Introduce Audio V6
Generate with ./newHal.sh 5.0
+ some typo fix in the .hal
+ some clang-tidy run
This new HAL is an exact copy of the V5.
It will be modified in following patches while R is developed.
Test: push V6 to device and check audio works fine
Bug: 134940862
Change-Id: Ic25d30175032e76e0d3208fb876451de9ae6b7b4
Merged-In: Ic25d30175032e76e0d3208fb876451de9ae6b7b4
Signed-off-by: Kevin Rocard <krocard@google.com>
diff --git a/audio/common/6.0/Android.bp b/audio/common/6.0/Android.bp
new file mode 100644
index 0000000..1a4e054
--- /dev/null
+++ b/audio/common/6.0/Android.bp
@@ -0,0 +1,17 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+ name: "android.hardware.audio.common@6.0",
+ root: "android.hardware",
+ vndk: {
+ enabled: true,
+ },
+ srcs: [
+ "types.hal",
+ ],
+ interfaces: [
+ "android.hidl.safe_union@1.0",
+ ],
+ gen_java: false,
+ gen_java_constants: true,
+}
diff --git a/audio/common/6.0/types.hal b/audio/common/6.0/types.hal
new file mode 100644
index 0000000..132f86d
--- /dev/null
+++ b/audio/common/6.0/types.hal
@@ -0,0 +1,1032 @@
+/*
+ * Copyright (C) 2019 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@6.0;
+
+import android.hidl.safe_union@1.0;
+
+/*
+ *
+ * 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.
+ */
+@export(name="")
+enum AudioHandleConsts : int32_t {
+ AUDIO_IO_HANDLE_NONE = 0,
+ AUDIO_MODULE_HANDLE_NONE = 0,
+ AUDIO_PORT_HANDLE_NONE = 0,
+ AUDIO_PATCH_HANDLE_NONE = 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 intended use case of a stream.
+ */
+@export(name="audio_stream_type_t", value_prefix="AUDIO_STREAM_")
+enum AudioStreamType : int32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioSystem.java
+ DEFAULT = -1,
+ MIN = 0,
+ VOICE_CALL = 0,
+ SYSTEM = 1,
+ RING = 2,
+ MUSIC = 3,
+ ALARM = 4,
+ NOTIFICATION = 5,
+ BLUETOOTH_SCO = 6,
+ ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be
+ // routed to speaker
+ DTMF = 8,
+ TTS = 9, // Transmitted Through Speaker. Plays over speaker
+ // only, silent on other devices
+ ACCESSIBILITY = 10, // For accessibility talk back prompts
+};
+
+@export(name="audio_source_t", value_prefix="AUDIO_SOURCE_")
+enum AudioSource : int32_t {
+ // 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
+ DEFAULT = 0,
+ MIC = 1,
+ VOICE_UPLINK = 2,
+ VOICE_DOWNLINK = 3,
+ VOICE_CALL = 4,
+ CAMCORDER = 5,
+ VOICE_RECOGNITION = 6,
+ 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.
+ */
+ REMOTE_SUBMIX = 8,
+ /**
+ * Source for unprocessed sound. Usage examples include level measurement
+ * and raw signal analysis.
+ */
+ UNPROCESSED = 9,
+ /**
+ * Source for capturing audio meant to be processed in real time and played back for live
+ * performance (e.g karaoke). The capture path will minimize latency and coupling with
+ * playback path.
+ */
+ VOICE_PERFORMANCE = 10,
+ /**
+ * Source for an echo canceller to capture the reference signal to be cancelled.
+ * The echo reference signal will be captured as close as possible to the DAC in order
+ * to include all post processing applied to the playback path.
+ */
+ ECHO_REFERENCE = 1997,
+ FM_TUNER = 1998,
+ HOTWORD = 1999,
+};
+
+typedef int32_t AudioSession;
+/**
+ * Special audio session values.
+ */
+@export(name="audio_session_t", value_prefix="AUDIO_SESSION_")
+enum AudioSessionConsts : int32_t {
+ /**
+ * Session for effects attached to a particular output stream
+ * (value must be less than 0)
+ */
+ 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)
+ */
+ OUTPUT_MIX = 0,
+ /**
+ * Application does not specify an explicit session ID to be used, and
+ * requests a new session ID to be allocated. Corresponds to
+ * AudioManager.AUDIO_SESSION_ID_GENERATE and
+ * AudioSystem.AUDIO_SESSION_ALLOCATE.
+ */
+ 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.
+ */
+ 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).
