Clarify AudioFormat sample rate and encoding documentation

import android.annotation.IntDef;

import android.annotation.IntDef;

import android.annotation.NonNull;

import android.annotation.NonNull;

import java.lang.annotation.Retention;

import java.lang.annotation.Retention;

import java.lang.annotation.RetentionPolicy;

import java.lang.annotation.RetentionPolicy;

import java.util.Arrays;

import java.util.Arrays;

/**

/**

 * The <code>AudioFormat</code> class is used to access a number of audio format and

 * The {@link AudioFormat} class is used to access a number of audio format and

 * channel configuration constants. They are for instance used

 * channel configuration constants. They are for instance used

 * in {@link AudioTrack} and {@link AudioRecord}, as valid values in individual parameters of

 * in {@link AudioTrack} and {@link AudioRecord}, as valid values in individual parameters of

 * constructors like {@link AudioTrack#AudioTrack(int, int, int, int, int, int)}, where the fourth

 * constructors like {@link AudioTrack#AudioTrack(int, int, int, int, int, int)}, where the fourth

 * parameter is one of the <code>AudioFormat.ENCODING_*</code> constants.

 * parameter is one of the <code>AudioFormat.ENCODING_*</code> constants.

 * The <code>AudioFormat</code> constants are also used in {@link MediaFormat} to specify

 * audio related values commonly used in media, such as for {@link MediaFormat#KEY_CHANNEL_MASK}.

 * <p>The {@link AudioFormat.Builder} class can be used to create instances of

 * <p>The {@link AudioFormat.Builder} class can be used to create instances of

 * the <code>AudioFormat</code> format class.

 * the <code>AudioFormat</code> format class.

 * Refer to

 * Refer to

 * <li><a href="#encoding">encoding</a>

 * <li><a href="#encoding">encoding</a>

 * <li><a href="#channelMask">channel masks</a>

 * <li><a href="#channelMask">channel masks</a>

 * </ol>

 * </ol>

 * <p>Closely associated with the <code>AudioFormat</code> is the notion of an

 * <a href="#audioFrame">audio frame</a>, which is used throughout the documentation

 * to represent the minimum size complete unit of audio data.

 *

 *

 * <h4 id="sampleRate">Sample rate</h4>

 * <h4 id="sampleRate">Sample rate</h4>

 * <p>Expressed in Hz, the sample rate in an <code>AudioFormat</code> instance expresses the number

 * <p>Expressed in Hz, the sample rate in an <code>AudioFormat</code> instance expresses the number

 * can be played on a device operating at a sample rate of 48000Hz; the sample rate conversion is

 * can be played on a device operating at a sample rate of 48000Hz; the sample rate conversion is

 * automatically handled by the platform, it will not play at 6x speed.

 * automatically handled by the platform, it will not play at 6x speed.

 *

 *

 * <p>As of API {@link android.os.Build.VERSION_CODES#MNC},

 * sample rates up to 192kHz are supported

 * for <code>AudioRecord</code> and <code>AudioTrack</code>, with sample rate conversion

 * performed as needed.

 * To improve efficiency and avoid lossy conversions, it is recommended to match the sample rate

 * for <code>AudioRecord</code> and <code>AudioTrack</code> to the endpoint device

 * sample rate, and limit the sample rate to no more than 48kHz unless there are special

 * device capabilities that warrant a higher rate.

 *

 * <h4 id="encoding">Encoding</h4>

 * <h4 id="encoding">Encoding</h4>

 * <p>For PCM audio, audio encoding is used to describe the bit representation of an audio data

 * <p>Audio encoding is used to describe the bit representation of audio data, which can be

 * sample; for example, the size as 8 bit, 16 bit, and the representation as integer or float.

 * either linear PCM or compressed audio, such as AC3 or DTS.

 * <br>For compressed formats, audio encoding is used to describe the compression scheme being used.

 * <p>For linear PCM, the audio encoding describes the sample size, 8 bits, 16 bits, or 32 bits,

 * and the sample representation, integer or float.

 * <ul>

 * <li> {@link #ENCODING_PCM_8BIT}: The audio sample is a 8 bit unsigned integer in the

 * range [0, 255], with a 128 offset for zero. This is typically stored as a Java byte in a

 * byte array or ByteBuffer. Since the Java byte is <em>signed</em>,

 * be careful with math operations and conversions as the most significant bit is inverted.

