Revert "Audio V4: Split system and vendor Audio.h"
This reverts commit 49c56de516b4e9556ae1b0af0ccfa7d695f5807b.
Reason for revert: Breaks the build of multiple devices
Change-Id: Iebd868467948b8afa5907462ccc0720cd9c4871e
diff --git a/audio/README b/audio/README
index f4b8555..2b81450 100644
--- a/audio/README
+++ b/audio/README
@@ -7,18 +7,15 @@
|-- common <== code common to audio core and effect API
| |-- 2.0
| | |-- default <== code that wraps the legacy API
-| | |-- legacy <== legacy API compatible with 2.0
| | `-- vts <== vts of 2.0 core and effect API common code
| |-- 4.0
| | |-- default
-| | |-- legacy
| | `-- vts
| |-- ... <== The future versions should continue this structure
| | |-- default
| | `-- vts
| `-- all_versions <== code common to all version of both core and effect API
| |-- default
-| | |-- legacy <== legacy API compatible with all versions
| `-- vts <== vts of core and effect API common version independent code
|
|-- core <== code relative to the core API
@@ -38,17 +35,13 @@
`-- effect <== idem for the effect API
|-- 2.0
| |-- default
- | |-- legacy <== legacy effect API compatible with 2.0
| `-- vts
|-- 4.0
| |-- default
- | |-- legacy
| `-- vts
|-- ...
| |-- default
- | |-- default
| `-- vts
`-- all_versions
|-- default
- |-- legacy
`-- vts
diff --git a/audio/common/2.0/default/Android.bp b/audio/common/2.0/default/Android.bp
index 123f8b3..ac66479 100644
--- a/audio/common/2.0/default/Android.bp
+++ b/audio/common/2.0/default/Android.bp
@@ -16,7 +16,10 @@
cc_library_shared {
name: "android.hardware.audio.common@2.0-util",
defaults: ["hidl_defaults"],
- vendor: true,
+ vendor_available: true,
+ vndk: {
+ enabled: true,
+ },
srcs: [
"HidlUtils.cpp",
],
@@ -38,7 +41,7 @@
],
header_libs: [
- "android.hardware.audio.common.legacy@2.0",
+ "libaudio_system_headers",
"libhardware_headers",
],
}
diff --git a/audio/common/2.0/legacy/Android.bp b/audio/common/2.0/legacy/Android.bp
deleted file mode 100644
index 2888c96..0000000
--- a/audio/common/2.0/legacy/Android.bp
+++ /dev/null
@@ -1,15 +0,0 @@
-cc_library_headers {
- name: "android.hardware.audio.common.legacy@2.0",
- vendor: true,
- header_libs: [
- "libhardware_headers",
- "android.hardware.audio.common.legacy@all-versions",
- ],
- export_header_lib_headers: [
- "libhardware_headers",
- "android.hardware.audio.common.legacy@all-versions",
- ],
-
- export_include_dirs: ["include"],
-}
-
diff --git a/audio/common/2.0/legacy/OWNERS b/audio/common/2.0/legacy/OWNERS
deleted file mode 100644
index 6fdc97c..0000000
--- a/audio/common/2.0/legacy/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-elaurent@google.com
-krocard@google.com
-mnaganov@google.com
diff --git a/audio/common/2.0/legacy/include/hardware/audio.h b/audio/common/2.0/legacy/include/hardware/audio.h
deleted file mode 100644
index 1ad3e0e..0000000
--- a/audio/common/2.0/legacy/include/hardware/audio.h
+++ /dev/null
@@ -1,709 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-#ifndef ANDROID_AUDIO_HAL_INTERFACE_H
-#define ANDROID_AUDIO_HAL_INTERFACE_H
-
-#include <stdint.h>
-#include <strings.h>
-#include <sys/cdefs.h>
-#include <sys/types.h>
-#include <time.h>
-
-#include <cutils/bitops.h>
-
-#include <hardware/audio_effect.h>
-#include <hardware/hardware.h>
-#include <system/audio.h>
-
-__BEGIN_DECLS
-
-/**
- * The id of this module
- */
-#define AUDIO_HARDWARE_MODULE_ID "audio"
-
-/**
- * Name of the audio devices to open
- */
-#define AUDIO_HARDWARE_INTERFACE "audio_hw_if"
-
-/* Use version 0.1 to be compatible with first generation of audio hw module with version_major
- * hardcoded to 1. No audio module API change.
- */
-#define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1)
-#define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1
-
-/* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0
- * will be considered of first generation API.
- */
-#define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0)
-#define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0)
-#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0)
-#define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0)
-#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_0
-/* Minimal audio HAL version supported by the audio framework */
-#define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0
-
-/**************************************/
-
-/**
- * standard audio parameters that the HAL may need to handle
- */
-
-/**
- * audio device parameters
- */
-
-/* TTY mode selection */
-#define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode"
-#define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off"
-#define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco"
-#define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco"
-#define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full"
-
-/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */
-#define AUDIO_PARAMETER_KEY_HAC "HACSetting"
-#define AUDIO_PARAMETER_VALUE_HAC_ON "ON"
-#define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF"
-
-/* A2DP sink address set by framework */
-#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
-
-/* A2DP source address set by framework */
-#define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"
-
-/* Bluetooth SCO wideband */
-#define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs"
-
-/**
- * audio stream parameters
- */
-
-/* Enable AANC */
-#define AUDIO_PARAMETER_KEY_AANC "aanc_enabled"
-
-/**************************************/
-
-/* common audio stream parameters and operations */
-struct audio_stream {
- /**
- * Return the sampling rate in Hz - eg. 44100.
- */
- uint32_t (*get_sample_rate)(const struct audio_stream* stream);
-
- /* currently unused - use set_parameters with key
- * AUDIO_PARAMETER_STREAM_SAMPLING_RATE
- */
- int (*set_sample_rate)(struct audio_stream* stream, uint32_t rate);
-
- /**
- * Return size of input/output buffer in bytes for this stream - eg. 4800.
- * It should be a multiple of the frame size. See also get_input_buffer_size.
- */
- size_t (*get_buffer_size)(const struct audio_stream* stream);
-
- /**
- * Return the channel mask -
- * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
- */
- audio_channel_mask_t (*get_channels)(const struct audio_stream* stream);
-
- /**
- * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
- */
- audio_format_t (*get_format)(const struct audio_stream* stream);
-
- /* currently unused - use set_parameters with key
- * AUDIO_PARAMETER_STREAM_FORMAT
- */
- int (*set_format)(struct audio_stream* stream, audio_format_t format);
-
- /**
- * Put the audio hardware input/output into standby mode.
- * Driver should exit from standby mode at the next I/O operation.
- * Returns 0 on success and <0 on failure.
- */
- int (*standby)(struct audio_stream* stream);
-
- /** dump the state of the audio input/output device */
- int (*dump)(const struct audio_stream* stream, int fd);
-
- /** Return the set of device(s) which this stream is connected to */
- audio_devices_t (*get_device)(const struct audio_stream* stream);
-
- /**
- * Currently unused - set_device() corresponds to set_parameters() with key
- * AUDIO_PARAMETER_STREAM_ROUTING for both input and output.
- * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by
- * input streams only.
- */
- int (*set_device)(struct audio_stream* stream, audio_devices_t device);
-
- /**
- * set/get audio stream parameters. The function accepts a list of
- * parameter key value pairs in the form: key1=value1;key2=value2;...
- *
- * Some keys are reserved for standard parameters (See AudioParameter class)
- *
- * If the implementation does not accept a parameter change while
- * the output is active but the parameter is acceptable otherwise, it must
- * return -ENOSYS.
- *
- * The audio flinger will put the stream in standby and then change the
- * parameter value.
- */
- int (*set_parameters)(struct audio_stream* stream, const char* kv_pairs);
-
- /*
- * Returns a pointer to a heap allocated string. The caller is responsible
- * for freeing the memory for it using free().
- */
- char* (*get_parameters)(const struct audio_stream* stream, const char* keys);
- int (*add_audio_effect)(const struct audio_stream* stream, effect_handle_t effect);
- int (*remove_audio_effect)(const struct audio_stream* stream, effect_handle_t effect);
-};
-typedef struct audio_stream audio_stream_t;
-
-/* type of asynchronous write callback events. Mutually exclusive */
-typedef enum {
- STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
- STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */
- STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */
-} stream_callback_event_t;
-
-typedef int (*stream_callback_t)(stream_callback_event_t event, void* param, void* cookie);
-
-/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
-typedef enum {
- AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
- AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
- from the current track has been played to
- give time for gapless track switch */
-} audio_drain_type_t;
-
-/**
- * audio_stream_out is the abstraction interface for the audio output hardware.
- *
- * It provides information about various properties of the audio output
- * hardware driver.
- */
-
-struct audio_stream_out {
- /**
- * Common methods of the audio stream out. This *must* be the first member of audio_stream_out
- * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
- * where it's known the audio_stream references an audio_stream_out.
- */
- struct audio_stream common;
-
- /**
- * Return the audio hardware driver estimated latency in milliseconds.
- */
- uint32_t (*get_latency)(const struct audio_stream_out* stream);
-
- /**
- * Use this method in situations where audio mixing is done in the
- * hardware. This method serves as a direct interface with hardware,
- * allowing you 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.
- */
- int (*set_volume)(struct audio_stream_out* stream, float left, float right);
-
- /**
- * Write audio buffer to driver. Returns number of bytes written, or a
- * negative status_t. If at least one frame was written successfully prior to the error,
- * it is suggested that the driver return that successful (short) byte count
- * and then return an error in the subsequent call.
- *
- * If set_callback() has previously been called to enable non-blocking mode
- * the write() is not allowed to block. It must write only the number of
- * bytes that currently fit in the driver/hardware buffer and then return
- * this byte count. If this is less than the requested write size the
- * callback function must be called when more space is available in the
- * driver/hardware buffer.
- */
- ssize_t (*write)(struct audio_stream_out* stream, const void* buffer, size_t bytes);
-
- /* return the number of audio frames written by the audio dsp to DAC since
- * the output has exited standby
- */
- int (*get_render_position)(const struct audio_stream_out* stream, uint32_t* dsp_frames);
-
- /**
- * 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.
- */
- int (*get_next_write_timestamp)(const struct audio_stream_out* stream, int64_t* timestamp);
-
- /**
- * set the callback function 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.
- */
- int (*set_callback)(struct audio_stream_out* stream, stream_callback_t callback, void* cookie);
-
- /**
- * 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 should be implemented for hardware with non-trivial
- * latency. In the pause state audio hardware could still be using power. User may
- * consider calling suspend after a timeout.
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*pause)(struct audio_stream_out* stream);
-
- /**
- * Notifies to the audio driver to resume playback following a pause.
- * Returns error if called without matching pause.
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*resume)(struct audio_stream_out* stream);
-
- /**
- * Requests notification when data buffered by the driver/hardware has
- * been played. If set_callback() has previously been called to enable
- * non-blocking mode, the drain() must not block, instead it should return
- * quickly and completion of the drain is notified through the callback.
- * If set_callback() has not been called, the drain() must block until
- * completion.
- * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
- * data has been played.
- * If type==AUDIO_DRAIN_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() call
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type);
-
- /**
- * Notifies to the audio driver to flush the queued data. Stream must already
- * be paused before calling flush().
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*flush)(struct audio_stream_out* stream);
-
- /**
- * 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 should 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 should correspond to N rather than N+M.
- * The terms 'recent' and 'small' are not defined.
- * They reflect the quality of the implementation.
- *
- * 3.0 and higher only.
- */
- int (*get_presentation_position)(const struct audio_stream_out* stream, uint64_t* frames,
- struct timespec* timestamp);
-
- /**
- * Called by the framework to start a stream operating in mmap mode.
- * create_mmap_buffer must be called before calling start()
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case of success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*start)(const struct audio_stream_out* stream);
-
- /**
- * Called by the framework to stop a stream operating in mmap mode.
