--- /dev/null
+/*
+ * Copyright (C) 2016 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.hardware.audio.common@2.0;
+
+/*
+ *
+ * IDs and Handles
+ *
+ */
+
+/*
+ * Handle type for identifying audio sources and sinks.
+ */
+typedef int32_t AudioIoHandle;
+
+/*
+ * Audio hw module handle functions or structures referencing a module.
+ */
+typedef int32_t AudioModuleHandle;
+
+/*
+ * Each port has a unique ID or handle allocated by policy manager.
+ */
+typedef int32_t AudioPortHandle;
+
+/*
+ * Each patch is identified by a handle at the interface used to create that
+ * patch. For instance, when a patch is created by the audio HAL, the HAL
+ * allocates and returns a handle. This handle is unique to a given audio HAL
+ * hardware module. But the same patch receives another system wide unique
+ * handle allocated by the framework. This unique handle is used for all
+ * transactions inside the framework.
+ */
+typedef int32_t AudioPatchHandle;
+
+/*
+ * A HW synchronization source returned by the audio HAL.
+ */
+typedef uint32_t AudioHwSync;
+
+/*
+ * Each port has a unique ID or handle allocated by policy manager.
+ */
+enum AudioHandleConsts {
+ AUDIO_IO_HANDLE_NONE = 0,
+ AUDIO_MODULE_HANDLE_NONE = 0,
+ AUDIO_PORT_HANDLE_NONE = 0,
+ AUDIO_PATCH_HANDLE_NONE = 0,
+ AUDIO_HW_SYNC_INVALID = 0
+};
+
+/*
+ * Commonly used structure for passing unique identifieds (UUID).
+ * For the definition of UUID, refer to ITU-T X.667 spec.
+ */
+struct Uuid {
+ uint32_t timeLow;
+ uint16_t timeMid;
+ uint16_t versionAndTimeHigh;
+ uint16_t variantAndClockSeqHigh;
+ uint8_t[6] node;
+};
+
+
+/*
+ *
+ * Audio streams
+ *
+ */
+
+/*
+ * Audio stream type describing the intented use case of a stream.
+ */
+enum AudioStreamType : int32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioSystem.java
+ // TODO: Synchronization should be done automatically by tools
+ AUDIO_STREAM_DEFAULT = -1,
+ AUDIO_STREAM_MIN = 0,
+ AUDIO_STREAM_VOICE_CALL = 0,
+ AUDIO_STREAM_SYSTEM = 1,
+ AUDIO_STREAM_RING = 2,
+ AUDIO_STREAM_MUSIC = 3,
+ AUDIO_STREAM_ALARM = 4,
+ AUDIO_STREAM_NOTIFICATION = 5,
+ AUDIO_STREAM_BLUETOOTH_SCO = 6,
+ AUDIO_STREAM_ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user
+ // and must be routed to speaker
+ AUDIO_STREAM_DTMF = 8,
+ AUDIO_STREAM_TTS = 9, // Transmitted Through Speaker
+ // Plays over speaker only, silent
+ // on other devices
+ AUDIO_STREAM_ACCESSIBILITY = 10, // For accessibility talk back prompts
+ AUDIO_STREAM_REROUTING = 11, // For dynamic policy output mixes
+ AUDIO_STREAM_PATCH = 12, // For internal audio flinger tracks.
+ // Fixed volume
+ AUDIO_STREAM_PUBLIC_CNT = AUDIO_STREAM_TTS + 1,
+ // Number of streams considered by audio policy for volume and routing
+ AUDIO_STREAM_FOR_POLICY_CNT = AUDIO_STREAM_PATCH,
+ AUDIO_STREAM_CNT = AUDIO_STREAM_PATCH + 1
+};
+
+/*
+ * The content type expresses the general category of the content --
+ * what is being played.
+ */
+enum AudioContentType {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioAttributes.java
+ AUDIO_CONTENT_TYPE_UNKNOWN = 0,
+ AUDIO_CONTENT_TYPE_SPEECH = 1,
+ AUDIO_CONTENT_TYPE_MUSIC = 2,
+ AUDIO_CONTENT_TYPE_MOVIE = 3,
+ AUDIO_CONTENT_TYPE_SONIFICATION = 4,
+
+ AUDIO_CONTENT_TYPE_CNT,
+ AUDIO_CONTENT_TYPE_MAX = AUDIO_CONTENT_TYPE_CNT - 1
+};
+
+/*
+ * The usage defines what is the purpose of the sound -- why somebody
+ * is playing it.
+ */
+enum AudioUsage {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioAttributes.java
+ AUDIO_USAGE_UNKNOWN = 0,
+ AUDIO_USAGE_MEDIA = 1,
+ AUDIO_USAGE_VOICE_COMMUNICATION = 2,
+ AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3,
+ AUDIO_USAGE_ALARM = 4,
+ AUDIO_USAGE_NOTIFICATION = 5,
+ AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE = 6,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST = 7,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT = 8,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED = 9,
+ AUDIO_USAGE_NOTIFICATION_EVENT = 10,
+ AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11,
+ AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12,
+ AUDIO_USAGE_ASSISTANCE_SONIFICATION = 13,
+ AUDIO_USAGE_GAME = 14,
+ AUDIO_USAGE_VIRTUAL_SOURCE = 15,
+
+ AUDIO_USAGE_CNT,
+ AUDIO_USAGE_MAX = AUDIO_USAGE_CNT - 1
+};
+
+/*
+ * Additional flags that can affect playback of the sound.
