Merge changes from topic "aidllize-audioflinger3"

* changes:
  Move business logic out of IAudioFlinger
  DeviceDescriptorBase parcels to media::AudioPort
  Add conversion between audio_port_v7 and media::AudioPort
  Extract union utilities
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index 8fb836d..f11f184 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -110,29 +110,6 @@
 }
 
 ////////////////////////////////////////////////////////////////////////////////////////////////////
-// Utilities for working with AIDL unions.
-// UNION_GET(obj, fieldname) returns a ConversionResult<T> containing either the strongly-typed
-//   value of the respective field, or BAD_VALUE if the union is not set to the requested field.
-// UNION_SET(obj, fieldname, value) sets the requested field to the given value.
-
-template<typename T, typename T::Tag tag>
-using UnionFieldType = std::decay_t<decltype(std::declval<T>().template get<tag>())>;
-
-template<typename T, typename T::Tag tag>
-ConversionResult<UnionFieldType<T, tag>> unionGetField(const T& u) {
-    if (u.getTag() != tag) {
-        return unexpected(BAD_VALUE);
-    }
-    return u.template get<tag>();
-}
-
-#define UNION_GET(u, field) \
-    unionGetField<std::decay_t<decltype(u)>, std::decay_t<decltype(u)>::Tag::field>(u)
-
-#define UNION_SET(u, field, value) \
-    (u).set<std::decay_t<decltype(u)>::Tag::field>(value)
-
-////////////////////////////////////////////////////////////////////////////////////////////////////
 
 enum class Direction {
     INPUT, OUTPUT
@@ -483,7 +460,7 @@
     return static_cast<media::audio::common::AudioFormat>(legacy);
 }
 
-ConversionResult<int> aidl2legacy_AudioGainMode_int(media::AudioGainMode aidl) {
+ConversionResult<audio_gain_mode_t> aidl2legacy_AudioGainMode_audio_gain_mode_t(media::AudioGainMode aidl) {
     switch (aidl) {
         case media::AudioGainMode::JOINT:
             return AUDIO_GAIN_MODE_JOINT;
@@ -496,7 +473,7 @@
     }
 }
 
-ConversionResult<media::AudioGainMode> legacy2aidl_int_AudioGainMode(int legacy) {
+ConversionResult<media::AudioGainMode> legacy2aidl_audio_gain_mode_t_AudioGainMode(audio_gain_mode_t legacy) {
     switch (legacy) {
         case AUDIO_GAIN_MODE_JOINT:
             return media::AudioGainMode::JOINT;
@@ -509,20 +486,20 @@
     }
 }
 
-ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t(int32_t aidl) {
-    return convertBitmask<audio_gain_mode_t, int32_t, int, media::AudioGainMode>(
-            aidl, aidl2legacy_AudioGainMode_int,
+ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl) {
+    return convertBitmask<audio_gain_mode_t, int32_t, audio_gain_mode_t, media::AudioGainMode>(
+            aidl, aidl2legacy_AudioGainMode_audio_gain_mode_t,
             // AudioGainMode is index-based.
             index2enum_index<media::AudioGainMode>,
             // AUDIO_GAIN_MODE_* constants are mask-based.
-            enumToMask_bitmask<audio_gain_mode_t, int>);
+            enumToMask_bitmask<audio_gain_mode_t, audio_gain_mode_t>);
 }
 
-ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t(audio_gain_mode_t legacy) {
-    return convertBitmask<int32_t, audio_gain_mode_t, media::AudioGainMode, int>(
-            legacy, legacy2aidl_int_AudioGainMode,
+ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_mask_int32_t(audio_gain_mode_t legacy) {
+    return convertBitmask<int32_t, audio_gain_mode_t, media::AudioGainMode, audio_gain_mode_t>(
+            legacy, legacy2aidl_audio_gain_mode_t_AudioGainMode,
             // AUDIO_GAIN_MODE_* constants are mask-based.
-            index2enum_bitmask<int>,
+            index2enum_bitmask<audio_gain_mode_t>,
             // AudioGainMode is index-based.
             enumToMask_index<int32_t, media::AudioGainMode>);
 }
@@ -541,7 +518,7 @@
         const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type) {
     audio_gain_config legacy;
     legacy.index = VALUE_OR_RETURN(convertIntegral<int>(aidl.index));
-    legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t(aidl.mode));
+    legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t_mask(aidl.mode));
     legacy.channel_mask =
             VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
     const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
@@ -563,7 +540,7 @@
         const audio_gain_config& legacy, audio_port_role_t role, audio_port_type_t type) {
     media::AudioGainConfig aidl;
     aidl.index = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.index));
-    aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t(legacy.mode));
+    aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_mask_int32_t(legacy.mode));
     aidl.channelMask =
             VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
     const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
@@ -977,7 +954,7 @@
     switch (role) {
         case media::AudioPortRole::NONE:
             // Just verify that the union is empty.
-            VALUE_OR_RETURN(UNION_GET(aidl, nothing));
+            VALUE_OR_RETURN(UNION_GET(aidl, unspecified));
             break;
 
         case media::AudioPortRole::SOURCE:
@@ -1004,7 +981,7 @@
 
     switch (role) {
         case AUDIO_PORT_ROLE_NONE:
-            UNION_SET(aidl, nothing, false);
+            UNION_SET(aidl, unspecified, false);
             break;
         case AUDIO_PORT_ROLE_SOURCE:
             // This is not a bug. A SOURCE role corresponds to the stream field.
@@ -1061,12 +1038,10 @@
         const media::AudioPortConfigExt& aidl, media::AudioPortType type,
         media::AudioPortRole role) {
     audio_port_config_ext legacy;
-    // Our way of representing a union in AIDL is to have multiple vectors and require that at most
-    // one of the them has size 1 and the rest are empty.
     switch (type) {
         case media::AudioPortType::NONE:
             // Just verify that the union is empty.
-            VALUE_OR_RETURN(UNION_GET(aidl, nothing));
+            VALUE_OR_RETURN(UNION_GET(aidl, unspecified));
             break;
         case media::AudioPortType::DEVICE:
             legacy.device = VALUE_OR_RETURN(
@@ -1092,7 +1067,7 @@
 
     switch (type) {
         case AUDIO_PORT_TYPE_NONE:
-            UNION_SET(aidl, nothing, false);
+            UNION_SET(aidl, unspecified, false);
             break;
         case AUDIO_PORT_TYPE_DEVICE:
             UNION_SET(aidl, device,
@@ -1829,4 +1804,282 @@
             enumToMask_index<int32_t, media::AudioEncapsulationMetadataType>);
 }
 