- * Must be called after start()
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case of success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*stop)(const struct audio_stream_out* stream);
-
- /**
- * Called by the framework to retrieve information on the mmap buffer used for audio
- * samples transfer.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[in] min_size_frames minimum buffer size requested. The actual buffer
- * size returned in struct audio_mmap_buffer_info can be larger.
- * \param[out] info address at which the mmap buffer information should be returned.
- *
- * \return 0 if the buffer was allocated.
- * -ENODEV in case of initialization error
- * -EINVAL if the requested buffer size is too large
- * -ENOSYS if called out of sequence (e.g. buffer already allocated)
- */
- int (*create_mmap_buffer)(const struct audio_stream_out* stream, int32_t min_size_frames,
- struct audio_mmap_buffer_info* info);
-
- /**
- * Called by the framework to read current read/write position in the mmap buffer
- * with associated time stamp.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[out] position address at which the mmap read/write position should be returned.
- *
- * \return 0 if the position is successfully returned.
- * -ENODATA if the position cannot be retrieved
- * -ENOSYS if called before create_mmap_buffer()
- */
- int (*get_mmap_position)(const struct audio_stream_out* stream,
- struct audio_mmap_position* position);
-};
-typedef struct audio_stream_out audio_stream_out_t;
-
-struct audio_stream_in {
- /**
- * Common methods of the audio stream in. This *must* be the first member of audio_stream_in
- * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
- * where it's known the audio_stream references an audio_stream_in.
- */
- struct audio_stream common;
-
- /** set the input gain for the audio driver. This method is for
- * for future use */
- int (*set_gain)(struct audio_stream_in* stream, float gain);
-
- /** Read audio buffer in from audio driver. Returns number of bytes read, or a
- * negative status_t. If at least one frame was read prior to the error,
- * read should return that byte count and then return an error in the subsequent call.
- */
- ssize_t (*read)(struct audio_stream_in* stream, void* buffer, size_t bytes);
-
- /**
- * 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.
- *
- * Unit: the number of input audio frames
- */
- uint32_t (*get_input_frames_lost)(struct audio_stream_in* stream);
-
- /**
- * Return a recent count of the number of audio frames received and
- * the clock time associated with that frame count.
- *
- * frames is the total frame count received. This should be as early in
- * the capture pipeline as possible. In general,
- * frames should be non-negative and should not go "backwards".
- *
- * time is the clock MONOTONIC time when frames was measured. In general,
- * time should be a positive quantity and should not go "backwards".
- *
- * The status returned is 0 on success, -ENOSYS if the device is not
- * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
- */
- int (*get_capture_position)(const struct audio_stream_in* stream, int64_t* frames,
- int64_t* time);
-
- /**
- * Called by the framework to start a stream operating in mmap mode.
- * create_mmap_buffer must be called before calling start()
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case off success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*start)(const struct audio_stream_in* stream);
-
- /**
- * Called by the framework to stop a stream operating in mmap mode.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case of success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*stop)(const struct audio_stream_in* stream);
-
- /**
- * Called by the framework to retrieve information on the mmap buffer used for audio
- * samples transfer.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[in] min_size_frames minimum buffer size requested. The actual buffer
- * size returned in struct audio_mmap_buffer_info can be larger.
- * \param[out] info address at which the mmap buffer information should be returned.
- *
- * \return 0 if the buffer was allocated.
- * -ENODEV in case of initialization error
- * -EINVAL if the requested buffer size is too large
- * -ENOSYS if called out of sequence (e.g. buffer already allocated)
- */
- int (*create_mmap_buffer)(const struct audio_stream_in* stream, int32_t min_size_frames,
- struct audio_mmap_buffer_info* info);
-
- /**
- * Called by the framework to read current read/write position in the mmap buffer
- * with associated time stamp.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[out] position address at which the mmap read/write position should be returned.
- *
- * \return 0 if the position is successfully returned.
- * -ENODATA if the position cannot be retreived
- * -ENOSYS if called before mmap_read_position()
- */
- int (*get_mmap_position)(const struct audio_stream_in* stream,
- struct audio_mmap_position* position);
-};
-typedef struct audio_stream_in audio_stream_in_t;
-
-/**
- * return the frame size (number of bytes per sample).
- *
- * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead.
- */
-__attribute__((__deprecated__)) static inline size_t audio_stream_frame_size(
- const struct audio_stream* s) {
- size_t chan_samp_sz;
- audio_format_t format = s->get_format(s);
-
- if (audio_has_proportional_frames(format)) {
- chan_samp_sz = audio_bytes_per_sample(format);
- return popcount(s->get_channels(s)) * chan_samp_sz;
- }
-
- return sizeof(int8_t);
-}
-
-/**
- * return the frame size (number of bytes per sample) of an output stream.
- */
-static inline size_t audio_stream_out_frame_size(const struct audio_stream_out* s) {
- size_t chan_samp_sz;
- audio_format_t format = s->common.get_format(&s->common);
-
- if (audio_has_proportional_frames(format)) {
- chan_samp_sz = audio_bytes_per_sample(format);
- return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
- }
-
- return sizeof(int8_t);
-}
-
-/**
- * return the frame size (number of bytes per sample) of an input stream.
- */
-static inline size_t audio_stream_in_frame_size(const struct audio_stream_in* s) {
- size_t chan_samp_sz;
- audio_format_t format = s->common.get_format(&s->common);
-
- if (audio_has_proportional_frames(format)) {
- chan_samp_sz = audio_bytes_per_sample(format);
- return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
- }
-
- return sizeof(int8_t);
-}
-
-/**********************************************************************/
-
-/**
- * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
- * and the fields of this data structure must begin with hw_module_t
- * followed by module specific information.
- */
-struct audio_module {
- struct hw_module_t common;
-};
-
-struct audio_hw_device {
- /**
- * Common methods of the audio device. This *must* be the first member of audio_hw_device
- * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts
- * where it's known the hw_device_t references an audio_hw_device.
- */
- struct hw_device_t common;
-
- /**
- * used by audio flinger to enumerate what devices are supported by
- * each audio_hw_device implementation.
- *
- * Return value is a bitmask of 1 or more values of audio_devices_t
- *
- * NOTE: audio HAL implementations starting with
- * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
- * All supported devices should be listed in audio_policy.conf
- * file and the audio policy manager must choose the appropriate
- * audio module based on information in this file.
- */
- uint32_t (*get_supported_devices)(const struct audio_hw_device* dev);
-
- /**
- * check to see if the audio hardware interface has been initialized.
- * returns 0 on success, -ENODEV on failure.
- */
- int (*init_check)(const struct audio_hw_device* dev);
-
- /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
- int (*set_voice_volume)(struct audio_hw_device* dev, float volume);
-
- /**
- * set the audio volume for all audio activities other than voice call.
- * Range between 0.0 and 1.0. If any value other than 0 is returned,
- * the software mixer will emulate this capability.
- */
- int (*set_master_volume)(struct audio_hw_device* dev, float volume);
-
- /**
- * Get the current master volume value for the HAL, if the HAL supports
- * master volume control. 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 may leave it set to NULL.
- */
- int (*get_master_volume)(struct audio_hw_device* dev, float* volume);
-
- /**
- * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
- * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
- * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
- */
- int (*set_mode)(struct audio_hw_device* dev, audio_mode_t mode);
-
- /* mic mute */
- int (*set_mic_mute)(struct audio_hw_device* dev, bool state);
- int (*get_mic_mute)(const struct audio_hw_device* dev, bool* state);
-
- /* set/get global audio parameters */
- int (*set_parameters)(struct audio_hw_device* dev, const char* kv_pairs);
-
- /*
- * Returns a pointer to a heap allocated string. The caller is responsible
- * for freeing the memory for it using free().
- */
- char* (*get_parameters)(const struct audio_hw_device* dev, const char* keys);
-
- /* Returns audio input buffer size according to parameters passed or
- * 0 if one of the parameters is not supported.
- * See also get_buffer_size which is for a particular stream.
- */
- size_t (*get_input_buffer_size)(const struct audio_hw_device* dev,
- const struct audio_config* config);
-
- /** This method creates and opens the audio hardware output stream.
- * The "address" parameter qualifies the "devices" audio device type if needed.
- * The format format depends on the device type:
- * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
- * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
- * - Other devices may use a number or any other string.
- */
-
- int (*open_output_stream)(struct audio_hw_device* dev, audio_io_handle_t handle,
- audio_devices_t devices, audio_output_flags_t flags,
- struct audio_config* config, struct audio_stream_out** stream_out,
- const char* address);
-
- void (*close_output_stream)(struct audio_hw_device* dev, struct audio_stream_out* stream_out);
-
- /** This method creates and opens the audio hardware input stream */
- int (*open_input_stream)(struct audio_hw_device* dev, audio_io_handle_t handle,
- audio_devices_t devices, struct audio_config* config,
- struct audio_stream_in** stream_in, audio_input_flags_t flags,
- const char* address, audio_source_t source);
-
- void (*close_input_stream)(struct audio_hw_device* dev, struct audio_stream_in* stream_in);
-
- /** This method dumps the state of the audio hardware */
- int (*dump)(const struct audio_hw_device* dev, int fd);
-
- /**
- * set the audio mute status for all audio activities. If any value other
- * than 0 is returned, the software mixer will emulate this capability.
- */
- int (*set_master_mute)(struct audio_hw_device* dev, bool mute);
-
- /**
- * 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. HALs which do not support this
- * method may leave it set to NULL.
- */
- int (*get_master_mute)(struct audio_hw_device* dev, bool* mute);
-
- /**
- * Routing control
- */
-
- /* Creates an audio patch between several source and sink ports.
- * The handle is allocated by the HAL and should be unique for this
- * audio HAL module. */
- int (*create_audio_patch)(struct audio_hw_device* dev, unsigned int num_sources,
- const struct audio_port_config* sources, unsigned int num_sinks,
- const struct audio_port_config* sinks, audio_patch_handle_t* handle);
-
- /* Release an audio patch */
- int (*release_audio_patch)(struct audio_hw_device* dev, audio_patch_handle_t handle);
-
- /* Fills 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, "port" contains possible attributes (sampling rates, formats,
- * channel masks, gain controllers...) for this port.
- */
- int (*get_audio_port)(struct audio_hw_device* dev, struct audio_port* port);
-
- /* Set audio port configuration */
- int (*set_audio_port_config)(struct audio_hw_device* dev,
- const struct audio_port_config* config);
-};
-typedef struct audio_hw_device audio_hw_device_t;
-
-/** convenience API for opening and closing a supported device */
-
-static inline int audio_hw_device_open(const struct hw_module_t* module,
- struct audio_hw_device** device) {
- return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, TO_HW_DEVICE_T_OPEN(device));
-}
-
-static inline int audio_hw_device_close(struct audio_hw_device* device) {
- return device->common.close(&device->common);
-}
-
-__END_DECLS
-
-#endif // ANDROID_AUDIO_INTERFACE_H
diff --git a/audio/common/2.0/legacy/include/system/audio-base.h b/audio/common/2.0/legacy/include/system/audio-base.h
deleted file mode 100644
index 53e524b..0000000
--- a/audio/common/2.0/legacy/include/system/audio-base.h
+++ /dev/null
@@ -1,434 +0,0 @@
-// This file is autogenerated by hidl-gen. Do not edit manually.