+ */
+enum AudioFlagsMask : uint32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioAttributes.java
+ AUDIO_FLAG_AUDIBILITY_ENFORCED = 0x1,
+ AUDIO_FLAG_SECURE = 0x2,
+ AUDIO_FLAG_SCO = 0x4,
+ AUDIO_FLAG_BEACON = 0x8,
+ AUDIO_FLAG_HW_AV_SYNC = 0x10,
+ AUDIO_FLAG_HW_HOTWORD = 0x20,
+ AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY = 0x40,
+ AUDIO_FLAG_BYPASS_MUTE = 0x80,
+ AUDIO_FLAG_LOW_LATENCY = 0x100
+};
+
+enum AudioSource {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/MediaRecorder.java,
+ // frameworks/av/services/audiopolicy/AudioPolicyService.cpp,
+ // system/media/audio_effects/include/audio_effects/audio_effects_conf.h
+ AUDIO_SOURCE_DEFAULT = 0,
+ AUDIO_SOURCE_MIC = 1,
+ AUDIO_SOURCE_VOICE_UPLINK = 2,
+ AUDIO_SOURCE_VOICE_DOWNLINK = 3,
+ AUDIO_SOURCE_VOICE_CALL = 4,
+ AUDIO_SOURCE_CAMCORDER = 5,
+ AUDIO_SOURCE_VOICE_RECOGNITION = 6,
+ AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
+ /*
+ * Source for the mix to be presented remotely. An example of remote
+ * presentation is Wifi Display where a dongle attached to a TV can be used
+ * to play the mix captured by this audio source.
+ */
+ AUDIO_SOURCE_REMOTE_SUBMIX = 8,
+ /*
+ * Source for unprocessed sound. Usage examples include level measurement
+ * and raw signal analysis.
+ */
+ AUDIO_SOURCE_UNPROCESSED = 9,
+
+ AUDIO_SOURCE_CNT,
+ AUDIO_SOURCE_MAX = AUDIO_SOURCE_CNT - 1,
+ AUDIO_SOURCE_FM_TUNER = 1998,
+ /*
+ * A low-priority, preemptible audio source for for background software
+ * hotword detection. Same tuning as AUDIO_SOURCE_VOICE_RECOGNITION.
+ * Used only internally to the framework.
+ */
+ AUDIO_SOURCE_HOTWORD = 1999
+};
+
+struct AudioAttributes {
+ AudioContentType contentType;
+ AudioUsage usage;
+ AudioSource source;
+ AudioFlagsMask flags;
+ string tags;
+};
+
+typedef int32_t AudioSession;
+/*
+ * Special audio session values.
+ */
+enum AudioSessionConsts : int32_t {
+ /*
+ * Session for effects attached to a particular output stream
+ * (value must be less than 0)
+ */
+ AUDIO_SESSION_OUTPUT_STAGE = -1,
+ /*
+ * Session for effects applied to output mix. These effects can
+ * be moved by audio policy manager to another output stream
+ * (value must be 0)
+ */
+ AUDIO_SESSION_OUTPUT_MIX = 0,
+ /*
+ * Application does not specify an explicit session ID to be used, and
+ * requests a new session ID to be allocated TODO use unique values for
+ * AUDIO_SESSION_OUTPUT_MIX and AUDIO_SESSION_ALLOCATE, after all uses have
+ * been updated from 0 to the appropriate symbol, and have been tested.
+ * Corresponds to AudioManager.AUDIO_SESSION_ID_GENERATE and
+ * AudioSystem.AUDIO_SESSION_ALLOCATE.
+ */
+ AUDIO_SESSION_ALLOCATE = 0,
+ /*
+ * For use with AudioRecord::start(), this indicates no trigger session.
+ * It is also used with output tracks and patch tracks, which never have a
+ * session.
+ */
+ AUDIO_SESSION_NONE = 0
+};
+
+/*
+ * Audio format is a 32-bit word that consists of:
+ * main format field (upper 8 bits)
+ * sub format field (lower 24 bits).
+ *
+ * The main format indicates the main codec type. The sub format field indicates
+ * options and parameters for each format. The sub format is mainly used for
+ * record to indicate for instance the requested bitrate or profile. It can
+ * also be used for certain formats to give informations not present in the
+ * encoded audio stream (e.g. octet alignement for AMR).
+ */
+enum AudioFormat : uint32_t {
+ AUDIO_FORMAT_INVALID = 0xFFFFFFFFUL,
+ AUDIO_FORMAT_DEFAULT = 0,
+ AUDIO_FORMAT_PCM = 0x00000000UL, /* DO NOT CHANGE */
+ AUDIO_FORMAT_MP3 = 0x01000000UL,
+ AUDIO_FORMAT_AMR_NB = 0x02000000UL,
+ AUDIO_FORMAT_AMR_WB = 0x03000000UL,
+ AUDIO_FORMAT_AAC = 0x04000000UL,
+ /* Deprecated, Use AUDIO_FORMAT_AAC_HE_V1*/
+ AUDIO_FORMAT_HE_AAC_V1 = 0x05000000UL,
+ /* Deprecated, Use AUDIO_FORMAT_AAC_HE_V2*/
+ AUDIO_FORMAT_HE_AAC_V2 = 0x06000000UL,
+ AUDIO_FORMAT_VORBIS = 0x07000000UL,
+ AUDIO_FORMAT_OPUS = 0x08000000UL,
+ AUDIO_FORMAT_AC3 = 0x09000000UL,
+ AUDIO_FORMAT_E_AC3 = 0x0A000000UL,
+ AUDIO_FORMAT_DTS = 0x0B000000UL,
+ AUDIO_FORMAT_DTS_HD = 0x0C000000UL,
+ // IEC61937 is encoded audio wrapped in 16-bit PCM.