+ */
+@export(name="audio_format_t", value_prefix="AUDIO_FORMAT_")
+enum AudioFormat : uint32_t {
+ INVALID = 0xFFFFFFFFUL,
+ DEFAULT = 0,
+ PCM = 0x00000000UL,
+ MP3 = 0x01000000UL,
+ AMR_NB = 0x02000000UL,
+ AMR_WB = 0x03000000UL,
+ AAC = 0x04000000UL,
+ /** Deprecated, Use AAC_HE_V1 */
+ HE_AAC_V1 = 0x05000000UL,
+ /** Deprecated, Use AAC_HE_V2 */
+ HE_AAC_V2 = 0x06000000UL,
+ VORBIS = 0x07000000UL,
+ OPUS = 0x08000000UL,
+ AC3 = 0x09000000UL,
+ E_AC3 = 0x0A000000UL,
+ DTS = 0x0B000000UL,
+ DTS_HD = 0x0C000000UL,
+ /** IEC61937 is encoded audio wrapped in 16-bit PCM. */
+ IEC61937 = 0x0D000000UL,
+ DOLBY_TRUEHD = 0x0E000000UL,
+ EVRC = 0x10000000UL,
+ EVRCB = 0x11000000UL,
+ EVRCWB = 0x12000000UL,
+ EVRCNW = 0x13000000UL,
+ AAC_ADIF = 0x14000000UL,
+ WMA = 0x15000000UL,
+ WMA_PRO = 0x16000000UL,
+ AMR_WB_PLUS = 0x17000000UL,
+ MP2 = 0x18000000UL,
+ QCELP = 0x19000000UL,
+ DSD = 0x1A000000UL,
+ FLAC = 0x1B000000UL,
+ ALAC = 0x1C000000UL,
+ APE = 0x1D000000UL,
+ AAC_ADTS = 0x1E000000UL,
+ SBC = 0x1F000000UL,
+ APTX = 0x20000000UL,
+ APTX_HD = 0x21000000UL,
+ AC4 = 0x22000000UL,
+ LDAC = 0x23000000UL,
+ /** Dolby Metadata-enhanced Audio Transmission */
+ MAT = 0x24000000UL,
+ AAC_LATM = 0x25000000UL,
+ CELT = 0x26000000UL,
+ APTX_ADAPTIVE = 0x27000000UL,
+ LHDC = 0x28000000UL,
+ LHDC_LL = 0x29000000UL,
+ APTX_TWSP = 0x2A000000UL,
+
+ /** Deprecated */
+ MAIN_MASK = 0xFF000000UL,
+ SUB_MASK = 0x00FFFFFFUL,
+
+ /* Subformats */
+ PCM_SUB_16_BIT = 0x1, // PCM signed 16 bits
+ PCM_SUB_8_BIT = 0x2, // PCM unsigned 8 bits
+ PCM_SUB_32_BIT = 0x3, // PCM signed .31 fixed point
+ PCM_SUB_8_24_BIT = 0x4, // PCM signed 8.23 fixed point
+ PCM_SUB_FLOAT = 0x5, // PCM single-precision float pt
+ PCM_SUB_24_BIT_PACKED = 0x6, // PCM signed .23 fix pt (3 bytes)
+
+ MP3_SUB_NONE = 0x0,
+
+ AMR_SUB_NONE = 0x0,
+
+ AAC_SUB_MAIN = 0x1,
+ AAC_SUB_LC = 0x2,
+ AAC_SUB_SSR = 0x4,
+ AAC_SUB_LTP = 0x8,
+ AAC_SUB_HE_V1 = 0x10,
+ AAC_SUB_SCALABLE = 0x20,
+ AAC_SUB_ERLC = 0x40,
+ AAC_SUB_LD = 0x80,
+ AAC_SUB_HE_V2 = 0x100,
+ AAC_SUB_ELD = 0x200,
+ AAC_SUB_XHE = 0x300,
+
+ VORBIS_SUB_NONE = 0x0,
+
+ E_AC3_SUB_JOC = 0x1,
+
+ MAT_SUB_1_0 = 0x1,
+ MAT_SUB_2_0 = 0x2,
+ MAT_SUB_2_1 = 0x3,
+
+ /* Aliases */
+ /** note != AudioFormat.ENCODING_PCM_16BIT */
+ PCM_16_BIT = (PCM | PCM_SUB_16_BIT),
+ /** note != AudioFormat.ENCODING_PCM_8BIT */
+ PCM_8_BIT = (PCM | PCM_SUB_8_BIT),
+ PCM_32_BIT = (PCM | PCM_SUB_32_BIT),
+ PCM_8_24_BIT = (PCM | PCM_SUB_8_24_BIT),
+ PCM_FLOAT = (PCM | PCM_SUB_FLOAT),
+ PCM_24_BIT_PACKED = (PCM | PCM_SUB_24_BIT_PACKED),
+ AAC_MAIN = (AAC | AAC_SUB_MAIN),
+ AAC_LC = (AAC | AAC_SUB_LC),
+ AAC_SSR = (AAC | AAC_SUB_SSR),
+ AAC_LTP = (AAC | AAC_SUB_LTP),
+ AAC_HE_V1 = (AAC | AAC_SUB_HE_V1),
+ AAC_SCALABLE = (AAC | AAC_SUB_SCALABLE),
+ AAC_ERLC = (AAC | AAC_SUB_ERLC),
+ AAC_LD = (AAC | AAC_SUB_LD),
+ AAC_HE_V2 = (AAC | AAC_SUB_HE_V2),
+ AAC_ELD = (AAC | AAC_SUB_ELD),
+ AAC_XHE = (AAC | AAC_SUB_XHE),
+ AAC_ADTS_MAIN = (AAC_ADTS | AAC_SUB_MAIN),
