diff --git a/audio/5.0/Android.bp b/audio/5.0/Android.bp
new file mode 100644
index 0000000..27c1ef5
--- /dev/null
+++ b/audio/5.0/Android.bp
@@ -0,0 +1,48 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+    name: "android.hardware.audio@5.0",
+    root: "android.hardware",
+    vndk: {
+        enabled: true,
+    },
+    srcs: [
+        "types.hal",
+        "IDevice.hal",
+        "IDevicesFactory.hal",
+        "IPrimaryDevice.hal",
+        "IStream.hal",
+        "IStreamIn.hal",
+        "IStreamOut.hal",
+        "IStreamOutCallback.hal",
+    ],
+    interfaces: [
+        "android.hardware.audio.common@5.0",
+        "android.hardware.audio.effect@5.0",
+        "android.hidl.base@1.0",
+    ],
+    types: [
+        "AudioDrain",
+        "AudioFrequencyResponsePoint",
+        "AudioMicrophoneChannelMapping",
+        "AudioMicrophoneCoordinate",
+        "AudioMicrophoneDirectionality",
+        "AudioMicrophoneLocation",
+        "DeviceAddress",
+        "MessageQueueFlagBits",
+        "MicrophoneInfo",
+        "MmapBufferFlag",
+        "MmapBufferInfo",
+        "MmapPosition",
+        "ParameterValue",
+        "PlaybackTrackMetadata",
+        "RecordTrackMetadata",
+        "Result",
+        "SinkMetadata",
+        "SourceMetadata",
+        "TimeSpec",
+    ],
+    gen_java: false,
+    gen_java_constants: true,
+}
+
diff --git a/audio/5.0/IDevice.hal b/audio/5.0/IDevice.hal
new file mode 100644
index 0000000..afb4fad
--- /dev/null
+++ b/audio/5.0/IDevice.hal
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+import IStreamIn;
+import IStreamOut;
+
+interface IDevice {
+    /**
+     * Returns whether the audio hardware interface has been initialized.
+     *
+     * @return retval OK on success, NOT_INITIALIZED on failure.
+     */
+    initCheck() generates (Result retval);
+
+    /**
+     * Sets the audio volume for all audio activities other than voice call. If
+     * NOT_SUPPORTED is returned, the software mixer will emulate this
+     * capability.
+     *
+     * @param volume 1.0f means unity, 0.0f is zero.
+     * @return retval operation completion status.
+     */
+    setMasterVolume(float volume) generates (Result retval);
+
+    /**
+     * Get the current master volume value for the HAL, if the HAL supports
+     * master volume control. For example, AudioFlinger will query this value
+     * from the primary audio HAL when the service starts and use the value for
+     * setting the initial master volume across all HALs. HALs which do not
+     * support this method must return NOT_SUPPORTED in 'retval'.
+     *
+     * @return retval operation completion status.
+     * @return volume 1.0f means unity, 0.0f is zero.
+     */
+    getMasterVolume() generates (Result retval, float volume);
+
+    /**
+     * Sets microphone muting state.
+     *
+     * @param mute whether microphone is muted.
+     * @return retval operation completion status.
+     */
+    setMicMute(bool mute) generates (Result retval);
+
+    /**
+     * Gets whether microphone is muted.
+     *
+     * @return retval operation completion status.
+     * @return mute whether microphone is muted.
+     */
+    getMicMute() generates (Result retval, bool mute);
+
+    /**
+     * Set the audio mute status for all audio activities. If the return value
+     * is NOT_SUPPORTED, the software mixer will emulate this capability.
+     *
+     * @param mute whether audio is muted.
+     * @return retval operation completion status.
+     */
+    setMasterMute(bool mute) generates (Result retval);
+
+    /**
+     * Get the current master mute status for the HAL, if the HAL supports
+     * master mute control. AudioFlinger will query this value from the primary
+     * audio HAL when the service starts and use the value for setting the
+     * initial master mute across all HALs. HAL must indicate that the feature
+     * is not supported by returning NOT_SUPPORTED status.
+     *
+     * @return retval operation completion status.
+     * @return mute whether audio is muted.
+     */
+    getMasterMute() generates (Result retval, bool mute);
+
+    /**
+     * Returns audio input buffer size according to parameters passed or
+     * INVALID_ARGUMENTS if one of the parameters is not supported.
+     *
+     * @param config audio configuration.
+     * @return retval operation completion status.
+     * @return bufferSize input buffer size in bytes.
+     */
+    getInputBufferSize(AudioConfig config)
+            generates (Result retval, uint64_t bufferSize);
+
+    /**
+     * This method creates and opens the audio hardware output stream.
+     * If the stream can not be opened with the proposed audio config,
+     * HAL must provide suggested values for the audio config.
+     *
+     * @param ioHandle handle assigned by AudioFlinger.
+     * @param device device type and (if needed) address.
+     * @param config stream configuration.
+     * @param flags additional flags.
+     * @param sourceMetadata Description of the audio that will be played.
+                             May be used by implementations to configure hardware effects.
+     * @return retval operation completion status.
+     * @return outStream created output stream.
+     * @return suggestedConfig in case of invalid parameters, suggested config.
+     */
+    openOutputStream(
+            AudioIoHandle ioHandle,
+            DeviceAddress device,
+            AudioConfig config,
+            bitfield<AudioOutputFlag> flags,
+            SourceMetadata sourceMetadata) generates (
+                    Result retval,
+                    IStreamOut outStream,
+                    AudioConfig suggestedConfig);
+
+    /**
+     * This method creates and opens the audio hardware input stream.
+     * If the stream can not be opened with the proposed audio config,
+     * HAL must provide suggested values for the audio config.
+     *
+     * @param ioHandle handle assigned by AudioFlinger.
+     * @param device device type and (if needed) address.
+     * @param config stream configuration.
+     * @param flags additional flags.
+     * @param sinkMetadata Description of the audio that is suggested by the client.
+     *                     May be used by implementations to configure hardware effects.
+     * @return retval operation completion status.
+     * @return inStream in case of success, created input stream.
+     * @return suggestedConfig in case of invalid parameters, suggested config.
+     */
+    openInputStream(
+            AudioIoHandle ioHandle,
+            DeviceAddress device,
+            AudioConfig config,
+            bitfield<AudioInputFlag> flags,
+            SinkMetadata sinkMetadata) generates (
+                    Result retval,
+                    IStreamIn inStream,
+                    AudioConfig suggestedConfig);
+
+    /**
+     * Returns whether HAL supports audio patches.
+     *
+     * @return supports true if audio patches are supported.
+     */
+    supportsAudioPatches() generates (bool supports);
+
+    /**
+     * Creates an audio patch between several source and sink ports.  The handle
+     * is allocated by the HAL and must be unique for this audio HAL module.
+     *
+     * @param sources patch sources.
+     * @param sinks patch sinks.
+     * @return retval operation completion status.
+     * @return patch created patch handle.
+     */
+    createAudioPatch(vec<AudioPortConfig> sources, vec<AudioPortConfig> sinks)
+            generates (Result retval, AudioPatchHandle patch);
+
+    /**
+     * Release an audio patch.
+     *
+     * @param patch patch handle.
+     * @return retval operation completion status.
+     */
+    releaseAudioPatch(AudioPatchHandle patch) generates (Result retval);
+
+    /**
+     * Returns the list of supported attributes for a given audio port.
+     *
+     * As input, 'port' contains the information (type, role, address etc...)
+     * needed by the HAL to identify the port.
+     *
+     * As output, 'resultPort' contains possible attributes (sampling rates,
+     * formats, channel masks, gain controllers...) for this port.
+     *
+     * @param port port identifier.
+     * @return retval operation completion status.
+     * @return resultPort port descriptor with all parameters filled up.
+     */
+    getAudioPort(AudioPort port)
+            generates (Result retval, AudioPort resultPort);
+
+    /**
+     * Set audio port configuration.
+     *
+     * @param config audio port configuration.
+     * @return retval operation completion status.
+     */
+    setAudioPortConfig(AudioPortConfig config) generates (Result retval);
+
+    /**
+     * Gets the HW synchronization source of the device. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_HW_AV_SYNC on the legacy HAL.
+     * Optional method
+     *
+     * @return retval operation completion status: OK or NOT_SUPPORTED.
+     * @return hwAvSync HW synchronization source
+     */
+    getHwAvSync() generates (Result retval, AudioHwSync hwAvSync);
+
+    /**
+     * Sets whether the screen is on. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_KEY_SCREEN_STATE on the legacy HAL.
+     * Optional method
+     *
+     * @param turnedOn whether the screen is turned on.
+     * @return retval operation completion status.
+     */
+    setScreenState(bool turnedOn) generates (Result retval);
+
+    /**
+     * Generic method for retrieving vendor-specific parameter values.
+     * The framework does not interpret the parameters, they are passed
+     * in an opaque manner between a vendor application and HAL.
+     *
+     * Multiple parameters can be retrieved at the same time.
+     * The implementation should return as many requested parameters
+     * as possible, even if one or more is not supported
+     *
+     * @param context provides more information about the request
+     * @param keys keys of the requested parameters
+     * @return retval operation completion status.
+     *         OK must be returned if keys is empty.
+     *         NOT_SUPPORTED must be returned if at least one key is unknown.
+     * @return parameters parameter key value pairs.
+     *         Must contain the value of all requested keys if retval == OK
+     */
+    getParameters(vec<ParameterValue> context, vec<string> keys)
+            generates (Result retval, vec<ParameterValue> parameters);
+
+    /**
+     * Generic method for setting vendor-specific parameter values.
+     * The framework does not interpret the parameters, they are passed
+     * in an opaque manner between a vendor application and HAL.
+     *
+     * Multiple parameters can be set at the same time though this is
+     * discouraged as it make failure analysis harder.
+     *
+     * If possible, a failed setParameters should not impact the platform state.
+     *
+     * @param context provides more information about the request
+     * @param parameters parameter key value pairs.
+     * @return retval operation completion status.
+     *         All parameters must be successfully set for OK to be returned
+     */
+    setParameters(vec<ParameterValue> context, vec<ParameterValue> parameters)
+            generates (Result retval);
+
+    /**
+     * Returns an array with available microphones in device.
+     *
+     * @return retval INVALID_STATE if the call is not successful,
+     *                OK otherwise.
+     *
+     * @return microphones array with microphones info
+     */
+    getMicrophones()
+         generates(Result retval, vec<MicrophoneInfo> microphones);
+
+    /**
+     * Notifies the device module about the connection state of an input/output
+     * device attached to it. Calling this method is equivalent to setting
+     * AUDIO_PARAMETER_DEVICE_[DIS]CONNECT on the legacy HAL.
+     *
+     * @param address audio device specification.
+     * @param connected whether the device is connected.
+     * @return retval operation completion status.
+     */
+    setConnectedState(DeviceAddress address, bool connected)
+            generates (Result retval);
+};
diff --git a/audio/5.0/IDevicesFactory.hal b/audio/5.0/IDevicesFactory.hal
new file mode 100644
index 0000000..92060e7
--- /dev/null
+++ b/audio/5.0/IDevicesFactory.hal
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+import IDevice;
+import IPrimaryDevice;
+
+/** This factory allows a HAL implementation to be split in multiple independent
+ *  devices (called module in the pre-treble API).
+ *  Note that this division is arbitrary and implementation are free
+ *  to only have a Primary.
+ *  The framework will query the devices according to audio_policy_configuration.xml
+ *
+ *  Each device name is arbitrary, provided by the vendor's audio_policy_configuration.xml
+ *  and only used to identify a device in this factory.
+ *  The framework must not interpret the name, treating it as a vendor opaque data
+ *  with the following exception:
+ *  - the "r_submix" device that must be present to support policyMixes (Eg: Android projected).
+ *    Note that this Device is included by default in a build derived from AOSP.
+ *
+ *  Note that on AOSP Oreo (including MR1) the "a2dp" module is not using this API
+ *  but is loaded directly from the system partition using the legacy API
+ *  due to limitations with the Bluetooth framework.
+ */
+interface IDevicesFactory {
+
+    /**
+     * Opens an audio device. To close the device, it is necessary to release
+     * references to the returned device object.
+     *
+     * @param device device name.
+     * @return retval operation completion status. Returns INVALID_ARGUMENTS
+     *         if there is no corresponding hardware module found,
+     *         NOT_INITIALIZED if an error occured while opening the hardware
+     *         module.
+     * @return result the interface for the created device.
+     */
+    openDevice(string device) generates (Result retval, IDevice result);
+
+    /**
+     * Opens the Primary audio device that must be present.
+     * This function is not optional and must return successfully the primary device.
+     *
+     * This device must have the name "primary".
+     *
+     * The telephony stack uses this device to control the audio during a voice call.
+     *
+     * @return retval operation completion status. Must be SUCCESS.
+     *         For debuging, return INVALID_ARGUMENTS if there is no corresponding
+     *         hardware module found, NOT_INITIALIZED if an error occurred
+     *         while opening the hardware module.
+     * @return result the interface for the created device.
+     */
+    openPrimaryDevice() generates (Result retval, IPrimaryDevice result);
+};
diff --git a/audio/5.0/IPrimaryDevice.hal b/audio/5.0/IPrimaryDevice.hal
new file mode 100644
index 0000000..60add5a
--- /dev/null
+++ b/audio/5.0/IPrimaryDevice.hal
@@ -0,0 +1,195 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+import IDevice;
+
+interface IPrimaryDevice extends IDevice {
+    /**
+     * Sets the audio volume of a voice call.
+     *
+     * @param volume 1.0f means unity, 0.0f is zero.
+     * @return retval operation completion status.
+     */
+    setVoiceVolume(float volume) generates (Result retval);
+
+    /**
+     * This method is used to notify the HAL about audio mode changes.
+     *
+     * @param mode new mode.
+     * @return retval operation completion status.
+     */
+    setMode(AudioMode mode) generates (Result retval);
+
+    /**
+     * Sets the name of the current BT SCO headset. Calling this method
+     * is equivalent to setting legacy "bt_headset_name" parameter.
+     * The BT SCO headset name must only be used for debugging purposes.
+     * Optional method
+     *
+     * @param name the name of the current BT SCO headset (can be empty).
+     * @return retval operation completion status.
+     */
+    setBtScoHeadsetDebugName(string name) generates (Result retval);
+
+    /**
+     * Gets whether BT SCO Noise Reduction and Echo Cancellation are enabled.
+     * Calling this method is equivalent to getting AUDIO_PARAMETER_KEY_BT_NREC
+     * on the legacy HAL.
+     *
+     * @return retval operation completion status.
+     * @return enabled whether BT SCO NR + EC are enabled.
+     */
+    getBtScoNrecEnabled() generates (Result retval, bool enabled);
+
+    /**
+     * Sets whether BT SCO Noise Reduction and Echo Cancellation are enabled.
+     * Calling this method is equivalent to setting AUDIO_PARAMETER_KEY_BT_NREC
+     * on the legacy HAL.
+     * Optional method
+     *
+     * @param enabled whether BT SCO NR + EC are enabled.
+     * @return retval operation completion status.
+     */
+    setBtScoNrecEnabled(bool enabled) generates (Result retval);
+
+    /**
+     * Gets whether BT SCO Wideband mode is enabled. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_KEY_BT_SCO_WB on the legacy HAL.
+     *
+     * @return retval operation completion status.
+     * @return enabled whether BT Wideband is enabled.
+     */
+    getBtScoWidebandEnabled() generates (Result retval, bool enabled);
+
+    /**
+     * Sets whether BT SCO Wideband mode is enabled. Calling this method is
+     * equivalent to setting AUDIO_PARAMETER_KEY_BT_SCO_WB on the legacy HAL.
+     * Optional method
+     *
+     * @param enabled whether BT Wideband is enabled.
+     * @return retval operation completion status.
+     */
+    setBtScoWidebandEnabled(bool enabled) generates (Result retval);
+
+    /**
+     * Gets whether BT HFP (Hands-Free Profile) is enabled. Calling this method
+     * is equivalent to getting "hfp_enable" parameter value on the legacy HAL.
+     *
+     * @return retval operation completion status.
+     * @return enabled whether BT HFP is enabled.
+     */
+    getBtHfpEnabled() generates (Result retval, bool enabled);
+
+    /**
+     * Sets whether BT HFP (Hands-Free Profile) is enabled. Calling this method
+     * is equivalent to setting "hfp_enable" parameter on the legacy HAL.
+     * Optional method
+     *
+     * @param enabled whether BT HFP is enabled.
+     * @return retval operation completion status.
+     */
+    setBtHfpEnabled(bool enabled) generates (Result retval);
+
+    /**
+     * Sets the sampling rate of BT HFP (Hands-Free Profile). Calling this
+     * method is equivalent to setting "hfp_set_sampling_rate" parameter
+     * on the legacy HAL.
