hybrid" codec. The codec 108 combines traditional channel-based audio data with metadata to create audio objects that facilitate the creation and transmission of audio adapted and optimized for presentation and reproduction in an environment (which may be different from the mixing environment). This enables a sound engineer to encode his or her intentions regarding how the final audio should be heard by the listener based on the listener's actual listening environment. Conventional channel-based audio codecs operate under the assumption that an audio program will be reproduced by an array of speakers at a predefined position relative to the listener. To create a complete multi-channel audio program, a sound engineer generally mixes a large number of separate audio streams (e.g., dialogue, music, sound effects) to create the overall desired impression. The audio mixing decisions are made by listening to the audio program as reproduced by an array of speakers at a predefined position, such as a particular 5.1 or 7.1 system in a specific theater. Finally, the mixed signal is provided as input to the audio codec. For reproduction, the correct spatial sound field is achieved only when the speakers are placed at the predefined position. A new form of audio coding, called audio object coding, provides distinct sound sources (audio objects) as inputs to an encoder in the form of separate audio streams. Examples of audio objects include dialogue tracks, individual instruments, individual sound effects, and other point sources. Each audio object is associated with spatial parameters, which may include (but are not limited to) sound position, sound width, and velocity information. The audio objects and associated parameters are then coded for distribution and storage. The final audio object mixing and presentation is performed at the receiving end of the audio distribution chain as part of the audio program reproduction. This step can be based on knowledge of the actual loudspeaker positions so that the result is an audio distribution system that can be customized for the user's specific listening conditions. These two coding forms (channel-based and object-based) are optimally implemented for different input signal conditions. Channel-based audio coding is generally more efficient for coding input signals that contain dense mixes of different audio sources and for diffuse sounds. In contrast, audio object coders are more efficient for coding a few highly directional sound sources. In one embodiment, the methods and components of system 100 include an audio encoding, distribution, and decoding system configured to generate one or more bitstreams that contain both conventional channel-based audio components and audio object coding components. Such a combined approach provides greater coding efficiency and presentation flexibility compared to methods that separately employ either channel-based or object-based coding alone. Other aspects of the illustrative embodiments include extending a predetermined channel-based audio codec in a backward-compatible manner to include audio object coding components. A new "extension layer" containing the audio object coding components is defined and added to the "base" or "backward-compatible" layer of the channel-based audio codec bitstream. This approach enables one or more bitstreams that include the extension layer to be processed by conventional decoders while providing an enhanced listener experience for users with the new decoder. An example of enhancing the user experience includes control of audio object presentation. An additional benefit of this approach is that audio objects can be added or modified anywhere along the distribution chain without being decoded / mixed / recoded as multi-channel audio coded by the channel-based audio codec. Regarding a reference frame, the spatial effect of an audio signal is crucial in providing an impressive experience to the listener. This means that the sound emitted from a specific area of the screen or the room should be played through speakers located at the same relative fixed points. Thereby, the main audio (single-track) metadata of a sound event described on a model basis is located through other parameters such as size, orientation, speed, and auditory dispersion that can also be described. Communicating fixed points, model-based, 3D, audio spatial descriptions requires a 3D coordinate system. The coordinate system used for transmission (Euclidean, spherical, etc.) is generally chosen for convenience or simplicity. However, other coordinate systems can also be used for rendering processing. In addition to the coordinate system, the reference frame needs to represent the fixed points of objects in space. For correctly reproducing position-based sounds in various environments, choosing an appropriate reference frame can be a crucial factor. With a non-egocentric reference frame, the audio source position is defined relative to features in the presentation environment, such as indoor walls and corners, standard speaker fixed points, and screen fixed points. In an egocentric reference frame, the fixed points represent the listener's perspective, such as "in front of me, slightly to the left", etc. Scientific research on spatial perception (audio and otherwise) has shown an almost universal egocentric perspective. However, for a cinema, for some reasons, a non-egocentric one is more suitable. For example, when the audio object is associated with an object on the screen, the accurate localization of the audio object is most important. Using a non-egocentric reference, for each listening position and for any screen size, the sound will be localized at the same relative position on the screen, such as one-third to the left of the middle of the screen. Another reason is that sound mixers tend to think and mix in non-egocentric terms and layout panning tools in a non-egocentric frame (indoor walls), and sound mixers expect them to be presented in that way, such as this sound should be on the screen, this sound should be off the screen or coming from the left wall, etc. Despite using a non-egocentric reference frame in a movie environment, there are situations where an egocentric reference frame of reference would be beneficial and more suitable. These include non-diegetic sounds, that is, those that do not appear in the "story space", such as ambient music, for which an egocentric consistent representation can be satisfactory. Another situation is near-field sound effects (such as a mosquito buzzing in the listener's left ear) which require an egocentric representation. Currently, there are no tools for presenting such a sound field without using headphones or very near-field speakers. In addition, infinitely distant sound sources (and the resulting plane waves) appear to come from a fixed egocentric position (such as 30 degrees to the left), and such sounds are easier to describe on an egocentric item compared to a non-egocentric one. In some cases, as long as it is possible to define a nominal listening position using a reference non-egocentric frame, while some examples require egocentric representations that are not yet possible to present. Although non-egocentric references can be more beneficial and suitable, due to many new features, including egocentric representations being more satisfactory in certain applications and listening environments, audio representations should be extensible. Examples of adaptive audio systems include hybrid spatial description methods that include recommended channel configurations for best fidelity and for diffuse or complex presentations, use of multi-point sources (such as stadium crowds, atmosphere) using egocentric references, plus non-egocentric, model-based sound descriptions to enable increased spatial resolution and scalability efficiently. System components Referring to FIG. 1, the original audio content data 102 is first processed in the preprocessing block 104. The preprocessing block 104 of the system 100 includes an object channel filtering component. In many cases, audio objects contain individual sound sources to enable independent sound panning. In some cases, such as when creating an audio program using natural or "produced" sounds, it may be necessary to record from multiple sound sources to extract individual sound objects. Examples include methods for isolating independent source signals from more complex signals. Unwanted components separated from the independent source signals may include (but are not limited to) other independent sound sources and background noise. In addition, reverb may be removed to restore the "dry" sound source. The preprocessor 104 also includes source separation and content type detection functions. This system provides automatic generation of metadata through input audio analysis. Metadata for location originates from multi-channel recordings by analyzing the relative levels of the inputs between channel pairs. Detection of content type (such as "speech" or "music") can be achieved through feature extraction and classification. Authoring tools The authoring tool block 106 includes features to optimize the creation of audio programs by input and decoding of the intentions of the sound engineer's creativity, once it is optimized for reproduction in virtually any reproduction environment, the intention of creativity allows him to create the final audio mix. This is accomplished through the use of audio objects and location data associated with and encoded with the original audio content. To correctly place the sound near the auditorium, the sound engineer needs control over how the sound will ultimately be presented based on the actual limitations and characteristics of the reproduction environment. The adaptive audio system provides this control by allowing the sound engineer to change how the audio content is designed and mixed through the use of audio objects and location data. An audio object can be regarded as a grouping of sound components that can be perceived as emanating from a particular physical location or a location within an auditorium. Such objects can be static, or they can move. In an adaptive audio system 100, the audio objects are controlled by metadata that, among other things, specifies the sound position at a given point in time. When the objects are monitored or played back behind the theater, they are rendered based on the position metadata of the existing loudspeakers and are not necessarily output to physical channels. Tracks during a session can be audio objects, as well as standard panning data analogous to the position metadata. As such, content placed on the screen can be panned in the same effective manner as channel-based content, but content placed in the surround, if desired, can be rendered on individual loudspeakers. While the use of audio objects provides the desired control for discrete sound effects, other aspects of the soundtrack operate really effectively in a channel-based environment. For example, many ambient sounds or reverberations benefit from being fed to a loudspeaker array. Although these can be regarded as objects wide enough to fill the array, it is advantageous to retain some channel-based functionality. In one embodiment, in addition to audio objects, the adaptive audio system supports "beds", where a bed is an effective channel-based submix or mix element (stems). Depending on the intent of the content creator, these can be sent for individual final rendering, or combined into a single bed. These beds can be created in different channel-based configurations (such as 5.1, 7.1) and can extend