/* * Copyright (C) 2008 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ package android.media; import android.annotation.IntDef; import android.annotation.NonNull; import java.lang.annotation.Retention; import java.lang.annotation.RetentionPolicy; import java.util.Arrays; /** * The {@link AudioFormat} class is used to access a number of audio format and * channel configuration constants. They are for instance used * 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 * parameter is one of the AudioFormat.ENCODING_* constants. * The AudioFormat constants are also used in {@link MediaFormat} to specify * audio related values commonly used in media, such as for {@link MediaFormat#KEY_CHANNEL_MASK}. *

The {@link AudioFormat.Builder} class can be used to create instances of * the AudioFormat format class. * Refer to * {@link AudioFormat.Builder} for documentation on the mechanics of the configuration and building * of such instances. Here we describe the main concepts that the AudioFormat class * allow you to convey in each instance, they are: *

    *
  1. sample rate *
  2. encoding *
  3. channel masks *
*

Closely associated with the AudioFormat is the notion of an * audio frame, which is used throughout the documentation * to represent the minimum size complete unit of audio data. * *

Sample rate

*

Expressed in Hz, the sample rate in an AudioFormat instance expresses the number * of audio samples for each channel per second in the content you are playing or recording. It is * not the sample rate * at which content is rendered or produced. For instance a sound at a media sample rate of 8000Hz * 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. * *

As of API {@link android.os.Build.VERSION_CODES#M}, * sample rates up to 192kHz are supported * for AudioRecord and AudioTrack, with sample rate conversion * performed as needed. * To improve efficiency and avoid lossy conversions, it is recommended to match the sample rate * for AudioRecord and AudioTrack 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. * *

Encoding

*

Audio encoding is used to describe the bit representation of audio data, which can be * either linear PCM or compressed audio, such as AC3 or DTS. *

For linear PCM, the audio encoding describes the sample size, 8 bits, 16 bits, or 32 bits, * and the sample representation, integer or float. *

*

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 AudioTrack, 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. *

When compressed audio is sent out through a direct AudioTrack, * it need not be written in exact multiples of the audio access unit; * this differs from MediaCodec input buffers. * *

Channel mask

*

Channel masks are used in AudioTrack and AudioRecord to describe * the samples and their arrangement in the audio frame. They are also used in the endpoint (e.g. * a USB audio interface, a DAC connected to headphones) to specify allowable configurations of a * particular device. *
As of API {@link android.os.Build.VERSION_CODES#M}, there are two types of channel masks: * channel position masks and channel index masks. * *

Channel position masks
* Channel position masks are the original Android channel masks, and are used since API * {@link android.os.Build.VERSION_CODES#BASE}. * For input and output, they imply a positional nature - the location of a speaker or a microphone * for recording or playback. *
For a channel position mask, each allowed channel position corresponds to a bit in the * channel mask. If that channel position is present in the audio frame, that bit is set, * otherwise it is zero. The order of the bits (from lsb to msb) corresponds to the order of that * position's sample in the audio frame. *
The canonical channel position masks by channel count are as follows: *
* * * * * * * * * *
channel countchannel position mask
1{@link #CHANNEL_OUT_MONO}
2{@link #CHANNEL_OUT_STEREO}
3{@link #CHANNEL_OUT_STEREO} | {@link #CHANNEL_OUT_FRONT_CENTER}
4{@link #CHANNEL_OUT_QUAD}
5{@link #CHANNEL_OUT_QUAD} | {@link #CHANNEL_OUT_FRONT_CENTER}
6{@link #CHANNEL_OUT_5POINT1}
7{@link #CHANNEL_OUT_5POINT1} | {@link #CHANNEL_OUT_BACK_CENTER}
8{@link #CHANNEL_OUT_7POINT1_SURROUND}
*
These masks are an ORed composite of individual channel masks. For example * {@link #CHANNEL_OUT_STEREO} is composed of {@link #CHANNEL_OUT_FRONT_LEFT} and * {@link #CHANNEL_OUT_FRONT_RIGHT}. * *
Channel index masks
* Channel index masks are introduced in API {@link android.os.Build.VERSION_CODES#M}. They allow * the selection of a particular channel from the source or sink endpoint by number, i.e. the first * channel, the second channel, and so forth. This avoids problems with artificially assigning * positions to channels of an endpoint, or figuring what the ith position bit is within * an endpoint's channel position mask etc. *
Here's an example where channel index masks address this confusion: dealing with a 4 channel * USB device. Using a position mask, and based on the channel count, this would be a * {@link #CHANNEL_OUT_QUAD} device, but really one is only interested in channel 0 * through channel 3. The USB device would then have the following individual bit channel masks: * {@link #CHANNEL_OUT_FRONT_LEFT}, * {@link #CHANNEL_OUT_FRONT_RIGHT}, {@link #CHANNEL_OUT_BACK_LEFT} * and {@link #CHANNEL_OUT_BACK_RIGHT}. But which is channel 0 and which is * channel 3? *
For a channel index mask, each channel number is represented as a bit in the mask, from the * lsb (channel 0) upwards to the msb, numerically this bit value is * 1 << channelNumber. * A set bit indicates that channel is present in the audio frame, otherwise it is cleared. * The order of the bits also correspond to that channel number's sample order in the audio frame. *
For the previous 4 channel USB device example, the device would have a channel index mask * 0xF. Suppose we wanted to select only the first and the third channels; this would * correspond to a channel index mask 0x5 (the first and third bits set). If an * AudioTrack uses this channel index mask, the audio frame would consist of two * samples, the first sample of each frame routed to channel 0, and the second sample of each frame * routed to channel 2. * The canonical channel index masks by channel count are given by the formula * (1 << channelCount) - 1. * *
Use cases
* *

