Audio: Copy HAL V6 into V7
This is an automated copy performed using copyHAL.sh script.
Bug: 142480271
Test: m
Change-Id: Ifd91cc0bb59608cd92d1d8e4e76c3abea0a8da5e
diff --git a/audio/7.0/Android.bp b/audio/7.0/Android.bp
new file mode 100644
index 0000000..d07ce12
--- /dev/null
+++ b/audio/7.0/Android.bp
@@ -0,0 +1,25 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+ name: "android.hardware.audio@7.0",
+ root: "android.hardware",
+ srcs: [
+ "types.hal",
+ "IDevice.hal",
+ "IDevicesFactory.hal",
+ "IPrimaryDevice.hal",
+ "IStream.hal",
+ "IStreamIn.hal",
+ "IStreamOut.hal",
+ "IStreamOutCallback.hal",
+ "IStreamOutEventCallback.hal",
+ ],
+ interfaces: [
+ "android.hardware.audio.common@7.0",
+ "android.hardware.audio.effect@7.0",
+ "android.hidl.base@1.0",
+ "android.hidl.safe_union@1.0",
+ ],
+ gen_java: false,
+ gen_java_constants: true,
+}
diff --git a/audio/7.0/IDevice.hal b/audio/7.0/IDevice.hal
new file mode 100644
index 0000000..7082d6b
--- /dev/null
+++ b/audio/7.0/IDevice.hal
@@ -0,0 +1,346 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+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 processing 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. Patch represents a connection
+ * between signal source(s) and signal sink(s). If HAL doesn't support
+ * patches natively (in hardware) then audio system will need to establish
+ * them in software.
+ *
+ * @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);
+
+ /**
+ * Updates an audio patch.
+ *
+ * Use of this function is preferred to releasing and re-creating a patch
+ * as the HAL module can figure out a way of switching the route without
+ * causing audio disruption.
+ *
+ * @param previousPatch handle of the previous patch to update.
+ * @param sources new patch sources.
+ * @param sinks new patch sinks.
+ * @return retval operation completion status.
+ * @return patch updated patch handle.
+ */
+ updateAudioPatch(
+ AudioPatchHandle previousPatch,
+ 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 NOT_SUPPORTED if there are no microphones on this device
+ * 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);
+
+ /**
+ * Called by the framework to deinitialize the device and free up
+ * all currently allocated resources. It is recommended to close
+ * the device on the client side as soon as it is becomes unused.
+ *
+ * Note that all streams must be closed by the client before
+ * attempting to close the device they belong to.
+ *
+ * @return retval OK in case the success.
+ * INVALID_STATE if the device was already closed
+ * or there are streams currently opened.
+ */
+ @exit
+ close() generates (Result retval);
+
+ /**
+ * Applies an audio effect to an audio device. The effect is inserted
+ * according to its insertion preference specified by INSERT_... EffectFlags
+ * in the EffectDescriptor.
+ *
+ * @param device identifies the sink or source device this effect must be applied to.
+ * "device" is the AudioPortHandle indicated for the device when the audio
+ * patch connecting that device was created.
+ * @param effectId effect ID (obtained from IEffectsFactory.createEffect) of
+ * the effect to add.
+ * @return retval operation completion status.
+ */
+ addDeviceEffect(AudioPortHandle device, uint64_t effectId) generates (Result retval);
+
+ /**
+ * Stops applying an audio effect to an audio device.
+ *
+ * @param device identifies the sink or source device this effect was applied to.
+ * "device" is the AudioPortHandle indicated for the device when the audio
+ * patch is created at the audio HAL.
+ * @param effectId effect ID (obtained from IEffectsFactory.createEffect) of
+ * the effect.
+ * @return retval operation completion status.
+ */
+ removeDeviceEffect(AudioPortHandle device, uint64_t effectId) generates (Result retval);
+};
diff --git a/audio/7.0/IDevicesFactory.hal b/audio/7.0/IDevicesFactory.hal
new file mode 100644
index 0000000..03549b4
--- /dev/null
+++ b/audio/7.0/IDevicesFactory.hal
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+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 occurred 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 debugging, 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/7.0/IPrimaryDevice.hal b/audio/7.0/IPrimaryDevice.hal
new file mode 100644
index 0000000..1427ae8
--- /dev/null
+++ b/audio/7.0/IPrimaryDevice.hal
@@ -0,0 +1,195 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+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/7.0/IStream.hal b/audio/7.0/IStream.hal
new file mode 100644
index 0000000..dacd3fd
--- /dev/null
+++ b/audio/7.0/IStream.hal
@@ -0,0 +1,317 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+import android.hardware.audio.effect@7.0::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 retval operation completion status.
+ * @return formats supported audio formats.
+ * Must be non empty if retval is OK.
+ */
+ getSupportedFormats() generates (Result retval, 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 preceded
+ * 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 success.
+ * 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.
+ * The size must be a positive value.
+ * @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 invalid
+ * 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 currently allocated resources. It is recommended to close
+ * the stream on the client side as soon as it is becomes unused.
+ *
+ * The client must ensure that this function is not called while
+ * audio data is being transferred through the stream's message queues.
+ *
+ * @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.
+ */
+ @exit
+ close() generates (Result retval);
+};
diff --git a/audio/7.0/IStreamIn.hal b/audio/7.0/IStreamIn.hal
new file mode 100644
index 0000000..15e4363
--- /dev/null
+++ b/audio/7.0/IStreamIn.hal
@@ -0,0 +1,199 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+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);
+
+ /**
+ * Specifies the logical microphone (for processing).
+ *
+ * If the feature is not supported an error should be returned
+ * If multiple microphones are present, this should be treated as a preference
+ * for their combined direction.
+ *
+ * Optional method
+ *
+ * @param Direction constant
+ * @return retval OK if the call is successful, an error code otherwise.
+ */
+ setMicrophoneDirection(MicrophoneDirection direction)
+ generates(Result retval);
+
+ /**
+ * Specifies the zoom factor for the selected microphone (for processing).
+ *
+ * If the feature is not supported an error should be returned
+ * If multiple microphones are present, this should be treated as a preference
+ * for their combined field dimension.
+ *
+ * Optional method
+ *
+ * @param the desired field dimension of microphone capture. Range is from -1 (wide angle),
+ * though 0 (no zoom) to 1 (maximum zoom).
+ *
+ * @return retval OK if the call is not successful, an error code otherwise.
+ */
+ setMicrophoneFieldDimension(float zoom) generates(Result retval);
+};
diff --git a/audio/7.0/IStreamOut.hal b/audio/7.0/IStreamOut.hal
new file mode 100644
index 0000000..208beb6
--- /dev/null
+++ b/audio/7.0/IStreamOut.hal
@@ -0,0 +1,378 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+import IStream;
+import IStreamOutCallback;
+import IStreamOutEventCallback;
+
+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);
+
+ /**
+ * Set the callback interface for notifying about an output stream event.
+ *
+ * Calling this method with a null pointer will result in releasing
+ * the local callback proxy on the server side and thus dereference
+ * the callback implementation on the client side.
+ *
+ * @return retval operation completion status.
+ */
+ setEventCallback(IStreamOutEventCallback callback)
+ 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);
+
+ /**
+ * Returns the Dual Mono mode presentation setting.
+ *
+ * Optional method
+ *
+ * @return retval operation completion status.
+ * @return mode current setting of Dual Mono mode.
+ */
+ getDualMonoMode() generates (Result retval, DualMonoMode mode);
+
+ /**
+ * Sets the Dual Mono mode presentation on the output device.
+ *
+ * The Dual Mono mode is generally applied to stereo audio streams
+ * where the left and right channels come from separate sources.
+ *
+ * Optional method
+ *
+ * @param mode selected Dual Mono mode.
+ * @return retval operation completion status.
+ */
+ setDualMonoMode(DualMonoMode mode) generates (Result retval);
+
+ /**
+ * Returns the Audio Description Mix level in dB.
+ *
+ * The level is applied to streams incorporating a secondary Audio
+ * Description stream. It specifies the relative level of mixing for
+ * the Audio Description with a reference to the Main Audio.
+ *
+ * Optional method
+ *
+ * The value of the relative level is in the range from negative infinity
+ * to +48.
+ *
+ * @return retval operation completion status.
+ * @return leveldB the current Audio Description Mix Level in dB.
+ */
+ getAudioDescriptionMixLevel() generates (Result retval, float leveldB);
+
+ /**
+ * Sets the Audio Description Mix level in dB.
+ *
+ * For streams incorporating a secondary Audio Description stream
+ * the relative level of mixing of the Audio Description to the Main Audio
+ * is controlled by this method.
+ *
+ * Optional method
+ *
+ * The value of the relative level must be in the range from negative
+ * infinity to +48.
+ *
+ * @param leveldB Audio Description Mix Level in dB
+ * @return retval operation completion status.
+ */
+ setAudioDescriptionMixLevel(float leveldB) generates (Result retval);
+
+ /**
+ * Retrieves current playback rate parameters.
+ *
+ * Optional method
+ *
+ * @return retval operation completion status.
+ * @return playbackRate current playback parameters
+ */
+ getPlaybackRateParameters()
+ generates (Result retval, PlaybackRate playbackRate);
+
+ /**
+ * Sets the playback rate parameters that control playback behavior.
+ * This is normally used when playing encoded content and decoding
+ * is performed in hardware. Otherwise, the framework can apply
+ * necessary transformations.
+ *
+ * Optional method
+ *
+ * If the HAL supports setting the playback rate, it is recommended
+ * to support speed and pitch values at least in the range
+ * from 0.5f to 2.0f, inclusive (see the definition of PlaybackRate struct).
+ *
+ * @param playbackRate playback parameters
+ * @return retval operation completion status.
+ */
+ setPlaybackRateParameters(PlaybackRate playbackRate)
+ generates (Result retval);
+};
diff --git a/audio/7.0/IStreamOutCallback.hal b/audio/7.0/IStreamOutCallback.hal
new file mode 100644
index 0000000..7b9d47f
--- /dev/null
+++ b/audio/7.0/IStreamOutCallback.hal
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+/**
+ * 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/7.0/IStreamOutEventCallback.hal b/audio/7.0/IStreamOutEventCallback.hal
new file mode 100644
index 0000000..52e65d3
--- /dev/null
+++ b/audio/7.0/IStreamOutEventCallback.hal
@@ -0,0 +1,140 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+/**
+ * Asynchronous stream out event callback interface. The interface provides
+ * a way for the HAL to notify platform when there are changes, e.g. codec
+ * format change, from the lower layer.
+ */
+interface IStreamOutEventCallback {
+ /**
+ * Codec format changed.
+ *
+ * onCodecFormatChanged returns an AudioMetadata object in read-only ByteString format.
+ * It represents the most recent codec format decoded by a HW audio decoder.
+ *
+ * Codec format is an optional message from HW audio decoders. It serves to
+ * notify the application about the codec format and audio objects contained
+ * within the compressed audio stream for control, informational,
+ * and display purposes.
+ *
+ * audioMetadata ByteString is convertible to an AudioMetadata object through
+ * both a C++ and a C API present in Metadata.h [1], or through a Java API present
+ * in AudioMetadata.java [2].
+ *
+ * The ByteString format is a stable format used for parcelling (marshalling) across
+ * JNI, AIDL, and HIDL interfaces. The test for R compatibility for native marshalling
+ * is TEST(metadata_tests, compatibility_R) [3]. The test for R compatibility for JNI
+ * marshalling is android.media.cts.AudioMetadataTest#testCompatibilityR [4].
+ *
+ * R (audio HAL 7.0) defined keys are as follows [2]:
+ * "bitrate", int32
+ * "channel-mask", int32
+ * "mime", string
+ * "sample-rate", int32
+ * "bit-width", int32
+ * "has-atmos", int32
+ * "audio-encoding", int32
+ *
+ * Parceling Format:
+ * All values are native endian order. [1]
+ *
+ * using type_size_t = uint32_t;
+ * using index_size_t = uint32_t;
+ * using datum_size_t = uint32_t;
+ *
+ * Permitted type indexes are
+ * TYPE_NONE = 0, // Reserved
+ * TYPE_INT32 = 1,
+ * TYPE_INT64 = 2,
+ * TYPE_FLOAT = 3,
+ * TYPE_DOUBLE = 4,
+ * TYPE_STRING = 5,
+ * TYPE_DATA = 6, // A data table of <String, Datum>
+ *
+ * Datum = {
+ * (type_size_t) Type (the type index from type_as_value<T>.)
+ * (datum_size_t) Size (size of the Payload)
+ * (byte string) Payload<Type>
+ * }
+ *
+ * The data is specified in native endian order.
+ * Since the size of the Payload is always present, unknown types may be skipped.
+ *
+ * Payload<Fixed-size Primitive_Value>
+ * [ sizeof(Primitive_Value) in raw bytes ]
+ *
+ * Example of Payload<Int32> of 123:
+ * Payload<Int32>
+ * [ value of 123 ] = 0x7b 0x00 0x00 0x00 123
+ *
+ * Payload<String>
+ * [ (index_size_t) length, not including zero terminator.]
+ * [ (length) raw bytes ]
+ *
+ * Example of Payload<String> of std::string("hi"):
+ * [ (index_size_t) length ] = 0x02 0x00 0x00 0x00 2 strlen("hi")
+ * [ raw bytes "hi" ] = 0x68 0x69 "hi"
+ *
+ * Payload<Data>
+ * [ (index_size_t) entries ]
+ * [ raw bytes (entry 1) Key (Payload<String>)
+ * Value (Datum)
+ * ... (until #entries) ]
+ *
+ * Example of Payload<Data> of {{"hello", "world"},
+ * {"value", (int32_t)1000}};
+ * [ (index_size_t) #entries ] = 0x02 0x00 0x00 0x00 2 entries
+ * Key (Payload<String>)
+ * [ index_size_t length ] = 0x05 0x00 0x00 0x00 5 strlen("hello")
+ * [ raw bytes "hello" ] = 0x68 0x65 0x6c 0x6c 0x6f "hello"
+ * Value (Datum)
+ * [ (type_size_t) type ] = 0x05 0x00 0x00 0x00 5 (TYPE_STRING)
+ * [ (datum_size_t) size ] = 0x09 0x00 0x00 0x00 sizeof(index_size_t) +
+ * strlen("world")
+ * Payload<String>
+ * [ (index_size_t) length ] = 0x05 0x00 0x00 0x00 5 strlen("world")
+ * [ raw bytes "world" ] = 0x77 0x6f 0x72 0x6c 0x64 "world"
+ * Key (Payload<String>)
+ * [ index_size_t length ] = 0x05 0x00 0x00 0x00 5 strlen("value")
+ * [ raw bytes "value" ] = 0x76 0x61 0x6c 0x75 0x65 "value"
+ * Value (Datum)
+ * [ (type_size_t) type ] = 0x01 0x00 0x00 0x00 1 (TYPE_INT32)
+ * [ (datum_size_t) size ] = 0x04 0x00 0x00 0x00 4 sizeof(int32_t)
+ * Payload<Int32>
+ * [ raw bytes 1000 ] = 0xe8 0x03 0x00 0x00 1000
+ *
+ * The contents of audioMetadata is a Payload<Data>.
+ * An implementation dependent detail is that the Keys are always
+ * stored sorted, so the byte string representation generated is unique.
+ *
+ * Vendor keys are allowed for informational and debugging purposes.
+ * Vendor keys should consist of the vendor company name followed
+ * by a dot; for example, "vendorCompany.someVolume" [2].
+ *
+ * [1] system/media/audio_utils/include/audio_utils/Metadata.h
+ * [2] frameworks/base/media/java/android/media/AudioMetadata.java
+ * [3] system/media/audio_utils/tests/metadata_tests.cpp
+ * [4] cts/tests/tests/media/src/android/media/cts/AudioMetadataTest.java
+ *
+ * @param audioMetadata is a buffer containing decoded format changes
+ * reported by codec. The buffer contains data that can be transformed
+ * to audio metadata, which is a C++ object based map.