+ConversionResult<audio_mix_latency_class_t>
+aidl2legacy_AudioMixLatencyClass_audio_mix_latency_class_t(
+        media::AudioMixLatencyClass aidl) {
+    switch (aidl) {
+        case media::AudioMixLatencyClass::LOW:
+            return AUDIO_LATENCY_LOW;
+        case media::AudioMixLatencyClass::NORMAL:
+            return AUDIO_LATENCY_NORMAL;
+    }
+    return unexpected(BAD_VALUE);
+}
+
+ConversionResult<media::AudioMixLatencyClass>
+legacy2aidl_audio_mix_latency_class_t_AudioMixLatencyClass(
+        audio_mix_latency_class_t legacy) {
+    switch (legacy) {
+        case AUDIO_LATENCY_LOW:
+            return media::AudioMixLatencyClass::LOW;
+        case AUDIO_LATENCY_NORMAL:
+            return media::AudioMixLatencyClass::NORMAL;
+    }
+    return unexpected(BAD_VALUE);
+}
+
+ConversionResult<audio_port_device_ext>
+aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(const media::AudioPortDeviceExt& aidl) {
+    audio_port_device_ext legacy;
+    legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
+    legacy.type = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.device.type));
+    RETURN_IF_ERROR(
+            aidl2legacy_string(aidl.device.address, legacy.address, sizeof(legacy.address)));
+    legacy.encapsulation_modes = VALUE_OR_RETURN(
+            aidl2legacy_AudioEncapsulationMode_mask(aidl.encapsulationModes));
+    legacy.encapsulation_metadata_types = VALUE_OR_RETURN(
+            aidl2legacy_AudioEncapsulationMetadataType_mask(aidl.encapsulationMetadataTypes));
+    return legacy;
+}
+
+ConversionResult<media::AudioPortDeviceExt>
+legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(const audio_port_device_ext& legacy) {
+    media::AudioPortDeviceExt aidl;
+    aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
+    aidl.device.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.type));
+    aidl.device.address = VALUE_OR_RETURN(
+            legacy2aidl_string(legacy.address, sizeof(legacy.address)));
+    aidl.encapsulationModes = VALUE_OR_RETURN(
+            legacy2aidl_AudioEncapsulationMode_mask(legacy.encapsulation_modes));
+    aidl.encapsulationMetadataTypes = VALUE_OR_RETURN(
+            legacy2aidl_AudioEncapsulationMetadataType_mask(legacy.encapsulation_metadata_types));
+    return aidl;
+}
+
+ConversionResult<audio_port_mix_ext>
+aidl2legacy_AudioPortMixExt_audio_port_mix_ext(const media::AudioPortMixExt& aidl) {
+    audio_port_mix_ext legacy;
+    legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
+    legacy.handle = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_io_handle_t(aidl.handle));
+    legacy.latency_class = VALUE_OR_RETURN(
+            aidl2legacy_AudioMixLatencyClass_audio_mix_latency_class_t(aidl.latencyClass));
+    return legacy;
+}
+
+ConversionResult<media::AudioPortMixExt>
+legacy2aidl_audio_port_mix_ext_AudioPortMixExt(const audio_port_mix_ext& legacy) {
+    media::AudioPortMixExt aidl;
+    aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
+    aidl.handle = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(legacy.handle));
+    aidl.latencyClass = VALUE_OR_RETURN(
+            legacy2aidl_audio_mix_latency_class_t_AudioMixLatencyClass(legacy.latency_class));
+    return aidl;
+}
+
+ConversionResult<audio_port_session_ext>
+aidl2legacy_AudioPortSessionExt_audio_port_session_ext(const media::AudioPortSessionExt& aidl) {
+    audio_port_session_ext legacy;
+    legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl.session));
+    return legacy;
+}
+
+ConversionResult<media::AudioPortSessionExt>
+legacy2aidl_audio_port_session_ext_AudioPortSessionExt(const audio_port_session_ext& legacy) {
+    media::AudioPortSessionExt aidl;
+    aidl.session = VALUE_OR_RETURN(legacy2aidl_audio_session_t_int32_t(legacy.session));
+    return aidl;
+}
+
+// This type is unnamed in the original definition, thus we name it here.
+using audio_port_v7_ext = decltype(audio_port_v7::ext);
+
+ConversionResult<audio_port_v7_ext> aidl2legacy_AudioPortExt(
+        const media::AudioPortExt& aidl, media::AudioPortType type) {
+    audio_port_v7_ext legacy;
+    switch (type) {
+        case media::AudioPortType::NONE:
+            // Just verify that the union is empty.
+            VALUE_OR_RETURN(UNION_GET(aidl, unspecified));
+            break;
+        case media::AudioPortType::DEVICE:
+            legacy.device = VALUE_OR_RETURN(
+                    aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(
+                            VALUE_OR_RETURN(UNION_GET(aidl, device))));
+            break;
+        case media::AudioPortType::MIX:
+            legacy.mix = VALUE_OR_RETURN(
+                    aidl2legacy_AudioPortMixExt_audio_port_mix_ext(
+                            VALUE_OR_RETURN(UNION_GET(aidl, mix))));
+            break;
+        case media::AudioPortType::SESSION:
+            legacy.session = VALUE_OR_RETURN(aidl2legacy_AudioPortSessionExt_audio_port_session_ext(
+                    VALUE_OR_RETURN(UNION_GET(aidl, session))));
+            break;
+        default:
+            LOG_ALWAYS_FATAL("Shouldn't get here");
+    }
+    return legacy;
+}
+
+ConversionResult<media::AudioPortExt> legacy2aidl_AudioPortExt(
+        const audio_port_v7_ext& legacy, audio_port_type_t type) {
+    media::AudioPortExt aidl;
+    switch (type) {
+        case AUDIO_PORT_TYPE_NONE:
+            UNION_SET(aidl, unspecified, false);
+            break;
+        case AUDIO_PORT_TYPE_DEVICE:
+            UNION_SET(aidl, device,
+                      VALUE_OR_RETURN(
+                              legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(legacy.device)));
+            break;
+        case AUDIO_PORT_TYPE_MIX:
+            UNION_SET(aidl, mix,
+                      VALUE_OR_RETURN(legacy2aidl_audio_port_mix_ext_AudioPortMixExt(legacy.mix)));
+            break;
+        case AUDIO_PORT_TYPE_SESSION:
+            UNION_SET(aidl, session,
+                      VALUE_OR_RETURN(legacy2aidl_audio_port_session_ext_AudioPortSessionExt(
+                              legacy.session)));
+            break;
+        default:
+            LOG_ALWAYS_FATAL("Shouldn't get here");
+    }
+    return aidl;
+}
+
+ConversionResult<audio_profile>
+aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl) {
+    audio_profile legacy;
+    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+
+    if (aidl.samplingRates.size() > std::size(legacy.sample_rates)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(
+            convertRange(aidl.samplingRates.begin(), aidl.samplingRates.end(), legacy.sample_rates,
+                         convertIntegral<int32_t, unsigned int>));
+    legacy.num_sample_rates = aidl.samplingRates.size();
+
+    if (aidl.channelMasks.size() > std::size(legacy.channel_masks)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(
+            convertRange(aidl.channelMasks.begin(), aidl.channelMasks.end(), legacy.channel_masks,
+                         aidl2legacy_int32_t_audio_channel_mask_t));
+    legacy.num_channel_masks = aidl.channelMasks.size();
+    return legacy;
+}
+
+ConversionResult<media::AudioProfile>
+legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy) {
+    media::AudioProfile aidl;
+    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+
+    if (legacy.num_sample_rates > std::size(legacy.sample_rates)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(
+            convertRange(legacy.sample_rates, legacy.sample_rates + legacy.num_sample_rates,
+                         std::back_inserter(aidl.samplingRates),
+                         convertIntegral<unsigned int, int32_t>));
+
+    if (legacy.num_channel_masks > std::size(legacy.channel_masks)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(
+            convertRange(legacy.channel_masks, legacy.channel_masks + legacy.num_channel_masks,
+                         std::back_inserter(aidl.channelMasks),
+                         legacy2aidl_audio_channel_mask_t_int32_t));
+    return aidl;
+}
+
+ConversionResult<audio_gain>
+aidl2legacy_AudioGain_audio_gain(const media::AudioGain& aidl) {
+    audio_gain legacy;
+    legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t_mask(aidl.mode));
+    legacy.channel_mask = VALUE_OR_RETURN(
+            aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
+    legacy.min_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.minValue));
+    legacy.max_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.maxValue));
+    legacy.default_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.defaultValue));
+    legacy.step_value = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.stepValue));
+    legacy.min_ramp_ms = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.minRampMs));
+    legacy.max_ramp_ms = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.maxRampMs));
+    return legacy;
+}
+
+ConversionResult<media::AudioGain>
+legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy) {
+    media::AudioGain aidl;
+    aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_mask_int32_t(legacy.mode));
+    aidl.channelMask = VALUE_OR_RETURN(
+            legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
+    aidl.minValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.min_value));
+    aidl.maxValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.max_value));
+    aidl.defaultValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.default_value));
+    aidl.stepValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.step_value));
+    aidl.minRampMs = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.min_ramp_ms));
+    aidl.maxRampMs = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.max_ramp_ms));
+    return aidl;
+}
+
+ConversionResult<audio_port_v7>
+aidl2legacy_AudioPort_audio_port_v7(const media::AudioPort& aidl) {
+    audio_port_v7 legacy;
+    legacy.id = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.id));
+    legacy.role = VALUE_OR_RETURN(aidl2legacy_AudioPortRole_audio_port_role_t(aidl.role));
+    legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioPortType_audio_port_type_t(aidl.type));
+    RETURN_IF_ERROR(aidl2legacy_string(aidl.name, legacy.name, sizeof(legacy.name)));
+
+    if (aidl.profiles.size() > std::size(legacy.audio_profiles)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(convertRange(aidl.profiles.begin(), aidl.profiles.end(), legacy.audio_profiles,
+                                 aidl2legacy_AudioProfile_audio_profile));
+    legacy.num_audio_profiles = aidl.profiles.size();
+
+    if (aidl.gains.size() > std::size(legacy.gains)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(convertRange(aidl.gains.begin(), aidl.gains.end(), legacy.gains,
+                                 aidl2legacy_AudioGain_audio_gain));
+    legacy.num_gains = aidl.gains.size();
+
+    legacy.active_config = VALUE_OR_RETURN(
+            aidl2legacy_AudioPortConfig_audio_port_config(aidl.activeConfig));
+    legacy.ext = VALUE_OR_RETURN(aidl2legacy_AudioPortExt(aidl.ext, aidl.type));
+    return legacy;
+}
+
+ConversionResult<media::AudioPort>
+legacy2aidl_audio_port_v7_AudioPort(const audio_port_v7& legacy) {
+    media::AudioPort aidl;
+    aidl.id = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.id));
+    aidl.role = VALUE_OR_RETURN(legacy2aidl_audio_port_role_t_AudioPortRole(legacy.role));
+    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_port_type_t_AudioPortType(legacy.type));
+    aidl.name = VALUE_OR_RETURN(legacy2aidl_string(legacy.name, sizeof(legacy.name)));
+
+    if (legacy.num_audio_profiles > std::size(legacy.audio_profiles)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(
+            convertRange(legacy.audio_profiles, legacy.audio_profiles + legacy.num_audio_profiles,
+                         std::back_inserter(aidl.profiles),
+                         legacy2aidl_audio_profile_AudioProfile));
+
+    if (legacy.num_gains > std::size(legacy.gains)) {
+        return unexpected(BAD_VALUE);
+    }
+    RETURN_IF_ERROR(
+            convertRange(legacy.gains, legacy.gains + legacy.num_gains,
+                         std::back_inserter(aidl.gains),
+                         legacy2aidl_audio_gain_AudioGain));
+
+    aidl.activeConfig = VALUE_OR_RETURN(
+            legacy2aidl_audio_port_config_AudioPortConfig(legacy.active_config));
+    aidl.ext = VALUE_OR_RETURN(legacy2aidl_AudioPortExt(legacy.ext, legacy.type));
+    return aidl;
+}
+
 }  // namespace android
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index db1a2a3..fa67898 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -288,6 +288,7 @@
         "aidl/android/media/AudioIoConfigEvent.aidl",
         "aidl/android/media/AudioIoDescriptor.aidl",
         "aidl/android/media/AudioIoFlags.aidl",
+        "aidl/android/media/AudioMixLatencyClass.aidl",
         "aidl/android/media/AudioMode.aidl",
         "aidl/android/media/AudioOffloadInfo.aidl",
         "aidl/android/media/AudioOutputFlags.aidl",
@@ -300,7 +301,11 @@
         "aidl/android/media/AudioPortConfigMixExt.aidl",
         "aidl/android/media/AudioPortConfigMixExtUseCase.aidl",
         "aidl/android/media/AudioPortConfigSessionExt.aidl",
+        "aidl/android/media/AudioPortDeviceExt.aidl",
+        "aidl/android/media/AudioPortExt.aidl",
+        "aidl/android/media/AudioPortMixExt.aidl",
         "aidl/android/media/AudioPortRole.aidl",
+        "aidl/android/media/AudioPortSessionExt.aidl",
         "aidl/android/media/AudioPortType.aidl",
         "aidl/android/media/AudioProfile.aidl",
         "aidl/android/media/AudioSourceType.aidl",
@@ -309,7 +314,6 @@
         "aidl/android/media/AudioUniqueIdUse.aidl",
         "aidl/android/media/AudioUsage.aidl",
         "aidl/android/media/AudioUuid.aidl",
-        "aidl/android/media/DeviceDescriptorBase.aidl",
         "aidl/android/media/EffectDescriptor.aidl",
     ],
     imports: [
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 5dfda09..786af53 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -24,77 +24,10 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/Parcel.h>
-#include <media/AudioValidator.h>
-#include <media/IAudioPolicyService.h>
-#include <mediautils/ServiceUtilities.h>
-#include <mediautils/TimeCheck.h>
 #include "IAudioFlinger.h"
 
