Convert all comments into "doxygen-ready" comments.
Bug: 36453077
Test: mma
Change-Id: I0b1f77dfae5d2258969e33d85ecf45401ffbdfaa
diff --git a/audio/2.0/IDevice.hal b/audio/2.0/IDevice.hal
index 2b5329b..62c2081 100644
--- a/audio/2.0/IDevice.hal
+++ b/audio/2.0/IDevice.hal
@@ -23,14 +23,14 @@
interface IDevice {
typedef android.hardware.audio@2.0::Result Result;
- /*
+ /**
* Returns whether the audio hardware interface has been initialized.
*
* @return retval OK on success, NOT_INITIALIZED on failure.
*/
initCheck() generates (Result retval);
- /*
+ /**
* Sets the audio volume for all audio activities other than voice call. If
* NOT_SUPPORTED is returned, the software mixer will emulate this
* capability.
@@ -40,7 +40,7 @@
*/
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
@@ -52,7 +52,7 @@
*/
getMasterVolume() generates (Result retval, float volume);
- /*
+ /**
* Sets microphone muting state.
*
* @param mute whether microphone is muted.
@@ -60,7 +60,7 @@
*/
setMicMute(bool mute) generates (Result retval);
- /*
+ /**
* Gets whether microphone is muted.
*
* @return retval operation completion status.
@@ -68,7 +68,7 @@
*/
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.
*
@@ -89,7 +89,7 @@
*/
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.
*
@@ -100,7 +100,7 @@
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.
@@ -122,7 +122,7 @@
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.
@@ -146,14 +146,14 @@
IStreamIn inStream,
AudioConfig suggestedConfig);
- /*
+ /**
* Returns whether HAL supports audio patches.
*
* @return supports true if audio patches are supported.
*/
supportsAudioPatches() generates (bool supports);
- /*
+ /**
* Creates an audio patch between several source and sink ports. The handle
* is allocated by the HAL and must be unique for this audio HAL module.
*
@@ -165,7 +165,7 @@
createAudioPatch(vec<AudioPortConfig> sources, vec<AudioPortConfig> sinks)
generates (Result retval, AudioPatchHandle patch);
- /*
+ /**
* Release an audio patch.
*
* @param patch patch handle.
@@ -173,7 +173,7 @@
*/
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...)
@@ -189,7 +189,7 @@
getAudioPort(AudioPort port)
generates (Result retval, AudioPort resultPort);
- /*
+ /**
* Set audio port configuration.
*
* @param config audio port configuration.
@@ -197,7 +197,7 @@
*/
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.
*
@@ -205,7 +205,7 @@
*/
getHwAvSync() generates (AudioHwSync hwAvSync);
- /*
+ /**
* Sets whether the screen is on. Calling this method is equivalent to
* setting AUDIO_PARAMETER_KEY_SCREEN_STATE on the legacy HAL.
*
@@ -214,7 +214,7 @@
*/
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.
@@ -226,7 +226,7 @@
getParameters(vec<string> keys)
generates (Result retval, vec<ParameterValue> parameters);
- /*
+ /**
* Generic method for setting vendor-specific parameter values.
* The framework does not interpret the parameters, they are passed
* in an opaque manner between a vendor application and HAL.
@@ -236,7 +236,7 @@
*/
setParameters(vec<ParameterValue> parameters) generates (Result retval);
- /*
+ /**
* Dumps information about the stream into the provided file descriptor.
* This is used for the dumpsys facility.
*
diff --git a/audio/2.0/IDevicesFactory.hal b/audio/2.0/IDevicesFactory.hal
index 0ef6bc5..6bbe7a1 100644
--- a/audio/2.0/IDevicesFactory.hal
+++ b/audio/2.0/IDevicesFactory.hal
@@ -30,7 +30,7 @@
STUB
};
- /*
+ /**
* Opens an audio device. To close the device, it is necessary to release
* references to the returned device object.
*
diff --git a/audio/2.0/IPrimaryDevice.hal b/audio/2.0/IPrimaryDevice.hal
index f1dd56e..adeb366 100644
--- a/audio/2.0/IPrimaryDevice.hal
+++ b/audio/2.0/IPrimaryDevice.hal
@@ -22,7 +22,7 @@
interface IPrimaryDevice extends IDevice {
typedef android.hardware.audio@2.0::Result Result;
- /*
+ /**
* Sets the audio volume of a voice call.
*
* @param volume 1.0f means unity, 0.0f is zero.
@@ -30,7 +30,7 @@
*/
setVoiceVolume(float volume) generates (Result retval);
- /*
+ /**
* This method is used to notify the HAL about audio mode changes.
*
* @param mode new mode.
@@ -38,7 +38,7 @@
*/
setMode(AudioMode mode) generates (Result retval);
- /*
+ /**
* Gets whether BT SCO Noise Reduction and Echo Cancellation are enabled.
* Calling this method is equivalent to getting AUDIO_PARAMETER_KEY_BT_NREC
* on the legacy HAL.
@@ -48,7 +48,7 @@
*/
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.
@@ -58,7 +58,7 @@
*/
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.
*
@@ -67,7 +67,7 @@
*/
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.
*
@@ -83,7 +83,7 @@
FULL
};
- /*
+ /**
* Gets current TTY mode selection. Calling this method is equivalent to
* getting AUDIO_PARAMETER_KEY_TTY_MODE on the legacy HAL.
*
@@ -92,7 +92,7 @@
*/
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.
*
@@ -101,7 +101,7 @@
*/
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.