to a wider format such as 9.1, and include overhead loudspeaker arrays. Figure 2 illustrates, in one embodiment, the combination of channel and object-based data to produce an adaptive audio mix. As shown in process 200, channel-based data 202, such as 5.1 or 7.1 surround sound data provided in the form of pulse code modulation (PCM) data, is combined with audio object data 204 to produce an adaptive audio mix 208. The audio object data 204 is produced by combining elements of the original channel-based data that are associated with certain metadata attached to the audio object location parameters. Conceptually, as shown in FIG. 2, the authoring work provides the ability to create an audio program that includes a combination of speaker channels grouped and object channels. For example, an audio program can include one or more selectable organized component groups (or tracks, such as stereo or 5.1 tracks) of speaker channels, descriptive metadata for one or more speaker channels, one or more object channels, and descriptive metadata for one or more object channels. Within an audio program, each speaker channel group and each object channel can represent the use of one or more different sampling rates. For example, applications in digital cinema (D-Cinema) support sampling rates of 48 kHz and 96 kHz, but other sampling rates can also be supported. In addition, acquisition, storage, and editing of channels with different sampling rates can also be supported. The creation of an audio program requires steps in sound design, which includes combining sound components as a sum of adjusted component levels to create a new, desired sound effect. Authoring tools adapted to the audio system enable the creation of sound effects as a collection of sound objects with relative positions using a spatial audio design graphical user interface. For example, a visual representation of a sound-producing object (such as a car) can be used as a template for fitting audio components (exhaust note, tire hum, engine noise), which are object channels that include the sound and a suitable spatial position (at the tailpipe, tires, hood). The individual object channels can then be linked and manipulated as a group. The authoring tool 106 includes some user interface components to allow the sound engineer to input control information and view mixing parameters, and to improve system functionality. The sound design and authoring process can also be improved by allowing object channels and speaker channels to be linked and manipulated as groups. An example is to combine an object channel with a dry sound source of a speaker channel that is spread out with a set of associated reverberation signals. The audio authoring tool 106 supports the ability to combine multiple audio channels, generally referred to as mixing. Multiple methods of mixing are supported and can include traditional level-based mixing and loudness-based mixing. In level-based mixing, broadband scaling is applied to the audio channels, and the scaled audio channels are then summed together. The broadband scaling factor for each channel is selected to control the absolute level of the resulting mixed signal and also to control the relative levels of the mixed channels within the mixed signal. In loudness-based mixing, frequency-dependent amplitude scaling is used to modify one or more input signals, where the frequency-dependent amplitude is selected to provide the desired perceived absolute and relative loudness while preserving the perceived timbre of the input sound. The creation tool allows the ability to create speaker channels and groupings of speaker channels. This allows metadata to be associated with each speaker grouping. Each speaker grouping can be tagged according to the content type. The content type is extensible via a text description. The content type can include (but is not limited to) dialogue, music, and sound effects. Each speaker channel grouping can be assigned unique instructions on how to upmix from one channel configuration to another, where upmix is defined as creating M audio channels from N channels, where M > N. The upmix instructions can include (but are not limited to) the following: an enable / disable flag to indicate whether upmixing is permitted; an upmix matrix to control the mapping between each input and output channel; and default enable and matrix settings can be assigned based on the content type, such as enabling upmixing only for music. Each speaker channel grouping can also be assigned unique instructions on how to downmix from one channel configuration to another, where downmix is defined as creating Y audio channels from X channels, where Y < X. The downmix instructions can include (but are not limited to) the following: a matrix to control the mapping between each input and output channel; and default matrix settings can be assigned based on the content type, such as dialogue should be downmixed to on-screen; sound effects should be downmixed off-screen. Each speaker channel can also be associated with a metadata flag during presentation to disable bass management. Examples include features enabling the creation of object channels and groupings of object channels. This invention allows metadata to be associated with each object channel grouping. Each object channel grouping can be tagged according to the content type. The content type is extensible via a text description, where the content type can include (but is not limited to) dialogue, music, and sound effects. Each object channel grouping can be assigned metadata to describe how the object should be presented. Location information is provided to indicate the desired sound source location. A frame of reference that is ego-centric or non-ego-centric can be used to indicate the location. An ego-centric reference is appropriate when the source location is referenced to the listener. For an ego-centric location, spherical coordinates are useful for location description. A non-ego-centric reference is the typical frame of reference used for movies or other audio / visual presentations, where the source location is referenced relative to objects in the presentation environment such as a visual display screen or room boundaries. Three-dimensional (3D) track information is provided to enable interpolation of the location or for use in other presentation decisions such as enabling a "snap to" mode. Dimension information is provided to indicate the desired apparent size of the audio source. Spatial quantization is provided to control the intention by a sound engineer or mixer through "locking to the nearest speaker" to have an object presented by exactly one speaker (with a potential sacrifice for spatial correctness). The limits of allowed spatial distortion are indicated through height and azimuth tolerance thresholds, such that if the thresholds are exceeded, the "locking" function does not occur. In addition to the distance threshold, a crossfade rate parameter is indicated to control how quickly a moving object will transfer or jump from one speaker to another when the desired position crosses between speakers. In one embodiment, dependent spatial metadata is used for some location metadata. For example, metadata can be automatically generated for "slave" objects, which are generated by the "master" object (which the slave object follows) associated with it. A time lag or relative speed can be assigned to the slave object. An apparatus can also be provided for a collection or grouping of objects to allow for the definition of a gravitational auditory center, such that objects can be presented to cause the perception of movement near another object. In such a case, one or more objects can rotate near an object or defined area, such as a dominant point or an indoor dry area. Even though the ultimate fixed-point information will be expressed relative to a fixed point in the room, such as relative to a fixed point relative to another object, the gravitational auditory center will then use this at the rendering stage to help determine the fixed-point information for the base sound of each suitable object. When an object is presented, it is assigned to one or more speakers based on the location metadata, and the fixed points of the reproducing speakers. Additional metadata can be associated with the object to limit the speakers that should be used. The limited use can prohibit the use of the indicated speakers or only prohibit the indicated speakers (allowing lower energy to enter the speaker or speakers for different applications). The limited set of speakers can include (but is not limited to) any named speaker or speaker zone (such as L, C, R, etc.) or speaker areas, such as: the front wall, the back wall, the left wall, the right wall, the ceiling, the floor, the speakers in the room, etc. Similarly, during the mixing of a particular desired set of sound components, it is possible for one or more sound components to become silent or "masked" due to the presence of other "masking" sound components. For example, when masking components are detected, they can be identified to the user via a graphical display. As described elsewhere, the audio program description can be adapted to be presented in a wide variety of speaker installations and channel configurations. When creating an audio program, it is important to monitor the sound effect of the rendering program in the expected reproduction configuration to verify that the desired result is achieved. This invention includes the ability to select a target reproduction configuration and monitor the results. In addition, the system can monitor the worst-case (i.e., highest) signal level, which will occur in each expected reproduction configuration, and provide an indication if clipping or limiting will occur. FIG. 3 is a block diagram illustrating a workflow for creating, packaging, and presenting adaptive audio content in one embodiment. The workflow 300 of FIG. 3 is separated into three distinct task groupings, labeled creation / authoring, packaging, and presentation. Generally, the music and object hybrid model as shown in FIG. 2 allows sound design, editing, pre-mixing, and final mixing to be performed in the same manner as they are today without overly adding current processing on top. In one embodiment, the adaptive audio functionality is provided in the form of software, firmware, or circuitry, which is used to interface with sound production and processing equipment, where such equipment can be a new hardware system or an upgrade to an existing system. For example, a plug-in application can be provided for a digital audio workstation to allow existing panning techniques within a sound setup and editing to remain unchanged. In this way, it is possible to lay down both music and objects within a workstation in a 5.1 or similar surround-equipped editing room. Object audio and metadata are recorded during sessions that are prepared for pre- and final mixing stages in a dubbing theater. As shown in FIG. 3, the creation or authoring task includes inputting mixing controls 302 to a mixing console or audio workstation 304 by a user (such as a sound engineer in the following example). In one embodiment, metadata is integrated into the mixing console surface, allowing operation of the microphone, panning, and audio processing of the track strip with both music or mixing elements and audio objects. The metadata can be edited using either the console surface or the workstation user interface, and the sound is monitored using a rendering and mastering unit (RMU) 306. The music and object audio data and associated metadata are