OpenAL is the Core Audio implementation of the open-source OpenAL standard for positional audio. It is built on top of the system-supplied 3D Mixer audio unit. All applications can use OpenAL, although it is best suited for games development.

The AVFoundation framework (AVFoundation.framework) provides the AVAudioPlayer class, a streamlined and simple Objective-C interface for audio playback. It also provides the AVAudioEngine class for more sophisticated audio handling.

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Core Audio uses the notion of proxy objects to represent such things as files, streams, audio players, and so on. When you want your application to work with an on-disk audio file, for example, the first step is to instantiate an audio file object of type AudioFileID. This object is declared as an opaque data structure in the AudioFile.h header file:

This sort of pattern is consistent throughout Core Audio, whether you are working with audio files, iPhone audio sessions (described in Audio Sessions: Cooperating with Core Audio), or even hardware devices.

In some cases, a property applies to an audio object as a whole. For example, to enable audio level metering in a playback audio queue object, you set the value of its kAudioQueueProperty_EnableLevelMetering property to true.

Implement the callback function. For example, you might implement the audio queue property listener callback to update the titles and the enabled/disabled state of buttons in your user interface, depending on whether an audio queue object is running or stopped.

In Core Audio, you use two universal data types to represent any audio data format. These types are the data structures AudioStreamBasicDescription (Listing 2-4) and AudioStreamPacketDescription (Listing 2-5), both declared in the CoreAudioTypes.h header file and described in Core Audio Data Types Reference.

In this structure, the mReserved member must always have a value of 0. Other members may have a value of 0 as well. For example, compressed audio formats use a varying number of bits per sample. For these formats, the value of the mBitsPerChannel member is 0.

The constants and data types used as values here are declared in the CoreAudioTypes.h header file and described in Core Audio Data Types Reference. Using the AudioUnitSampleType data type here (and the AudioSampleType data type when handling audio I/O) ensures that an ASBD is platform agnostic.

The previous chapter defined a packet as a collection of one or more frames. It is the smallest meaningful set of frames for a given audio data format. For this reason, it is the best unit of audio data to represent a unit of time in an audio file. Synchronization in Core Audio works by counting packets. You can use packets to calculate useful audio data buffer sizes, as shown in Listing 2-8.

To use VBR or VFR formats in Core Audio, you use the audio stream packet description structure (Listing 2-5). Each such structure describes a single packet in a sound file. To record or play a VBR or VFR sound file, you need an array of these structures, one for each packet in the file.

In CBR and VBR formats (that is, in all commonly used formats), the number of packets per second is fixed for a given audio file or stream. There is a useful implication here: the packetization implies a unit of time for the format. You can use packetization when calculating a practical audio data buffer size for your application. For example, the following method determines the number of packets to read to fill a buffer with a given duration of audio data:

To convert audio data from one format to another, you use an audio converter. You can make simple conversions such as changing sample rate or interleaving/deinterleaving. You can also perform complex conversions such as compressing or decompressing audio. Three types of conversions are available:

When you use Audio Queue Services (described in Recording and Playback using Audio Queue Services), you get the appropriate converter automatically. Audio Codec Services (Mac only) lets you create specialized audio codecs, such as for handling digital rights management (DRM) or proprietary audio formats. After you create a custom codec, you can use an audio converter to access and use it.

When you use an audio converter explicitly in OS X, you call a conversion function with a particular converter instance, specifying where to find the input data and where to write the output. Most conversions require a callback function to periodically supply input data to the converter. For examples of how to use audio converters, see SimpleSDK/ConvertFile and the AFConvert command-line tool in Services/AudioFileTools in the Core Audio SDK.

Whenever you want to work with sound files in your application, you can use Audio File Services, one of the mid-level services in Core Audio. Audio File Services provides a powerful abstraction for accessing audio data and metadata contained in files, and for creating sound files.

In iOS you typically use Audio File Services to read audio data from, and write it to, sound files. Reading and writing are essentially mirror images of each other when you use Audio File Services. Both operations block until completion, and both can work using bytes or packets. However, unless you have special requirements, always use packets.

First, you need a callback for property changes in your audio file stream object. At a minimum, you write this callback to respond to changes in the kAudioFileStreamProperty_ReadyToProducePackets property. The typical scenario for using this property is as follows:

When enough audio data packets are parsed to send them along to your application, Audio File Stream Services sets the kAudioFileStreamProperty_ReadyToProducePackets property to true (actually, to a value of 1) in your audio file stream object.

Second, you need a callback for the audio data. Audio File Stream Services calls this callback whenever it has collected a set of complete audio data packets. You define this callback to handle the received audio. Typically, you play it back immediately by sending it along to Audio Queue Services. For more on playback, see the next section, Recording and Playback using Audio Queue Services.

With answers in hand, you employ the audio session interface (declared in AudioToolbox/AudioServices.h) to configure your audio session and your application. Table 2-2 describes the three programmatic features provided by this interface.

A category is a key that identifies a set of audio behaviors for your application. By setting a category, you indicate your audio intentions to iOS, such as whether your audio should continue when the screen locks.

You can query the audio session to discover characteristics of the device your application is running on, such as hardware sample rate, number of hardware channels, and whether audio input is available.

This set of behaviors is specified by the default audio session category, namely kAudioSessionCategory_SoloAmbientSound. iOS provides categories for a wide range of audio needs, ranging from user-interface sound effects to simultaneous audio input and output, as you would use for a VOIP (voice over Internet protocol) application. You can specify the category you want at launch and while your application runs.

Audio session default behavior is enough to get you started in iPhone audio development. Except for certain special cases, however, the default behavior is unsuitable for a shipping application, as described next.

One feature conspicuously absent from a default audio session is the ability to reactivate itself following an interruption. An audio session has two primary states: active and inactive. Audio can work in your application only when the audio session is active.

Upon launch, your default audio session is active. However, if a phone call comes in, your session is immediately deactivated and your audio stops. This is called an interruption. If the user chooses to ignore the phone call, your application continues running. But with your audio session inactive, audio does not work.

If you use OpenAL, the I/O audio unit, or Audio Queue Services for audio in your application, you must write an interruption listener callback function and explicitly reactivate your audio session when an interruption ends. Audio Session Programming Guide provides details and code examples.

A recording application on an iOS-based device can record only if hardware audio input is available. To test this, you use the audio session kAudioSessionProperty_AudioInputAvailable property. This is important when your application is running on devices like the iPod touch (2nd generation), which gain audio input only when appropriate accessory hardware is attached. Listing 2-10 shows how to perform the test.

When you add audio session support to your application, you can still run your app in the Simulator for development and testing. However, the Simulator does not simulate session behavior. To test the behavior of your audio session code, you need to run on a device.

Note: Ignoring Audio Session Services will not prevent your application from running, but your app may not behave the way you want it to. In most cases, you should not ship an iPhone or iPod touch application that uses audio without using this interface, as described earlier in this section.

The AVAudioPlayer class provides a simple Objective-C interface for audio playback. If your application does not require stereo positioning or precise synchronization, and if you are not playing audio captured from a network stream, Apple recommends that you use this class for playback.

Unlike OpenAL, the I/O audio unit, and Audio Queue Services, the AVAudioPlayer class does not require that you use Audio Session Services. An audio player reactivates itself following an interruption. If you want the behavior specified by the default audio session category (see Audio Sessions: Cooperating with Core Audio), such as having your audio stop when the screen locks, you can successfully use the default audio session with an audio player. 589ccfa754

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