+ */
+ oneway onCodecFormatChanged(vec<uint8_t> audioMetadata);
+};
diff --git a/audio/7.0/config/Android.bp b/audio/7.0/config/Android.bp
new file mode 100644
index 0000000..015c424
--- /dev/null
+++ b/audio/7.0/config/Android.bp
@@ -0,0 +1,5 @@
+xsd_config {
+ name: "audio_policy_configuration_V7_0",
+ srcs: ["audio_policy_configuration.xsd"],
+ package_name: "audio.policy.configuration.V7_0",
+}
diff --git a/audio/7.0/config/api/current.txt b/audio/7.0/config/api/current.txt
new file mode 100644
index 0000000..98c5eac
--- /dev/null
+++ b/audio/7.0/config/api/current.txt
@@ -0,0 +1,435 @@
+// Signature format: 2.0
+package audio.policy.configuration.V7_0 {
+
+ public class AttachedDevices {
+ ctor public AttachedDevices();
+ method public java.util.List<java.lang.String> getItem();
+ }
+
+ public enum AudioDevice {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_AMBIENT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_AUX_DIGITAL;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_BACK_MIC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_BLUETOOTH_BLE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_BUILTIN_MIC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_BUS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_COMMUNICATION;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_DEFAULT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_ECHO_REFERENCE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_FM_TUNER;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_HDMI;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_HDMI_ARC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_IP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_LINE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_LOOPBACK;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_PROXY;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_SPDIF;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_STUB;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_TELEPHONY_RX;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_TV_TUNER;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_USB_ACCESSORY;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_USB_DEVICE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_USB_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_VOICE_CALL;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_WIRED_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_NONE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_AUX_LINE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_BUS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_DEFAULT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_EARPIECE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_ECHO_CANCELLER;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_FM;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_HDMI;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_HDMI_ARC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_HEARING_AID;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_IP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_LINE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_PROXY;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_SPDIF;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_SPEAKER;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_STUB;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_TELEPHONY_TX;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_USB_DEVICE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_USB_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+
+ public enum AudioFormat {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADIF;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_ELD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_ERLC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_HE_V1;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_HE_V2;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_LC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_LD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_LTP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_MAIN;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_SCALABLE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_SSR;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ADTS_XHE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ELD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_ERLC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_HE_V1;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_HE_V2;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LATM;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LATM_HE_V1;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LATM_HE_V2;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LATM_LC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_LTP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_MAIN;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_SCALABLE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_SSR;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AAC_XHE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AC3;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AC4;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_ALAC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AMR_NB;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AMR_WB;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_AMR_WB_PLUS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APTX;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APTX_ADAPTIVE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APTX_HD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APTX_TWSP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_CELT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DOLBY_TRUEHD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DSD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DTS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DTS_HD;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_EVRC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_EVRCB;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_EVRCNW;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_EVRCWB;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_E_AC3;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_E_AC3_JOC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_FLAC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_HE_AAC_V1;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_HE_AAC_V2;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_IEC61937;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_LDAC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_LHDC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_LHDC_LL;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_MAT_1_0;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_MAT_2_0;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_MAT_2_1;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_MP2;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_MP3;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_OPUS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_PCM_16_BIT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_PCM_32_BIT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_PCM_8_24_BIT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_PCM_8_BIT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_PCM_FLOAT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_QCELP;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_SBC;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_VORBIS;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_WMA;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_WMA_PRO;
+ }
+
+ public class AudioPolicyConfiguration {
+ ctor public AudioPolicyConfiguration();
+ method public audio.policy.configuration.V7_0.GlobalConfiguration getGlobalConfiguration();
+ method public java.util.List<audio.policy.configuration.V7_0.Modules> getModules();
+ method public audio.policy.configuration.V7_0.SurroundSound getSurroundSound();
+ method public audio.policy.configuration.V7_0.Version getVersion();
+ method public java.util.List<audio.policy.configuration.V7_0.Volumes> getVolumes();
+ method public void setGlobalConfiguration(audio.policy.configuration.V7_0.GlobalConfiguration);
+ method public void setSurroundSound(audio.policy.configuration.V7_0.SurroundSound);
+ method public void setVersion(audio.policy.configuration.V7_0.Version);
+ }
+
+ public enum AudioUsage {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ALARM;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANT;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_GAME;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_MEDIA;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_NOTIFICATION;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_UNKNOWN;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VIRTUAL_SOURCE;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VOICE_COMMUNICATION;
+ enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+ }
+
+ public enum DeviceCategory {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.DeviceCategory DEVICE_CATEGORY_EARPIECE;
+ enum_constant public static final audio.policy.configuration.V7_0.DeviceCategory DEVICE_CATEGORY_EXT_MEDIA;
+ enum_constant public static final audio.policy.configuration.V7_0.DeviceCategory DEVICE_CATEGORY_HEADSET;
+ enum_constant public static final audio.policy.configuration.V7_0.DeviceCategory DEVICE_CATEGORY_HEARING_AID;
+ enum_constant public static final audio.policy.configuration.V7_0.DeviceCategory DEVICE_CATEGORY_SPEAKER;
+ }
+
+ public class DevicePorts {
+ ctor public DevicePorts();
+ method public java.util.List<audio.policy.configuration.V7_0.DevicePorts.DevicePort> getDevicePort();
+ }
+
+ public static class DevicePorts.DevicePort {
+ ctor public DevicePorts.DevicePort();
+ method public String getAddress();
+ method public java.util.List<audio.policy.configuration.V7_0.AudioFormat> getEncodedFormats();
+ method public audio.policy.configuration.V7_0.Gains getGains();
+ method public java.util.List<audio.policy.configuration.V7_0.Profile> getProfile();
+ method public audio.policy.configuration.V7_0.Role getRole();
+ method public String getTagName();
+ method public String getType();
+ method public boolean get_default();
+ method public void setAddress(String);
+ method public void setEncodedFormats(java.util.List<audio.policy.configuration.V7_0.AudioFormat>);
+ method public void setGains(audio.policy.configuration.V7_0.Gains);
+ method public void setRole(audio.policy.configuration.V7_0.Role);
+ method public void setTagName(String);
+ method public void setType(String);
+ method public void set_default(boolean);
+ }
+
+ public enum EngineSuffix {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.EngineSuffix _default;
+ enum_constant public static final audio.policy.configuration.V7_0.EngineSuffix configurable;
+ }
+
+ public enum GainMode {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.GainMode AUDIO_GAIN_MODE_CHANNELS;
+ enum_constant public static final audio.policy.configuration.V7_0.GainMode AUDIO_GAIN_MODE_JOINT;
+ enum_constant public static final audio.policy.configuration.V7_0.GainMode AUDIO_GAIN_MODE_RAMP;
+ }
+
+ public class Gains {
+ ctor public Gains();
+ method public java.util.List<audio.policy.configuration.V7_0.Gains.Gain> getGain();
+ }
+
+ public static class Gains.Gain {
+ ctor public Gains.Gain();
+ method public String getChannel_mask();
+ method public int getDefaultValueMB();
+ method public int getMaxRampMs();
+ method public int getMaxValueMB();
+ method public int getMinRampMs();
+ method public int getMinValueMB();
+ method public audio.policy.configuration.V7_0.GainMode getMode();
+ method public String getName();
+ method public int getStepValueMB();
+ method public boolean getUseForVolume();
+ method public void setChannel_mask(String);
+ method public void setDefaultValueMB(int);
+ method public void setMaxRampMs(int);
+ method public void setMaxValueMB(int);
+ method public void setMinRampMs(int);
+ method public void setMinValueMB(int);
+ method public void setMode(audio.policy.configuration.V7_0.GainMode);
+ method public void setName(String);
+ method public void setStepValueMB(int);
+ method public void setUseForVolume(boolean);
+ }
+
+ public class GlobalConfiguration {
+ ctor public GlobalConfiguration();
+ method public boolean getCall_screen_mode_supported();
+ method public audio.policy.configuration.V7_0.EngineSuffix getEngine_library();
+ method public boolean getSpeaker_drc_enabled();
+ method public void setCall_screen_mode_supported(boolean);
+ method public void setEngine_library(audio.policy.configuration.V7_0.EngineSuffix);
+ method public void setSpeaker_drc_enabled(boolean);
+ }
+
+ public enum HalVersion {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.HalVersion _2_0;
+ enum_constant public static final audio.policy.configuration.V7_0.HalVersion _3_0;
+ }
+
+ public class MixPorts {
+ ctor public MixPorts();
+ method public java.util.List<audio.policy.configuration.V7_0.MixPorts.MixPort> getMixPort();
+ }
+
+ public static class MixPorts.MixPort {
+ ctor public MixPorts.MixPort();
+ method public String getFlags();
+ method public audio.policy.configuration.V7_0.Gains getGains();
+ method public long getMaxActiveCount();
+ method public long getMaxOpenCount();
+ method public String getName();
+ method public java.util.List<audio.policy.configuration.V7_0.AudioUsage> getPreferredUsage();
+ method public java.util.List<audio.policy.configuration.V7_0.Profile> getProfile();
+ method public audio.policy.configuration.V7_0.Role getRole();
+ method public void setFlags(String);
+ method public void setGains(audio.policy.configuration.V7_0.Gains);
+ method public void setMaxActiveCount(long);
+ method public void setMaxOpenCount(long);
+ method public void setName(String);
+ method public void setPreferredUsage(java.util.List<audio.policy.configuration.V7_0.AudioUsage>);
+ method public void setRole(audio.policy.configuration.V7_0.Role);
+ }
+
+ public enum MixType {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.MixType mix;
+ enum_constant public static final audio.policy.configuration.V7_0.MixType mux;
+ }
+
+ public class Modules {
+ ctor public Modules();
+ method public java.util.List<audio.policy.configuration.V7_0.Modules.Module> getModule();
+ }
+
+ public static class Modules.Module {
+ ctor public Modules.Module();
+ method public audio.policy.configuration.V7_0.AttachedDevices getAttachedDevices();
+ method public String getDefaultOutputDevice();
+ method public audio.policy.configuration.V7_0.DevicePorts getDevicePorts();
+ method public audio.policy.configuration.V7_0.HalVersion getHalVersion();
+ method public audio.policy.configuration.V7_0.MixPorts getMixPorts();
+ method public String getName();
+ method public audio.policy.configuration.V7_0.Routes getRoutes();
+ method public void setAttachedDevices(audio.policy.configuration.V7_0.AttachedDevices);
+ method public void setDefaultOutputDevice(String);
+ method public void setDevicePorts(audio.policy.configuration.V7_0.DevicePorts);
+ method public void setHalVersion(audio.policy.configuration.V7_0.HalVersion);
+ method public void setMixPorts(audio.policy.configuration.V7_0.MixPorts);
+ method public void setName(String);
+ method public void setRoutes(audio.policy.configuration.V7_0.Routes);
+ }
+
+ public class Profile {
+ ctor public Profile();
+ method public String getChannelMasks();
+ method public String getFormat();
+ method public String getName();
+ method public String getSamplingRates();
+ method public void setChannelMasks(String);
+ method public void setFormat(String);
+ method public void setName(String);
+ method public void setSamplingRates(String);
+ }
+
+ public class Reference {
+ ctor public Reference();
+ method public String getName();
+ method public java.util.List<java.lang.String> getPoint();
+ method public void setName(String);
+ }
+
+ public enum Role {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.Role sink;
+ enum_constant public static final audio.policy.configuration.V7_0.Role source;
+ }
+
+ public class Routes {
+ ctor public Routes();
+ method public java.util.List<audio.policy.configuration.V7_0.Routes.Route> getRoute();
+ }
+
+ public static class Routes.Route {
+ ctor public Routes.Route();
+ method public String getSink();
+ method public String getSources();
+ method public audio.policy.configuration.V7_0.MixType getType();
+ method public void setSink(String);
+ method public void setSources(String);
+ method public void setType(audio.policy.configuration.V7_0.MixType);
+ }
+
+ public enum Stream {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ACCESSIBILITY;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ALARM;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ASSISTANT;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_BLUETOOTH_SCO;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_DTMF;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ENFORCED_AUDIBLE;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_MUSIC;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_NOTIFICATION;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_PATCH;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_REROUTING;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_RING;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_SYSTEM;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_TTS;
+ enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_VOICE_CALL;
+ }
+
+ public class SurroundFormats {
+ ctor public SurroundFormats();
+ method public java.util.List<audio.policy.configuration.V7_0.SurroundFormats.Format> getFormat();
+ }
+
+ public static class SurroundFormats.Format {
+ ctor public SurroundFormats.Format();
+ method public audio.policy.configuration.V7_0.AudioFormat getName();
+ method public java.util.List<audio.policy.configuration.V7_0.AudioFormat> getSubformats();
+ method public void setName(audio.policy.configuration.V7_0.AudioFormat);
+ method public void setSubformats(java.util.List<audio.policy.configuration.V7_0.AudioFormat>);
+ }
+
+ public class SurroundSound {
+ ctor public SurroundSound();
+ method public audio.policy.configuration.V7_0.SurroundFormats getFormats();
+ method public void setFormats(audio.policy.configuration.V7_0.SurroundFormats);
+ }
+
+ public enum Version {
+ method public String getRawName();
+ enum_constant public static final audio.policy.configuration.V7_0.Version _1_0;
+ }
+
+ public class Volume {
+ ctor public Volume();
+ method public audio.policy.configuration.V7_0.DeviceCategory getDeviceCategory();
+ method public java.util.List<java.lang.String> getPoint();
+ method public String getRef();
+ method public audio.policy.configuration.V7_0.Stream getStream();
+ method public void setDeviceCategory(audio.policy.configuration.V7_0.DeviceCategory);
+ method public void setRef(String);
+ method public void setStream(audio.policy.configuration.V7_0.Stream);
+ }
+
+ public class Volumes {
+ ctor public Volumes();
+ method public java.util.List<audio.policy.configuration.V7_0.Reference> getReference();
+ method public java.util.List<audio.policy.configuration.V7_0.Volume> getVolume();
+ }
+
+ public class XmlParser {
+ ctor public XmlParser();
+ method public static audio.policy.configuration.V7_0.AudioPolicyConfiguration read(java.io.InputStream) throws javax.xml.datatype.DatatypeConfigurationException, java.io.IOException, org.xmlpull.v1.XmlPullParserException;
+ method public static String readText(org.xmlpull.v1.XmlPullParser) throws java.io.IOException, org.xmlpull.v1.XmlPullParserException;
+ method public static void skip(org.xmlpull.v1.XmlPullParser) throws java.io.IOException, org.xmlpull.v1.XmlPullParserException;
+ }
+
+}
+
diff --git a/audio/7.0/config/api/last_current.txt b/audio/7.0/config/api/last_current.txt
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/audio/7.0/config/api/last_current.txt
diff --git a/audio/7.0/config/api/last_removed.txt b/audio/7.0/config/api/last_removed.txt
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/audio/7.0/config/api/last_removed.txt
diff --git a/audio/7.0/config/api/removed.txt b/audio/7.0/config/api/removed.txt
new file mode 100644
index 0000000..d802177
--- /dev/null
+++ b/audio/7.0/config/api/removed.txt
@@ -0,0 +1 @@
+// Signature format: 2.0
diff --git a/audio/7.0/config/audio_policy_configuration.xsd b/audio/7.0/config/audio_policy_configuration.xsd
new file mode 100644
index 0000000..19c6f70
--- /dev/null
+++ b/audio/7.0/config/audio_policy_configuration.xsd
@@ -0,0 +1,634 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<!-- 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" minOccurs="0" />
+ </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:attribute name="call_screen_mode_supported" type="xs:boolean" use="optional"/>
+ <xs:attribute name="engine_library" type="engineSuffix" use="optional"/>
+ </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_HDMI_ARC"/>
+ <xs:enumeration value="AUDIO_DEVICE_IN_ECHO_REFERENCE"/>
+ <xs:enumeration value="AUDIO_DEVICE_IN_DEFAULT"/>
+ <xs:enumeration value="AUDIO_DEVICE_IN_STUB"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="vendorExtension">
+ <!-- Vendor extension names must be prefixed by "VX_" to distinguish them from AOSP values.
+ Vendor are encouraged to namespace their module names to avoid conflicts.
+ Example for an hypothetical Google virtual reality device:
+ <devicePort tagName="VR" type="VX_GOOGLE_VR" role="sink">
+ -->
+ <xs:restriction base="xs:string">
+ <xs:pattern value="VX_[_a-zA-Z0-9]+"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="extendableAudioDevice">
+ <xs:union memberTypes="audioDevice vendorExtension"/>
+ </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:enumeration value="AUDIO_FORMAT_AAC_LATM"/>
+ <xs:enumeration value="AUDIO_FORMAT_AAC_LATM_LC"/>
+ <xs:enumeration value="AUDIO_FORMAT_AAC_LATM_HE_V1"/>
+ <xs:enumeration value="AUDIO_FORMAT_AAC_LATM_HE_V2"/>
+ <xs:enumeration value="AUDIO_FORMAT_CELT"/>
+ <xs:enumeration value="AUDIO_FORMAT_APTX_ADAPTIVE"/>
+ <xs:enumeration value="AUDIO_FORMAT_LHDC"/>
+ <xs:enumeration value="AUDIO_FORMAT_LHDC_LL"/>
+ <xs:enumeration value="AUDIO_FORMAT_APTX_TWSP"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="extendableAudioFormat">
+ <xs:union memberTypes="audioFormat vendorExtension"/>
+ </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="extendableAudioFormat" 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:attribute name="useForVolume" type="xs:boolean" 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="extendableAudioDevice" 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:attribute name="encodedFormats" type="audioFormatsList" use="optional"
+ default="" />
+ </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_ASSISTANT"/>
+ <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="audioFormatsList">
+ <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="audioFormatsList" />
+ </xs:complexType>
+ </xs:element>
+ </xs:sequence>
+ </xs:complexType>
+ <xs:simpleType name="engineSuffix">
+ <xs:restriction base="xs:string">
+ <xs:enumeration value="default"/>
+ <xs:enumeration value="configurable"/>
+ </xs:restriction>
+ </xs:simpleType>
+</xs:schema>
diff --git a/audio/7.0/types.hal b/audio/7.0/types.hal
new file mode 100644
index 0000000..b0b0843
--- /dev/null
+++ b/audio/7.0/types.hal
@@ -0,0 +1,357 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio@7.0;
+
+import android.hardware.audio.common@7.0;
+
+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
+};
+
+struct ParameterValue {
+ string key;
+ string value;
+};
+
+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
+};
+
+/*
+ * 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;
+};
+
+/**
+ * Constants used by the HAL to determine how to select microphones and process those inputs in
+ * order to optimize for capture in the specified direction.
+ *
+ * MicrophoneDirection Constants are defined in MicrophoneDirection.java.
+ */
+@export(name="audio_microphone_direction_t", value_prefix="MIC_DIRECTION_")
+enum MicrophoneDirection : int32_t {
+ /**
+ * Don't do any directionality processing of the activated microphone(s).
+ */
+ UNSPECIFIED = 0,
+ /**
+ * Optimize capture for audio coming from the screen-side of the device.
+ */
+ FRONT = 1,
+ /**
+ * Optimize capture for audio coming from the side of the device opposite the screen.
+ */
+ BACK = 2,
+ /**
+ * Optimize capture for audio coming from an off-device microphone.
+ */
+ EXTERNAL = 3,
+};
+
+
+/* Dual Mono handling is used when a stereo audio stream
+ * contains separate audio content on the left and right channels.
+ * Such information about the content of the stream may be found, for example,
+ * in ITU T-REC-J.94-201610 A.6.2.3 Component descriptor.
+ */
+@export(name="audio_dual_mono_mode_t", value_prefix="AUDIO_DUAL_MONO_MODE_")
+enum DualMonoMode : int32_t {
+ // Need to be in sync with DUAL_MONO_MODE* constants in
+ // frameworks/base/media/java/android/media/AudioTrack.java
+ /**
+ * Disable any Dual Mono presentation effect.
+ *
+ */
+ OFF = 0,
+ /**
+ * This mode indicates that a stereo stream should be presented
+ * with the left and right audio channels blended together
+ * and delivered to both channels.
+ *
+ * Behavior for non-stereo streams is implementation defined.
+ * A suggested guideline is that the left-right stereo symmetric
+ * channels are pairwise blended, the other channels such as center
+ * are left alone.
+ */
+ LR = 1,
+ /**
+ * This mode indicates that a stereo stream should be presented
+ * with the left audio channel replicated into the right audio channel.
+ *
+ * Behavior for non-stereo streams is implementation defined.
+ * A suggested guideline is that all channels with left-right
+ * stereo symmetry will have the left channel position replicated
+ * into the right channel position. The center channels (with no
+ * left/right symmetry) or unbalanced channels are left alone.
+ */
+ LL = 2,
+ /**
+ * This mode indicates that a stereo stream should be presented
+ * with the right audio channel replicated into the left audio channel.
+ *
+ * Behavior for non-stereo streams is implementation defined.
+ * A suggested guideline is that all channels with left-right
+ * stereo symmetry will have the right channel position replicated
+ * into the left channel position. The center channels (with no
+ * left/right symmetry) or unbalanced channels are left alone.
+ */
+ RR = 3,
+};
+
+/**
+ * Algorithms used for timestretching (preserving pitch while playing audio
+ * content at different speed).
+ */
+@export(name="audio_timestretch_stretch_mode_t", value_prefix="AUDIO_TIMESTRETCH_STRETCH_")
+enum TimestretchMode : int32_t {
+ // Need to be in sync with AUDIO_STRETCH_MODE_* constants in
+ // frameworks/base/media/java/android/media/PlaybackParams.java
+ DEFAULT = 0,
+ /** Selects timestretch algorithm best suitable for voice (speech) content. */
+ VOICE = 1,
+};
+
+/**
+ * Behavior when the values for speed and / or pitch are out
+ * of applicable range.
+ */
+@export(name="audio_timestretch_fallback_mode_t", value_prefix="AUDIO_TIMESTRETCH_FALLBACK_")
+enum TimestretchFallbackMode : int32_t {
+ // Need to be in sync with AUDIO_FALLBACK_MODE_* constants in
+ // frameworks/base/media/java/android/media/PlaybackParams.java
+ /** Play silence for parameter values that are out of range. */
+ MUTE = 1,
+ /** Return an error while trying to set the parameters. */
+ FAIL = 2,
+};
+
+/**
+ * Parameters determining playback behavior. They are used to speed up or
+ * slow down playback and / or change the tonal frequency of the audio content
+ * (pitch).
+ */
+struct PlaybackRate {
+ /**
+ * Speed factor (multiplier). Normal speed has the value of 1.0f.
+ * Values less than 1.0f slow down playback, value greater than 1.0f
+ * speed it up.
+ */
+ float speed;
+ /**
+ * Pitch factor (multiplier). Setting pitch value to 1.0f together
+ * with changing playback speed preserves the pitch, this is often
+ * called "timestretching." Setting the pitch value equal to speed produces
+ * the same effect as playing audio content at different sampling rate.
+ */
+ float pitch;
+ /**
+ * Selects the algorithm used for timestretching (preserving pitch while
+ * playing audio at different speed).
+ */
+ TimestretchMode timestretchMode;
+ /**
+ * Selects the behavior when the specified values for speed and / or pitch
+ * are out of applicable range.
+ */
+ TimestretchFallbackMode fallbackMode;
+};
diff --git a/audio/common/7.0/Android.bp b/audio/common/7.0/Android.bp
new file mode 100644
index 0000000..eef563f
--- /dev/null
+++ b/audio/common/7.0/Android.bp
@@ -0,0 +1,14 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+ name: "android.hardware.audio.common@7.0",
+ root: "android.hardware",
+ srcs: [
+ "types.hal",
+ ],
+ interfaces: [
+ "android.hidl.safe_union@1.0",
+ ],
+ gen_java: true,
+ gen_java_constants: true,
+}
diff --git a/audio/common/7.0/types.hal b/audio/common/7.0/types.hal
new file mode 100644
index 0000000..2288eb1
--- /dev/null
+++ b/audio/common/7.0/types.hal
@@ -0,0 +1,1191 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.common@7.0;
+
+import android.hidl.safe_union@1.0;
+
+/*
+ *
+ * IDs and Handles
+ *
+ */
+
+/**
+ * Handle type for identifying audio sources and sinks.
+ */
+typedef int32_t AudioIoHandle;
+
+/**
+ * Audio hw module handle functions or structures referencing a module.
+ */
+typedef int32_t AudioModuleHandle;
+
+/**
+ * Each port has a unique ID or handle allocated by policy manager.
+ */
+typedef int32_t AudioPortHandle;
+
+/**
+ * Each patch is identified by a handle at the interface used to create that
+ * patch. For instance, when a patch is created by the audio HAL, the HAL
+ * allocates and returns a handle. This handle is unique to a given audio HAL
+ * hardware module. But the same patch receives another system wide unique
+ * handle allocated by the framework. This unique handle is used for all
+ * transactions inside the framework.
+ */
+typedef int32_t AudioPatchHandle;
+
+/**
+ * A HW synchronization source returned by the audio HAL.
+ */
+typedef uint32_t AudioHwSync;
+
+/**
+ * Each port has a unique ID or handle allocated by policy manager.
+ */
+@export(name="")
+enum AudioHandleConsts : int32_t {
+ AUDIO_IO_HANDLE_NONE = 0,
+ AUDIO_MODULE_HANDLE_NONE = 0,
+ AUDIO_PORT_HANDLE_NONE = 0,
+ AUDIO_PATCH_HANDLE_NONE = 0,
+};
+
+/**
+ * Commonly used structure for passing unique identifieds (UUID).
+ * For the definition of UUID, refer to ITU-T X.667 spec.