+     * Optional method
+     *
+     * @param sampleRateHz sample rate in Hz.
+     * @return retval operation completion status.
+     */
+    setBtHfpSampleRate(uint32_t sampleRateHz) generates (Result retval);
+
+    /**
+     * Sets the current output volume Hz for BT HFP (Hands-Free Profile).
+     * Calling this method is equivalent to setting "hfp_volume" parameter value
+     * on the legacy HAL (except that legacy HAL implementations expect
+     * an integer value in the range from 0 to 15.)
+     * Optional method
+     *
+     * @param volume 1.0f means unity, 0.0f is zero.
+     * @return retval operation completion status.
+     */
+    setBtHfpVolume(float volume) generates (Result retval);
+
+    enum TtyMode : int32_t {
+        OFF,
+        VCO,
+        HCO,
+        FULL
+    };
+
+    /**
+     * Gets current TTY mode selection. Calling this method is equivalent to
+     * getting AUDIO_PARAMETER_KEY_TTY_MODE on the legacy HAL.
+     *
+     * @return retval operation completion status.
+     * @return mode TTY mode.
+     */
+    getTtyMode() generates (Result retval, TtyMode mode);
+
+    /**
+     * Sets current TTY mode. Calling this method is equivalent to setting
+     * AUDIO_PARAMETER_KEY_TTY_MODE on the legacy HAL.
+     *
+     * @param mode TTY mode.
+     * @return retval operation completion status.
+     */
+    setTtyMode(TtyMode mode) generates (Result retval);
+
+    /**
+     * Gets whether Hearing Aid Compatibility - Telecoil (HAC-T) mode is
+     * enabled. Calling this method is equivalent to getting
+     * AUDIO_PARAMETER_KEY_HAC on the legacy HAL.
+     *
+     * @return retval operation completion status.
+     * @return enabled whether HAC mode is enabled.
+     */
+    getHacEnabled() generates (Result retval, bool enabled);
+
+    /**
+     * Sets whether Hearing Aid Compatibility - Telecoil (HAC-T) mode is
+     * enabled. Calling this method is equivalent to setting
+     * AUDIO_PARAMETER_KEY_HAC on the legacy HAL.
+     * Optional method
+     *
+     * @param enabled whether HAC mode is enabled.
+     * @return retval operation completion status.
+     */
+    setHacEnabled(bool enabled) generates (Result retval);
+
+    enum Rotation : int32_t {
+        DEG_0,
+        DEG_90,
+        DEG_180,
+        DEG_270
+    };
+
+    /**
+     * Updates HAL on the current rotation of the device relative to natural
+     * orientation. Calling this method is equivalent to setting legacy
+     * parameter "rotation".
+     *
+     * @param rotation rotation in degrees relative to natural device
+     *     orientation.
+     * @return retval operation completion status.
+     */
+    updateRotation(Rotation rotation) generates (Result retval);
+};
diff --git a/audio/5.0/IStream.hal b/audio/5.0/IStream.hal
new file mode 100644
index 0000000..0f975b6
--- /dev/null
+++ b/audio/5.0/IStream.hal
@@ -0,0 +1,310 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+import android.hardware.audio.effect@5.0::IEffect;
+
+interface IStream {
+    /**
+     * Return the frame size (number of bytes per sample).
+     *
+     * @return frameSize frame size in bytes.
+     */
+    getFrameSize() generates (uint64_t frameSize);
+
+    /**
+     * Return the frame count of the buffer. Calling this method is equivalent
+     * to getting AUDIO_PARAMETER_STREAM_FRAME_COUNT on the legacy HAL.
+     *
+     * @return count frame count.
+     */
+    getFrameCount() generates (uint64_t count);
+
+    /**
+     * Return the size of input/output buffer in bytes for this stream.
+     * It must be a multiple of the frame size.
+     *
+     * @return buffer buffer size in bytes.
+     */
+    getBufferSize() generates (uint64_t bufferSize);
+
+    /**
+     * Return the sampling rate in Hz.
+     *
+     * @return sampleRateHz sample rate in Hz.
+     */
+    getSampleRate() generates (uint32_t sampleRateHz);
+
+    /**
+     * Return supported native sampling rates of the stream for a given format.
+     * A supported native sample rate is a sample rate that can be efficiently
+     * played by the hardware (typically without sample-rate conversions).
+     *
+     * This function is only called for dynamic profile. If called for
+     * non-dynamic profile is should return NOT_SUPPORTED or the same list
+     * as in audio_policy_configuration.xml.
+     *
+     * Calling this method is equivalent to getting
+     * AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES on the legacy HAL.
+     *
+     *
+     * @param format audio format for which the sample rates are supported.
+     * @return retval operation completion status.
+     *                Must be OK if the format is supported.
+     * @return sampleRateHz supported sample rates.
+     */
+    getSupportedSampleRates(AudioFormat format)
+            generates (Result retval, vec<uint32_t> sampleRates);
+
+    /**
+     * Sets the sampling rate of the stream. Calling this method is equivalent
+     * to setting AUDIO_PARAMETER_STREAM_SAMPLING_RATE on the legacy HAL.
+     * Optional method. If implemented, only called on a stopped stream.
+     *
+     * @param sampleRateHz sample rate in Hz.
+     * @return retval operation completion status.
+     */
+    setSampleRate(uint32_t sampleRateHz) generates (Result retval);
+
+    /**
+     * Return the channel mask of the stream.
+     *
+     * @return mask channel mask.
+     */
+    getChannelMask() generates (bitfield<AudioChannelMask> mask);
+
+    /**
+     * Return supported channel masks of the stream. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_STREAM_SUP_CHANNELS on the legacy
+     * HAL.
+     *
+     * @param format audio format for which the channel masks are supported.
+     * @return retval operation completion status.
+     *                Must be OK if the format is supported.
+     * @return masks supported audio masks.
+     */
+    getSupportedChannelMasks(AudioFormat format)
+            generates (Result retval, vec<bitfield<AudioChannelMask>> masks);
+
+    /**
+     * Sets the channel mask of the stream. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_STREAM_CHANNELS on the legacy HAL.
+     * Optional method
+     *
+     * @param format audio format.
+     * @return retval operation completion status.
+     */
+    setChannelMask(bitfield<AudioChannelMask> mask) generates (Result retval);
+
+    /**
+     * Return the audio format of the stream.
+     *
+     * @return format audio format.
+     */
+    getFormat() generates (AudioFormat format);
+
+    /**
+     * Return supported audio formats of the stream. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_STREAM_SUP_FORMATS on the legacy
+     * HAL.
+     *
+     * @return formats supported audio formats.
+     */
+    getSupportedFormats() generates (vec<AudioFormat> formats);
+
+    /**
+     * Sets the audio format of the stream. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_STREAM_FORMAT on the legacy HAL.
+     * Optional method
+     *
+     * @param format audio format.
+     * @return retval operation completion status.
+     */
+    setFormat(AudioFormat format) generates (Result retval);
+
+    /**
+     * Convenience method for retrieving several stream parameters in
+     * one transaction.
+     *
+     * @return sampleRateHz sample rate in Hz.
+     * @return mask channel mask.
+     * @return format audio format.
+     */
+    getAudioProperties() generates (
+            uint32_t sampleRateHz, bitfield<AudioChannelMask> mask, AudioFormat format);
+
+    /**
+     * Applies audio effect to the stream.
+     *
+     * @param effectId effect ID (obtained from IEffectsFactory.createEffect) of
+     *                 the effect to apply.
+     * @return retval operation completion status.
+     */
+    addEffect(uint64_t effectId) generates (Result retval);
+
+    /**
+     * Stops application of the effect to the stream.
+     *
+     * @param effectId effect ID (obtained from IEffectsFactory.createEffect) of
+     *                 the effect to remove.
+     * @return retval operation completion status.
+     */
+    removeEffect(uint64_t effectId) generates (Result retval);
+
+    /**
+     * Put the audio hardware input/output into standby mode.
+     * Driver must exit from standby mode at the next I/O operation.
+     *
+     * @return retval operation completion status.
+     */
+    standby() generates (Result retval);
+
+    /**
+     * Return the set of devices which this stream is connected to.
+     * Optional method
+     *
+     * @return retval operation completion status: OK or NOT_SUPPORTED.
+     * @return device set of devices which this stream is connected to.
+     */
+    getDevices() generates (Result retval, vec<DeviceAddress> devices);
+
+    /**
+     * Connects the stream to one or multiple devices.
+     *
+     * This method must only be used for HALs that do not support
+     * 'IDevice.createAudioPatch' method. Calling this method is
+     * equivalent to setting AUDIO_PARAMETER_STREAM_ROUTING preceeded
+     * with a device address in the legacy HAL interface.
+     *
+     * @param address device to connect the stream to.
+     * @return retval operation completion status.
+     */
+    setDevices(vec<DeviceAddress> devices) generates (Result retval);
+
+    /**
+     * Sets the HW synchronization source. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_STREAM_HW_AV_SYNC on the legacy HAL.
+     * Optional method
+     *
+     * @param hwAvSync HW synchronization source
+     * @return retval operation completion status.
+     */
+    setHwAvSync(AudioHwSync hwAvSync) generates (Result retval);
+
+    /**
+     * Generic method for retrieving vendor-specific parameter values.
+     * The framework does not interpret the parameters, they are passed
+     * in an opaque manner between a vendor application and HAL.
+     *
+     * Multiple parameters can be retrieved at the same time.
+     * The implementation should return as many requested parameters
+     * as possible, even if one or more is not supported
+     *
+     * @param context provides more information about the request
+     * @param keys keys of the requested parameters
+     * @return retval operation completion status.
+     *         OK must be returned if keys is empty.
+     *         NOT_SUPPORTED must be returned if at least one key is unknown.
+     * @return parameters parameter key value pairs.
+     *         Must contain the value of all requested keys if retval == OK
+     */
+    getParameters(vec<ParameterValue> context, vec<string> keys)
+            generates (Result retval, vec<ParameterValue> parameters);
+
+    /**
+     * Generic method for setting vendor-specific parameter values.
+     * The framework does not interpret the parameters, they are passed
+     * in an opaque manner between a vendor application and HAL.
+     *
+     * Multiple parameters can be set at the same time though this is
+     * discouraged as it make failure analysis harder.
+     *
+     * If possible, a failed setParameters should not impact the platform state.
+     *
+     * @param context provides more information about the request
+     * @param parameters parameter key value pairs.
+     * @return retval operation completion status.
+     *         All parameters must be successfully set for OK to be returned
+     */
+    setParameters(vec<ParameterValue> context, vec<ParameterValue> parameters)
+            generates (Result retval);
+
+    /**
+     * Called by the framework to start a stream operating in mmap mode.
+     * createMmapBuffer() must be called before calling start().
+     * Function only implemented by streams operating in mmap mode.
+     *
+     * @return retval OK in case the success.
+     *                NOT_SUPPORTED on non mmap mode streams
+     *                INVALID_STATE if called out of sequence
+     */
+    start() generates (Result retval);
+
+    /**
+     * Called by the framework to stop a stream operating in mmap mode.
+     * Function only implemented by streams operating in mmap mode.
+     *
+     * @return retval OK in case the succes.
+     *                NOT_SUPPORTED on non mmap mode streams
+     *                INVALID_STATE if called out of sequence
+     */
+    stop() generates (Result retval) ;
+
+    /**
+     * Called by the framework to retrieve information on the mmap buffer used for audio
+     * samples transfer.
+     * Function only implemented by streams operating in mmap mode.
+     *
+     * @param minSizeFrames minimum buffer size requested. The actual buffer
+     *                     size returned in struct MmapBufferInfo can be larger.
+     * @return retval OK in case the success.
+     *                NOT_SUPPORTED on non mmap mode streams
+     *                NOT_INITIALIZED in case of memory allocation error
+     *                INVALID_ARGUMENTS if the requested buffer size is too large
+     *                INVALID_STATE if called out of sequence
+     * @return info    a MmapBufferInfo struct containing information on the MMMAP buffer created.
+     */
+    createMmapBuffer(int32_t minSizeFrames)
+            generates (Result retval, MmapBufferInfo info);
+
+    /**
+     * Called by the framework to read current read/write position in the mmap buffer
+     * with associated time stamp.
+     * Function only implemented by streams operating in mmap mode.
+     *
+     * @return retval OK in case the success.
+     *                NOT_SUPPORTED on non mmap mode streams
+     *                INVALID_STATE if called out of sequence
+     * @return position a MmapPosition struct containing current HW read/write position in frames
+     *                  with associated time stamp.
+     */
+    getMmapPosition()
+            generates (Result retval, MmapPosition position);
+
+    /**
+     * Called by the framework to deinitialize the stream and free up
+     * all the currently allocated resources. It is recommended to close
+     * the stream on the client side as soon as it is becomes unused.
+     *
+     * @return retval OK in case the success.
+     *                NOT_SUPPORTED if called on IStream instead of input or
+     *                              output stream interface.
+     *                INVALID_STATE if the stream was already closed.
+     */
+    close() generates (Result retval);
+};
diff --git a/audio/5.0/IStreamIn.hal b/audio/5.0/IStreamIn.hal
new file mode 100644
index 0000000..d33cfdc
--- /dev/null
+++ b/audio/5.0/IStreamIn.hal
@@ -0,0 +1,168 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+import IStream;
+
+interface IStreamIn extends IStream {
+    /**
+     * Returns the source descriptor of the input stream. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_STREAM_INPUT_SOURCE on the legacy
+     * HAL.
+     * Optional method
+     *
+     * @return retval operation completion status.
+     * @return source audio source.
+     */
+    getAudioSource() generates (Result retval, AudioSource source);
+
+    /**
+     * Set the input gain for the audio driver.
+     * Optional method
+     *
+     * @param gain 1.0f is unity, 0.0f is zero.
+     * @result retval operation completion status.
+     */
+    setGain(float gain) generates (Result retval);
+
+    /**
+     * Commands that can be executed on the driver reader thread.
+     */
+    enum ReadCommand : int32_t {
+        READ,
+        GET_CAPTURE_POSITION
+    };
+
+    /**
+     * Data structure passed to the driver for executing commands
+     * on the driver reader thread.
+     */
+    struct ReadParameters {
+        ReadCommand command;  // discriminator
+        union Params {
+            uint64_t read;    // READ command, amount of bytes to read, >= 0.
+            // No parameters for GET_CAPTURE_POSITION.
+        } params;
+    };
+
+    /**
+     * Data structure passed back to the client via status message queue
+     * of 'read' operation.
+     *
+     * Possible values of 'retval' field:
+     *  - OK, read operation was successful;
+     *  - INVALID_ARGUMENTS, stream was not configured properly;
+     *  - INVALID_STATE, stream is in a state that doesn't allow reads.
+     */
+    struct ReadStatus {
+        Result retval;
+        ReadCommand replyTo;  // discriminator
+        union Reply {
+            uint64_t read;    // READ command, amount of bytes read, >= 0.
+            struct CapturePosition { // same as generated by getCapturePosition.
+                uint64_t frames;
+                uint64_t time;
+            } capturePosition;
+        } reply;
+    };
+
+    /**
+     * Called when the metadata of the stream's sink has been changed.
+     * @param sinkMetadata Description of the audio that is suggested by the clients.
+     */
+    updateSinkMetadata(SinkMetadata sinkMetadata);
+
+    /**
+     * Set up required transports for receiving audio buffers from the driver.
+     *
+     * The transport consists of three message queues:
+     *  -- command queue is used to instruct the reader thread what operation
+     *     to perform;
+     *  -- data queue is used for passing audio data from the driver
+     *     to the client;
+     *  -- status queue is used for reporting operation status
+     *     (e.g. amount of bytes actually read or error code).
+     *
+     * The driver operates on a dedicated thread. The client must ensure that
+     * the thread is given an appropriate priority and assigned to correct
+     * scheduler and cgroup. For this purpose, the method returns identifiers
+     * of the driver thread.
+     *
+     * @param frameSize the size of a single frame, in bytes.
+     * @param framesCount the number of frames in a buffer.
+     * @param threadPriority priority of the driver thread.
+     * @return retval OK if both message queues were created successfully.
+     *                INVALID_STATE if the method was already called.
+     *                INVALID_ARGUMENTS if there was a problem setting up
+     *                                  the queues.
+     * @return commandMQ a message queue used for passing commands.
+     * @return dataMQ a message queue used for passing audio data in the format
+     *                specified at the stream opening.
+     * @return statusMQ a message queue used for passing status from the driver
+     *                  using ReadStatus structures.
+     * @return threadInfo identifiers of the driver's dedicated thread.