@@ -111,7 +111,7 @@
*/
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.
diff --git a/audio/2.0/IStream.hal b/audio/2.0/IStream.hal
index 8de7851..2b9cc06 100644
--- a/audio/2.0/IStream.hal
+++ b/audio/2.0/IStream.hal
@@ -22,14 +22,14 @@
interface IStream {
typedef android.hardware.audio@2.0::Result Result;
- /*
+ /**
* Return the frame size (number of bytes per sample).
*
* @return frameSize frame size in bytes.
*/
getFrameSize() generates (uint64_t frameSize);
- /*
+ /**
* Return the frame count of the buffer. Calling this method is equivalent
* to getting AUDIO_PARAMETER_STREAM_FRAME_COUNT on the legacy HAL.
*
@@ -37,7 +37,7 @@
*/
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.
*
@@ -45,14 +45,14 @@
*/
getBufferSize() generates (uint64_t bufferSize);
- /*
+ /**
* Return the sampling rate in Hz.
*
* @return sampleRateHz sample rate in Hz.
*/
getSampleRate() generates (uint32_t sampleRateHz);
- /*
+ /**
* Return supported sampling rates of the stream. Calling this method is
* equivalent to getting AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES on the
* legacy HAL.
@@ -61,7 +61,7 @@
*/
getSupportedSampleRates() generates (vec<uint32_t> sampleRates);
- /*
+ /**
* Sets the sampling rate of the stream. Calling this method is equivalent
* to setting AUDIO_PARAMETER_STREAM_SAMPLING_RATE on the legacy HAL.
*
@@ -70,14 +70,14 @@
*/
setSampleRate(uint32_t sampleRateHz) generates (Result retval);
- /*
+ /**
* Return the channel mask of the stream.
*
* @return mask channel mask.
*/
getChannelMask() generates (AudioChannelMask mask);
- /*
+ /**
* Return supported channel masks of the stream. Calling this method is
* equivalent to getting AUDIO_PARAMETER_STREAM_SUP_CHANNELS on the legacy
* HAL.
@@ -86,7 +86,7 @@
*/
getSupportedChannelMasks() generates (vec<AudioChannelMask> masks);
- /*
+ /**
* Sets the channel mask of the stream. Calling this method is equivalent to
* setting AUDIO_PARAMETER_STREAM_CHANNELS on the legacy HAL.
*
@@ -95,14 +95,14 @@
*/
setChannelMask(AudioChannelMask mask) generates (Result retval);
- /*
+ /**
* Return the audio format of the stream.
*
* @return format audio format.
*/
getFormat() generates (AudioFormat format);
- /*
+ /**
* Return supported audio formats of the stream. Calling this method is
* equivalent to getting AUDIO_PARAMETER_STREAM_SUP_FORMATS on the legacy
* HAL.
@@ -111,7 +111,7 @@
*/
getSupportedFormats() generates (vec<AudioFormat> formats);
- /*
+ /**
* Sets the audio format of the stream. Calling this method is equivalent to
* setting AUDIO_PARAMETER_STREAM_FORMAT on the legacy HAL.
*
@@ -120,7 +120,7 @@
*/
setFormat(AudioFormat format) generates (Result retval);
- /*
+ /**
* Convenience method for retrieving several stream parameters in
* one transaction.
*
@@ -131,7 +131,7 @@
getAudioProperties() generates (
uint32_t sampleRateHz, AudioChannelMask mask, AudioFormat format);
- /*
+ /**
* Applies audio effect to the stream.
*
* @param effectId effect ID (obtained from IEffectsFactory.createEffect) of
@@ -140,7 +140,7 @@
*/
addEffect(uint64_t effectId) generates (Result retval);
- /*
+ /**
* Stops application of the effect to the stream.
*
* @param effectId effect ID (obtained from IEffectsFactory.createEffect) of
@@ -149,7 +149,7 @@
*/
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.
*
@@ -157,14 +157,14 @@
*/
standby() generates (Result retval);
- /*
+ /**
* Return the set of device(s) which this stream is connected to.
*
* @return device set of device(s) which this stream is connected to.
*/
getDevice() generates (AudioDevice device);
- /*
+ /**
* Connects the stream to the device.
*
* This method must only be used for HALs that do not support
@@ -177,7 +177,7 @@
*/
setDevice(DeviceAddress address) generates (Result retval);
- /*
+ /**
* Notifies the stream about device connection state. Calling this method is
* equivalent to setting AUDIO_PARAMETER_DEVICE_[DIS]CONNECT on the legacy
* HAL.
@@ -189,7 +189,7 @@
setConnectedState(DeviceAddress address, bool connected)
generates (Result retval);
- /*
+ /**
* Sets the HW synchronization source. Calling this method is equivalent to
* setting AUDIO_PARAMETER_STREAM_HW_AV_SYNC on the legacy HAL.
*
@@ -198,7 +198,7 @@
*/
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.
@@ -210,7 +210,7 @@
getParameters(vec<string> keys)
generates (Result retval, vec<ParameterValue> parameters);
- /*
+ /**
* Generic method for setting vendor-specific parameter values.
* The framework does not interpret the parameters, they are passed
* in an opaque manner between a vendor application and HAL.
@@ -220,7 +220,7 @@
*/
setParameters(vec<ParameterValue> parameters) generates (Result retval);
- /*
+ /**
* Dumps information about the stream into the provided file descriptor.
* This is used for the dumpsys facility.
*
@@ -228,7 +228,7 @@
*/
debugDump(handle fd);
- /*
+ /**
* Called by the framework to start a stream operating in mmap mode.