 namespace android {
 
-enum {
-    CREATE_TRACK = IBinder::FIRST_CALL_TRANSACTION,
-    CREATE_RECORD,
-    SAMPLE_RATE,
-    RESERVED,   // obsolete, was CHANNEL_COUNT
-    FORMAT,
-    FRAME_COUNT,
-    LATENCY,
-    SET_MASTER_VOLUME,
-    SET_MASTER_MUTE,
-    MASTER_VOLUME,
-    MASTER_MUTE,
-    SET_STREAM_VOLUME,
-    SET_STREAM_MUTE,
-    STREAM_VOLUME,
-    STREAM_MUTE,
-    SET_MODE,
-    SET_MIC_MUTE,
-    GET_MIC_MUTE,
-    SET_RECORD_SILENCED,
-    SET_PARAMETERS,
-    GET_PARAMETERS,
-    REGISTER_CLIENT,
-    GET_INPUTBUFFERSIZE,
-    OPEN_OUTPUT,
-    OPEN_DUPLICATE_OUTPUT,
-    CLOSE_OUTPUT,
-    SUSPEND_OUTPUT,
-    RESTORE_OUTPUT,
-    OPEN_INPUT,
-    CLOSE_INPUT,
-    INVALIDATE_STREAM,
-    SET_VOICE_VOLUME,
-    GET_RENDER_POSITION,
-    GET_INPUT_FRAMES_LOST,
-    NEW_AUDIO_UNIQUE_ID,
-    ACQUIRE_AUDIO_SESSION_ID,
-    RELEASE_AUDIO_SESSION_ID,
-    QUERY_NUM_EFFECTS,
-    QUERY_EFFECT,
-    GET_EFFECT_DESCRIPTOR,
-    CREATE_EFFECT,
-    MOVE_EFFECTS,
-    LOAD_HW_MODULE,
-    GET_PRIMARY_OUTPUT_SAMPLING_RATE,
-    GET_PRIMARY_OUTPUT_FRAME_COUNT,
-    SET_LOW_RAM_DEVICE,
-    LIST_AUDIO_PORTS,
-    GET_AUDIO_PORT,
-    CREATE_AUDIO_PATCH,
-    RELEASE_AUDIO_PATCH,
-    LIST_AUDIO_PATCHES,
-    SET_AUDIO_PORT_CONFIG,
-    GET_AUDIO_HW_SYNC_FOR_SESSION,
-    SYSTEM_READY,
-    FRAME_COUNT_HAL,
-    GET_MICROPHONES,
-    SET_MASTER_BALANCE,
-    GET_MASTER_BALANCE,
-    SET_EFFECT_SUSPENDED,
-    SET_AUDIO_HAL_PIDS
-};
-
 #define MAX_ITEMS_PER_LIST 1024
 