+ */
+struct Uuid {
+ uint32_t timeLow;
+ uint16_t timeMid;
+ uint16_t versionAndTimeHigh;
+ uint16_t variantAndClockSeqHigh;
+ uint8_t[6] node;
+};
+
+
+/*
+ *
+ * Audio streams
+ *
+ */
+
+/**
+ * Audio stream type describing the intended use case of a stream.
+ */
+@export(name="audio_stream_type_t", value_prefix="AUDIO_STREAM_")
+enum AudioStreamType : int32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioSystem.java
+ /** Used to identify the default audio stream volume. */
+ DEFAULT = -1,
+ /** Specifies the minimum value for use in checks and loops. */
+ MIN = 0,
+ /** Used to identify the volume of audio streams for phone calls. */
+ VOICE_CALL = 0,
+ /** Used to identify the volume of audio streams for system sounds. */
+ SYSTEM = 1,
+ /**
+ * Used to identify the volume of audio streams for the phone ring
+ * and message alerts.
+ */
+ RING = 2,
+ /** Used to identify the volume of audio streams for music playback. */
+ MUSIC = 3,
+ /** Used to identify the volume of audio streams for alarms. */
+ ALARM = 4,
+ /** Used to identify the volume of audio streams for notifications. */
+ NOTIFICATION = 5,
+ /**
+ * Used to identify the volume of audio streams for phone calls
+ * when connected on bluetooth.
+ */
+ BLUETOOTH_SCO = 6,
+ /**
+ * Used to identify the volume of audio streams for enforced system
+ * sounds in certain countries (e.g camera in Japan). */
+ ENFORCED_AUDIBLE = 7,
+ /** Used to identify the volume of audio streams for DTMF tones. */
+ DTMF = 8,
+ /**
+ * Used to identify the volume of audio streams exclusively transmitted
+ * through the speaker (TTS) of the device.
+ */
+ TTS = 9,
+ /**
+ * Used to identify the volume of audio streams for accessibility prompts.
+ */
+ ACCESSIBILITY = 10,
+ /** Used to identify the volume of audio streams for virtual assistant. */
+ ASSISTANT = 11,
+};
+
+@export(name="audio_source_t", value_prefix="AUDIO_SOURCE_")
+enum AudioSource : int32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/MediaRecorder.java,
+ // system/media/audio_effects/include/audio_effects/audio_effects_conf.h
+ /** Default audio source. */
+ DEFAULT = 0,
+ /** Microphone audio source. */
+ MIC = 1,
+ /** Voice call uplink (Tx) audio source. */
+ VOICE_UPLINK = 2,
+ /** Voice call downlink (Rx) audio source. */
+ VOICE_DOWNLINK = 3,
+ /** Voice call uplink + downlink audio source. */
+ VOICE_CALL = 4,
+ /**
+ * Microphone audio source tuned for video recording, with the same
+ * orientation as the camera if available.
+ */
+ CAMCORDER = 5,
+ /** Microphone audio source tuned for voice recognition. */
+ VOICE_RECOGNITION = 6,
+ /**
+ * Microphone audio source tuned for voice communications such as VoIP. It
+ * will for instance take advantage of echo cancellation or automatic gain
+ * control if available.
+ */
+ VOICE_COMMUNICATION = 7,
+ /**
+ * Source for the mix to be presented remotely. An example of remote
+ * presentation is Wifi Display where a dongle attached to a TV can be used
+ * to play the mix captured by this audio source.
+ */
+ REMOTE_SUBMIX = 8,
+ /**
+ * Source for unprocessed sound. Usage examples include level measurement
+ * and raw signal analysis.
+ */
+ UNPROCESSED = 9,
+ /**
+ * Source for capturing audio meant to be processed in real time and played back for live
+ * performance (e.g karaoke). The capture path will minimize latency and coupling with
+ * playback path.
+ */
+ VOICE_PERFORMANCE = 10,
+ /**
+ * Source for an echo canceller to capture the reference signal to be cancelled.
+ * The echo reference signal will be captured as close as possible to the DAC in order
+ * to include all post processing applied to the playback path.
+ */
+ ECHO_REFERENCE = 1997,
+ /** Virtual source for the built-in FM tuner. */
+ FM_TUNER = 1998,
+ /** Virtual source for the last captured hotword. */
+ HOTWORD = 1999,
+};
+
+typedef int32_t AudioSession;
+/**
+ * Special audio session values.
+ */
+@export(name="audio_session_t", value_prefix="AUDIO_SESSION_")
+enum AudioSessionConsts : int32_t {
+ /**
+ * Session for effects attached to a particular sink or source audio device
+ * (e.g an effect only applied to a speaker)
+ */
+ DEVICE = -2,
+ /**
+ * Session for effects attached to a particular output stream
+ * (value must be less than 0)
+ */
+ OUTPUT_STAGE = -1,
+ /**
+ * Session for effects applied to output mix. These effects can
+ * be moved by audio policy manager to another output stream
+ * (value must be 0)
+ */
+ OUTPUT_MIX = 0,
+ /**
+ * Application does not specify an explicit session ID to be used, and
+ * requests a new session ID to be allocated. Corresponds to
+ * AudioManager.AUDIO_SESSION_ID_GENERATE and
+ * AudioSystem.AUDIO_SESSION_ALLOCATE.
+ */
+ ALLOCATE = 0,
+ /**
+ * For use with AudioRecord::start(), this indicates no trigger session.
+ * It is also used with output tracks and patch tracks, which never have a
+ * session.
+ */
+ NONE = 0
+};
+
+/**
+ * Audio format is a 32-bit word that consists of:
+ * main format field (upper 8 bits)
+ * sub format field (lower 24 bits).
+ *
+ * The main format indicates the main codec type. The sub format field indicates
+ * options and parameters for each format. The sub format is mainly used for
+ * record to indicate for instance the requested bitrate or profile. It can
+ * also be used for certain formats to give informations not present in the
+ * encoded audio stream (e.g. octet alignement for AMR).
+ */
+@export(name="audio_format_t", value_prefix="AUDIO_FORMAT_")
+enum AudioFormat : uint32_t {
+ INVALID = 0xFFFFFFFFUL,
+ DEFAULT = 0,
+ PCM = 0x00000000UL,
+ MP3 = 0x01000000UL,
+ AMR_NB = 0x02000000UL,
+ AMR_WB = 0x03000000UL,
+ AAC = 0x04000000UL,
+ /** Deprecated, Use AAC_HE_V1 */
+ HE_AAC_V1 = 0x05000000UL,
+ /** Deprecated, Use AAC_HE_V2 */
+ HE_AAC_V2 = 0x06000000UL,
+ VORBIS = 0x07000000UL,
+ OPUS = 0x08000000UL,
+ AC3 = 0x09000000UL,
+ E_AC3 = 0x0A000000UL,
+ DTS = 0x0B000000UL,
+ DTS_HD = 0x0C000000UL,
+ /** IEC61937 is encoded audio wrapped in 16-bit PCM. */
+ IEC61937 = 0x0D000000UL,
+ DOLBY_TRUEHD = 0x0E000000UL,
+ EVRC = 0x10000000UL,
+ EVRCB = 0x11000000UL,
+ EVRCWB = 0x12000000UL,
+ EVRCNW = 0x13000000UL,
+ AAC_ADIF = 0x14000000UL,
+ WMA = 0x15000000UL,
+ WMA_PRO = 0x16000000UL,
+ AMR_WB_PLUS = 0x17000000UL,
+ MP2 = 0x18000000UL,
+ QCELP = 0x19000000UL,
+ DSD = 0x1A000000UL,
+ FLAC = 0x1B000000UL,
+ ALAC = 0x1C000000UL,
+ APE = 0x1D000000UL,
+ AAC_ADTS = 0x1E000000UL,
+ SBC = 0x1F000000UL,
+ APTX = 0x20000000UL,
+ APTX_HD = 0x21000000UL,
+ AC4 = 0x22000000UL,
+ LDAC = 0x23000000UL,
+ /** Dolby Metadata-enhanced Audio Transmission */
+ MAT = 0x24000000UL,
+ AAC_LATM = 0x25000000UL,
+ CELT = 0x26000000UL,
+ APTX_ADAPTIVE = 0x27000000UL,
+ LHDC = 0x28000000UL,
+ LHDC_LL = 0x29000000UL,
+ APTX_TWSP = 0x2A000000UL,
+
+ /** Deprecated */
+ MAIN_MASK = 0xFF000000UL,
+ SUB_MASK = 0x00FFFFFFUL,
+
+ /* Subformats */
+ PCM_SUB_16_BIT = 0x1, // PCM signed 16 bits
+ PCM_SUB_8_BIT = 0x2, // PCM unsigned 8 bits
+ PCM_SUB_32_BIT = 0x3, // PCM signed .31 fixed point
+ PCM_SUB_8_24_BIT = 0x4, // PCM signed 8.23 fixed point
+ PCM_SUB_FLOAT = 0x5, // PCM single-precision float pt
+ PCM_SUB_24_BIT_PACKED = 0x6, // PCM signed .23 fix pt (3 bytes)
+
+ MP3_SUB_NONE = 0x0,
+
+ AMR_SUB_NONE = 0x0,
+
+ AAC_SUB_MAIN = 0x1,
+ AAC_SUB_LC = 0x2,
+ AAC_SUB_SSR = 0x4,
+ AAC_SUB_LTP = 0x8,
+ AAC_SUB_HE_V1 = 0x10,
+ AAC_SUB_SCALABLE = 0x20,
+ AAC_SUB_ERLC = 0x40,
+ AAC_SUB_LD = 0x80,
+ AAC_SUB_HE_V2 = 0x100,
+ AAC_SUB_ELD = 0x200,
+ AAC_SUB_XHE = 0x300,
+
+ VORBIS_SUB_NONE = 0x0,
+
+ E_AC3_SUB_JOC = 0x1,
+
+ MAT_SUB_1_0 = 0x1,
+ MAT_SUB_2_0 = 0x2,
+ MAT_SUB_2_1 = 0x3,
+
+ /* Aliases */
+ /** note != AudioFormat.ENCODING_PCM_16BIT */
+ PCM_16_BIT = (PCM | PCM_SUB_16_BIT),
+ /** note != AudioFormat.ENCODING_PCM_8BIT */
+ PCM_8_BIT = (PCM | PCM_SUB_8_BIT),
+ PCM_32_BIT = (PCM | PCM_SUB_32_BIT),
+ PCM_8_24_BIT = (PCM | PCM_SUB_8_24_BIT),
+ PCM_FLOAT = (PCM | PCM_SUB_FLOAT),
+ PCM_24_BIT_PACKED = (PCM | PCM_SUB_24_BIT_PACKED),
+ AAC_MAIN = (AAC | AAC_SUB_MAIN),
+ AAC_LC = (AAC | AAC_SUB_LC),
+ AAC_SSR = (AAC | AAC_SUB_SSR),
+ AAC_LTP = (AAC | AAC_SUB_LTP),
+ AAC_HE_V1 = (AAC | AAC_SUB_HE_V1),
+ AAC_SCALABLE = (AAC | AAC_SUB_SCALABLE),
+ AAC_ERLC = (AAC | AAC_SUB_ERLC),
+ AAC_LD = (AAC | AAC_SUB_LD),
+ AAC_HE_V2 = (AAC | AAC_SUB_HE_V2),
+ AAC_ELD = (AAC | AAC_SUB_ELD),
+ AAC_XHE = (AAC | AAC_SUB_XHE),
+ AAC_ADTS_MAIN = (AAC_ADTS | AAC_SUB_MAIN),
+ AAC_ADTS_LC = (AAC_ADTS | AAC_SUB_LC),
+ AAC_ADTS_SSR = (AAC_ADTS | AAC_SUB_SSR),
+ AAC_ADTS_LTP = (AAC_ADTS | AAC_SUB_LTP),
+ AAC_ADTS_HE_V1 = (AAC_ADTS | AAC_SUB_HE_V1),
+ AAC_ADTS_SCALABLE = (AAC_ADTS | AAC_SUB_SCALABLE),
+ AAC_ADTS_ERLC = (AAC_ADTS | AAC_SUB_ERLC),
+ AAC_ADTS_LD = (AAC_ADTS | AAC_SUB_LD),
+ AAC_ADTS_HE_V2 = (AAC_ADTS | AAC_SUB_HE_V2),
+ AAC_ADTS_ELD = (AAC_ADTS | AAC_SUB_ELD),
+ AAC_ADTS_XHE = (AAC_ADTS | AAC_SUB_XHE),
+ E_AC3_JOC = (E_AC3 | E_AC3_SUB_JOC),
+ MAT_1_0 = (MAT | MAT_SUB_1_0),
+ MAT_2_0 = (MAT | MAT_SUB_2_0),
+ MAT_2_1 = (MAT | MAT_SUB_2_1),
+ AAC_LATM_LC = (AAC_LATM | AAC_SUB_LC),
+ AAC_LATM_HE_V1 = (AAC_LATM | AAC_SUB_HE_V1),
+ AAC_LATM_HE_V2 = (AAC_LATM | AAC_SUB_HE_V2),
+};
+
+/**
+ * Usage of these values highlights places in the code that use 2- or 8- channel
+ * assumptions.
+ */
+@export(name="")
+enum FixedChannelCount : int32_t {
+ FCC_2 = 2, // This is typically due to legacy implementation of stereo I/O
+ FCC_8 = 8 // This is typically due to audio mixer and resampler limitations
+};
+
+/**
+ * A channel mask per se only defines the presence or absence of a channel, not
+ * the order.
+ *
+ * The channel order convention is that channels are interleaved in order from
+ * least significant channel mask bit to most significant channel mask bit,
+ * with unused bits skipped. For example for stereo, LEFT would be first,
+ * followed by RIGHT.
+ * Any exceptions to this convention are noted at the appropriate API.
+ *
+ * AudioChannelMask is an opaque type and its internal layout should not be
+ * assumed as it may change in the future. Instead, always use functions
+ * to examine it.
+ *
+ * These are the current representations:
+ *
+ * REPRESENTATION_POSITION
+ * is a channel mask representation for position assignment. Each low-order
+ * bit corresponds to the spatial position of a transducer (output), or
+ * interpretation of channel (input). The user of a channel mask needs to
+ * know the context of whether it is for output or input. The constants
+ * OUT_* or IN_* apply to the bits portion. It is not permitted for no bits
+ * to be set.
+ *
+ * REPRESENTATION_INDEX
+ * is a channel mask representation for index assignment. Each low-order
+ * bit corresponds to a selected channel. There is no platform
+ * interpretation of the various bits. There is no concept of output or
+ * input. It is not permitted for no bits to be set.
+ *
+ * All other representations are reserved for future use.
+ *
+ * Warning: current representation distinguishes between input and output, but
+ * this will not the be case in future revisions of the platform. Wherever there
+ * is an ambiguity between input and output that is currently resolved by
+ * checking the channel mask, the implementer should look for ways to fix it
+ * with additional information outside of the mask.
+ */
+@export(name="", value_prefix="AUDIO_CHANNEL_")
+enum AudioChannelMask : uint32_t {
+ /** must be 0 for compatibility */
+ REPRESENTATION_POSITION = 0,
+ /** 1 is reserved for future use */
+ REPRESENTATION_INDEX = 2,
+ /* 3 is reserved for future use */
+
+ /** These can be a complete value of AudioChannelMask */
+ NONE = 0x0,
+ INVALID = 0xC0000000,
+
+ /*
+ * These can be the bits portion of an AudioChannelMask
+ * with representation REPRESENTATION_POSITION.
+ */
+
+ /** output channels */
+ OUT_FRONT_LEFT = 0x1,
+ OUT_FRONT_RIGHT = 0x2,
+ OUT_FRONT_CENTER = 0x4,
+ OUT_LOW_FREQUENCY = 0x8,
+ OUT_BACK_LEFT = 0x10,
+ OUT_BACK_RIGHT = 0x20,
+ OUT_FRONT_LEFT_OF_CENTER = 0x40,
+ OUT_FRONT_RIGHT_OF_CENTER = 0x80,
+ OUT_BACK_CENTER = 0x100,
+ OUT_SIDE_LEFT = 0x200,
+ OUT_SIDE_RIGHT = 0x400,
+ OUT_TOP_CENTER = 0x800,
+ OUT_TOP_FRONT_LEFT = 0x1000,
+ OUT_TOP_FRONT_CENTER = 0x2000,
+ OUT_TOP_FRONT_RIGHT = 0x4000,
+ OUT_TOP_BACK_LEFT = 0x8000,
+ OUT_TOP_BACK_CENTER = 0x10000,
+ OUT_TOP_BACK_RIGHT = 0x20000,
+ OUT_TOP_SIDE_LEFT = 0x40000,
+ OUT_TOP_SIDE_RIGHT = 0x80000,
+
+ /**
+ * Haptic channel characteristics are specific to a device and
+ * only used to play device specific resources (eg: ringtones).
+ * The HAL can freely map A and B to haptic controllers, the
+ * framework shall not interpret those values and forward them
+ * from the device audio assets.
+ */
+ OUT_HAPTIC_A = 0x20000000,
+ OUT_HAPTIC_B = 0x10000000,
+
+ OUT_MONO = OUT_FRONT_LEFT,
+ OUT_STEREO = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT),
+ OUT_2POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_LOW_FREQUENCY),
+ OUT_2POINT0POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_2POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT |
+ OUT_LOW_FREQUENCY),
+ OUT_3POINT0POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_FRONT_CENTER |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_3POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_FRONT_CENTER |
+ OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT |
+ OUT_LOW_FREQUENCY),
+ OUT_QUAD = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT),
+ OUT_QUAD_BACK = OUT_QUAD,
+ /** like OUT_QUAD_BACK with *_SIDE_* instead of *_BACK_* */
+ OUT_QUAD_SIDE = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
+ OUT_SURROUND = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_BACK_CENTER),
+ OUT_PENTA = (OUT_QUAD | OUT_FRONT_CENTER),
+ OUT_5POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT),
+ OUT_5POINT1_BACK = OUT_5POINT1,
+ /** like OUT_5POINT1_BACK with *_SIDE_* instead of *_BACK_* */
+ OUT_5POINT1_SIDE = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
+ OUT_5POINT1POINT2 = (OUT_5POINT1 | OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_5POINT1POINT4 = (OUT_5POINT1 |
+ OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_RIGHT |
+ OUT_TOP_BACK_LEFT | OUT_TOP_BACK_RIGHT),
+ OUT_6POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT |
+ OUT_BACK_CENTER),
+ /** matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND */
+ OUT_7POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
+ OUT_BACK_LEFT | OUT_BACK_RIGHT |
+ OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
+ OUT_7POINT1POINT2 = (OUT_7POINT1 | OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
+ OUT_7POINT1POINT4 = (OUT_7POINT1 |
+ OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_RIGHT |
+ OUT_TOP_BACK_LEFT | OUT_TOP_BACK_RIGHT),
+ OUT_MONO_HAPTIC_A = (OUT_FRONT_LEFT | OUT_HAPTIC_A),
+ OUT_STEREO_HAPTIC_A = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_HAPTIC_A),
+ OUT_HAPTIC_AB = (OUT_HAPTIC_A | OUT_HAPTIC_B),
+ OUT_MONO_HAPTIC_AB = (OUT_FRONT_LEFT | OUT_HAPTIC_A | OUT_HAPTIC_B),
+ OUT_STEREO_HAPTIC_AB = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
+ OUT_HAPTIC_A | OUT_HAPTIC_B),
+ // Note that the 2.0 OUT_ALL* have been moved to helper functions
+
+ /* These are bits only, not complete values */
+
+ /** input channels */
+ IN_LEFT = 0x4,
+ IN_RIGHT = 0x8,
+ IN_FRONT = 0x10,
+ IN_BACK = 0x20,
+ IN_LEFT_PROCESSED = 0x40,
+ IN_RIGHT_PROCESSED = 0x80,
+ IN_FRONT_PROCESSED = 0x100,
+ IN_BACK_PROCESSED = 0x200,
+ IN_PRESSURE = 0x400,
+ IN_X_AXIS = 0x800,
+ IN_Y_AXIS = 0x1000,
+ IN_Z_AXIS = 0x2000,
+ IN_BACK_LEFT = 0x10000,
+ IN_BACK_RIGHT = 0x20000,
+ IN_CENTER = 0x40000,
+ IN_LOW_FREQUENCY = 0x100000,
+ IN_TOP_LEFT = 0x200000,
+ IN_TOP_RIGHT = 0x400000,
+
+ IN_VOICE_UPLINK = 0x4000,
+ IN_VOICE_DNLINK = 0x8000,
+
+ IN_MONO = IN_FRONT,
+ IN_STEREO = (IN_LEFT | IN_RIGHT),
+ IN_FRONT_BACK = (IN_FRONT | IN_BACK),
+ IN_6 = (IN_LEFT | IN_RIGHT |
+ IN_FRONT | IN_BACK |
+ IN_LEFT_PROCESSED | IN_RIGHT_PROCESSED),
+ IN_2POINT0POINT2 = (IN_LEFT | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT),
+ IN_2POINT1POINT2 = (IN_LEFT | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT |
+ IN_LOW_FREQUENCY),
+ IN_3POINT0POINT2 = (IN_LEFT | IN_CENTER | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT),
+ IN_3POINT1POINT2 = (IN_LEFT | IN_CENTER | IN_RIGHT |
+ IN_TOP_LEFT | IN_TOP_RIGHT | IN_LOW_FREQUENCY),
+ IN_5POINT1 = (IN_LEFT | IN_CENTER | IN_RIGHT |
+ IN_BACK_LEFT | IN_BACK_RIGHT | IN_LOW_FREQUENCY),
+ IN_VOICE_UPLINK_MONO = (IN_VOICE_UPLINK | IN_MONO),
+ IN_VOICE_DNLINK_MONO = (IN_VOICE_DNLINK | IN_MONO),
+ IN_VOICE_CALL_MONO = (IN_VOICE_UPLINK_MONO |
+ IN_VOICE_DNLINK_MONO),
+ // Note that the 2.0 IN_ALL* have been moved to helper functions
+
+ COUNT_MAX = 30,
+ INDEX_HDR = REPRESENTATION_INDEX << COUNT_MAX,
+ INDEX_MASK_1 = INDEX_HDR | ((1 << 1) - 1),
+ INDEX_MASK_2 = INDEX_HDR | ((1 << 2) - 1),
+ INDEX_MASK_3 = INDEX_HDR | ((1 << 3) - 1),
+ INDEX_MASK_4 = INDEX_HDR | ((1 << 4) - 1),
+ INDEX_MASK_5 = INDEX_HDR | ((1 << 5) - 1),
+ INDEX_MASK_6 = INDEX_HDR | ((1 << 6) - 1),
+ INDEX_MASK_7 = INDEX_HDR | ((1 << 7) - 1),
+ INDEX_MASK_8 = INDEX_HDR | ((1 << 8) - 1),
+ INDEX_MASK_9 = INDEX_HDR | ((1 << 9) - 1),
+ INDEX_MASK_10 = INDEX_HDR | ((1 << 10) - 1),
+ INDEX_MASK_11 = INDEX_HDR | ((1 << 11) - 1),
+ INDEX_MASK_12 = INDEX_HDR | ((1 << 12) - 1),
+ INDEX_MASK_13 = INDEX_HDR | ((1 << 13) - 1),
+ INDEX_MASK_14 = INDEX_HDR | ((1 << 14) - 1),
+ INDEX_MASK_15 = INDEX_HDR | ((1 << 15) - 1),
+ INDEX_MASK_16 = INDEX_HDR | ((1 << 16) - 1),
+ INDEX_MASK_17 = INDEX_HDR | ((1 << 17) - 1),
+ INDEX_MASK_18 = INDEX_HDR | ((1 << 18) - 1),
+ INDEX_MASK_19 = INDEX_HDR | ((1 << 19) - 1),
+ INDEX_MASK_20 = INDEX_HDR | ((1 << 20) - 1),
+ INDEX_MASK_21 = INDEX_HDR | ((1 << 21) - 1),
+ INDEX_MASK_22 = INDEX_HDR | ((1 << 22) - 1),
+ INDEX_MASK_23 = INDEX_HDR | ((1 << 23) - 1),
+ INDEX_MASK_24 = INDEX_HDR | ((1 << 24) - 1),
+};
+
+/**
+ * Major modes for a mobile device. The current mode setting affects audio
+ * routing.