* createMmapBuffer() must be called before calling start().
* Function only implemented by streams operating in mmap mode.
@@ -249,7 +249,7 @@
*/
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.
@@ -266,7 +266,7 @@
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.
@@ -280,7 +280,7 @@
getMmapPosition()
generates (Result retval, MmapPosition position);
- /*
+ /**
* Called by the framework to deinitialize the stream and free up
* all the currently allocated resources. It is recommended to close
* the stream on the client side as soon as it is becomes unused.
diff --git a/audio/2.0/IStreamIn.hal b/audio/2.0/IStreamIn.hal
index 6f1f9df..6b79f48 100644
--- a/audio/2.0/IStreamIn.hal
+++ b/audio/2.0/IStreamIn.hal
@@ -22,7 +22,7 @@
interface IStreamIn extends IStream {
typedef android.hardware.audio@2.0::Result Result;
- /*
+ /**
* Returns the source descriptor of the input stream. Calling this method is
* equivalent to getting AUDIO_PARAMETER_STREAM_INPUT_SOURCE on the legacy
* HAL.
@@ -32,7 +32,7 @@
*/
getAudioSource() generates (Result retval, AudioSource source);
- /*
+ /**
* Set the input gain for the audio driver.
*
* @param gain 1.0f is unity, 0.0f is zero.
@@ -40,7 +40,7 @@
*/
setGain(float gain) generates (Result retval);
- /*
+ /**
* Commands that can be executed on the driver reader thread.
*/
enum ReadCommand : int32_t {
@@ -48,7 +48,7 @@
GET_CAPTURE_POSITION
};
- /*
+ /**
* Data structure passed to the driver for executing commands
* on the driver reader thread.
*/
@@ -60,7 +60,7 @@
} params;
};
- /*
+ /**
* Data structure passed back to the client via status message queue
* of 'read' operation.
*
@@ -81,7 +81,7 @@
} reply;
};
- /*
+ /**
* Set up required transports for receiving audio buffers from the driver.
*
* The transport consists of three message queues:
@@ -119,7 +119,7 @@
fmq_sync<ReadStatus> statusMQ,
ThreadInfo threadInfo);
- /*
+ /**
* Return the amount of input frames lost in the audio driver since the last
* call of this function.
*
diff --git a/audio/2.0/IStreamOut.hal b/audio/2.0/IStreamOut.hal
index 9ee32c5..84b7447 100644
--- a/audio/2.0/IStreamOut.hal
+++ b/audio/2.0/IStreamOut.hal
@@ -23,14 +23,14 @@
interface IStreamOut extends IStream {
typedef android.hardware.audio@2.0::Result Result;
- /*
+ /**
* Return the audio hardware driver estimated latency in milliseconds.
*
* @return latencyMs latency in milliseconds.
*/
getLatency() generates (uint32_t latencyMs);
- /*
+ /**
* This method is used in situations where audio mixing is done in the
* hardware. This method serves as a direct interface with hardware,
* allowing to directly set the volume as apposed to via the framework.
@@ -43,7 +43,7 @@
*/
setVolume(float left, float right) generates (Result retval);
- /*
+ /**
* Commands that can be executed on the driver writer thread.
*/
enum WriteCommand : int32_t {
@@ -52,7 +52,7 @@
GET_LATENCY
};
- /*
+ /**
* Data structure passed back to the client via status message queue
* of 'write' operation.
*
@@ -75,7 +75,7 @@
} reply;
};
- /*
+ /**
* Set up required transports for passing audio buffers to the driver.
*
* The transport consists of three message queues:
@@ -112,7 +112,7 @@
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.
*
@@ -121,7 +121,7 @@
*/
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.
@@ -131,7 +131,7 @@
*/
getNextWriteTimestamp() generates (Result retval, int64_t timestampUs);
- /*
+ /**
* Set the callback interface for notifying completion of non-blocking
* write and drain.
*
@@ -146,7 +146,7 @@
*/
setCallback(IStreamOutCallback callback) generates (Result retval);
- /*
+ /**
* Clears the callback previously set via 'setCallback' method.
*
* Warning: failure to call this method results in callback implementation
@@ -156,7 +156,7 @@
*/
clearCallback() generates (Result retval);
- /*
+ /**
* Returns whether HAL supports pausing and resuming of streams.
*
* @return supportsPause true if pausing is supported.
@@ -179,7 +179,7 @@
*/
pause() generates (Result retval);
- /*
+ /**
* Notifies to the audio driver to resume playback following a pause.
* Returns error INVALID_STATE if called without matching pause.
*
@@ -189,7 +189,7 @@
*/
resume() generates (Result retval);
- /*
+ /**
* Returns whether HAL supports draining of streams.
*
* @return supports true if draining is supported.
@@ -220,7 +220,7 @@
*/
drain(AudioDrain type) generates (Result retval);
- /*
+ /**
* Notifies to the audio driver to flush the queued data. Stream must
* already be paused before calling 'flush'.
*
@@ -230,7 +230,7 @@
*/
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
diff --git a/audio/2.0/IStreamOutCallback.hal b/audio/2.0/IStreamOutCallback.hal
index cdb38de..01e123c 100644
--- a/audio/2.0/IStreamOutCallback.hal
+++ b/audio/2.0/IStreamOutCallback.hal
@@ -16,21 +16,21 @@
package android.hardware.audio@2.0;
-/*
+/**
* 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/2.0/types.hal b/audio/2.0/types.hal
index 8e9ff14..93118c2 100644
--- a/audio/2.0/types.hal
+++ b/audio/2.0/types.hal
@@ -28,14 +28,16 @@
@export(name="audio_drain_type_t", value_prefix="AUDIO_DRAIN_")
enum AudioDrain : int32_t {
- /* drain() returns when all data has been played. */
+ /** 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. */
+ /**
+ * 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 {
@@ -43,7 +45,7 @@
uint64_t tvNSec; // nanoseconds
};
-/*
+/**
* IEEE 802 MAC address.