recorded during a master session to create a "print master" that includes an adaptive audio mix 310 and any other rendered deliverables (such as a surround 7.1 or 5.1 theater mix) 308. Existing authoring tools (such as a digital audio workstation like Pro Tools) can be used to allow a sound engineer to label individual audio tracks during a mixing session. This concept is extended in an embodiment by allowing the user to label individual sub-segments within a track to help find or quickly identify audio objects. The user interface of the mixing console for enabling the definition and creation of metadata can be implemented through graphical user interface elements, physical controls (such as sliders or knobs), or any combination thereof. During the packaging stage, to ensure the integrity of the audio content for transmission to a digital cinema packaging facility, the print master file is packaged, chopped, and optionally encrypted using an industry-standard MXF wrapper. This step can be performed by a digital cinema processor (DCP) 312 or any suitable audio processor depending on the ultimate playback environment, such as a standard surround sound equipped theater 318, an audio-enabled theater 320 adapted, or any other playback environment. As shown in FIG. 3, the processor 312 outputs suitable audio signals 314 and 316 depending on the display environment. In one embodiment, the adapted audio print master includes an adapted audio mix mixed with a standard DCI (DCI-compliant) compliant pulse code modulation (PCM). The PCM mix can be presented by the presentation and master control unit in the dubbing theater, or by a separate mix if desired. The PCM audio forms a standard main audio track file within the digital cinema processor 312, and the adapted audio forms an additional track file. Such track files can conform to existing industry standards and be ignored by DCI-compliant servers that cannot use them. In an exemplary movie playback environment, a DCP containing an adapted audio track file is recognized by the server as a valid package and is absorbed by the server and then flows to an adapted audio movie processor. A system having both linear PCM and adapted audio files available can switch between them when needed. For distribution to the display stage, the adapted audio solution allows the transmission of a single type of package to be sent to the cinema. The DCP package contains both PCM and adapted audio files. The use of security keys, such as a key delivery message (KDM), can be combined to enable the secure delivery of movie content, or other similar content. As shown in FIG. 3, the adapted audio methodology is implemented through the audio workstation 304 by enabling the sound engineer to express his or her intentions regarding the presentation and audio content playback. By controlling certain input controls, the engineer can specify how and where audio objects and sound components are played back depending on the listening environment. In response to the engineer's mixing input 302, metadata is generated at the audio workstation 304 to provide a presentation queue that controls spatial parameters (such as position, velocity, intensity, timbre, etc.) and which specific speakers or speaker groups in the listening environment play individual sounds during the presentation. The metadata is associated with the individual audio data at the workstation 304 or RMU 306 for packaging and transmission by the DCP 312. The graphical user interface and software tools that provide control of the workstation 304 by the engineer include at least part of the authoring tool 106 of FIG. 1. Hybrid audio codec As shown in FIG. 1, system 100 includes a hybrid audio codec 108. This component includes an audio encoding, distribution, and decoding system configured to generate a single bitstream containing both conventional channel-based audio components and audio object decoding components. The hybrid audio decoding system is built around the channel-based encoding system, which is configured to generate a single (unified) bitstream that is simultaneously compatible with (i.e., can be decoded by) a first decoder configured to decode audio data encoded according to a first encoding protocol (channel-based) and one or more secondary decoders configured to decode audio data encoded according to one or more secondary encoding protocols (object-based). The bitstream can include encoded data (in data bursts) that can be decoded by the first decoder (and ignored by any secondary decoder) and encoded data (e.g., bursts of other data) that can be decoded by one or more secondary decoders (and ignored by the first decoder). The decoded audio and associated information (metadata) from the first and one or more secondary decoders can then be combined in a way that presents both channel-based and object-based information simultaneously to recreate an environment, channels, spatial information, and a rendition of the objects (i.e., within a 3D space or listening environment) in the hybrid decoding system. Codec 108 generates a bitstream that contains decoded audio information and information related to multiple sets of channel positions (speakers). In one embodiment, one set of channel positions is fixed and used for the channel-based encoding protocol, while another set of channel positions is adaptable and used for the audio object-based encoding protocol, such that the channel configuration for an audio object can change as a function of time (depending on where the object is placed in the sound field). Thereby, the hybrid audio decoding system can carry information about two sets of speaker locations for reproduction, where one set is fixed and a subset of the other. Devices that support traditional decoded audio information will decode and present the audio information from the fixed subset, while devices that can support a larger set of devices can decode and present additional decoded audio information that is time-varying and assigned from the larger set to different speakers. Additionally, the system is not dependent on the first and one or more secondary decoders being present simultaneously within the system and / or device. Thus, traditional and / or existing devices / systems that include only a decoder that supports the first protocol will produce a fully compatible sound field for presentation via a traditional channel-based reproduction system. In this case, portions of the hybrid bitstream protocol that are unknown or not supported (i.e., the audio information represented by the secondary encoding protocol) will be ignored by the decoder of the system or device that supports the first hybrid encoding protocol. In another embodiment, the codec 108 is configured to operate in a mode where the first encoding subsystem (supporting the first protocol) includes a combined representation of all the audio field information (channels and objects) that represents both the first and one or more secondary encoders present in the hybrid encoder in the subsystem. This ensures backward compatibility of the hybrid bitstream with decoders that only support the first protocol by allowing audio objects (commonly carried in one or more secondary encoder protocols) to be represented and presented within a decoder that only supports the first protocol. In another embodiment, the codec 108 includes two or more encoding subsystems, where each of these subsystems is configured to encode audio data according to different protocols and is configured to combine the outputs of the subsystems to generate a hybrid format (unified) bitstream. In one of the benefits, embodiments are for the ability to hybrid decode audio bitstreams to be carried in a wide range of content distribution systems, where each of the distribution systems conventionally only supports data encoded according to the first encoding protocol. This eliminates the need for any system and / or transport level protocol modifications / changes for specific support of hybrid decoding systems. Audio coding systems generally utilize standardized bitstream elements to enable additional (optional) data transmission within the bitstream itself. The additional (optional) data is generally skipped (i.e., ignored) during the decoding of the audio encoded in the bitstream, but can be used for non-decoding purposes. Different audio decoding standards express these additional data fields using unique nomenclature. This general type of bitstream element can include (but is not limited to) auxiliary data, ignored fields, data stream elements, padding elements, supplementary data, and substream elements. Unless otherwise noted, the use of the expression "auxiliary data" in this document does not imply a specific type or format of additional data, but is preferably interpreted as a general expression that encompasses any or all of the examples associated with the present invention. Data channels enabled via the "auxiliary" bitstream element of the first encoding protocol within the combined hybrid decoding system bitstream can carry one or more secondary (independent or dependent) audio bitstreams (encoded according to one or more secondary encoding protocols). The one or more secondary audio bitstreams can be divided into N sampling blocks and multiplexed into the "auxiliary data" field of the first bitstream. The first bitstream is decodable by a suitable (complementary) decoder. Additionally, the auxiliary data of the first bitstream can be extracted, recombined into one or more secondary audio bitstreams, decoded by a processor supporting the syntax of the one or more secondary bitstreams, and then combined and presented together or independently. Furthermore, it is also possible to reverse the roles of the first and second bitstreams, such that the data blocks of the first bitstream are multiplexed into the auxiliary data of the second bitstream. The bitstream components associated with the sub-encoding protocol also carry and convey information (metadata) characteristics of the underlying audio, which may include (but is not limited to) the desired sound source location, velocity, and size. This metadata is utilized during the decoding and presentation processing to re-establish the appropriate (i.e., original) position for the audio objects associated within the applicable bitstream. It is also possible to carry the above metadata within the bitstream components associated with the first encoding protocol, which can be applied to the audio objects contained in one or more sub-bitstreams that occur in the hybrid stream. The bitstream components associated with one or both of the first and second encoding protocols of the hybrid decoding system carry / convey context metadata that identifies spatial parameters (i.e., the nature of the signal attributes itself) and further describes the information of the underlying audio nature type in a specific audio category form, which is carried within the hybrid decoded audio bitstream. Such metadata can indicate, for example, the performance of a spoken dialogue, music, a dialogue in music, applause, singing voices, etc., and can be utilized to adaptively modify the interconnected behavior upstream or downstream of the pre- or post-processing modules of the hybrid decoding system. In one embodiment, the codec 108 is configured to operate with a shared or common bit pool, where the bits available for decoding are "shared" among all or some of the encoding subsystems that support one or more protocols. To optimize the overall audio quality of the unified bitstream, such a codec can distribute the available bits (from the common "shared" bit pool) among the encoding subsystems. For example, during a first time interval, the codec can assign more available bits to the first encoding subsystem and fewer available bits to the remaining subsystems, while during a second time interval, the codec can assign fewer available bits to the first encoding subsystem and more available bits to the remaining subsystems. For example, the decision of how to assign bits among the encoding subsystems can depend on the results of a statistical analysis of the shared bit pool and / or the analysis of the audio content encoded by each subsystem. The codec can then distribute the bits from the shared pool in such a way that it maintains a fixed frame length / bit rate for the unified bitstream over a specific time interval by multiplexing the outputs of the encoding subsystems. In some cases, the frame length / bit rate for the unified bitstream may also vary over a specific time interval. In an alternative embodiment, the codec 108 generates a unified bitstream that includes data encoded according to a first encoding protocol that is configured and transmitted as an independent substream of an encoded data stream (which will be decoded by a decoder that supports the first encoding protocol), and data encoded according to a second protocol that is transmitted as a substream of an independent or dependent encoded data stream (which will be ignored by a decoder that supports the first protocol). More generally, in a class of embodiments, the codec generates a unified bitstream that includes two or more independent or dependent substreams (where each substream includes data encoded according to a different or identical encoding protocol). In yet another alternative embodiment, the codec 108 generates a unified bitstream that includes data encoded according to a first encoding protocol that is configured and transmitted with a unified bitstream identifier (which will be decoded by a decoder that supports the first encoding protocol associated with the unified bitstream identifier), and data encoded according to a second protocol that is configured and transmitted with the unified bitstream identifier, which will be ignored by a decoder that supports the first protocol. More generally, in a class of embodiments, the codec generates a unified bitstream that includes two or more substreams (where each substream includes data encoded according to a different or identical encoding protocol and where each carries a unified bitstream identifier). Methods and systems for creating the above unified substreams provide the ability to explicitly signal (to the decoder) which interleaving and / or protocols have been utilized within the hybrid bitstream (e.g., whether AUX data, SKIP, DSE, or substreaming methods as described in the present invention have been utilized). The hybrid decoding system is configured to support de-interleaving / de-multiplexing and re-interleaving / re-multiplexing of bitstreams that support one or more secondary protocols to a first bitstream (supporting the first protocol) at any processing point found throughout the media delivery system. The hybrid codec is also configured to be able to encode an audio input stream at different sampling rates for a single bitstream. This provides a means for efficiently decoding and distributing audio sources that contain signals with substantially different bandwidths. For example, dialogue tracks generally have a substantially lower bandwidth than music and sound effect tracks. Presentation In one embodiment, an adaptive audio system allows multiple (e.g., up to 128) tracks to be encapsulated, typically as a combination of background music and objects. The basic format of the audio data for the adaptive audio system includes a number of independent mono audio streams. Each stream has associated metadata specifying whether the stream is a channel-based stream or an object-based stream. Channel-based streams have presentation information encoded by channel names or labels; and object-based streams have fixed-point information encoded by a mathematical formula encoded in further associated metadata. The original independent audio streams are then encapsulated into a single serial bitstream that contains all the audio data in an ordered pattern. The adaptive data arrangement allows sounds to be presented according to a reference non-egocentric frame, where the ultimate presentation fixed-point of the sound is based on the reproduction environment to correspond to the intentions of the sound engineer. Thereby, a sound can be specified with a reference frame originating from within the reproduction room (e.g., the middle of the left wall), rather than a specific-labeled speaker or speaker group (e.g., left surround). The object position metadata includes a non-egocentric frame of appropriate reference information that is required to correctly play the sound using the speaker positions available in the room set up to play the adaptive audio content. The renderer employs a bitstream encoded audio track and processes the content according to the signal type. The background music is fed into an array, which potentially requires different delay and quantization processing than individual objects. This processing supports presenting this background music and objects to multiple (up to 64) speaker outputs. FIG. 4 is a block diagram of the presentation stage of an adaptive audio system in one embodiment. As shown in system 400 of FIG. 4, some input signals, such as up to 128 audio tracks that contain adaptive audio signals 402, are provided by some components (such as RMU 306 and processor 312) of the establishment, authoring, and encapsulation stages of system 300. These signals include channel-based background music and objects utilized by the renderer 404. The channel-based audio (background music) and objects are input to the level manager 406 which provides control over the output levels or amplitudes of different audio components. Certain audio components may be processed by the array correction component 408. The adaptive audio signals then pass through the B-chain processing component 410, which generates some (e.g., up to 64) speaker feed output signals. Generally, the B-chain feed refers to the signals processed by the power amplifiers, switches, and speakers, as the content relative to the A-chain that makes up the sound track on the film. In one embodiment, the renderer 404 runs a rendering algorithm that intelligently uses the surround speakers in a theater to their best ability. By improving the power handling and frequency response of the surround speakers, and maintaining the same monitoring reference level for each output channel or speaker in the theater, objects panning between the screen and the surround speakers can maintain their sound pressure level and importantly have a closer timbre match without increasing the overall sound pressure level in the theater. A properly specified array of surround speakers will typically have sufficient headroom to reproduce the maximum dynamic range available within a surround 7.1 or 5.1 band (i.e., a reference level of 20 dB or more), however unlike a single surround speaker will not have as much headroom as a multi-channel screen speaker. As a result, there are instances such as when an object placed in the sound field will require a greater sound pressure than can be achieved using a single surround speaker. In these cases, the renderer will spread the sound across a suitable number of speakers to achieve the required sound pressure level. The adaptive audio system improves the quality and power handling of the surround speakers to provide an improvement in faithful rendering. Support for bass management of the surround speakers is provided through the use of an optional rear subwoofer, which allows each surround speaker to achieve improved power handling and at the same time potentially utilize a smaller speaker enclosure. It also allows the addition of side surround speakers closer to the screen than current implementations to ensure that objects can smoothly transfer from the screen to the surroundings. As some rendering processes use metadata for the定点 information of specific audio objects, the system 400 provides a comprehensive, flexible approach for content creators to move beyond the limitations of existing systems. As previously mentioned, current systems establish and distribute audio fixed to specific speaker positions (parts of the playback audio) with limited knowledge of the type of content being delivered in the audio essence. The adaptive audio system 100 provides new mixing methods that include options for both speaker-position specific audio (left channel, right channel, etc.) and object-oriented audio components, the object-oriented audio components having generalized spatial information that can include (but is not limited to) position, size, and velocity. The mixing methods provide fidelity (provided by fixed speaker positions) and flexibility in the rendering (generalized audio objects). The system also provides additional useful information regarding the audio content paired with the audio essence by the content creator at the time of content creation. This information can be used during rendering in very powerful ways to provide powerful, detailed information on the attributes of the audio. Such attributes can include (but are not limited to) content type (dialogue, music, sound effects, Foley, background / environment, etc.), spatial attributes (3D position, 3D size, velocity), and rendering information (locked to speaker positions, channel weights, gain, bass management information, etc.). It should be noted that there is an unclear expression "定点信息" in the original Chinese text, which may need to be further clarified in the actual context for more accurate translation. The above translation tries to make the best guess based on the overall meaning.The adaptive audio system described herein provides powerful information that can be used for rendering by a wide variety of different numbers of endpoints. The situation of many optimal rendering technology applications depends very much on the endpoint device. For example, home theater systems and soundbars can have 2, 3, 5, 7 or even 9 separate speakers. Many other types of systems, such as TVs, computers and music docks, only have two speakers and almost all commonly used devices have stereo headphone outputs (PCs, laptops, tablets, mobile phones, music players, etc.). However, for currently distributed traditional audio (mono, stereo, 5.1, 7.1 channels), endpoint devices usually need to make overly simplistic decisions and compromise on the rendering and playback of audio in the form of channels / speakers that are currently distributed. In addition, little or no information is transmitted about the actual content being distributed (dialogue, music, environment, etc.) and little or no information about the content of the creator's intention for audio playback. However, the adaptive audio system 100 provides this information and potentially accesses audio objects, which can be used to create a compelling next-generation user experience. Through unique and powerful metadata and an adaptive audio transmission format, system 100 allows content creators to embed the spatial intent of the mix within the bitstream of metadata such as location, size, speed, etc. This allows for great flexibility in the spatial reproduction of audio. From a spatial presentation standpoint, the adaptive audio enabled mix is adapted to the exact location of the speakers in a particular room to avoid