+ */
+@export(name="audio_mode_t", value_prefix="AUDIO_MODE_")
+enum AudioMode : int32_t {
+ NORMAL = 0,
+ RINGTONE = 1,
+ /** Calls handled by the telephony stack (Eg: PSTN). */
+ IN_CALL = 2,
+ /** Calls handled by apps (Eg: Hangout). */
+ IN_COMMUNICATION = 3,
+ /** Call screening in progress. */
+ CALL_SCREEN = 4,
+};
+
+@export(name="", value_prefix="AUDIO_DEVICE_")
+enum AudioDevice : uint32_t {
+ NONE = 0x0,
+ /** reserved bits */
+ BIT_IN = 0x80000000,
+ BIT_DEFAULT = 0x40000000,
+ /** output devices */
+ OUT_EARPIECE = 0x1,
+ OUT_SPEAKER = 0x2,
+ OUT_WIRED_HEADSET = 0x4,
+ OUT_WIRED_HEADPHONE = 0x8,
+ OUT_BLUETOOTH_SCO = 0x10,
+ OUT_BLUETOOTH_SCO_HEADSET = 0x20,
+ OUT_BLUETOOTH_SCO_CARKIT = 0x40,
+ OUT_BLUETOOTH_A2DP = 0x80,
+ OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
+ OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
+ OUT_AUX_DIGITAL = 0x400,
+ OUT_HDMI = OUT_AUX_DIGITAL,
+ /** uses an analog connection (multiplexed over the USB pins for instance) */
+ OUT_ANLG_DOCK_HEADSET = 0x800,
+ OUT_DGTL_DOCK_HEADSET = 0x1000,
+ /** USB accessory mode: Android device is USB device and dock is USB host */
+ OUT_USB_ACCESSORY = 0x2000,
+ /** USB host mode: Android device is USB host and dock is USB device */
+ OUT_USB_DEVICE = 0x4000,
+ OUT_REMOTE_SUBMIX = 0x8000,
+ /** Telephony voice TX path */
+ OUT_TELEPHONY_TX = 0x10000,
+ /** Analog jack with line impedance detected */
+ OUT_LINE = 0x20000,
+ /** HDMI Audio Return Channel */
+ OUT_HDMI_ARC = 0x40000,
+ /** S/PDIF out */
+ OUT_SPDIF = 0x80000,
+ /** FM transmitter out */
+ OUT_FM = 0x100000,
+ /** Line out for av devices */
+ OUT_AUX_LINE = 0x200000,
+ /** limited-output speaker device for acoustic safety */
+ OUT_SPEAKER_SAFE = 0x400000,
+ OUT_IP = 0x800000,
+ /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
+ OUT_BUS = 0x1000000,
+ OUT_PROXY = 0x2000000,
+ OUT_USB_HEADSET = 0x4000000,
+ OUT_HEARING_AID = 0x8000000,
+ OUT_ECHO_CANCELLER = 0x10000000,
+ OUT_DEFAULT = BIT_DEFAULT,
+ // Note that the 2.0 OUT_ALL* have been moved to helper functions
+
+ /** input devices */
+ IN_COMMUNICATION = BIT_IN | 0x1,
+ IN_AMBIENT = BIT_IN | 0x2,
+ IN_BUILTIN_MIC = BIT_IN | 0x4,
+ IN_BLUETOOTH_SCO_HEADSET = BIT_IN | 0x8,
+ IN_WIRED_HEADSET = BIT_IN | 0x10,
+ IN_AUX_DIGITAL = BIT_IN | 0x20,
+ IN_HDMI = IN_AUX_DIGITAL,
+ /** Telephony voice RX path */
+ IN_VOICE_CALL = BIT_IN | 0x40,
+ IN_TELEPHONY_RX = IN_VOICE_CALL,
+ IN_BACK_MIC = BIT_IN | 0x80,
+ IN_REMOTE_SUBMIX = BIT_IN | 0x100,
+ IN_ANLG_DOCK_HEADSET = BIT_IN | 0x200,
+ IN_DGTL_DOCK_HEADSET = BIT_IN | 0x400,
+ IN_USB_ACCESSORY = BIT_IN | 0x800,
+ IN_USB_DEVICE = BIT_IN | 0x1000,
+ /** FM tuner input */
+ IN_FM_TUNER = BIT_IN | 0x2000,
+ /** TV tuner input */
+ IN_TV_TUNER = BIT_IN | 0x4000,
+ /** Analog jack with line impedance detected */
+ IN_LINE = BIT_IN | 0x8000,
+ /** S/PDIF in */
+ IN_SPDIF = BIT_IN | 0x10000,
+ IN_BLUETOOTH_A2DP = BIT_IN | 0x20000,
+ IN_LOOPBACK = BIT_IN | 0x40000,
+ IN_IP = BIT_IN | 0x80000,
+ /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
+ IN_BUS = BIT_IN | 0x100000,
+ IN_PROXY = BIT_IN | 0x1000000,
+ IN_USB_HEADSET = BIT_IN | 0x2000000,
+ IN_BLUETOOTH_BLE = BIT_IN | 0x4000000,
+ IN_ECHO_REFERENCE = BIT_IN | 0x10000000,
+ IN_DEFAULT = BIT_IN | BIT_DEFAULT,
+
+ // Note that the 2.0 IN_ALL* have been moved to helper functions
+};
+
+/**
+ * IEEE 802 MAC address.
+ */
+typedef uint8_t[6] MacAddress;
+
+/**
+ * Specifies a device address 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;
+ /** Arbitrary BUS device unique address. Should not be interpreted by the framework. */
+ string busAddress;
+ /** Arbitrary REMOTE_SUBMIX device unique address. Should not be interpreted by the HAL. */
+ string rSubmixAddress;
+};
+
+/**
+ * The audio output flags serve two purposes:
+ *
+ * - when an AudioTrack is created they indicate a "wish" to be connected to an
+ * output stream with attributes corresponding to the specified flags;
+ *
+ * - when present in an output profile descriptor listed for a particular audio
+ * hardware module, they indicate that an output stream can be opened that
+ * supports the attributes indicated by the flags.
+ *
+ * The audio policy manager will try to match the flags in the request
+ * (when getOuput() is called) to an available output stream.
+ */
+@export(name="audio_output_flags_t", value_prefix="AUDIO_OUTPUT_FLAG_")
+enum AudioOutputFlag : int32_t {
+ NONE = 0x0, // no attributes
+ DIRECT = 0x1, // this output directly connects a track
+ // to one output stream: no software mixer
+ PRIMARY = 0x2, // this output is the primary output of the device. It is
+ // unique and must be present. It is opened by default and
+ // receives routing, audio mode and volume controls related
+ // to voice calls.
+ FAST = 0x4, // output supports "fast tracks", defined elsewhere
+ DEEP_BUFFER = 0x8, // use deep audio buffers
+ COMPRESS_OFFLOAD = 0x10, // offload playback of compressed streams to
+ // hardware codec
+ NON_BLOCKING = 0x20, // use non-blocking write
+ HW_AV_SYNC = 0x40, // output uses a hardware A/V sync
+ TTS = 0x80, // output for streams transmitted through speaker at a
+ // sample rate high enough to accommodate lower-range
+ // ultrasonic p/b
+ RAW = 0x100, // minimize signal processing
+ SYNC = 0x200, // synchronize I/O streams
+ IEC958_NONAUDIO = 0x400, // Audio stream contains compressed audio in SPDIF
+ // data bursts, not PCM.
+ DIRECT_PCM = 0x2000, // Audio stream containing PCM data that needs
+ // to pass through compress path for DSP post proc.
+ MMAP_NOIRQ = 0x4000, // output operates in MMAP no IRQ mode.
+ VOIP_RX = 0x8000, // preferred output for VoIP calls.
+ /** preferred output for call music */
+ INCALL_MUSIC = 0x10000,
+};
+
+/**
+ * The audio input flags are analogous to audio output flags.
+ * Currently they are used only when an AudioRecord is created,
+ * to indicate a preference to be connected to an input stream with
+ * attributes corresponding to the specified flags.
+ */
+@export(name="audio_input_flags_t", value_prefix="AUDIO_INPUT_FLAG_")
+enum AudioInputFlag : int32_t {
+ NONE = 0x0, // no attributes
+ FAST = 0x1, // prefer an input that supports "fast tracks"
+ HW_HOTWORD = 0x2, // prefer an input that captures from hw hotword source
+ RAW = 0x4, // minimize signal processing
+ SYNC = 0x8, // synchronize I/O streams
+ MMAP_NOIRQ = 0x10, // input operates in MMAP no IRQ mode.
+ VOIP_TX = 0x20, // preferred input for VoIP calls.
+ HW_AV_SYNC = 0x40, // input connected to an output that uses a hardware A/V sync
+ DIRECT = 0x80, // for acquiring encoded streams
+};
+
+@export(name="audio_usage_t", value_prefix="AUDIO_USAGE_")
+enum AudioUsage : int32_t {
+ // These values must kept in sync with
+ // frameworks/base/media/java/android/media/AudioAttributes.java
+ // Note that not all framework values are exposed
+ /**
+ * Usage value to use when the usage is unknown.
+ */
+ UNKNOWN = 0,
+ /**
+ * Usage value to use when the usage is media, such as music, or movie
+ * soundtracks.
+ */
+ MEDIA = 1,
+ /**
+ * Usage value to use when the usage is voice communications, such as
+ * telephony or VoIP.
+ */
+ VOICE_COMMUNICATION = 2,
+ /**
+ * Usage value to use when the usage is in-call signalling, such as with
+ * a "busy" beep, or DTMF tones.
+ */
+ VOICE_COMMUNICATION_SIGNALLING = 3,
+ /**
+ * Usage value to use when the usage is an alarm (e.g. wake-up alarm).
+ */
+ ALARM = 4,
+ /**
+ * Usage value to use when the usage is a generic notification.
+ */
+ NOTIFICATION = 5,
+ /**
+ * Usage value to use when the usage is telephony ringtone.
+ */
+ NOTIFICATION_TELEPHONY_RINGTONE = 6,
+ /**
+ * Usage value to use when the usage is for accessibility, such as with
+ * a screen reader.
+ */
+ ASSISTANCE_ACCESSIBILITY = 11,
+ /**
+ * Usage value to use when the usage is driving or navigation directions.
+ */
+ ASSISTANCE_NAVIGATION_GUIDANCE = 12,
+ /**
+ * Usage value to use when the usage is sonification, such as with user
+ * interface sounds.
+ */
+ ASSISTANCE_SONIFICATION = 13,
+ /**
+ * Usage value to use when the usage is for game audio.
+ */
+ GAME = 14,
+ /**
+ * Usage value to use when feeding audio to the platform and replacing
+ * "traditional" audio source, such as audio capture devices.
+ */
+ VIRTUAL_SOURCE = 15,
+ /**
+ * Usage value to use for audio responses to user queries, audio
+ * instructions or help utterances.
+ */
+ ASSISTANT = 16,
+ /**
+ * Usage value to use for assistant voice interaction with remote caller
+ * on Cell and VoIP calls.
+ */
+ CALL_ASSISTANT = 17,
+ /**
+ * Usage value to use when the usage is an emergency.
+ */
+ EMERGENCY = 1000,
+ /**
+ * Usage value to use when the usage is a safety sound.
+ */
+ SAFETY = 1001,
+ /**
+ * Usage value to use when the usage is a vehicle status.
+ */
+ VEHICLE_STATUS = 1002,
+ /**
+ * Usage value to use when the usage is an announcement.
+ */
+ ANNOUNCEMENT = 1003,
+};
+
+/** Type of audio generated by an application. */
+@export(name="audio_content_type_t", value_prefix="AUDIO_CONTENT_TYPE_")
+enum AudioContentType : uint32_t {
+ // Do not change these values without updating their counterparts
+ // in frameworks/base/media/java/android/media/AudioAttributes.java
+ /**
+ * Content type value to use when the content type is unknown, or other than
+ * the ones defined.
+ */
+ UNKNOWN = 0,
+ /**
+ * Content type value to use when the content type is speech.
+ */
+ SPEECH = 1,
+ /**
+ * Content type value to use when the content type is music.
+ */
+ MUSIC = 2,
+ /**
+ * Content type value to use when the content type is a soundtrack,
+ * typically accompanying a movie or TV program.
+ */
+ MOVIE = 3,
+ /**
+ * Content type value to use when the content type is a sound used to
+ * accompany a user action, such as a beep or sound effect expressing a key
+ * click, or event, such as the type of a sound for a bonus being received
+ * in a game. These sounds are mostly synthesized or short Foley sounds.
+ */
+ SONIFICATION = 4,
+};
+
+/** Encapsulation mode used for sending audio compressed data. */
+@export(name="audio_encapsulation_mode_t", value_prefix="AUDIO_ENCAPSULATION_MODE_")
+enum AudioEncapsulationMode : int32_t {
+ // Do not change these values without updating their counterparts
+ // in frameworks/base/media/java/android/media/AudioTrack.java
+ /**
+ * No encapsulation mode for metadata.
+ */
+ NONE = 0,
+ /**
+ * Elementary stream payload with metadata
+ */
+ ELEMENTARY_STREAM = 1,
+ /**
+ * Handle-based payload with metadata
+ */
+ HANDLE = 2,
+};
+
+/**
+ * Additional information about the stream passed to hardware decoders.
+ */
+struct AudioOffloadInfo {
+ uint32_t sampleRateHz;
+ bitfield<AudioChannelMask> channelMask;
+ AudioFormat format;
+ AudioStreamType streamType;
+ uint32_t bitRatePerSecond;
+ int64_t durationMicroseconds; // -1 if unknown
+ bool hasVideo;
+ bool isStreaming;
+ uint32_t bitWidth;
+ uint32_t bufferSize;
+ AudioUsage usage;
+ AudioEncapsulationMode encapsulationMode;
+ int32_t contentId;
+ int32_t syncId;
+};
+
+/**
+ * Commonly used audio stream configuration parameters.
+ */
+struct AudioConfig {
+ uint32_t sampleRateHz;
+ bitfield<AudioChannelMask> channelMask;
+ AudioFormat format;
+ AudioOffloadInfo offloadInfo;
+ uint64_t frameCount;
+};
+
+/** 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;
+ /**
+ * Indicates the destination of an input stream, can be left unspecified.
+ */
+ safe_union Destination {
+ Monostate unspecified;
+ DeviceAddress device;
+ };
+ Destination destination;
+};
+
+/** Metadatas of the sink of a StreamIn. */
+struct SinkMetadata {
+ vec<RecordTrackMetadata> tracks;
+};
+
+
+/*
+ *
+ * Volume control
+ *
+ */
+
+/**
+ * Type of gain control exposed by an audio port.
+ */
+@export(name="", value_prefix="AUDIO_GAIN_MODE_")
+enum AudioGainMode : uint32_t {
+ JOINT = 0x1, // supports joint channel gain control
+ CHANNELS = 0x2, // supports separate channel gain control
+ RAMP = 0x4 // supports gain ramps
+};
+
+/**
+ * An audio_gain struct is a representation of a gain stage.
+ * A gain stage is always attached to an audio port.
+ */
+struct AudioGain {
+ bitfield<AudioGainMode> mode;
+ bitfield<AudioChannelMask> channelMask; // channels which gain an be controlled
+ int32_t minValue; // minimum gain value in millibels
+ int32_t maxValue; // maximum gain value in millibels
+ int32_t defaultValue; // default gain value in millibels
+ uint32_t stepValue; // gain step in millibels
+ uint32_t minRampMs; // minimum ramp duration in ms
+ uint32_t maxRampMs; // maximum ramp duration in ms
+};
+
+/**
+ * The gain configuration structure is used to get or set the gain values of a
+ * given port.
+ */
+struct AudioGainConfig {
+ int32_t index; // index of the corresponding AudioGain in AudioPort.gains
+ AudioGainMode mode;
+ AudioChannelMask channelMask; // channels which gain value follows
+ /**
+ * 4 = sizeof(AudioChannelMask),
+ * 8 is not "FCC_8", so it won't need to be changed for > 8 channels.
+ * Gain values in millibels for each channel ordered from LSb to MSb in
+ * channel mask. The number of values is 1 in joint mode or
+ * popcount(channel_mask).
+ */
+ int32_t[4 * 8] values;
+ uint32_t rampDurationMs; // ramp duration in ms
+};
+
+
+/*
+ *
+ * Routing control
+ *
+ */
+
+/*
+ * Types defined here are used to describe an audio source or sink at internal
+ * framework interfaces (audio policy, patch panel) or at the audio HAL.
+ * Sink and sources are grouped in a concept of “audio port” representing an
+ * audio end point at the edge of the system managed by the module exposing
+ * the interface.
+ */
+
+/** Audio port role: either source or sink */
+@export(name="audio_port_role_t", value_prefix="AUDIO_PORT_ROLE_")
+enum AudioPortRole : int32_t {
+ NONE,
+ SOURCE,
+ SINK,
+};
+
+/**
+ * Audio port type indicates if it is a session (e.g AudioTrack), a mix (e.g
+ * PlaybackThread output) or a physical device (e.g OUT_SPEAKER)
+ */
+@export(name="audio_port_type_t", value_prefix="AUDIO_PORT_TYPE_")
+enum AudioPortType : int32_t {
+ NONE,
+ DEVICE,
+ MIX,
+ SESSION,
+};
+
+/**
+ * Extension for audio port configuration structure when the audio port is a
+ * hardware device.