-// Source: android.hardware.audio.common@2.0
-// Root: android.hardware:hardware/interfaces
-
-#ifndef HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_
-#define HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-enum {
- AUDIO_IO_HANDLE_NONE = 0,
- AUDIO_MODULE_HANDLE_NONE = 0,
- AUDIO_PORT_HANDLE_NONE = 0,
- AUDIO_PATCH_HANDLE_NONE = 0,
-};
-
-typedef enum {
- AUDIO_STREAM_DEFAULT = -1, // (-1)
- AUDIO_STREAM_MIN = 0,
- AUDIO_STREAM_VOICE_CALL = 0,
- AUDIO_STREAM_SYSTEM = 1,
- AUDIO_STREAM_RING = 2,
- AUDIO_STREAM_MUSIC = 3,
- AUDIO_STREAM_ALARM = 4,
- AUDIO_STREAM_NOTIFICATION = 5,
- AUDIO_STREAM_BLUETOOTH_SCO = 6,
- AUDIO_STREAM_ENFORCED_AUDIBLE = 7,
- AUDIO_STREAM_DTMF = 8,
- AUDIO_STREAM_TTS = 9,
- AUDIO_STREAM_ACCESSIBILITY = 10,
- AUDIO_STREAM_REROUTING = 11,
- AUDIO_STREAM_PATCH = 12,
- AUDIO_STREAM_PUBLIC_CNT = 11, // (ACCESSIBILITY + 1)
- AUDIO_STREAM_FOR_POLICY_CNT = 12, // PATCH
- AUDIO_STREAM_CNT = 13, // (PATCH + 1)
-} audio_stream_type_t;
-
-typedef enum {
- AUDIO_SOURCE_DEFAULT = 0,
- AUDIO_SOURCE_MIC = 1,
- AUDIO_SOURCE_VOICE_UPLINK = 2,
- AUDIO_SOURCE_VOICE_DOWNLINK = 3,
- AUDIO_SOURCE_VOICE_CALL = 4,
- AUDIO_SOURCE_CAMCORDER = 5,
- AUDIO_SOURCE_VOICE_RECOGNITION = 6,
- AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
- AUDIO_SOURCE_REMOTE_SUBMIX = 8,
- AUDIO_SOURCE_UNPROCESSED = 9,
- AUDIO_SOURCE_CNT = 10,
- AUDIO_SOURCE_MAX = 9, // (CNT - 1)
- AUDIO_SOURCE_FM_TUNER = 1998,
- AUDIO_SOURCE_HOTWORD = 1999,
-} audio_source_t;
-
-typedef enum {
- AUDIO_SESSION_OUTPUT_STAGE = -1, // (-1)
- AUDIO_SESSION_OUTPUT_MIX = 0,
- AUDIO_SESSION_ALLOCATE = 0,
- AUDIO_SESSION_NONE = 0,
-} audio_session_t;
-
-typedef enum {
- AUDIO_FORMAT_INVALID = 4294967295u, // 0xFFFFFFFFUL
- AUDIO_FORMAT_DEFAULT = 0u, // 0
- AUDIO_FORMAT_PCM = 0u, // 0x00000000UL
- AUDIO_FORMAT_MP3 = 16777216u, // 0x01000000UL
- AUDIO_FORMAT_AMR_NB = 33554432u, // 0x02000000UL
- AUDIO_FORMAT_AMR_WB = 50331648u, // 0x03000000UL
- AUDIO_FORMAT_AAC = 67108864u, // 0x04000000UL
- AUDIO_FORMAT_HE_AAC_V1 = 83886080u, // 0x05000000UL
- AUDIO_FORMAT_HE_AAC_V2 = 100663296u, // 0x06000000UL
- AUDIO_FORMAT_VORBIS = 117440512u, // 0x07000000UL
- AUDIO_FORMAT_OPUS = 134217728u, // 0x08000000UL
- AUDIO_FORMAT_AC3 = 150994944u, // 0x09000000UL
- AUDIO_FORMAT_E_AC3 = 167772160u, // 0x0A000000UL
- AUDIO_FORMAT_DTS = 184549376u, // 0x0B000000UL
- AUDIO_FORMAT_DTS_HD = 201326592u, // 0x0C000000UL
- AUDIO_FORMAT_IEC61937 = 218103808u, // 0x0D000000UL
- AUDIO_FORMAT_DOLBY_TRUEHD = 234881024u, // 0x0E000000UL
- AUDIO_FORMAT_EVRC = 268435456u, // 0x10000000UL
- AUDIO_FORMAT_EVRCB = 285212672u, // 0x11000000UL
- AUDIO_FORMAT_EVRCWB = 301989888u, // 0x12000000UL
- AUDIO_FORMAT_EVRCNW = 318767104u, // 0x13000000UL
- AUDIO_FORMAT_AAC_ADIF = 335544320u, // 0x14000000UL
- AUDIO_FORMAT_WMA = 352321536u, // 0x15000000UL
- AUDIO_FORMAT_WMA_PRO = 369098752u, // 0x16000000UL
- AUDIO_FORMAT_AMR_WB_PLUS = 385875968u, // 0x17000000UL
- AUDIO_FORMAT_MP2 = 402653184u, // 0x18000000UL
- AUDIO_FORMAT_QCELP = 419430400u, // 0x19000000UL
- AUDIO_FORMAT_DSD = 436207616u, // 0x1A000000UL
- AUDIO_FORMAT_FLAC = 452984832u, // 0x1B000000UL
- AUDIO_FORMAT_ALAC = 469762048u, // 0x1C000000UL
- AUDIO_FORMAT_APE = 486539264u, // 0x1D000000UL
- AUDIO_FORMAT_AAC_ADTS = 503316480u, // 0x1E000000UL
- AUDIO_FORMAT_SBC = 520093696u, // 0x1F000000UL
- AUDIO_FORMAT_APTX = 536870912u, // 0x20000000UL
- AUDIO_FORMAT_APTX_HD = 553648128u, // 0x21000000UL
- AUDIO_FORMAT_AC4 = 570425344u, // 0x22000000UL
- AUDIO_FORMAT_LDAC = 587202560u, // 0x23000000UL
- AUDIO_FORMAT_MAIN_MASK = 4278190080u, // 0xFF000000UL
- AUDIO_FORMAT_SUB_MASK = 16777215u, // 0x00FFFFFFUL
- AUDIO_FORMAT_PCM_SUB_16_BIT = 1u, // 0x1
- AUDIO_FORMAT_PCM_SUB_8_BIT = 2u, // 0x2
- AUDIO_FORMAT_PCM_SUB_32_BIT = 3u, // 0x3
- AUDIO_FORMAT_PCM_SUB_8_24_BIT = 4u, // 0x4
- AUDIO_FORMAT_PCM_SUB_FLOAT = 5u, // 0x5
- AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED = 6u, // 0x6
- AUDIO_FORMAT_MP3_SUB_NONE = 0u, // 0x0
- AUDIO_FORMAT_AMR_SUB_NONE = 0u, // 0x0
- AUDIO_FORMAT_AAC_SUB_MAIN = 1u, // 0x1
- AUDIO_FORMAT_AAC_SUB_LC = 2u, // 0x2
- AUDIO_FORMAT_AAC_SUB_SSR = 4u, // 0x4
- AUDIO_FORMAT_AAC_SUB_LTP = 8u, // 0x8
- AUDIO_FORMAT_AAC_SUB_HE_V1 = 16u, // 0x10
- AUDIO_FORMAT_AAC_SUB_SCALABLE = 32u, // 0x20
- AUDIO_FORMAT_AAC_SUB_ERLC = 64u, // 0x40
- AUDIO_FORMAT_AAC_SUB_LD = 128u, // 0x80
- AUDIO_FORMAT_AAC_SUB_HE_V2 = 256u, // 0x100
- AUDIO_FORMAT_AAC_SUB_ELD = 512u, // 0x200
- AUDIO_FORMAT_VORBIS_SUB_NONE = 0u, // 0x0
- AUDIO_FORMAT_PCM_16_BIT = 1u, // (PCM | PCM_SUB_16_BIT)
- AUDIO_FORMAT_PCM_8_BIT = 2u, // (PCM | PCM_SUB_8_BIT)
- AUDIO_FORMAT_PCM_32_BIT = 3u, // (PCM | PCM_SUB_32_BIT)
- AUDIO_FORMAT_PCM_8_24_BIT = 4u, // (PCM | PCM_SUB_8_24_BIT)
- AUDIO_FORMAT_PCM_FLOAT = 5u, // (PCM | PCM_SUB_FLOAT)
- AUDIO_FORMAT_PCM_24_BIT_PACKED = 6u, // (PCM | PCM_SUB_24_BIT_PACKED)
- AUDIO_FORMAT_AAC_MAIN = 67108865u, // (AAC | AAC_SUB_MAIN)
- AUDIO_FORMAT_AAC_LC = 67108866u, // (AAC | AAC_SUB_LC)
- AUDIO_FORMAT_AAC_SSR = 67108868u, // (AAC | AAC_SUB_SSR)
- AUDIO_FORMAT_AAC_LTP = 67108872u, // (AAC | AAC_SUB_LTP)
- AUDIO_FORMAT_AAC_HE_V1 = 67108880u, // (AAC | AAC_SUB_HE_V1)
- AUDIO_FORMAT_AAC_SCALABLE = 67108896u, // (AAC | AAC_SUB_SCALABLE)
- AUDIO_FORMAT_AAC_ERLC = 67108928u, // (AAC | AAC_SUB_ERLC)
- AUDIO_FORMAT_AAC_LD = 67108992u, // (AAC | AAC_SUB_LD)
- AUDIO_FORMAT_AAC_HE_V2 = 67109120u, // (AAC | AAC_SUB_HE_V2)
- AUDIO_FORMAT_AAC_ELD = 67109376u, // (AAC | AAC_SUB_ELD)
- AUDIO_FORMAT_AAC_ADTS_MAIN = 503316481u, // (AAC_ADTS | AAC_SUB_MAIN)
- AUDIO_FORMAT_AAC_ADTS_LC = 503316482u, // (AAC_ADTS | AAC_SUB_LC)
- AUDIO_FORMAT_AAC_ADTS_SSR = 503316484u, // (AAC_ADTS | AAC_SUB_SSR)
- AUDIO_FORMAT_AAC_ADTS_LTP = 503316488u, // (AAC_ADTS | AAC_SUB_LTP)
- AUDIO_FORMAT_AAC_ADTS_HE_V1 = 503316496u, // (AAC_ADTS | AAC_SUB_HE_V1)
- AUDIO_FORMAT_AAC_ADTS_SCALABLE = 503316512u, // (AAC_ADTS | AAC_SUB_SCALABLE)
- AUDIO_FORMAT_AAC_ADTS_ERLC = 503316544u, // (AAC_ADTS | AAC_SUB_ERLC)
- AUDIO_FORMAT_AAC_ADTS_LD = 503316608u, // (AAC_ADTS | AAC_SUB_LD)
- AUDIO_FORMAT_AAC_ADTS_HE_V2 = 503316736u, // (AAC_ADTS | AAC_SUB_HE_V2)
- AUDIO_FORMAT_AAC_ADTS_ELD = 503316992u, // (AAC_ADTS | AAC_SUB_ELD)
-} audio_format_t;
-
-enum {
- FCC_2 = 2,
- FCC_8 = 8,
-};
-
-enum {
- AUDIO_CHANNEL_REPRESENTATION_POSITION = 0u, // 0
- AUDIO_CHANNEL_REPRESENTATION_INDEX = 2u, // 2
- AUDIO_CHANNEL_NONE = 0u, // 0x0
- AUDIO_CHANNEL_INVALID = 3221225472u, // 0xC0000000
- AUDIO_CHANNEL_OUT_FRONT_LEFT = 1u, // 0x1
- AUDIO_CHANNEL_OUT_FRONT_RIGHT = 2u, // 0x2
- AUDIO_CHANNEL_OUT_FRONT_CENTER = 4u, // 0x4
- AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 8u, // 0x8
- AUDIO_CHANNEL_OUT_BACK_LEFT = 16u, // 0x10
- AUDIO_CHANNEL_OUT_BACK_RIGHT = 32u, // 0x20
- AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 64u, // 0x40
- AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 128u, // 0x80
- AUDIO_CHANNEL_OUT_BACK_CENTER = 256u, // 0x100
- AUDIO_CHANNEL_OUT_SIDE_LEFT = 512u, // 0x200
- AUDIO_CHANNEL_OUT_SIDE_RIGHT = 1024u, // 0x400
- AUDIO_CHANNEL_OUT_TOP_CENTER = 2048u, // 0x800
- AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 4096u, // 0x1000
- AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 8192u, // 0x2000
- AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 16384u, // 0x4000
- AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 32768u, // 0x8000
- AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 65536u, // 0x10000
- AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 131072u, // 0x20000
- AUDIO_CHANNEL_OUT_MONO = 1u, // OUT_FRONT_LEFT
- AUDIO_CHANNEL_OUT_STEREO = 3u, // (OUT_FRONT_LEFT | OUT_FRONT_RIGHT)
- AUDIO_CHANNEL_OUT_2POINT1 = 11u, // ((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_LOW_FREQUENCY)
- AUDIO_CHANNEL_OUT_QUAD =
- 51u, // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_BACK_LEFT) | OUT_BACK_RIGHT)
- AUDIO_CHANNEL_OUT_QUAD_BACK = 51u, // OUT_QUAD
- AUDIO_CHANNEL_OUT_QUAD_SIDE =
- 1539u, // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_SIDE_LEFT) | OUT_SIDE_RIGHT)
- AUDIO_CHANNEL_OUT_SURROUND =
- 263u, // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) | OUT_BACK_CENTER)
- AUDIO_CHANNEL_OUT_PENTA = 55u, // (OUT_QUAD | OUT_FRONT_CENTER)
- AUDIO_CHANNEL_OUT_5POINT1 = 63u, // (((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER)
- // | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | OUT_BACK_RIGHT)
- AUDIO_CHANNEL_OUT_5POINT1_BACK = 63u, // OUT_5POINT1
- AUDIO_CHANNEL_OUT_5POINT1_SIDE = 1551u, // (((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) |
- // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) |
- // OUT_SIDE_LEFT) | OUT_SIDE_RIGHT)
- AUDIO_CHANNEL_OUT_6POINT1 = 319u, // ((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) |
- // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) |
- // OUT_BACK_RIGHT) | OUT_BACK_CENTER)
- AUDIO_CHANNEL_OUT_7POINT1 = 1599u, // (((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) |
- // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) |
- // OUT_BACK_RIGHT) | OUT_SIDE_LEFT) | OUT_SIDE_RIGHT)
- AUDIO_CHANNEL_OUT_ALL =
- 262143u, // (((((((((((((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) |
- // OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | OUT_BACK_RIGHT) |
- // OUT_FRONT_LEFT_OF_CENTER) | OUT_FRONT_RIGHT_OF_CENTER) | OUT_BACK_CENTER) |
- // OUT_SIDE_LEFT) | OUT_SIDE_RIGHT) | OUT_TOP_CENTER) | OUT_TOP_FRONT_LEFT) |
- // OUT_TOP_FRONT_CENTER) | OUT_TOP_FRONT_RIGHT) | OUT_TOP_BACK_LEFT) |
- // OUT_TOP_BACK_CENTER) | OUT_TOP_BACK_RIGHT)
- AUDIO_CHANNEL_IN_LEFT = 4u, // 0x4
- AUDIO_CHANNEL_IN_RIGHT = 8u, // 0x8
- AUDIO_CHANNEL_IN_FRONT = 16u, // 0x10
- AUDIO_CHANNEL_IN_BACK = 32u, // 0x20
- AUDIO_CHANNEL_IN_LEFT_PROCESSED = 64u, // 0x40
- AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 128u, // 0x80
- AUDIO_CHANNEL_IN_FRONT_PROCESSED = 256u, // 0x100
- AUDIO_CHANNEL_IN_BACK_PROCESSED = 512u, // 0x200
- AUDIO_CHANNEL_IN_PRESSURE = 1024u, // 0x400
- AUDIO_CHANNEL_IN_X_AXIS = 2048u, // 0x800
- AUDIO_CHANNEL_IN_Y_AXIS = 4096u, // 0x1000
- AUDIO_CHANNEL_IN_Z_AXIS = 8192u, // 0x2000
- AUDIO_CHANNEL_IN_VOICE_UPLINK = 16384u, // 0x4000
- AUDIO_CHANNEL_IN_VOICE_DNLINK = 32768u, // 0x8000
- AUDIO_CHANNEL_IN_MONO = 16u, // IN_FRONT
- AUDIO_CHANNEL_IN_STEREO = 12u, // (IN_LEFT | IN_RIGHT)
- AUDIO_CHANNEL_IN_FRONT_BACK = 48u, // (IN_FRONT | IN_BACK)
- AUDIO_CHANNEL_IN_6 = 252u, // (((((IN_LEFT | IN_RIGHT) | IN_FRONT) | IN_BACK) |
- // IN_LEFT_PROCESSED) | IN_RIGHT_PROCESSED)
- AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO = 16400u, // (IN_VOICE_UPLINK | IN_MONO)
- AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO = 32784u, // (IN_VOICE_DNLINK | IN_MONO)
- AUDIO_CHANNEL_IN_VOICE_CALL_MONO = 49168u, // (IN_VOICE_UPLINK_MONO | IN_VOICE_DNLINK_MONO)
- AUDIO_CHANNEL_IN_ALL =
- 65532u, // (((((((((((((IN_LEFT | IN_RIGHT) | IN_FRONT) | IN_BACK) | IN_LEFT_PROCESSED) |
- // IN_RIGHT_PROCESSED) | IN_FRONT_PROCESSED) | IN_BACK_PROCESSED) | IN_PRESSURE) |
- // IN_X_AXIS) | IN_Y_AXIS) | IN_Z_AXIS) | IN_VOICE_UPLINK) | IN_VOICE_DNLINK)
- AUDIO_CHANNEL_COUNT_MAX = 30u, // 30
- AUDIO_CHANNEL_INDEX_HDR = 2147483648u, // (REPRESENTATION_INDEX << COUNT_MAX)
- AUDIO_CHANNEL_INDEX_MASK_1 = 2147483649u, // (INDEX_HDR | ((1 << 1) - 1))
- AUDIO_CHANNEL_INDEX_MASK_2 = 2147483651u, // (INDEX_HDR | ((1 << 2) - 1))
- AUDIO_CHANNEL_INDEX_MASK_3 = 2147483655u, // (INDEX_HDR | ((1 << 3) - 1))
- AUDIO_CHANNEL_INDEX_MASK_4 = 2147483663u, // (INDEX_HDR | ((1 << 4) - 1))
- AUDIO_CHANNEL_INDEX_MASK_5 = 2147483679u, // (INDEX_HDR | ((1 << 5) - 1))
- AUDIO_CHANNEL_INDEX_MASK_6 = 2147483711u, // (INDEX_HDR | ((1 << 6) - 1))
- AUDIO_CHANNEL_INDEX_MASK_7 = 2147483775u, // (INDEX_HDR | ((1 << 7) - 1))
- AUDIO_CHANNEL_INDEX_MASK_8 = 2147483903u, // (INDEX_HDR | ((1 << 8) - 1))
-};
-
-enum {
- AUDIO_INTERLEAVE_LEFT = 0,
- AUDIO_INTERLEAVE_RIGHT = 1,
-};
-
-typedef enum {
- AUDIO_MODE_INVALID = -2, // (-2)
- AUDIO_MODE_CURRENT = -1, // (-1)
- AUDIO_MODE_NORMAL = 0,
- AUDIO_MODE_RINGTONE = 1,
- AUDIO_MODE_IN_CALL = 2,
- AUDIO_MODE_IN_COMMUNICATION = 3,
- AUDIO_MODE_CNT = 4,
- AUDIO_MODE_MAX = 3, // (CNT - 1)
-} audio_mode_t;
-
-enum {
- AUDIO_DEVICE_NONE = 0u, // 0x0
- AUDIO_DEVICE_BIT_IN = 2147483648u, // 0x80000000
- AUDIO_DEVICE_BIT_DEFAULT = 1073741824u, // 0x40000000
- AUDIO_DEVICE_OUT_EARPIECE = 1u, // 0x1
- AUDIO_DEVICE_OUT_SPEAKER = 2u, // 0x2
- AUDIO_DEVICE_OUT_WIRED_HEADSET = 4u, // 0x4
- AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 8u, // 0x8
- AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 16u, // 0x10
- AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 32u, // 0x20
- AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 64u, // 0x40
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 128u, // 0x80
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 256u, // 0x100
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 512u, // 0x200
- AUDIO_DEVICE_OUT_AUX_DIGITAL = 1024u, // 0x400
- AUDIO_DEVICE_OUT_HDMI = 1024u, // OUT_AUX_DIGITAL
- AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 2048u, // 0x800
- AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 4096u, // 0x1000
- AUDIO_DEVICE_OUT_USB_ACCESSORY = 8192u, // 0x2000
- AUDIO_DEVICE_OUT_USB_DEVICE = 16384u, // 0x4000
- AUDIO_DEVICE_OUT_REMOTE_SUBMIX = 32768u, // 0x8000
- AUDIO_DEVICE_OUT_TELEPHONY_TX = 65536u, // 0x10000
- AUDIO_DEVICE_OUT_LINE = 131072u, // 0x20000
- AUDIO_DEVICE_OUT_HDMI_ARC = 262144u, // 0x40000
- AUDIO_DEVICE_OUT_SPDIF = 524288u, // 0x80000
- AUDIO_DEVICE_OUT_FM = 1048576u, // 0x100000
- AUDIO_DEVICE_OUT_AUX_LINE = 2097152u, // 0x200000