Audio Frame

*

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 channel mask, * and whose sample contents are specified by the encoding. * 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 { //--------------------------------------------------------- // Constants //-------------------- /** Invalid audio data format */ public static final int ENCODING_INVALID = 0; /** Default audio data format */ public static final int ENCODING_DEFAULT = 1; // These values must be kept in sync with core/jni/android_media_AudioFormat.h // Also sync av/services/audiopolicy/managerdefault/ConfigParsingUtils.h /** Audio data format: PCM 16 bit per sample. Guaranteed to be supported by devices. */ public static final int ENCODING_PCM_16BIT = 2; /** Audio data format: PCM 8 bit per sample. Not guaranteed to be supported by devices. */ public static final int ENCODING_PCM_8BIT = 3; /** Audio data format: single-precision floating-point per sample */ public static final int ENCODING_PCM_FLOAT = 4; /** Audio data format: AC-3 compressed */ public static final int ENCODING_AC3 = 5; /** Audio data format: E-AC-3 compressed */ public static final int ENCODING_E_AC3 = 6; /** Audio data format: DTS compressed */ public static final int ENCODING_DTS = 7; /** Audio data format: DTS HD compressed */ public static final int ENCODING_DTS_HD = 8; /** Audio data format: MP3 compressed * @hide * */ public static final int ENCODING_MP3 = 9; /** Audio data format: AAC LC compressed * @hide * */ public static final int ENCODING_AAC_LC = 10; /** Audio data format: AAC HE V1 compressed * @hide * */ public static final int ENCODING_AAC_HE_V1 = 11; /** Audio data format: AAC HE V2 compressed * @hide * */ public static final int ENCODING_AAC_HE_V2 = 12; /** Audio data format: AMRNB * @hide * */ public static final int ENCODING_AMRNB = 100; /** Audio data format: AMRWB * @hide * */ public static final int ENCODING_AMRWB = 101; /** Audio data format: EVRC * @hide * */ public static final int ENCODING_EVRC = 102; /** Audio data format: EVRCB * @hide * */ public static final int ENCODING_EVRCB = 103; /** Audio data format: EVRCWB * @hide * */ public static final int ENCODING_EVRCWB = 104; /** Audio data format: EVRCNW * @hide * */ public static final int ENCODING_EVRCNW = 105; /** Invalid audio channel configuration */ /** @deprecated Use {@link #CHANNEL_INVALID} instead. */ @Deprecated public static final int CHANNEL_CONFIGURATION_INVALID = 0; /** Default audio channel configuration */ /** @deprecated Use {@link #CHANNEL_OUT_DEFAULT} or {@link #CHANNEL_IN_DEFAULT} instead. */ @Deprecated public static final int CHANNEL_CONFIGURATION_DEFAULT = 1; /** Mono audio configuration */ /** @deprecated Use {@link #CHANNEL_OUT_MONO} or {@link #CHANNEL_IN_MONO} instead. */ @Deprecated public static final int CHANNEL_CONFIGURATION_MONO = 2; /** Stereo (2 channel) audio configuration */ /** @deprecated Use {@link #CHANNEL_OUT_STEREO} or {@link #CHANNEL_IN_STEREO} instead. */ @Deprecated public static final int CHANNEL_CONFIGURATION_STEREO = 3; /** Invalid audio channel mask */ public static final int CHANNEL_INVALID = 0; /** Default audio channel mask */ public static final int CHANNEL_OUT_DEFAULT = 1; // Output channel mask definitions below are translated to the native values defined in // in /system/media/audio/include/system/audio.h in the JNI code of AudioTrack public static final int CHANNEL_OUT_FRONT_LEFT = 0x4; public static final int CHANNEL_OUT_FRONT_RIGHT = 0x8; public static final int CHANNEL_OUT_FRONT_CENTER = 0x10; public static final int CHANNEL_OUT_LOW_FREQUENCY = 0x20; public static final int CHANNEL_OUT_BACK_LEFT = 0x40; public static final int CHANNEL_OUT_BACK_RIGHT = 0x80; public static final int CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100; public static final int CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200; public static final int CHANNEL_OUT_BACK_CENTER = 0x400; public static final int CHANNEL_OUT_SIDE_LEFT = 0x800; public static final int