 ConversionResult<media::CreateTrackRequest> IAudioFlinger::CreateTrackInput::toAidl() const {
@@ -988,106 +921,6 @@
 status_t BnAudioFlinger::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
-    // make sure transactions reserved to AudioPolicyManager do not come from other processes
-    switch (code) {
-        case SET_STREAM_VOLUME:
-        case SET_STREAM_MUTE:
-        case OPEN_OUTPUT:
-        case OPEN_DUPLICATE_OUTPUT:
-        case CLOSE_OUTPUT:
-        case SUSPEND_OUTPUT:
-        case RESTORE_OUTPUT:
-        case OPEN_INPUT:
-        case CLOSE_INPUT:
-        case INVALIDATE_STREAM:
-        case SET_VOICE_VOLUME:
-        case MOVE_EFFECTS:
-        case SET_EFFECT_SUSPENDED:
-        case LOAD_HW_MODULE:
-        case LIST_AUDIO_PORTS:
-        case GET_AUDIO_PORT:
-        case CREATE_AUDIO_PATCH:
-        case RELEASE_AUDIO_PATCH:
-        case LIST_AUDIO_PATCHES:
-        case SET_AUDIO_PORT_CONFIG:
-        case SET_RECORD_SILENCED:
-            ALOGW("%s: transaction %d received from PID %d",
-                  __func__, code, IPCThreadState::self()->getCallingPid());
-            // return status only for non void methods
-            switch (code) {
-                case SET_RECORD_SILENCED:
-                case SET_EFFECT_SUSPENDED:
-                    break;
-                default:
-                    reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
-                    break;
-            }
-            return OK;
-        default:
-            break;
-    }
-
-    // make sure the following transactions come from system components
-    switch (code) {
-        case SET_MASTER_VOLUME:
-        case SET_MASTER_MUTE:
-        case SET_MODE:
-        case SET_MIC_MUTE:
-        case SET_LOW_RAM_DEVICE:
-        case SYSTEM_READY:
-        case SET_AUDIO_HAL_PIDS: {
-            if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
-                ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
-                      __func__, code, IPCThreadState::self()->getCallingPid(),
-                      IPCThreadState::self()->getCallingUid());
-                // return status only for non void methods
-                switch (code) {
-                    case SYSTEM_READY:
-                        break;
-                    default:
-                        reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
-                        break;
-                }
-                return OK;
-            }
-        } break;
-        default:
-            break;
-    }
-
-    // List of relevant events that trigger log merging.
-    // Log merging should activate during audio activity of any kind. This are considered the
-    // most relevant events.
-    // TODO should select more wisely the items from the list
-    switch (code) {
-        case CREATE_TRACK:
-        case CREATE_RECORD:
-        case SET_MASTER_VOLUME:
-        case SET_MASTER_MUTE:
-        case SET_MIC_MUTE:
-        case SET_PARAMETERS:
-        case CREATE_EFFECT:
-        case SYSTEM_READY: {
-            requestLogMerge();
-            break;
-        }
-        default:
-            break;
-    }
-
-    std::string tag("IAudioFlinger command " + std::to_string(code));
-    TimeCheck check(tag.c_str());
-
-    // Make sure we connect to Audio Policy Service before calling into AudioFlinger:
-    //  - AudioFlinger can call into Audio Policy Service with its global mutex held
-    //  - If this is the first time Audio Policy Service is queried from inside audioserver process
-    //  this will trigger Audio Policy Manager initialization.
-    //  - Audio Policy Manager initialization calls into AudioFlinger which will try to lock
-    //  its global mutex and a deadlock will occur.
-    if (IPCThreadState::self()->getCallingPid() != getpid()) {
-        AudioSystem::get_audio_policy_service();
-    }
-
     switch (code) {
         case CREATE_TRACK: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
@@ -1496,10 +1329,6 @@
                 ALOGE("b/23905951");
                 return status;
             }
-            status = AudioValidator::validateAudioPort(port);
-            if (status == NO_ERROR) {
-                status = getAudioPort(&port);
-            }
             reply->writeInt32(status);
             if (status == NO_ERROR) {
                 reply->write(&port, sizeof(struct audio_port_v7));
@@ -1519,10 +1348,6 @@
                 ALOGE("b/23905951");
                 return status;
             }
-            status = AudioValidator::validateAudioPatch(patch);
-            if (status == NO_ERROR) {
-                status = createAudioPatch(&patch, &handle);
-            }
             reply->writeInt32(status);
             if (status == NO_ERROR) {
                 reply->write(&handle, sizeof(audio_patch_handle_t));
@@ -1571,10 +1396,6 @@
             if (status != NO_ERROR) {
                 return status;
             }
-            status = AudioValidator::validateAudioPortConfig(config);
-            if (status == NO_ERROR) {
-                status = setAudioPortConfig(&config);
-            }
             reply->writeInt32(status);
             return NO_ERROR;
         } break;
diff --git a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl b/media/libaudioclient/aidl/android/media/AudioMixLatencyClass.aidl
similarity index 62%
copy from media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
copy to media/libaudioclient/aidl/android/media/AudioMixLatencyClass.aidl
index aa0f149..d70b364 100644
--- a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioMixLatencyClass.aidl
@@ -13,22 +13,13 @@
  * See the License for the specific language governing permissions and
  * limitations under the License.
  */
-
 package android.media;
 