+ AUDIO_FORMAT_IEC61937 = 0x0D000000UL,
+ AUDIO_FORMAT_DOLBY_TRUEHD = 0x0E000000UL,
+ AUDIO_FORMAT_MAIN_MASK = 0xFF000000UL, /* Deprecated */
+ AUDIO_FORMAT_SUB_MASK = 0x00FFFFFFUL,
+
+ /* Subformats */
+ AUDIO_FORMAT_PCM_SUB_16_BIT = 0x1, // PCM signed 16 bits
+ AUDIO_FORMAT_PCM_SUB_8_BIT = 0x2, // PCM unsigned 8 bits
+ AUDIO_FORMAT_PCM_SUB_32_BIT = 0x3, // PCM signed .31 fixed point
+ AUDIO_FORMAT_PCM_SUB_8_24_BIT = 0x4, // PCM signed 8.23 fixed point
+ AUDIO_FORMAT_PCM_SUB_FLOAT = 0x5, // PCM single-precision float pt
+ AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED = 0x6, // PCM signed .23 fix pt (3 bytes)
+
+ AUDIO_FORMAT_MP3_SUB_NONE = 0x0,
+
+ AUDIO_FORMAT_AMR_SUB_NONE = 0x0,
+
+ AUDIO_FORMAT_AAC_SUB_MAIN = 0x1,
+ AUDIO_FORMAT_AAC_SUB_LC = 0x2,
+ AUDIO_FORMAT_AAC_SUB_SSR = 0x4,
+ AUDIO_FORMAT_AAC_SUB_LTP = 0x8,
+ AUDIO_FORMAT_AAC_SUB_HE_V1 = 0x10,
+ AUDIO_FORMAT_AAC_SUB_SCALABLE = 0x20,
+ AUDIO_FORMAT_AAC_SUB_ERLC = 0x40,
+ AUDIO_FORMAT_AAC_SUB_LD = 0x80,
+ AUDIO_FORMAT_AAC_SUB_HE_V2 = 0x100,
+ AUDIO_FORMAT_AAC_SUB_ELD = 0x200,
+
+ AUDIO_FORMAT_VORBIS_SUB_NONE = 0x0,
+
+ /* Aliases */
+ /* note != AudioFormat.ENCODING_PCM_16BIT */
+ AUDIO_FORMAT_PCM_16_BIT = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_16_BIT),
+ /* note != AudioFormat.ENCODING_PCM_8BIT */
+ AUDIO_FORMAT_PCM_8_BIT = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_8_BIT),
+ AUDIO_FORMAT_PCM_32_BIT = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_32_BIT),
+ AUDIO_FORMAT_PCM_8_24_BIT = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_8_24_BIT),
+ AUDIO_FORMAT_PCM_FLOAT = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_FLOAT),
+ AUDIO_FORMAT_PCM_24_BIT_PACKED = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED),
+ AUDIO_FORMAT_AAC_MAIN = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_MAIN),
+ AUDIO_FORMAT_AAC_LC = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_LC),
+ AUDIO_FORMAT_AAC_SSR = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_SSR),
+ AUDIO_FORMAT_AAC_LTP = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_LTP),
+ AUDIO_FORMAT_AAC_HE_V1 = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_HE_V1),
+ AUDIO_FORMAT_AAC_SCALABLE = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_SCALABLE),
+ AUDIO_FORMAT_AAC_ERLC = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_ERLC),
+ AUDIO_FORMAT_AAC_LD = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_LD),
+ AUDIO_FORMAT_AAC_HE_V2 = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_HE_V2),
+ AUDIO_FORMAT_AAC_ELD = (AUDIO_FORMAT_AAC |
+ AUDIO_FORMAT_AAC_SUB_ELD)
+};
+
+/*
+ * Usage of these values highlights places in the code that use 2- or 8- channel
+ * assumptions.
+ */
+enum FixedChannelCount {
+ FCC_2 = 2, // This is typically due to legacy implementation of stereo I/O
+ FCC_8 = 8 // This is typically due to audio mixer and resampler limitations
+};
+
+/*
+ * A channel mask per se only defines the presence or absence of a channel, not
+ * the order. See AUDIO_INTERLEAVE_* for the platform convention of order.
+ *
+ * AudioChannelMask is an opaque type and its internal layout should not be
+ * assumed as it may change in the future. Instead, always use functions
+ * to examine it.
+ *
+ * These are the current representations:
+ *
+ * AUDIO_CHANNEL_REPRESENTATION_POSITION
+ * is a channel mask representation for position assignment. Each low-order
+ * bit corresponds to the spatial position of a transducer (output), or
+ * interpretation of channel (input). The user of a channel mask needs to
+ * know the context of whether it is for output or input. The constants
+ * AUDIO_CHANNEL_OUT_* or AUDIO_CHANNEL_IN_* apply to the bits portion. It
+ * is not permitted for no bits to be set.
+ *
+ * AUDIO_CHANNEL_REPRESENTATION_INDEX
+ * is a channel mask representation for index assignment. Each low-order
+ * bit corresponds to a selected channel. There is no platform
+ * interpretation of the various bits. There is no concept of output or
+ * input. It is not permitted for no bits to be set.
+ *
+ * All other representations are reserved for future use.
+ *
+ * Warning: current representation distinguishes between input and output, but
+ * this will not the be case in future revisions of the platform. Wherever there
+ * is an ambiguity between input and output that is currently resolved by
+ * checking the channel mask, the implementer should look for ways to fix it
+ * with additional information outside of the mask.