+ AAC_ADTS_LC = (AAC_ADTS | AAC_SUB_LC),
+ AAC_ADTS_SSR = (AAC_ADTS | AAC_SUB_SSR),
+ AAC_ADTS_LTP = (AAC_ADTS | AAC_SUB_LTP),
+ AAC_ADTS_HE_V1 = (AAC_ADTS | AAC_SUB_HE_V1),
+ AAC_ADTS_SCALABLE = (AAC_ADTS | AAC_SUB_SCALABLE),
+ AAC_ADTS_ERLC = (AAC_ADTS | AAC_SUB_ERLC),
+ AAC_ADTS_LD = (AAC_ADTS | AAC_SUB_LD),
+ AAC_ADTS_HE_V2 = (AAC_ADTS | AAC_SUB_HE_V2),
+ AAC_ADTS_ELD = (AAC_ADTS | AAC_SUB_ELD),
+ AAC_ADTS_XHE = (AAC_ADTS | AAC_SUB_XHE),
+ E_AC3_JOC = (E_AC3 | E_AC3_SUB_JOC),
+ MAT_1_0 = (MAT | MAT_SUB_1_0),
+ MAT_2_0 = (MAT | MAT_SUB_2_0),
+ MAT_2_1 = (MAT | MAT_SUB_2_1),
+ AAC_LATM_LC = (AAC_LATM | AAC_SUB_LC),
+ AAC_LATM_HE_V1 = (AAC_LATM | AAC_SUB_HE_V1),
+ AAC_LATM_HE_V2 = (AAC_LATM | AAC_SUB_HE_V2),
+};
+
+/**
+ * Usage of these values highlights places in the code that use 2- or 8- channel
+ * assumptions.
+ */
+@export(name="")
+enum FixedChannelCount : int32_t {
+ 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.
+ *
+ * The channel order convention is that channels are interleaved in order from
+ * least significant channel mask bit to most significant channel mask bit,
+ * with unused bits skipped. For example for stereo, LEFT would be first,
+ * followed by RIGHT.
+ * Any exceptions to this convention are noted at the appropriate API.
+ *
+ * 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:
+ *
+ * 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
+ * OUT_* or IN_* apply to the bits portion. It is not permitted for no bits
+ * to be set.
+ *
+ * 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.
+ */
+@export(name="", value_prefix="AUDIO_CHANNEL_")
+enum AudioChannelMask : uint32_t {
+ /** must be 0 for compatibility */
+ REPRESENTATION_POSITION = 0,
+ /** 1 is reserved for future use */
+ REPRESENTATION_INDEX = 2,
+ /* 3 is reserved for future use */
+
+ /** These can be a complete value of AudioChannelMask */
+ NONE = 0x0,
+ INVALID = 0xC0000000,
+
+ /*
+ * These can be the bits portion of an AudioChannelMask
+ * with representation REPRESENTATION_POSITION.
+ */
+
+ /** output channels */
+ OUT_FRONT_LEFT = 0x1,
+ OUT_FRONT_RIGHT = 0x2,
+ OUT_FRONT_CENTER = 0x4,
+ OUT_LOW_FREQUENCY = 0x8,
+ OUT_BACK_LEFT = 0x10,
+ OUT_BACK_RIGHT = 0x20,
+ OUT_FRONT_LEFT_OF_CENTER = 0x40,
+ OUT_FRONT_RIGHT_OF_CENTER = 0x80,
+ OUT_BACK_CENTER = 0x100,
+ OUT_SIDE_LEFT = 0x200,
+ OUT_SIDE_RIGHT = 0x400,
+ OUT_TOP_CENTER = 0x800,
+ OUT_TOP_FRONT_LEFT = 0x1000,
+ OUT_TOP_FRONT_CENTER = 0x2000,
+ OUT_TOP_FRONT_RIGHT = 0x4000,
+ OUT_TOP_BACK_LEFT = 0x8000,
+ OUT_TOP_BACK_CENTER = 0x10000,
+ OUT_TOP_BACK_RIGHT = 0x20000,
+ OUT_TOP_SIDE_LEFT = 0x40000,
+ OUT_TOP_SIDE_RIGHT = 0x80000,
+
+ /**
+ * Haptic channel characteristics are specific to a device and
+ * only used to play device specific resources (eg: ringtones).
+ * The HAL can freely map A and B to haptic controllers, the
+ * framework shall not interpret those values and forward them
+ * from the device audio assets.
