Audio V4: Copy 2.0 .hal files in 4.0

That will ease review of the changes.
The only changes are:
 - replace all @2.0 by 4.0
   $ sed -i 's/@2\.0/@4\.0/g' */4.0/*.hal
 - replace all licence 2016 by 2018
   $ sed -i 's/2016/2018/g' */4.0/*.hal

Moving the .hal in a subfolder forces the package name and the
namespace to change.

This mean that the audio HAL 2.0 and 4.0 will not be consider
different version of the same HAL but two different HALs.

As a result to minimize code change due to tight deadline,
keep the 4.0 core in the audio folder.

Bug: 38184704
Test: hardware/interfaces/update-makefiles.sh
Change-Id: I7c7a826270c9933091f037b795806787e1284583
Signed-off-by: Kevin Rocard <krocard@google.com>

8 files changed, 1251 insertions, 0 deletions

diff --git a/audio/4.0/IDevice.hal b/audio/4.0/IDevice.hal
new file mode 100644
index 000000000..e3f36f6e2
--- /dev/null
+++ b/audio/4.0/IDevice.hal
@@ -0,0 +1,246 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+import IStreamIn;
+import IStreamOut;
+
+interface IDevice {
+    typedef android.hardware.audio@4.0::Result Result;
+
+    /**
+     * 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.
+     * @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,
+            AudioOutputFlag flags) 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 source source specification.
+     * @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,
+            AudioInputFlag flags,
+            AudioSource source) 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.
+     *
+     * @return hwAvSync HW synchronization source
+     */
+    getHwAvSync() generates (AudioHwSync hwAvSync);
+
+    /**
+     * Sets whether the screen is on. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_KEY_SCREEN_STATE on the legacy HAL.
+     *
+     * @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.
+     *
+     * @param keys parameter keys.
+     * @return retval operation completion status.
+     * @return parameters parameter key value pairs.
+     */
+    getParameters(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.
+     *
+     * @param parameters parameter key value pairs.
+     * @return retval operation completion status.
+     */
+    setParameters(vec<ParameterValue> parameters) generates (Result retval);
+
+    /**
+     * Dumps information about the stream into the provided file descriptor.
+     * This is used for the dumpsys facility.
+     *
+     * @param fd dump file descriptor.
+     */
+    debugDump(handle fd);
+};
diff --git a/audio/4.0/IDevicesFactory.hal b/audio/4.0/IDevicesFactory.hal
new file mode 100644
index 000000000..5a6242236
--- /dev/null
+++ b/audio/4.0/IDevicesFactory.hal
@@ -0,0 +1,45 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+import IDevice;
+
+interface IDevicesFactory {
+    typedef android.hardware.audio@4.0::Result Result;
+
+    enum Device : int32_t {
+        PRIMARY,
+        A2DP,
+        USB,
+        R_SUBMIX,
+        STUB
+    };
+
+    /**
+     * Opens an audio device. To close the device, it is necessary to release
+     * references to the returned device object.
+     *
+     * @param device device type.
+     * @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(Device device) generates (Result retval, IDevice result);
+};
diff --git a/audio/4.0/IPrimaryDevice.hal b/audio/4.0/IPrimaryDevice.hal
new file mode 100644
index 000000000..60330fbed
--- /dev/null
+++ b/audio/4.0/IPrimaryDevice.hal
@@ -0,0 +1,123 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+import IDevice;
+
+interface IPrimaryDevice extends IDevice {
+    typedef android.hardware.audio@4.0::Result Result;
+
+    /**
+     * 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);
+
+    /**
+     * 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.
+     *
+     * @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.
+     *
+     * @param enabled whether BT Wideband is enabled.
+     * @return retval operation completion status.
+     */
+    setBtScoWidebandEnabled(bool enabled) 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.
+     *
+     * @param enabled whether HAC mode is enabled.
+     * @return retval operation completion status.
+     */
+    setHacEnabled(bool enabled) generates (Result retval);
+};
diff --git a/audio/4.0/IStream.hal b/audio/4.0/IStream.hal
new file mode 100644
index 000000000..162e0fd0d
--- /dev/null
+++ b/audio/4.0/IStream.hal
@@ -0,0 +1,294 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+import android.hardware.audio.effect@4.0::IEffect;
+
+interface IStream {
+    typedef android.hardware.audio@4.0::Result Result;
+
+    /**
+     * 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 sampling rates of the stream. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES on the
+     * legacy HAL.
+     *
+     * @return sampleRateHz supported sample rates.
+     */
+    getSupportedSampleRates() generates (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.
+     *
+     * @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 (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.
+     *
+     * @return masks supported audio masks.
+     */