-import android.media.AudioPort;
-import android.media.AudioPortConfig;
-import android.media.AudioDevice;
-
 /**
  * {@hide}
  */
-parcelable DeviceDescriptorBase {
-    AudioPort port;
-    AudioPortConfig portConfig;
-    AudioDevice device;
-    /** Bitmask, indexed by AudioEncapsulationMode. */
-    int encapsulationModes;
-    /** Bitmask, indexed by AudioEncapsulationMetadataType. */
-    int encapsulationMetadataTypes;
+@Backing(type="int")
+enum AudioMixLatencyClass {
+    LOW = 0,
+    NORMAL = 1,
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioPort.aidl b/media/libaudioclient/aidl/android/media/AudioPort.aidl
index 1aa532b..123aeb0 100644
--- a/media/libaudioclient/aidl/android/media/AudioPort.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPort.aidl
@@ -17,6 +17,8 @@
 package android.media;
 
 import android.media.AudioGain;
+import android.media.AudioPortConfig;
+import android.media.AudioPortExt;
 import android.media.AudioPortRole;
 import android.media.AudioPortType;
 import android.media.AudioProfile;
@@ -25,11 +27,18 @@
  * {@hide}
  */
 parcelable AudioPort {
-    /** Gain controllers. */
-    AudioGain[] gains;
-    @utf8InCpp String name;
-    AudioPortType type;
+    /** Port unique ID. Interpreted as audio_port_handle_t. */
+    int id;
+    /** Sink or source. */
     AudioPortRole role;
+    /** Device, mix ... */
+    AudioPortType type;
+    @utf8InCpp String name;
     /** AudioProfiles supported by this port (format, Rates, Channels). */
     AudioProfile[] profiles;
+    /** Gain controllers. */
+    AudioGain[] gains;
+    /** Current audio port configuration. */
+    AudioPortConfig activeConfig;
+    AudioPortExt ext;
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
index 38da4f5..5d635b6 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
@@ -29,7 +29,7 @@
      * TODO(ytai): replace with the canonical representation for an empty union, as soon as it is
      *             established.
      */
-    boolean nothing;
+    boolean unspecified;
     /** Device specific info. */
     AudioPortConfigDeviceExt device;
     /** Mix specific info. */
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
index 9e5e081..c61f044 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
@@ -29,7 +29,7 @@
      * TODO(ytai): replace with the canonical representation for an empty union, as soon as it is
      *             established.
      */
-    boolean nothing;
+    boolean unspecified;
     /** This to be set if the containing config has the AudioPortRole::SOURCE role. */
     AudioStreamType stream;
     /** This to be set if the containing config has the AudioPortRole::SINK role. */
diff --git a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl b/media/libaudioclient/aidl/android/media/AudioPortDeviceExt.aidl
similarity index 85%
rename from media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
rename to media/libaudioclient/aidl/android/media/AudioPortDeviceExt.aidl
index aa0f149..b758f23 100644
--- a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortDeviceExt.aidl
@@ -16,16 +16,14 @@
 
 package android.media;
 
-import android.media.AudioPort;
-import android.media.AudioPortConfig;
 import android.media.AudioDevice;
 
 /**
  * {@hide}
  */
-parcelable DeviceDescriptorBase {
-    AudioPort port;
-    AudioPortConfig portConfig;
+parcelable AudioPortDeviceExt {
+    /** Module the device is attached to. Interpreted as audio_module_handle_t. */
+    int hwModule;
     AudioDevice device;
     /** Bitmask, indexed by AudioEncapsulationMode. */
     int encapsulationModes;
diff --git a/media/libaudioclient/aidl/android/media/AudioPortExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortExt.aidl
new file mode 100644
index 0000000..453784b
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortExt.aidl
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2020 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.media.AudioPortDeviceExt;
+import android.media.AudioPortMixExt;
+import android.media.AudioPortSessionExt;
+
+/**
+ * {@hide}
+ */
+union AudioPortExt {
+    /**
+     * This represents an empty union. Value is ignored.
+     * TODO(ytai): replace with the canonical representation for an empty union, as soon as it is
+     *             established.
+     */
+    boolean unspecified;
+    /** Device specific info. */
+    AudioPortDeviceExt device;
+    /** Mix specific info. */
+    AudioPortMixExt mix;
+    /** Session specific info. */
+    AudioPortSessionExt session;
+}
diff --git a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl b/media/libaudioclient/aidl/android/media/AudioPortMixExt.aidl
similarity index 62%
copy from media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
copy to media/libaudioclient/aidl/android/media/AudioPortMixExt.aidl
index aa0f149..62cdb8e 100644
--- a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortMixExt.aidl
@@ -16,19 +16,16 @@
 
 package android.media;
 
-import android.media.AudioPort;
-import android.media.AudioPortConfig;
-import android.media.AudioDevice;
+import android.media.AudioMixLatencyClass;
 
 /**
  * {@hide}
  */
-parcelable DeviceDescriptorBase {
-    AudioPort port;
-    AudioPortConfig portConfig;
-    AudioDevice device;
-    /** Bitmask, indexed by AudioEncapsulationMode. */
-    int encapsulationModes;
-    /** Bitmask, indexed by AudioEncapsulationMetadataType. */
-    int encapsulationMetadataTypes;
+parcelable AudioPortMixExt {
+    /** Module the stream is attached to. Interpreted as audio_module_handle_t. */
+    int hwModule;
+    /** I/O handle of the input/output stream. Interpreted as audio_io_handle_t. */
+    int handle;
+    /** Latency class */
+    AudioMixLatencyClass latencyClass;
 }
diff --git a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl b/media/libaudioclient/aidl/android/media/AudioPortSessionExt.aidl
similarity index 62%
copy from media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
copy to media/libaudioclient/aidl/android/media/AudioPortSessionExt.aidl
index aa0f149..dbca168 100644
--- a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortSessionExt.aidl
@@ -16,19 +16,10 @@
 
 package android.media;
 
-import android.media.AudioPort;
-import android.media.AudioPortConfig;
-import android.media.AudioDevice;
-
 /**
  * {@hide}
  */
-parcelable DeviceDescriptorBase {
-    AudioPort port;
-    AudioPortConfig portConfig;
-    AudioDevice device;
-    /** Bitmask, indexed by AudioEncapsulationMode. */
-    int encapsulationModes;
-    /** Bitmask, indexed by AudioEncapsulationMetadataType. */
-    int encapsulationMetadataTypes;
+parcelable AudioPortSessionExt {
+    /** Audio session. Interpreted as audio_session_t. */
+    int session;
 }
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index 4518adb..06b12e9 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -17,7 +17,7 @@
 package android.media;
 
 import android.media.AudioConfig;
-import android.media.DeviceDescriptorBase;
+import android.media.AudioPort;
 
 /**
  * {@hide}
@@ -26,7 +26,8 @@
     /** Interpreted as audio_module_handle_t. */
     int module;
     AudioConfig config;
-    DeviceDescriptorBase device;
+    /** Type must be DEVICE. */
+    AudioPort device;
     /** Bitmask, indexed by AudioOutputFlag. */
     int flags;
 }
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index 894e56e..2dc471b 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -28,12 +28,20 @@
 #include <android/media/AudioEncapsulationMode.h>
 #include <android/media/AudioEncapsulationMetadataType.h>
 #include <android/media/AudioFlag.h>
+#include <android/media/AudioGain.h>
 #include <android/media/AudioGainMode.h>
 #include <android/media/AudioInputFlags.h>
 #include <android/media/AudioIoConfigEvent.h>
 #include <android/media/AudioIoDescriptor.h>
+#include <android/media/AudioMixLatencyClass.h>
 #include <android/media/AudioOutputFlags.h>
+#include <android/media/AudioPort.h>
 #include <android/media/AudioPortConfigType.h>
+#include <android/media/AudioPortDeviceExt.h>
+#include <android/media/AudioPortExt.h>
+#include <android/media/AudioPortMixExt.h>
+#include <android/media/AudioPortSessionExt.h>
+#include <android/media/AudioProfile.h>
 #include <android/media/AudioTimestampInternal.h>
 #include <android/media/EffectDescriptor.h>
 
@@ -110,11 +118,13 @@
 ConversionResult<media::audio::common::AudioFormat> legacy2aidl_audio_format_t_AudioFormat(
         audio_format_t legacy);
 