+     */
+    prepareForReading(uint32_t frameSize, uint32_t framesCount)
+    generates (
+            Result retval,
+            fmq_sync<ReadParameters> commandMQ,
+            fmq_sync<uint8_t> dataMQ,
+            fmq_sync<ReadStatus> statusMQ,
+            ThreadInfo threadInfo);
+
+    /**
+     * Return the amount of input frames lost in the audio driver since the last
+     * call of this function.
+     *
+     * Audio driver is expected to reset the value to 0 and restart counting
+     * upon returning the current value by this function call. Such loss
+     * typically occurs when the user space process is blocked longer than the
+     * capacity of audio driver buffers.
+     *
+     * @return framesLost the number of input audio frames lost.
+     */
+    getInputFramesLost() generates (uint32_t framesLost);
+
+    /**
+     * Return a recent count of the number of audio frames received and the
+     * clock time associated with that frame count.
+     *
+     * @return retval INVALID_STATE if the device is not ready/available,
+     *                NOT_SUPPORTED if the command is not supported,
+     *                OK otherwise.
+     * @return frames the total frame count received. This must be as early in
+     *                the capture pipeline as possible. In general, frames
+     *                must be non-negative and must not go "backwards".
+     * @return time is the clock monotonic time when frames was measured. In
+     *              general, time must be a positive quantity and must not
+     *              go "backwards".
+     */
+    getCapturePosition()
+            generates (Result retval, uint64_t frames, uint64_t time);
+
+    /**
+     * Returns an array with active microphones in the stream.
+     *
+     * @return retval INVALID_STATE if the call is not successful,
+     *                OK otherwise.
+     *
+     * @return microphones array with microphones info
+     */
+    getActiveMicrophones()
+               generates(Result retval, vec<MicrophoneInfo> microphones);
+};
diff --git a/audio/5.0/IStreamOut.hal b/audio/5.0/IStreamOut.hal
new file mode 100644
index 0000000..119a642
--- /dev/null
+++ b/audio/5.0/IStreamOut.hal
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+import IStream;
+import IStreamOutCallback;
+
+interface IStreamOut extends IStream {
+    /**
+     * Return the audio hardware driver estimated latency in milliseconds.
+     *
+     * @return latencyMs latency in milliseconds.
+     */
+    getLatency() generates (uint32_t latencyMs);
+
+    /**
+     * This method is used in situations where audio mixing is done in the
+     * hardware. This method serves as a direct interface with hardware,
+     * allowing to directly set the volume as apposed to via the framework.
+     * This method might produce multiple PCM outputs or hardware accelerated
+     * codecs, such as MP3 or AAC.
+     * Optional method
+     *
+     * @param left left channel attenuation, 1.0f is unity, 0.0f is zero.
+     * @param right right channel attenuation, 1.0f is unity, 0.0f is zero.
+     * @return retval operation completion status.
+     *        If a volume is outside [0,1], return INVALID_ARGUMENTS
+     */
+    setVolume(float left, float right) generates (Result retval);
+
+    /**
+     * Commands that can be executed on the driver writer thread.
+     */
+    enum WriteCommand : int32_t {
+        WRITE,
+        GET_PRESENTATION_POSITION,
+        GET_LATENCY
+    };
+
+    /**
+     * Data structure passed back to the client via status message queue
+     * of 'write' operation.
+     *
+     * Possible values of 'retval' field:
+     *  - OK, write operation was successful;
+     *  - INVALID_ARGUMENTS, stream was not configured properly;
+     *  - INVALID_STATE, stream is in a state that doesn't allow writes;
+     *  - INVALID_OPERATION, retrieving presentation position isn't supported.
+     */
+    struct WriteStatus {
+        Result retval;
+        WriteCommand replyTo;  // discriminator
+        union Reply {
+            uint64_t written;  // WRITE command, amount of bytes written, >= 0.
+            struct PresentationPosition {  // same as generated by
+                uint64_t frames;           // getPresentationPosition.
+                TimeSpec timeStamp;
+            } presentationPosition;
+            uint32_t latencyMs; // Same as generated by getLatency.
+        } reply;
+    };
+
+    /**
+     * Called when the metadata of the stream's source has been changed.
+     * @param sourceMetadata Description of the audio that is played by the clients.
+     */
+    updateSourceMetadata(SourceMetadata sourceMetadata);
+
+    /**
+     * Set up required transports for passing audio buffers to the driver.
+     *
+     * The transport consists of three message queues:
+     *  -- command queue is used to instruct the writer thread what operation
+     *     to perform;
+     *  -- data queue is used for passing audio data from the client
+     *     to the driver;
+     *  -- status queue is used for reporting operation status
+     *     (e.g. amount of bytes actually written or error code).
+     *
+     * The driver operates on a dedicated thread. The client must ensure that
+     * the thread is given an appropriate priority and assigned to correct
+     * scheduler and cgroup. For this purpose, the method returns identifiers
+     * of the driver thread.
+     *
+     * @param frameSize the size of a single frame, in bytes.
+     * @param framesCount the number of frames in a buffer.
+     * @return retval OK if both message queues were created successfully.
+     *                INVALID_STATE if the method was already called.
+     *                INVALID_ARGUMENTS if there was a problem setting up
+     *                                  the queues.
+     * @return commandMQ a message queue used for passing commands.
+     * @return dataMQ a message queue used for passing audio data in the format
+     *                specified at the stream opening.
+     * @return statusMQ a message queue used for passing status from the driver
+     *                  using WriteStatus structures.
+     * @return threadInfo identifiers of the driver's dedicated thread.
+     */
+    prepareForWriting(uint32_t frameSize, uint32_t framesCount)
+    generates (
+            Result retval,
+            fmq_sync<WriteCommand> commandMQ,
+            fmq_sync<uint8_t> dataMQ,
+            fmq_sync<WriteStatus> statusMQ,
+            ThreadInfo threadInfo);
+
+    /**
+     * Return the number of audio frames written by the audio DSP to DAC since
+     * the output has exited standby.
+     * Optional method
+     *
+     * @return retval operation completion status.
+     * @return dspFrames number of audio frames written.
+     */
+    getRenderPosition() generates (Result retval, uint32_t dspFrames);
+
+    /**
+     * Get the local time at which the next write to the audio driver will be
+     * presented. The units are microseconds, where the epoch is decided by the
+     * local audio HAL.
+     * Optional method
+     *
+     * @return retval operation completion status.
+     * @return timestampUs time of the next write.
+     */
+    getNextWriteTimestamp() generates (Result retval, int64_t timestampUs);
+
+    /**
+     * Set the callback interface for notifying completion of non-blocking
+     * write and drain.
+     *
+     * Calling this function implies that all future 'write' and 'drain'
+     * must be non-blocking and use the callback to signal completion.
+     *
+     * 'clearCallback' method needs to be called in order to release the local
+     * callback proxy on the server side and thus dereference the callback
+     * implementation on the client side.
+     *
+     * @return retval operation completion status.
+     */
+    setCallback(IStreamOutCallback callback) generates (Result retval);
+
+    /**
+     * Clears the callback previously set via 'setCallback' method.
+     *
+     * Warning: failure to call this method results in callback implementation
+     * on the client side being held until the HAL server termination.
+     *
+     * If no callback was previously set, the method should be a no-op
+     * and return OK.
+     *
+     * @return retval operation completion status: OK or NOT_SUPPORTED.
+     */
+    clearCallback() generates (Result retval);
+
+    /**
+     * Returns whether HAL supports pausing and resuming of streams.
+     *
+     * @return supportsPause true if pausing is supported.
+     * @return supportsResume true if resume is supported.
+     */
+    supportsPauseAndResume()
+            generates (bool supportsPause, bool supportsResume);
+
+    /**
+     * Notifies to the audio driver to stop playback however the queued buffers
+     * are retained by the hardware. Useful for implementing pause/resume. Empty
+     * implementation if not supported however must be implemented for hardware
+     * with non-trivial latency. In the pause state, some audio hardware may
+     * still be using power. Client code may consider calling 'suspend' after a
+     * timeout to prevent that excess power usage.
+     *
+     * Implementation of this function is mandatory for offloaded playback.
+     *
+     * @return retval operation completion status.
+     */
+    pause() generates (Result retval);
+
+    /**
+     * Notifies to the audio driver to resume playback following a pause.
+     * Returns error INVALID_STATE if called without matching pause.
+     *
+     * Implementation of this function is mandatory for offloaded playback.
+     *
+     * @return retval operation completion status.
+     */
+    resume() generates (Result retval);
+
+    /**
+     * Returns whether HAL supports draining of streams.
+     *
+     * @return supports true if draining is supported.
+     */
+    supportsDrain() generates (bool supports);
+
+    /**
+     * Requests notification when data buffered by the driver/hardware has been
+     * played. If 'setCallback' has previously been called to enable
+     * non-blocking mode, then 'drain' must not block, instead it must return
+     * quickly and completion of the drain is notified through the callback. If
+     * 'setCallback' has not been called, then 'drain' must block until
+     * completion.
+     *
+     * If 'type' is 'ALL', the drain completes when all previously written data
+     * has been played.
+     *
+     * If 'type' is 'EARLY_NOTIFY', the drain completes shortly before all data
+     * for the current track has played to allow time for the framework to
+     * perform a gapless track switch.
+     *
+     * Drain must return immediately on 'stop' and 'flush' calls.
+     *
+     * Implementation of this function is mandatory for offloaded playback.
+     *
+     * @param type type of drain.
+     * @return retval operation completion status.
+     */
+    drain(AudioDrain type) generates (Result retval);
+
+    /**
+     * Notifies to the audio driver to flush the queued data. Stream must
+     * already be paused before calling 'flush'.
+     * Optional method
+     *
+     * Implementation of this function is mandatory for offloaded playback.
+     *
+     * @return retval operation completion status.
+     */
+    flush() generates (Result retval);
+
+    /**
+     * Return a recent count of the number of audio frames presented to an
+     * external observer. This excludes frames which have been written but are
+     * still in the pipeline. The count is not reset to zero when output enters
+     * standby. Also returns the value of CLOCK_MONOTONIC as of this
+     * presentation count. The returned count is expected to be 'recent', but
+     * does not need to be the most recent possible value. However, the
+     * associated time must correspond to whatever count is returned.
+     *
+     * Example: assume that N+M frames have been presented, where M is a 'small'
+     * number. Then it is permissible to return N instead of N+M, and the
+     * timestamp must correspond to N rather than N+M. The terms 'recent' and
+     * 'small' are not defined. They reflect the quality of the implementation.
+     *
+     * Optional method
+     *
+     * @return retval operation completion status.
+     * @return frames count of presented audio frames.
+     * @return timeStamp associated clock time.
+     */
+    getPresentationPosition()
+            generates (Result retval, uint64_t frames, TimeSpec timeStamp);
+
+    /**
+     * Selects a presentation for decoding from a next generation media stream
+     * (as defined per ETSI TS 103 190-2) and a program within the presentation.
+     * Optional method
+     *
+     * @param presentationId selected audio presentation.
+     * @param programId refinement for the presentation.
+     * @return retval operation completion status.
+     */
+    selectPresentation(int32_t presentationId, int32_t programId)
+            generates (Result retval);
+};
diff --git a/audio/5.0/IStreamOutCallback.hal b/audio/5.0/IStreamOutCallback.hal
new file mode 100644
index 0000000..c52a040
--- /dev/null
+++ b/audio/5.0/IStreamOutCallback.hal
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+/**
+ * Asynchronous write callback interface.
+ */
+interface IStreamOutCallback {
+    /**
+     * Non blocking write completed.
+     */
+    oneway onWriteReady();
+
+    /**
+     * Drain completed.
+     */
+    oneway onDrainReady();
+
+    /**
+     * Stream hit an error.
+     */
+    oneway onError();
+};
diff --git a/audio/5.0/config/audio_policy_configuration.xsd b/audio/5.0/config/audio_policy_configuration.xsd
new file mode 100644
index 0000000..311b9c1
--- /dev/null
+++ b/audio/5.0/config/audio_policy_configuration.xsd
@@ -0,0 +1,595 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2017 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.
+-->
+<!-- TODO: define a targetNamespace. Note that it will break retrocompatibility -->
+<xs:schema version="2.0"
+           elementFormDefault="qualified"
+           attributeFormDefault="unqualified"
+           xmlns:xs="http://www.w3.org/2001/XMLSchema">
+    <!-- List the config versions supported by audio policy. -->
+    <xs:simpleType name="version">
+        <xs:restriction base="xs:decimal">
+            <xs:enumeration value="1.0"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:simpleType name="halVersion">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Version of the interface the hal implements.
+            </xs:documentation>
+        </xs:annotation>
+        <xs:restriction base="xs:decimal">
+            <!-- List of HAL versions supported by the framework. -->
+            <xs:enumeration value="2.0"/>
+            <xs:enumeration value="3.0"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:element name="audioPolicyConfiguration">
+        <xs:complexType>
+            <xs:sequence>
+                <xs:element name="globalConfiguration" type="globalConfiguration"/>
+                <xs:element name="modules" type="modules" maxOccurs="unbounded"/>
+                <xs:element name="volumes" type="volumes" maxOccurs="unbounded"/>
+                <xs:element name="surroundSound" type="surroundSound" />
+            </xs:sequence>
+            <xs:attribute name="version" type="version"/>
+        </xs:complexType>
+        <xs:key name="moduleNameKey">
+            <xs:selector xpath="modules/module"/>
+            <xs:field xpath="@name"/>
+        </xs:key>
+        <xs:unique name="volumeTargetUniqueness">
+            <xs:selector xpath="volumes/volume"/>
+            <xs:field xpath="@stream"/>
+            <xs:field xpath="@deviceCategory"/>
+        </xs:unique>
+        <xs:key name="volumeCurveNameKey">
+            <xs:selector xpath="volumes/reference"/>
+            <xs:field xpath="@name"/>
+        </xs:key>
+        <xs:keyref name="volumeCurveRef" refer="volumeCurveNameKey">
+            <xs:selector xpath="volumes/volume"/>
+            <xs:field xpath="@ref"/>
+        </xs:keyref>
+    </xs:element>
+    <xs:complexType name="globalConfiguration">
+        <xs:attribute name="speaker_drc_enabled" type="xs:boolean" use="required"/>
+    </xs:complexType>
+    <xs:complexType name="modules">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                There should be one section per audio HW module present on the platform.
+                Each <module/> contains two mandatory tags: “halVersion” and “name”.
+                The module "name" is the same as in previous .conf file.
+                Each module must contain the following sections:
+                 - <devicePorts/>: a list of device descriptors for all
+                   input and output devices accessible via this module.
+                   This contains both permanently attached devices and removable devices.
+                 - <mixPorts/>: listing all output and input streams exposed by the audio HAL
+                 - <routes/>: list of possible connections between input
+                   and output devices or between stream and devices.
+                   A <route/> is defined by a set of 3 attributes:
+                        -"type": mux|mix means all sources are mutual exclusive (mux) or can be mixed (mix)
+                        -"sink": the sink involved in this route
+                        -"sources": all the sources than can be connected to the sink via this route
+                 - <attachedDevices/>: permanently attached devices.
+                   The attachedDevices section is a list of devices names.
+                   Their names correspond to device names defined in "devicePorts" section.