- AUDIO_DEVICE_OUT_SPEAKER_SAFE = 4194304u, // 0x400000
- AUDIO_DEVICE_OUT_IP = 8388608u, // 0x800000
- AUDIO_DEVICE_OUT_BUS = 16777216u, // 0x1000000
- AUDIO_DEVICE_OUT_PROXY = 33554432u, // 0x2000000
- AUDIO_DEVICE_OUT_USB_HEADSET = 67108864u, // 0x4000000
- AUDIO_DEVICE_OUT_DEFAULT = 1073741824u, // BIT_DEFAULT
- AUDIO_DEVICE_OUT_ALL =
- 1207959551u, // (((((((((((((((((((((((((((OUT_EARPIECE | OUT_SPEAKER) | OUT_WIRED_HEADSET)
- // | OUT_WIRED_HEADPHONE) | OUT_BLUETOOTH_SCO) | OUT_BLUETOOTH_SCO_HEADSET) |
- // OUT_BLUETOOTH_SCO_CARKIT) | OUT_BLUETOOTH_A2DP) |
- // OUT_BLUETOOTH_A2DP_HEADPHONES) | OUT_BLUETOOTH_A2DP_SPEAKER) | OUT_HDMI) |
- // OUT_ANLG_DOCK_HEADSET) | OUT_DGTL_DOCK_HEADSET) | OUT_USB_ACCESSORY) |
- // OUT_USB_DEVICE) | OUT_REMOTE_SUBMIX) | OUT_TELEPHONY_TX) | OUT_LINE) |
- // OUT_HDMI_ARC) | OUT_SPDIF) | OUT_FM) | OUT_AUX_LINE) | OUT_SPEAKER_SAFE) |
- // OUT_IP) | OUT_BUS) | OUT_PROXY) | OUT_USB_HEADSET) | OUT_DEFAULT)
- AUDIO_DEVICE_OUT_ALL_A2DP = 896u, // ((OUT_BLUETOOTH_A2DP | OUT_BLUETOOTH_A2DP_HEADPHONES) |
- // OUT_BLUETOOTH_A2DP_SPEAKER)
- AUDIO_DEVICE_OUT_ALL_SCO =
- 112u, // ((OUT_BLUETOOTH_SCO | OUT_BLUETOOTH_SCO_HEADSET) | OUT_BLUETOOTH_SCO_CARKIT)
- AUDIO_DEVICE_OUT_ALL_USB =
- 67133440u, // ((OUT_USB_ACCESSORY | OUT_USB_DEVICE) | OUT_USB_HEADSET)
- AUDIO_DEVICE_IN_COMMUNICATION = 2147483649u, // (BIT_IN | 0x1)
- AUDIO_DEVICE_IN_AMBIENT = 2147483650u, // (BIT_IN | 0x2)
- AUDIO_DEVICE_IN_BUILTIN_MIC = 2147483652u, // (BIT_IN | 0x4)
- AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = 2147483656u, // (BIT_IN | 0x8)
- AUDIO_DEVICE_IN_WIRED_HEADSET = 2147483664u, // (BIT_IN | 0x10)
- AUDIO_DEVICE_IN_AUX_DIGITAL = 2147483680u, // (BIT_IN | 0x20)
- AUDIO_DEVICE_IN_HDMI = 2147483680u, // IN_AUX_DIGITAL
- AUDIO_DEVICE_IN_VOICE_CALL = 2147483712u, // (BIT_IN | 0x40)
- AUDIO_DEVICE_IN_TELEPHONY_RX = 2147483712u, // IN_VOICE_CALL
- AUDIO_DEVICE_IN_BACK_MIC = 2147483776u, // (BIT_IN | 0x80)
- AUDIO_DEVICE_IN_REMOTE_SUBMIX = 2147483904u, // (BIT_IN | 0x100)
- AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET = 2147484160u, // (BIT_IN | 0x200)
- AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET = 2147484672u, // (BIT_IN | 0x400)
- AUDIO_DEVICE_IN_USB_ACCESSORY = 2147485696u, // (BIT_IN | 0x800)
- AUDIO_DEVICE_IN_USB_DEVICE = 2147487744u, // (BIT_IN | 0x1000)
- AUDIO_DEVICE_IN_FM_TUNER = 2147491840u, // (BIT_IN | 0x2000)
- AUDIO_DEVICE_IN_TV_TUNER = 2147500032u, // (BIT_IN | 0x4000)
- AUDIO_DEVICE_IN_LINE = 2147516416u, // (BIT_IN | 0x8000)
- AUDIO_DEVICE_IN_SPDIF = 2147549184u, // (BIT_IN | 0x10000)
- AUDIO_DEVICE_IN_BLUETOOTH_A2DP = 2147614720u, // (BIT_IN | 0x20000)
- AUDIO_DEVICE_IN_LOOPBACK = 2147745792u, // (BIT_IN | 0x40000)
- AUDIO_DEVICE_IN_IP = 2148007936u, // (BIT_IN | 0x80000)
- AUDIO_DEVICE_IN_BUS = 2148532224u, // (BIT_IN | 0x100000)
- AUDIO_DEVICE_IN_PROXY = 2164260864u, // (BIT_IN | 0x1000000)
- AUDIO_DEVICE_IN_USB_HEADSET = 2181038080u, // (BIT_IN | 0x2000000)
- AUDIO_DEVICE_IN_DEFAULT = 3221225472u, // (BIT_IN | BIT_DEFAULT)
- AUDIO_DEVICE_IN_ALL =
- 3273654271u, // (((((((((((((((((((((((IN_COMMUNICATION | IN_AMBIENT) | IN_BUILTIN_MIC) |
- // IN_BLUETOOTH_SCO_HEADSET) | IN_WIRED_HEADSET) | IN_HDMI) | IN_TELEPHONY_RX)
- // | IN_BACK_MIC) | IN_REMOTE_SUBMIX) | IN_ANLG_DOCK_HEADSET) |
- // IN_DGTL_DOCK_HEADSET) | IN_USB_ACCESSORY) | IN_USB_DEVICE) | IN_FM_TUNER) |
- // IN_TV_TUNER) | IN_LINE) | IN_SPDIF) | IN_BLUETOOTH_A2DP) | IN_LOOPBACK) |
- // IN_IP) | IN_BUS) | IN_PROXY) | IN_USB_HEADSET) | IN_DEFAULT)
- AUDIO_DEVICE_IN_ALL_SCO = 2147483656u, // IN_BLUETOOTH_SCO_HEADSET
- AUDIO_DEVICE_IN_ALL_USB = 2181044224u, // ((IN_USB_ACCESSORY | IN_USB_DEVICE) | IN_USB_HEADSET)
-};
-
-typedef enum {
- AUDIO_OUTPUT_FLAG_NONE = 0, // 0x0
- AUDIO_OUTPUT_FLAG_DIRECT = 1, // 0x1
- AUDIO_OUTPUT_FLAG_PRIMARY = 2, // 0x2
- AUDIO_OUTPUT_FLAG_FAST = 4, // 0x4
- AUDIO_OUTPUT_FLAG_DEEP_BUFFER = 8, // 0x8
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD = 16, // 0x10
- AUDIO_OUTPUT_FLAG_NON_BLOCKING = 32, // 0x20
- AUDIO_OUTPUT_FLAG_HW_AV_SYNC = 64, // 0x40
- AUDIO_OUTPUT_FLAG_TTS = 128, // 0x80
- AUDIO_OUTPUT_FLAG_RAW = 256, // 0x100
- AUDIO_OUTPUT_FLAG_SYNC = 512, // 0x200
- AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO = 1024, // 0x400
- AUDIO_OUTPUT_FLAG_DIRECT_PCM = 8192, // 0x2000
- AUDIO_OUTPUT_FLAG_MMAP_NOIRQ = 16384, // 0x4000
- AUDIO_OUTPUT_FLAG_VOIP_RX = 32768, // 0x8000
-} audio_output_flags_t;
-
-typedef enum {
- AUDIO_INPUT_FLAG_NONE = 0, // 0x0
- AUDIO_INPUT_FLAG_FAST = 1, // 0x1
- AUDIO_INPUT_FLAG_HW_HOTWORD = 2, // 0x2
- AUDIO_INPUT_FLAG_RAW = 4, // 0x4
- AUDIO_INPUT_FLAG_SYNC = 8, // 0x8
- AUDIO_INPUT_FLAG_MMAP_NOIRQ = 16, // 0x10
- AUDIO_INPUT_FLAG_VOIP_TX = 32, // 0x20
-} audio_input_flags_t;
-
-typedef enum {
- AUDIO_USAGE_UNKNOWN = 0,
- AUDIO_USAGE_MEDIA = 1,
- AUDIO_USAGE_VOICE_COMMUNICATION = 2,
- AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3,
- AUDIO_USAGE_ALARM = 4,
- AUDIO_USAGE_NOTIFICATION = 5,
- AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE = 6,
- AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST = 7,
- AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT = 8,
- AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED = 9,
- AUDIO_USAGE_NOTIFICATION_EVENT = 10,
- AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11,
- AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12,
- AUDIO_USAGE_ASSISTANCE_SONIFICATION = 13,
- AUDIO_USAGE_GAME = 14,
- AUDIO_USAGE_VIRTUAL_SOURCE = 15,
- AUDIO_USAGE_ASSISTANT = 16,
- AUDIO_USAGE_CNT = 17,
- AUDIO_USAGE_MAX = 16, // (CNT - 1)
-} audio_usage_t;
-
-enum {
- AUDIO_GAIN_MODE_JOINT = 1u, // 0x1
- AUDIO_GAIN_MODE_CHANNELS = 2u, // 0x2
- AUDIO_GAIN_MODE_RAMP = 4u, // 0x4
-};
-
-typedef enum {
- AUDIO_PORT_ROLE_NONE = 0,
- AUDIO_PORT_ROLE_SOURCE = 1,
- AUDIO_PORT_ROLE_SINK = 2,
-} audio_port_role_t;
-
-typedef enum {
- AUDIO_PORT_TYPE_NONE = 0,
- AUDIO_PORT_TYPE_DEVICE = 1,
- AUDIO_PORT_TYPE_MIX = 2,
- AUDIO_PORT_TYPE_SESSION = 3,
-} audio_port_type_t;
-
-enum {
- AUDIO_PORT_CONFIG_SAMPLE_RATE = 1u, // 0x1
- AUDIO_PORT_CONFIG_CHANNEL_MASK = 2u, // 0x2
- AUDIO_PORT_CONFIG_FORMAT = 4u, // 0x4
- AUDIO_PORT_CONFIG_GAIN = 8u, // 0x8
- AUDIO_PORT_CONFIG_ALL = 15u, // (((SAMPLE_RATE | CHANNEL_MASK) | FORMAT) | GAIN)
-};
-
-typedef enum {
- AUDIO_LATENCY_LOW = 0,
- AUDIO_LATENCY_NORMAL = 1,
-} audio_mix_latency_class_t;
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_
diff --git a/audio/common/2.0/legacy/include/system/audio_effect-base.h b/audio/common/2.0/legacy/include/system/audio_effect-base.h
deleted file mode 100644
index cd17f55..0000000
--- a/audio/common/2.0/legacy/include/system/audio_effect-base.h
+++ /dev/null
@@ -1,101 +0,0 @@
-// This file is autogenerated by hidl-gen. Do not edit manually.