*/
typedef uint8_t[6] MacAddress;
@@ -53,7 +55,7 @@
string value;
};
-/*
+/**
* Specifies a device in case when several devices of the same type
* can be connected (e.g. BT A2DP, USB).
*/
@@ -71,7 +73,7 @@
string rSubmixAddress; // used for REMOTE_SUBMIX
};
-/*
+/**
* Mmap buffer descriptor returned by IStream.createMmapBuffer().
* Used by streams opened in mmap mode.
*/
@@ -81,7 +83,7 @@
int32_t burstSizeFrames; // transfer size granularity in frames
};
-/*
+/**
* Mmap buffer read/write position returned by IStream.getMmapPosition().
* Used by streams opened in mmap mode.
*/
@@ -90,7 +92,7 @@
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.
*/
diff --git a/audio/common/2.0/types.hal b/audio/common/2.0/types.hal
index dd7281d..7c49795 100644
--- a/audio/common/2.0/types.hal
+++ b/audio/common/2.0/types.hal
@@ -16,28 +16,28 @@
package android.hardware.audio.common@2.0;
-/*
+/**
*
* 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
@@ -47,12 +47,12 @@
*/
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="")
@@ -63,7 +63,7 @@
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.
*/
@@ -76,13 +76,13 @@
};
-/*
+/**
*
* Audio streams
*
*/
-/*
+/**
* Audio stream type describing the intented use case of a stream.
*/
@export(name="audio_stream_type_t", value_prefix="AUDIO_STREAM_")
@@ -126,13 +126,13 @@
CAMCORDER = 5,
VOICE_RECOGNITION = 6,
VOICE_COMMUNICATION = 7,
- /*
+ /**
* Source for the mix to be presented remotely. An example of remote
* presentation is Wifi Display where a dongle attached to a TV can be used
* to play the mix captured by this audio source.
*/
REMOTE_SUBMIX = 8,
- /*
+ /**
* Source for unprocessed sound. Usage examples include level measurement
* and raw signal analysis.
*/
@@ -141,7 +141,7 @@
CNT,
MAX = CNT - 1,
FM_TUNER = 1998,
- /*
+ /**
* A low-priority, preemptible audio source for for background software
* hotword detection. Same tuning as VOICE_RECOGNITION. Used only
* internally by the framework.
@@ -150,30 +150,30 @@
};
typedef int32_t AudioSession;
-/*
+/**
* Special audio session values.
*/
@export(name="audio_session_t", value_prefix="AUDIO_SESSION_")
enum AudioSessionConsts : int32_t {
- /*
+ /**
* Session for effects attached to a particular output stream
* (value must be less than 0)
*/
OUTPUT_STAGE = -1,
- /*
+ /**
* Session for effects applied to output mix. These effects can
* be moved by audio policy manager to another output stream
* (value must be 0)
*/
OUTPUT_MIX = 0,
- /*
+ /**
* Application does not specify an explicit session ID to be used, and
* requests a new session ID to be allocated. Corresponds to
* AudioManager.AUDIO_SESSION_ID_GENERATE and
* AudioSystem.AUDIO_SESSION_ALLOCATE.
*/
ALLOCATE = 0,
- /*
+ /**
* For use with AudioRecord::start(), this indicates no trigger session.
* It is also used with output tracks and patch tracks, which never have a
* session.
@@ -181,7 +181,7 @@
NONE = 0
};
-/*
+/**
* Audio format is a 32-bit word that consists of:
* main format field (upper 8 bits)
* sub format field (lower 24 bits).
@@ -196,13 +196,13 @@
enum AudioFormat : uint32_t {
INVALID = 0xFFFFFFFFUL,
DEFAULT = 0,
- PCM = 0x00000000UL, /* DO NOT CHANGE */
+ PCM = 0x00000000UL, /** DO NOT CHANGE */
MP3 = 0x01000000UL,
AMR_NB = 0x02000000UL,
AMR_WB = 0x03000000UL,
AAC = 0x04000000UL,
- HE_AAC_V1 = 0x05000000UL, /* Deprecated, Use AAC_HE_V1*/
- HE_AAC_V2 = 0x06000000UL, /* Deprecated, Use AAC_HE_V2*/
+ HE_AAC_V1 = 0x05000000UL, /** Deprecated, Use AAC_HE_V1 */
+ HE_AAC_V2 = 0x06000000UL, /** Deprecated, Use AAC_HE_V2 */
VORBIS = 0x07000000UL,
OPUS = 0x08000000UL,
AC3 = 0x09000000UL,
@@ -232,10 +232,10 @@
APTX_HD = 0x21000000UL,
AC4 = 0x22000000UL,
LDAC = 0x23000000UL,
- MAIN_MASK = 0xFF000000UL, /* Deprecated */
+ MAIN_MASK = 0xFF000000UL, /** Deprecated */
SUB_MASK = 0x00FFFFFFUL,
- /* Subformats */
+ /** 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
@@ -260,10 +260,10 @@
VORBIS_SUB_NONE = 0x0,
- /* Aliases */
- /* note != AudioFormat.ENCODING_PCM_16BIT */
+ /** Aliases */
+ /** note != AudioFormat.ENCODING_PCM_16BIT */
PCM_16_BIT = (PCM | PCM_SUB_16_BIT),
- /* note != AudioFormat.ENCODING_PCM_8BIT */
+ /** 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),
@@ -291,7 +291,7 @@
AAC_ADTS_ELD = (AAC_ADTS | AAC_SUB_ELD)
};
-/*
+/**
* Usage of these values highlights places in the code that use 2- or 8- channel
* assumptions.