+    getSupportedChannelMasks() generates (vec<AudioChannelMask> masks);
+
+    /**
+     * Sets the channel mask of the stream. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_STREAM_CHANNELS on the legacy HAL.
+     *
+     * @param format audio format.
+     * @return retval operation completion status.
+     */
+    setChannelMask(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.
+     *
+     * @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, 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 device(s) which this stream is connected to.
+     *
+     * @return device set of device(s) which this stream is connected to.
+     */
+    getDevice() generates (AudioDevice device);
+
+    /**
+     * Connects the stream to the device.
+     *
+     * 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 in the legacy HAL
+     * interface.
+     *
+     * @param address device to connect the stream to.
+     * @return retval operation completion status.
+     */
+    setDevice(DeviceAddress address) generates (Result retval);
+
+    /**
+     * Notifies the stream about device connection state. 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);
+
+    /**
+     * Sets the HW synchronization source. Calling this method is equivalent to
+     * setting AUDIO_PARAMETER_STREAM_HW_AV_SYNC on the legacy HAL.
+     *
+     * @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.
+     *
+     * @param keys parameter keys.
+     * @return retval operation completion status.
+     * @return parameters parameter key value pairs.
+     */
+    getParameters(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.
+     *
+     * @param parameters parameter key value pairs.
+     * @return retval operation completion status.
+     */
+    setParameters(vec<ParameterValue> parameters) generates (Result retval);
+
+    /**
+     * Dumps information about the stream into the provided file descriptor.
+     * This is used for the dumpsys facility.
+     *
+     * @param fd dump file descriptor.
+     */
+    debugDump(handle fd);
+
+    /**
+     * 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/4.0/IStreamIn.hal b/audio/4.0/IStreamIn.hal
new file mode 100644
index 000000000..306aeb131
--- /dev/null
+++ b/audio/4.0/IStreamIn.hal
@@ -0,0 +1,151 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+import IStream;
+
+interface IStreamIn extends IStream {
+    typedef android.hardware.audio@4.0::Result Result;
+
+    /**
+     * Returns the source descriptor of the input stream. Calling this method is
+     * equivalent to getting AUDIO_PARAMETER_STREAM_INPUT_SOURCE on the legacy
+     * HAL.
+     *
+     * @return retval operation completion status.
+     * @return source audio source.
+     */
+    getAudioSource() generates (Result retval, AudioSource source);
+
+    /**
+     * Set the input gain for the audio driver.
+     *
+     * @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;
+    };
+
+    /**
+     * 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);
+};
diff --git a/audio/4.0/IStreamOut.hal b/audio/4.0/IStreamOut.hal
new file mode 100644
index 000000000..22b91f050
--- /dev/null
+++ b/audio/4.0/IStreamOut.hal
@@ -0,0 +1,253 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+import IStream;
+import IStreamOutCallback;
+
+interface IStreamOut extends IStream {
+    typedef android.hardware.audio@4.0::Result Result;
+
+    /**
+     * 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.
+     *
+     * @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.
+     */
+    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;
+    };
+
+    /**
+     * 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.
+     *
+     * @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.
+     *
+     * @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.
+     *
+     * @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'.
+     *
+     * 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.
+     *
+     * @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);
+};
diff --git a/audio/4.0/IStreamOutCallback.hal b/audio/4.0/IStreamOutCallback.hal
new file mode 100644
index 000000000..9a19d325c
--- /dev/null
+++ b/audio/4.0/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@4.0;
+
+/**
+ * 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/4.0/types.hal b/audio/4.0/types.hal
new file mode 100644
index 000000000..cd9a39406
--- /dev/null
+++ b/audio/4.0/types.hal
@@ -0,0 +1,102 @@
+/*
+ * 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@4.0;
+
+import android.hardware.audio.common@4.0;
+
+enum Result : int32_t {
+    OK,
+    NOT_INITIALIZED,
+    INVALID_ARGUMENTS,
+    INVALID_STATE,
+    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
+};
+
+/**
+ * Mmap buffer descriptor returned by IStream.createMmapBuffer().
+ * Used by streams opened in mmap mode.
+ */
+struct MmapBufferInfo {
+    memory  sharedMemory;         // mmap memory buffer
+    int32_t bufferSizeFrames;     // total buffer size in frames
+    int32_t burstSizeFrames;      // transfer size granularity in frames
+};
+
+/**
+ * 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
+};