-ConversionResult<int> aidl2legacy_AudioGainMode_int(media::AudioGainMode aidl);
-ConversionResult<media::AudioGainMode> legacy2aidl_int_AudioGainMode(int legacy);
+ConversionResult<audio_gain_mode_t>
+aidl2legacy_AudioGainMode_audio_gain_mode_t(media::AudioGainMode aidl);
+ConversionResult<media::AudioGainMode>
+legacy2aidl_audio_gain_mode_t_AudioGainMode(audio_gain_mode_t legacy);
 
-ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t(int32_t aidl);
-ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t(audio_gain_mode_t legacy);
+ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_mask_int32_t(audio_gain_mode_t legacy);
 
 ConversionResult<audio_devices_t> aidl2legacy_int32_t_audio_devices_t(int32_t aidl);
 ConversionResult<int32_t> legacy2aidl_audio_devices_t_int32_t(audio_devices_t legacy);
@@ -279,4 +289,41 @@
 ConversionResult<int32_t>
 legacy2aidl_AudioEncapsulationMetadataType_mask(uint32_t legacy);
 
+ConversionResult<audio_mix_latency_class_t>
+aidl2legacy_AudioMixLatencyClass_audio_mix_latency_class_t(
+        media::AudioMixLatencyClass aidl);
+ConversionResult<media::AudioMixLatencyClass>
+legacy2aidl_audio_mix_latency_class_t_AudioMixLatencyClass(
+        audio_mix_latency_class_t legacy);
+
+ConversionResult<audio_port_device_ext>
+aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(const media::AudioPortDeviceExt& aidl);
+ConversionResult<media::AudioPortDeviceExt>
+legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(const audio_port_device_ext& legacy);
+
+ConversionResult<audio_port_mix_ext>
+aidl2legacy_AudioPortMixExt_audio_port_mix_ext(const media::AudioPortMixExt& aidl);
+ConversionResult<media::AudioPortMixExt>
+legacy2aidl_audio_port_mix_ext_AudioPortMixExt(const audio_port_mix_ext& legacy);
+
+ConversionResult<audio_port_session_ext>
+aidl2legacy_AudioPortSessionExt_audio_port_session_ext(const media::AudioPortSessionExt& aidl);
+ConversionResult<media::AudioPortSessionExt>
+legacy2aidl_audio_port_session_ext_AudioPortSessionExt(const audio_port_session_ext& legacy);
+
+ConversionResult<audio_profile>
+aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl);
+ConversionResult<media::AudioProfile>
+legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy);
+
+ConversionResult<audio_gain>
+aidl2legacy_AudioGain_audio_gain(const media::AudioGain& aidl);
+ConversionResult<media::AudioGain>
+legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy);
+
+ConversionResult<audio_port_v7>
+aidl2legacy_AudioPort_audio_port_v7(const media::AudioPort& aidl);
+ConversionResult<media::AudioPort>
+legacy2aidl_audio_port_v7_AudioPort(const audio_port_v7& legacy);
+
 }  // namespace android
diff --git a/media/libaudioclient/include/media/AidlConversionUtil.h b/media/libaudioclient/include/media/AidlConversionUtil.h
index 00e5ff2..6bfb743 100644
--- a/media/libaudioclient/include/media/AidlConversionUtil.h
+++ b/media/libaudioclient/include/media/AidlConversionUtil.h
@@ -82,6 +82,20 @@
 }
 
 /**
+ * A generic template that helps convert containers of convertible types, using iterators.
+ */
+template<typename InputIterator, typename OutputIterator, typename Func>
+status_t convertRange(InputIterator start,
+                      InputIterator end,
+                      OutputIterator out,
+                      const Func& itemConversion) {
+    for (InputIterator iter = start; iter != end; ++iter, ++out) {
+        *out = VALUE_OR_RETURN_STATUS(itemConversion(*iter));
+    }
+    return OK;
+}
+
+/**
  * A generic template that helps convert containers of convertible types.
  */
 template<typename OutputContainer, typename InputContainer, typename Func>
@@ -95,4 +109,27 @@
     return output;
 }
 
+////////////////////////////////////////////////////////////////////////////////////////////////////
+// Utilities for working with AIDL unions.
+// UNION_GET(obj, fieldname) returns a ConversionResult<T> containing either the strongly-typed
+//   value of the respective field, or BAD_VALUE if the union is not set to the requested field.
+// UNION_SET(obj, fieldname, value) sets the requested field to the given value.
+
+template<typename T, typename T::Tag tag>
+using UnionFieldType = std::decay_t<decltype(std::declval<T>().template get<tag>())>;
+
+template<typename T, typename T::Tag tag>
+ConversionResult<UnionFieldType<T, tag>> unionGetField(const T& u) {
+    if (u.getTag() != tag) {
+        return base::unexpected(BAD_VALUE);
+    }
+    return u.template get<tag>();
+}
+
+#define UNION_GET(u, field) \
+    unionGetField<std::decay_t<decltype(u)>, std::decay_t<decltype(u)>::Tag::field>(u)
+
+#define UNION_SET(u, field, value) \
+    (u).set<std::decay_t<decltype(u)>::Tag::field>(value)
+
 }  // namespace android
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 11d341e..911a34f 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -333,6 +333,70 @@
     virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
 
     virtual status_t setAudioHalPids(const std::vector<pid_t>& pids) = 0;
+
+protected:
+    enum {
+        CREATE_TRACK = IBinder::FIRST_CALL_TRANSACTION,
+        CREATE_RECORD,
+        SAMPLE_RATE,
+        RESERVED,   // obsolete, was CHANNEL_COUNT
+        FORMAT,
+        FRAME_COUNT,
+        LATENCY,
+        SET_MASTER_VOLUME,
+        SET_MASTER_MUTE,
+        MASTER_VOLUME,
+        MASTER_MUTE,
+        SET_STREAM_VOLUME,
+        SET_STREAM_MUTE,
+        STREAM_VOLUME,
+        STREAM_MUTE,
+        SET_MODE,
+        SET_MIC_MUTE,
+        GET_MIC_MUTE,
+        SET_RECORD_SILENCED,
+        SET_PARAMETERS,
+        GET_PARAMETERS,
+        REGISTER_CLIENT,
+        GET_INPUTBUFFERSIZE,
+        OPEN_OUTPUT,
+        OPEN_DUPLICATE_OUTPUT,
+        CLOSE_OUTPUT,
+        SUSPEND_OUTPUT,
+        RESTORE_OUTPUT,
+        OPEN_INPUT,
+        CLOSE_INPUT,
+        INVALIDATE_STREAM,
+        SET_VOICE_VOLUME,
+        GET_RENDER_POSITION,
+        GET_INPUT_FRAMES_LOST,
+        NEW_AUDIO_UNIQUE_ID,
+        ACQUIRE_AUDIO_SESSION_ID,
+        RELEASE_AUDIO_SESSION_ID,
+        QUERY_NUM_EFFECTS,
+        QUERY_EFFECT,
+        GET_EFFECT_DESCRIPTOR,
+        CREATE_EFFECT,
+        MOVE_EFFECTS,
+        LOAD_HW_MODULE,
+        GET_PRIMARY_OUTPUT_SAMPLING_RATE,
+        GET_PRIMARY_OUTPUT_FRAME_COUNT,
+        SET_LOW_RAM_DEVICE,
+        LIST_AUDIO_PORTS,
+        GET_AUDIO_PORT,
+        CREATE_AUDIO_PATCH,
+        RELEASE_AUDIO_PATCH,
+        LIST_AUDIO_PATCHES,
+        SET_AUDIO_PORT_CONFIG,
+        GET_AUDIO_HW_SYNC_FOR_SESSION,
+        SYSTEM_READY,
+        FRAME_COUNT_HAL,
+        GET_MICROPHONES,
+        SET_MASTER_BALANCE,
+        GET_MASTER_BALANCE,
+        SET_EFFECT_SUSPENDED,
+        SET_AUDIO_HAL_PIDS
+    };
 };
 