*/
@@ -301,7 +301,7 @@
FCC_8 = 8 // This is typically due to audio mixer and resampler limitations
};
-/*
+/**
* A channel mask per se only defines the presence or absence of a channel, not
* the order. See AUDIO_INTERLEAVE_* for the platform convention of order.
*
@@ -335,21 +335,21 @@
*/
@export(name="", value_prefix="AUDIO_CHANNEL_")
enum AudioChannelMask : uint32_t {
- REPRESENTATION_POSITION = 0, /* must be 0 for compatibility */
- /* 1 is reserved for future use */
+ REPRESENTATION_POSITION = 0, /** must be 0 for compatibility */
+ /** 1 is reserved for future use */
REPRESENTATION_INDEX = 2,
- /* 3 is reserved for future use */
+ /** 3 is reserved for future use */
- /* These can be a complete value of AudioChannelMask */
+ /** 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 */
+ /** output channels */
OUT_FRONT_LEFT = 0x1,
OUT_FRONT_RIGHT = 0x2,
OUT_FRONT_CENTER = 0x4,
@@ -375,7 +375,7 @@
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_* */
+ /** 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 |
@@ -385,7 +385,7 @@
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_* */
+ /** 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),
@@ -393,7 +393,7 @@
OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
OUT_BACK_LEFT | OUT_BACK_RIGHT |
OUT_BACK_CENTER),
- /* matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND */
+ /** 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 |
@@ -408,9 +408,9 @@
OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_CENTER | OUT_TOP_FRONT_RIGHT |
OUT_TOP_BACK_LEFT | OUT_TOP_BACK_CENTER | OUT_TOP_BACK_RIGHT),
- /* These are bits only, not complete values */
+ /** These are bits only, not complete values */
- /* input channels */
+ /** input channels */
IN_LEFT = 0x4,
IN_RIGHT = 0x8,
IN_FRONT = 0x10,
@@ -456,7 +456,7 @@
};
-/*
+/**
* Expresses the convention when stereo audio samples are stored interleaved
* in an array. This should improve readability by allowing code to use
* symbolic indices instead of hard-coded [0] and [1].
@@ -472,7 +472,7 @@
RIGHT = 1,
};
-/*
+/**
* Major modes for a mobile device. The current mode setting affects audio
* routing.
*/
@@ -492,10 +492,10 @@
@export(name="", value_prefix="AUDIO_DEVICE_")
enum AudioDevice : uint32_t {
NONE = 0x0,
- /* reserved bits */
+ /** reserved bits */
BIT_IN = 0x80000000,
BIT_DEFAULT = 0x40000000,
- /* output devices */
+ /** output devices */
OUT_EARPIECE = 0x1,
OUT_SPEAKER = 0x2,
OUT_WIRED_HEADSET = 0x4,
@@ -508,30 +508,30 @@
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) */
+ /** 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 */
+ /** 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 */
+ /** 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 */
+ /** Telephony voice TX path */
OUT_TELEPHONY_TX = 0x10000,
- /* Analog jack with line impedance detected */
+ /** Analog jack with line impedance detected */
OUT_LINE = 0x20000,
- /* HDMI Audio Return Channel */
+ /** HDMI Audio Return Channel */
OUT_HDMI_ARC = 0x40000,
- /* S/PDIF out */
+ /** S/PDIF out */
OUT_SPDIF = 0x80000,
- /* FM transmitter out */
+ /** FM transmitter out */
OUT_FM = 0x100000,
- /* Line out for av devices */
+ /** Line out for av devices */
OUT_AUX_LINE = 0x200000,
- /* limited-output speaker device for acoustic safety */
+ /** 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) */
+ /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
OUT_BUS = 0x1000000,
OUT_PROXY = 0x2000000,
OUT_USB_HEADSET = 0x4000000,
@@ -571,7 +571,7 @@
OUT_BLUETOOTH_SCO_HEADSET |
OUT_BLUETOOTH_SCO_CARKIT),
OUT_ALL_USB = (OUT_USB_ACCESSORY | OUT_USB_DEVICE | OUT_USB_HEADSET),
- /* input devices */
+ /** input devices */
IN_COMMUNICATION = BIT_IN | 0x1,
IN_AMBIENT = BIT_IN | 0x2,
IN_BUILTIN_MIC = BIT_IN | 0x4,
@@ -579,7 +579,7 @@
IN_WIRED_HEADSET = BIT_IN | 0x10,
IN_AUX_DIGITAL = BIT_IN | 0x20,
IN_HDMI = IN_AUX_DIGITAL,
- /* Telephony voice RX path */
+ /** Telephony voice RX path */
IN_VOICE_CALL = BIT_IN | 0x40,
IN_TELEPHONY_RX = IN_VOICE_CALL,
IN_BACK_MIC = BIT_IN | 0x80,
@@ -588,18 +588,18 @@
IN_DGTL_DOCK_HEADSET = BIT_IN | 0x400,
IN_USB_ACCESSORY = BIT_IN | 0x800,
IN_USB_DEVICE = BIT_IN | 0x1000,
- /* FM tuner input */
+ /** FM tuner input */
IN_FM_TUNER = BIT_IN | 0x2000,
- /* TV tuner input */
+ /** TV tuner input */
IN_TV_TUNER = BIT_IN | 0x4000,
- /* Analog jack with line impedance detected */
+ /** Analog jack with line impedance detected */
IN_LINE = BIT_IN | 0x8000,
- /* S/PDIF in */
+ /** 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) */
+ /** 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,
@@ -633,7 +633,7 @@
IN_ALL_USB = (IN_USB_ACCESSORY | IN_USB_DEVICE | IN_USB_HEADSET),
};
-/*
+/**
* The audio output flags serve two purposes:
*
* - when an AudioTrack is created they indicate a "wish" to be connected to an
@@ -674,7 +674,7 @@
VOIP_RX = 0x8000, // preferred output for VoIP calls.