-// Source: android.hardware.audio.effect@2.0
-// Root: android.hardware:hardware/interfaces
-
-#ifndef HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_
-#define HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-enum {
- EFFECT_FLAG_TYPE_SHIFT = 0,
- EFFECT_FLAG_TYPE_SIZE = 3,
- EFFECT_FLAG_TYPE_MASK = 7, // (((1 << TYPE_SIZE) - 1) << TYPE_SHIFT)
- EFFECT_FLAG_TYPE_INSERT = 0, // (0 << TYPE_SHIFT)
- EFFECT_FLAG_TYPE_AUXILIARY = 1, // (1 << TYPE_SHIFT)
- EFFECT_FLAG_TYPE_REPLACE = 2, // (2 << TYPE_SHIFT)
- EFFECT_FLAG_TYPE_PRE_PROC = 3, // (3 << TYPE_SHIFT)
- EFFECT_FLAG_TYPE_POST_PROC = 4, // (4 << TYPE_SHIFT)
- EFFECT_FLAG_INSERT_SHIFT = 3, // (TYPE_SHIFT + TYPE_SIZE)
- EFFECT_FLAG_INSERT_SIZE = 3,
- EFFECT_FLAG_INSERT_MASK = 56, // (((1 << INSERT_SIZE) - 1) << INSERT_SHIFT)
- EFFECT_FLAG_INSERT_ANY = 0, // (0 << INSERT_SHIFT)
- EFFECT_FLAG_INSERT_FIRST = 8, // (1 << INSERT_SHIFT)
- EFFECT_FLAG_INSERT_LAST = 16, // (2 << INSERT_SHIFT)
- EFFECT_FLAG_INSERT_EXCLUSIVE = 24, // (3 << INSERT_SHIFT)
- EFFECT_FLAG_VOLUME_SHIFT = 6, // (INSERT_SHIFT + INSERT_SIZE)
- EFFECT_FLAG_VOLUME_SIZE = 3,
- EFFECT_FLAG_VOLUME_MASK = 448, // (((1 << VOLUME_SIZE) - 1) << VOLUME_SHIFT)
- EFFECT_FLAG_VOLUME_CTRL = 64, // (1 << VOLUME_SHIFT)
- EFFECT_FLAG_VOLUME_IND = 128, // (2 << VOLUME_SHIFT)
- EFFECT_FLAG_VOLUME_NONE = 0, // (0 << VOLUME_SHIFT)
- EFFECT_FLAG_DEVICE_SHIFT = 9, // (VOLUME_SHIFT + VOLUME_SIZE)
- EFFECT_FLAG_DEVICE_SIZE = 3,
- EFFECT_FLAG_DEVICE_MASK = 3584, // (((1 << DEVICE_SIZE) - 1) << DEVICE_SHIFT)
- EFFECT_FLAG_DEVICE_IND = 512, // (1 << DEVICE_SHIFT)
- EFFECT_FLAG_DEVICE_NONE = 0, // (0 << DEVICE_SHIFT)
- EFFECT_FLAG_INPUT_SHIFT = 12, // (DEVICE_SHIFT + DEVICE_SIZE)
- EFFECT_FLAG_INPUT_SIZE = 2,
- EFFECT_FLAG_INPUT_MASK = 12288, // (((1 << INPUT_SIZE) - 1) << INPUT_SHIFT)
- EFFECT_FLAG_INPUT_DIRECT = 4096, // (1 << INPUT_SHIFT)
- EFFECT_FLAG_INPUT_PROVIDER = 8192, // (2 << INPUT_SHIFT)
- EFFECT_FLAG_INPUT_BOTH = 12288, // (3 << INPUT_SHIFT)
- EFFECT_FLAG_OUTPUT_SHIFT = 14, // (INPUT_SHIFT + INPUT_SIZE)
- EFFECT_FLAG_OUTPUT_SIZE = 2,
- EFFECT_FLAG_OUTPUT_MASK = 49152, // (((1 << OUTPUT_SIZE) - 1) << OUTPUT_SHIFT)
- EFFECT_FLAG_OUTPUT_DIRECT = 16384, // (1 << OUTPUT_SHIFT)
- EFFECT_FLAG_OUTPUT_PROVIDER = 32768, // (2 << OUTPUT_SHIFT)
- EFFECT_FLAG_OUTPUT_BOTH = 49152, // (3 << OUTPUT_SHIFT)
- EFFECT_FLAG_HW_ACC_SHIFT = 16, // (OUTPUT_SHIFT + OUTPUT_SIZE)
- EFFECT_FLAG_HW_ACC_SIZE = 2,
- EFFECT_FLAG_HW_ACC_MASK = 196608, // (((1 << HW_ACC_SIZE) - 1) << HW_ACC_SHIFT)
- EFFECT_FLAG_HW_ACC_SIMPLE = 65536, // (1 << HW_ACC_SHIFT)
- EFFECT_FLAG_HW_ACC_TUNNEL = 131072, // (2 << HW_ACC_SHIFT)
- EFFECT_FLAG_AUDIO_MODE_SHIFT = 18, // (HW_ACC_SHIFT + HW_ACC_SIZE)
- EFFECT_FLAG_AUDIO_MODE_SIZE = 2,
- EFFECT_FLAG_AUDIO_MODE_MASK = 786432, // (((1 << AUDIO_MODE_SIZE) - 1) << AUDIO_MODE_SHIFT)
- EFFECT_FLAG_AUDIO_MODE_IND = 262144, // (1 << AUDIO_MODE_SHIFT)
- EFFECT_FLAG_AUDIO_MODE_NONE = 0, // (0 << AUDIO_MODE_SHIFT)
- EFFECT_FLAG_AUDIO_SOURCE_SHIFT = 20, // (AUDIO_MODE_SHIFT + AUDIO_MODE_SIZE)
- EFFECT_FLAG_AUDIO_SOURCE_SIZE = 2,
- EFFECT_FLAG_AUDIO_SOURCE_MASK =
- 3145728, // (((1 << AUDIO_SOURCE_SIZE) - 1) << AUDIO_SOURCE_SHIFT)
- EFFECT_FLAG_AUDIO_SOURCE_IND = 1048576, // (1 << AUDIO_SOURCE_SHIFT)
- EFFECT_FLAG_AUDIO_SOURCE_NONE = 0, // (0 << AUDIO_SOURCE_SHIFT)
- EFFECT_FLAG_OFFLOAD_SHIFT = 22, // (AUDIO_SOURCE_SHIFT + AUDIO_SOURCE_SIZE)
- EFFECT_FLAG_OFFLOAD_SIZE = 1,
- EFFECT_FLAG_OFFLOAD_MASK = 4194304, // (((1 << OFFLOAD_SIZE) - 1) << OFFLOAD_SHIFT)
- EFFECT_FLAG_OFFLOAD_SUPPORTED = 4194304, // (1 << OFFLOAD_SHIFT)
- EFFECT_FLAG_NO_PROCESS_SHIFT = 23, // (OFFLOAD_SHIFT + OFFLOAD_SIZE)
- EFFECT_FLAG_NO_PROCESS_SIZE = 1,
- EFFECT_FLAG_NO_PROCESS_MASK = 8388608, // (((1 << NO_PROCESS_SIZE) - 1) << NO_PROCESS_SHIFT)
- EFFECT_FLAG_NO_PROCESS = 8388608, // (1 << NO_PROCESS_SHIFT)
-};
-
-typedef enum {
- EFFECT_BUFFER_ACCESS_WRITE = 0,
- EFFECT_BUFFER_ACCESS_READ = 1,
- EFFECT_BUFFER_ACCESS_ACCUMULATE = 2,
-} effect_buffer_access_e;
-
-enum {
- EFFECT_CONFIG_BUFFER = 1, // 0x0001
- EFFECT_CONFIG_SMP_RATE = 2, // 0x0002
- EFFECT_CONFIG_CHANNELS = 4, // 0x0004
- EFFECT_CONFIG_FORMAT = 8, // 0x0008
- EFFECT_CONFIG_ACC_MODE = 16, // 0x0010
- EFFECT_CONFIG_ALL = 31, // ((((BUFFER | SMP_RATE) | CHANNELS) | FORMAT) | ACC_MODE)
-};
-
-typedef enum {
- EFFECT_FEATURE_AUX_CHANNELS = 0,
- EFFECT_FEATURE_CNT = 1,
-} effect_feature_e;
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_
diff --git a/audio/common/all-versions/default/Android.bp b/audio/common/all-versions/default/Android.bp
index 9b82f05..8f6b74c 100644
--- a/audio/common/all-versions/default/Android.bp
+++ b/audio/common/all-versions/default/Android.bp
@@ -16,7 +16,10 @@
cc_library_shared {
name: "android.hardware.audio.common-util",
defaults: ["hidl_defaults"],
- vendor: true,
+ vendor_available: true,
+ vndk: {
+ enabled: true,
+ },
srcs: [
"EffectMap.cpp",
],
@@ -30,7 +33,7 @@
],
header_libs: [
- "android.hardware.audio.common.legacy@2.0",
+ "libaudio_system_headers",
"libhardware_headers",
],
}
diff --git a/audio/common/all-versions/legacy/Android.bp b/audio/common/all-versions/legacy/Android.bp
deleted file mode 100644
index 2fb01dd..0000000
--- a/audio/common/all-versions/legacy/Android.bp
+++ /dev/null
@@ -1,8 +0,0 @@
-cc_library_headers {
- name: "android.hardware.audio.common.legacy@all-versions",
- vendor: true,
- export_include_dirs: ["include"],
- header_libs: ["libcutils_headers"],
- export_header_lib_headers: ["libcutils_headers"],
-}
-
diff --git a/audio/common/all-versions/legacy/OWNERS b/audio/common/all-versions/legacy/OWNERS
deleted file mode 100644
index 6fdc97c..0000000
--- a/audio/common/all-versions/legacy/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-elaurent@google.com
-krocard@google.com
-mnaganov@google.com
diff --git a/audio/common/all-versions/legacy/include/hardware/audio.h b/audio/common/all-versions/legacy/include/hardware/audio.h
deleted file mode 100644
index 1ad3e0e..0000000
--- a/audio/common/all-versions/legacy/include/hardware/audio.h
+++ /dev/null
@@ -1,709 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-#ifndef ANDROID_AUDIO_HAL_INTERFACE_H
-#define ANDROID_AUDIO_HAL_INTERFACE_H
-
-#include <stdint.h>
-#include <strings.h>
-#include <sys/cdefs.h>
-#include <sys/types.h>
-#include <time.h>
-
-#include <cutils/bitops.h>
-
-#include <hardware/audio_effect.h>
-#include <hardware/hardware.h>
-#include <system/audio.h>
-
-__BEGIN_DECLS
-
-/**
- * The id of this module
- */
-#define AUDIO_HARDWARE_MODULE_ID "audio"
-
-/**
- * Name of the audio devices to open
- */
-#define AUDIO_HARDWARE_INTERFACE "audio_hw_if"
-
-/* Use version 0.1 to be compatible with first generation of audio hw module with version_major
- * hardcoded to 1. No audio module API change.
- */
-#define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1)
-#define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1
-
-/* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0
- * will be considered of first generation API.
- */
-#define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0)
-#define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0)
-#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0)
-#define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0)
-#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_0
-/* Minimal audio HAL version supported by the audio framework */
-#define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0
-
-/**************************************/
-
-/**
- * standard audio parameters that the HAL may need to handle
- */
-
-/**
- * audio device parameters
- */
-
-/* TTY mode selection */
-#define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode"
-#define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off"
-#define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco"
-#define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco"
-#define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full"
-
-/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */
-#define AUDIO_PARAMETER_KEY_HAC "HACSetting"
-#define AUDIO_PARAMETER_VALUE_HAC_ON "ON"
-#define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF"
-
-/* A2DP sink address set by framework */
-#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
-
-/* A2DP source address set by framework */
-#define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"
-
-/* Bluetooth SCO wideband */
-#define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs"
-
-/**
- * audio stream parameters
- */
-
-/* Enable AANC */
-#define AUDIO_PARAMETER_KEY_AANC "aanc_enabled"
-
-/**************************************/
-
-/* common audio stream parameters and operations */
-struct audio_stream {
- /**
- * Return the sampling rate in Hz - eg. 44100.
- */
- uint32_t (*get_sample_rate)(const struct audio_stream* stream);
-
- /* currently unused - use set_parameters with key
- * AUDIO_PARAMETER_STREAM_SAMPLING_RATE
- */
- int (*set_sample_rate)(struct audio_stream* stream, uint32_t rate);
-
- /**
- * Return size of input/output buffer in bytes for this stream - eg. 4800.
- * It should be a multiple of the frame size. See also get_input_buffer_size.
- */
- size_t (*get_buffer_size)(const struct audio_stream* stream);
-
- /**
- * Return the channel mask -
- * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
- */
- audio_channel_mask_t (*get_channels)(const struct audio_stream* stream);
-
- /**
- * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
- */
- audio_format_t (*get_format)(const struct audio_stream* stream);
-
- /* currently unused - use set_parameters with key
- * AUDIO_PARAMETER_STREAM_FORMAT
- */
- int (*set_format)(struct audio_stream* stream, audio_format_t format);
-
- /**
- * Put the audio hardware input/output into standby mode.
- * Driver should exit from standby mode at the next I/O operation.
- * Returns 0 on success and <0 on failure.
- */
- int (*standby)(struct audio_stream* stream);
-
- /** dump the state of the audio input/output device */
- int (*dump)(const struct audio_stream* stream, int fd);
-
- /** Return the set of device(s) which this stream is connected to */
- audio_devices_t (*get_device)(const struct audio_stream* stream);
-
- /**
- * Currently unused - set_device() corresponds to set_parameters() with key
- * AUDIO_PARAMETER_STREAM_ROUTING for both input and output.
- * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by
- * input streams only.
- */
- int (*set_device)(struct audio_stream* stream, audio_devices_t device);
-
- /**
- * set/get audio stream parameters. The function accepts a list of
- * parameter key value pairs in the form: key1=value1;key2=value2;...
- *
- * Some keys are reserved for standard parameters (See AudioParameter class)
- *
- * If the implementation does not accept a parameter change while
- * the output is active but the parameter is acceptable otherwise, it must
- * return -ENOSYS.
- *
- * The audio flinger will put the stream in standby and then change the
- * parameter value.
- */
- int (*set_parameters)(struct audio_stream* stream, const char* kv_pairs);
-
- /*
- * Returns a pointer to a heap allocated string. The caller is responsible
- * for freeing the memory for it using free().
