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WARNING

At the time of this writing, there is a bug when playing a sound file. The file plays correctly the first time, but with a small delay every time thereafter. Let’s hope that this is fixed in a future release.

Settings and the Audio Codec

The Flash Authoring tool offers the option to modify audio files directly in the library. You can change compression, convert from stereo to mono, and choose a sample rate without requiring an external audio tool. Settings are set globally in the Publish Settings panel and can be overwritten for individual files in the library.

If you own Soundbooth, or another external audio application, you can launch it for an individual sound from within the development tools and make changes, which will be applied to the sound in your project. You can, for instance, change the track from stereo to mono or adjust the volume.

In Flash Professional, select the track in the library, click the top pull-down menu, and select “Edit with” to launch the audio editing application. In Flash Builder, single-click the asset, right-click, and select “Open with” to launch the sound application.

The most professional approach, of course, is to work in an audio application directly, as you have more control over your sound design: all files can be opened together and you can set uniform settings such as volume. Prepare your audio carefully beforehand to remove any unnecessary bytes. For background music, write a small file which loops, rather than a long track.

Compression

Supported compressed formats are MP3 (MPEG-1 Audio Layer 3), AAC (Advanced Audio Coding), WAV (Waveform Audio File Format), and AIFF (Audio Interchange File Format).

MP3 can be imported dynamically using the Sound object. MP3 adds a problematic small silence at the beginning of the track. MP3 encodes incoming audio data in blocks. If the data does not fill up a complete block, the encoder adds padding at the beginning and the end of the track. Read André Michelle’s blog on the issue, and a potential solution, at http://blog.andre-michelle.com/2010/playback-mp3-loop-gapless/.

AAC audio can also be loaded dynamically using the NetStream class. AAC is considered the successor of the MP3 format. It is particularly interesting to us because it is hardware-decoded in AIR for Android:

import flash.net.NetConnection;

import flash.net.NetStream;

var connection:NetConnection = new NetConnection();

connection.connect(null);

var stream:NetStream = new NetStream(connection);

var client:Object = new Object();

client.onMetaData = onMetaData;

stream.client = client;

stream.play("someAudio.m4a");

To control or manipulate an AAC file, refer to the section Playing Sounds. Here is some sample code:

var mySound:SoundTransform;

stream.play("someAudio.m4a");

mySound = stream.soundTransform;

// change volume

mySound.volume = 0.75;

stream.soundTransform = mySound;

NOTE

AAC or HE-AAC (High Efficiency AAC) is a lossless form of compression. It can support up to 96 kHz and 48 channels, and is high-efficiency (HE), high-fidelity (HiFi), and low-bandwidth. It uses Spectral Band Replication (SBR) and Parametric Stereo (PS) to increase coding efficiency and low use of bandwidth, in particular for streaming.

You can embed WAV or AIFF files in your project or library. Or you can use one of the third-party tools mentioned earlier.

Supported uncompressed settings are Adaptive Differential Pulse Code Modulation (ADPCM), and Raw, which uses no compression at all. Uncompressed formats must be embedded.

Bit rate

The bit rate represents the amount of data encoded for one second of a sound file. The higher the bit rate, the better the audio fidelity, but the bigger the file size. For mobile applications, consider reducing the bit rate that you would normally choose for desktop applications.

Bit rate is represented in kilobits per second (kbps), and ranges from 8 to 160 kbps. The default audio publish setting in Flash Professional is 16 kbps Mono.

Sampling rate

The sampling rate is the number