+ */
+struct AudioPortConfigDeviceExt {
+ AudioModuleHandle hwModule; // module the device is attached to
+ AudioDevice type; // device type (e.g OUT_SPEAKER)
+ uint8_t[32] address; // device address. "" if N/A
+};
+
+/**
+ * Extension for audio port configuration structure when the audio port is an
+ * audio session.
+ */
+struct AudioPortConfigSessionExt {
+ AudioSession session;
+};
+
+/**
+ * Flags indicating which fields are to be considered in AudioPortConfig.
+ */
+@export(name="", value_prefix="AUDIO_PORT_CONFIG_")
+enum AudioPortConfigMask : uint32_t {
+ SAMPLE_RATE = 0x1,
+ CHANNEL_MASK = 0x2,
+ FORMAT = 0x4,
+ GAIN = 0x8,
+};
+
+/**
+ * Audio port configuration structure used to specify a particular configuration
+ * of an audio port.
+ */
+struct AudioPortConfig {
+ AudioPortHandle id;
+ bitfield<AudioPortConfigMask> configMask;
+ uint32_t sampleRateHz;
+ bitfield<AudioChannelMask> channelMask;
+ AudioFormat format;
+ AudioGainConfig gain;
+ AudioPortType type; // type is used as a discriminator for Ext union
+ AudioPortRole role; // role is used as a discriminator for UseCase union
+ union Ext {
+ AudioPortConfigDeviceExt device;
+ struct AudioPortConfigMixExt {
+ AudioModuleHandle hwModule; // module the stream is attached to
+ AudioIoHandle ioHandle; // I/O handle of the input/output stream
+ union UseCase {
+ AudioStreamType stream;
+ AudioSource source;
+ } useCase;
+ } mix;
+ AudioPortConfigSessionExt session;
+ } ext;
+};
+
+/**
+ * Extension for audio port structure when the audio port is a hardware device.
+ */
+struct AudioPortDeviceExt {
+ AudioModuleHandle hwModule; // module the device is attached to
+ AudioDevice type;
+ /** 32 byte string identifying the port. */
+ uint8_t[32] address;
+};
+
+/**
+ * Latency class of the audio mix.
+ */
+@export(name="audio_mix_latency_class_t", value_prefix="AUDIO_LATENCY_")
+enum AudioMixLatencyClass : int32_t {
+ LOW,
+ NORMAL
+};
+
+struct AudioPortMixExt {
+ AudioModuleHandle hwModule; // module the stream is attached to
+ AudioIoHandle ioHandle; // I/O handle of the stream
+ AudioMixLatencyClass latencyClass;
+};
+
+/**
+ * Extension for audio port structure when the audio port is an audio session.
+ */
+struct AudioPortSessionExt {
+ AudioSession session;
+};
+
+struct AudioPort {
+ AudioPortHandle id;
+ AudioPortRole role;
+ string name;
+ vec<uint32_t> sampleRates;
+ vec<bitfield<AudioChannelMask>> channelMasks;
+ vec<AudioFormat> formats;
+ vec<AudioGain> gains;
+ AudioPortConfig activeConfig; // current audio port configuration
+ AudioPortType type; // type is used as a discriminator
+ union Ext {
+ AudioPortDeviceExt device;
+ AudioPortMixExt mix;
+ AudioPortSessionExt session;
+ } ext;
+};
+
+struct ThreadInfo {
+ int64_t pid;
+ int64_t tid;
+};
diff --git a/audio/common/all-versions/default/Android.bp b/audio/common/all-versions/default/Android.bp
index 0eb4a71..5caddcf 100644
--- a/audio/common/all-versions/default/Android.bp
+++ b/audio/common/all-versions/default/Android.bp
@@ -117,3 +117,16 @@
"-include common/all-versions/VersionMacro.h",
]
}
+
+cc_library_shared {
+ name: "android.hardware.audio.common@7.0-util",
+ defaults: ["android.hardware.audio.common-util_default"],
+ shared_libs: [
+ "android.hardware.audio.common@7.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
diff --git a/audio/core/all-versions/default/Android.bp b/audio/core/all-versions/default/Android.bp
index 8fdb70d..46f64b7 100644
--- a/audio/core/all-versions/default/Android.bp
+++ b/audio/core/all-versions/default/Android.bp
@@ -123,3 +123,18 @@
name: "android.hardware.audio@6.0-impl",
defaults: ["android.hardware.audio@6.0-impl_default"],
}
+
+cc_library_shared {
+ name: "android.hardware.audio@7.0-impl",
+ defaults: ["android.hardware.audio-impl_default"],
+ shared_libs: [
+ "android.hardware.audio@7.0",
+ "android.hardware.audio.common@7.0",
+ "android.hardware.audio.common@7.0-util",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ],
+}
diff --git a/audio/core/all-versions/vts/functional/7.0/AudioPrimaryHidlHalTest.cpp b/audio/core/all-versions/vts/functional/7.0/AudioPrimaryHidlHalTest.cpp
new file mode 100644
index 0000000..33efa6f
--- /dev/null
+++ b/audio/core/all-versions/vts/functional/7.0/AudioPrimaryHidlHalTest.cpp
@@ -0,0 +1,18 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// pull in all the <= 6.0 tests
+#include "6.0/AudioPrimaryHidlHalTest.cpp"
diff --git a/audio/core/all-versions/vts/functional/Android.bp b/audio/core/all-versions/vts/functional/Android.bp
index 2d5e8a5..598944a 100644
--- a/audio/core/all-versions/vts/functional/Android.bp
+++ b/audio/core/all-versions/vts/functional/Android.bp
@@ -128,3 +128,26 @@
// TODO(b/146104851): Add auto-gen rules and remove it.
test_config: "VtsHalAudioV6_0TargetTest.xml",
}
+
+cc_test {
+ name: "VtsHalAudioV7_0TargetTest",
+ defaults: ["VtsHalAudioTargetTest_defaults"],
+ srcs: [
+ "7.0/AudioPrimaryHidlHalTest.cpp",
+ ],
+ static_libs: [
+ "android.hardware.audio@7.0",
+ "android.hardware.audio.common@7.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ],
+ data: [
+ ":audio_policy_configuration_V7_0",
+ ],
+ // Use test_config for vts suite.
+ // TODO(b/146104851): Add auto-gen rules and remove it.
+ test_config: "VtsHalAudioV7_0TargetTest.xml",
+}
diff --git a/audio/core/all-versions/vts/functional/VtsHalAudioV7_0TargetTest.xml b/audio/core/all-versions/vts/functional/VtsHalAudioV7_0TargetTest.xml
new file mode 100644
index 0000000..6635f31
--- /dev/null
+++ b/audio/core/all-versions/vts/functional/VtsHalAudioV7_0TargetTest.xml
@@ -0,0 +1,38 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Runs VtsHalAudioV7_0TargetTest.">
+ <option name="test-suite-tag" value="apct" />
+ <option name="test-suite-tag" value="apct-native" />
+
+ <target_preparer class="com.android.tradefed.targetprep.RootTargetPreparer"/>
+ <target_preparer class="com.android.tradefed.targetprep.StopServicesSetup"/>
+
+ <target_preparer class="com.android.tradefed.targetprep.RunCommandTargetPreparer">
+ <option name="run-command" value="setprop vts.native_server.on 1"/>
+ <option name="teardown-command" value="setprop vts.native_server.on 0"/>
+ </target_preparer>
+
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="VtsHalAudioV7_0TargetTest->/data/local/tmp/VtsHalAudioV7_0TargetTest" />
+ <option name="push" value="audio_policy_configuration_V7_0.xsd->/data/local/tmp/audio_policy_configuration_V7_0.xsd" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="VtsHalAudioV7_0TargetTest" />
+ </test>
+</configuration>
diff --git a/audio/effect/7.0/Android.bp b/audio/effect/7.0/Android.bp
new file mode 100644
index 0000000..c113782
--- /dev/null
+++ b/audio/effect/7.0/Android.bp
@@ -0,0 +1,30 @@
+// This file is autogenerated by hidl-gen -Landroidbp.
+
+hidl_interface {
+ name: "android.hardware.audio.effect@7.0",
+ root: "android.hardware",
+ srcs: [
+ "types.hal",
+ "IAcousticEchoCancelerEffect.hal",
+ "IAutomaticGainControlEffect.hal",
+ "IBassBoostEffect.hal",
+ "IDownmixEffect.hal",
+ "IEffect.hal",
+ "IEffectBufferProviderCallback.hal",
+ "IEffectsFactory.hal",
+ "IEnvironmentalReverbEffect.hal",
+ "IEqualizerEffect.hal",
+ "ILoudnessEnhancerEffect.hal",
+ "INoiseSuppressionEffect.hal",
+ "IPresetReverbEffect.hal",
+ "IVirtualizerEffect.hal",
+ "IVisualizerEffect.hal",
+ ],
+ interfaces: [
+ "android.hardware.audio.common@7.0",
+ "android.hidl.base@1.0",
+ "android.hidl.safe_union@1.0",
+ ],
+ gen_java: false,
+ gen_java_constants: true,
+}
diff --git a/audio/effect/7.0/IAcousticEchoCancelerEffect.hal b/audio/effect/7.0/IAcousticEchoCancelerEffect.hal
new file mode 100644
index 0000000..2bc2a7f
--- /dev/null
+++ b/audio/effect/7.0/IAcousticEchoCancelerEffect.hal
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IAcousticEchoCancelerEffect extends IEffect {
+ /**
+ * Sets echo delay value in milliseconds.
+ */
+ setEchoDelay(uint32_t echoDelayMs) generates (Result retval);
+
+ /**
+ * Gets echo delay value in milliseconds.
+ */
+ getEchoDelay() generates (Result retval, uint32_t echoDelayMs);
+};
diff --git a/audio/effect/7.0/IAutomaticGainControlEffect.hal b/audio/effect/7.0/IAutomaticGainControlEffect.hal
new file mode 100644
index 0000000..8ffa659
--- /dev/null
+++ b/audio/effect/7.0/IAutomaticGainControlEffect.hal
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IAutomaticGainControlEffect extends IEffect {
+ /**
+ * Sets target level in millibels.
+ */
+ setTargetLevel(int16_t targetLevelMb) generates (Result retval);
+
+ /**
+ * Gets target level.
+ */
+ getTargetLevel() generates (Result retval, int16_t targetLevelMb);
+
+ /**
+ * Sets gain in the compression range in millibels.
+ */
+ setCompGain(int16_t compGainMb) generates (Result retval);
+
+ /**
+ * Gets gain in the compression range.
+ */
+ getCompGain() generates (Result retval, int16_t compGainMb);
+
+ /**
+ * Enables or disables limiter.
+ */
+ setLimiterEnabled(bool enabled) generates (Result retval);
+
+ /**
+ * Returns whether limiter is enabled.
+ */
+ isLimiterEnabled() generates (Result retval, bool enabled);
+
+ struct AllProperties {
+ int16_t targetLevelMb;
+ int16_t compGainMb;
+ bool limiterEnabled;
+ };
+
+ /**
+ * Sets all properties at once.
+ */
+ setAllProperties(AllProperties properties) generates (Result retval);
+
+ /**
+ * Gets all properties at once.
+ */
+ getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/7.0/IBassBoostEffect.hal b/audio/effect/7.0/IBassBoostEffect.hal
new file mode 100644
index 0000000..d8d049e
--- /dev/null
+++ b/audio/effect/7.0/IBassBoostEffect.hal
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IBassBoostEffect extends IEffect {
+ /**
+ * Returns whether setting bass boost strength is supported.
+ */
+ isStrengthSupported() generates (Result retval, bool strengthSupported);
+
+ enum StrengthRange : uint16_t {
+ MIN = 0,
+ MAX = 1000
+ };
+
+ /**
+ * Sets bass boost strength.
+ *
+ * @param strength strength of the effect. The valid range for strength
+ * strength is [0, 1000], where 0 per mille designates the
+ * mildest effect and 1000 per mille designates the
+ * strongest.
+ * @return retval operation completion status.
+ */
+ setStrength(uint16_t strength) generates (Result retval);
+
+ /**
+ * Gets virtualization strength.
+ */
+ getStrength() generates (Result retval, uint16_t strength);
+};
diff --git a/audio/effect/7.0/IDownmixEffect.hal b/audio/effect/7.0/IDownmixEffect.hal
new file mode 100644
index 0000000..2035430
--- /dev/null
+++ b/audio/effect/7.0/IDownmixEffect.hal
@@ -0,0 +1,37 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IDownmixEffect extends IEffect {
+ enum Type : int32_t {
+ STRIP, // throw away the extra channels
+ FOLD // mix the extra channels with FL/FR
+ };
+
+ /**
+ * Sets the current downmix preset.
+ */
+ setType(Type preset) generates (Result retval);
+
+ /**
+ * Gets the current downmix preset.
+ */
+ getType() generates (Result retval, Type preset);
+};
diff --git a/audio/effect/7.0/IEffect.hal b/audio/effect/7.0/IEffect.hal
new file mode 100644
index 0000000..5b176dc
--- /dev/null
+++ b/audio/effect/7.0/IEffect.hal
@@ -0,0 +1,421 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffectBufferProviderCallback;
+
+interface IEffect {
+ /**
+ * Initialize effect engine--all configurations return to default.
+ *
+ * @return retval operation completion status.
+ */
+ @entry
+ init() generates (Result retval);
+
+ /**
+ * Apply new audio parameters configurations for input and output buffers.
+ * The provider callbacks may be empty, but in this case the buffer
+ * must be provided in the EffectConfig structure.
+ *
+ * @param config configuration descriptor.
+ * @param inputBufferProvider optional buffer provider reference.
+ * @param outputBufferProvider optional buffer provider reference.
+ * @return retval operation completion status.
+ */
+ setConfig(EffectConfig config,
+ IEffectBufferProviderCallback inputBufferProvider,
+ IEffectBufferProviderCallback outputBufferProvider)
+ generates (Result retval);
+
+ /**
+ * Reset the effect engine. Keep configuration but resets state and buffer
+ * content.
+ *
+ * @return retval operation completion status.
+ */
+ reset() generates (Result retval);
+
+ /**
+ * Enable processing.
+ *
+ * @return retval operation completion status.
+ */
+ @callflow(next={"prepareForProcessing"})
+ enable() generates (Result retval);
+
+ /**
+ * Disable processing.
+ *
+ * @return retval operation completion status.
+ */
+ @callflow(next={"close"})
+ disable() generates (Result retval);
+
+ /**
+ * Set the rendering device the audio output path is connected to. The
+ * effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its
+ * descriptor to receive this command when the device changes.
+ *
+ * Note: this method is only supported for effects inserted into
+ * the output chain.
+ *
+ * @param device output device specification.
+ * @return retval operation completion status.
+ */
+ setDevice(bitfield<AudioDevice> device) generates (Result retval);
+
+ /**
+ * Set and get volume. Used by audio framework to delegate volume control to
+ * effect engine. The effect implementation must set EFFECT_FLAG_VOLUME_CTRL
+ * flag in its descriptor to receive this command. The effect engine must
+ * return the volume that should be applied before the effect is
+ * processed. The overall volume (the volume actually applied by the effect
+ * engine multiplied by the returned value) should match the value indicated
+ * in the command.
+ *
+ * @param volumes vector containing volume for each channel defined in
+ * EffectConfig for output buffer expressed in 8.24 fixed
+ * point format.
+ * @return result updated volume values.
+ * @return retval operation completion status.
+ */
+ setAndGetVolume(vec<uint32_t> volumes)
+ generates (Result retval, vec<uint32_t> result);
+
+ /**
+ * Notify the effect of the volume change. The effect implementation must
+ * set EFFECT_FLAG_VOLUME_IND flag in its descriptor to receive this
+ * command.
+ *
+ * @param volumes vector containing volume for each channel defined in
+ * EffectConfig for output buffer expressed in 8.24 fixed
+ * point format.
+ * @return retval operation completion status.
+ */
+ volumeChangeNotification(vec<uint32_t> volumes)
+ generates (Result retval);
+
+ /**
+ * Set the audio mode. The effect implementation must set
+ * EFFECT_FLAG_AUDIO_MODE_IND flag in its descriptor to receive this command
+ * when the audio mode changes.
+ *
+ * @param mode desired audio mode.
+ * @return retval operation completion status.
+ */
+ setAudioMode(AudioMode mode) generates (Result retval);
+
+ /**
+ * Apply new audio parameters configurations for input and output buffers of
+ * reverse stream. An example of reverse stream is the echo reference
+ * supplied to an Acoustic Echo Canceler.
+ *
+ * @param config configuration descriptor.
+ * @param inputBufferProvider optional buffer provider reference.
+ * @param outputBufferProvider optional buffer provider reference.
+ * @return retval operation completion status.
+ */
+ setConfigReverse(EffectConfig config,
+ IEffectBufferProviderCallback inputBufferProvider,
+ IEffectBufferProviderCallback outputBufferProvider)
+ generates (Result retval);
+
+ /**
+ * Set the capture device the audio input path is connected to. The effect
+ * implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to
+ * receive this command when the device changes.
+ *
+ * Note: this method is only supported for effects inserted into
+ * the input chain.
+ *
+ * @param device input device specification.
+ * @return retval operation completion status.
+ */
+ setInputDevice(bitfield<AudioDevice> device) generates (Result retval);
+
+ /**
+ * Read audio parameters configurations for input and output buffers.
+ *
+ * @return retval operation completion status.
+ * @return config configuration descriptor.
+ */
+ getConfig() generates (Result retval, EffectConfig config);
+
+ /**
+ * Read audio parameters configurations for input and output buffers of
+ * reverse stream.
+ *
+ * @return retval operation completion status.
+ * @return config configuration descriptor.
+ */
+ getConfigReverse() generates (Result retval, EffectConfig config);
+
+ /**
+ * Queries for supported combinations of main and auxiliary channels
+ * (e.g. for a multi-microphone noise suppressor).
+ *
+ * @param maxConfigs maximum number of the combinations to return.
+ * @return retval absence of the feature support is indicated using
+ * NOT_SUPPORTED code. RESULT_TOO_BIG is returned if
+ * the number of supported combinations exceeds 'maxConfigs'.
+ * @return result list of configuration descriptors.
+ */
+ getSupportedAuxChannelsConfigs(uint32_t maxConfigs)
+ generates (Result retval, vec<EffectAuxChannelsConfig> result);
+
+ /**
+ * Retrieves the current configuration of main and auxiliary channels.
+ *
+ * @return retval absence of the feature support is indicated using
+ * NOT_SUPPORTED code.
+ * @return result configuration descriptor.
+ */
+ getAuxChannelsConfig()
+ generates (Result retval, EffectAuxChannelsConfig result);
+
+ /**
+ * Sets the current configuration of main and auxiliary channels.
+ *
+ * @return retval operation completion status; absence of the feature
+ * support is indicated using NOT_SUPPORTED code.
+ */
+ setAuxChannelsConfig(EffectAuxChannelsConfig config)
+ generates (Result retval);
+
+ /**
+ * Set the audio source the capture path is configured for (Camcorder, voice
+ * recognition...).
+ *
+ * Note: this method is only supported for effects inserted into
+ * the input chain.
+ *
+ * @param source source descriptor.
+ * @return retval operation completion status.
+ */
+ setAudioSource(AudioSource source) generates (Result retval);
+
+ /**
+ * This command indicates if the playback thread the effect is attached to
+ * is offloaded or not, and updates the I/O handle of the playback thread
+ * the effect is attached to.
+ *
+ * @param param effect offload descriptor.
+ * @return retval operation completion status.
+ */
+ offload(EffectOffloadParameter param) generates (Result retval);
+
+ /**
+ * Returns the effect descriptor.
+ *
+ * @return retval operation completion status.
+ * @return descriptor effect descriptor.
+ */
+ getDescriptor() generates (Result retval, EffectDescriptor descriptor);
+
+ /**
+ * Set up required transports for passing audio buffers to the effect.
+ *
+ * The transport consists of shared memory and a message queue for reporting
+ * effect processing operation status. The shared memory is set up
+ * separately using 'setProcessBuffers' method.
+ *
+ * Processing is requested by setting 'REQUEST_PROCESS' or
+ * 'REQUEST_PROCESS_REVERSE' EventFlags associated with the status message
+ * queue. The result of processing may be one of the following:
+ * OK if there were no errors during processing;
+ * INVALID_ARGUMENTS if audio buffers are invalid;
+ * INVALID_STATE if the engine has finished the disable phase;
+ * NOT_INITIALIZED if the audio buffers were not set;
+ * NOT_SUPPORTED if the requested processing type is not supported by
+ * the effect.
+ *
+ * @return retval OK if both message queues were created successfully.
+ * INVALID_STATE if the method was already called.