+ */
+enum AudioChannelMask : uint32_t {
+ AUDIO_CHANNEL_REPRESENTATION_POSITION = 0, /* must be 0 for compatibility */
+ /* 1 is reserved for future use */
+ AUDIO_CHANNEL_REPRESENTATION_INDEX = 2,
+ /* 3 is reserved for future use */
+
+ /* These can be a complete value of AudioChannelMask */
+ AUDIO_CHANNEL_NONE = 0x0,
+ AUDIO_CHANNEL_INVALID = 0xC0000000,
+
+ /*
+ * These can be the bits portion of an AudioChannelMask
+ * with representation AUDIO_CHANNEL_REPRESENTATION_POSITION.
+ */
+
+ /* output channels */
+ AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1,
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2,
+ AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4,
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8,
+ AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10,
+ AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20,
+ AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40,
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80,
+ AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100,
+ AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200,
+ AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400,
+ AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800,
+ AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000,
+ AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000,
+ AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000,
+ AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000,
+ AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000,
+ AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000,
+
+ AUDIO_CHANNEL_OUT_MONO = AUDIO_CHANNEL_OUT_FRONT_LEFT,
+ AUDIO_CHANNEL_OUT_STEREO = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT),
+ AUDIO_CHANNEL_OUT_QUAD = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+ AUDIO_CHANNEL_OUT_BACK_LEFT |
+ AUDIO_CHANNEL_OUT_BACK_RIGHT),
+ AUDIO_CHANNEL_OUT_QUAD_BACK = AUDIO_CHANNEL_OUT_QUAD,
+ /* like AUDIO_CHANNEL_OUT_QUAD_BACK with *_SIDE_* instead of *_BACK_* */
+ AUDIO_CHANNEL_OUT_QUAD_SIDE = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+ AUDIO_CHANNEL_OUT_SIDE_LEFT |
+ AUDIO_CHANNEL_OUT_SIDE_RIGHT),
+ AUDIO_CHANNEL_OUT_5POINT1 = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+ AUDIO_CHANNEL_OUT_FRONT_CENTER |
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+ AUDIO_CHANNEL_OUT_BACK_LEFT |
+ AUDIO_CHANNEL_OUT_BACK_RIGHT),
+ AUDIO_CHANNEL_OUT_5POINT1_BACK = AUDIO_CHANNEL_OUT_5POINT1,
+ /* like AUDIO_CHANNEL_OUT_5POINT1_BACK with *_SIDE_* instead of *_BACK_* */
+ AUDIO_CHANNEL_OUT_5POINT1_SIDE = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+ AUDIO_CHANNEL_OUT_FRONT_CENTER |
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+ AUDIO_CHANNEL_OUT_SIDE_LEFT |
+ AUDIO_CHANNEL_OUT_SIDE_RIGHT),
+ /* matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND */
+ AUDIO_CHANNEL_OUT_7POINT1 = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+ AUDIO_CHANNEL_OUT_FRONT_CENTER |
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+ AUDIO_CHANNEL_OUT_BACK_LEFT |
+ AUDIO_CHANNEL_OUT_BACK_RIGHT |
+ AUDIO_CHANNEL_OUT_SIDE_LEFT |
+ AUDIO_CHANNEL_OUT_SIDE_RIGHT),
+ AUDIO_CHANNEL_OUT_ALL = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+ AUDIO_CHANNEL_OUT_FRONT_CENTER |
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+ AUDIO_CHANNEL_OUT_BACK_LEFT |
+ AUDIO_CHANNEL_OUT_BACK_RIGHT |
+ AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER |
+ AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER |
+ AUDIO_CHANNEL_OUT_BACK_CENTER|
+ AUDIO_CHANNEL_OUT_SIDE_LEFT|
+ AUDIO_CHANNEL_OUT_SIDE_RIGHT|
+ AUDIO_CHANNEL_OUT_TOP_CENTER|
+ AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT|
+ AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER|
+ AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT|
+ AUDIO_CHANNEL_OUT_TOP_BACK_LEFT|
+ AUDIO_CHANNEL_OUT_TOP_BACK_CENTER|
+ AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT),
+
+ /* These are bits only, not complete values */
+
+ /* input channels */
+ AUDIO_CHANNEL_IN_LEFT = 0x4,
+ AUDIO_CHANNEL_IN_RIGHT = 0x8,
+ AUDIO_CHANNEL_IN_FRONT = 0x10,
+ AUDIO_CHANNEL_IN_BACK = 0x20,
+ AUDIO_CHANNEL_IN_LEFT_PROCESSED = 0x40,
+ AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 0x80,
+ AUDIO_CHANNEL_IN_FRONT_PROCESSED = 0x100,
+ AUDIO_CHANNEL_IN_BACK_PROCESSED = 0x200,
+ AUDIO_CHANNEL_IN_PRESSURE = 0x400,
+ AUDIO_CHANNEL_IN_X_AXIS = 0x800,
+ AUDIO_CHANNEL_IN_Y_AXIS = 0x1000,
+ AUDIO_CHANNEL_IN_Z_AXIS = 0x2000,
+ AUDIO_CHANNEL_IN_VOICE_UPLINK = 0x4000,
+ AUDIO_CHANNEL_IN_VOICE_DNLINK = 0x8000,
+
+ AUDIO_CHANNEL_IN_MONO = AUDIO_CHANNEL_IN_FRONT,
+ AUDIO_CHANNEL_IN_STEREO = (AUDIO_CHANNEL_IN_LEFT | AUDIO_CHANNEL_IN_RIGHT),
+ AUDIO_CHANNEL_IN_FRONT_BACK = (AUDIO_CHANNEL_IN_FRONT |
+ AUDIO_CHANNEL_IN_BACK),
+ AUDIO_CHANNEL_IN_ALL = (AUDIO_CHANNEL_IN_LEFT |
+ AUDIO_CHANNEL_IN_RIGHT |
+ AUDIO_CHANNEL_IN_FRONT |
+ AUDIO_CHANNEL_IN_BACK|
+ AUDIO_CHANNEL_IN_LEFT_PROCESSED |
+ AUDIO_CHANNEL_IN_RIGHT_PROCESSED |
+ AUDIO_CHANNEL_IN_FRONT_PROCESSED |
+ AUDIO_CHANNEL_IN_BACK_PROCESSED|
+ AUDIO_CHANNEL_IN_PRESSURE |
+ AUDIO_CHANNEL_IN_X_AXIS |
+ AUDIO_CHANNEL_IN_Y_AXIS |
+ AUDIO_CHANNEL_IN_Z_AXIS |
+ AUDIO_CHANNEL_IN_VOICE_UPLINK |
+ AUDIO_CHANNEL_IN_VOICE_DNLINK),
+
+ /*
+ * Expresses the convention when stereo audio samples are stored interleaved
+ * in an array. This should improve readability by allowing code to use
+ * symbolic indices instead of hard-coded [0] and [1].
