Audio HAL V5: Introduce HAL V5, equal to V4 for now
Port audio HAL V5 to AOSP for BT HAL.
The implementation is not ported as that would require to port an
additional ~30 patches and is not needed by the BT team.
Bug: 118203066
Test: Compile
Change-Id: If99a5645d19c9780019704ea4f51f8114d83ee8e
Merged-In: If99a5645d19c9780019704ea4f51f8114d83ee8f
Signed-off-by: Kevin Rocard <krocard@google.com>
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;
+};