+ * INVALID_ARGUMENTS if there was a problem setting up
+ * the queue.
+ * @return statusMQ a message queue used for passing status from the effect.
+ */
+ @callflow(next={"setProcessBuffers"})
+ prepareForProcessing() generates (Result retval, fmq_sync<Result> statusMQ);
+
+ /**
+ * Set up input and output buffers for processing audio data. The effect
+ * may modify both the input and the output buffer during the operation.
+ * Buffers may be set multiple times during effect lifetime.
+ *
+ * The input and the output buffer may be reused between different effects,
+ * and the input buffer may be used as an output buffer. Buffers are
+ * distinguished using 'AudioBuffer.id' field.
+ *
+ * @param inBuffer input audio buffer.
+ * @param outBuffer output audio buffer.
+ * @return retval OK if both buffers were mapped successfully.
+ * INVALID_ARGUMENTS if there was a problem with mapping
+ * any of the buffers.
+ */
+ setProcessBuffers(AudioBuffer inBuffer, AudioBuffer outBuffer)
+ generates (Result retval);
+
+ /**
+ * Execute a vendor specific command on the effect. The command code
+ * and data, as well as result data are not interpreted by Android
+ * Framework and are passed as-is between the application and the effect.
+ *
+ * The effect must use standard POSIX.1-2001 error codes for the operation
+ * completion status.
+ *
+ * Use this method only if the effect is provided by a third party, and
+ * there is no interface defined for it. This method only works for effects
+ * implemented in software.
+ *
+ * @param commandId the ID of the command.
+ * @param data command data.
+ * @param resultMaxSize maximum size in bytes of the result; can be 0.
+ * @return status command completion status.
+ * @return result result data.
+ */
+ command(uint32_t commandId, vec<uint8_t> data, uint32_t resultMaxSize)
+ generates (int32_t status, vec<uint8_t> result);
+
+ /**
+ * Set a vendor-specific parameter and apply it immediately. The parameter
+ * code and data are not interpreted by Android Framework and are passed
+ * as-is between the application and the effect.
+ *
+ * The effect must use INVALID_ARGUMENTS return code if the parameter ID is
+ * unknown or if provided parameter data is invalid. If the effect does not
+ * support setting vendor-specific parameters, it must return NOT_SUPPORTED.
+ *
+ * Use this method only if the effect is provided by a third party, and
+ * there is no interface defined for it. This method only works for effects
+ * implemented in software.
+ *
+ * @param parameter identifying data of the parameter.
+ * @param value the value of the parameter.
+ * @return retval operation completion status.
+ */
+ setParameter(vec<uint8_t> parameter, vec<uint8_t> value)
+ generates (Result retval);
+
+ /**
+ * Get a vendor-specific parameter value. The parameter code and returned
+ * data are not interpreted by Android Framework and are passed as-is
+ * between the application and the effect.
+ *
+ * The effect must use INVALID_ARGUMENTS return code if the parameter ID is
+ * unknown. If the effect does not support setting vendor-specific
+ * parameters, it must return NOT_SUPPORTED.
+ *
+ * Use this method only if the effect is provided by a third party, and
+ * there is no interface defined for it. This method only works for effects
+ * implemented in software.
+ *
+ * @param parameter identifying data of the parameter.
+ * @param valueMaxSize maximum size in bytes of the value.
+ * @return retval operation completion status.
+ * @return result the value of the parameter.
+ */
+ getParameter(vec<uint8_t> parameter, uint32_t valueMaxSize)
+ generates (Result retval, vec<uint8_t> value);
+
+ /**
+ * Get supported configs for a vendor-specific feature. The configs returned
+ * are not interpreted by Android Framework and are passed as-is between the
+ * application and the effect.
+ *
+ * The effect must use INVALID_ARGUMENTS return code if the feature ID is
+ * unknown. If the effect does not support getting vendor-specific feature
+ * configs, it must return NOT_SUPPORTED. If the feature is supported but
+ * the total number of supported configurations exceeds the maximum number
+ * indicated by the caller, the method must return RESULT_TOO_BIG.
+ *
+ * Use this method only if the effect is provided by a third party, and
+ * there is no interface defined for it. This method only works for effects
+ * implemented in software.
+ *
+ * @param featureId feature identifier.
+ * @param maxConfigs maximum number of configs to return.
+ * @param configSize size of each config in bytes.
+ * @return retval operation completion status.
+ * @return configsCount number of configs returned.
+ * @return configsData data for all the configs returned.
+ */
+ getSupportedConfigsForFeature(
+ uint32_t featureId,
+ uint32_t maxConfigs,
+ uint32_t configSize) generates (
+ Result retval,
+ uint32_t configsCount,
+ vec<uint8_t> configsData);
+
+ /**
+ * Get the current config for a vendor-specific feature. The config returned
+ * is not interpreted by Android Framework and is passed as-is between the
+ * application and the effect.
+ *
+ * The effect must use INVALID_ARGUMENTS return code if the feature ID is
+ * unknown. If the effect does not support getting vendor-specific
+ * feature configs, it must return NOT_SUPPORTED.
+ *
+ * Use this method only if the effect is provided by a third party, and
+ * there is no interface defined for it. This method only works for effects
+ * implemented in software.
+ *
+ * @param featureId feature identifier.
+ * @param configSize size of the config in bytes.
+ * @return retval operation completion status.
+ * @return configData config data.
+ */
+ getCurrentConfigForFeature(uint32_t featureId, uint32_t configSize)
+ generates (Result retval, vec<uint8_t> configData);
+
+ /**
+ * Set the current config for a vendor-specific feature. The config data
+ * is not interpreted by Android Framework and is passed as-is between the
+ * application and the effect.
+ *
+ * The effect must use INVALID_ARGUMENTS return code if the feature ID is
+ * unknown. If the effect does not support getting vendor-specific
+ * feature configs, it must return NOT_SUPPORTED.
+ *
+ * Use this method only if the effect is provided by a third party, and
+ * there is no interface defined for it. This method only works for effects
+ * implemented in software.
+ *
+ * @param featureId feature identifier.
+ * @param configData config data.
+ * @return retval operation completion status.
+ */
+ setCurrentConfigForFeature(uint32_t featureId, vec<uint8_t> configData)
+ generates (Result retval);
+
+ /**
+ * Called by the framework to deinitialize the effect and free up
+ * all currently allocated resources. It is recommended to close
+ * the effect on the client side as soon as it is becomes unused.
+ *
+ * The client must ensure that this function is not called while
+ * audio data is being transferred through the effect's message queues.
+ *
+ * @return retval OK in case the success.
+ * INVALID_STATE if the effect was already closed.
+ */
+ @exit
+ close() generates (Result retval);
+};
diff --git a/audio/effect/7.0/IEffectBufferProviderCallback.hal b/audio/effect/7.0/IEffectBufferProviderCallback.hal
new file mode 100644
index 0000000..d18f7df
--- /dev/null
+++ b/audio/effect/7.0/IEffectBufferProviderCallback.hal
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+/**
+ * This callback interface contains functions that can be used by the effect
+ * engine 'process' function to exchange input and output audio buffers.
+ */
+interface IEffectBufferProviderCallback {
+ /**
+ * Called to retrieve a buffer where data should read from by 'process'
+ * function.
+ *
+ * @return buffer audio buffer for processing
+ */
+ getBuffer() generates (AudioBuffer buffer);
+
+ /**
+ * Called to provide a buffer with the data written by 'process' function.
+ *
+ * @param buffer audio buffer for processing
+ */
+ putBuffer(AudioBuffer buffer);
+};
diff --git a/audio/effect/7.0/IEffectsFactory.hal b/audio/effect/7.0/IEffectsFactory.hal
new file mode 100644
index 0000000..337251c
--- /dev/null
+++ b/audio/effect/7.0/IEffectsFactory.hal
@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IEffectsFactory {
+ /**
+ * Returns descriptors of different effects in all loaded libraries.
+ *
+ * @return retval operation completion status.
+ * @return result list of effect descriptors.
+ */
+ getAllDescriptors() generates(Result retval, vec<EffectDescriptor> result);
+
+ /**
+ * Returns a descriptor of a particular effect.
+ *
+ * @return retval operation completion status.
+ * @return result effect descriptor.
+ */
+ getDescriptor(Uuid uid) generates(Result retval, EffectDescriptor result);
+
+ /**
+ * Creates an effect engine of the specified type. To release the effect
+ * engine, it is necessary to release references to the returned effect
+ * object.
+ *
+ * @param uid effect uuid.
+ * @param session audio session to which this effect instance will be
+ * attached. All effects created with the same session ID
+ * are connected in series and process the same signal
+ * stream.
+ * @param ioHandle identifies the output or input stream this effect is
+ * directed to in audio HAL.
+ * @param device identifies the sink or source device this effect is directed to in the
+ * audio HAL. Must be specified if session is AudioSessionConsts.DEVICE.
+ * "device" is the AudioPortHandle used for the device when the audio
+ * patch is created at the audio HAL.
+ * @return retval operation completion status.
+ * @return result the interface for the created effect.
+ * @return effectId the unique ID of the effect to be used with
+ * IStream::addEffect and IStream::removeEffect methods.
+ */
+ createEffect(Uuid uid, AudioSession session, AudioIoHandle ioHandle, AudioPortHandle device)
+ generates (Result retval, IEffect result, uint64_t effectId);
+};
diff --git a/audio/effect/7.0/IEnvironmentalReverbEffect.hal b/audio/effect/7.0/IEnvironmentalReverbEffect.hal
new file mode 100644
index 0000000..e02cfbc
--- /dev/null
+++ b/audio/effect/7.0/IEnvironmentalReverbEffect.hal
@@ -0,0 +1,178 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IEnvironmentalReverbEffect extends IEffect {
+ /**
+ * Sets whether the effect should be bypassed.
+ */
+ setBypass(bool bypass) generates (Result retval);
+
+ /**
+ * Gets whether the effect should be bypassed.
+ */
+ getBypass() generates (Result retval, bool bypass);
+
+ enum ParamRange : int16_t {
+ ROOM_LEVEL_MIN = -6000,
+ ROOM_LEVEL_MAX = 0,
+ ROOM_HF_LEVEL_MIN = -4000,
+ ROOM_HF_LEVEL_MAX = 0,
+ DECAY_TIME_MIN = 100,
+ DECAY_TIME_MAX = 20000,
+ DECAY_HF_RATIO_MIN = 100,
+ DECAY_HF_RATIO_MAX = 1000,
+ REFLECTIONS_LEVEL_MIN = -6000,
+ REFLECTIONS_LEVEL_MAX = 0,
+ REFLECTIONS_DELAY_MIN = 0,
+ REFLECTIONS_DELAY_MAX = 65,
+ REVERB_LEVEL_MIN = -6000,
+ REVERB_LEVEL_MAX = 0,
+ REVERB_DELAY_MIN = 0,
+ REVERB_DELAY_MAX = 65,
+ DIFFUSION_MIN = 0,
+ DIFFUSION_MAX = 1000,
+ DENSITY_MIN = 0,
+ DENSITY_MAX = 1000
+ };
+
+ /**
+ * Sets the room level.
+ */
+ setRoomLevel(int16_t roomLevel) generates (Result retval);
+
+ /**
+ * Gets the room level.
+ */
+ getRoomLevel() generates (Result retval, int16_t roomLevel);
+
+ /**
+ * Sets the room high frequencies level.
+ */
+ setRoomHfLevel(int16_t roomHfLevel) generates (Result retval);
+
+ /**
+ * Gets the room high frequencies level.
+ */
+ getRoomHfLevel() generates (Result retval, int16_t roomHfLevel);
+
+ /**
+ * Sets the room decay time.
+ */
+ setDecayTime(uint32_t decayTime) generates (Result retval);
+
+ /**
+ * Gets the room decay time.
+ */
+ getDecayTime() generates (Result retval, uint32_t decayTime);
+
+ /**
+ * Sets the ratio of high frequencies decay.
+ */
+ setDecayHfRatio(int16_t decayHfRatio) generates (Result retval);
+
+ /**
+ * Gets the ratio of high frequencies decay.
+ */
+ getDecayHfRatio() generates (Result retval, int16_t decayHfRatio);
+
+ /**
+ * Sets the level of reflections in the room.
+ */
+ setReflectionsLevel(int16_t reflectionsLevel) generates (Result retval);
+
+ /**
+ * Gets the level of reflections in the room.
+ */
+ getReflectionsLevel() generates (Result retval, int16_t reflectionsLevel);
+
+ /**
+ * Sets the reflections delay in the room.
+ */
+ setReflectionsDelay(uint32_t reflectionsDelay) generates (Result retval);
+
+ /**
+ * Gets the reflections delay in the room.
+ */
+ getReflectionsDelay() generates (Result retval, uint32_t reflectionsDelay);
+
+ /**
+ * Sets the reverb level of the room.
+ */
+ setReverbLevel(int16_t reverbLevel) generates (Result retval);
+
+ /**
+ * Gets the reverb level of the room.
+ */
+ getReverbLevel() generates (Result retval, int16_t reverbLevel);
+
+ /**
+ * Sets the reverb delay of the room.
+ */
+ setReverbDelay(uint32_t reverDelay) generates (Result retval);
+
+ /**
+ * Gets the reverb delay of the room.
+ */
+ getReverbDelay() generates (Result retval, uint32_t reverbDelay);
+
+ /**
+ * Sets room diffusion.
+ */
+ setDiffusion(int16_t diffusion) generates (Result retval);
+
+ /**
+ * Gets room diffusion.
+ */
+ getDiffusion() generates (Result retval, int16_t diffusion);
+
+ /**
+ * Sets room wall density.
+ */
+ setDensity(int16_t density) generates (Result retval);
+
+ /**
+ * Gets room wall density.
+ */
+ getDensity() generates (Result retval, int16_t density);
+
+ struct AllProperties {
+ int16_t roomLevel; // in millibels, range -6000 to 0
+ int16_t roomHfLevel; // in millibels, range -4000 to 0
+ uint32_t decayTime; // in milliseconds, range 100 to 20000
+ int16_t decayHfRatio; // in permilles, range 100 to 1000
+ int16_t reflectionsLevel; // in millibels, range -6000 to 0
+ uint32_t reflectionsDelay; // in milliseconds, range 0 to 65
+ int16_t reverbLevel; // in millibels, range -6000 to 0
+ uint32_t reverbDelay; // in milliseconds, range 0 to 65
+ int16_t diffusion; // in permilles, range 0 to 1000
+ int16_t density; // in permilles, range 0 to 1000
+ };
+
+ /**
+ * Sets all properties at once.
+ */
+ setAllProperties(AllProperties properties) generates (Result retval);
+
+ /**
+ * Gets all properties at once.
+ */
+ getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/7.0/IEqualizerEffect.hal b/audio/effect/7.0/IEqualizerEffect.hal
new file mode 100644
index 0000000..e7d7ae1
--- /dev/null
+++ b/audio/effect/7.0/IEqualizerEffect.hal
@@ -0,0 +1,93 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IEqualizerEffect extends IEffect {
+ /**
+ * Gets the number of frequency bands that the equalizer supports.
+ */
+ getNumBands() generates (Result retval, uint16_t numBands);
+
+ /**
+ * Returns the minimum and maximum band levels supported.
+ */
+ getLevelRange()
+ generates (Result retval, int16_t minLevel, int16_t maxLevel);
+
+ /**
+ * Sets the gain for the given equalizer band.
+ */
+ setBandLevel(uint16_t band, int16_t level) generates (Result retval);
+
+ /**
+ * Gets the gain for the given equalizer band.
+ */
+ getBandLevel(uint16_t band) generates (Result retval, int16_t level);
+
+ /**
+ * Gets the center frequency of the given band, in milliHertz.
+ */
+ getBandCenterFrequency(uint16_t band)
+ generates (Result retval, uint32_t centerFreqmHz);
+
+ /**
+ * Gets the frequency range of the given frequency band, in milliHertz.
+ */
+ getBandFrequencyRange(uint16_t band)
+ generates (Result retval, uint32_t minFreqmHz, uint32_t maxFreqmHz);
+
+ /**
+ * Gets the band that has the most effect on the given frequency
+ * in milliHertz.
+ */
+ getBandForFrequency(uint32_t freqmHz)
+ generates (Result retval, uint16_t band);
+
+ /**
+ * Gets the names of all presets the equalizer supports.
+ */
+ getPresetNames() generates (Result retval, vec<string> names);
+
+ /**
+ * Sets the current preset using the index of the preset in the names
+ * vector returned via 'getPresetNames'.
+ */
+ setCurrentPreset(uint16_t preset) generates (Result retval);
+
+ /**
+ * Gets the current preset.
+ */
+ getCurrentPreset() generates (Result retval, uint16_t preset);
+
+ struct AllProperties {
+ uint16_t curPreset;
+ vec<int16_t> bandLevels;
+ };
+
+ /**
+ * Sets all properties at once.
+ */
+ setAllProperties(AllProperties properties) generates (Result retval);
+
+ /**
+ * Gets all properties at once.
+ */
+ getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/7.0/ILoudnessEnhancerEffect.hal b/audio/effect/7.0/ILoudnessEnhancerEffect.hal
new file mode 100644
index 0000000..0304f20
--- /dev/null
+++ b/audio/effect/7.0/ILoudnessEnhancerEffect.hal
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface ILoudnessEnhancerEffect extends IEffect {
+ /**
+ * Sets target gain expressed in millibels.
+ */
+ setTargetGain(int32_t targetGainMb) generates (Result retval);
+
+ /**
+ * Gets target gain expressed in millibels.
+ */
+ getTargetGain() generates (Result retval, int32_t targetGainMb);
+};
diff --git a/audio/effect/7.0/INoiseSuppressionEffect.hal b/audio/effect/7.0/INoiseSuppressionEffect.hal
new file mode 100644
index 0000000..2c6210c
--- /dev/null
+++ b/audio/effect/7.0/INoiseSuppressionEffect.hal
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface INoiseSuppressionEffect extends IEffect {
+ enum Level : int32_t {
+ LOW,
+ MEDIUM,
+ HIGH
+ };
+
+ /**
+ * Sets suppression level.
+ */
+ setSuppressionLevel(Level level) generates (Result retval);
+
+ /**
+ * Gets suppression level.
+ */
+ getSuppressionLevel() generates (Result retval, Level level);
+
+ enum Type : int32_t {
+ SINGLE_CHANNEL,
+ MULTI_CHANNEL
+ };
+
+ /**
+ * Set suppression type.
+ */
+ setSuppressionType(Type type) generates (Result retval);
+
+ /**
+ * Get suppression type.
+ */
+ getSuppressionType() generates (Result retval, Type type);
+
+ struct AllProperties {
+ Level level;
+ Type type;
+ };
+
+ /**
+ * Sets all properties at once.
+ */
+ setAllProperties(AllProperties properties) generates (Result retval);
+
+ /**
+ * Gets all properties at once.
+ */
+ getAllProperties() generates (Result retval, AllProperties properties);
+};
diff --git a/audio/effect/7.0/IPresetReverbEffect.hal b/audio/effect/7.0/IPresetReverbEffect.hal
new file mode 100644
index 0000000..da61d24
--- /dev/null
+++ b/audio/effect/7.0/IPresetReverbEffect.hal
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IPresetReverbEffect extends IEffect {
+ enum Preset : int32_t {
+ NONE, // no reverb or reflections
+ SMALLROOM, // a small room less than five meters in length
+ MEDIUMROOM, // a medium room with a length of ten meters or less
+ LARGEROOM, // a large-sized room suitable for live performances
+ MEDIUMHALL, // a medium-sized hall
+ LARGEHALL, // a large-sized hall suitable for a full orchestra
+ PLATE, // synthesis of the traditional plate reverb
+ LAST = PLATE
+ };
+
+ /**
+ * Sets the current preset.
+ */
+ setPreset(Preset preset) generates (Result retval);
+
+ /**
+ * Gets the current preset.
+ */
+ getPreset() generates (Result retval, Preset preset);
+};
diff --git a/audio/effect/7.0/IVirtualizerEffect.hal b/audio/effect/7.0/IVirtualizerEffect.hal
new file mode 100644
index 0000000..0e6ff54
--- /dev/null
+++ b/audio/effect/7.0/IVirtualizerEffect.hal
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IVirtualizerEffect extends IEffect {
+ /**
+ * Returns whether setting virtualization strength is supported.
+ */
+ isStrengthSupported() generates (bool strengthSupported);
+
+ enum StrengthRange : uint16_t {
+ MIN = 0,
+ MAX = 1000
+ };
+
+ /**
+ * Sets virtualization strength.
+ *
+ * @param strength strength of the effect. The valid range for strength
+ * strength is [0, 1000], where 0 per mille designates the
+ * mildest effect and 1000 per mille designates the
+ * strongest.
+ * @return retval operation completion status.
+ */
+ setStrength(uint16_t strength) generates (Result retval);
+
+ /**
+ * Gets virtualization strength.