+ */
+ OUT_HAPTIC_A = 0x20000000,
+ OUT_HAPTIC_B = 0x10000000,
+
+ OUT_MONO = OUT_FRONT_LEFT,
+ OUT_STEREO = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT),
+ OUT_2POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_LOW_FREQUENCY),
+ OUT_2POINT0POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_2POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT |
+ OUT_LOW_FREQUENCY),
+ OUT_3POINT0POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_FRONT_CENTER |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_3POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_FRONT_CENTER |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT |
+ OUT_LOW_FREQUENCY),
+ OUT_QUAD = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT),
+ OUT_QUAD_BACK = OUT_QUAD,
+ /** like OUT_QUAD_BACK with *_SIDE_* instead of *_BACK_* */
+ OUT_QUAD_SIDE = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
+ OUT_SURROUND = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_BACK_CENTER),
+ OUT_PENTA = (OUT_QUAD | OUT_FRONT_CENTER),
+ OUT_5POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT),
+ OUT_5POINT1_BACK = OUT_5POINT1,
+ /** like OUT_5POINT1_BACK with *_SIDE_* instead of *_BACK_* */
+ OUT_5POINT1_SIDE = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
+ OUT_5POINT1POINT2 = (OUT_5POINT1 | OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_5POINT1POINT4 = (OUT_5POINT1 |
+ OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_RIGHT |
+ OUT_TOP_BACK_LEFT | OUT_TOP_BACK_RIGHT),
+ OUT_6POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT |
+ OUT_BACK_CENTER),
+ /** matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND */
+ OUT_7POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT |
+ OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
+ OUT_7POINT1POINT2 = (OUT_7POINT1 | OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_7POINT1POINT4 = (OUT_7POINT1 |
+ OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_RIGHT |
+ OUT_TOP_BACK_LEFT | OUT_TOP_BACK_RIGHT),
+ OUT_MONO_HAPTIC_A = (OUT_FRONT_LEFT | OUT_HAPTIC_A),
+ OUT_STEREO_HAPTIC_A = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_HAPTIC_A),
+ OUT_HAPTIC_AB = (OUT_HAPTIC_A | OUT_HAPTIC_B),
+ OUT_MONO_HAPTIC_AB = (OUT_FRONT_LEFT | OUT_HAPTIC_A | OUT_HAPTIC_B),
+ OUT_STEREO_HAPTIC_AB = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_HAPTIC_A | OUT_HAPTIC_B),
+ // Note that the 2.0 OUT_ALL* have been moved to helper functions
+
+ /* These are bits only, not complete values */
+
+ /** input channels */
+ IN_LEFT = 0x4,
+ IN_RIGHT = 0x8,
+ IN_FRONT = 0x10,
+ IN_BACK = 0x20,
+ IN_LEFT_PROCESSED = 0x40,
+ IN_RIGHT_PROCESSED = 0x80,
+ IN_FRONT_PROCESSED = 0x100,
+ IN_BACK_PROCESSED = 0x200,
+ IN_PRESSURE = 0x400,
+ IN_X_AXIS = 0x800,
+ IN_Y_AXIS = 0x1000,
+ IN_Z_AXIS = 0x2000,
+ IN_BACK_LEFT = 0x10000,
+ IN_BACK_RIGHT = 0x20000,
+ IN_CENTER = 0x40000,
+ IN_LOW_FREQUENCY = 0x100000,
+ IN_TOP_LEFT = 0x200000,
+ IN_TOP_RIGHT = 0x400000,
+
+ IN_VOICE_UPLINK = 0x4000,
+ IN_VOICE_DNLINK = 0x8000,
+
+ IN_MONO = IN_FRONT,
+ IN_STEREO = (IN_LEFT | IN_RIGHT),
+ IN_FRONT_BACK = (IN_FRONT | IN_BACK),
+ IN_6 = (IN_LEFT | IN_RIGHT |
+ IN_FRONT | IN_BACK |
+ IN_LEFT_PROCESSED | IN_RIGHT_PROCESSED),
+ IN_2POINT0POINT2 = (IN_LEFT | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT),
+ IN_2POINT1POINT2 = (IN_LEFT | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT |
+ IN_LOW_FREQUENCY),