spatial distortion that occurs when the geometry of the reproduction system is different from the presentation system. In current audio reproduction systems where only the audio for the speaker channels is transmitted, the intent of the content creator is unknown. System 100 uses metadata that is passed throughout the creation and distribution pipeline. The adaptive audio-aware reproduction system can use this metadata information to reproduce the content in a manner that matches the original intent of the content creator. Similarly, the mix can be adapted to the exact hardware configuration of the reproduction system. Currently, there are many different possible speaker configurations and types in presentation setups, such as TVs, home theaters, soundbars, portable music docking stations, etc. When these current systems are transmitted channel-specific audio information (i.e., left and right channel audio or multi-channel audio), the system must process the audio to properly match the capabilities of the presentation setup. An example is standard stereo audio being sent to a soundbar with more than two speakers. In current audio reproduction where only the audio for the speaker channels is transmitted, the intent of the content creator is unknown. Through the use of metadata passed throughout the creation and distribution pipeline, the adaptive audio-aware reproduction system can use this information to reproduce the content in a manner that matches the original intent of the content creator. For example, some soundbars have side firing speakers to create a sense of enclosure. With adaptive audio, spatial information and content type (such as ambient sound effects) can be used by the soundbar to only transmit the appropriate audio to these side firing speakers. The adaptive audio system allows for infinite speaker interpolation in systems in all front / back, left / right, up / down, near / far dimensions. In current audio reproduction systems, there is no information on how to process audio where it may be desired to localize the audio so that it is perceived by the listener between two speakers. Currently, with audio that is only assigned to specific speakers, a spatial quantization factor is introduced. With adaptive audio, the spatial location of the audio can be correctly known and reproduced accordingly on the audio reproduction system. Regarding the rendering of headphones, the creator's intention is achieved by matching the head-related transfer functions (HRTFs) for spatial localization. When audio is reproduced on headphones, spatial virtualization can be achieved by the application of HRTFs, which process the audio and add perceptual cues that create the perception of audio playing in a 3D space rather than on the headphones. The correctness of spatial reproduction depends on the selection of appropriate HRTFs (which can vary based on several factors including spatial location). An adaptive audio system provides the ability to use spatial information to enable the selection of one or a continuously varying number of HRTFs to greatly improve the reproduction experience. The ability of an adaptive audio system to convey spatial information can be used not only by content creators to create compelling entertainment experiences (movies, TV, music, etc.), but spatial information can also indicate to the listener where they are located relative to physical objects such as buildings or relevant geographical points. This would allow users to interact with a virtualized audio experience that is related to the real world, i.e., augmented reality. When only object audio data is not available, spatial upmixing is also enabled by the example by performing enhanced upmixing by reading metadata. Knowing the positions of all objects and their types allows the upmixer to better distinguish components within the channel base track. Existing upmixing algorithms have to infer information such as audio content type (speech, music, environmental sound effects) and the positions of different components within the audio stream to create high-quality upmixing with minimal or no audible artifacts. Many times the inferred information can be incorrect or inappropriate. With adaptive audio, additional information obtained from metadata about, for example, audio content type, spatial location, velocity, audio object size, etc., can be used by the upmixing algorithm to create high-quality reproduction results. By correctly positioning the audio objects on the screen to the visual element system, the audio is also spatially matched to the video. In this case, a compelling audio / video reproduction experience is possible, especially for large screen sizes. An example is having the dialogue in a movie or TV show spatially coincide with the person or character speaking on the screen. For normal speaker-channel-based audio, there is no simple way to determine where the dialogue should be spatially located to match the on-screen position of the person or character. With the audio information obtained with adaptive audio, such audio / visual alignment can be achieved. The visual position and audio spatial alignment can also be used for non-character / dialogue objects such as cars, trucks, animations, etc. Since the knowledge of the spatial intent of the mix by adapting the audio metadata means that the mix can adapt to any speaker configuration, spatial masking processing is facilitated by system 100. However, due to reproduction system limitations, anyone at the same or nearly the same fixed point risks downmixing an object. For example, if there is no surround channel, an object panned to the left rear may be downmixed to the left front, but if a louder component occurs simultaneously in the left front, the downmixed object will be masked and disappear from the mix. Using adaptive audio metadata, spatial masking is anticipated by the renderer, and the spatial and / or loudness downmixing parameters for each object can be adjusted so that all mixed audio objects remain perceptible as in the original mix. Because the renderer understands the spatial relationship between the mix and the reproduction system, it has the ability to "lock" an object to the nearest speaker instead of a phantom image between two or more speakers. While this method slightly distorts the spatial representation of the mix, it also allows the renderer to avoid accidental phantom images. For example, if the angular position of the left speaker in the mixing stage does not match the angular position of the left speaker in the reproduction system, using the function of locking to the nearest speaker can prevent the reproduction system from reproducing an invariant phantom image of the left channel in the mixing stage. Regarding content processing, the adaptive audio system 100 allows content creators to create individual audio objects and add information about the content that can be delivered to the playback system. This allows for a great deal of flexibility in audio processing prior to playback. From the viewpoints of content processing and presentation, the adaptive audio system enables processing to adapt to object types. For example, dialogue enhancement can only be applied to dialogue objects. Dialogue enhancement refers to methods of processing audio that includes dialogue so that the audibility and / or intelligibility of the dialogue is increased and / or improved. In many cases, the audio processing applied to dialogue is not suitable for non-dialogue audio content (i.e., music, environmental sound effects, etc.) and can result in undesirable audible artifacts. With adaptive audio, an audio object can contain only the dialogue in a segment of content, and it can be tagged accordingly so that the presentation solution can selectively apply dialogue enhancement only to the dialogue content. Additionally, if the audio object is only dialogue (and typically not a mix of dialogue and other content), then the dialogue enhancement processing can specifically process the dialogue (thereby restricting any processing from being applied to any other content). Similarly, bass management (filtering, attenuation, gain) can be targeted in specific objects based on their type. Bass management refers to selectively isolating and only processing the bass (or lower) frequencies in the content of a particular segment. This is a "blind" process with current audio systems and delivery mechanisms that is applied to all audio. With adaptive audio, the specific audio objects for which bass management is appropriate can be identified by metadata, and the presentation processing can be appropriately applied. The adaptive audio system 100 also provides upmixing for object-based dynamic range compression and selection. Traditional audio tracks have the same duration as the content itself, while audio objects can occur only during a limited number of times in the content. Metadata associated with the object can include information about its average and peak signal amplitudes and its onset or attack time (especially for transient material). This information will allow the compressor to better adapt its compression and time constants (attack, release, etc.) to better fit the content. For selected upmixing, the content creator can choose to indicate in the adaptive audio bitstream whether an object should be upmixed. This information allows the adaptive audio renderer and upmixer to distinguish which audio objects can be safely upmixed while respecting the creator's intent. The embodiments also allow the adaptive audio system to select a better rendering algorithm from some available rendering algorithms and / or surround sound formats. Examples of available rendering algorithms include: stereo, stereo bipolar, high-fidelity stereo reproduction, wave field synthesis (WFS; Wave Field Synthesis), multichannel panning, raw stems with position metadata. Others include dual balance and vector-based amplitude panning. The stereo distribution format uses a two-channel representation of the sound field in terms of the left and right ear signals. Stereo information can be recorded via in-ear recording or synthesized using the HRTF model. Reproduction of the stereo representation is generally done on headphones or by using crosstalk cancellation. Reproduction on any speaker setting would require signal analysis to determine the associated sound field and / or signal source. The stereo bipolar rendering method is a transaural cross-talk cancellation process to reproduce stereo signals on stereo speakers (e.g., + and -10 degrees off center). High-fidelity stereo reproduction is a (distribution format and rendering method) that is encoded in the form of four channels called B-format. The first channel W is an omnidirectional pressure signal; the second channel X is a directional pressure gradient containing front and rear information; the third channel Y contains left and right and Z contains up and down. These channels define the first-order sampling of the complete sound field at a point. High-fidelity stereo reproduction uses all available speakers to reconstruct the sampled (or synthesized) sound field within the speaker array so that when some speakers are pushing, others are pulling. Wave field synthesis is a sound reproduction rendering method that is based on the precise construction of the desired sound wave field by secondary sources. WFS is based on Huygens' principle and is implemented with an array of speakers (tens or hundreds) that surround the listening space and operate in a coordinated, phased manner to reconstruct each individual sound wave. Multi-channel panning