};
-/*
+/**
* 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
@@ -717,7 +717,7 @@
MAX = CNT - 1,
};
-/*
+/**
* Additional information about the stream passed to hardware decoders.
*/
struct AudioOffloadInfo {
@@ -734,7 +734,7 @@
AudioUsage usage;
};
-/*
+/**
* Commonly used audio stream configuration parameters.
*/
struct AudioConfig {
@@ -746,13 +746,13 @@
};
-/*
+/**
*
* Volume control
*
*/
-/*
+/**
* Type of gain control exposed by an audio port.
*/
@export(name="", value_prefix="AUDIO_GAIN_MODE_")
@@ -762,7 +762,7 @@
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.
*/
@@ -777,7 +777,7 @@
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.
*/
@@ -785,7 +785,7 @@
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
@@ -797,13 +797,13 @@
};
-/*
+/**
*
* 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
@@ -811,7 +811,7 @@
* the interface.
*/
-/* Audio port role: either source or sink */
+/** Audio port role: either source or sink */
@export(name="audio_port_role_t", value_prefix="AUDIO_PORT_ROLE_")
enum AudioPortRole : int32_t {
NONE,
@@ -819,7 +819,7 @@
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)
*/
@@ -831,7 +831,7 @@
SESSION,
};
-/*
+/**
* Extension for audio port configuration structure when the audio port is a
* hardware device.
*/
@@ -841,7 +841,7 @@
uint8_t[32] address; // device address. "" if N/A
};
-/*
+/**
* Extension for audio port configuration structure when the audio port is an
* audio session.
*/
@@ -849,7 +849,7 @@
AudioSession session;
};
-/*
+/**
* Flags indicating which fields are to be considered in AudioPortConfig.
*/
@export(name="", value_prefix="AUDIO_PORT_CONFIG_")
@@ -861,7 +861,7 @@
ALL = SAMPLE_RATE | CHANNEL_MASK | FORMAT | GAIN
};
-/*
+/**
* Audio port configuration structure used to specify a particular configuration
* of an audio port.
*/
@@ -888,7 +888,7 @@
} ext;
};
-/*
+/**
* Extension for audio port structure when the audio port is a hardware device.
*/
struct AudioPortDeviceExt {
@@ -897,7 +897,7 @@
uint8_t[32] address;
};
-/*
+/**
* Latency class of the audio mix.
*/
@export(name="audio_mix_latency_class_t", value_prefix="AUDIO_LATENCY_")
@@ -912,7 +912,7 @@
AudioMixLatencyClass latencyClass;
};
-/*
+/**
* Extension for audio port structure when the audio port is an audio session.
*/
struct AudioPortSessionExt {
diff --git a/audio/effect/2.0/IAcousticEchoCancelerEffect.hal b/audio/effect/2.0/IAcousticEchoCancelerEffect.hal
index 9e2e0c3..b5f94a5 100644
--- a/audio/effect/2.0/IAcousticEchoCancelerEffect.hal
+++ b/audio/effect/2.0/IAcousticEchoCancelerEffect.hal
@@ -20,12 +20,12 @@
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/2.0/IAutomaticGainControlEffect.hal b/audio/effect/2.0/IAutomaticGainControlEffect.hal
index a02002d..b8ca7e3 100644
--- a/audio/effect/2.0/IAutomaticGainControlEffect.hal
+++ b/audio/effect/2.0/IAutomaticGainControlEffect.hal
@@ -20,32 +20,32 @@
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);
@@ -56,12 +56,12 @@
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/2.0/IBassBoostEffect.hal b/audio/effect/2.0/IBassBoostEffect.hal
index bcf7b7d..db6a297 100644
--- a/audio/effect/2.0/IBassBoostEffect.hal
+++ b/audio/effect/2.0/IBassBoostEffect.hal
@@ -20,7 +20,7 @@
import IEffect;
interface IBassBoostEffect extends IEffect {
- /*
+ /**
* Returns whether setting bass boost strength is supported.
*/
isStrengthSupported() generates (Result retval, bool strengthSupported);
@@ -30,7 +30,7 @@
MAX = 1000
};
- /*
+ /**
* Sets bass boost strength.
*
* @param strength strength of the effect. The valid range for strength
@@ -41,7 +41,7 @@
*/
setStrength(uint16_t strength) generates (Result retval);
- /*
+ /**
* Gets virtualization strength.