+ IN_3POINT0POINT2 = (IN_LEFT | IN_CENTER | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT),
+ IN_3POINT1POINT2 = (IN_LEFT | IN_CENTER | IN_RIGHT |
+ IN_TOP_LEFT | IN_TOP_RIGHT | IN_LOW_FREQUENCY),
+ IN_5POINT1 = (IN_LEFT | IN_CENTER | IN_RIGHT |
+ IN_BACK_LEFT | IN_BACK_RIGHT | IN_LOW_FREQUENCY),
+ IN_VOICE_UPLINK_MONO = (IN_VOICE_UPLINK | IN_MONO),
+ IN_VOICE_DNLINK_MONO = (IN_VOICE_DNLINK | IN_MONO),
+ IN_VOICE_CALL_MONO = (IN_VOICE_UPLINK_MONO |
+ IN_VOICE_DNLINK_MONO),
+ // Note that the 2.0 IN_ALL* have been moved to helper functions
+
+ COUNT_MAX = 30,
+ INDEX_HDR = REPRESENTATION_INDEX << COUNT_MAX,
+ INDEX_MASK_1 = INDEX_HDR | ((1 << 1) - 1),
+ INDEX_MASK_2 = INDEX_HDR | ((1 << 2) - 1),
+ INDEX_MASK_3 = INDEX_HDR | ((1 << 3) - 1),
+ INDEX_MASK_4 = INDEX_HDR | ((1 << 4) - 1),
+ INDEX_MASK_5 = INDEX_HDR | ((1 << 5) - 1),
+ INDEX_MASK_6 = INDEX_HDR | ((1 << 6) - 1),
+ INDEX_MASK_7 = INDEX_HDR | ((1 << 7) - 1),
+ INDEX_MASK_8 = INDEX_HDR | ((1 << 8) - 1),
+ INDEX_MASK_9 = INDEX_HDR | ((1 << 9) - 1),
+ INDEX_MASK_10 = INDEX_HDR | ((1 << 10) - 1),
+ INDEX_MASK_11 = INDEX_HDR | ((1 << 11) - 1),
+ INDEX_MASK_12 = INDEX_HDR | ((1 << 12) - 1),
+ INDEX_MASK_13 = INDEX_HDR | ((1 << 13) - 1),
+ INDEX_MASK_14 = INDEX_HDR | ((1 << 14) - 1),
+ INDEX_MASK_15 = INDEX_HDR | ((1 << 15) - 1),
+ INDEX_MASK_16 = INDEX_HDR | ((1 << 16) - 1),
+ INDEX_MASK_17 = INDEX_HDR | ((1 << 17) - 1),
+ INDEX_MASK_18 = INDEX_HDR | ((1 << 18) - 1),
+ INDEX_MASK_19 = INDEX_HDR | ((1 << 19) - 1),
+ INDEX_MASK_20 = INDEX_HDR | ((1 << 20) - 1),
+ INDEX_MASK_21 = INDEX_HDR | ((1 << 21) - 1),
+ INDEX_MASK_22 = INDEX_HDR | ((1 << 22) - 1),
+ INDEX_MASK_23 = INDEX_HDR | ((1 << 23) - 1),
+ INDEX_MASK_24 = INDEX_HDR | ((1 << 24) - 1),
+};
+
+/**
+ * Major modes for a mobile device. The current mode setting affects audio
+ * routing.
+ */
+@export(name="audio_mode_t", value_prefix="AUDIO_MODE_")
+enum AudioMode : int32_t {
+ NORMAL = 0,
+ RINGTONE = 1,
+ /** Calls handled by the telephony stack (Eg: PSTN). */
+ IN_CALL = 2,
+ /** Calls handled by apps (Eg: Hangout). */
+ IN_COMMUNICATION = 3,
+};
+
+@export(name="", value_prefix="AUDIO_DEVICE_")
+enum AudioDevice : uint32_t {
+ NONE = 0x0,
+ /** reserved bits */
+ BIT_IN = 0x80000000,
+ BIT_DEFAULT = 0x40000000,
+ /** output devices */
+ OUT_EARPIECE = 0x1,
+ OUT_SPEAKER = 0x2,
+ OUT_WIRED_HEADSET = 0x4,
+ OUT_WIRED_HEADPHONE = 0x8,
+ OUT_BLUETOOTH_SCO = 0x10,
+ OUT_BLUETOOTH_SCO_HEADSET = 0x20,
+ OUT_BLUETOOTH_SCO_CARKIT = 0x40,
+ OUT_BLUETOOTH_A2DP = 0x80,
+ OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
+ OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
+ OUT_AUX_DIGITAL = 0x400,
+ OUT_HDMI = OUT_AUX_DIGITAL,
+ /** uses an analog connection (multiplexed over the USB pins for instance) */
+ OUT_ANLG_DOCK_HEADSET = 0x800,
+ OUT_DGTL_DOCK_HEADSET = 0x1000,
+ /** USB accessory mode: Android device is USB device and dock is USB host */
+ OUT_USB_ACCESSORY = 0x2000,
+ /** USB host mode: Android device is USB host and dock is USB device */
+ OUT_USB_DEVICE = 0x4000,
+ OUT_REMOTE_SUBMIX = 0x8000,
+ /** Telephony voice TX path */
+ OUT_TELEPHONY_TX = 0x10000,
+ /** Analog jack with line impedance detected */
+ OUT_LINE = 0x20000,
+ /** HDMI Audio Return Channel */
+ OUT_HDMI_ARC = 0x40000,
+ /** S/PDIF out */
+ OUT_SPDIF = 0x80000,
+ /** FM transmitter out */
+ OUT_FM = 0x100000,
+ /** Line out for av devices */
+ OUT_AUX_LINE = 0x200000,
+ /** limited-output speaker device for acoustic safety */