+                 - <defaultOutputDevice/> is the device to be used when no policy rule applies
+            </xs:documentation>
+        </xs:annotation>
+        <xs:sequence>
+            <xs:element name="module" maxOccurs="unbounded">
+                <xs:complexType>
+                    <xs:sequence>
+                        <xs:element name="attachedDevices" type="attachedDevices" minOccurs="0">
+                            <xs:unique name="attachedDevicesUniqueness">
+                                <xs:selector xpath="item"/>
+                                <xs:field xpath="."/>
+                            </xs:unique>
+                        </xs:element>
+                        <xs:element name="defaultOutputDevice" type="xs:token" minOccurs="0"/>
+                        <xs:element name="mixPorts" type="mixPorts" minOccurs="0"/>
+                        <xs:element name="devicePorts" type="devicePorts" minOccurs="0"/>
+                        <xs:element name="routes" type="routes" minOccurs="0"/>
+                    </xs:sequence>
+                    <xs:attribute name="name" type="xs:string" use="required"/>
+                    <xs:attribute name="halVersion" type="halVersion" use="required"/>
+                </xs:complexType>
+                <xs:unique name="mixPortNameUniqueness">
+                    <xs:selector xpath="mixPorts/mixPort"/>
+                    <xs:field xpath="@name"/>
+                </xs:unique>
+                <xs:key name="devicePortNameKey">
+                    <xs:selector xpath="devicePorts/devicePort"/>
+                    <xs:field xpath="@tagName"/>
+                </xs:key>
+                <xs:unique name="devicePortUniqueness">
+                    <xs:selector xpath="devicePorts/devicePort"/>
+                    <xs:field xpath="@type"/>
+                    <xs:field xpath="@address"/>
+                </xs:unique>
+                <xs:keyref name="defaultOutputDeviceRef" refer="devicePortNameKey">
+                    <xs:selector xpath="defaultOutputDevice"/>
+                    <xs:field xpath="."/>
+                </xs:keyref>
+                <xs:keyref name="attachedDeviceRef" refer="devicePortNameKey">
+                    <xs:selector xpath="attachedDevices/item"/>
+                    <xs:field xpath="."/>
+                </xs:keyref>
+                <!-- The following 3 constraints try to make sure each sink port
+                     is reference in one an only one route. -->
+                <xs:key name="routeSinkKey">
+                    <!-- predicate [@type='sink'] does not work in xsd 1.0 -->
+                    <xs:selector xpath="devicePorts/devicePort|mixPorts/mixPort"/>
+                    <xs:field xpath="@tagName|@name"/>
+                </xs:key>
+                <xs:keyref name="routeSinkRef" refer="routeSinkKey">
+                    <xs:selector xpath="routes/route"/>
+                    <xs:field xpath="@sink"/>
+                </xs:keyref>
+                <xs:unique name="routeUniqueness">
+                    <xs:selector xpath="routes/route"/>
+                    <xs:field xpath="@sink"/>
+                </xs:unique>
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+    <xs:complexType name="attachedDevices">
+        <xs:sequence>
+            <xs:element name="item" type="xs:token" minOccurs="0" maxOccurs="unbounded"/>
+        </xs:sequence>
+    </xs:complexType>
+    <!-- TODO: separate values by space for better xsd validations. -->
+    <xs:simpleType name="audioInOutFlags">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                "|" separated list of audio_output_flags_t or audio_input_flags_t.
+            </xs:documentation>
+        </xs:annotation>
+        <xs:restriction base="xs:string">
+            <xs:pattern value="|[_A-Z]+(\|[_A-Z]+)*"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:simpleType name="role">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="sink"/>
+            <xs:enumeration value="source"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:complexType name="mixPorts">
+        <xs:sequence>
+            <xs:element name="mixPort" minOccurs="0" maxOccurs="unbounded">
+                <xs:complexType>
+                    <xs:sequence>
+                        <xs:element name="profile" type="profile" minOccurs="0" maxOccurs="unbounded"/>
+                        <xs:element name="gains" type="gains" minOccurs="0"/>
+                    </xs:sequence>
+                    <xs:attribute name="name" type="xs:token" use="required"/>
+                    <xs:attribute name="role" type="role" use="required"/>
+                    <xs:attribute name="flags" type="audioInOutFlags"/>
+                    <xs:attribute name="maxOpenCount" type="xs:unsignedInt"/>
+                    <xs:attribute name="maxActiveCount" type="xs:unsignedInt"/>
+                    <xs:attribute name="preferredUsage" type="audioUsageList">
+                        <xs:annotation>
+                            <xs:documentation xml:lang="en">
+                                When choosing the mixPort of an audio track, the audioPolicy
+                                first considers the mixPorts with a preferredUsage including
+                                the track AudioUsage preferred .
+                                If non support the track format, the other mixPorts are considered.
+                                Eg: a <mixPort preferredUsage="AUDIO_USAGE_MEDIA" /> will receive
+                                    the audio of all apps playing with a MEDIA usage.
+                                    It may receive audio from ALARM if there are no audio compatible
+                                    <mixPort preferredUsage="AUDIO_USAGE_ALARM" />.
+                             </xs:documentation>
+                        </xs:annotation>
+                    </xs:attribute>
+                </xs:complexType>
+                <xs:unique name="mixPortProfileUniqueness">
+                    <xs:selector xpath="profile"/>
+                    <xs:field xpath="format"/>
+                    <xs:field xpath="samplingRate"/>
+                    <xs:field xpath="channelMasks"/>
+                </xs:unique>
+                <xs:unique name="mixPortGainUniqueness">
+                    <xs:selector xpath="gains/gain"/>
+                    <xs:field xpath="@name"/>
+                </xs:unique>
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+    <!-- Enum values of audio_device_t in audio.h
+         TODO: generate from hidl to avoid manual sync.
+         TODO: separate source and sink in the xml for better xsd validations. -->
+    <xs:simpleType name="audioDevice">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_DEVICE_NONE"/>
+
+            <xs:enumeration value="AUDIO_DEVICE_OUT_EARPIECE"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_SPEAKER"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_WIRED_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_WIRED_HEADPHONE"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BLUETOOTH_SCO"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_AUX_DIGITAL"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_HDMI"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_USB_ACCESSORY"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_USB_DEVICE"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_REMOTE_SUBMIX"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_TELEPHONY_TX"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_LINE"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_HDMI_ARC"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_SPDIF"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_FM"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_AUX_LINE"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_SPEAKER_SAFE"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_IP"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_BUS"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_PROXY"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_USB_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_HEARING_AID"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_ECHO_CANCELLER"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_DEFAULT"/>
+            <xs:enumeration value="AUDIO_DEVICE_OUT_STUB"/>
+
+            <!-- Due to the xml format, IN types can not be a separated from OUT types -->
+            <xs:enumeration value="AUDIO_DEVICE_IN_COMMUNICATION"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_AMBIENT"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_BUILTIN_MIC"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_WIRED_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_AUX_DIGITAL"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_HDMI"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_VOICE_CALL"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_TELEPHONY_RX"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_BACK_MIC"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_REMOTE_SUBMIX"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_USB_ACCESSORY"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_USB_DEVICE"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_FM_TUNER"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_TV_TUNER"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_LINE"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_SPDIF"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_BLUETOOTH_A2DP"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_LOOPBACK"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_IP"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_BUS"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_PROXY"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_USB_HEADSET"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_BLUETOOTH_BLE"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_DEFAULT"/>
+            <xs:enumeration value="AUDIO_DEVICE_IN_STUB"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <!-- Enum values of audio_format_t in audio.h
+         TODO: generate from hidl to avoid manual sync. -->
+    <xs:simpleType name="audioFormat">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_FORMAT_PCM_16_BIT" />
+            <xs:enumeration value="AUDIO_FORMAT_PCM_8_BIT"/>
+            <xs:enumeration value="AUDIO_FORMAT_PCM_32_BIT"/>
+            <xs:enumeration value="AUDIO_FORMAT_PCM_8_24_BIT"/>
+            <xs:enumeration value="AUDIO_FORMAT_PCM_FLOAT"/>
+            <xs:enumeration value="AUDIO_FORMAT_PCM_24_BIT_PACKED"/>
+            <xs:enumeration value="AUDIO_FORMAT_MP3"/>
+            <xs:enumeration value="AUDIO_FORMAT_AMR_NB"/>
+            <xs:enumeration value="AUDIO_FORMAT_AMR_WB"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_MAIN"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_LC"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_SSR"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_LTP"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_HE_V1"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_SCALABLE"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ERLC"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_LD"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_HE_V2"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ELD"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_MAIN"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_LC"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_SSR"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_LTP"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_HE_V1"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_SCALABLE"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_ERLC"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_LD"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_HE_V2"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_ELD"/>
+            <xs:enumeration value="AUDIO_FORMAT_VORBIS"/>
+            <xs:enumeration value="AUDIO_FORMAT_HE_AAC_V1"/>
+            <xs:enumeration value="AUDIO_FORMAT_HE_AAC_V2"/>
+            <xs:enumeration value="AUDIO_FORMAT_OPUS"/>
+            <xs:enumeration value="AUDIO_FORMAT_AC3"/>
+            <xs:enumeration value="AUDIO_FORMAT_E_AC3"/>
+            <xs:enumeration value="AUDIO_FORMAT_DTS"/>
+            <xs:enumeration value="AUDIO_FORMAT_DTS_HD"/>
+            <xs:enumeration value="AUDIO_FORMAT_IEC61937"/>
+            <xs:enumeration value="AUDIO_FORMAT_DOLBY_TRUEHD"/>
+            <xs:enumeration value="AUDIO_FORMAT_EVRC"/>
+            <xs:enumeration value="AUDIO_FORMAT_EVRCB"/>
+            <xs:enumeration value="AUDIO_FORMAT_EVRCWB"/>
+            <xs:enumeration value="AUDIO_FORMAT_EVRCNW"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADIF"/>
+            <xs:enumeration value="AUDIO_FORMAT_WMA"/>
+            <xs:enumeration value="AUDIO_FORMAT_WMA_PRO"/>
+            <xs:enumeration value="AUDIO_FORMAT_AMR_WB_PLUS"/>
+            <xs:enumeration value="AUDIO_FORMAT_MP2"/>
+            <xs:enumeration value="AUDIO_FORMAT_QCELP"/>
+            <xs:enumeration value="AUDIO_FORMAT_DSD"/>
+            <xs:enumeration value="AUDIO_FORMAT_FLAC"/>
+            <xs:enumeration value="AUDIO_FORMAT_ALAC"/>
+            <xs:enumeration value="AUDIO_FORMAT_APE"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS"/>
+            <xs:enumeration value="AUDIO_FORMAT_SBC"/>
+            <xs:enumeration value="AUDIO_FORMAT_APTX"/>
+            <xs:enumeration value="AUDIO_FORMAT_APTX_HD"/>
+            <xs:enumeration value="AUDIO_FORMAT_AC4"/>
+            <xs:enumeration value="AUDIO_FORMAT_LDAC"/>
+            <xs:enumeration value="AUDIO_FORMAT_E_AC3_JOC"/>
+            <xs:enumeration value="AUDIO_FORMAT_MAT_1_0"/>
+            <xs:enumeration value="AUDIO_FORMAT_MAT_2_0"/>
+            <xs:enumeration value="AUDIO_FORMAT_MAT_2_1"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_XHE"/>
+            <xs:enumeration value="AUDIO_FORMAT_AAC_ADTS_XHE"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <!-- Enum values of audio::common::4_0::AudioUsage
+         TODO: generate from HIDL to avoid manual sync. -->
+    <xs:simpleType name="audioUsage">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_USAGE_UNKNOWN" />
+            <xs:enumeration value="AUDIO_USAGE_MEDIA" />
+            <xs:enumeration value="AUDIO_USAGE_VOICE_COMMUNICATION" />
+            <xs:enumeration value="AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING" />
+            <xs:enumeration value="AUDIO_USAGE_ALARM" />
+            <xs:enumeration value="AUDIO_USAGE_NOTIFICATION" />
+            <xs:enumeration value="AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE" />
+            <xs:enumeration value="AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY" />
+            <xs:enumeration value="AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE" />
+            <xs:enumeration value="AUDIO_USAGE_ASSISTANCE_SONIFICATION" />
+            <xs:enumeration value="AUDIO_USAGE_GAME" />
+            <xs:enumeration value="AUDIO_USAGE_VIRTUAL_SOURCE" />
+            <xs:enumeration value="AUDIO_USAGE_ASSISTANT" />
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:simpleType name="audioUsageList">
+        <xs:list itemType="audioUsage"/>
+    </xs:simpleType>
+    <!-- TODO: Change to a space separated list to xsd enforce correctness. -->
+    <xs:simpleType name="samplingRates">
+        <xs:restriction base="xs:string">
+            <xs:pattern value="[0-9]+(,[0-9]+)*"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <!-- TODO: Change to a space separated list to xsd enforce correctness. -->
+    <xs:simpleType name="channelMask">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Comma (",") separated list of channel flags
+                from audio_channel_mask_t.
+            </xs:documentation>
+        </xs:annotation>
+        <xs:restriction base="xs:string">
+            <xs:pattern value="[_A-Z][_A-Z0-9]*(,[_A-Z][_A-Z0-9]*)*"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:complexType name="profile">
+        <xs:attribute name="name" type="xs:token" use="optional"/>
+        <xs:attribute name="format" type="audioFormat" use="optional"/>
+        <xs:attribute name="samplingRates" type="samplingRates" use="optional"/>
+        <xs:attribute name="channelMasks" type="channelMask" use="optional"/>
+    </xs:complexType>
+    <xs:simpleType name="gainMode">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_GAIN_MODE_JOINT"/>
+            <xs:enumeration value="AUDIO_GAIN_MODE_CHANNELS"/>
+            <xs:enumeration value="AUDIO_GAIN_MODE_RAMP"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:complexType name="gains">
+        <xs:sequence>
+            <xs:element name="gain" minOccurs="0" maxOccurs="unbounded">
+                <xs:complexType>
+                    <xs:attribute name="name" type="xs:token" use="required"/>
+                    <xs:attribute name="mode" type="gainMode" use="required"/>
+                    <xs:attribute name="channel_mask" type="channelMask" use="optional"/>
+                    <xs:attribute name="minValueMB" type="xs:int" use="optional"/>
+                    <xs:attribute name="maxValueMB" type="xs:int" use="optional"/>
+                    <xs:attribute name="defaultValueMB" type="xs:int" use="optional"/>
+                    <xs:attribute name="stepValueMB" type="xs:int" use="optional"/>
+                    <xs:attribute name="minRampMs" type="xs:int" use="optional"/>
+                    <xs:attribute name="maxRampMs" type="xs:int" use="optional"/>
+                </xs:complexType>
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+    <xs:complexType name="devicePorts">
+        <xs:sequence>
+            <xs:element name="devicePort" minOccurs="0" maxOccurs="unbounded">
+                <xs:complexType>
+                    <xs:sequence>
+                        <xs:element name="profile" type="profile" minOccurs="0" maxOccurs="unbounded"/>
+                        <xs:element name="gains" type="gains" minOccurs="0"/>
+                    </xs:sequence>
+                    <xs:attribute name="tagName" type="xs:token" use="required"/>
+                    <xs:attribute name="type" type="audioDevice" use="required"/>
+                    <xs:attribute name="role" type="role" use="required"/>
+                    <xs:attribute name="address" type="xs:string" use="optional" default=""/>
+                    <!-- Note that XSD 1.0 can not check that a type only has one default. -->
+                    <xs:attribute name="default" type="xs:boolean" use="optional">
+                        <xs:annotation>
+                            <xs:documentation xml:lang="en">
+                                The default device will be used if multiple have the same type
+                                and no explicit route request exists for a specific device of
+                                that type.
+                            </xs:documentation>
+                        </xs:annotation>
+                    </xs:attribute>
+                </xs:complexType>
+                <xs:unique name="devicePortProfileUniqueness">
+                    <xs:selector xpath="profile"/>
+                    <xs:field xpath="format"/>
+                    <xs:field xpath="samplingRate"/>
+                    <xs:field xpath="channelMasks"/>
+                </xs:unique>
+                <xs:unique name="devicePortGainUniqueness">
+                    <xs:selector xpath="gains/gain"/>
+                    <xs:field xpath="@name"/>
+                </xs:unique>
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+    <xs:simpleType name="mixType">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="mix"/>
+            <xs:enumeration value="mux"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:complexType name="routes">
+        <xs:sequence>
+            <xs:element name="route" minOccurs="0" maxOccurs="unbounded">
+                <xs:annotation>
+                    <xs:documentation xml:lang="en">
+                        List all available sources for a given sink.
+                    </xs:documentation>
+                </xs:annotation>
+                <xs:complexType>
+                    <xs:attribute name="type" type="mixType" use="required"/>
+                    <xs:attribute name="sink" type="xs:string" use="required"/>
+                    <xs:attribute name="sources" type="xs:string" use="required"/>
+                </xs:complexType>
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+    <xs:complexType name="volumes">
+        <xs:sequence>
+            <xs:element name="volume" type="volume" minOccurs="0" maxOccurs="unbounded"/>
+            <xs:element name="reference" type="reference" minOccurs="0" maxOccurs="unbounded">
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+    <!-- TODO: Always require a ref for better xsd validations.
+               Currently a volume could have no points nor ref
+               as it can not be forbidden by xsd 1.0.-->
+    <xs:simpleType name="volumePoint">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Comma separated pair of number.
+                The fist one is the framework level (between 0 and 100).
+                The second one is the volume to send to the HAL.
+                The framework will interpolate volumes not specified.
+                Their MUST be at least 2 points specified.