 
@@ -345,9 +409,6 @@
                                     const Parcel& data,
                                     Parcel* reply,
                                     uint32_t flags = 0);
-
-    // Requests media.log to start merging log buffers
-    virtual void requestLogMerge() = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
index c59e966..56343d8 100644
--- a/media/libaudiofoundation/AudioGain.cpp
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -139,7 +139,8 @@
     parcelable->index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mIndex));
     parcelable->useInChannelMask = mUseInChannelMask;
     parcelable->useForVolume = mUseForVolume;
-    parcelable->mode = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_gain_mode_t_int32_t(mGain.mode));
+    parcelable->mode = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_gain_mode_t_mask_int32_t(mGain.mode));
     parcelable->channelMask = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
     parcelable->minValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_value));
@@ -162,7 +163,8 @@
     mIndex = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.index));
     mUseInChannelMask = parcelable.useInChannelMask;
     mUseForVolume = parcelable.useForVolume;
-    mGain.mode = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_gain_mode_t(parcelable.mode));
+    mGain.mode = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_int32_t_audio_gain_mode_t_mask(parcelable.mode));
     mGain.channel_mask = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_channel_mask_t(parcelable.channelMask));
     mGain.min_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.minValue));
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
index 559c711..6b63675 100644
--- a/media/libaudiofoundation/AudioPort.cpp
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -291,7 +291,7 @@
     parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
     parcelable->gain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.index));
     parcelable->gain.mode = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_gain_mode_t_int32_t(mGain.mode));
+            legacy2aidl_audio_gain_mode_t_mask_int32_t(mGain.mode));
     parcelable->gain.channelMask = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
     parcelable->gain.rampDurationMs = VALUE_OR_RETURN_STATUS(
@@ -315,7 +315,7 @@
     mId = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_port_handle_t(parcelable.id));
     mGain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.gain.index));
     mGain.mode = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_int32_t_audio_gain_mode_t(parcelable.gain.mode));
+            aidl2legacy_int32_t_audio_gain_mode_t_mask(parcelable.gain.mode));
     mGain.channel_mask = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_channel_mask_t(parcelable.gain.channelMask));
     mGain.ramp_duration_ms = VALUE_OR_RETURN_STATUS(
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
index 6261559..a3e9589 100644
--- a/media/libaudiofoundation/DeviceDescriptorBase.cpp
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -159,41 +159,49 @@
 
 status_t DeviceDescriptorBase::writeToParcel(Parcel *parcel) const
 {
-    media::DeviceDescriptorBase parcelable;
+    media::AudioPort parcelable;
     return writeToParcelable(&parcelable)
         ?: parcelable.writeToParcel(parcel);
 }
 
-status_t DeviceDescriptorBase::writeToParcelable(media::DeviceDescriptorBase* parcelable) const {
-    AudioPort::writeToParcelable(&parcelable->port);
-    AudioPortConfig::writeToParcelable(&parcelable->portConfig);
-    parcelable->device = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_AudioDeviceTypeAddress(mDeviceTypeAddr));
-    parcelable->encapsulationModes = VALUE_OR_RETURN_STATUS(
+status_t DeviceDescriptorBase::writeToParcelable(media::AudioPort* parcelable) const {
+    AudioPort::writeToParcelable(parcelable);
+    AudioPortConfig::writeToParcelable(&parcelable->activeConfig);
+    parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
+
+    media::AudioPortDeviceExt ext;
+    ext.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(mDeviceTypeAddr));
+    ext.encapsulationModes = VALUE_OR_RETURN_STATUS(
             legacy2aidl_AudioEncapsulationMode_mask(mEncapsulationModes));
-    parcelable->encapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
+    ext.encapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
             legacy2aidl_AudioEncapsulationMetadataType_mask(mEncapsulationMetadataTypes));
+    UNION_SET(parcelable->ext, device, std::move(ext));
     return OK;
 }
 
 status_t DeviceDescriptorBase::readFromParcel(const Parcel *parcel) {
-    media::DeviceDescriptorBase parcelable;
+    media::AudioPort parcelable;
     return parcelable.readFromParcel(parcel)
         ?: readFromParcelable(parcelable);
 }
 
-status_t DeviceDescriptorBase::readFromParcelable(const media::DeviceDescriptorBase& parcelable) {
-    status_t status = AudioPort::readFromParcelable(parcelable.port)
-                      ?: AudioPortConfig::readFromParcelable(parcelable.portConfig);
+status_t DeviceDescriptorBase::readFromParcelable(const media::AudioPort& parcelable) {
+    if (parcelable.type != media::AudioPortType::DEVICE) {
+        return BAD_VALUE;
+    }
+    status_t status = AudioPort::readFromParcelable(parcelable)
+                      ?: AudioPortConfig::readFromParcelable(parcelable.activeConfig);
     if (status != OK) {
         return status;
     }
+
+    media::AudioPortDeviceExt ext = VALUE_OR_RETURN_STATUS(UNION_GET(parcelable.ext, device));
     mDeviceTypeAddr = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioDeviceTypeAddress(parcelable.device));
+            aidl2legacy_AudioDeviceTypeAddress(ext.device));
     mEncapsulationModes = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioEncapsulationMode_mask(parcelable.encapsulationModes));
+            aidl2legacy_AudioEncapsulationMode_mask(ext.encapsulationModes));
     mEncapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioEncapsulationMetadataType_mask(parcelable.encapsulationMetadataTypes));
+            aidl2legacy_AudioEncapsulationMetadataType_mask(ext.encapsulationMetadataTypes));
     return OK;
 }
 
@@ -219,7 +227,7 @@
 }
 
 ConversionResult<sp<DeviceDescriptorBase>>
-aidl2legacy_DeviceDescriptorBase(const media::DeviceDescriptorBase& aidl) {
+aidl2legacy_DeviceDescriptorBase(const media::AudioPort& aidl) {
     sp<DeviceDescriptorBase> result = new DeviceDescriptorBase(AUDIO_DEVICE_NONE);
     status_t status = result->readFromParcelable(aidl);
     if (status != OK) {
@@ -228,9 +236,9 @@
     return result;
 }
 
-ConversionResult<media::DeviceDescriptorBase>
+ConversionResult<media::AudioPort>
 legacy2aidl_DeviceDescriptorBase(const sp<DeviceDescriptorBase>& legacy) {
-    media::DeviceDescriptorBase aidl;
+    media::AudioPort aidl;
     status_t status = legacy->writeToParcelable(&aidl);
     if (status != OK) {
         return base::unexpected(status);
diff --git a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
index 8a920b7..140ce36 100644
--- a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
+++ b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
@@ -18,7 +18,7 @@
 
 #include <vector>
 
-#include <android/media/DeviceDescriptorBase.h>
+#include <android/media/AudioPort.h>
 #include <binder/Parcel.h>
 #include <binder/Parcelable.h>
 #include <media/AudioContainers.h>
@@ -77,8 +77,8 @@
     status_t writeToParcel(Parcel* parcel) const override;
     status_t readFromParcel(const Parcel* parcel) override;
 