is a distribution format and / or a rendering method, and can refer to, for example, channel-based audio. In this case, sounds are represented as a number of discrete sources to be reproduced through an equal number of speakers from a defined listener perspective. Content creators / mixers can create virtual images by panning signals between adjacent channels to provide direction cues; early reflections, reverberation, etc. can be mixed into multiple channels to provide direction and environmental cues. Original mix elements with position metadata are in a distribution format and can also refer to, for example, object-based audio. In this format, the position and environmental metadata represent distinct, "close mic’ed" sound sources. Virtual sources are presented based on the metadata and the reproduction equipment and listening environment. The adaptive audio format is a hybrid of the multi-channel panning format and the original mix element format. The rendering method in the present embodiment is multi-channel panning. For audio channels, rendering (panning) occurs at creation time, while for object rendering (panning) occurs at reproduction. Metadata and adaptive audio transport formats As described above, metadata is generated during the build phase to encode certain position information for audio objects and to accompany the audio program to assist in rendering the audio program, and in particular, to describe the audio program in a way that enables the rendering of the audio program on a variety of reproduction equipment and reproduction environments. Metadata is generated for a given program and for editors and mixers who create, collect, edit, and manipulate audio during post-production. An important feature of the adaptive audio format is the ability to control how audio is translated to the reproduction system and an environment different from the mix. In particular, a given cinema may have lower capabilities than the mix environment. Adaptive audio renderers are designed so that the best use of the equipment can be obtained to recreate the mixer's intent. In addition, adaptive audio creation tools allow the mixer to preview and adjust how the mix will be presented in various reproduction configurations. All metadata values can be adjusted in the reproduction environment and speaker configuration. For example, different mix levels for a given audio component can be specified based on the reproduction configuration or mode. In one embodiment, the list of adjustable reproduction modes is extensible and includes the following: (1) channel-based only reproduction: 5.1, 7.1, 7.1 (height), 9.1; and (2) distributed speaker reproduction: 3D, 2D (no height). In one embodiment, metadata controls or specifies different aspects of the adaptive audio content and organizes it based on different types, which includes: program metadata, audio metadata, and presentation metadata (for channels and objects). Each type of metadata includes one or more metadata items that provide values for features referenced by an identifier (ID). FIG. 5 lists the metadata types and associated metadata components for an adaptive audio system in one embodiment. As shown in Table 500 of FIG. 5, the first type of metadata is program metadata, which includes metadata components such as a specific frame rate, track count, extendable channel description, and mixing stage description. The frame rate metadata component specifies the audio content frame rate in units of frames per second (fps). Since the audio is provided as a full track (the duration of a reel or overall feature) rather than an audio segment (the duration of an object), the original audio format does not need to include frames for audio or metadata. The original format really needs to carry all the information required to enable the adaptive audio encoder to frame the audio and metadata, including the actual frame rate. Table 1 shows the ID, example values, and descriptions of the frame rate metadata component. The track count metadata component indicates the number of audio tracks in a frame. Example adaptive audio decoders / processors can support up to 128 synchronous audio tracks, while the adaptive audio format will support any number of tracks. Table 2 shows the ID, example values, and descriptions of the track count metadata component. Channel-based audio can be assigned to non-standard channels, and the extendable channel description metadata component enables mixing to use new channel positions. For each extended channel, the following metadata should be provided as shown in Table 3: The mixing stage description metadata component specifies the frequency at which half of the passband power is generated at a particular speaker. Table 4 shows the ID, example values, and descriptions of the mixing stage description metadata, where LF = low frequency; HF = high frequency; 3dB point = the edge of the speaker passband. As shown in FIG. 5, the second type of metadata is audio metadata. Each channel-based or object-based audio component consists of audio essence and metadata. The audio essence is a mono audio stream carried on one of a number of audio tracks. The associated metadata describes how the audio essence is stored (audio metadata, such as sampling rate) or how it should be presented (presentation metadata, such as the desired audio source location). Generally, the audio track persists throughout the duration of the audio program. The program editor or mixer is responsible for assigning audio components to tracks. The use of tracks is expected to be sparse, i.e., the median usage of synchronous tracks can be as low as 16 to 32. In a typical implementation, audio will be transmitted using an efficient lossless encoder. However, alternative implementations are possible, such as transmitting unencoded audio data or distorted decoded audio data. In a typical implementation, the format consists of up to 128 audio tracks, each with a single sampling rate and a single decoding system. Each track maintains a characteristic duration (without explicit reel support). The mapping of objects to tracks (time multiplexing) is the responsibility of the content creator (mixer). As shown in FIG. 3, the audio metadata includes a sampling rate component, bit depth, and decoding system. Table 5 shows the ID, example values, and descriptions of the sampling rate metadata components. Table 6 shows the ID, example values, and bit depth metadata components (for PCM and lossless compression). Table 7 shows the ID, example values, and descriptions of the decoding system metadata components. As shown in FIG. 5, the third type of metadata is presentation metadata. Regardless of the reproduction environment, the presented metadata specifically helps the renderer to match as closely as possible the values intended by the original mixer. This set of metadata components is different from channel-based audio and object-based audio. As shown in Table 8, the first presentation metadata field selects between two types, audio ~ channel-based or object-based. The presentation metadata for channel-based audio contains a position metadata component that specifies the audio source location as one or more speaker locations. Table 9 shows the ID and values of the position metadata components for the channel-based case. As shown in Table 10, the presentation metadata for channel-based audio contains presentation control components that specify certain characteristics associated with the reproduction of channel-based audio. For object-based audio, the metadata includes analog components for channel-based audio. Table 11 provides the IDs and values of the components for object location metadata. The object location is described in one of three ways: three-dimensional coordinates; planar and two-dimensional coordinates; or line and one-dimensional coordinates. The rendering method can adapt based on the type of location information. Table 12 shows the IDs and values of the components for object rendering control metadata. These values provide additional means to control or optimize the rendering for object-based audio. In one embodiment, the metadata described above and illustrated in FIG. 5 is generated and stored as one or more files that are associated with or indexed to the corresponding audio content such that the audio stream is processed by the metadata generated by the mixer through an adaptive audio system translation. It should be noted that the above metadata is a set of exemplary IDs, values, and definitions and other or additional metadata components may be included for use within the adaptive audio system. In one embodiment, two (or more) sets of metadata components are associated with each channel and object-based audio stream. The first set of metadata is applied to a plurality of audio streams for a first condition of the reproduction environment, and the second set of metadata is applied to a plurality of audio streams for a second condition of the reproduction environment. Based on the conditions of the reproduction environment, the second or subsequent set of metadata components replaces the first set of metadata components for a given audio stream. The conditions include factors such as room dimensions, shape, material composition within the room, current occupancy and density of people within the room, ambient noise characteristics, ambient light characteristics, and any other factors that may affect sound or even the atmosphere of the reproduction environment. Post-production and mastering The rendering stage 110 of the adaptive audio processing system 100 may include audio post-production steps that result in the establishment of a final mix. In a film application, the three main sound categories used in film mixing are dialogue, music, and sound effects. Sound effects consist of sounds that are not dialogue or music (e.g., ambient noise, background / scene noise). Sound effects can be recorded or synthesized by sound designers or they can originate from a sound effects library. A subgroup of sound effects that includes specific noise sources (e.g., footsteps, doors, etc.) is known as production sounds and is performed by production actors. Different types of sounds are marked and panned by recording engineers accordingly. Figure 6 illustrates an example workflow for post-production in an adaptive audio system. As shown in system 600, all of the individual sound components of music, dialogue, action sounds, and special effects in the final mix 606 are brought together into the dubbing theater, and the rerecording mixer 604 uses premixes (also known as "mix minus") to create mix elements, such as dialogue, music, special effects, action sounds, and background sounds, along with the individual sound objects and location information. In addition to forming the final mix 606, the music and all of the mix elements can be used as the basis for creating the language versions of the movie for dubbing. Each mix element consists of a channel base and some audio objects with metadata. The mix elements are combined to form the final mix. Using object panning information from both the audio workstation and the mixing console, the presentation and master unit 608 presents the audio to speaker positions in the dubbing theater. This presentation allows the mixer to hear how the channel base music and audio objects are combined and also provides the ability to present to different configurations. The mixer can use conditional metadata, which is keyed to the relevant profiles, to control how the content is presented to the surround channels. In this way, the mixer retains full control