CHANNEL_OUT_SIDE_RIGHT = 0x1000; /** @hide */ public static final int CHANNEL_OUT_TOP_CENTER = 0x2000; /** @hide */ public static final int CHANNEL_OUT_TOP_FRONT_LEFT = 0x4000; /** @hide */ public static final int CHANNEL_OUT_TOP_FRONT_CENTER = 0x8000; /** @hide */ public static final int CHANNEL_OUT_TOP_FRONT_RIGHT = 0x10000; /** @hide */ public static final int CHANNEL_OUT_TOP_BACK_LEFT = 0x20000; /** @hide */ public static final int CHANNEL_OUT_TOP_BACK_CENTER = 0x40000; /** @hide */ public static final int CHANNEL_OUT_TOP_BACK_RIGHT = 0x80000; public static final int CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT; public static final int CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT); // aka QUAD_BACK public static final int CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT); /** @hide */ public static final int CHANNEL_OUT_QUAD_SIDE = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_SIDE_LEFT | CHANNEL_OUT_SIDE_RIGHT); public static final int CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER); // aka 5POINT1_BACK public static final int CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT); /** @hide */ public static final int CHANNEL_OUT_5POINT1_SIDE = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_SIDE_LEFT | CHANNEL_OUT_SIDE_RIGHT); // different from AUDIO_CHANNEL_OUT_7POINT1 used internally, and not accepted by AudioRecord. /** @deprecated Not the typical 7.1 surround configuration. Use {@link #CHANNEL_OUT_7POINT1_SURROUND} instead. */ @Deprecated public static final int CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER); // matches AUDIO_CHANNEL_OUT_7POINT1 public static final int CHANNEL_OUT_7POINT1_SURROUND = ( CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_SIDE_LEFT | CHANNEL_OUT_SIDE_RIGHT | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | CHANNEL_OUT_LOW_FREQUENCY); // CHANNEL_OUT_ALL is not yet defined; if added then it should match AUDIO_CHANNEL_OUT_ALL /** * @hide * Return the input channel mask corresponding to an output channel mask. * This can be used for submix rerouting for the mask of the recorder to map to that of the mix. * @param outMask a combination of the CHANNEL_OUT_* definitions, but not CHANNEL_OUT_DEFAULT * @return a combination of CHANNEL_IN_* definitions matching an output channel mask * @throws IllegalArgumentException */ public static int inChannelMaskFromOutChannelMask(int outMask) throws IllegalArgumentException { if (outMask == CHANNEL_OUT_DEFAULT) { throw new IllegalArgumentException( "Illegal CHANNEL_OUT_DEFAULT channel mask for input."); } switch (channelCountFromOutChannelMask(outMask)) { case 1: return CHANNEL_IN_MONO; case 2: return CHANNEL_IN_STEREO; default: throw new IllegalArgumentException("Unsupported channel configuration for input."); } } /** * @hide * Return the number of channels from an input channel mask * @param mask a combination of the CHANNEL_IN_* definitions, even CHANNEL_IN_DEFAULT * @return number of channels for the mask */ public static int channelCountFromInChannelMask(int mask) { return Integer.bitCount(mask); } /** * @hide * Return the number of channels from an output channel mask * @param mask a combination of the CHANNEL_OUT_* definitions, but not CHANNEL_OUT_DEFAULT * @return number of channels for the mask */ public static int channelCountFromOutChannelMask(int mask) { return Integer.bitCount(mask); } /** * @hide * Return a channel mask ready to be used by native code * @param mask a combination of the CHANNEL_OUT_* definitions, but not CHANNEL_OUT_DEFAULT * @return a native channel mask */ public static int convertChannelOutMaskToNativeMask(int javaMask) { return (javaMask >> 2); } /** * @hide * Return a java output channel mask * @param mask a native channel mask * @return a combination of the CHANNEL_OUT_* definitions */ public static int convertNativeChannelMaskToOutMask(int nativeMask) { return (nativeMask << 2); } public static final int CHANNEL_IN_DEFAULT = 1; // These directly match native public static final