+            </xs:documentation>
+        </xs:annotation>
+        <xs:restriction base="xs:string">
+            <xs:pattern value="([0-9]{1,2}|100),-?[0-9]+"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <!-- Enum values of audio_stream_type_t in audio-base.h
+         TODO: generate from hidl to avoid manual sync. -->
+    <xs:simpleType name="stream">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_STREAM_VOICE_CALL"/>
+            <xs:enumeration value="AUDIO_STREAM_SYSTEM"/>
+            <xs:enumeration value="AUDIO_STREAM_RING"/>
+            <xs:enumeration value="AUDIO_STREAM_MUSIC"/>
+            <xs:enumeration value="AUDIO_STREAM_ALARM"/>
+            <xs:enumeration value="AUDIO_STREAM_NOTIFICATION"/>
+            <xs:enumeration value="AUDIO_STREAM_BLUETOOTH_SCO"/>
+            <xs:enumeration value="AUDIO_STREAM_ENFORCED_AUDIBLE"/>
+            <xs:enumeration value="AUDIO_STREAM_DTMF"/>
+            <xs:enumeration value="AUDIO_STREAM_TTS"/>
+            <xs:enumeration value="AUDIO_STREAM_ACCESSIBILITY"/>
+            <xs:enumeration value="AUDIO_STREAM_REROUTING"/>
+            <xs:enumeration value="AUDIO_STREAM_PATCH"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <!-- Enum values of device_category from Volume.h.
+         TODO: generate from hidl to avoid manual sync. -->
+    <xs:simpleType name="deviceCategory">
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="DEVICE_CATEGORY_HEADSET"/>
+            <xs:enumeration value="DEVICE_CATEGORY_SPEAKER"/>
+            <xs:enumeration value="DEVICE_CATEGORY_EARPIECE"/>
+            <xs:enumeration value="DEVICE_CATEGORY_EXT_MEDIA"/>
+            <xs:enumeration value="DEVICE_CATEGORY_HEARING_AID"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:complexType name="volume">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Volume section defines a volume curve for a given use case and device category.
+                It contains a list of points of this curve expressing the attenuation in Millibels
+                for a given volume index from 0 to 100.
+                <volume stream="AUDIO_STREAM_MUSIC" deviceCategory="DEVICE_CATEGORY_SPEAKER">
+                    <point>0,-9600</point>
+                    <point>100,0</point>
+                </volume>
+
+                It may also reference a reference/@name to avoid duplicating curves.
+                <volume stream="AUDIO_STREAM_MUSIC" deviceCategory="DEVICE_CATEGORY_SPEAKER"
+                        ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+                <reference name="DEFAULT_MEDIA_VOLUME_CURVE">
+                    <point>0,-9600</point>
+                    <point>100,0</point>
+                </reference>
+            </xs:documentation>
+        </xs:annotation>
+        <xs:sequence>
+            <xs:element name="point" type="volumePoint" minOccurs="0" maxOccurs="unbounded"/>
+        </xs:sequence>
+        <xs:attribute name="stream" type="stream"/>
+        <xs:attribute name="deviceCategory" type="deviceCategory"/>
+        <xs:attribute name="ref" type="xs:token" use="optional"/>
+    </xs:complexType>
+    <xs:complexType name="reference">
+        <xs:sequence>
+            <xs:element name="point" type="volumePoint" minOccurs="2" maxOccurs="unbounded"/>
+        </xs:sequence>
+        <xs:attribute name="name" type="xs:token" use="required"/>
+    </xs:complexType>
+    <xs:complexType name="surroundSound">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Surround Sound section provides configuration related to handling of
+                multi-channel formats.
+            </xs:documentation>
+        </xs:annotation>
+        <xs:sequence>
+            <xs:element name="formats" type="surroundFormats"/>
+        </xs:sequence>
+    </xs:complexType>
+    <xs:simpleType name="surroundFormatsList">
+        <xs:list itemType="audioFormat" />
+    </xs:simpleType>
+    <xs:complexType name="surroundFormats">
+        <xs:sequence>
+            <xs:element name="format" minOccurs="0" maxOccurs="unbounded">
+                <xs:complexType>
+                    <xs:attribute name="name" type="audioFormat" use="required"/>
+                    <xs:attribute name="subformats" type="surroundFormatsList" />
+                </xs:complexType>
+            </xs:element>
+        </xs:sequence>
+    </xs:complexType>
+</xs:schema>
diff --git a/audio/5.0/types.hal b/audio/5.0/types.hal
new file mode 100644
index 0000000..f58bfb0
--- /dev/null
+++ b/audio/5.0/types.hal
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+import android.hardware.audio.common@5.0;
+
+enum Result : int32_t {
+    OK,
+    NOT_INITIALIZED,
+    INVALID_ARGUMENTS,
+    INVALID_STATE,
+    /**
+     * Methods marked as "Optional method" must return this result value
+     * if the operation is not supported by HAL.
+     */
+    NOT_SUPPORTED
+};
+
+@export(name="audio_drain_type_t", value_prefix="AUDIO_DRAIN_")
+enum AudioDrain : int32_t {
+    /** drain() returns when all data has been played. */
+    ALL,
+    /**
+     * drain() returns a short time before all data from the current track has
+     * been played to give time for gapless track switch.
+     */
+    EARLY_NOTIFY
+};
+
+/**
+ * A substitute for POSIX timespec.
+ */
+struct TimeSpec {
+    uint64_t tvSec;   // seconds
+    uint64_t tvNSec;  // nanoseconds
+};
+
+/**
+ * IEEE 802 MAC address.
+ */
+typedef uint8_t[6] MacAddress;
+
+struct ParameterValue {
+    string key;
+    string value;
+};
+
+/**
+ * Specifies a device 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;
+    string busAddress;      // used for BUS
+    string rSubmixAddress;  // used for REMOTE_SUBMIX
+};
+
+enum MmapBufferFlag : uint32_t {
+    NONE    = 0x0,
+    /**
+     * If the buffer can be securely shared to untrusted applications
+     * through the AAudio exclusive mode.
+     * Only set this flag if applications are restricted from accessing the
+     * memory surrounding the audio data buffer by a kernel mechanism.
+     * See Linux kernel's dma_buf.
+     */
+    APPLICATION_SHAREABLE    = 0x1,
+};
+
+/**
+ * Mmap buffer descriptor returned by IStream.createMmapBuffer().
+ * Used by streams opened in mmap mode.
+ */
+struct MmapBufferInfo {
+    /** Mmap memory buffer */
+    memory  sharedMemory;
+    /** Total buffer size in frames */
+    uint32_t bufferSizeFrames;
+    /** Transfer size granularity in frames */
+    uint32_t burstSizeFrames;
+    /** Attributes describing the buffer. */
+    bitfield<MmapBufferFlag> flags;
+};
+
+/**
+ * Mmap buffer read/write position returned by IStream.getMmapPosition().
+ * Used by streams opened in mmap mode.
+ */
+struct MmapPosition {
+    int64_t  timeNanoseconds; // time stamp in ns, CLOCK_MONOTONIC
+    int32_t  positionFrames;  // increasing 32 bit frame count reset when IStream.stop() is called
+};
+
+/**
+ * The message queue flags used to synchronize reads and writes from
+ * message queues used by StreamIn and StreamOut.
+ */
+enum MessageQueueFlagBits : uint32_t {
+    NOT_EMPTY = 1 << 0,
+    NOT_FULL = 1 << 1
+};
+
+/** 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;
+};
+
+/** Metadatas of the source of a StreamIn. */
+struct SinkMetadata {
+    vec<RecordTrackMetadata> tracks;
+};
+
+/*
+ * Microphone information
+ *
+ */
+
+/**
+ * A 3D point used to represent position or orientation of a microphone.
+ *
+ * Position: Coordinates of the microphone's capsule, in meters, from the
+ * bottom-left-back corner of the bounding box of android device in natural
+ * orientation (PORTRAIT for phones, LANDSCAPE for tablets, tvs, etc).
+ * The orientation musth match the reported by the api Display.getRotation().
+ *
+ * Orientation: Normalized vector to signal the main orientation of the
+ * microphone's capsule. Magnitude = sqrt(x^2 + y^2 + z^2) = 1
+ */
+struct AudioMicrophoneCoordinate {
+    float x;
+    float y;
+    float z;
+};
+
+/**
+ * Enum to identify the type of channel mapping for active microphones.
+ * Used channels further identify if the microphone has any significative
+ * process (e.g. High Pass Filtering, dynamic compression)
+ * Simple processing as constant gain adjustment must be DIRECT.
+ */
+enum AudioMicrophoneChannelMapping : uint32_t {
+    UNUSED      = 0, /* Channel not used */
+    DIRECT      = 1, /* Channel used and signal not processed */
+    PROCESSED   = 2, /* Channel used and signal has some process */
+};
+
+/**
+ * Enum to identify locations of microphones in regards to the body of the
+ * android device.
+ */
+enum AudioMicrophoneLocation : uint32_t {
+    UNKNOWN             = 0,
+    MAINBODY            = 1,
+    MAINBODY_MOVABLE    = 2,
+    PERIPHERAL          = 3,
+};
+
+/**
+ * Identifier to help group related microphones together
+ * e.g. microphone arrays should belong to the same group
+ */
+typedef int32_t AudioMicrophoneGroup;
+
+/**
+ * Enum with standard polar patterns of microphones
+ */
+enum AudioMicrophoneDirectionality : uint32_t {
+    UNKNOWN         = 0,
+    OMNI            = 1,
+    BI_DIRECTIONAL  = 2,
+    CARDIOID        = 3,
+    HYPER_CARDIOID  = 4,
+    SUPER_CARDIOID  = 5,
+};
+
+/**
+ * A (frequency, level) pair. Used to represent frequency response.
+ */
+struct AudioFrequencyResponsePoint {
+    /** In Hz */
+    float frequency;
+    /** In dB */
+    float level;
+};
+
+/**
+ * Structure used by the HAL to describe microphone's characteristics
+ * Used by StreamIn and Device
+ */
+struct MicrophoneInfo {
+    /** Unique alphanumeric id for microphone. Guaranteed to be the same
+     * even after rebooting.
+     */
+    string                                  deviceId;
+    /**
+     * Device specific information
+     */
+    DeviceAddress                           deviceAddress;
+    /** Each element of the vector must describe the channel with the same
+     *  index.
+     */
+    vec<AudioMicrophoneChannelMapping>      channelMapping;
+    /** Location of the microphone in regard to the body of the device */
+    AudioMicrophoneLocation                 location;
+    /** Identifier to help group related microphones together
+     *  e.g. microphone arrays should belong to the same group
+     */
+    AudioMicrophoneGroup                    group;
+    /** Index of this microphone within the group.
+     *  (group, index) must be unique within the same device.
+     */
+    uint32_t                                indexInTheGroup;
+    /** Level in dBFS produced by a 1000 Hz tone at 94 dB SPL */
+    float                                   sensitivity;
+    /** Level in dB of the max SPL supported at 1000 Hz */
+    float                                   maxSpl;
+    /** Level in dB of the min SPL supported at 1000 Hz */
+    float                                   minSpl;
+    /** Standard polar pattern of the microphone */
+    AudioMicrophoneDirectionality           directionality;
+    /** Vector with ordered frequency responses (from low to high frequencies)
+     *  with the frequency response of the microphone.
+     *  Levels are in dB, relative to level at 1000 Hz
+     */
+    vec<AudioFrequencyResponsePoint>        frequencyResponse;
+    /** Position of the microphone's capsule in meters, from the
+     *  bottom-left-back corner of the bounding box of device.
+     */
+    AudioMicrophoneCoordinate               position;
+    /** Normalized point to signal the main orientation of the microphone's
+     *  capsule. sqrt(x^2 + y^2 + z^2) = 1
+     */
+    AudioMicrophoneCoordinate               orientation;
+};
diff --git a/audio/common/5.0/Android.bp b/audio/common/5.0/Android.bp
new file mode 100644
index 0000000..663f847
--- /dev/null
+++ b/audio/common/5.0/Android.bp
@@ -0,0 +1,48 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+    name: "android.hardware.audio.common@5.0",
+    root: "android.hardware",
+    vndk: {
+        enabled: true,
+    },
+    srcs: [
+        "types.hal",
+    ],
+    types: [
+        "AudioChannelMask",
+        "AudioConfig",
+        "AudioContentType",
+        "AudioDevice",
+        "AudioFormat",
+        "AudioGain",
+        "AudioGainConfig",
+        "AudioGainMode",
+        "AudioHandleConsts",
+        "AudioInputFlag",
+        "AudioMixLatencyClass",
+        "AudioMode",
+        "AudioOffloadInfo",
+        "AudioOutputFlag",
+        "AudioPort",
+        "AudioPortConfig",
+        "AudioPortConfigDeviceExt",
+        "AudioPortConfigMask",
+        "AudioPortConfigSessionExt",
+        "AudioPortDeviceExt",
+        "AudioPortMixExt",
+        "AudioPortRole",
+        "AudioPortSessionExt",
+        "AudioPortType",
+        "AudioSessionConsts",
+        "AudioSource",
+        "AudioStreamType",
+        "AudioUsage",
+        "FixedChannelCount",
+        "ThreadInfo",
+        "Uuid",
+    ],
+    gen_java: false,
+    gen_java_constants: true,
+}
+
diff --git a/audio/common/5.0/types.hal b/audio/common/5.0/types.hal
new file mode 100644
index 0000000..eb09b1f
--- /dev/null
+++ b/audio/common/5.0/types.hal
@@ -0,0 +1,903 @@
+/*
+ * Copyright (C) 2018 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@5.0;
+
+/*
+ *
+ *  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,
+
+    FM_TUNER            = 1998,
+};
+
+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,
+    /** 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),
+};
+
+/**
+ * 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.  See AUDIO_INTERLEAVE_* for the platform convention of order.
+ *
+ * AudioChannelMask is an opaque type and its internal layout should not be
+ * assumed as it may change in the future.  Instead, always use functions
+ * to examine it.
+ *
+ * These are the current representations:
+ *
+ *   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,
+
+    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_CENTER | OUT_FRONT_RIGHT |
+                         OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+    OUT_3POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_CENTER | OUT_FRONT_RIGHT |
+                        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),
+    // 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)
+};
+
+/**
+ * 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_DEFAULT               = BIT_IN | BIT_DEFAULT,
+
+    // Note that the 2.0 IN_ALL* have been moved to helper functions
+};
+
+/**
+ * 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;
+};
+
+
+/*
+ *
+ *  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/effect/5.0/Android.bp b/audio/effect/5.0/Android.bp
new file mode 100644
index 0000000..d60d94a
--- /dev/null
+++ b/audio/effect/5.0/Android.bp
@@ -0,0 +1,47 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+    name: "android.hardware.audio.effect@5.0",
+    root: "android.hardware",
+    vndk: {
+        enabled: true,
+    },
+    srcs: [
+        "types.hal",
+        "IAcousticEchoCancelerEffect.hal",
+        "IAutomaticGainControlEffect.hal",
+        "IBassBoostEffect.hal",
+        "IDownmixEffect.hal",
+        "IEffect.hal",
+        "IEffectBufferProviderCallback.hal",
+        "IEffectsFactory.hal",
+        "IEnvironmentalReverbEffect.hal",
+        "IEqualizerEffect.hal",
+        "ILoudnessEnhancerEffect.hal",
+        "INoiseSuppressionEffect.hal",
+        "IPresetReverbEffect.hal",
+        "IVirtualizerEffect.hal",
+        "IVisualizerEffect.hal",
+    ],
+    interfaces: [
+        "android.hardware.audio.common@5.0",
+        "android.hidl.base@1.0",
+    ],
+    types: [
+        "AudioBuffer",
+        "EffectAuxChannelsConfig",
+        "EffectBufferAccess",
+        "EffectBufferConfig",
+        "EffectConfig",
+        "EffectConfigParameters",
+        "EffectDescriptor",
+        "EffectFeature",
+        "EffectFlags",
+        "EffectOffloadParameter",
+        "MessageQueueFlagBits",
+        "Result",
+    ],
+    gen_java: false,
+    gen_java_constants: true,
+}
+
diff --git a/audio/effect/5.0/IAcousticEchoCancelerEffect.hal b/audio/effect/5.0/IAcousticEchoCancelerEffect.hal
new file mode 100644
index 0000000..61cdc95
--- /dev/null
+++ b/audio/effect/5.0/IAcousticEchoCancelerEffect.hal
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IAcousticEchoCancelerEffect extends IEffect {
+    /**
+     * Sets echo delay value in milliseconds.
+     */
+    setEchoDelay(uint32_t echoDelayMs) generates (Result retval);
+
+    /**
+     * Gets echo delay value in milliseconds.
+     */
+    getEchoDelay() generates (Result retval, uint32_t echoDelayMs);
+};
diff --git a/audio/effect/5.0/IAutomaticGainControlEffect.hal b/audio/effect/5.0/IAutomaticGainControlEffect.hal
new file mode 100644
index 0000000..ae354ec
--- /dev/null
+++ b/audio/effect/5.0/IAutomaticGainControlEffect.hal
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IAutomaticGainControlEffect extends IEffect {
+    /**
+     * Sets target level in millibels.