+ */
+ getStrength() generates (Result retval, uint16_t strength);
+
+ struct SpeakerAngle {
+ /** Speaker channel mask */
+ bitfield<AudioChannelMask> mask;
+ // all angles are expressed in degrees and
+ // are relative to the listener.
+ int16_t azimuth; // 0 is the direction the listener faces
+ // 180 is behind the listener
+ // -90 is to their left
+ int16_t elevation; // 0 is the horizontal plane
+ // +90 is above the listener, -90 is below
+ };
+ /**
+ * Retrieves virtual speaker angles for the given channel mask on the
+ * specified device.
+ */
+ getVirtualSpeakerAngles(bitfield<AudioChannelMask> mask, AudioDevice device)
+ generates (Result retval, vec<SpeakerAngle> speakerAngles);
+
+ /**
+ * Forces the virtualizer effect for the given output device.
+ */
+ forceVirtualizationMode(AudioDevice device) generates (Result retval);
+
+ /**
+ * Returns audio device reflecting the current virtualization mode,
+ * AUDIO_DEVICE_NONE when not virtualizing.
+ */
+ getVirtualizationMode() generates (Result retval, AudioDevice device);
+};
diff --git a/audio/effect/7.0/IVisualizerEffect.hal b/audio/effect/7.0/IVisualizerEffect.hal
new file mode 100644
index 0000000..b4e8659
--- /dev/null
+++ b/audio/effect/7.0/IVisualizerEffect.hal
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+import IEffect;
+
+interface IVisualizerEffect extends IEffect {
+ enum CaptureSizeRange : int32_t {
+ MAX = 1024, // maximum capture size in samples
+ MIN = 128 // minimum capture size in samples
+ };
+
+ /**
+ * Sets the number PCM samples in the capture.
+ */
+ setCaptureSize(uint16_t captureSize) generates (Result retval);
+
+ /**
+ * Gets the number PCM samples in the capture.
+ */
+ getCaptureSize() generates (Result retval, uint16_t captureSize);
+
+ enum ScalingMode : int32_t {
+ // Keep in sync with SCALING_MODE_... in
+ // frameworks/base/media/java/android/media/audiofx/Visualizer.java
+ NORMALIZED = 0,
+ AS_PLAYED = 1
+ };
+
+ /**
+ * Specifies the way the captured data is scaled.
+ */
+ setScalingMode(ScalingMode scalingMode) generates (Result retval);
+
+ /**
+ * Retrieves the way the captured data is scaled.
+ */
+ getScalingMode() generates (Result retval, ScalingMode scalingMode);
+
+ /**
+ * Informs the visualizer about the downstream latency.
+ */
+ setLatency(uint32_t latencyMs) generates (Result retval);
+
+ /**
+ * Gets the downstream latency.
+ */
+ getLatency() generates (Result retval, uint32_t latencyMs);
+
+ enum MeasurementMode : int32_t {
+ // Keep in sync with MEASUREMENT_MODE_... in
+ // frameworks/base/media/java/android/media/audiofx/Visualizer.java
+ NONE = 0x0,
+ PEAK_RMS = 0x1
+ };
+
+ /**
+ * Specifies which measurements are to be made.
+ */
+ setMeasurementMode(MeasurementMode measurementMode)
+ generates (Result retval);
+
+ /**
+ * Retrieves which measurements are to be made.
+ */
+ getMeasurementMode() generates (
+ Result retval, MeasurementMode measurementMode);
+
+ /**
+ * Retrieves the latest PCM snapshot captured by the visualizer engine. The
+ * number of samples to capture is specified by 'setCaptureSize' parameter.
+ *
+ * @return retval operation completion status.
+ * @return samples samples in 8 bit unsigned format (0 = 0x80)
+ */
+ capture() generates (Result retval, vec<uint8_t> samples);
+
+ struct Measurement {
+ MeasurementMode mode; // discriminator
+ union Values {
+ struct PeakAndRms {
+ int32_t peakMb; // millibels
+ int32_t rmsMb; // millibels
+ } peakAndRms;
+ } value;
+ };
+ /**
+ * Retrieves the latest measurements. The measurements to be made
+ * are specified by 'setMeasurementMode' parameter.
+ *
+ * @return retval operation completion status.
+ * @return result measurement.
+ */
+ measure() generates (Result retval, Measurement result);
+};
diff --git a/audio/effect/7.0/types.hal b/audio/effect/7.0/types.hal
new file mode 100644
index 0000000..7f5a382
--- /dev/null
+++ b/audio/effect/7.0/types.hal
@@ -0,0 +1,301 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.audio.effect@7.0;
+
+import android.hardware.audio.common@7.0;
+
+enum Result : int32_t {
+ OK,
+ NOT_INITIALIZED,
+ INVALID_ARGUMENTS,
+ INVALID_STATE,
+ NOT_SUPPORTED,
+ RESULT_TOO_BIG
+};
+
+/**
+ * Effect engine capabilities/requirements flags.
+ *
+ * Definitions for flags field of effect descriptor.
+ *
+ * +----------------+--------+--------------------------------------------------
+ * | description | bits | values
+ * +----------------+--------+--------------------------------------------------
+ * | connection | 0..2 | 0 insert: after track process
+ * | mode | | 1 auxiliary: connect to track auxiliary
+ * | | | output and use send level
+ * | | | 2 replace: replaces track process function;
+ * | | | must implement SRC, volume and mono to stereo.
+ * | | | 3 pre processing: applied below audio HAL on in
+ * | | | 4 post processing: applied below audio HAL on out
+ * | | | 5 - 7 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | insertion | 3..5 | 0 none
+ * | preference | | 1 first of the chain
+ * | | | 2 last of the chain
+ * | | | 3 exclusive (only effect in the insert chain)
+ * | | | 4..7 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Volume | 6..8 | 0 none
+ * | management | | 1 implements volume control
+ * | | | 2 requires volume indication
+ * | | | 3 monitors requested volume
+ * | | | 4 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Device | 9..11 | 0 none
+ * | indication | | 1 requires device updates
+ * | | | 2, 4 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Sample input | 12..13 | 1 direct: process() function or
+ * | mode | | EFFECT_CMD_SET_CONFIG command must specify
+ * | | | a buffer descriptor
+ * | | | 2 provider: process() function uses the
+ * | | | bufferProvider indicated by the
+ * | | | EFFECT_CMD_SET_CONFIG command to request input.
+ * | | | buffers.
+ * | | | 3 both: both input modes are supported
+ * +----------------+--------+--------------------------------------------------
+ * | Sample output | 14..15 | 1 direct: process() function or
+ * | mode | | EFFECT_CMD_SET_CONFIG command must specify
+ * | | | a buffer descriptor
+ * | | | 2 provider: process() function uses the
+ * | | | bufferProvider indicated by the
+ * | | | EFFECT_CMD_SET_CONFIG command to request output
+ * | | | buffers.
+ * | | | 3 both: both output modes are supported
+ * +----------------+--------+--------------------------------------------------
+ * | Hardware | 16..17 | 0 No hardware acceleration
+ * | acceleration | | 1 non tunneled hw acceleration: the process()
+ * | | | function reads the samples, send them to HW
+ * | | | accelerated effect processor, reads back
+ * | | | the processed samples and returns them
+ * | | | to the output buffer.
+ * | | | 2 tunneled hw acceleration: the process()
+ * | | | function is transparent. The effect interface
+ * | | | is only used to control the effect engine.
+ * | | | This mode is relevant for global effects
+ * | | | actually applied by the audio hardware on
+ * | | | the output stream.
+ * +----------------+--------+--------------------------------------------------
+ * | Audio Mode | 18..19 | 0 none
+ * | indication | | 1 requires audio mode updates
+ * | | | 2..3 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Audio source | 20..21 | 0 none
+ * | indication | | 1 requires audio source updates
+ * | | | 2..3 reserved
+ * +----------------+--------+--------------------------------------------------
+ * | Effect offload | 22 | 0 The effect cannot be offloaded to an audio DSP
+ * | supported | | 1 The effect can be offloaded to an audio DSP
+ * +----------------+--------+--------------------------------------------------
+ * | Process | 23 | 0 The effect implements a process function.
+ * | function | | 1 The effect does not implement a process
+ * | not | | function: enabling the effect has no impact
+ * | implemented | | on latency or CPU load.
+ * | | | Effect implementations setting this flag do not
+ * | | | have to implement a process function.
+ * +----------------+--------+--------------------------------------------------
+ */
+@export(name="", value_prefix="EFFECT_FLAG_")
+enum EffectFlags : int32_t {
+ // Insert mode
+ TYPE_SHIFT = 0,
+ TYPE_SIZE = 3,
+ TYPE_MASK = ((1 << TYPE_SIZE) -1) << TYPE_SHIFT,
+ TYPE_INSERT = 0 << TYPE_SHIFT,
+ TYPE_AUXILIARY = 1 << TYPE_SHIFT,
+ TYPE_REPLACE = 2 << TYPE_SHIFT,
+ TYPE_PRE_PROC = 3 << TYPE_SHIFT,
+ TYPE_POST_PROC = 4 << TYPE_SHIFT,
+
+ // Insert preference
+ INSERT_SHIFT = TYPE_SHIFT + TYPE_SIZE,
+ INSERT_SIZE = 3,
+ INSERT_MASK = ((1 << INSERT_SIZE) -1) << INSERT_SHIFT,
+ INSERT_ANY = 0 << INSERT_SHIFT,
+ INSERT_FIRST = 1 << INSERT_SHIFT,
+ INSERT_LAST = 2 << INSERT_SHIFT,
+ INSERT_EXCLUSIVE = 3 << INSERT_SHIFT,
+
+ // Volume control
+ VOLUME_SHIFT = INSERT_SHIFT + INSERT_SIZE,
+ VOLUME_SIZE = 3,
+ VOLUME_MASK = ((1 << VOLUME_SIZE) -1) << VOLUME_SHIFT,
+ VOLUME_CTRL = 1 << VOLUME_SHIFT,
+ VOLUME_IND = 2 << VOLUME_SHIFT,
+ VOLUME_MONITOR = 3 << VOLUME_SHIFT,
+ VOLUME_NONE = 0 << VOLUME_SHIFT,
+
+ // Device indication
+ DEVICE_SHIFT = VOLUME_SHIFT + VOLUME_SIZE,
+ DEVICE_SIZE = 3,
+ DEVICE_MASK = ((1 << DEVICE_SIZE) -1) << DEVICE_SHIFT,
+ DEVICE_IND = 1 << DEVICE_SHIFT,
+ DEVICE_NONE = 0 << DEVICE_SHIFT,
+
+ // Sample input modes
+ INPUT_SHIFT = DEVICE_SHIFT + DEVICE_SIZE,
+ INPUT_SIZE = 2,
+ INPUT_MASK = ((1 << INPUT_SIZE) -1) << INPUT_SHIFT,
+ INPUT_DIRECT = 1 << INPUT_SHIFT,
+ INPUT_PROVIDER = 2 << INPUT_SHIFT,
+ INPUT_BOTH = 3 << INPUT_SHIFT,
+
+ // Sample output modes
+ OUTPUT_SHIFT = INPUT_SHIFT + INPUT_SIZE,
+ OUTPUT_SIZE = 2,
+ OUTPUT_MASK = ((1 << OUTPUT_SIZE) -1) << OUTPUT_SHIFT,
+ OUTPUT_DIRECT = 1 << OUTPUT_SHIFT,
+ OUTPUT_PROVIDER = 2 << OUTPUT_SHIFT,
+ OUTPUT_BOTH = 3 << OUTPUT_SHIFT,
+
+ // Hardware acceleration mode
+ HW_ACC_SHIFT = OUTPUT_SHIFT + OUTPUT_SIZE,
+ HW_ACC_SIZE = 2,
+ HW_ACC_MASK = ((1 << HW_ACC_SIZE) -1) << HW_ACC_SHIFT,
+ HW_ACC_SIMPLE = 1 << HW_ACC_SHIFT,
+ HW_ACC_TUNNEL = 2 << HW_ACC_SHIFT,
+
+ // Audio mode indication
+ AUDIO_MODE_SHIFT = HW_ACC_SHIFT + HW_ACC_SIZE,
+ AUDIO_MODE_SIZE = 2,
+ AUDIO_MODE_MASK = ((1 << AUDIO_MODE_SIZE) -1) << AUDIO_MODE_SHIFT,
+ AUDIO_MODE_IND = 1 << AUDIO_MODE_SHIFT,
+ AUDIO_MODE_NONE = 0 << AUDIO_MODE_SHIFT,
+
+ // Audio source indication
+ AUDIO_SOURCE_SHIFT = AUDIO_MODE_SHIFT + AUDIO_MODE_SIZE,
+ AUDIO_SOURCE_SIZE = 2,
+ AUDIO_SOURCE_MASK = ((1 << AUDIO_SOURCE_SIZE) -1) << AUDIO_SOURCE_SHIFT,
+ AUDIO_SOURCE_IND = 1 << AUDIO_SOURCE_SHIFT,
+ AUDIO_SOURCE_NONE = 0 << AUDIO_SOURCE_SHIFT,
+
+ // Effect offload indication
+ OFFLOAD_SHIFT = AUDIO_SOURCE_SHIFT + AUDIO_SOURCE_SIZE,
+ OFFLOAD_SIZE = 1,
+ OFFLOAD_MASK = ((1 << OFFLOAD_SIZE) -1) << OFFLOAD_SHIFT,
+ OFFLOAD_SUPPORTED = 1 << OFFLOAD_SHIFT,
+
+ // Effect has no process indication
+ NO_PROCESS_SHIFT = OFFLOAD_SHIFT + OFFLOAD_SIZE,
+ NO_PROCESS_SIZE = 1,
+ NO_PROCESS_MASK = ((1 << NO_PROCESS_SIZE) -1) << NO_PROCESS_SHIFT,
+ NO_PROCESS = 1 << NO_PROCESS_SHIFT
+};
+
+/**
+ * The effect descriptor contains necessary information to facilitate the
+ * enumeration of the effect engines present in a library.
+ */
+struct EffectDescriptor {
+ Uuid type; // UUID of to the OpenSL ES interface implemented
+ // by this effect
+ Uuid uuid; // UUID for this particular implementation
+ bitfield<EffectFlags> flags; // effect engine capabilities/requirements flags
+ uint16_t cpuLoad; // CPU load indication expressed in 0.1 MIPS units
+ // as estimated on an ARM9E core (ARMv5TE) with 0 WS
+ uint16_t memoryUsage; // data memory usage expressed in KB and includes
+ // only dynamically allocated memory
+ uint8_t[64] name; // human readable effect name
+ uint8_t[64] implementor; // human readable effect implementor name
+};
+
+/**
+ * A buffer is a chunk of audio data for processing. Multi-channel audio is
+ * always interleaved. The channel order is from LSB to MSB with regard to the
+ * channel mask definition in audio.h, audio_channel_mask_t, e.g.:
+ * Stereo: L, R; 5.1: FL, FR, FC, LFE, BL, BR.
+ *
+ * The buffer size is expressed in frame count, a frame being composed of
+ * samples for all channels at a given time. Frame size for unspecified format
+ * (AUDIO_FORMAT_OTHER) is 8 bit by definition.
+ */
+struct AudioBuffer {
+ uint64_t id;
+ uint32_t frameCount;
+ memory data;
+};
+
+@export(name="effect_buffer_access_e", value_prefix="EFFECT_BUFFER_")
+enum EffectBufferAccess : int32_t {
+ ACCESS_WRITE,
+ ACCESS_READ,
+ ACCESS_ACCUMULATE
+};
+
+/**
+ * Determines what fields of EffectBufferConfig need to be considered.
+ */
+@export(name="", value_prefix="EFFECT_CONFIG_")
+enum EffectConfigParameters : int32_t {
+ BUFFER = 0x0001, // buffer field
+ SMP_RATE = 0x0002, // samplingRate
+ CHANNELS = 0x0004, // channels
+ FORMAT = 0x0008, // format
+ ACC_MODE = 0x0010, // accessMode
+ // Note that the 2.0 ALL have been moved to an helper function
+};
+
+/**
+ * The buffer config structure specifies the input or output audio format
+ * to be used by the effect engine.
+ */
+struct EffectBufferConfig {
+ AudioBuffer buffer;
+ uint32_t samplingRateHz;
+ bitfield<AudioChannelMask> channels;
+ AudioFormat format;
+ EffectBufferAccess accessMode;
+ bitfield<EffectConfigParameters> mask;
+};
+
+struct EffectConfig {
+ EffectBufferConfig inputCfg;
+ EffectBufferConfig outputCfg;
+};
+
+@export(name="effect_feature_e", value_prefix="EFFECT_FEATURE_")
+enum EffectFeature : int32_t {
+ AUX_CHANNELS, // supports auxiliary channels
+ // (e.g. dual mic noise suppressor)
+ CNT
+};
+
+struct EffectAuxChannelsConfig {
+ bitfield<AudioChannelMask> mainChannels; // channel mask for main channels
+ bitfield<AudioChannelMask> auxChannels; // channel mask for auxiliary channels
+};
+
+struct EffectOffloadParameter {
+ bool isOffload; // true if the playback thread the effect
+ // is attached to is offloaded
+ AudioIoHandle ioHandle; // io handle of the playback thread
+ // the effect is attached to
+};
+
+/**
+ * The message queue flags used to synchronize reads and writes from
+ * the status message queue used by effects.