int CHANNEL_IN_LEFT = 0x4; public static final int CHANNEL_IN_RIGHT = 0x8; public static final int CHANNEL_IN_FRONT = 0x10; public static final int CHANNEL_IN_BACK = 0x20; public static final int CHANNEL_IN_LEFT_PROCESSED = 0x40; public static final int CHANNEL_IN_RIGHT_PROCESSED = 0x80; public static final int CHANNEL_IN_FRONT_PROCESSED = 0x100; public static final int CHANNEL_IN_BACK_PROCESSED = 0x200; public static final int CHANNEL_IN_PRESSURE = 0x400; public static final int CHANNEL_IN_X_AXIS = 0x800; public static final int CHANNEL_IN_Y_AXIS = 0x1000; public static final int CHANNEL_IN_Z_AXIS = 0x2000; public static final int CHANNEL_IN_VOICE_UPLINK = 0x4000; public static final int CHANNEL_IN_VOICE_DNLINK = 0x8000; public static final int CHANNEL_IN_MONO = CHANNEL_IN_FRONT; public static final int CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT); /** @hide */ public static final int CHANNEL_IN_FRONT_BACK = CHANNEL_IN_FRONT | CHANNEL_IN_BACK; /** @hide */ public static final int CHANNEL_IN_5POINT1 = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK | CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED); // CHANNEL_IN_ALL is not yet defined; if added then it should match AUDIO_CHANNEL_IN_ALL /** @hide */ public static int getBytesPerSample(int audioFormat) { switch (audioFormat) { case ENCODING_PCM_8BIT: return 1; case ENCODING_PCM_16BIT: case ENCODING_DEFAULT: return 2; case ENCODING_PCM_FLOAT: return 4; case ENCODING_AMRNB: return 32; case ENCODING_AMRWB: return 61; case ENCODING_EVRC: case ENCODING_EVRCB: case ENCODING_EVRCWB: case ENCODING_EVRCNW: return 23; case ENCODING_INVALID: default: throw new IllegalArgumentException("Bad audio format " + audioFormat); } } /** @hide */ public static boolean isValidEncoding(int audioFormat) { switch (audioFormat) { case ENCODING_PCM_8BIT: case ENCODING_PCM_16BIT: case ENCODING_PCM_FLOAT: case ENCODING_AC3: case ENCODING_E_AC3: case ENCODING_DTS: case ENCODING_DTS_HD: case ENCODING_MP3: case ENCODING_AAC_LC: case ENCODING_AAC_HE_V1: case ENCODING_AAC_HE_V2: case ENCODING_AMRNB: case ENCODING_AMRWB: case ENCODING_EVRC: case ENCODING_EVRCB: case ENCODING_EVRCWB: case ENCODING_EVRCNW: return true; default: return false; } } /** @hide */ public static boolean isPublicEncoding(int audioFormat) { switch (audioFormat) { case ENCODING_PCM_8BIT: case ENCODING_PCM_16BIT: case ENCODING_PCM_FLOAT: case ENCODING_AC3: case ENCODING_E_AC3: case ENCODING_DTS: case ENCODING_DTS_HD: return true; default: return false; } } /** @hide */ public static boolean isEncodingLinearPcm(int audioFormat) { switch (audioFormat) { case ENCODING_PCM_8BIT: case ENCODING_PCM_16BIT: case ENCODING_PCM_FLOAT: case ENCODING_DEFAULT: return true; case ENCODING_AC3: case ENCODING_E_AC3: case ENCODING_DTS: case ENCODING_DTS_HD: case ENCODING_MP3: case ENCODING_AAC_LC: case ENCODING_AAC_HE_V1: case ENCODING_AAC_HE_V2: case ENCODING_AMRNB: case ENCODING_AMRWB: case ENCODING_EVRC: case ENCODING_EVRCB: case ENCODING_EVRCWB: case ENCODING_EVRCNW: return false; case ENCODING_INVALID: default: throw new IllegalArgumentException("Bad audio format " + audioFormat); } } /** * Returns an array of public encoding values extracted from an array of * encoding values. * @hide */ public static int[] filterPublicFormats(int[] formats) { if (formats == null) { return null; } int[] myCopy = Arrays.copyOf(formats, formats.length); int size = 0; for (int i = 0; i < myCopy.length; i++) { if (isPublicEncoding(myCopy[i])) { if (size != i) { myCopy[size] = myCopy[i]; } size++; } } return Arrays.copyOf(myCopy, size); } /** @removed */ public AudioFormat() { throw new UnsupportedOperationException("There is no valid usage of this constructor"); } /** * Private constructor with an ignored argument to differentiate from the removed default ctor * @param ignoredArgument */ private AudioFormat(int ignoredArgument) { } /** * Constructor used by the JNI */ // Update sound trigger JNI in core/jni/android_hardware_SoundTrigger.cpp when modifying this // constructor private AudioFormat(int encoding, int sampleRate, int channelMask, int channelIndexMask) { mEncoding = encoding; mSampleRate = sampleRate; mChannelMask = channelMask; mChannelIndexMask = channelIndexMask; mPropertySetMask = AUDIO_FORMAT_HAS_PROPERTY_ENCODING | AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE | AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK | AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK; } /** @hide */ public final static int AUDIO_FORMAT_HAS_PROPERTY_NONE = 0x0; /** @hide */ public final static int AUDIO_FORMAT_HAS_PROPERTY_ENCODING = 0x1 << 0; /** @hide */ public final static int AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE = 0x1 << 1; /** @hide */ public final static int AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK = 0x1 << 2; /** @hide */ public final static int AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK = 0x1 << 3; private int mEncoding; private int mSampleRate; private int mChannelMask; private int mChannelIndexMask; private int mPropertySetMask; /** * Return the encoding. * See the section on encodings for more information about the different * types of supported audio encoding. * @return one of the values that can be set in {@link Builder#setEncoding(int)} or * {@link AudioFormat#ENCODING_INVALID} if not set. */ public int getEncoding() { if ((mPropertySetMask & AUDIO_FORMAT_HAS_PROPERTY_ENCODING) == 0) { return ENCODING_INVALID; } return mEncoding; } /** * Return the sample rate. * @return one of the values that can be set in {@link Builder#setSampleRate(int)} or * 0 if not set. */ public int getSampleRate() { if ((mPropertySetMask & AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE) == 0) { return 0; } return mSampleRate; } /** * Return the channel mask. * See the section on channel masks for more information about * the difference between index-based masks(as returned by {@link #getChannelIndexMask()}) and * the position-based mask returned by this function. * @return one of the values that can be set in {@link Builder#setChannelMask(int)} or * {@link AudioFormat#CHANNEL_INVALID} if not set. */ public int getChannelMask() { if ((mPropertySetMask & AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) == 0) { return CHANNEL_INVALID; } return mChannelMask; } /** * Return the channel index mask. * See the section on channel masks for more information about * the difference between index-based masks, and position-based masks (as returned * by {@link #getChannelMask()}). * @return one of the values that can be set in {@link Builder#setChannelIndexMask(int)} or * {@link AudioFormat#CHANNEL_INVALID} if not set or an invalid mask was used. */ public int getChannelIndexMask() { if ((mPropertySetMask & AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) == 0) { return CHANNEL_INVALID; } return mChannelIndexMask; } /** * Return the channel count. * @return the channel count derived from the channel position mask or the channel index mask. * Zero is returned if both the channel position mask and the channel index mask are not set. */ public int getChannelCount() { final int channelIndexCount = Integer.bitCount(getChannelIndexMask()); int channelCount = channelCountFromOutChannelMask(getChannelMask()); if (channelCount == 0) { channelCount = channelIndexCount; } else if (channelCount != channelIndexCount && channelIndexCount != 0) { channelCount = 0; // position and index channel count mismatch } return channelCount; } /** @hide */ public int getPropertySetMask() { return mPropertySetMask; } /** * Builder class for {@link AudioFormat} objects. * Use this class to configure and create an AudioFormat instance. By setting format * characteristics such as audio encoding, channel mask or sample rate, you indicate which * of those are to vary from the default behavior on this device wherever this audio format * is used. See {@link AudioFormat} for a complete description of the different parameters that * can be used to configure an AudioFormat instance. *