*/
getStrength() generates (Result retval, uint16_t strength);
diff --git a/audio/effect/2.0/IEffect.hal b/audio/effect/2.0/IEffect.hal
index d254e8c..332e2df 100644
--- a/audio/effect/2.0/IEffect.hal
+++ b/audio/effect/2.0/IEffect.hal
@@ -20,7 +20,7 @@
import IEffectBufferProviderCallback;
interface IEffect {
- /*
+ /**
* Initialize effect engine--all configurations return to default.
*
* @return retval operation completion status.
@@ -29,7 +29,7 @@
@callflow(next={"*"})
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.
@@ -45,7 +45,7 @@
IEffectBufferProviderCallback outputBufferProvider)
generates (Result retval);
- /*
+ /**
* Reset the effect engine. Keep configuration but resets state and buffer
* content.
*
@@ -54,7 +54,7 @@
@callflow(next={"*"})
reset() generates (Result retval);
- /*
+ /**
* Enable processing.
*
* @return retval operation completion status.
@@ -62,7 +62,7 @@
@callflow(next={"prepareForProcessing"})
enable() generates (Result retval);
- /*
+ /**
* Disable processing.
*
* @return retval operation completion status.
@@ -70,7 +70,7 @@
@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.
@@ -84,7 +84,7 @@
@callflow(next={"*"})
setDevice(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
@@ -103,7 +103,7 @@
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.
@@ -116,7 +116,7 @@
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.
@@ -127,7 +127,7 @@
@callflow(next={"*"})
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.
@@ -143,7 +143,7 @@
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.
@@ -157,7 +157,7 @@
@callflow(next={"*"})
setInputDevice(AudioDevice device) generates (Result retval);
- /*
+ /**
* Read audio parameters configurations for input and output buffers.
*
* @return retval operation completion status.
@@ -166,7 +166,7 @@
@callflow(next={"*"})
getConfig() generates (Result retval, EffectConfig config);
- /*
+ /**
* Read audio parameters configurations for input and output buffers of
* reverse stream.
*
@@ -176,7 +176,7 @@
@callflow(next={"*"})
getConfigReverse() generates (Result retval, EffectConfig config);
- /*
+ /**
* Queries for supported combinations of main and auxiliary channels
* (e.g. for a multi-microphone noise suppressor).
*
@@ -190,7 +190,7 @@
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
@@ -201,7 +201,7 @@
getAuxChannelsConfig()
generates (Result retval, EffectAuxChannelsConfig result);
- /*
+ /**
* Sets the current configuration of main and auxiliary channels.
*
* @return retval operation completion status; absence of the feature
@@ -211,7 +211,7 @@
setAuxChannelsConfig(EffectAuxChannelsConfig config)
generates (Result retval);
- /*
+ /**
* Set the audio source the capture path is configured for (Camcorder, voice
* recognition...).
*
@@ -224,7 +224,7 @@
@callflow(next={"*"})
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.
@@ -235,7 +235,7 @@
@callflow(next={"*"})
offload(EffectOffloadParameter param) generates (Result retval);
- /*
+ /**
* Returns the effect descriptor.
*
* @return retval operation completion status.
@@ -244,7 +244,7 @@
@callflow(next={"*"})
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
@@ -270,7 +270,7 @@
@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.
@@ -289,7 +289,7 @@
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.
@@ -310,7 +310,7 @@
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.
@@ -331,7 +331,7 @@
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.
@@ -353,7 +353,7 @@
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.
@@ -384,7 +384,7 @@
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.
@@ -406,7 +406,7 @@
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.
@@ -426,7 +426,7 @@
setCurrentConfigForFeature(uint32_t featureId, vec<uint8_t> configData)
generates (Result retval);
- /*
+ /**
* Called by the framework to deinitialize the effect and free up
* all the currently allocated resources. It is recommended to close
* the effect on the client side as soon as it is becomes unused.
diff --git a/audio/effect/2.0/IEffectBufferProviderCallback.hal b/audio/effect/2.0/IEffectBufferProviderCallback.hal
index 53f4d6e..6ab33c3 100644
--- a/audio/effect/2.0/IEffectBufferProviderCallback.hal
+++ b/audio/effect/2.0/IEffectBufferProviderCallback.hal
@@ -16,12 +16,12 @@
package android.hardware.audio.effect@2.0;
-/*
+/**
* 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.
*
@@ -29,7 +29,7 @@
*/
getBuffer() generates (AudioBuffer buffer);
- /*
+ /**
* Called to provide a buffer with the data written by 'process' function.
*
* @param buffer audio buffer for processing
diff --git a/audio/effect/2.0/IEffectsFactory.hal b/audio/effect/2.0/IEffectsFactory.hal
index c82b4a2..3d300ec 100644
--- a/audio/effect/2.0/IEffectsFactory.hal
+++ b/audio/effect/2.0/IEffectsFactory.hal
@@ -20,7 +20,7 @@
import IEffect;
interface IEffectsFactory {
- /*
+ /**
* Returns descriptors of different effects in all loaded libraries.
*
* @return retval operation completion status.
@@ -28,7 +28,7 @@
*/
getAllDescriptors() generates(Result retval, vec<EffectDescriptor> result);
- /*
+ /**
* Returns a descriptor of a particular effect.
*
* @return retval operation completion status.
@@ -36,7 +36,7 @@
*/
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.
@@ -56,7 +56,7 @@
createEffect(Uuid uid, AudioSession session, AudioIoHandle ioHandle)
generates (Result retval, IEffect result, uint64_t effectId);
- /*
+ /**
* Dumps information about effects into the provided file descriptor.
* This is used for the dumpsys facility.