+ *
+ * For multi-channel beyond stereo, the platform convention is that channels
+ * are interleaved in order from least significant channel mask bit to most
+ * significant channel mask bit, with unused bits skipped. Any exceptions
+ * to this convention will be noted at the appropriate API.
+ */
+ AUDIO_INTERLEAVE_LEFT = 0,
+ AUDIO_INTERLEAVE_RIGHT = 1,
+
+ AUDIO_CHANNEL_COUNT_MAX = 30,
+ AUDIO_CHANNEL_INDEX_HDR =
+ AUDIO_CHANNEL_REPRESENTATION_INDEX << AUDIO_CHANNEL_COUNT_MAX,
+ AUDIO_CHANNEL_INDEX_MASK_1 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 1) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_2 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 2) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_3 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 3) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_4 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 4) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_5 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 5) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_6 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 6) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_7 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 7) - 1),
+ AUDIO_CHANNEL_INDEX_MASK_8 = AUDIO_CHANNEL_INDEX_HDR | ((1 << 8) - 1)
+};
+
+/*
+ * Major modes for a mobile device. The current mode setting affects audio
+ * routing.
+ */
+enum AudioMode {
+ AUDIO_MODE_INVALID = -2,
+ AUDIO_MODE_CURRENT = -1,
+ AUDIO_MODE_NORMAL = 0,
+ AUDIO_MODE_RINGTONE = 1,
+ AUDIO_MODE_IN_CALL = 2,
+ AUDIO_MODE_IN_COMMUNICATION = 3,
+
+ AUDIO_MODE_CNT,
+ AUDIO_MODE_MAX = AUDIO_MODE_CNT - 1,
+};
+
+enum AudioDevice {
+ AUDIO_DEVICE_NONE = 0x0,
+ /* reserved bits */
+ AUDIO_DEVICE_BIT_IN = 0x80000000,
+ AUDIO_DEVICE_BIT_DEFAULT = 0x40000000,
+ /* output devices */
+ AUDIO_DEVICE_OUT_EARPIECE = 0x1,
+ AUDIO_DEVICE_OUT_SPEAKER = 0x2,
+ AUDIO_DEVICE_OUT_WIRED_HEADSET = 0x4,
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 0x8,
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 0x10,
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
+ AUDIO_DEVICE_OUT_AUX_DIGITAL = 0x400,
+ AUDIO_DEVICE_OUT_HDMI = AUDIO_DEVICE_OUT_AUX_DIGITAL,
+ /* uses an analog connection (multiplexed over the USB pins for instance) */
+ AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800,
+ AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,
+ /* USB accessory mode: Android device is USB device and dock is USB host */
+ AUDIO_DEVICE_OUT_USB_ACCESSORY = 0x2000,
+ /* USB host mode: Android device is USB host and dock is USB device */
+ AUDIO_DEVICE_OUT_USB_DEVICE = 0x4000,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX = 0x8000,
+ /* Telephony voice TX path */
+ AUDIO_DEVICE_OUT_TELEPHONY_TX = 0x10000,
+ /* Analog jack with line impedance detected */
+ AUDIO_DEVICE_OUT_LINE = 0x20000,
+ /* HDMI Audio Return Channel */
+ AUDIO_DEVICE_OUT_HDMI_ARC = 0x40000,
+ /* S/PDIF out */
+ AUDIO_DEVICE_OUT_SPDIF = 0x80000,
+ /* FM transmitter out */
+ AUDIO_DEVICE_OUT_FM = 0x100000,
+ /* Line out for av devices */
+ AUDIO_DEVICE_OUT_AUX_LINE = 0x200000,
+ /* limited-output speaker device for acoustic safety */
+ AUDIO_DEVICE_OUT_SPEAKER_SAFE = 0x400000,
+ AUDIO_DEVICE_OUT_IP = 0x800000,
+ /* audio bus implemented by the audio system (e.g an MOST stereo channel) */
+ AUDIO_DEVICE_OUT_BUS = 0x1000000,
+ AUDIO_DEVICE_OUT_DEFAULT = AUDIO_DEVICE_BIT_DEFAULT,
+ AUDIO_DEVICE_OUT_ALL = (AUDIO_DEVICE_OUT_EARPIECE |
+ AUDIO_DEVICE_OUT_SPEAKER |
+ AUDIO_DEVICE_OUT_WIRED_HEADSET |
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO |
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER |
+ AUDIO_DEVICE_OUT_HDMI |
+ AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
+ AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET |
+ AUDIO_DEVICE_OUT_USB_ACCESSORY |