+     */
+    setTargetLevel(int16_t targetLevelMb) generates (Result retval);
+
+    /**
+     * Gets target level.
+     */
+    getTargetLevel() generates (Result retval, int16_t targetLevelMb);
+
+    /**
+     * Sets gain in the compression range in millibels.
+     */
+    setCompGain(int16_t compGainMb) generates (Result retval);
+
+    /**
+     * Gets gain in the compression range.
+     */
+    getCompGain() generates (Result retval, int16_t compGainMb);
+
+    /**
+     * Enables or disables limiter.
+     */
+    setLimiterEnabled(bool enabled) generates (Result retval);
+
+    /**
+     * Returns whether limiter is enabled.
+     */
+    isLimiterEnabled() generates (Result retval, bool enabled);
+
+    struct AllProperties {
+        int16_t targetLevelMb;
+        int16_t compGainMb;
+        bool limiterEnabled;
+    };
+
+    /**
+     * Sets all properties at once.
+     */
+    setAllProperties(AllProperties properties) generates (Result retval);
+
+    /**
+     * Gets all properties at once.
+     */
+    getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/5.0/IBassBoostEffect.hal b/audio/effect/5.0/IBassBoostEffect.hal
new file mode 100644
index 0000000..4b3b629
--- /dev/null
+++ b/audio/effect/5.0/IBassBoostEffect.hal
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IBassBoostEffect extends IEffect {
+    /**
+     * Returns whether setting bass boost strength is supported.
+     */
+    isStrengthSupported() generates (Result retval, bool strengthSupported);
+
+    enum StrengthRange : uint16_t {
+        MIN = 0,
+        MAX = 1000
+    };
+
+    /**
+     * Sets bass boost strength.
+     *
+     * @param strength strength of the effect. The valid range for strength
+     *                 strength is [0, 1000], where 0 per mille designates the
+     *                 mildest effect and 1000 per mille designates the
+     *                 strongest.
+     * @return retval operation completion status.
+     */
+    setStrength(uint16_t strength) generates (Result retval);
+
+    /**
+     * Gets virtualization strength.
+     */
+    getStrength() generates (Result retval, uint16_t strength);
+};
diff --git a/audio/effect/5.0/IDownmixEffect.hal b/audio/effect/5.0/IDownmixEffect.hal
new file mode 100644
index 0000000..43d4b23
--- /dev/null
+++ b/audio/effect/5.0/IDownmixEffect.hal
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IDownmixEffect extends IEffect {
+    enum Type : int32_t {
+        STRIP, // throw away the extra channels
+        FOLD   // mix the extra channels with FL/FR
+    };
+
+    /**
+     * Sets the current downmix preset.
+     */
+    setType(Type preset) generates (Result retval);
+
+    /**
+     * Gets the current downmix preset.
+     */
+    getType() generates (Result retval, Type preset);
+};
diff --git a/audio/effect/5.0/IEffect.hal b/audio/effect/5.0/IEffect.hal
new file mode 100644
index 0000000..9c50963
--- /dev/null
+++ b/audio/effect/5.0/IEffect.hal
@@ -0,0 +1,418 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffectBufferProviderCallback;
+
+interface IEffect {
+    /**
+     * Initialize effect engine--all configurations return to default.
+     *
+     * @return retval operation completion status.
+     */
+    @entry
+    init() generates (Result retval);
+
+    /**
+     * Apply new audio parameters configurations for input and output buffers.
+     * The provider callbacks may be empty, but in this case the buffer
+     * must be provided in the EffectConfig structure.
+     *
+     * @param config configuration descriptor.
+     * @param inputBufferProvider optional buffer provider reference.
+     * @param outputBufferProvider optional buffer provider reference.
+     * @return retval operation completion status.
+     */
+    setConfig(EffectConfig config,
+            IEffectBufferProviderCallback inputBufferProvider,
+            IEffectBufferProviderCallback outputBufferProvider)
+            generates (Result retval);
+
+    /**
+     * Reset the effect engine. Keep configuration but resets state and buffer
+     * content.
+     *
+     * @return retval operation completion status.
+     */
+    reset() generates (Result retval);
+
+    /**
+     * Enable processing.
+     *
+     * @return retval operation completion status.
+     */
+    @callflow(next={"prepareForProcessing"})
+    enable() generates (Result retval);
+
+    /**
+     * Disable processing.
+     *
+     * @return retval operation completion status.
+     */
+    @callflow(next={"close"})
+    disable() generates (Result retval);
+
+    /**
+     * Set the rendering device the audio output path is connected to.  The
+     * effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its
+     * descriptor to receive this command when the device changes.
+     *
+     * Note: this method is only supported for effects inserted into
+     *       the output chain.
+     *
+     * @param device output device specification.
+     * @return retval operation completion status.
+     */
+    setDevice(bitfield<AudioDevice> device) generates (Result retval);
+
+    /**
+     * Set and get volume. Used by audio framework to delegate volume control to
+     * effect engine. The effect implementation must set EFFECT_FLAG_VOLUME_CTRL
+     * flag in its descriptor to receive this command. The effect engine must
+     * return the volume that should be applied before the effect is
+     * processed. The overall volume (the volume actually applied by the effect
+     * engine multiplied by the returned value) should match the value indicated
+     * in the command.
+     *
+     * @param volumes vector containing volume for each channel defined in
+     *                EffectConfig for output buffer expressed in 8.24 fixed
+     *                point format.
+     * @return result updated volume values.
+     * @return retval operation completion status.
+     */
+    setAndGetVolume(vec<uint32_t> volumes)
+            generates (Result retval, vec<uint32_t> result);
+
+    /**
+     * Notify the effect of the volume change. The effect implementation must
+     * set EFFECT_FLAG_VOLUME_IND flag in its descriptor to receive this
+     * command.
+     *
+     * @param volumes vector containing volume for each channel defined in
+     *                EffectConfig for output buffer expressed in 8.24 fixed
+     *                point format.
+     * @return retval operation completion status.
+     */
+    volumeChangeNotification(vec<uint32_t> volumes)
+            generates (Result retval);
+
+    /**
+     * Set the audio mode. The effect implementation must set
+     * EFFECT_FLAG_AUDIO_MODE_IND flag in its descriptor to receive this command
+     * when the audio mode changes.
+     *
+     * @param mode desired audio mode.
+     * @return retval operation completion status.
+     */
+    setAudioMode(AudioMode mode) generates (Result retval);
+
+    /**
+     * Apply new audio parameters configurations for input and output buffers of
+     * reverse stream.  An example of reverse stream is the echo reference
+     * supplied to an Acoustic Echo Canceler.
+     *
+     * @param config configuration descriptor.
+     * @param inputBufferProvider optional buffer provider reference.
+     * @param outputBufferProvider optional buffer provider reference.
+     * @return retval operation completion status.
+     */
+    setConfigReverse(EffectConfig config,
+            IEffectBufferProviderCallback inputBufferProvider,
+            IEffectBufferProviderCallback outputBufferProvider)
+            generates (Result retval);
+
+    /**
+     * Set the capture device the audio input path is connected to. The effect
+     * implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to
+     * receive this command when the device changes.
+     *
+     * Note: this method is only supported for effects inserted into
+     *       the input chain.
+     *
+     * @param device input device specification.
+     * @return retval operation completion status.
+     */
+    setInputDevice(bitfield<AudioDevice> device) generates (Result retval);
+
+    /**
+     * Read audio parameters configurations for input and output buffers.
+     *
+     * @return retval operation completion status.
+     * @return config configuration descriptor.
+     */
+    getConfig() generates (Result retval, EffectConfig config);
+
+    /**
+     * Read audio parameters configurations for input and output buffers of
+     * reverse stream.
+     *
+     * @return retval operation completion status.
+     * @return config configuration descriptor.
+     */
+    getConfigReverse() generates (Result retval, EffectConfig config);
+
+    /**
+     * Queries for supported combinations of main and auxiliary channels
+     * (e.g. for a multi-microphone noise suppressor).
+     *
+     * @param maxConfigs maximum number of the combinations to return.
+     * @return retval absence of the feature support is indicated using
+     *                NOT_SUPPORTED code. RESULT_TOO_BIG is returned if
+     *                the number of supported combinations exceeds 'maxConfigs'.
+     * @return result list of configuration descriptors.
+     */
+    getSupportedAuxChannelsConfigs(uint32_t maxConfigs)
+            generates (Result retval, vec<EffectAuxChannelsConfig> result);
+
+    /**
+     * Retrieves the current configuration of main and auxiliary channels.
+     *
+     * @return retval absence of the feature support is indicated using
+     *                NOT_SUPPORTED code.
+     * @return result configuration descriptor.
+     */
+    getAuxChannelsConfig()
+            generates (Result retval, EffectAuxChannelsConfig result);
+
+    /**
+     * Sets the current configuration of main and auxiliary channels.
+     *
+     * @return retval operation completion status; absence of the feature
+     *                support is indicated using NOT_SUPPORTED code.
+     */
+    setAuxChannelsConfig(EffectAuxChannelsConfig config)
+            generates (Result retval);
+
+    /**
+     * Set the audio source the capture path is configured for (Camcorder, voice
+     * recognition...).
+     *
+     * Note: this method is only supported for effects inserted into
+     *       the input chain.
+     *
+     * @param source source descriptor.
+     * @return retval operation completion status.
+     */
+    setAudioSource(AudioSource source) generates (Result retval);
+
+    /**
+     * This command indicates if the playback thread the effect is attached to
+     * is offloaded or not, and updates the I/O handle of the playback thread
+     * the effect is attached to.
+     *
+     * @param param effect offload descriptor.
+     * @return retval operation completion status.
+     */
+    offload(EffectOffloadParameter param) generates (Result retval);
+
+    /**
+     * Returns the effect descriptor.
+     *
+     * @return retval operation completion status.
+     * @return descriptor effect descriptor.
+     */
+    getDescriptor() generates (Result retval, EffectDescriptor descriptor);
+
+    /**
+     * Set up required transports for passing audio buffers to the effect.
+     *
+     * The transport consists of shared memory and a message queue for reporting
+     * effect processing operation status. The shared memory is set up
+     * separately using 'setProcessBuffers' method.
+     *
+     * Processing is requested by setting 'REQUEST_PROCESS' or
+     * 'REQUEST_PROCESS_REVERSE' EventFlags associated with the status message
+     * queue. The result of processing may be one of the following:
+     *   OK if there were no errors during processing;
+     *   INVALID_ARGUMENTS if audio buffers are invalid;
+     *   INVALID_STATE if the engine has finished the disable phase;
+     *   NOT_INITIALIZED if the audio buffers were not set;
+     *   NOT_SUPPORTED if the requested processing type is not supported by
+     *                 the effect.
+     *
+     * @return retval OK if both message queues were created successfully.
+     *                INVALID_STATE if the method was already called.
+     *                INVALID_ARGUMENTS if there was a problem setting up
+     *                                  the queue.
+     * @return statusMQ a message queue used for passing status from the effect.
+     */
+    @callflow(next={"setProcessBuffers"})
+    prepareForProcessing() generates (Result retval, fmq_sync<Result> statusMQ);
+
+    /**
+     * Set up input and output buffers for processing audio data. The effect
+     * may modify both the input and the output buffer during the operation.
+     * Buffers may be set multiple times during effect lifetime.
+     *
+     * The input and the output buffer may be reused between different effects,
+     * and the input buffer may be used as an output buffer. Buffers are
+     * distinguished using 'AudioBuffer.id' field.
+     *
+     * @param inBuffer input audio buffer.
+     * @param outBuffer output audio buffer.
+     * @return retval OK if both buffers were mapped successfully.
+     *                INVALID_ARGUMENTS if there was a problem with mapping
+     *                                  any of the buffers.
+     */
+    setProcessBuffers(AudioBuffer inBuffer, AudioBuffer outBuffer)
+            generates (Result retval);
+
+    /**
+     * Execute a vendor specific command on the effect. The command code
+     * and data, as well as result data are not interpreted by Android
+     * Framework and are passed as-is between the application and the effect.
+     *
+     * The effect must use standard POSIX.1-2001 error codes for the operation
+     * completion status.
+     *
+     * Use this method only if the effect is provided by a third party, and
+     * there is no interface defined for it. This method only works for effects
+     * implemented in software.
+     *
+     * @param commandId the ID of the command.
+     * @param data command data.
+     * @param resultMaxSize maximum size in bytes of the result; can be 0.
+     * @return status command completion status.
+     * @return result result data.
+     */
+    command(uint32_t commandId, vec<uint8_t> data, uint32_t resultMaxSize)
+            generates (int32_t status, vec<uint8_t> result);
+
+    /**
+     * Set a vendor-specific parameter and apply it immediately. The parameter
+     * code and data are not interpreted by Android Framework and are passed
+     * as-is between the application and the effect.
+     *
+     * The effect must use INVALID_ARGUMENTS return code if the parameter ID is
+     * unknown or if provided parameter data is invalid. If the effect does not
+     * support setting vendor-specific parameters, it must return NOT_SUPPORTED.
+     *
+     * Use this method only if the effect is provided by a third party, and
+     * there is no interface defined for it. This method only works for effects
+     * implemented in software.
+     *
+     * @param parameter identifying data of the parameter.
+     * @param value the value of the parameter.
+     * @return retval operation completion status.
+     */
+    setParameter(vec<uint8_t> parameter, vec<uint8_t> value)
+            generates (Result retval);
+
+    /**
+     * Get a vendor-specific parameter value. The parameter code and returned
+     * data are not interpreted by Android Framework and are passed as-is
+     * between the application and the effect.
+     *
+     * The effect must use INVALID_ARGUMENTS return code if the parameter ID is
+     * unknown. If the effect does not support setting vendor-specific
+     * parameters, it must return NOT_SUPPORTED.
+     *
+     * Use this method only if the effect is provided by a third party, and
+     * there is no interface defined for it.  This method only works for effects
+     * implemented in software.
+     *
+     * @param parameter identifying data of the parameter.
+     * @param valueMaxSize maximum size in bytes of the value.
+     * @return retval operation completion status.
+     * @return result the value of the parameter.
+     */
+    getParameter(vec<uint8_t> parameter, uint32_t valueMaxSize)
+            generates (Result retval, vec<uint8_t> value);
+
+    /**
+     * Get supported configs for a vendor-specific feature. The configs returned
+     * are not interpreted by Android Framework and are passed as-is between the
+     * application and the effect.
+     *
+     * The effect must use INVALID_ARGUMENTS return code if the feature ID is
+     * unknown. If the effect does not support getting vendor-specific feature
+     * configs, it must return NOT_SUPPORTED. If the feature is supported but
+     * the total number of supported configurations exceeds the maximum number
+     * indicated by the caller, the method must return RESULT_TOO_BIG.
+     *
+     * Use this method only if the effect is provided by a third party, and
+     * there is no interface defined for it.  This method only works for effects
+     * implemented in software.
+     *
+     * @param featureId feature identifier.
+     * @param maxConfigs maximum number of configs to return.
+     * @param configSize size of each config in bytes.
+     * @return retval operation completion status.
+     * @return configsCount number of configs returned.
+     * @return configsData data for all the configs returned.
+     */
+    getSupportedConfigsForFeature(
+            uint32_t featureId,
+            uint32_t maxConfigs,
+            uint32_t configSize) generates (
+                    Result retval,
+                    uint32_t configsCount,
+                    vec<uint8_t> configsData);
+
+    /**
+     * Get the current config for a vendor-specific feature. The config returned
+     * is not interpreted by Android Framework and is passed as-is between the
+     * application and the effect.
+     *
+     * The effect must use INVALID_ARGUMENTS return code if the feature ID is
+     * unknown. If the effect does not support getting vendor-specific
+     * feature configs, it must return NOT_SUPPORTED.
+     *
+     * Use this method only if the effect is provided by a third party, and
+     * there is no interface defined for it.  This method only works for effects
+     * implemented in software.
+     *
+     * @param featureId feature identifier.
+     * @param configSize size of the config in bytes.
+     * @return retval operation completion status.
+     * @return configData config data.
+     */
+    getCurrentConfigForFeature(uint32_t featureId, uint32_t configSize)
+            generates (Result retval, vec<uint8_t> configData);
+
+    /**
+     * Set the current config for a vendor-specific feature. The config data
+     * is not interpreted by Android Framework and is passed as-is between the
+     * application and the effect.
+     *
+     * The effect must use INVALID_ARGUMENTS return code if the feature ID is
+     * unknown. If the effect does not support getting vendor-specific
+     * feature configs, it must return NOT_SUPPORTED.