- */
- char* (*get_parameters)(const struct audio_stream* stream, const char* keys);
- int (*add_audio_effect)(const struct audio_stream* stream, effect_handle_t effect);
- int (*remove_audio_effect)(const struct audio_stream* stream, effect_handle_t effect);
-};
-typedef struct audio_stream audio_stream_t;
-
-/* type of asynchronous write callback events. Mutually exclusive */
-typedef enum {
- STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
- STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */
- STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */
-} stream_callback_event_t;
-
-typedef int (*stream_callback_t)(stream_callback_event_t event, void* param, void* cookie);
-
-/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
-typedef enum {
- AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
- AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
- from the current track has been played to
- give time for gapless track switch */
-} audio_drain_type_t;
-
-/**
- * audio_stream_out is the abstraction interface for the audio output hardware.
- *
- * It provides information about various properties of the audio output
- * hardware driver.
- */
-
-struct audio_stream_out {
- /**
- * Common methods of the audio stream out. This *must* be the first member of audio_stream_out
- * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
- * where it's known the audio_stream references an audio_stream_out.
- */
- struct audio_stream common;
-
- /**
- * Return the audio hardware driver estimated latency in milliseconds.
- */
- uint32_t (*get_latency)(const struct audio_stream_out* stream);
-
- /**
- * Use this method in situations where audio mixing is done in the
- * hardware. This method serves as a direct interface with hardware,
- * allowing you 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.
- */
- int (*set_volume)(struct audio_stream_out* stream, float left, float right);
-
- /**
- * Write audio buffer to driver. Returns number of bytes written, or a
- * negative status_t. If at least one frame was written successfully prior to the error,
- * it is suggested that the driver return that successful (short) byte count
- * and then return an error in the subsequent call.
- *
- * If set_callback() has previously been called to enable non-blocking mode
- * the write() is not allowed to block. It must write only the number of
- * bytes that currently fit in the driver/hardware buffer and then return
- * this byte count. If this is less than the requested write size the
- * callback function must be called when more space is available in the
- * driver/hardware buffer.
- */
- ssize_t (*write)(struct audio_stream_out* stream, const void* buffer, size_t bytes);
-
- /* return the number of audio frames written by the audio dsp to DAC since
- * the output has exited standby
- */
- int (*get_render_position)(const struct audio_stream_out* stream, uint32_t* dsp_frames);
-
- /**
- * 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.
- */
- int (*get_next_write_timestamp)(const struct audio_stream_out* stream, int64_t* timestamp);
-
- /**
- * set the callback function 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.
- */
- int (*set_callback)(struct audio_stream_out* stream, stream_callback_t callback, void* cookie);
-
- /**
- * 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 should be implemented for hardware with non-trivial
- * latency. In the pause state audio hardware could still be using power. User may
- * consider calling suspend after a timeout.
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*pause)(struct audio_stream_out* stream);
-
- /**
- * Notifies to the audio driver to resume playback following a pause.
- * Returns error if called without matching pause.
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*resume)(struct audio_stream_out* stream);
-
- /**
- * Requests notification when data buffered by the driver/hardware has
- * been played. If set_callback() has previously been called to enable
- * non-blocking mode, the drain() must not block, instead it should return
- * quickly and completion of the drain is notified through the callback.
- * If set_callback() has not been called, the drain() must block until
- * completion.
- * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
- * data has been played.
- * If type==AUDIO_DRAIN_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() call
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type);
-
- /**
- * Notifies to the audio driver to flush the queued data. Stream must already
- * be paused before calling flush().
- *
- * Implementation of this function is mandatory for offloaded playback.
- */
- int (*flush)(struct audio_stream_out* stream);
-
- /**
- * 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 should 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 should correspond to N rather than N+M.
- * The terms 'recent' and 'small' are not defined.
- * They reflect the quality of the implementation.
- *
- * 3.0 and higher only.
- */
- int (*get_presentation_position)(const struct audio_stream_out* stream, uint64_t* frames,
- struct timespec* timestamp);
-
- /**
- * Called by the framework to start a stream operating in mmap mode.
- * create_mmap_buffer must be called before calling start()
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case of success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*start)(const struct audio_stream_out* stream);
-
- /**
- * Called by the framework to stop a stream operating in mmap mode.
- * Must be called after start()
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case of success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*stop)(const struct audio_stream_out* stream);
-
- /**
- * Called by the framework to retrieve information on the mmap buffer used for audio
- * samples transfer.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[in] min_size_frames minimum buffer size requested. The actual buffer
- * size returned in struct audio_mmap_buffer_info can be larger.
- * \param[out] info address at which the mmap buffer information should be returned.
- *
- * \return 0 if the buffer was allocated.
- * -ENODEV in case of initialization error
- * -EINVAL if the requested buffer size is too large
- * -ENOSYS if called out of sequence (e.g. buffer already allocated)
- */
- int (*create_mmap_buffer)(const struct audio_stream_out* stream, int32_t min_size_frames,
- struct audio_mmap_buffer_info* info);
-
- /**
- * Called by the framework to read current read/write position in the mmap buffer
- * with associated time stamp.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[out] position address at which the mmap read/write position should be returned.
- *
- * \return 0 if the position is successfully returned.
- * -ENODATA if the position cannot be retrieved
- * -ENOSYS if called before create_mmap_buffer()
- */
- int (*get_mmap_position)(const struct audio_stream_out* stream,
- struct audio_mmap_position* position);
-};
-typedef struct audio_stream_out audio_stream_out_t;
-
-struct audio_stream_in {
- /**
- * Common methods of the audio stream in. This *must* be the first member of audio_stream_in
- * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
- * where it's known the audio_stream references an audio_stream_in.
- */
- struct audio_stream common;
-
- /** set the input gain for the audio driver. This method is for
- * for future use */
- int (*set_gain)(struct audio_stream_in* stream, float gain);
-
- /** Read audio buffer in from audio driver. Returns number of bytes read, or a
- * negative status_t. If at least one frame was read prior to the error,
- * read should return that byte count and then return an error in the subsequent call.
- */
- ssize_t (*read)(struct audio_stream_in* stream, void* buffer, size_t bytes);
-
- /**
- * 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.
- *
- * Unit: the number of input audio frames
- */
- uint32_t (*get_input_frames_lost)(struct audio_stream_in* stream);
-
- /**
- * Return a recent count of the number of audio frames received and
- * the clock time associated with that frame count.
- *
- * frames is the total frame count received. This should be as early in
- * the capture pipeline as possible. In general,
- * frames should be non-negative and should not go "backwards".
- *
- * time is the clock MONOTONIC time when frames was measured. In general,
- * time should be a positive quantity and should not go "backwards".
- *
- * The status returned is 0 on success, -ENOSYS if the device is not
- * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
- */
- int (*get_capture_position)(const struct audio_stream_in* stream, int64_t* frames,
- int64_t* time);
-
- /**
- * Called by the framework to start a stream operating in mmap mode.
- * create_mmap_buffer must be called before calling start()
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case off success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*start)(const struct audio_stream_in* stream);
-
- /**
- * Called by the framework to stop a stream operating in mmap mode.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \return 0 in case of success.
- * -ENOSYS if called out of sequence or on non mmap stream
- */
- int (*stop)(const struct audio_stream_in* stream);
-
- /**
- * Called by the framework to retrieve information on the mmap buffer used for audio
- * samples transfer.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[in] min_size_frames minimum buffer size requested. The actual buffer
- * size returned in struct audio_mmap_buffer_info can be larger.
- * \param[out] info address at which the mmap buffer information should be returned.
- *
- * \return 0 if the buffer was allocated.
- * -ENODEV in case of initialization error
- * -EINVAL if the requested buffer size is too large
- * -ENOSYS if called out of sequence (e.g. buffer already allocated)
- */
- int (*create_mmap_buffer)(const struct audio_stream_in* stream, int32_t min_size_frames,
- struct audio_mmap_buffer_info* info);
-
- /**
- * Called by the framework to read current read/write position in the mmap buffer
- * with associated time stamp.
- *
- * \note Function only implemented by streams operating in mmap mode.
- *
- * \param[in] stream the stream object.
- * \param[out] position address at which the mmap read/write position should be returned.
- *
- * \return 0 if the position is successfully returned.
- * -ENODATA if the position cannot be retreived
- * -ENOSYS if called before mmap_read_position()
- */
- int (*get_mmap_position)(const struct audio_stream_in* stream,
- struct audio_mmap_position* position);
-};
-typedef struct audio_stream_in audio_stream_in_t;
-
-/**
- * return the frame size (number of bytes per sample).
- *
- * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead.
- */
-__attribute__((__deprecated__)) static inline size_t audio_stream_frame_size(
- const struct audio_stream* s) {
- size_t chan_samp_sz;
- audio_format_t format = s->get_format(s);
-
- if (audio_has_proportional_frames(format)) {
- chan_samp_sz = audio_bytes_per_sample(format);
- return popcount(s->get_channels(s)) * chan_samp_sz;
- }
-
- return sizeof(int8_t);
-}
-
-/**
- * return the frame size (number of bytes per sample) of an output stream.
- */
-static inline size_t audio_stream_out_frame_size(const struct audio_stream_out* s) {
- size_t chan_samp_sz;
- audio_format_t format = s->common.get_format(&s->common);
-
- if (audio_has_proportional_frames(format)) {
- chan_samp_sz = audio_bytes_per_sample(format);
- return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
- }
-
- return sizeof(int8_t);
-}
-
-/**
- * return the frame size (number of bytes per sample) of an input stream.
- */
-static inline size_t audio_stream_in_frame_size(const struct audio_stream_in* s) {
- size_t chan_samp_sz;
- audio_format_t format = s->common.get_format(&s->common);
-
- if (audio_has_proportional_frames(format)) {
- chan_samp_sz = audio_bytes_per_sample(format);
- return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
- }
-
- return sizeof(int8_t);
-}
-
-/**********************************************************************/
-
-/**
- * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
- * and the fields of this data structure must begin with hw_module_t
- * followed by module specific information.
- */
-struct audio_module {
- struct hw_module_t common;
-};
-
-struct audio_hw_device {
- /**
- * Common methods of the audio device. This *must* be the first member of audio_hw_device
- * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts
- * where it's known the hw_device_t references an audio_hw_device.
- */
- struct hw_device_t common;
-
- /**
- * used by audio flinger to enumerate what devices are supported by
- * each audio_hw_device implementation.
- *
- * Return value is a bitmask of 1 or more values of audio_devices_t
- *
- * NOTE: audio HAL implementations starting with
- * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
- * All supported devices should be listed in audio_policy.conf
- * file and the audio policy manager must choose the appropriate
- * audio module based on information in this file.
- */
- uint32_t (*get_supported_devices)(const struct audio_hw_device* dev);
-
- /**
- * check to see if the audio hardware interface has been initialized.
- * returns 0 on success, -ENODEV on failure.
- */
- int (*init_check)(const struct audio_hw_device* dev);
-
- /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
- int (*set_voice_volume)(struct audio_hw_device* dev, float volume);
-
- /**
- * set the audio volume for all audio activities other than voice call.
- * Range between 0.0 and 1.0. If any value other than 0 is returned,
- * the software mixer will emulate this capability.
- */
- int (*set_master_volume)(struct audio_hw_device* dev, float volume);
-
- /**
- * Get the current master volume value for the HAL, if the HAL supports
- * master volume control. 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 may leave it set to NULL.
- */
- int (*get_master_volume)(struct audio_hw_device* dev, float* volume);
-
- /**
- * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
- * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
- * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
- */
- int (*set_mode)(struct audio_hw_device* dev, audio_mode_t mode);
-
- /* mic mute */
- int (*set_mic_mute)(struct audio_hw_device* dev, bool state);
- int (*get_mic_mute)(const struct audio_hw_device* dev, bool* state);
-
- /* set/get global audio parameters */
- int (*set_parameters)(struct audio_hw_device* dev, const char* kv_pairs);
-
- /*
- * Returns a pointer to a heap allocated string. The caller is responsible
- * for freeing the memory for it using free().
- */
- char* (*get_parameters)(const struct audio_hw_device* dev, const char* keys);
-
- /* Returns audio input buffer size according to parameters passed or
- * 0 if one of the parameters is not supported.