 * </li>

 * <li> {@link #ENCODING_PCM_16BIT}: The audio sample is a 16 bit signed integer

 * typically stored as a Java short in a short array, but when the short

 * is stored in a ByteBuffer, it is native endian (as compared to the default Java big endian).

 * The short has full range from [-32768, 32767],

 * and is sometimes interpreted as fixed point Q.15 data.

 * </li>

 * <li> {@link #ENCODING_PCM_FLOAT}: Introduced in

 * API {@link android.os.Build.VERSION_CODES#LOLLIPOP}, this encoding specifies that

 * the audio sample is a 32 bit IEEE single precision float. The sample can be

 * manipulated as a Java float in a float array, though within a ByteBuffer

 * it is stored in native endian byte order.

 * The nominal range of <code>ENCODING_PCM_FLOAT</code> audio data is [-1.0, 1.0].

 * It is implementation dependent whether the positive maximum of 1.0 is included

 * in the interval. Values outside of the nominal range are clamped before

 * sending to the endpoint device. Beware that

 * the handling of NaN is undefined; subnormals may be treated as zero; and

 * infinities are generally clamped just like other values for <code>AudioTrack</code>

 * – try to avoid infinities because they can easily generate a NaN.

 * <br>

 * To achieve higher audio bit depth than a signed 16 bit integer short,

 * it is recommended to use <code>ENCODING_PCM_FLOAT</code> for audio capture, processing,

 * and playback.

 * Floats are efficiently manipulated by modern CPUs,

 * have greater precision than 24 bit signed integers,

 * and have greater dynamic range than 32 bit signed integers.

 * <code>AudioRecord</code> as of API {@link android.os.Build.VERSION_CODES#MNC} and

 * <code>AudioTrack</code> as of API {@link android.os.Build.VERSION_CODES#LOLLIPOP}

 * support <code>ENCODING_PCM_FLOAT</code>.

 * </li>

 * </ul>

 * <p>For compressed audio, the encoding specifies the method of compression,

 * for example {@link #ENCODING_AC3} and {@link #ENCODING_DTS}. The compressed

 * audio data is typically stored as bytes in

 * a byte array or ByteBuffer. When a compressed audio encoding is specified

 * for an <code>AudioTrack</code>, it creates a direct (non-mixed) track

 * for output to an endpoint (such as HDMI) capable of decoding the compressed audio.

 * For (most) other endpoints, which are not capable of decoding such compressed audio,

 * you will need to decode the data first, typically by creating a {@link MediaCodec}.

 * Alternatively, one may use {@link MediaPlayer} for playback of compressed

 * audio files or streams.

 * <p>When compressed audio is sent out through a direct <code>AudioTrack</code>,

 * it need not be written in exact multiples of the audio access unit;

 * this differs from <code>MediaCodec</code> input buffers.

 *

 *

 * <h4 id="channelMask">Channel mask</h4>

 * <h4 id="channelMask">Channel mask</h4>

 * <p>Channel masks are used in <code>AudioTrack</code> and <code>AudioRecord</code> to describe

 * <p>Channel masks are used in <code>AudioTrack</code> and <code>AudioRecord</code> to describe

 *  about position it corresponds to, in which case the channel index mask is <code>0xC</code>.

 *  about position it corresponds to, in which case the channel index mask is <code>0xC</code>.

 *  Multichannel <code>AudioRecord</code> sessions should use channel index masks.

 *  Multichannel <code>AudioRecord</code> sessions should use channel index masks.

 * </ul>

 * </ul>

 * <h4 id="audioFrame">Audio Frame</h4>

 * <p>For linear PCM, an audio frame consists of a set of samples captured at the same time,

 * whose count and

 * channel association are given by the <a href="#channelMask">channel mask</a>,

 * and whose sample contents are specified by the <a href="#encoding">encoding</a>.

 * For example, a stereo 16 bit PCM frame consists of

 * two 16 bit linear PCM samples, with a frame size of 4 bytes.

 * For compressed audio, an audio frame may alternately

 * refer to an access unit of compressed data bytes that is logically grouped together for

 * decoding and bitstream access (e.g. {@link MediaCodec}),

 * or a single byte of compressed data (e.g. {@link AudioTrack#getBufferSizeInFrames()

 * AudioTrack.getBufferSizeInFrames()}),

 * or the linear PCM frame result from decoding the compressed data

 * (e.g.{@link AudioTrack#getPlaybackHeadPosition()

 * AudioTrack.getPlaybackHeadPosition()}),

 * depending on the context where audio frame is used.

 */

 */

public class AudioFormat {

public class AudioFormat {