over how the movie is reproduced in all expandable environments. The monitoring step can include allowing the mixer to hear and evaluate the immediate content generated in one or both of the rerecording step 604 and the final mix step 606. During the mastering session, the mix elements, objects, and metadata are brought to the adaptive audio package 614, which is generated by the print host 610. This package also contains a backward-compatible (legacy 5.1 or 7.1) surround sound theater mix 612. The presentation / master unit (RMU) 608 can present these outputs (if desired); thereby eliminating any additional workflow steps for generating the existing channel base deliverables. In one embodiment, the audio files are packaged using the standard material exchange format (MXF; Material Exchange Format). The adaptive audio mix master file can also be used to generate other deliverables, such as consumer multichannel or stereo mixes. The intelligent profiles and conditional metadata allow for controlled presentation, which can significantly reduce the time required to create such mixes. In one embodiment, the packaging system can be used to create digital movie packages for deliverables that include an adaptive audio mix. The audio track files can be locked together to help prevent synchronization errors associated with the adaptive audio track files. During packaging, certain areas require the addition of track files, such as hearing impaired (HI) or visually impaired narration (VI-N) tracks to the main audio track file. In one embodiment, a speaker array in a reproduction environment may include any number of surround sound speakers placed and designated according to established surround sound standards. Any number of additional speakers for the correct rendering of object-based audio content may also be placed based on the conditions of the reproduction environment. These additional speakers may be installed by a sound engineer, and this installation is provided to the system in the form of an installation file that is used by the system's object-based components for rendering adaptive audio to specific speakers or within the overall speaker array. This installation file includes at least a list of speaker designations and mappings to individual speaker channels, information about speaker grouping, and a runtime mapping based on the relative position of the speakers to the reproduction environment. The runtime mapping is utilized by the system's locking function that presents the source object-based audio content to a specific speaker that is closest to the perceived point of sound as intended by the sound engineer. Figure 7 is an exemplary flowchart of a digital movie packaging process for using an adaptive audio file in one embodiment. As shown in workflow 700, an audio file containing both an adaptive audio file and a 5.1 or 7.1 surround sound audio file is input to the packaging / encryption block 704. In one embodiment, for the creation of a digital movie package in block 706, a PCM MXF file (with appropriate additional tracks appended) is encrypted according to current practice using SMPTE specifications. The adaptive audio MXF package is an auxiliary track file and may optionally be encrypted using a symmetric content key in each SMPTE specification. This single DCP 708 can then be transmitted to any Digital Cinema Initiatives (DCI) compliant server. In general, any improperly equipped installation will simply ignore the additional track file containing the adaptive audio track and will use the existing main audio track file for standard reproduction. An installation equipped with an appropriate adaptive audio processor will be able to absorb and replay the adaptive audio track where applicable and revert to the standard audio track when needed. The packaging / encryption component 704 may also directly provide input to the key distribution for KDM block 710 that generates the appropriate security keys for use in the digital movie server. Other movie components or files, such as subtitles 714 and video 716, may be packaged and encrypted along with the audio file 702. In this case, certain processing steps may be included, such as compression 712 in the case of the video file 716. Regarding content management, the adaptive audio system 100 allows content creators to create individual audio objects and add content that can be delivered to the playback system. This allows for a great deal of flexibility in the content management of audio. From the perspective of content management, the adaptive audio method enables several different features. These include changing the content language by replacing dialogue objects for reasons such as space savings, download efficiency, geographical reproduction adaptation, etc. Movies, TV shows, and other entertainment programs are generally distributed internationally. This usually requires changing the language on content segments depending on where they are to be played (French for a movie shown in France, German for a TV show shown in Germany, etc.). Currently, this usually requires completely separate audio tracks to be created, packaged, and distributed. With adaptive audio and its inherent audio object concept, the dialogue for a content segment can be an independent audio object. This allows the language of the content to be simply changed without updating or changing the components of other audio tracks (such as music, sound effects, etc.). This applies not only to foreign languages but also to inappropriate language, targeted advertising, etc. for certain audiences (e.g., children's TV shows, in-flight movies, etc.). Installation and Equipment Considerations The adaptive audio file format and associated processing allow for changes in how a theater is equipped, calibrated, and maintained. With the introduction of many potential speaker outputs, each individual equalization and balancing requires intelligent and timely automatic room equalization, which can be implemented through the ability to manually adjust any automatic room equalization. In one embodiment, the adaptive audio system uses an optimal 1 / 12 octave band equalization engine. Up to 64 outputs can be processed to more accurately balance the sound in the theater. The system also allows for scheduled monitoring of individual speaker outputs, from the movie processor output all the way to the sound reproduced in the auditorium. Local or network alerts can be established to ensure appropriate action is taken. The flexible presentation system can automatically remove damaged speakers or amplifiers from the playback chain and nearby, so allowing the performance to continue. The movie processor can be connected to a digital movie server with an existing 8xAES main audio connection, as well as an Ethernet for streaming adaptive audio data. Surround reproduction of 7.1 or 5.1 content uses an existing PCM connection. The adaptive audio data is streamed over Ethernet to the movie processor for decoding and presentation, and the communication between the server and the movie processor allows for the identification and synchronization of audio. In any event of an adaptive audio track reproduction problem, the sound reverts to Dolby Surround 7.1 or 5.1 PCM audio. Although embodiments have been shown in the context of 5.1 and 7.1 surround sound systems, it should be noted that other current and future surround configurations can be used to link to embodiments including 9.1, 11.1, and 13.1 and above. Adaptive audio system design allows both content creators and presenters to determine how audio content is presented in different reproduction speaker configurations. The ideal number of speaker channels to use will vary depending on the size of the room. The recommended speaker placement thus depends on many factors, such as size, composition, seating configuration, environment, average audience size, etc. The examples or representative speaker configurations and layouts provided herein are for illustrative purposes only and are not intended to limit the scope of any claimed embodiments. The recommended layout for speakers in an adaptive audio system remains compatible with existing cinema systems, which is extremely important so as not to compromise the reproduction of existing 5.1 and 7.1 channel base formats. In order to preserve the intent of adaptive audio sound engineers, as well as the intent of 7.1 and 5.1 content mixers, the positions of the existing screen channels should not be changed too radically by introducing new speaker anchor points that enhance or emphasize them. Instead of using all available 64 output channels, the adaptive audio format can be correctly presented in a cinema for speaker configurations such as 7.1, thus allowing the use of the format (and associated benefits) in existing theaters without changing the amplifier or speaker configuration. Depending on the theater design, different speaker anchor points can have different utilities, and thus there is currently no industry-specific ideal number or placement of channels. Adaptive audio is prepared to truly adapt and be able to play back correctly in various auditoriums, whether they have a limited number of reproduction channels or many channels with a flexible configuration. Figure 8 is a top view 800 of a recommended layout of speaker anchor points for an adaptive audio system in a typical auditorium, and Figure 9 is a front view 900 of a recommended layout of speaker anchor points for a screen in the auditorium. The reference position referred to herein corresponds to the position two-thirds of the distance from the screen to the back wall, on the center line of the screen. The standard screen speaker 801 is shown in its usual position relative to the screen. Studies on enhancing perception in the screen plane have shown that additional speakers 804 behind the screen, such as left center (Lc) and right center (Rc) screen speakers (in the anchor points of the left and right additional channels in the 70mm film format), can be beneficial in creating a smooth pan across the screen. Such optional speakers are thus recommended especially in auditoriums with wide screens greater than 12m (40ft.). All screen speakers should be angled so that they are aligned towards the reference position. Regarding the middle of the room, the recommended placement of the subwoofer 810 behind the screen should remain unchanged, including maintaining an asymmetric speaker placement to prevent the excitation of standing waves. Additional subwoofers 816 can be placed at the back of the theater. The surround speakers 802 should be individually wired to the amplifier rack and individually amplified with dedicated channels that may have power amplification to match the power reserve of the speakers according to the manual specifications. Ideally, specific surround speakers should be used to handle the increased SPL for each individual speaker and also have a relatively wide frequency response if possible. As a rule of thumb for an average-sized theater, the spacing of the surround speakers should be between 2 and 3 m (6'6" and 9'9"), with the left and right surround speakers placed symmetrically. However, the spacing of the surround speakers is most effectively considered in terms of the angle towards a given listener from the adjacent speakers. For optimal reproduction throughout the auditorium, referring to each of the four corners of the main listening area, the angular distance between adjacent speakers should be 30 degrees or less. Good results can be achieved with an interval up to 50 degrees. For each surround zone, the speakers adjacent to