+     *
+     * Use this method only if the effect is provided by a third party, and
+     * there is no interface defined for it.  This method only works for effects
+     * implemented in software.
+     *
+     * @param featureId feature identifier.
+     * @param configData config data.
+     * @return retval operation completion status.
+     */
+    setCurrentConfigForFeature(uint32_t featureId, vec<uint8_t> configData)
+            generates (Result retval);
+
+    /**
+     * Called by the framework to deinitialize the effect and free up
+     * all the currently allocated resources. It is recommended to close
+     * the effect on the client side as soon as it is becomes unused.
+     *
+     * @return retval OK in case the success.
+     *                INVALID_STATE if the effect was already closed.
+     */
+    @exit
+    close() generates (Result retval);
+};
diff --git a/audio/effect/5.0/IEffectBufferProviderCallback.hal b/audio/effect/5.0/IEffectBufferProviderCallback.hal
new file mode 100644
index 0000000..aabd15e
--- /dev/null
+++ b/audio/effect/5.0/IEffectBufferProviderCallback.hal
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+/**
+ * This callback interface contains functions that can be used by the effect
+ * engine 'process' function to exchange input and output audio buffers.
+ */
+interface IEffectBufferProviderCallback {
+    /**
+     * Called to retrieve a buffer where data should read from by 'process'
+     * function.
+     *
+     * @return buffer audio buffer for processing
+     */
+    getBuffer() generates (AudioBuffer buffer);
+
+    /**
+     * Called to provide a buffer with the data written by 'process' function.
+     *
+     * @param buffer audio buffer for processing
+     */
+    putBuffer(AudioBuffer buffer);
+};
diff --git a/audio/effect/5.0/IEffectsFactory.hal b/audio/effect/5.0/IEffectsFactory.hal
new file mode 100644
index 0000000..a88b381
--- /dev/null
+++ b/audio/effect/5.0/IEffectsFactory.hal
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IEffectsFactory {
+    /**
+     * Returns descriptors of different effects in all loaded libraries.
+     *
+     * @return retval operation completion status.
+     * @return result list of effect descriptors.
+     */
+    getAllDescriptors() generates(Result retval, vec<EffectDescriptor> result);
+
+    /**
+     * Returns a descriptor of a particular effect.
+     *
+     * @return retval operation completion status.
+     * @return result effect descriptor.
+     */
+    getDescriptor(Uuid uid) generates(Result retval, EffectDescriptor result);
+
+    /**
+     * Creates an effect engine of the specified type.  To release the effect
+     * engine, it is necessary to release references to the returned effect
+     * object.
+     *
+     * @param uid effect uuid.
+     * @param session audio session to which this effect instance will be
+     *                attached.  All effects created with the same session ID
+     *                are connected in series and process the same signal
+     *                stream.
+     * @param ioHandle identifies the output or input stream this effect is
+     *                 directed to in audio HAL.
+     * @return retval operation completion status.
+     * @return result the interface for the created effect.
+     * @return effectId the unique ID of the effect to be used with
+     *                  IStream::addEffect and IStream::removeEffect methods.
+     */
+    createEffect(Uuid uid, AudioSession session, AudioIoHandle ioHandle)
+        generates (Result retval, IEffect result, uint64_t effectId);
+};
diff --git a/audio/effect/5.0/IEnvironmentalReverbEffect.hal b/audio/effect/5.0/IEnvironmentalReverbEffect.hal
new file mode 100644
index 0000000..ffc4312
--- /dev/null
+++ b/audio/effect/5.0/IEnvironmentalReverbEffect.hal
@@ -0,0 +1,178 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IEnvironmentalReverbEffect extends IEffect {
+    /**
+     * Sets whether the effect should be bypassed.
+     */
+    setBypass(bool bypass) generates (Result retval);
+
+    /**
+     * Gets whether the effect should be bypassed.
+     */
+    getBypass() generates (Result retval, bool bypass);
+
+    enum ParamRange : int16_t {
+        ROOM_LEVEL_MIN = -6000,
+        ROOM_LEVEL_MAX = 0,
+        ROOM_HF_LEVEL_MIN = -4000,
+        ROOM_HF_LEVEL_MAX = 0,
+        DECAY_TIME_MIN = 100,
+        DECAY_TIME_MAX = 20000,
+        DECAY_HF_RATIO_MIN = 100,
+        DECAY_HF_RATIO_MAX = 1000,
+        REFLECTIONS_LEVEL_MIN = -6000,
+        REFLECTIONS_LEVEL_MAX = 0,
+        REFLECTIONS_DELAY_MIN = 0,
+        REFLECTIONS_DELAY_MAX = 65,
+        REVERB_LEVEL_MIN = -6000,
+        REVERB_LEVEL_MAX = 0,
+        REVERB_DELAY_MIN = 0,
+        REVERB_DELAY_MAX = 65,
+        DIFFUSION_MIN = 0,
+        DIFFUSION_MAX = 1000,
+        DENSITY_MIN = 0,
+        DENSITY_MAX = 1000
+    };
+
+    /**
+     * Sets the room level.
+     */
+    setRoomLevel(int16_t roomLevel) generates (Result retval);
+
+    /**
+     * Gets the room level.
+     */
+    getRoomLevel() generates (Result retval, int16_t roomLevel);
+
+    /**
+     * Sets the room high frequencies level.
+     */
+    setRoomHfLevel(int16_t roomHfLevel) generates (Result retval);
+
+    /**
+     * Gets the room high frequencies level.
+     */
+    getRoomHfLevel() generates (Result retval, int16_t roomHfLevel);
+
+    /**
+     * Sets the room decay time.
+     */
+    setDecayTime(uint32_t decayTime) generates (Result retval);
+
+    /**
+     * Gets the room decay time.
+     */
+    getDecayTime() generates (Result retval, uint32_t decayTime);
+
+    /**
+     * Sets the ratio of high frequencies decay.
+     */
+    setDecayHfRatio(int16_t decayHfRatio) generates (Result retval);
+
+    /**
+     * Gets the ratio of high frequencies decay.
+     */
+    getDecayHfRatio() generates (Result retval, int16_t decayHfRatio);
+
+    /**
+     * Sets the level of reflections in the room.
+     */
+    setReflectionsLevel(int16_t reflectionsLevel) generates (Result retval);
+
+    /**
+     * Gets the level of reflections in the room.
+     */
+    getReflectionsLevel() generates (Result retval, int16_t reflectionsLevel);
+
+    /**
+     * Sets the reflections delay in the room.
+     */
+    setReflectionsDelay(uint32_t reflectionsDelay) generates (Result retval);
+
+    /**
+     * Gets the reflections delay in the room.
+     */
+    getReflectionsDelay() generates (Result retval, uint32_t reflectionsDelay);
+
+    /**
+     * Sets the reverb level of the room.
+     */
+    setReverbLevel(int16_t reverbLevel) generates (Result retval);
+
+    /**
+     * Gets the reverb level of the room.
+     */
+    getReverbLevel() generates (Result retval, int16_t reverbLevel);
+
+    /**
+     * Sets the reverb delay of the room.
+     */
+    setReverbDelay(uint32_t reverDelay) generates (Result retval);
+
+    /**
+     * Gets the reverb delay of the room.
+     */
+    getReverbDelay() generates (Result retval, uint32_t reverbDelay);
+
+    /**
+     * Sets room diffusion.
+     */
+    setDiffusion(int16_t diffusion) generates (Result retval);
+
+    /**
+     * Gets room diffusion.
+     */
+    getDiffusion() generates (Result retval, int16_t diffusion);
+
+    /**
+     * Sets room wall density.
+     */
+    setDensity(int16_t density) generates (Result retval);
+
+    /**
+     * Gets room wall density.
+     */
+    getDensity() generates (Result retval, int16_t density);
+
+    struct AllProperties {
+        int16_t  roomLevel;         // in millibels,    range -6000 to 0
+        int16_t  roomHfLevel;       // in millibels,    range -4000 to 0
+        uint32_t decayTime;         // in milliseconds, range 100 to 20000
+        int16_t  decayHfRatio;      // in permilles,    range 100 to 1000
+        int16_t  reflectionsLevel;  // in millibels,    range -6000 to 0
+        uint32_t reflectionsDelay;  // in milliseconds, range 0 to 65
+        int16_t  reverbLevel;       // in millibels,    range -6000 to 0
+        uint32_t reverbDelay;       // in milliseconds, range 0 to 65
+        int16_t  diffusion;         // in permilles,    range 0 to 1000
+        int16_t  density;           // in permilles,    range 0 to 1000
+    };
+
+    /**
+     * Sets all properties at once.
+     */
+    setAllProperties(AllProperties properties) generates (Result retval);
+
+    /**
+     * Gets all properties at once.
+     */
+    getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/5.0/IEqualizerEffect.hal b/audio/effect/5.0/IEqualizerEffect.hal
new file mode 100644
index 0000000..c9829f3
--- /dev/null
+++ b/audio/effect/5.0/IEqualizerEffect.hal
@@ -0,0 +1,93 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IEqualizerEffect extends IEffect {
+    /**
+     * Gets the number of frequency bands that the equalizer supports.
+     */
+    getNumBands() generates (Result retval, uint16_t numBands);
+
+    /**
+     * Returns the minimum and maximum band levels supported.
+     */
+    getLevelRange()
+            generates (Result retval, int16_t minLevel, int16_t maxLevel);
+
+    /**
+     * Sets the gain for the given equalizer band.
+     */
+    setBandLevel(uint16_t band, int16_t level) generates (Result retval);
+
+    /**
+     * Gets the gain for the given equalizer band.
+     */
+    getBandLevel(uint16_t band) generates (Result retval, int16_t level);
+
+    /**
+     * Gets the center frequency of the given band, in milliHertz.
+     */
+    getBandCenterFrequency(uint16_t band)
+            generates (Result retval, uint32_t centerFreqmHz);
+
+    /**
+     * Gets the frequency range of the given frequency band, in milliHertz.
+     */
+    getBandFrequencyRange(uint16_t band)
+            generates (Result retval, uint32_t minFreqmHz, uint32_t maxFreqmHz);
+
+    /**
+     * Gets the band that has the most effect on the given frequency
+     * in milliHertz.
+     */
+    getBandForFrequency(uint32_t freqmHz)
+            generates (Result retval, uint16_t band);
+
+    /**
+     * Gets the names of all presets the equalizer supports.
+     */
+    getPresetNames() generates (Result retval, vec<string> names);
+
+    /**
+     * Sets the current preset using the index of the preset in the names
+     * vector returned via 'getPresetNames'.
+     */
+    setCurrentPreset(uint16_t preset) generates (Result retval);
+
+    /**
+     * Gets the current preset.
+     */
+    getCurrentPreset() generates (Result retval, uint16_t preset);
+
+    struct AllProperties {
+        uint16_t curPreset;
+        vec<int16_t> bandLevels;
+    };
+
+    /**
+     * Sets all properties at once.
+     */
+    setAllProperties(AllProperties properties) generates (Result retval);
+
+    /**
+     * Gets all properties at once.
+     */
+    getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/5.0/ILoudnessEnhancerEffect.hal b/audio/effect/5.0/ILoudnessEnhancerEffect.hal
new file mode 100644
index 0000000..eaf8949
--- /dev/null
+++ b/audio/effect/5.0/ILoudnessEnhancerEffect.hal
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface ILoudnessEnhancerEffect extends IEffect {
+    /**
+     * Sets target gain expressed in millibels.
+     */
+    setTargetGain(int32_t targetGainMb) generates (Result retval);
+
+    /**
+     * Gets target gain expressed in millibels.
+     */
+    getTargetGain() generates (Result retval, int32_t targetGainMb);
+};
diff --git a/audio/effect/5.0/INoiseSuppressionEffect.hal b/audio/effect/5.0/INoiseSuppressionEffect.hal
new file mode 100644
index 0000000..5de794f
--- /dev/null
+++ b/audio/effect/5.0/INoiseSuppressionEffect.hal
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface INoiseSuppressionEffect extends IEffect {
+    enum Level : int32_t {
+        LOW,
+        MEDIUM,
+        HIGH
+    };
+
+    /**
+     * Sets suppression level.
+     */
+    setSuppressionLevel(Level level) generates (Result retval);
+
+    /**
+     * Gets suppression level.
+     */
+    getSuppressionLevel() generates (Result retval, Level level);
+
+    enum Type : int32_t {
+        SINGLE_CHANNEL,
+        MULTI_CHANNEL
+    };
+
+    /**
+     * Set suppression type.
+     */
+    setSuppressionType(Type type) generates (Result retval);
+
+    /**
+     * Get suppression type.
+     */
+    getSuppressionType() generates (Result retval, Type type);
+
+    struct AllProperties {
+        Level level;
+        Type type;
+    };
+
+    /**
+     * Sets all properties at once.
+     */
+    setAllProperties(AllProperties properties) generates (Result retval);
+
+    /**
+     * Gets all properties at once.
+     */
+    getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/5.0/IPresetReverbEffect.hal b/audio/effect/5.0/IPresetReverbEffect.hal
new file mode 100644
index 0000000..af3b3ac
--- /dev/null
+++ b/audio/effect/5.0/IPresetReverbEffect.hal
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IPresetReverbEffect extends IEffect {
+    enum Preset : int32_t {
+        NONE,        // no reverb or reflections
+        SMALLROOM,   // a small room less than five meters in length
+        MEDIUMROOM,  // a medium room with a length of ten meters or less
+        LARGEROOM,   // a large-sized room suitable for live performances
+        MEDIUMHALL,  // a medium-sized hall
+        LARGEHALL,   // a large-sized hall suitable for a full orchestra
+        PLATE,       // synthesis of the traditional plate reverb
+        LAST = PLATE
+    };
+
+    /**
+     * Sets the current preset.
+     */
+    setPreset(Preset preset) generates (Result retval);
+
+    /**
+     * Gets the current preset.
+     */
+    getPreset() generates (Result retval, Preset preset);
+};
diff --git a/audio/effect/5.0/IVirtualizerEffect.hal b/audio/effect/5.0/IVirtualizerEffect.hal
new file mode 100644
index 0000000..9402958
--- /dev/null
+++ b/audio/effect/5.0/IVirtualizerEffect.hal
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IVirtualizerEffect extends IEffect {
+    /**
+     * Returns whether setting virtualization strength is supported.
+     */
+    isStrengthSupported() generates (bool strengthSupported);
+
+    enum StrengthRange : uint16_t {
+        MIN = 0,
+        MAX = 1000
+    };
+
+    /**
+     * Sets virtualization strength.
+     *
+     * @param strength strength of the effect. The valid range for strength
+     *                 strength is [0, 1000], where 0 per mille designates the
+     *                 mildest effect and 1000 per mille designates the
+     *                 strongest.
+     * @return retval operation completion status.
+     */
+    setStrength(uint16_t strength) generates (Result retval);
+
+    /**
+     * Gets virtualization strength.
+     */
+    getStrength() generates (Result retval, uint16_t strength);
+
+    struct SpeakerAngle {
+        /** Speaker channel mask */
+        bitfield<AudioChannelMask> mask;
+        // all angles are expressed in degrees and
+        // are relative to the listener.
+        int16_t azimuth; // 0 is the direction the listener faces
+                         // 180 is behind the listener
+                         // -90 is to their left
+        int16_t elevation; // 0 is the horizontal plane
+                           // +90 is above the listener, -90 is below
+    };
+    /**
+     * Retrieves virtual speaker angles for the given channel mask on the
+     * specified device.
+     */
+    getVirtualSpeakerAngles(bitfield<AudioChannelMask> mask, AudioDevice device)
+            generates (Result retval, vec<SpeakerAngle> speakerAngles);
+
+    /**
+     * Forces the virtualizer effect for the given output device.
+     */
+    forceVirtualizationMode(AudioDevice device) generates (Result retval);
+
+    /**
+     * Returns audio device reflecting the current virtualization mode,
+     * AUDIO_DEVICE_NONE when not virtualizing.
+     */
+    getVirtualizationMode() generates (Result retval, AudioDevice device);
+};
diff --git a/audio/effect/5.0/IVisualizerEffect.hal b/audio/effect/5.0/IVisualizerEffect.hal
new file mode 100644
index 0000000..38752a9
--- /dev/null
+++ b/audio/effect/5.0/IVisualizerEffect.hal
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+import IEffect;
+
+interface IVisualizerEffect extends IEffect {
+    enum CaptureSizeRange : int32_t {
+        MAX = 1024,  // maximum capture size in samples
+        MIN = 128    // minimum capture size in samples
+    };
+
+    /**
+     * Sets the number PCM samples in the capture.
+     */
+    setCaptureSize(uint16_t captureSize) generates (Result retval);
+
+    /**
+     * Gets the number PCM samples in the capture.
