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;
+};