-    status_t writeToParcelable(media::DeviceDescriptorBase* parcelable) const;
-    status_t readFromParcelable(const media::DeviceDescriptorBase& parcelable);
+    status_t writeToParcelable(media::AudioPort* parcelable) const;
+    status_t readFromParcelable(const media::AudioPort& parcelable);
 
 protected:
     AudioDeviceTypeAddr mDeviceTypeAddr;
@@ -113,8 +113,8 @@
 
 // Conversion routines, according to AidlConversion.h conventions.
 ConversionResult<sp<DeviceDescriptorBase>>
-aidl2legacy_DeviceDescriptorBase(const media::DeviceDescriptorBase& aidl);
-ConversionResult<media::DeviceDescriptorBase>
+aidl2legacy_DeviceDescriptorBase(const media::AudioPort& aidl);
+ConversionResult<media::AudioPort>
 legacy2aidl_DeviceDescriptorBase(const sp<DeviceDescriptorBase>& legacy);
 
 } // namespace android
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 9ba99bc..e7a12df 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -31,6 +31,7 @@
 #include <sys/resource.h>
 #include <thread>
 
+
 #include <android/os/IExternalVibratorService.h>
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
@@ -41,8 +42,10 @@
 #include <media/audiohal/DevicesFactoryHalInterface.h>
 #include <media/audiohal/EffectsFactoryHalInterface.h>
 #include <media/AudioParameter.h>
+#include <media/IAudioPolicyService.h>
 #include <media/MediaMetricsItem.h>
 #include <media/TypeConverter.h>
+#include <mediautils/TimeCheck.h>
 #include <memunreachable/memunreachable.h>
 #include <utils/String16.h>
 #include <utils/threads.h>
@@ -69,6 +72,7 @@
 
 #include <media/IMediaLogService.h>
 #include <media/AidlConversion.h>
+#include <media/AudioValidator.h>
 #include <media/nbaio/Pipe.h>
 #include <media/nbaio/PipeReader.h>
 #include <mediautils/BatteryNotifier.h>
@@ -2335,6 +2339,11 @@
 {
     ALOGV(__func__);
 
+    status_t status = AudioValidator::validateAudioPortConfig(*config);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
     audio_module_handle_t module;
     if (config->type == AUDIO_PORT_TYPE_DEVICE) {
         module = config->ext.device.hw_module;
@@ -4036,6 +4045,106 @@
 status_t AudioFlinger::onTransact(
         uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
+    // make sure transactions reserved to AudioPolicyManager do not come from other processes
+    switch (code) {
+        case SET_STREAM_VOLUME:
+        case SET_STREAM_MUTE:
+        case OPEN_OUTPUT:
+        case OPEN_DUPLICATE_OUTPUT:
+        case CLOSE_OUTPUT:
+        case SUSPEND_OUTPUT:
+        case RESTORE_OUTPUT:
+        case OPEN_INPUT:
+        case CLOSE_INPUT:
+        case INVALIDATE_STREAM:
+        case SET_VOICE_VOLUME:
+        case MOVE_EFFECTS:
+        case SET_EFFECT_SUSPENDED:
+        case LOAD_HW_MODULE:
+        case LIST_AUDIO_PORTS:
+        case GET_AUDIO_PORT:
+        case CREATE_AUDIO_PATCH:
+        case RELEASE_AUDIO_PATCH:
+        case LIST_AUDIO_PATCHES:
+        case SET_AUDIO_PORT_CONFIG:
+        case SET_RECORD_SILENCED:
+            ALOGW("%s: transaction %d received from PID %d",
+                  __func__, code, IPCThreadState::self()->getCallingPid());
+            // return status only for non void methods
+            switch (code) {
+                case SET_RECORD_SILENCED:
+                case SET_EFFECT_SUSPENDED:
+                    break;
+                default:
+                    reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+                    break;
+            }
+            return OK;
+        default:
+            break;
+    }
+
+    // make sure the following transactions come from system components
+    switch (code) {
+        case SET_MASTER_VOLUME:
+        case SET_MASTER_MUTE:
+        case SET_MODE:
+        case SET_MIC_MUTE:
+        case SET_LOW_RAM_DEVICE:
+        case SYSTEM_READY:
+        case SET_AUDIO_HAL_PIDS: {
+            if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
+                ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
+                      __func__, code, IPCThreadState::self()->getCallingPid(),
+                      IPCThreadState::self()->getCallingUid());
+                // return status only for non void methods
+                switch (code) {
+                    case SYSTEM_READY:
+                        break;
+                    default:
+                        reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+                        break;
+                }
+                return OK;
+            }
+        } break;
+        default:
+            break;
+    }
+
+    // List of relevant events that trigger log merging.
+    // Log merging should activate during audio activity of any kind. This are considered the
+    // most relevant events.
+    // TODO should select more wisely the items from the list
+    switch (code) {
+        case CREATE_TRACK:
+        case CREATE_RECORD:
+        case SET_MASTER_VOLUME:
+        case SET_MASTER_MUTE:
+        case SET_MIC_MUTE:
+        case SET_PARAMETERS:
+        case CREATE_EFFECT:
+        case SYSTEM_READY: {
+            requestLogMerge();
+            break;
+        }
+        default:
+            break;
+    }
+
+    std::string tag("IAudioFlinger command " + std::to_string(code));
+    TimeCheck check(tag.c_str());
+
+    // Make sure we connect to Audio Policy Service before calling into AudioFlinger:
+    //  - AudioFlinger can call into Audio Policy Service with its global mutex held
+    //  - If this is the first time Audio Policy Service is queried from inside audioserver process
+    //  this will trigger Audio Policy Manager initialization.
+    //  - Audio Policy Manager initialization calls into AudioFlinger which will try to lock
+    //  its global mutex and a deadlock will occur.
+    if (IPCThreadState::self()->getCallingPid() != getpid()) {
+        AudioSystem::get_audio_policy_service();
+    }
+
     return BnAudioFlinger::onTransact(code, data, reply, flags);
 }
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6dfc48f..a2e50f8 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -519,6 +519,7 @@
     const sp<MediaLogNotifier> mMediaLogNotifier;
 
     // This is a helper that is called during incoming binder calls.
+    // Requests media.log to start merging log buffers
     void requestLogMerge();
 
     class TrackHandle;
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index b956b96..1e11660 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -25,6 +25,7 @@
 
 #include "AudioFlinger.h"
 #include <media/AudioParameter.h>
+#include <media/AudioValidator.h>
 #include <media/DeviceDescriptorBase.h>
 #include <media/PatchBuilder.h>
 #include <mediautils/ServiceUtilities.h>
@@ -56,6 +57,11 @@
 
 /* Get supported attributes for a given audio port */
 status_t AudioFlinger::getAudioPort(struct audio_port_v7 *port) {
+    status_t status = AudioValidator::validateAudioPort(*port);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
     Mutex::Autolock _l(mLock);
     return mPatchPanel.getAudioPort(port);
 }
@@ -64,6 +70,11 @@
 status_t AudioFlinger::createAudioPatch(const struct audio_patch *patch,
                                    audio_patch_handle_t *handle)
 {
+    status_t status = AudioValidator::validateAudioPatch(*patch);
+    if (status != NO_ERROR) {
+        return status;
+    }
+
     Mutex::Autolock _l(mLock);
     return mPatchPanel.createAudioPatch(patch, handle);
 }