+ OUT_SPEAKER_SAFE = 0x400000,
+ OUT_IP = 0x800000,
+ /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
+ OUT_BUS = 0x1000000,
+ OUT_PROXY = 0x2000000,
+ OUT_USB_HEADSET = 0x4000000,
+ OUT_HEARING_AID = 0x8000000,
+ OUT_ECHO_CANCELLER = 0x10000000,
+ OUT_DEFAULT = BIT_DEFAULT,
+ // Note that the 2.0 OUT_ALL* have been moved to helper functions
+
+ /** input devices */
+ IN_COMMUNICATION = BIT_IN | 0x1,
+ IN_AMBIENT = BIT_IN | 0x2,
+ IN_BUILTIN_MIC = BIT_IN | 0x4,
+ IN_BLUETOOTH_SCO_HEADSET = BIT_IN | 0x8,
+ IN_WIRED_HEADSET = BIT_IN | 0x10,
+ IN_AUX_DIGITAL = BIT_IN | 0x20,
+ IN_HDMI = IN_AUX_DIGITAL,
+ /** Telephony voice RX path */
+ IN_VOICE_CALL = BIT_IN | 0x40,
+ IN_TELEPHONY_RX = IN_VOICE_CALL,
+ IN_BACK_MIC = BIT_IN | 0x80,
+ IN_REMOTE_SUBMIX = BIT_IN | 0x100,
+ IN_ANLG_DOCK_HEADSET = BIT_IN | 0x200,
+ IN_DGTL_DOCK_HEADSET = BIT_IN | 0x400,
+ IN_USB_ACCESSORY = BIT_IN | 0x800,
+ IN_USB_DEVICE = BIT_IN | 0x1000,
+ /** FM tuner input */
+ IN_FM_TUNER = BIT_IN | 0x2000,
+ /** TV tuner input */
+ IN_TV_TUNER = BIT_IN | 0x4000,
+ /** Analog jack with line impedance detected */
+ IN_LINE = BIT_IN | 0x8000,
+ /** S/PDIF in */
+ IN_SPDIF = BIT_IN | 0x10000,
+ IN_BLUETOOTH_A2DP = BIT_IN | 0x20000,
+ IN_LOOPBACK = BIT_IN | 0x40000,
+ IN_IP = BIT_IN | 0x80000,
+ /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
+ IN_BUS = BIT_IN | 0x100000,
+ IN_PROXY = BIT_IN | 0x1000000,
+ IN_USB_HEADSET = BIT_IN | 0x2000000,
+ IN_BLUETOOTH_BLE = BIT_IN | 0x4000000,
+ IN_ECHO_REFERENCE = BIT_IN | 0x10000000,
+ IN_DEFAULT = BIT_IN | BIT_DEFAULT,
+
+ // Note that the 2.0 IN_ALL* have been moved to helper functions
+};
+
+/**
+ * IEEE 802 MAC address.
+ */
+typedef uint8_t[6] MacAddress;
+
+/**
+ * Specifies a device address in case when several devices of the same type
+ * can be connected (e.g. BT A2DP, USB).
+ */
+struct DeviceAddress {
+ AudioDevice device; // discriminator
+ union Address {
+ MacAddress mac; // used for BLUETOOTH_A2DP_*
+ uint8_t[4] ipv4; // used for IP
+ struct Alsa {
+ int32_t card;
+ int32_t device;
+ } alsa; // used for USB_*
+ } address;
+ /** Arbitrary BUS device unique address. Should not be interpreted by the framework. */
+ string busAddress;
+ /** Arbitrary REMOTE_SUBMIX device unique address. Should not be interpreted by the HAL. */
+ string rSubmixAddress;
+};
+
+/**
+ * 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.
+ */
+@export(name="audio_output_flags_t", value_prefix="AUDIO_OUTPUT_FLAG_")
+enum AudioOutputFlag : int32_t {
+ NONE = 0x0, // no attributes
+ DIRECT = 0x1, // this output directly connects a track
+ // to one output stream: no software mixer
+ 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.
+ FAST = 0x4, // output supports "fast tracks", defined elsewhere
+ DEEP_BUFFER = 0x8, // use deep audio buffers
+ COMPRESS_OFFLOAD = 0x10, // offload playback of compressed streams to
+ // hardware codec
+ NON_BLOCKING = 0x20, // use non-blocking write
+ HW_AV_SYNC = 0x40, // output uses a hardware A/V sync
+ TTS = 0x80, // output for streams transmitted through speaker at a
+ // sample rate high enough to accommodate lower-range
+ // ultrasonic p/b
+ RAW = 0x100, // minimize signal processing
+ SYNC = 0x200, // synchronize I/O streams
+ IEC958_NONAUDIO = 0x400, // Audio stream contains compressed audio in SPDIF
+ // data bursts, not PCM.
+ DIRECT_PCM = 0x2000, // Audio stream containing PCM data that needs
+ // to pass through compress path for DSP post proc.