+ AUDIO_DEVICE_OUT_USB_DEVICE |
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX |
+ AUDIO_DEVICE_OUT_TELEPHONY_TX |
+ AUDIO_DEVICE_OUT_LINE |
+ AUDIO_DEVICE_OUT_HDMI_ARC |
+ AUDIO_DEVICE_OUT_SPDIF |
+ AUDIO_DEVICE_OUT_FM |
+ AUDIO_DEVICE_OUT_AUX_LINE |
+ AUDIO_DEVICE_OUT_SPEAKER_SAFE |
+ AUDIO_DEVICE_OUT_IP |
+ AUDIO_DEVICE_OUT_BUS |
+ AUDIO_DEVICE_OUT_DEFAULT),
+ AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+ AUDIO_DEVICE_OUT_ALL_SCO = (AUDIO_DEVICE_OUT_BLUETOOTH_SCO |
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+ AUDIO_DEVICE_OUT_ALL_USB = (AUDIO_DEVICE_OUT_USB_ACCESSORY |
+ AUDIO_DEVICE_OUT_USB_DEVICE),
+ /* input devices */
+ AUDIO_DEVICE_IN_COMMUNICATION = AUDIO_DEVICE_BIT_IN | 0x1,
+ AUDIO_DEVICE_IN_AMBIENT = AUDIO_DEVICE_BIT_IN | 0x2,
+ AUDIO_DEVICE_IN_BUILTIN_MIC = AUDIO_DEVICE_BIT_IN | 0x4,
+ AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = AUDIO_DEVICE_BIT_IN | 0x8,
+ AUDIO_DEVICE_IN_WIRED_HEADSET = AUDIO_DEVICE_BIT_IN | 0x10,
+ AUDIO_DEVICE_IN_AUX_DIGITAL = AUDIO_DEVICE_BIT_IN | 0x20,
+ AUDIO_DEVICE_IN_HDMI = AUDIO_DEVICE_IN_AUX_DIGITAL,
+ /* Telephony voice RX path */
+ AUDIO_DEVICE_IN_VOICE_CALL = AUDIO_DEVICE_BIT_IN | 0x40,
+ AUDIO_DEVICE_IN_TELEPHONY_RX = AUDIO_DEVICE_IN_VOICE_CALL,
+ AUDIO_DEVICE_IN_BACK_MIC = AUDIO_DEVICE_BIT_IN | 0x80,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX = AUDIO_DEVICE_BIT_IN | 0x100,
+ AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET = AUDIO_DEVICE_BIT_IN | 0x200,
+ AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET = AUDIO_DEVICE_BIT_IN | 0x400,
+ AUDIO_DEVICE_IN_USB_ACCESSORY = AUDIO_DEVICE_BIT_IN | 0x800,
+ AUDIO_DEVICE_IN_USB_DEVICE = AUDIO_DEVICE_BIT_IN | 0x1000,
+ /* FM tuner input */
+ AUDIO_DEVICE_IN_FM_TUNER = AUDIO_DEVICE_BIT_IN | 0x2000,
+ /* TV tuner input */
+ AUDIO_DEVICE_IN_TV_TUNER = AUDIO_DEVICE_BIT_IN | 0x4000,
+ /* Analog jack with line impedance detected */
+ AUDIO_DEVICE_IN_LINE = AUDIO_DEVICE_BIT_IN | 0x8000,
+ /* S/PDIF in */
+ AUDIO_DEVICE_IN_SPDIF = AUDIO_DEVICE_BIT_IN | 0x10000,
+ AUDIO_DEVICE_IN_BLUETOOTH_A2DP = AUDIO_DEVICE_BIT_IN | 0x20000,
+ AUDIO_DEVICE_IN_LOOPBACK = AUDIO_DEVICE_BIT_IN | 0x40000,
+ AUDIO_DEVICE_IN_IP = AUDIO_DEVICE_BIT_IN | 0x80000,
+ /* audio bus implemented by the audio system (e.g an MOST stereo channel) */
+ AUDIO_DEVICE_IN_BUS = AUDIO_DEVICE_BIT_IN | 0x100000,
+ AUDIO_DEVICE_IN_DEFAULT = AUDIO_DEVICE_BIT_IN |
+ AUDIO_DEVICE_BIT_DEFAULT,
+
+ AUDIO_DEVICE_IN_ALL = (AUDIO_DEVICE_IN_COMMUNICATION |
+ AUDIO_DEVICE_IN_AMBIENT |
+ AUDIO_DEVICE_IN_BUILTIN_MIC |
+ AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET |
+ AUDIO_DEVICE_IN_WIRED_HEADSET |
+ AUDIO_DEVICE_IN_HDMI |
+ AUDIO_DEVICE_IN_TELEPHONY_RX |
+ AUDIO_DEVICE_IN_BACK_MIC |
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX |
+ AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET |
+ AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET |
+ AUDIO_DEVICE_IN_USB_ACCESSORY |
+ AUDIO_DEVICE_IN_USB_DEVICE |
+ AUDIO_DEVICE_IN_FM_TUNER |
+ AUDIO_DEVICE_IN_TV_TUNER |
+ AUDIO_DEVICE_IN_LINE |
+ AUDIO_DEVICE_IN_SPDIF |
+ AUDIO_DEVICE_IN_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_IN_LOOPBACK |
+ AUDIO_DEVICE_IN_IP |
+ AUDIO_DEVICE_IN_BUS |
+ AUDIO_DEVICE_IN_DEFAULT),
+ AUDIO_DEVICE_IN_ALL_SCO = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET,
+ AUDIO_DEVICE_IN_ALL_USB = (AUDIO_DEVICE_IN_USB_ACCESSORY |
+ AUDIO_DEVICE_IN_USB_DEVICE),
+};
+
+/*
+ * The audio output flags serve two purposes:
+ *
+ * - when an AudioTrack is created they indicate a "wish" to be connected to an
+ * output stream with attributes corresponding to the specified flags;
+ *
+ * - when present in an output profile descriptor listed for a particular audio
+ * hardware module, they indicate that an output stream can be opened that
+ * supports the attributes indicated by the flags.