+ */
+enum MessageQueueFlagBits : uint32_t {
+ DONE_PROCESSING = 1 << 0,
+ REQUEST_PROCESS = 1 << 1,
+ REQUEST_PROCESS_REVERSE = 1 << 2,
+ REQUEST_QUIT = 1 << 3,
+ REQUEST_PROCESS_ALL =
+ REQUEST_PROCESS | REQUEST_PROCESS_REVERSE | REQUEST_QUIT
+};
diff --git a/audio/effect/7.0/xml/Android.bp b/audio/effect/7.0/xml/Android.bp
new file mode 100644
index 0000000..dc12e63
--- /dev/null
+++ b/audio/effect/7.0/xml/Android.bp
@@ -0,0 +1,5 @@
+xsd_config {
+ name: "audio_effects_conf_V7_0",
+ srcs: ["audio_effects_conf.xsd"],
+ package_name: "audio.effects.V7_0",
+}
diff --git a/audio/effect/7.0/xml/api/current.txt b/audio/effect/7.0/xml/api/current.txt
new file mode 100644
index 0000000..34cb541
--- /dev/null
+++ b/audio/effect/7.0/xml/api/current.txt
@@ -0,0 +1,208 @@
+// Signature format: 2.0
+package audio.effects.V7_0 {
+
+ public class AudioEffectsConf {
+ ctor public AudioEffectsConf();
+ method public audio.effects.V7_0.AudioEffectsConf.DeviceEffects getDeviceEffects();
+ method public audio.effects.V7_0.EffectsType getEffects();
+ method public audio.effects.V7_0.LibrariesType getLibraries();
+ method public audio.effects.V7_0.AudioEffectsConf.Postprocess getPostprocess();
+ method public audio.effects.V7_0.AudioEffectsConf.Preprocess getPreprocess();
+ method public audio.effects.V7_0.VersionType getVersion();
+ method public void setDeviceEffects(audio.effects.V7_0.AudioEffectsConf.DeviceEffects);
+ method public void setEffects(audio.effects.V7_0.EffectsType);
+ method public void setLibraries(audio.effects.V7_0.LibrariesType);
+ method public void setPostprocess(audio.effects.V7_0.AudioEffectsConf.Postprocess);
+ method public void setPreprocess(audio.effects.V7_0.AudioEffectsConf.Preprocess);
+ method public void setVersion(audio.effects.V7_0.VersionType);
+ }
+
+ public static class AudioEffectsConf.DeviceEffects {
+ ctor public AudioEffectsConf.DeviceEffects();
+ method public java.util.List<audio.effects.V7_0.DeviceProcessType> getDevicePort();
+ }
+
+ public static class AudioEffectsConf.Postprocess {
+ ctor public AudioEffectsConf.Postprocess();
+ method public java.util.List<audio.effects.V7_0.StreamPostprocessType> getStream();
+ }
+
+ public static class AudioEffectsConf.Preprocess {
+ ctor public AudioEffectsConf.Preprocess();
+ method public java.util.List<audio.effects.V7_0.StreamPreprocessType> getStream();
+ }
+
+ public class DeviceProcessType extends audio.effects.V7_0.StreamProcessingType {
+ ctor public DeviceProcessType();
+ method public String getAddress();
+ method public audio.effects.V7_0.DeviceType getType();
+ method public void setAddress(String);
+ method public void setType(audio.effects.V7_0.DeviceType);
+ }
+
+ public enum DeviceType {
+ method public String getRawName();
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_AUX_DIGITAL;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_BACK_MIC;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_BLUETOOTH_BLE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_BUILTIN_MIC;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_BUS;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_COMMUNICATION;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_ECHO_REFERENCE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_FM_TUNER;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_HDMI;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_HDMI_ARC;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_IP;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_LINE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_LOOPBACK;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_PROXY;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_SPDIF;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_TELEPHONY_RX;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_TV_TUNER;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_USB_ACCESSORY;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_USB_DEVICE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_USB_HEADSET;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_VOICE_CALL;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_IN_WIRED_HEADSET;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_AUX_LINE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_BUS;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_EARPIECE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_ECHO_CANCELLER;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_FM;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_HDMI;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_HDMI_ARC;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_HEARING_AID;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_IP;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_LINE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_PROXY;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_SPDIF;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_SPEAKER;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_TELEPHONY_TX;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_USB_DEVICE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_USB_HEADSET;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ enum_constant public static final audio.effects.V7_0.DeviceType AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+
+ public class EffectImplType {
+ ctor public EffectImplType();
+ method public String getLibrary();
+ method public String getUuid();
+ method public void setLibrary(String);
+ method public void setUuid(String);
+ }
+
+ public class EffectProxyType extends audio.effects.V7_0.EffectType {
+ ctor public EffectProxyType();
+ method public audio.effects.V7_0.EffectImplType getLibhw();
+ method public audio.effects.V7_0.EffectImplType getLibsw();
+ method public void setLibhw(audio.effects.V7_0.EffectImplType);
+ method public void setLibsw(audio.effects.V7_0.EffectImplType);
+ }
+
+ public class EffectType extends audio.effects.V7_0.EffectImplType {
+ ctor public EffectType();
+ method public String getName();
+ method public void setName(String);
+ }
+
+ public class EffectsType {
+ ctor public EffectsType();
+ method public java.util.List<audio.effects.V7_0.EffectProxyType> getEffectProxy_optional();
+ method public java.util.List<audio.effects.V7_0.EffectType> getEffect_optional();
+ }
+
+ public class LibrariesType {
+ ctor public LibrariesType();
+ method public java.util.List<audio.effects.V7_0.LibrariesType.Library> getLibrary();
+ }
+
+ public static class LibrariesType.Library {
+ ctor public LibrariesType.Library();
+ method public String getName();
+ method public String getPath();
+ method public void setName(String);
+ method public void setPath(String);
+ }
+
+ public enum StreamInputType {
+ method public String getRawName();
+ enum_constant public static final audio.effects.V7_0.StreamInputType camcorder;
+ enum_constant public static final audio.effects.V7_0.StreamInputType echo_reference;
+ enum_constant public static final audio.effects.V7_0.StreamInputType fm_tuner;
+ enum_constant public static final audio.effects.V7_0.StreamInputType mic;
+ enum_constant public static final audio.effects.V7_0.StreamInputType unprocessed;
+ enum_constant public static final audio.effects.V7_0.StreamInputType voice_call;
+ enum_constant public static final audio.effects.V7_0.StreamInputType voice_communication;
+ enum_constant public static final audio.effects.V7_0.StreamInputType voice_downlink;
+ enum_constant public static final audio.effects.V7_0.StreamInputType voice_performance;
+ enum_constant public static final audio.effects.V7_0.StreamInputType voice_recognition;
+ enum_constant public static final audio.effects.V7_0.StreamInputType voice_uplink;
+ }
+
+ public enum StreamOutputType {
+ method public String getRawName();
+ enum_constant public static final audio.effects.V7_0.StreamOutputType alarm;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType assistant;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType bluetooth_sco;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType dtmf;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType enforced_audible;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType music;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType notification;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType ring;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType system;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType tts;
+ enum_constant public static final audio.effects.V7_0.StreamOutputType voice_call;
+ }
+
+ public class StreamPostprocessType extends audio.effects.V7_0.StreamProcessingType {
+ ctor public StreamPostprocessType();
+ method public audio.effects.V7_0.StreamOutputType getType();
+ method public void setType(audio.effects.V7_0.StreamOutputType);
+ }
+
+ public class StreamPreprocessType extends audio.effects.V7_0.StreamProcessingType {
+ ctor public StreamPreprocessType();
+ method public audio.effects.V7_0.StreamInputType getType();
+ method public void setType(audio.effects.V7_0.StreamInputType);
+ }
+
+ public class StreamProcessingType {
+ ctor public StreamProcessingType();
+ method public java.util.List<audio.effects.V7_0.StreamProcessingType.Apply> getApply();
+ }
+
+ public static class StreamProcessingType.Apply {
+ ctor public StreamProcessingType.Apply();
+ method public String getEffect();
+ method public void setEffect(String);
+ }
+
+ public enum VersionType {
+ method public String getRawName();
+ enum_constant public static final audio.effects.V7_0.VersionType _2_0;
+ }
+
+ public class XmlParser {
+ ctor public XmlParser();
+ method public static audio.effects.V7_0.AudioEffectsConf read(java.io.InputStream) throws javax.xml.datatype.DatatypeConfigurationException, java.io.IOException, org.xmlpull.v1.XmlPullParserException;
+ method public static String readText(org.xmlpull.v1.XmlPullParser) throws java.io.IOException, org.xmlpull.v1.XmlPullParserException;
+ method public static void skip(org.xmlpull.v1.XmlPullParser) throws java.io.IOException, org.xmlpull.v1.XmlPullParserException;
+ }
+
+}
+
diff --git a/audio/effect/7.0/xml/api/last_current.txt b/audio/effect/7.0/xml/api/last_current.txt
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/audio/effect/7.0/xml/api/last_current.txt
diff --git a/audio/effect/7.0/xml/api/last_removed.txt b/audio/effect/7.0/xml/api/last_removed.txt
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/audio/effect/7.0/xml/api/last_removed.txt
diff --git a/audio/effect/7.0/xml/api/removed.txt b/audio/effect/7.0/xml/api/removed.txt
new file mode 100644
index 0000000..d802177
--- /dev/null
+++ b/audio/effect/7.0/xml/api/removed.txt
@@ -0,0 +1 @@
+// Signature format: 2.0
diff --git a/audio/effect/7.0/xml/audio_effects_conf.xsd b/audio/effect/7.0/xml/audio_effects_conf.xsd
new file mode 100644
index 0000000..94f9f76
--- /dev/null
+++ b/audio/effect/7.0/xml/audio_effects_conf.xsd
@@ -0,0 +1,323 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+ http://www.apache.org/licenses/LICENSE-2.0
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<xs:schema xmlns:xs="http://www.w3.org/2001/XMLSchema"
+ targetNamespace="http://schemas.android.com/audio/audio_effects_conf/v2_0"
+ xmlns:aec="http://schemas.android.com/audio/audio_effects_conf/v2_0"
+ elementFormDefault="qualified">
+ <!-- Simple types -->
+ <xs:simpleType name="versionType">
+ <xs:restriction base="xs:decimal">
+ <xs:enumeration value="2.0"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="uuidType">
+ <xs:restriction base="xs:string">
+ <xs:pattern value="[0-9A-Fa-f]{8}-[0-9A-Fa-f]{4}-[0-9A-Fa-f]{4}-[0-9A-Fa-f]{4}-[0-9A-Fa-f]{12}"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="streamInputType">
+ <xs:restriction base="xs:string">
+ <xs:enumeration value="mic"/>
+ <xs:enumeration value="voice_uplink"/>
+ <xs:enumeration value="voice_downlink"/>
+ <xs:enumeration value="voice_call"/>
+ <xs:enumeration value="camcorder"/>
+ <xs:enumeration value="voice_recognition"/>
+ <xs:enumeration value="voice_communication"/>
+ <xs:enumeration value="unprocessed"/>
+ <xs:enumeration value="voice_performance"/>
+ <xs:enumeration value="echo_reference"/>
+ <xs:enumeration value="fm_tuner"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="streamOutputType">
+ <xs:restriction base="xs:string">
+ <xs:enumeration value="voice_call"/>
+ <xs:enumeration value="system"/>
+ <xs:enumeration value="ring"/>
+ <xs:enumeration value="music"/>
+ <xs:enumeration value="alarm"/>
+ <xs:enumeration value="notification"/>
+ <xs:enumeration value="bluetooth_sco"/>
+ <xs:enumeration value="enforced_audible"/>
+ <xs:enumeration value="dtmf"/>
+ <xs:enumeration value="tts"/>
+ <xs:enumeration value="assistant"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="relativePathType">
+ <xs:restriction base="xs:string">
+ <xs:pattern value="[^/].*"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <xs:simpleType name="deviceType">
+ <xs:restriction base="xs:string">
+ <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_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"/>
+ <!-- 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_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_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_HDMI_ARC"/>
+ <xs:enumeration value="AUDIO_DEVICE_IN_ECHO_REFERENCE"/>
+ </xs:restriction>
+ </xs:simpleType>
+ <!-- Complex types -->
+ <xs:complexType name="librariesType">
+ <xs:annotation>
+ <xs:documentation xml:lang="en">
+ List of effect libraries to load. Each library element must have "name" and
+ "path" attributes. The latter is giving the path of the library .so file
+ relative to the standard effect folders: /(vendor|odm|system)/lib(64)?/soundfx/
+ Example for a library in "/vendor/lib/soundfx/lib.so":
+ <library name="name" path="lib.so"/>
+ </xs:documentation>
+ </xs:annotation>
+ <xs:sequence>
+ <xs:element name="library" minOccurs="0" maxOccurs="unbounded">
+ <xs:complexType>
+ <xs:attribute name="name" type="xs:string" use="required"/>
+ <xs:attribute name="path" type="aec:relativePathType" use="required"/>
+ </xs:complexType>
+ </xs:element>
+ </xs:sequence>
+ </xs:complexType>
+ <xs:complexType name="effectImplType">
+ <xs:attribute name="library" type="xs:string" use="required"/>
+ <xs:attribute name="uuid" type="aec:uuidType" use="required"/>
+ </xs:complexType>
+ <xs:complexType name="effectType">
+ <xs:complexContent>
+ <xs:extension base="aec:effectImplType">
+ <xs:attribute name="name" type="xs:string" use="required"/>
+ </xs:extension>
+ </xs:complexContent>
+ </xs:complexType>
+ <xs:complexType name="effectProxyType">
+ <xs:complexContent>
+ <xs:extension base="aec:effectType">
+ <xs:sequence>
+ <xs:element name="libsw" type="aec:effectImplType"/>
+ <xs:element name="libhw" type="aec:effectImplType"/>
+ </xs:sequence>
+ </xs:extension>
+ </xs:complexContent>
+ </xs:complexType>
+ <xs:complexType name="effectsType">
+ <xs:annotation>
+ <xs:documentation xml:lang="en">
+ List of effects to load. Each effect element must contain "name",
+ "library", and "uuid" attrs. The value of the "library" attr must
+ correspond to the name of a "library" element. The name of the effect
+ element is indicative, only the value of the "uuid" element designates
+ the effect for the audio framework. The uuid is the implementation
+ specific UUID as specified by the effect vendor. This is not the generic
+ effect type UUID.
+ For effect proxy implementations, SW and HW implementations of the effect
+ can be specified.
+ Example:
+ <effect name="name" library="lib" uuid="uuuu"/>
+ <effectProxy name="proxied" library="proxy" uuid="xxxx">
+ <libsw library="sw_bundle" uuid="yyyy"/>
+ <libhw library="offload_bundle" uuid="zzzz"/>
+ </effectProxy>
+ </xs:documentation>
+ </xs:annotation>
+ <xs:choice maxOccurs="unbounded">
+ <xs:element name="effect" type="aec:effectType" minOccurs="0" maxOccurs="unbounded"/>
+ <xs:element name="effectProxy" type="aec:effectProxyType" minOccurs="0" maxOccurs="unbounded"/>
+ </xs:choice>
+ </xs:complexType>
+ <xs:complexType name="streamProcessingType">
+ <xs:sequence>
+ <xs:element name="apply" minOccurs="0" maxOccurs="unbounded">
+ <xs:complexType>
+ <xs:attribute name="effect" type="xs:string" use="required"/>
+ </xs:complexType>
+ </xs:element>
+ </xs:sequence>
+ </xs:complexType>
+ <xs:complexType name="streamPreprocessType">
+ <xs:annotation>
+ <xs:documentation xml:lang="en">
+ Audio preprocessing configuration. The processing configuration consists
+ of a list of elements each describing processing settings for a given
+ input stream. Valid input stream types are listed in "streamInputType".
+ Each stream element contains a list of "apply" elements. The value of the
+ "effect" attr must correspond to the name of an "effect" element.
+ Example:
+ <stream type="voice_communication">
+ <apply effect="effect1"/>
+ <apply effect="effect2"/>
+ </stream>
+ </xs:documentation>
+ </xs:annotation>
+ <xs:complexContent>
+ <xs:extension base="aec:streamProcessingType">
+ <xs:attribute name="type" type="aec:streamInputType" use="required"/>
+ </xs:extension>
+ </xs:complexContent>
+ </xs:complexType>
+ <xs:complexType name="streamPostprocessType">
+ <xs:annotation>
+ <xs:documentation xml:lang="en">
+ Audio postprocessing configuration. The processing configuration consists
+ of a list of elements each describing processing settings for a given
+ output stream. Valid output stream types are listed in "streamOutputType".
+ Each stream element contains a list of "apply" elements. The value of the
+ "effect" attr must correspond to the name of an "effect" element.
+ Example:
+ <stream type="music">
+ <apply effect="effect1"/>
+ </stream>
+ </xs:documentation>
+ </xs:annotation>
+ <xs:complexContent>
+ <xs:extension base="aec:streamProcessingType">
+ <xs:attribute name="type" type="aec:streamOutputType" use="required"/>
+ </xs:extension>
+ </xs:complexContent>
+ </xs:complexType>
+ <xs:complexType name="deviceProcessType">
+ <xs:annotation>
+ <xs:documentation xml:lang="en">
+ Audio Device Effects configuration. The processing configuration consists
+ of a list of effects to be automatically added on a device Port when involved in an audio
+ patch.
+ Valid device type are listed in "deviceType" and shall be aligned.
+ Each stream element contains a list of "apply" elements. The value of the
+ "effect" attr must correspond to the name of an "effect" element.
+ Note that if the device is involved in a hardware patch, the effect must be hardware
+ accelerated.
+ Example:
+ <devicePort address="BUS00_USAGE_MAIN" type="AUDIO_DEVICE_OUT_BUS">
+ <apply effect="equalizer"/>
+ <apply effect="effect2"/>
+ </devicePort>
+ </xs:documentation>
+ </xs:annotation>
+ <xs:complexContent>
+ <xs:extension base="aec:streamProcessingType">
+ <xs:attribute name="address" type="xs:string" use="required"/>
+ <xs:attribute name="type" type="aec:deviceType" use="required"/>
+ </xs:extension>
+ </xs:complexContent>
+ </xs:complexType>
+ <!-- Root element -->
+ <xs:element name="audio_effects_conf">
+ <xs:complexType>
+ <xs:sequence>
+ <xs:element name="libraries" type="aec:librariesType"/>
+ <xs:element name="effects" type="aec:effectsType"/>
+ <xs:element name="postprocess" minOccurs="0" maxOccurs="1">
+ <xs:complexType>
+ <xs:sequence>
+ <xs:element name="stream" type="aec:streamPostprocessType" minOccurs="0" maxOccurs="unbounded"/>
+ </xs:sequence>
+ </xs:complexType>
+ </xs:element>
+ <xs:element name="preprocess" minOccurs="0" maxOccurs="1">
+ <xs:complexType>
+ <xs:sequence>
+ <xs:element name="stream" type="aec:streamPreprocessType" minOccurs="0" maxOccurs="unbounded"/>
+ </xs:sequence>
+ </xs:complexType>
+ </xs:element>
+ <xs:element name="deviceEffects" minOccurs="0" maxOccurs="1">
+ <xs:complexType>
+ <xs:sequence>
+ <xs:element name="devicePort" type="aec:deviceProcessType" minOccurs="0" maxOccurs="unbounded"/>
+ </xs:sequence>
+ </xs:complexType>
+ </xs:element>
+ </xs:sequence>
+ <xs:attribute name="version" type="aec:versionType" use="required"/>
+ </xs:complexType>
+ <!-- Keys and references -->
+ <xs:key name="libraryName">
+ <xs:selector xpath="aec:libraries/aec:library"/>
+ <xs:field xpath="@name"/>
+ </xs:key>
+ <xs:keyref name="libraryNameRef1" refer="aec:libraryName">
+ <xs:selector xpath="aec:effects/aec:effect"/>
+ <xs:field xpath="@library"/>
+ </xs:keyref>
+ <xs:keyref name="libraryNameRef2" refer="aec:libraryName">
+ <xs:selector xpath="aec:effects/aec:effect/aec:libsw"/>
+ <xs:field xpath="@library"/>
+ </xs:keyref>
+ <xs:keyref name="libraryNameRef3" refer="aec:libraryName">
+ <xs:selector xpath="aec:effects/aec:effect/aec:libhw"/>
+ <xs:field xpath="@library"/>
+ </xs:keyref>
+ <xs:key name="effectName">
+ <xs:selector xpath="aec:effects/aec:effect|aec:effects/aec:effectProxy"/>
+ <xs:field xpath="@name"/>
+ </xs:key>
+ <xs:keyref name="effectNamePreRef" refer="aec:effectName">
+ <xs:selector xpath="aec:preprocess/aec:stream/aec:apply"/>
+ <xs:field xpath="@effect"/>
+ </xs:keyref>
+ <xs:keyref name="effectNamePostRef" refer="aec:effectName">
+ <xs:selector xpath="aec:postprocess/aec:stream/aec:apply"/>
+ <xs:field xpath="@effect"/>
+ </xs:keyref>
+ </xs:element>
+</xs:schema>
diff --git a/audio/effect/all-versions/default/Android.bp b/audio/effect/all-versions/default/Android.bp
index d9bb78b..01392bd 100644
--- a/audio/effect/all-versions/default/Android.bp
+++ b/audio/effect/all-versions/default/Android.bp
@@ -103,3 +103,18 @@
"-include common/all-versions/VersionMacro.h",
]
}
+
+cc_library_shared {
+ name: "android.hardware.audio.effect@7.0-impl",
+ defaults: ["android.hardware.audio.effect-impl_default"],
+ shared_libs: [
+ "android.hardware.audio.common@7.0",
+ "android.hardware.audio.common@7.0-util",
+ "android.hardware.audio.effect@7.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
diff --git a/audio/effect/all-versions/vts/functional/Android.bp b/audio/effect/all-versions/vts/functional/Android.bp
index 309aa9d..6a1d6a3 100644
--- a/audio/effect/all-versions/vts/functional/Android.bp
+++ b/audio/effect/all-versions/vts/functional/Android.bp
@@ -113,3 +113,23 @@
"-include common/all-versions/VersionMacro.h",
]
}
+
+cc_test {
+ name: "VtsHalAudioEffectV7_0TargetTest",
+ defaults: ["VtsHalAudioEffectTargetTest_default"],
+ // Use test_config for vts suite.
+ // TODO(b/146104851): Add auto-gen rules and remove it.
+ test_config: "VtsHalAudioEffectV7_0TargetTest.xml",
+ static_libs: [
+ "android.hardware.audio.common@7.0",
+ "android.hardware.audio.effect@7.0",
+ ],
+ data: [
+ ":audio_effects_conf_V7_0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
diff --git a/audio/effect/all-versions/vts/functional/VtsHalAudioEffectV7_0TargetTest.xml b/audio/effect/all-versions/vts/functional/VtsHalAudioEffectV7_0TargetTest.xml
new file mode 100644
index 0000000..e609756
--- /dev/null
+++ b/audio/effect/all-versions/vts/functional/VtsHalAudioEffectV7_0TargetTest.xml
@@ -0,0 +1,38 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2019 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.
+-->
+<configuration description="Runs VtsHalAudioEffectV7_0TargetTest.">
+ <option name="test-suite-tag" value="apct" />
+ <option name="test-suite-tag" value="apct-native" />
+
+ <target_preparer class="com.android.tradefed.targetprep.RootTargetPreparer"/>
+ <target_preparer class="com.android.tradefed.targetprep.StopServicesSetup"/>
+
+ <target_preparer class="com.android.tradefed.targetprep.RunCommandTargetPreparer">
+ <option name="run-command" value="setprop vts.native_server.on 1"/>
+ <option name="teardown-command" value="setprop vts.native_server.on 0"/>
+ </target_preparer>
+
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="VtsHalAudioEffectV7_0TargetTest->/data/local/tmp/VtsHalAudioEffectV7_0TargetTest" />
+ <option name="push" value="audio_effects_conf_V7_0.xsd->/data/local/tmp/audio_effects_conf_V7_0.xsd" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="VtsHalAudioEffectV7_0TargetTest" />
+ </test>
+</configuration>