+ MMAP_NOIRQ = 0x4000, // output operates in MMAP no IRQ mode.
+ VOIP_RX = 0x8000, // preferred output for VoIP calls.
+ /** preferred output for call music */
+ INCALL_MUSIC = 0x10000,
+};
+
+/**
+ * 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.
+ */
+@export(name="audio_input_flags_t", value_prefix="AUDIO_INPUT_FLAG_")
+enum AudioInputFlag : int32_t {
+ NONE = 0x0, // no attributes
+ FAST = 0x1, // prefer an input that supports "fast tracks"
+ HW_HOTWORD = 0x2, // prefer an input that captures from hw hotword source
+ RAW = 0x4, // minimize signal processing
+ SYNC = 0x8, // synchronize I/O streams
+ MMAP_NOIRQ = 0x10, // input operates in MMAP no IRQ mode.
+ VOIP_TX = 0x20, // preferred input for VoIP calls.
+ HW_AV_SYNC = 0x40, // input connected to an output that uses a hardware A/V sync
+};
+
+@export(name="audio_usage_t", value_prefix="AUDIO_USAGE_")
+enum AudioUsage : int32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioAttributes.java
+ // Note that not all framework values are exposed
+ UNKNOWN = 0,
+ MEDIA = 1,
+ VOICE_COMMUNICATION = 2,
+ VOICE_COMMUNICATION_SIGNALLING = 3,
+ ALARM = 4,
+ NOTIFICATION = 5,
+ NOTIFICATION_TELEPHONY_RINGTONE = 6,
+ ASSISTANCE_ACCESSIBILITY = 11,
+ ASSISTANCE_NAVIGATION_GUIDANCE = 12,
+ ASSISTANCE_SONIFICATION = 13,
+ GAME = 14,
+ VIRTUAL_SOURCE = 15,
+ ASSISTANT = 16,
+};
+
+/** Type of audio generated by an application. */
+@export(name="audio_content_type_t", value_prefix="AUDIO_CONTENT_TYPE_")
+enum AudioContentType : uint32_t {
+ // Do not change these values without updating their counterparts
+ // in frameworks/base/media/java/android/media/AudioAttributes.java
+ UNKNOWN = 0,
+ SPEECH = 1,
+ MUSIC = 2,
+ MOVIE = 3,
+ SONIFICATION = 4,
+};
+
+/**
+ * Additional information about the stream passed to hardware decoders.
+ */
+struct AudioOffloadInfo {
+ uint32_t sampleRateHz;
+ bitfield<AudioChannelMask> channelMask;
+ AudioFormat format;
+ AudioStreamType streamType;
+ uint32_t bitRatePerSecond;
+ int64_t durationMicroseconds; // -1 if unknown
+ bool hasVideo;
+ bool isStreaming;
+ uint32_t bitWidth;
+ uint32_t bufferSize;
+ AudioUsage usage;
+};
+
+/**
+ * Commonly used audio stream configuration parameters.
+ */
+struct AudioConfig {
+ uint32_t sampleRateHz;
+ bitfield<AudioChannelMask> channelMask;
+ AudioFormat format;
+ AudioOffloadInfo offloadInfo;
+ uint64_t frameCount;
+};
+
+/** Metadata of a playback track for a StreamOut. */
+struct PlaybackTrackMetadata {
+ AudioUsage usage;
+ AudioContentType contentType;
+ /**
+ * Positive linear gain applied to the track samples. 0 being muted and 1 is no attenuation,
+ * 2 means double amplification...
+ * Must not be negative.
+ */
+ float gain;
+};
+
+/** Metadatas of the source of a StreamOut. */
+struct SourceMetadata {
+ vec<PlaybackTrackMetadata> tracks;
+};
+
+/** Metadata of a record track for a StreamIn. */
+struct RecordTrackMetadata {
+ AudioSource source;
+ /**
+ * Positive linear gain applied to the track samples. 0 being muted and 1 is no attenuation,
+ * 2 means double amplification...
+ * Must not be negative.
+ */
+ float gain;
+ /**
+ * Indicates the destination of an input stream, can be left unspecified.
+ */
+ safe_union Destination {
+ Monostate unspecified;
+ DeviceAddress device;
+ };
+ Destination destination;
+};
+
+/** Metadatas of the sink of a StreamIn. */
+struct SinkMetadata {
+ vec<RecordTrackMetadata> tracks;
+};
+
+
+/*
+ *
+ * Volume control
+ *
+ */
+
+/**
+ * Type of gain control exposed by an audio port.
+ */
+@export(name="", value_prefix="AUDIO_GAIN_MODE_")
+enum AudioGainMode : uint32_t {
+ JOINT = 0x1, // supports joint channel gain control
+ CHANNELS = 0x2, // supports separate channel gain control
+ 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 {
+ bitfield<AudioGainMode> mode;
+ bitfield<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 */
+@export(name="audio_port_role_t", value_prefix="AUDIO_PORT_ROLE_")
+enum AudioPortRole : int32_t {
+ NONE,
+ SOURCE,
+ 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 OUT_SPEAKER)
+ */
+@export(name="audio_port_type_t", value_prefix="AUDIO_PORT_TYPE_")
+enum AudioPortType : int32_t {
+ NONE,
+ DEVICE,
+ MIX,
+ 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 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.