the seating area should maintain an equal linear spacing where possible. The linear spacing outside the listening area, such as between the front row and the screen, can be slightly larger. Figure 11 shows an example of the positioning of the upper surround speaker 808 and the side surround speaker 806 relative to the reference position in one embodiment. The additional side surround speakers 806 should be embedded closer to the screen than the currently recommended distance which actually starts at about one-third of the distance from the front to the back of the auditorium. These speakers are not used as side surrounds during the reproduction of Dolby Surround 7.1 or 5.1 soundtracks, but will enable smooth transitions and improved tonal matching when panning an object from the screen speakers to the surround zones. To maximize the spatial impression, the surround array should be placed as low as practical, subject to the following limitations: the vertical placement of the surround speakers in front of the array should be reasonably close to the height of the auditory center of the screen speakers and high enough to maintain good coverage across the seating area according to the directivity of the speakers. The vertical placement of the surround speakers should be such that they form a straight line from the front to the back and (generally) tilted upwards so that as the seating elevation increases, the relative elevation of the surround speakers over the listener is maintained towards the back of the cinema, as shown in Figure 10, which is a side view of a recommended speaker positioning layout example for use with an audio system in a typical auditorium. In practice, this can be most simply achieved by choosing the elevation for the frontmost and rearmost side surround speakers and placing the remaining speakers on the line between these points. To provide the best coverage of each speaker over the seating area, under the guidelines defined regarding spacing, position, angle, etc., the side surrounds 806, the rear speakers 816, and the upper surrounds 808 should be aligned towards the reference position in the theater. Examples of an Adaptive Audio Movie System and Format achieve improved audience immersion and parity with current systems by providing powerful new creative tools to sound mixers, and provide a new movie processor featuring a flexible rendering engine that optimizes audio quality and surround sound for each in-room speaker layout and characteristics. Additionally, the system maintains backward compatibility and minimizes impact on current production and distribution workflows. While the examples have been described with respect to exemplary and implementation in a movie environment where adaptive audio content is associated with movie content for use in a digital movie processing system, it should be noted that the examples may also be implemented in non-theater environments. Adaptive audio content, including object-based audio and channel-based audio, may be used in conjunction with any relevant content (with respect to audio, video, graphics, etc.), or may constitute standalone audio content. The reproduction environment may be any suitable listening environment from headphones or any near-field monitors to small or large rooms, cars, outdoor stages, concert halls, etc. Aspects of system 100 may be implemented in a network environment for suitable computer-based sound processing of digital or digitized audio files. Portions of the adaptive audio system may include one or more networks, the network including any desired number of individual machines, including one or more routers (not shown) that are used to buffer and send data transmitted between computers. Such networks may be built on various different network protocols and may be the Internet, a wide area network (WAN), a local area network (LAN), or any combination thereof. In embodiments where the network includes the Internet, one or more machines may be configured to access the Internet through a web browser program. One or more components, blocks, processes, or other functional components may be implemented through execution of a processor-based computing device of the control system. It should be noted that the various functions disclosed herein may be described in terms of any number of hardware, firmware, and / or data and / or instructions (in terms of their behavior, register transfers, logic components, and / or other characteristics, which are embodied in various machine-readable or computer-readable media). Computer-readable media in which such formatted data and / or instructions may be embodied include (but are not limited to) non-volatile storage media in any form, such as optical, magnetic, or semiconductor storage media. Unless the context clearly requires otherwise, throughout the description and the claims, the words "comprise", "comprising", and the like are to be construed in an inclusive sense as opposed to an exclusive or exhaustive sense; that is to say, in the sense of "including, but not limited to". The use of the singular or plural number is also to include the plural or singular respectively. Additionally, the words "herein", "hereinafter", "above", "below", and words of similar import refer to this application as a whole and not to any particular portion of this application. When the word "or" is used to refer to a list of two or more items, that word covers all of the following interpretations: any of the items in the list, all of the items in the list, and any combination of the items in the list. When one or more implementations have been described by way of example and with respect to specific embodiments, it is to be understood that one or more implementations are not limited to the disclosed embodiments. On the contrary, it is intended to cover various modifications and similar arrangements that would be apparent to those of ordinary skill in the art. Accordingly, the scope of the appended claims should be accorded the broadest interpretation so as to encompass all such modifications and similar arrangements. 300: Workflow 302: Input Mix Control 304: Mixing Console / Audio Workstation 306: Presentation and Mastering Unit 308: Other Presentational Deliverables 310: Adaptive Audio Mixing 312: Digital Movie Processor 314: Audio Signal 316: Audio Signal 318: Theater 320: Adaptive Audio Enabled Theater

Claims

1. A system for processing audio signals, comprising an editing component configured to: receive a plurality of audio signals; generate an adaptive audio mix comprising a plurality of mono audio streams and metadata associated with each of the audio streams, and indicating a reproduction point for each mono audio stream, wherein at least some of the plurality of mono audio streams are identified as channel-based audio, and the remainder of the plurality of mono audio streams are identified as object-based audio, wherein the reproduction point of the channel-based mono audio streams is included in the designation of speakers in a speaker array, and the reproduction point of the object-based mono audio streams is included in a point in three-dimensional space, and wherein each object-based mono... The audio stream is presented in at least one specific speaker of the speaker array; and the plurality of mono audio streams and the metadata in the bit stream are encapsulated for transmission to a presentation system configured to present the plurality of mono audio streams to a plurality of speaker feeds corresponding to speakers in a reproduction environment, wherein the speakers of the speaker array are positioned in a specific location in the reproduction environment, and wherein metadata elements associated with each respective object-based mono audio stream indicate whether the reproduction point of the object-based mono audio stream is defined relative to a reference screen.

2. A system for processing audio signals, comprising a presentation system configured to: receive a bitstream to be adapted for an audio mix encapsulation, the bitstream comprising a plurality of mono audio streams and metadata associated with each of the audio streams and indicating a reproduction point for each mono audio stream, wherein at least some of the plurality of mono audio streams are identified as channel-based audio and the remainder of the plurality of mono audio streams are identified as object-based audio, and wherein the reproduction point of the channel-based mono audio streams is included in the designation of speakers in a speaker array, and the reproduction point of the object-based mono audio streams includes... A fixed point contained in three-dimensional space, and the audio stream of each object-based mono channel therein is presented in at least one specific speaker of the speaker array; and the plurality of mono audio streams are presented to a plurality of speaker feeds corresponding to speakers in the reproduction environment, wherein the speakers of the speaker array are placed at specific locations within the reproduction environment, and wherein metadata elements associated with each respective object-based mono audio stream indicate whether the reproduction fixed point of the object-based mono audio stream is defined relative to a reference screen.

3. The system as described in claim 2, wherein when the reproduction point of the object-based mono audio stream is defined relative to a reference screen, the reproduction point is shifted in response to information about the reference screen and the reproduction screen, and the object-based mono audio stream is presented in response to the shifted reproduction point.

4. A method for presenting an audio signal, comprising: receiving a bitstream to be adapted to an audio mix encapsulation, the bitstream comprising a plurality of mono audio streams and metadata associated with each of the audio streams, and indicating a reproduction point for each of the mono audio streams, wherein at least some of the plurality of mono audio streams are identified as channel-based audio and the remainder of the plurality of mono audio streams are identified as object-based audio, and wherein the reproduction point of the channel-based mono audio streams is included in the designation of a loudspeaker in a loudspeaker array, and the reproduction point of the object-based mono audio streams is... The method includes a fixed point in three-dimensional space, and the audio stream of each object-based mono channel is presented in at least one specific speaker of the speaker array; and the plurality of mono audio streams are presented to a plurality of speaker feeds corresponding to speakers in the reproduction environment, wherein the speakers of the speaker array are placed at specific locations within the reproduction environment, and wherein metadata elements associated with each respective object-based mono audio stream indicate whether the reproduction fixed point of the object-based mono audio stream is defined relative to a reference screen.

5. The method of claim 4, wherein when the reproduction point of the object-based mono audio stream is defined relative to a reference screen, the reproduction point is shifted in response to information about the reference screen and the reproduction screen, and the object-based mono audio stream is presented in response to the shifted reproduction point.

6. A non-transitory computer-readable storage medium comprising a sequence of instructions, wherein when executed by a system for processing audio signals, the sequence of instructions causes the system to perform the method described in claim 4.

Images