+ *
+ * The audio policy manager will try to match the flags in the request
+ * (when getOuput() is called) to an available output stream.
+ */
+enum AudioOutputFlag {
+ AUDIO_OUTPUT_FLAG_NONE = 0x0, // no attributes
+ AUDIO_OUTPUT_FLAG_DIRECT = 0x1, // this output directly connects a track
+ // to one output stream: no software mixer
+ AUDIO_OUTPUT_FLAG_PRIMARY = 0x2, // this output is the primary output of
+ // the device. It is unique and must be
+ // present. It is opened by default and
+ // receives routing, audio mode and volume
+ // controls related to voice calls.
+ AUDIO_OUTPUT_FLAG_FAST = 0x4, // output supports "fast tracks",
+ // defined elsewhere
+ AUDIO_OUTPUT_FLAG_DEEP_BUFFER = 0x8, // use deep audio buffers
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD = 0x10, // offload playback of compressed
+ // streams to hardware codec
+ AUDIO_OUTPUT_FLAG_NON_BLOCKING = 0x20, // use non-blocking write
+ AUDIO_OUTPUT_FLAG_HW_AV_SYNC = 0x40, // output uses a hardware A/V sync
+ AUDIO_OUTPUT_FLAG_TTS = 0x80, // output for streams transmitted through
+ // speaker at a sample rate high enough
+ // to accommodate lower-range ultrasonic p/b
+ AUDIO_OUTPUT_FLAG_RAW = 0x100, // minimize signal processing
+ AUDIO_OUTPUT_FLAG_SYNC = 0x200, // synchronize I/O streams
+ AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO = 0x400, // Audio stream contains
+ // compressed audio in
+ // SPDIF data bursts, not PCM.
+};
+
+/*
+ * The audio input flags are analogous to audio output flags.
+ * Currently they are used only when an AudioRecord is created,
+ * to indicate a preference to be connected to an input stream with
+ * attributes corresponding to the specified flags.
+ */
+enum AudioInputFlags {
+ AUDIO_INPUT_FLAG_NONE = 0x0, // no attributes
+ AUDIO_INPUT_FLAG_FAST = 0x1, // prefer an input that supports
+ // "fast tracks"
+ AUDIO_INPUT_FLAG_HW_HOTWORD = 0x2, // prefer an input that captures from
+ // hw hotword source
+ AUDIO_INPUT_FLAG_RAW = 0x4, // minimize signal processing
+ AUDIO_INPUT_FLAG_SYNC = 0x8, // synchronize I/O streams
+};
+
+/*
+ * Additional information about the stream passed to hardware decoders.
+ */
+struct AudioOffloadInfo {
+ uint32_t sampleRateHz;
+ AudioChannelMask channelMask;
+ AudioFormat format;
+ AudioStreamType streamType;
+ uint32_t bitRatePerSecond;
+ int64_t durationMicroseconds; // -1 if unknown
+ bool hasVideo;
+ bool isStreaming;
+};
+
+/*
+ * Commonly used audio stream configuration parameters.
+ */
+struct AudioConfig {
+ uint32_t sampleRateHz;
+ AudioChannelMask channelMask;
+ AudioFormat format;
+ AudioOffloadInfo offloadInfo;
+ uint64_t frameCount;
+};
+
+
+/*
+ *
+ * Volume control
+ *
+ */
+
+/*
+ * Type of gain control exposed by an audio port.
+ */
+enum AudioGainMode : uint32_t {
+ AUDIO_GAIN_MODE_JOINT = 0x1, // supports joint channel gain control
+ AUDIO_GAIN_MODE_CHANNELS = 0x2, // supports separate channel gain control
+ AUDIO_GAIN_MODE_RAMP = 0x4 // supports gain ramps
+};
+
+/*
+ * An audio_gain struct is a representation of a gain stage.
+ * A gain stage is always attached to an audio port.
+ */
+struct AudioGain {
+ AudioGainMode mode;
+ AudioChannelMask channelMask; // channels which gain an be controlled
+ int32_t minValue; // minimum gain value in millibels
+ int32_t maxValue; // maximum gain value in millibels
+ int32_t defaultValue; // default gain value in millibels
+ uint32_t stepValue; // gain step in millibels
+ uint32_t minRampMs; // minimum ramp duration in ms
+ uint32_t maxRampMs; // maximum ramp duration in ms
+};
+
+/*
+ * The gain configuration structure is used to get or set the gain values of a
+ * given port.
+ */
+struct AudioGainConfig {
+ int32_t index; // index of the corresponding AudioGain in AudioPort.gains
+ AudioGainMode mode;
+ AudioChannelMask channelMask; // channels which gain value follows
+ /*
+ * 4 = sizeof(AudioChannelMask),
+ * 8 is not "FCC_8", so it won't need to be changed for > 8 channels.