+ */
+@export(name="", value_prefix="AUDIO_PORT_CONFIG_")
+enum AudioPortConfigMask : uint32_t {
+ SAMPLE_RATE = 0x1,
+ CHANNEL_MASK = 0x2,
+ FORMAT = 0x4,
+ GAIN = 0x8,
+};
+
+/**
+ * Audio port configuration structure used to specify a particular configuration
+ * of an audio port.
+ */
+struct AudioPortConfig {
+ AudioPortHandle id;
+ bitfield<AudioPortConfigMask> configMask;
+ uint32_t sampleRateHz;
+ bitfield<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;
+ } useCase;
+ } mix;
+ AudioPortConfigSessionExt session;
+ } ext;
+};
+
+/**
+ * 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;
+ /** 32 byte string identifying the port. */
+ uint8_t[32] address;
+};
+
+/**
+ * Latency class of the audio mix.
+ */
+@export(name="audio_mix_latency_class_t", value_prefix="AUDIO_LATENCY_")
+enum AudioMixLatencyClass : int32_t {
+ LOW,
+ 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<bitfield<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;
+ } ext;
+};
+
+struct ThreadInfo {
+ int64_t pid;
+ int64_t tid;
+};
diff --git a/audio/common/all-versions/default/Android.bp b/audio/common/all-versions/default/Android.bp
index e630d93..0eb4a71 100644
--- a/audio/common/all-versions/default/Android.bp
+++ b/audio/common/all-versions/default/Android.bp
@@ -104,3 +104,16 @@
"-include common/all-versions/VersionMacro.h",
]
}
+
+cc_library_shared {
+ name: "android.hardware.audio.common@6.0-util",
+ defaults: ["android.hardware.audio.common-util_default"],
+ shared_libs: [
+ "android.hardware.audio.common@6.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=6",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
diff --git a/audio/common/all-versions/default/service/Android.bp b/audio/common/all-versions/default/service/Android.bp
index f5a6f42..4565730 100644
--- a/audio/common/all-versions/default/service/Android.bp
+++ b/audio/common/all-versions/default/service/Android.bp
@@ -27,12 +27,15 @@
"android.hardware.audio@2.0",
"android.hardware.audio@4.0",
"android.hardware.audio@5.0",
+ "android.hardware.audio@6.0",
"android.hardware.audio.common@2.0",
"android.hardware.audio.common@4.0",
"android.hardware.audio.common@5.0",
+ "android.hardware.audio.common@6.0",
"android.hardware.audio.effect@2.0",
"android.hardware.audio.effect@4.0",
"android.hardware.audio.effect@5.0",
+ "android.hardware.audio.effect@6.0",
"android.hardware.bluetooth.a2dp@1.0",
"android.hardware.bluetooth.audio@2.0",
"android.hardware.soundtrigger@2.0",
diff --git a/audio/common/all-versions/default/service/service.cpp b/audio/common/all-versions/default/service/service.cpp
index e8a9e23..2730f3b 100644
--- a/audio/common/all-versions/default/service/service.cpp
+++ b/audio/common/all-versions/default/service/service.cpp
@@ -19,9 +19,11 @@
#include <android/hardware/audio/2.0/IDevicesFactory.h>
#include <android/hardware/audio/4.0/IDevicesFactory.h>
#include <android/hardware/audio/5.0/IDevicesFactory.h>
+#include <android/hardware/audio/6.0/IDevicesFactory.h>
#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
#include <android/hardware/audio/effect/5.0/IEffectsFactory.h>
+#include <android/hardware/audio/effect/6.0/IEffectsFactory.h>
#include <android/hardware/bluetooth/a2dp/1.0/IBluetoothAudioOffload.h>
#include <android/hardware/bluetooth/audio/2.0/IBluetoothAudioProvidersFactory.h>
#include <android/hardware/soundtrigger/2.0/ISoundTriggerHw.h>
@@ -62,12 +64,14 @@
// Keep versions on a separate line for easier parsing
// clang-format off
LOG_ALWAYS_FATAL_IF((registerPassthroughServiceImplementations<
+ audio::V6_0::IDevicesFactory,
audio::V5_0::IDevicesFactory,
audio::V4_0::IDevicesFactory,
audio::V2_0::IDevicesFactory>()),
"Could not register audio core API");
LOG_ALWAYS_FATAL_IF((registerPassthroughServiceImplementations<
+ audio::effect::V6_0::IEffectsFactory,
audio::effect::V5_0::IEffectsFactory,
audio::effect::V4_0::IEffectsFactory,
audio::effect::V2_0::IEffectsFactory>()),