java.lang.Objectjavax.sound.sampled.AudioFormat
AudioFormat is the class that specifies a particular arrangement of data in a sound stream.
By examing the information stored in the audio format, you can discover how to interpret the bits in the
binary sound data.
Every data line has an audio format associated with its data stream. The audio format of a source (playback) data line indicates
what kind of data the data line expects to receive for output. For a target (capture) data line, the audio format specifies the kind
of the data that can be read from the line.
Sound files also have audio formats, of course. The AudioFileFormat
class encapsulates an AudioFormat in addition to other,
file-specific information. Similarly, an AudioInputStream has an
AudioFormat.
The AudioFormat class accommodates a number of common sound-file encoding techniques, including
pulse-code modulation (PCM), mu-law encoding, and a-law encoding. These encoding techniques are predefined,
but service providers can create new encoding types.
The encoding that a specific format uses is named by its encoding field.
In addition to the encoding, the audio format includes other properties that further specify the exact arrangement of the data. These include the number of channels, sample rate, sample size, byte order, frame rate, and frame size. Sounds may have different numbers of audio channels: one for mono, two for stereo. The sample rate measures how many "snapshots" (samples) of the sound pressure are taken per second, per channel. (If the sound is stereo rather than mono, two samples are actually measured at each instant of time: one for the left channel, and another for the right channel; however, the sample rate still measures the number per channel, so the rate is the same regardless of the number of channels. This is the standard use of the term.) The sample size indicates how many bits are used to store each snapshot; 8 and 16 are typical values. For 16-bit samples (or any other sample size larger than a byte), byte order is important; the bytes in each sample are arranged in either the "little-endian" or "big-endian" style. For encodings like PCM, a frame consists of the set of samples for all channels at a given point in time, and so the size of a frame (in bytes) is always equal to the size of a sample (in bytes) times the number of channels. However, with some other sorts of encodings a frame can contain a bundle of compressed data for a whole series of samples, as well as additional, non-sample data. For such encodings, the sample rate and sample size refer to the data after it is decoded into PCM, and so they are completely different from the frame rate and frame size.
An AudioFormat object can include a set of
properties. A property is a pair of key and value: the key
is of type String, the associated property
value is an arbitrary object. Properties specify
additional format specifications, like the bit rate for
compressed formats. Properties are mainly used as a means
to transport additional information of the audio format
to and from the service providers. Therefore, properties
are ignored in the #matches(AudioFormat) method.
However, methods which rely on the installed service
providers, like (AudioFormat, AudioFormat) isConversionSupported may consider
properties, depending on the respective service provider
implementation.
The following table lists some common properties which service providers should use, if applicable:
| Property key | Value type | Description |
|---|---|---|
| "bitrate" | Integer | average bit rate in bits per second |
| "vbr" | Boolean | true, if the file is encoded in variable bit
rate (VBR) |
| "quality" | Integer | encoding/conversion quality, 1..100 |
Vendors of service providers (plugins) are encouraged to seek information about other already established properties in third party plugins, and follow the same conventions.
Kara - KytleFlorian - Bomers1.3 - | Nested Class Summary: | ||
|---|---|---|
| public static class | AudioFormat.Encoding | The Encoding class names the specific type of data representation
used for an audio stream. The encoding includes aspects of the
sound format other than the number of channels, sample rate, sample size,
frame rate, frame size, and byte order.
One ubiquitous type of audio encoding is pulse-code modulation (PCM), which is simply a linear (proportional) representation of the sound waveform. With PCM, the number stored in each sample is proportional to the instantaneous amplitude of the sound pressure at that point in time. The numbers are frequently signed or unsigned integers. Besides PCM, other encodings include mu-law and a-law, which are nonlinear mappings of the sound amplitude that are often used for recording speech.
You can use a predefined encoding by referring to one of the static
objects created by this class, such as PCM_SIGNED or
PCM_UNSIGNED. Service providers can create new encodings, such as
compressed audio formats or floating-point PCM samples, and make
these available through the
The |
| Field Summary | ||
|---|---|---|
| protected AudioFormat.Encoding | encoding | The audio encoding technique used by this format. |
| protected float | sampleRate | The number of samples played or recorded per second, for sounds that have this format. |
| protected int | sampleSizeInBits | The number of bits in each sample of a sound that has this format. |
| protected int | channels | The number of audio channels in this format (1 for mono, 2 for stereo). |
| protected int | frameSize | The number of bytes in each frame of a sound that has this format. |
| protected float | frameRate | The number of frames played or recorded per second, for sounds that have this format. |
| protected boolean | bigEndian | Indicates whether the audio data is stored in big-endian or little-endian order. |
| Constructor: |
|---|
AudioFormat with a linear PCM encoding and
the given parameters. The frame size is set to the number of bytes
required to contain one sample from each channel, and the frame rate
is set to the sample rate.
|
AudioFormat with the given parameters.
The encoding specifies the convention used to represent the data.
The other parameters are further explained in the class description .
|
AudioFormat with the given parameters.
The encoding specifies the convention used to represent the data.
The other parameters are further explained in the class description .
|
| Method from javax.sound.sampled.AudioFormat Summary: |
|---|
| getChannels, getEncoding, getFrameRate, getFrameSize, getProperty, getSampleRate, getSampleSizeInBits, isBigEndian, matches, properties, toString |
| Methods from java.lang.Object: |
|---|
| clone, equals, finalize, getClass, hashCode, notify, notifyAll, toString, wait, wait, wait |
| Method from javax.sound.sampled.AudioFormat Detail: |
|---|
AudioSystem.NOT_SPECIFIED means that any (positive) number of channels is
acceptable. |
|
AudioSystem.NOT_SPECIFIED means that any frame rate is
acceptable. AudioSystem.NOT_SPECIFIED is also returned when
the frame rate is not defined for this audio format. |
AudioSystem.NOT_SPECIFIED means that any frame size is
acceptable. AudioSystem.NOT_SPECIFIED is also returned when
the frame size is not defined for this audio format. |
If the specified property is not defined for a
particular file format, this method returns
|
AudioSystem.NOT_SPECIFIED means that any sample rate is
acceptable. AudioSystem.NOT_SPECIFIED is also returned when
the sample rate is not defined for this audio format. |
AudioSystem.NOT_SPECIFIED means that any sample size is
acceptable. AudioSystem.NOT_SPECIFIED is also returned when
the sample size is not defined for this audio format. |
|
AudioSystem.NOT_SPECIFIED,
which any sample rate will match. The frame rates must
similarly be equal, unless the specified format has the frame rate
value AudioSystem.NOT_SPECIFIED. The byte order (big-endian or little-endian)
must match if the sample size is greater than one byte. |
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