*
diff --git a/audio/effect/2.0/IEnvironmentalReverbEffect.hal b/audio/effect/2.0/IEnvironmentalReverbEffect.hal
index d9b1ee6..dca89f9 100644
--- a/audio/effect/2.0/IEnvironmentalReverbEffect.hal
+++ b/audio/effect/2.0/IEnvironmentalReverbEffect.hal
@@ -20,12 +20,12 @@
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);
@@ -53,102 +53,102 @@
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 frequences level.
*/
setRoomHfLevel(int16_t roomHfLevel) generates (Result retval);
- /*
+ /**
* Gets the room high frequences 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 frequences decay.
*/
setDecayHfRatio(int16_t decayHfRatio) generates (Result retval);
- /*
+ /**
* Gets the ratio of high frequences 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);
@@ -166,12 +166,12 @@
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/2.0/IEqualizerEffect.hal b/audio/effect/2.0/IEqualizerEffect.hal
index b8fa177..1528e0d 100644
--- a/audio/effect/2.0/IEqualizerEffect.hal
+++ b/audio/effect/2.0/IEqualizerEffect.hal
@@ -20,58 +20,58 @@
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);
@@ -81,12 +81,12 @@
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/2.0/ILoudnessEnhancerEffect.hal b/audio/effect/2.0/ILoudnessEnhancerEffect.hal
index 3e1ee4e..adeb1c8 100644
--- a/audio/effect/2.0/ILoudnessEnhancerEffect.hal
+++ b/audio/effect/2.0/ILoudnessEnhancerEffect.hal
@@ -20,12 +20,12 @@
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/2.0/INoiseSuppressionEffect.hal b/audio/effect/2.0/INoiseSuppressionEffect.hal
index ae2bfb5..6617a1e 100644
--- a/audio/effect/2.0/INoiseSuppressionEffect.hal
+++ b/audio/effect/2.0/INoiseSuppressionEffect.hal
@@ -26,12 +26,12 @@
HIGH
};
- /*
+ /**
* Sets suppression level.
*/
setSuppressionLevel(Level level) generates (Result retval);
- /*
+ /**
* Gets suppression level.
*/
getSuppressionLevel() generates (Result retval, Level level);
@@ -41,12 +41,12 @@
MULTI_CHANNEL
};
- /*
+ /**
* Set suppression type.
*/
setSuppressionType(Type type) generates (Result retval);
- /*
+ /**
* Get suppression type.
*/
getSuppressionType() generates (Result retval, Type type);
@@ -56,12 +56,12 @@
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/2.0/IVirtualizerEffect.hal b/audio/effect/2.0/IVirtualizerEffect.hal
index 2b7116c..49b49a0 100644
--- a/audio/effect/2.0/IVirtualizerEffect.hal
+++ b/audio/effect/2.0/IVirtualizerEffect.hal
@@ -20,7 +20,7 @@
import IEffect;
interface IVirtualizerEffect extends IEffect {
- /*
+ /**
* Returns whether setting virtualization strength is supported.
*/
isStrengthSupported() generates (bool strengthSupported);
@@ -30,7 +30,7 @@
MAX = 1000
};
- /*
+ /**
* Sets virtualization strength.
*
* @param strength strength of the effect. The valid range for strength
@@ -41,7 +41,7 @@
*/
setStrength(uint16_t strength) generates (Result retval);
- /*
+ /**
* Gets virtualization strength.
*/
getStrength() generates (Result retval, uint16_t strength);
@@ -56,19 +56,19 @@
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(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.
*/
diff --git a/audio/effect/2.0/IVisualizerEffect.hal b/audio/effect/2.0/IVisualizerEffect.hal
index 79dc9ee..fd3edbd 100644
--- a/audio/effect/2.0/IVisualizerEffect.hal
+++ b/audio/effect/2.0/IVisualizerEffect.hal
@@ -25,12 +25,12 @@
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);
@@ -42,22 +42,22 @@
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);
@@ -69,19 +69,19 @@
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.
*
@@ -99,7 +99,7 @@
} peakAndRms;
} value;
};
- /*
+ /**
* Retrieves the lastest measurements. The measurements to be made
* are specified by 'setMeasurementMode' parameter.
*
diff --git a/audio/effect/2.0/types.hal b/audio/effect/2.0/types.hal
index 0626ec5..2c5e4ef 100644
--- a/audio/effect/2.0/types.hal
+++ b/audio/effect/2.0/types.hal
@@ -27,7 +27,7 @@
RESULT_TOO_BIG
};
-/*
+/**
* Effect engine capabilities/requirements flags.
*
* Definitions for flags field of effect descriptor.
@@ -195,7 +195,7 @@
NO_PROCESS = 1 << NO_PROCESS_SHIFT
};
-/*
+/**
* The effect descriptor contains necessary information to facilitate the
* enumeration of the effect engines present in a library.
*/
@@ -212,7 +212,7 @@
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.:
@@ -235,7 +235,7 @@
ACCESS_ACCUMULATE
};
-/*
+/**
* Determines what fields of EffectBufferConfig need to be considered.
*/
@export(name="", value_prefix="EFFECT_CONFIG_")
@@ -248,7 +248,7 @@
ALL = BUFFER | SMP_RATE | CHANNELS | FORMAT | ACC_MODE
};
-/*
+/**
* The buffer config structure specifies the input or output audio format
* to be used by the effect engine.
*/
@@ -285,7 +285,7 @@
// the effect is attached to
};
-/*
+/**
* The message queue flags used to synchronize reads and writes from
* the status message queue used by effects.
*/