{@link AudioFormat} is for instance used in * {@link AudioTrack#AudioTrack(AudioAttributes, AudioFormat, int, int, int)}. In this * constructor, every format characteristic set on the Builder (e.g. with * {@link #setSampleRate(int)}) will alter the default values used by an * AudioTrack. In this case for audio playback with AudioTrack, the * sample rate set in the Builder would override the platform output sample rate * which would otherwise be selected by default. */ public static class Builder { private int mEncoding = ENCODING_INVALID; private int mSampleRate = 0; private int mChannelMask = CHANNEL_INVALID; private int mChannelIndexMask = 0; private int mPropertySetMask = AUDIO_FORMAT_HAS_PROPERTY_NONE; /** * Constructs a new Builder with none of the format characteristics set. */ public Builder() { } /** * Constructs a new Builder from a given {@link AudioFormat}. * @param af the {@link AudioFormat} object whose data will be reused in the new Builder. */ public Builder(AudioFormat af) { mEncoding = af.mEncoding; mSampleRate = af.mSampleRate; mChannelMask = af.mChannelMask; mChannelIndexMask = af.mChannelIndexMask; mPropertySetMask = af.mPropertySetMask; } /** * Combines all of the format characteristics that have been set and return a new * {@link AudioFormat} object. * @return a new {@link AudioFormat} object */ public AudioFormat build() { AudioFormat af = new AudioFormat(1980/*ignored*/); af.mEncoding = mEncoding; af.mSampleRate = mSampleRate; af.mChannelMask = mChannelMask; af.mChannelIndexMask = mChannelIndexMask; af.mPropertySetMask = mPropertySetMask; return af; } /** * Sets the data encoding format. * @param encoding one of {@link AudioFormat#ENCODING_DEFAULT}, * {@link AudioFormat#ENCODING_PCM_8BIT}, * {@link AudioFormat#ENCODING_PCM_16BIT}, * {@link AudioFormat#ENCODING_PCM_FLOAT}, * {@link AudioFormat#ENCODING_AC3}, * {@link AudioFormat#ENCODING_E_AC3}. * {@link AudioFormat#ENCODING_DTS}, * {@link AudioFormat#ENCODING_DTS_HD}. * @return the same Builder instance. * @throws java.lang.IllegalArgumentException */ public Builder setEncoding(@Encoding int encoding) throws IllegalArgumentException { switch (encoding) { case ENCODING_DEFAULT: mEncoding = ENCODING_PCM_16BIT; break; case ENCODING_PCM_8BIT: case ENCODING_PCM_16BIT: case ENCODING_PCM_FLOAT: case ENCODING_AC3: case ENCODING_E_AC3: case ENCODING_DTS: case ENCODING_DTS_HD: case ENCODING_AMRNB: case ENCODING_AMRWB: case ENCODING_EVRC: case ENCODING_EVRCB: case ENCODING_EVRCWB: case ENCODING_EVRCNW: mEncoding = encoding; break; case ENCODING_INVALID: default: throw new IllegalArgumentException("Invalid encoding " + encoding); } mPropertySetMask |= AUDIO_FORMAT_HAS_PROPERTY_ENCODING; return this; } /** * Sets the channel position mask. * The channel position mask specifies the association between audio samples in a frame * with named endpoint channels. The samples in the frame correspond to the * named set bits in the channel position mask, in ascending bit order. * See {@link #setChannelIndexMask(int)} to specify channels * based on endpoint numbered channels. This 192000)) { throw new IllegalArgumentException("Invalid sample rate " + sampleRate); } mSampleRate = sampleRate; mPropertySetMask |= AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE; return this; } } @Override public String toString () { return new String("AudioFormat:" + " props=" + mPropertySetMask + " enc=" + mEncoding + " chan=0x" + Integer.toHexString(mChannelMask).toUpperCase() + " chan_index=0x" + Integer.toHexString(mChannelIndexMask).toUpperCase() + " rate=" + mSampleRate); } /** @hide */ @IntDef({ ENCODING_DEFAULT, ENCODING_PCM_8BIT, ENCODING_PCM_16BIT, ENCODING_PCM_FLOAT, ENCODING_AC3, ENCODING_E_AC3, ENCODING_DTS, ENCODING_DTS_HD, ENCODING_AMRNB, ENCODING_AMRWB, ENCODING_EVRC, ENCODING_EVRCB, ENCODING_EVRCWB, ENCODING_EVRCNW }) @Retention(RetentionPolicy.SOURCE) public @interface Encoding {} }