+     */
+    getCaptureSize() generates (Result retval, uint16_t captureSize);
+
+    enum ScalingMode : int32_t {
+        // Keep in sync with SCALING_MODE_... in
+        // frameworks/base/media/java/android/media/audiofx/Visualizer.java
+        NORMALIZED = 0,
+        AS_PLAYED = 1
+    };
+
+    /**
+     * Specifies the way the captured data is scaled.
+     */
+    setScalingMode(ScalingMode scalingMode) generates (Result retval);
+
+    /**
+     * Retrieves the way the captured data is scaled.
+     */
+    getScalingMode() generates (Result retval, ScalingMode scalingMode);
+
+    /**
+     * Informs the visualizer about the downstream latency.
+     */
+    setLatency(uint32_t latencyMs) generates (Result retval);
+
+    /**
+     * Gets the downstream latency.
+     */
+    getLatency() generates (Result retval, uint32_t latencyMs);
+
+    enum MeasurementMode : int32_t {
+        // Keep in sync with MEASUREMENT_MODE_... in
+        // frameworks/base/media/java/android/media/audiofx/Visualizer.java
+        NONE = 0x0,
+        PEAK_RMS = 0x1
+    };
+
+    /**
+     * Specifies which measurements are to be made.
+     */
+    setMeasurementMode(MeasurementMode measurementMode)
+            generates (Result retval);
+
+    /**
+     * Retrieves which measurements are to be made.
+     */
+    getMeasurementMode() generates (
+            Result retval, MeasurementMode measurementMode);
+
+    /**
+     * Retrieves the latest PCM snapshot captured by the visualizer engine.  The
+     * number of samples to capture is specified by 'setCaptureSize' parameter.
+     *
+     * @return retval operation completion status.
+     * @return samples samples in 8 bit unsigned format (0 = 0x80)
+     */
+    capture() generates (Result retval, vec<uint8_t> samples);
+
+    struct Measurement {
+        MeasurementMode mode;    // discriminator
+        union Values {
+            struct PeakAndRms {
+                int32_t peakMb;  // millibels
+                int32_t rmsMb;   // millibels
+            } peakAndRms;
+        } value;
+    };
+    /**
+     * Retrieves the latest measurements. The measurements to be made
+     * are specified by 'setMeasurementMode' parameter.
+     *
+     * @return retval operation completion status.
+     * @return result measurement.
+     */
+    measure() generates (Result retval, Measurement result);
+};
diff --git a/audio/effect/5.0/types.hal b/audio/effect/5.0/types.hal
new file mode 100644
index 0000000..84e1108
--- /dev/null
+++ b/audio/effect/5.0/types.hal
@@ -0,0 +1,299 @@
+/*
+ * Copyright (C) 2018 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.effect@5.0;
+
+import android.hardware.audio.common@5.0;
+
+enum Result : int32_t {
+    OK,
+    NOT_INITIALIZED,
+    INVALID_ARGUMENTS,
+    INVALID_STATE,
+    NOT_SUPPORTED,
+    RESULT_TOO_BIG
+};
+
+/**
+ * Effect engine capabilities/requirements flags.
+ *
+ * Definitions for flags field of effect descriptor.
+ *
+ * +----------------+--------+--------------------------------------------------
+ * | description    | bits   | values
+ * +----------------+--------+--------------------------------------------------
+ * | connection     | 0..2   | 0 insert: after track process
+ * | mode           |        | 1 auxiliary: connect to track auxiliary
+ * |                |        |  output and use send level
+ * |                |        | 2 replace: replaces track process function;
+ * |                |        |   must implement SRC, volume and mono to stereo.
+ * |                |        | 3 pre processing: applied below audio HAL on in
+ * |                |        | 4 post processing: applied below audio HAL on out
+ * |                |        | 5 - 7 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | insertion      | 3..5   | 0 none
+ * | preference     |        | 1 first of the chain
+ * |                |        | 2 last of the chain
+ * |                |        | 3 exclusive (only effect in the insert chain)
+ * |                |        | 4..7 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Volume         | 6..8   | 0 none
+ * | management     |        | 1 implements volume control
+ * |                |        | 2 requires volume indication
+ * |                |        | 4 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Device         | 9..11  | 0 none
+ * | indication     |        | 1 requires device updates
+ * |                |        | 2, 4 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Sample input   | 12..13 | 1 direct: process() function or
+ * | mode           |        |   EFFECT_CMD_SET_CONFIG command must specify
+ * |                |        |   a buffer descriptor
+ * |                |        | 2 provider: process() function uses the
+ * |                |        |   bufferProvider indicated by the
+ * |                |        |   EFFECT_CMD_SET_CONFIG command to request input.
+ * |                |        |   buffers.
+ * |                |        | 3 both: both input modes are supported
+ * +----------------+--------+--------------------------------------------------
+ * | Sample output  | 14..15 | 1 direct: process() function or
+ * | mode           |        |   EFFECT_CMD_SET_CONFIG command must specify
+ * |                |        |   a buffer descriptor
+ * |                |        | 2 provider: process() function uses the
+ * |                |        |   bufferProvider indicated by the
+ * |                |        |   EFFECT_CMD_SET_CONFIG command to request output
+ * |                |        |   buffers.
+ * |                |        | 3 both: both output modes are supported
+ * +----------------+--------+--------------------------------------------------
+ * | Hardware       | 16..17 | 0 No hardware acceleration
+ * | acceleration   |        | 1 non tunneled hw acceleration: the process()
+ * |                |        |   function reads the samples, send them to HW
+ * |                |        |   accelerated effect processor, reads back
+ * |                |        |   the processed samples and returns them
+ * |                |        |   to the output buffer.
+ * |                |        | 2 tunneled hw acceleration: the process()
+ * |                |        |   function is transparent. The effect interface
+ * |                |        |   is only used to control the effect engine.
+ * |                |        |   This mode is relevant for global effects
+ * |                |        |   actually applied by the audio hardware on
+ * |                |        |   the output stream.
+ * +----------------+--------+--------------------------------------------------
+ * | Audio Mode     | 18..19 | 0 none
+ * | indication     |        | 1 requires audio mode updates
+ * |                |        | 2..3 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Audio source   | 20..21 | 0 none
+ * | indication     |        | 1 requires audio source updates
+ * |                |        | 2..3 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Effect offload | 22     | 0 The effect cannot be offloaded to an audio DSP
+ * | supported      |        | 1 The effect can be offloaded to an audio DSP
+ * +----------------+--------+--------------------------------------------------
+ * | Process        | 23     | 0 The effect implements a process function.
+ * | function       |        | 1 The effect does not implement a process
+ * | not            |        |   function: enabling the effect has no impact
+ * | implemented    |        |   on latency or CPU load.
+ * |                |        |   Effect implementations setting this flag do not
+ * |                |        |   have to implement a process function.
+ * +----------------+--------+--------------------------------------------------
+ */
+@export(name="", value_prefix="EFFECT_FLAG_")
+enum EffectFlags : int32_t {
+    // Insert mode
+    TYPE_SHIFT = 0,
+    TYPE_SIZE = 3,
+    TYPE_MASK = ((1 << TYPE_SIZE) -1) << TYPE_SHIFT,
+    TYPE_INSERT = 0 << TYPE_SHIFT,
+    TYPE_AUXILIARY = 1 << TYPE_SHIFT,
+    TYPE_REPLACE = 2 << TYPE_SHIFT,
+    TYPE_PRE_PROC = 3 << TYPE_SHIFT,
+    TYPE_POST_PROC = 4 << TYPE_SHIFT,
+
+    // Insert preference
+    INSERT_SHIFT = TYPE_SHIFT + TYPE_SIZE,
+    INSERT_SIZE = 3,
+    INSERT_MASK = ((1 << INSERT_SIZE) -1) << INSERT_SHIFT,
+    INSERT_ANY = 0 << INSERT_SHIFT,
+    INSERT_FIRST = 1 << INSERT_SHIFT,
+    INSERT_LAST = 2 << INSERT_SHIFT,
+    INSERT_EXCLUSIVE = 3 << INSERT_SHIFT,
+
+    // Volume control
+    VOLUME_SHIFT = INSERT_SHIFT + INSERT_SIZE,
+    VOLUME_SIZE = 3,
+    VOLUME_MASK = ((1 << VOLUME_SIZE) -1) << VOLUME_SHIFT,
+    VOLUME_CTRL = 1 << VOLUME_SHIFT,
+    VOLUME_IND = 2 << VOLUME_SHIFT,
+    VOLUME_NONE = 0 << VOLUME_SHIFT,
+
+    // Device indication
+    DEVICE_SHIFT = VOLUME_SHIFT + VOLUME_SIZE,
+    DEVICE_SIZE = 3,
+    DEVICE_MASK = ((1 << DEVICE_SIZE) -1) << DEVICE_SHIFT,
+    DEVICE_IND = 1 << DEVICE_SHIFT,
+    DEVICE_NONE = 0 << DEVICE_SHIFT,
+
+    // Sample input modes
+    INPUT_SHIFT = DEVICE_SHIFT + DEVICE_SIZE,
+    INPUT_SIZE = 2,
+    INPUT_MASK = ((1 << INPUT_SIZE) -1) << INPUT_SHIFT,
+    INPUT_DIRECT = 1 << INPUT_SHIFT,
+    INPUT_PROVIDER = 2 << INPUT_SHIFT,
+    INPUT_BOTH = 3 << INPUT_SHIFT,
+
+    // Sample output modes
+    OUTPUT_SHIFT = INPUT_SHIFT + INPUT_SIZE,
+    OUTPUT_SIZE = 2,
+    OUTPUT_MASK = ((1 << OUTPUT_SIZE) -1) << OUTPUT_SHIFT,
+    OUTPUT_DIRECT = 1 << OUTPUT_SHIFT,
+    OUTPUT_PROVIDER = 2 << OUTPUT_SHIFT,
+    OUTPUT_BOTH = 3 << OUTPUT_SHIFT,
+
+    // Hardware acceleration mode
+    HW_ACC_SHIFT = OUTPUT_SHIFT + OUTPUT_SIZE,
+    HW_ACC_SIZE = 2,
+    HW_ACC_MASK = ((1 << HW_ACC_SIZE) -1) << HW_ACC_SHIFT,
+    HW_ACC_SIMPLE = 1 << HW_ACC_SHIFT,
+    HW_ACC_TUNNEL = 2 << HW_ACC_SHIFT,
+
+    // Audio mode indication
+    AUDIO_MODE_SHIFT = HW_ACC_SHIFT + HW_ACC_SIZE,
+    AUDIO_MODE_SIZE = 2,
+    AUDIO_MODE_MASK = ((1 << AUDIO_MODE_SIZE) -1) << AUDIO_MODE_SHIFT,
+    AUDIO_MODE_IND = 1 << AUDIO_MODE_SHIFT,
+    AUDIO_MODE_NONE = 0 << AUDIO_MODE_SHIFT,
+
+    // Audio source indication
+    AUDIO_SOURCE_SHIFT = AUDIO_MODE_SHIFT + AUDIO_MODE_SIZE,
+    AUDIO_SOURCE_SIZE = 2,
+    AUDIO_SOURCE_MASK = ((1 << AUDIO_SOURCE_SIZE) -1) << AUDIO_SOURCE_SHIFT,
+    AUDIO_SOURCE_IND = 1 << AUDIO_SOURCE_SHIFT,
+    AUDIO_SOURCE_NONE = 0 << AUDIO_SOURCE_SHIFT,
+
+    // Effect offload indication
+    OFFLOAD_SHIFT = AUDIO_SOURCE_SHIFT + AUDIO_SOURCE_SIZE,
+    OFFLOAD_SIZE = 1,
+    OFFLOAD_MASK = ((1 << OFFLOAD_SIZE) -1) << OFFLOAD_SHIFT,
+    OFFLOAD_SUPPORTED = 1 << OFFLOAD_SHIFT,
+
+    // Effect has no process indication
+    NO_PROCESS_SHIFT = OFFLOAD_SHIFT + OFFLOAD_SIZE,
+    NO_PROCESS_SIZE = 1,
+    NO_PROCESS_MASK = ((1 << NO_PROCESS_SIZE) -1) << NO_PROCESS_SHIFT,
+    NO_PROCESS = 1 << NO_PROCESS_SHIFT
+};
+
+/**
+ * The effect descriptor contains necessary information to facilitate the
+ * enumeration of the effect engines present in a library.
+ */
+struct EffectDescriptor {
+    Uuid type;                   // UUID of to the OpenSL ES interface implemented
+                                 // by this effect
+    Uuid uuid;                   // UUID for this particular implementation
+    bitfield<EffectFlags> flags; // effect engine capabilities/requirements flags
+    uint16_t cpuLoad;            // CPU load indication expressed in 0.1 MIPS units
+                                 // as estimated on an ARM9E core (ARMv5TE) with 0 WS
+    uint16_t memoryUsage;        // data memory usage expressed in KB and includes
+                                 // only dynamically allocated memory
+    uint8_t[64] name;            // human readable effect name
+    uint8_t[64] implementor;     // human readable effect implementor name
+};
+
+/**
+ * A buffer is a chunk of audio data for processing.  Multi-channel audio is
+ * always interleaved. The channel order is from LSB to MSB with regard to the
+ * channel mask definition in audio.h, audio_channel_mask_t, e.g.:
+ * Stereo: L, R; 5.1: FL, FR, FC, LFE, BL, BR.
+ *
+ * The buffer size is expressed in frame count, a frame being composed of
+ * samples for all channels at a given time. Frame size for unspecified format
+ * (AUDIO_FORMAT_OTHER) is 8 bit by definition.
+ */
+struct AudioBuffer {
+    uint64_t id;
+    uint32_t frameCount;
+    memory data;
+};
+
+@export(name="effect_buffer_access_e", value_prefix="EFFECT_BUFFER_")
+enum EffectBufferAccess : int32_t {
+    ACCESS_WRITE,
+    ACCESS_READ,
+    ACCESS_ACCUMULATE
+};
+
+/**
+ * Determines what fields of EffectBufferConfig need to be considered.
+ */
+@export(name="", value_prefix="EFFECT_CONFIG_")
+enum EffectConfigParameters : int32_t {
+    BUFFER = 0x0001,    // buffer field
+    SMP_RATE = 0x0002,  // samplingRate
+    CHANNELS = 0x0004,  // channels
+    FORMAT = 0x0008,    // format
+    ACC_MODE = 0x0010,  // accessMode
+    // Note that the 2.0 ALL have been moved to an helper function
+};
+
+/**
+ * The buffer config structure specifies the input or output audio format
+ * to be used by the effect engine.
+ */
+struct EffectBufferConfig {
+    AudioBuffer buffer;
+    uint32_t samplingRateHz;
+    bitfield<AudioChannelMask> channels;
+    AudioFormat format;
+    EffectBufferAccess accessMode;
+    bitfield<EffectConfigParameters> mask;
+};
+
+struct EffectConfig {
+    EffectBufferConfig inputCfg;
+    EffectBufferConfig outputCfg;
+};
+
+@export(name="effect_feature_e", value_prefix="EFFECT_FEATURE_")
+enum EffectFeature : int32_t {
+    AUX_CHANNELS, // supports auxiliary channels
+                  // (e.g. dual mic noise suppressor)
+    CNT
+};
+
+struct EffectAuxChannelsConfig {
+    bitfield<AudioChannelMask> mainChannels;  // channel mask for main channels
+    bitfield<AudioChannelMask> auxChannels;   // channel mask for auxiliary channels
+};
+
+struct EffectOffloadParameter {
+    bool isOffload;          // true if the playback thread the effect
+                             // is attached to is offloaded
+    AudioIoHandle ioHandle;  // io handle of the playback thread
+                             // the effect is attached to
+};
+
+/**
+ * The message queue flags used to synchronize reads and writes from
+ * the status message queue used by effects.
+ */
+enum MessageQueueFlagBits : uint32_t {
+    DONE_PROCESSING = 1 << 0,
+    REQUEST_PROCESS = 1 << 1,
+    REQUEST_PROCESS_REVERSE = 1 << 2,
+    REQUEST_QUIT = 1 << 3,
+    REQUEST_PROCESS_ALL =
+        REQUEST_PROCESS | REQUEST_PROCESS_REVERSE | REQUEST_QUIT
+};
diff --git a/audio/effect/5.0/xml/audio_effects_conf.xsd b/audio/effect/5.0/xml/audio_effects_conf.xsd
new file mode 120000
index 0000000..9d85fa7
--- /dev/null
+++ b/audio/effect/5.0/xml/audio_effects_conf.xsd
@@ -0,0 +1 @@
+../../2.0/xml/audio_effects_conf.xsd
\ No newline at end of file