+ * Gain values in millibels for each channel ordered from LSb to MSb in
+ * channel mask. The number of values is 1 in joint mode or
+ * popcount(channel_mask).
+ */
+ int32_t[4 * 8] values;
+ uint32_t rampDurationMs; // ramp duration in ms
+};
+
+
+/*
+ *
+ * Routing control
+ *
+ */
+
+/*
+ * Types defined here are used to describe an audio source or sink at internal
+ * framework interfaces (audio policy, patch panel) or at the audio HAL.
+ * Sink and sources are grouped in a concept of “audio port” representing an
+ * audio end point at the edge of the system managed by the module exposing
+ * the interface.
+ */
+
+/* Audio port role: either source or sink */
+enum AudioPortRole {
+ AUDIO_PORT_ROLE_NONE,
+ AUDIO_PORT_ROLE_SOURCE,
+ AUDIO_PORT_ROLE_SINK,
+};
+
+/*
+ * Audio port type indicates if it is a session (e.g AudioTrack),
+ * a mix (e.g PlaybackThread output) or a physical device
+ * (e.g AUDIO_DEVICE_OUT_SPEAKER)
+ */
+enum AudioPortType {
+ AUDIO_PORT_TYPE_NONE,
+ AUDIO_PORT_TYPE_DEVICE,
+ AUDIO_PORT_TYPE_MIX,
+ AUDIO_PORT_TYPE_SESSION,
+};
+
+/*
+ * Extension for audio port configuration structure when the audio port is a
+ * hardware device.
+ */
+struct AudioPortConfigDeviceExt {
+ AudioModuleHandle hwModule; // module the device is attached to
+ AudioDevice type; // device type (e.g AUDIO_DEVICE_OUT_SPEAKER)
+ uint8_t[32] address; // device address. "" if N/A
+};
+
+/*
+ * Extension for audio port configuration structure when the audio port is an
+ * audio session.
+ */
+struct AudioPortConfigSessionExt {
+ AudioSession session;
+};
+
+/*
+ * Flags indicating which fields are to be considered in AudioPortConfig.
+ */
+enum AudioPortConfigMask : uint32_t {
+ AUDIO_PORT_CONFIG_SAMPLE_RATE = 0x1,
+ AUDIO_PORT_CONFIG_CHANNEL_MASK = 0x2,
+ AUDIO_PORT_CONFIG_FORMAT = 0x4,
+ AUDIO_PORT_CONFIG_GAIN = 0x8,
+ AUDIO_PORT_CONFIG_ALL = AUDIO_PORT_CONFIG_SAMPLE_RATE |
+ AUDIO_PORT_CONFIG_CHANNEL_MASK |
+ AUDIO_PORT_CONFIG_FORMAT |
+ AUDIO_PORT_CONFIG_GAIN
+};
+
+/*
+ * Audio port configuration structure used to specify a particular configuration
+ * of an audio port.
+ */
+struct AudioPortConfig {
+ AudioPortHandle id;
+ AudioPortConfigMask configMask;
+ uint32_t sampleRateHz;
+ AudioChannelMask channelMask;
+ AudioFormat format;
+ AudioGainConfig gain;
+ AudioPortType type; // type is used as a discriminator for Ext union
+ AudioPortRole role; // role is used as a discriminator for UseCase union
+ union Ext {
+ AudioPortConfigDeviceExt device;
+ struct AudioPortConfigMixExt {
+ AudioModuleHandle hwModule; // module the stream is attached to
+ AudioIoHandle ioHandle; // I/O handle of the input/output stream
+ union UseCase {
+ AudioStreamType stream;
+ AudioSource source;
+ };
+ } mix;
+ AudioPortConfigSessionExt session;
+ };
+};
+
+/*
+ * Extension for audio port structure when the audio port is a hardware device.
+ */
+struct AudioPortDeviceExt {
+ AudioModuleHandle hwModule; // module the device is attached to
+ AudioDevice type;
+ uint8_t[32] address;
+};
+
+/*
+ * Latency class of the audio mix.
+ */
+enum AudioMixLatencyClass {
+ AUDIO_LATENCY_LOW,
+ AUDIO_LATENCY_NORMAL
+} ;
+
+struct AudioPortMixExt {
+ AudioModuleHandle hwModule; // module the stream is attached to
+ AudioIoHandle ioHandle; // I/O handle of the stream
+ AudioMixLatencyClass latencyClass;
+};
+
+/*
+ * Extension for audio port structure when the audio port is an audio session.
+ */
+struct AudioPortSessionExt {
+ AudioSession session;
+};
+
+struct AudioPort {
+ AudioPortHandle id;
+ AudioPortRole role;
+ string name;
+ vec<uint32_t> sampleRates;
+ vec<AudioChannelMask> channelMasks;
+ vec<AudioFormat> formats;
+ vec<AudioGain> gains;
+ AudioPortConfig activeConfig; // current audio port configuration
+ AudioPortType type; // type is used as a discriminator
+ union Ext {
+ AudioPortDeviceExt device;
+ AudioPortMixExt mix;
+ AudioPortSessionExt session;
+ };
+};
+
+/*
+ * An audio patch represents a connection between one or more source ports and
+ * one or more sink ports. Patches are connected and disconnected by audio
+ * policy manager or by applications via framework APIs.
+ */
+struct AudioPatch {
+ AudioPatchHandle id;
+ vec<AudioPortConfig> sources;
+ vec<AudioPortConfig> sinks;
+};
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