- * See also get_buffer_size which is for a particular stream.
- */
- size_t (*get_input_buffer_size)(const struct audio_hw_device* dev,
- const struct audio_config* config);
-
- /** This method creates and opens the audio hardware output stream.
- * The "address" parameter qualifies the "devices" audio device type if needed.
- * The format format depends on the device type:
- * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
- * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
- * - Other devices may use a number or any other string.
- */
-
- int (*open_output_stream)(struct audio_hw_device* dev, audio_io_handle_t handle,
- audio_devices_t devices, audio_output_flags_t flags,
- struct audio_config* config, struct audio_stream_out** stream_out,
- const char* address);
-
- void (*close_output_stream)(struct audio_hw_device* dev, struct audio_stream_out* stream_out);
-
- /** This method creates and opens the audio hardware input stream */
- int (*open_input_stream)(struct audio_hw_device* dev, audio_io_handle_t handle,
- audio_devices_t devices, struct audio_config* config,
- struct audio_stream_in** stream_in, audio_input_flags_t flags,
- const char* address, audio_source_t source);
-
- void (*close_input_stream)(struct audio_hw_device* dev, struct audio_stream_in* stream_in);
-
- /** This method dumps the state of the audio hardware */
- int (*dump)(const struct audio_hw_device* dev, int fd);
-
- /**
- * set the audio mute status for all audio activities. If any value other
- * than 0 is returned, the software mixer will emulate this capability.
- */
- int (*set_master_mute)(struct audio_hw_device* dev, bool mute);
-
- /**
- * 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. HALs which do not support this
- * method may leave it set to NULL.
- */
- int (*get_master_mute)(struct audio_hw_device* dev, bool* mute);
-
- /**
- * Routing control
- */
-
- /* Creates an audio patch between several source and sink ports.
- * The handle is allocated by the HAL and should be unique for this
- * audio HAL module. */
- int (*create_audio_patch)(struct audio_hw_device* dev, unsigned int num_sources,
- const struct audio_port_config* sources, unsigned int num_sinks,
- const struct audio_port_config* sinks, audio_patch_handle_t* handle);
-
- /* Release an audio patch */
- int (*release_audio_patch)(struct audio_hw_device* dev, audio_patch_handle_t handle);
-
- /* Fills 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, "port" contains possible attributes (sampling rates, formats,
- * channel masks, gain controllers...) for this port.
- */
- int (*get_audio_port)(struct audio_hw_device* dev, struct audio_port* port);
-
- /* Set audio port configuration */
- int (*set_audio_port_config)(struct audio_hw_device* dev,
- const struct audio_port_config* config);
-};
-typedef struct audio_hw_device audio_hw_device_t;
-
-/** convenience API for opening and closing a supported device */
-
-static inline int audio_hw_device_open(const struct hw_module_t* module,
- struct audio_hw_device** device) {
- return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, TO_HW_DEVICE_T_OPEN(device));
-}
-
-static inline int audio_hw_device_close(struct audio_hw_device* device) {
- return device->common.close(&device->common);
-}
-
-__END_DECLS
-
-#endif // ANDROID_AUDIO_INTERFACE_H
diff --git a/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h b/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h
deleted file mode 100644
index aa16654..0000000
--- a/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright (C) 2014 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.
- */
-
-/* This file contains shared utility functions to handle the tinyalsa
- * implementation for Android internal audio, generally in the hardware layer.
- * Some routines may log a fatal error on failure, as noted.
- */
-
-#ifndef ANDROID_AUDIO_ALSAOPS_H
-#define ANDROID_AUDIO_ALSAOPS_H
-
-#include <log/log.h>
-
-#include <system/audio.h>
-#include <tinyalsa/asoundlib.h>
-
-__BEGIN_DECLS
-
-/* Converts audio_format to pcm_format.
- * Parameters:
- * format the audio_format_t to convert
- *
- * Logs a fatal error if format is not a valid convertible audio_format_t.
- */
-static inline enum pcm_format pcm_format_from_audio_format(audio_format_t format) {
- switch (format) {
-#if HAVE_BIG_ENDIAN
- case AUDIO_FORMAT_PCM_16_BIT:
- return PCM_FORMAT_S16_BE;
- case AUDIO_FORMAT_PCM_24_BIT_PACKED:
- return PCM_FORMAT_S24_3BE;
- case AUDIO_FORMAT_PCM_32_BIT:
- return PCM_FORMAT_S32_BE;
- case AUDIO_FORMAT_PCM_8_24_BIT:
- return PCM_FORMAT_S24_BE;
-#else
- case AUDIO_FORMAT_PCM_16_BIT:
- return PCM_FORMAT_S16_LE;
- case AUDIO_FORMAT_PCM_24_BIT_PACKED:
- return PCM_FORMAT_S24_3LE;
- case AUDIO_FORMAT_PCM_32_BIT:
- return PCM_FORMAT_S32_LE;
- case AUDIO_FORMAT_PCM_8_24_BIT:
- return PCM_FORMAT_S24_LE;
-#endif
- case AUDIO_FORMAT_PCM_FLOAT: /* there is no equivalent for float */
- default:
- LOG_ALWAYS_FATAL("pcm_format_from_audio_format: invalid audio format %#x", format);
- return 0;
- }
-}
-
-/* Converts pcm_format to audio_format.
- * Parameters:
- * format the pcm_format to convert
- *
- * Logs a fatal error if format is not a valid convertible pcm_format.
- */
-static inline audio_format_t audio_format_from_pcm_format(enum pcm_format format) {
- switch (format) {
-#if HAVE_BIG_ENDIAN
- case PCM_FORMAT_S16_BE:
- return AUDIO_FORMAT_PCM_16_BIT;
- case PCM_FORMAT_S24_3BE:
- return AUDIO_FORMAT_PCM_24_BIT_PACKED;
- case PCM_FORMAT_S24_BE:
- return AUDIO_FORMAT_PCM_8_24_BIT;
- case PCM_FORMAT_S32_BE:
- return AUDIO_FORMAT_PCM_32_BIT;
-#else
- case PCM_FORMAT_S16_LE:
- return AUDIO_FORMAT_PCM_16_BIT;
- case PCM_FORMAT_S24_3LE:
- return AUDIO_FORMAT_PCM_24_BIT_PACKED;
- case PCM_FORMAT_S24_LE:
- return AUDIO_FORMAT_PCM_8_24_BIT;
- case PCM_FORMAT_S32_LE:
- return AUDIO_FORMAT_PCM_32_BIT;
-#endif
- default:
- LOG_ALWAYS_FATAL("audio_format_from_pcm_format: invalid pcm format %#x", format);
- return 0;
- }
-}
-
-__END_DECLS
-
-#endif /* ANDROID_AUDIO_ALSAOPS_H */
diff --git a/audio/common/all-versions/legacy/include/hardware/audio_effect.h b/audio/common/all-versions/legacy/include/hardware/audio_effect.h
deleted file mode 100644
index b91c60a..0000000
--- a/audio/common/all-versions/legacy/include/hardware/audio_effect.h
+++ /dev/null
@@ -1,295 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-#ifndef ANDROID_AUDIO_EFFECT_H
-#define ANDROID_AUDIO_EFFECT_H
-
-#include <errno.h>
-#include <stdint.h>
-#include <strings.h>
-#include <sys/cdefs.h>
-#include <sys/types.h>
-
-#include <cutils/bitops.h>
-
-#include <system/audio_effect.h>
-
-__BEGIN_DECLS
-
-/////////////////////////////////////////////////
-// Common Definitions
-/////////////////////////////////////////////////
-
-#define EFFECT_MAKE_API_VERSION(M, m) (((M) << 16) | ((m)&0xFFFF))
-#define EFFECT_API_VERSION_MAJOR(v) ((v) >> 16)
-#define EFFECT_API_VERSION_MINOR(v) ((m)&0xFFFF)
-
-/////////////////////////////////////////////////
-// Effect control interface
-/////////////////////////////////////////////////
-
-// Effect control interface version 2.0
-#define EFFECT_CONTROL_API_VERSION EFFECT_MAKE_API_VERSION(2, 0)
-
-// Effect control interface structure: effect_interface_s
-// The effect control interface is exposed by each effect engine implementation. It consists of
-// a set of functions controlling the configuration, activation and process of the engine.
-// The functions are grouped in a structure of type effect_interface_s.
-//
-// Effect control interface handle: effect_handle_t
-// The effect_handle_t serves two purposes regarding the implementation of the effect engine:
-// - 1 it is the address of a pointer to an effect_interface_s structure where the functions
-// of the effect control API for a particular effect are located.
-// - 2 it is the address of the context of a particular effect instance.
-// A typical implementation in the effect library would define a structure as follows:
-// struct effect_module_s {
-// const struct effect_interface_s *itfe;
-// effect_config_t config;
-// effect_context_t context;
-// }
-// The implementation of EffectCreate() function would then allocate a structure of this
-// type and return its address as effect_handle_t
-typedef struct effect_interface_s** effect_handle_t;
-
-// Effect control interface definition
-struct effect_interface_s {
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: process
- //
- // Description: Effect process function. Takes input samples as specified
- // (count and location) in input buffer descriptor and output processed
- // samples as specified in output buffer descriptor. If the buffer descriptor
- // is not specified the function must use either the buffer or the
- // buffer provider function installed by the EFFECT_CMD_SET_CONFIG command.
- // The effect framework will call the process() function after the EFFECT_CMD_ENABLE
- // command is received and until the EFFECT_CMD_DISABLE is received. When the engine
- // receives the EFFECT_CMD_DISABLE command it should turn off the effect gracefully
- // and when done indicate that it is OK to stop calling the process() function by
- // returning the -ENODATA status.
- //
- // NOTE: the process() function implementation should be "real-time safe" that is
- // it should not perform blocking calls: malloc/free, sleep, read/write/open/close,
- // pthread_cond_wait/pthread_mutex_lock...
- //
- // Input:
- // self: handle to the effect interface this function
- // is called on.
- // inBuffer: buffer descriptor indicating where to read samples to process.
- // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command.
- //
- // outBuffer: buffer descriptor indicating where to write processed samples.
- // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command.
- //
- // Output:
- // returned value: 0 successful operation
- // -ENODATA the engine has finished the disable phase and the framework
- // can stop calling process()
- // -EINVAL invalid interface handle or
- // invalid input/output buffer description
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*process)(effect_handle_t self, audio_buffer_t* inBuffer, audio_buffer_t* outBuffer);
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: command
- //
- // Description: Send a command and receive a response to/from effect engine.
- //
- // Input:
- // self: handle to the effect interface this function
- // is called on.
- // cmdCode: command code: the command can be a standardized command defined in
- // effect_command_e (see below) or a proprietary command.
- // cmdSize: size of command in bytes
- // pCmdData: pointer to command data
- // pReplyData: pointer to reply data
- //
- // Input/Output:
- // replySize: maximum size of reply data as input
- // actual size of reply data as output
- //
- // Output:
- // returned value: 0 successful operation
- // -EINVAL invalid interface handle or
- // invalid command/reply size or format according to
- // command code
- // The return code should be restricted to indicate problems related to this API
- // specification. Status related to the execution of a particular command should be
- // indicated as part of the reply field.
- //
- // *pReplyData updated with command response
- //
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*command)(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize, void* pCmdData,
- uint32_t* replySize, void* pReplyData);
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: get_descriptor
- //
- // Description: Returns the effect descriptor
- //
- // Input:
- // self: handle to the effect interface this function
- // is called on.
- //
- // Input/Output:
- // pDescriptor: address where to return the effect descriptor.
- //
- // Output:
- // returned value: 0 successful operation.
- // -EINVAL invalid interface handle or invalid pDescriptor
- // *pDescriptor: updated with the effect descriptor.
- //
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*get_descriptor)(effect_handle_t self, effect_descriptor_t* pDescriptor);
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: process_reverse
- //
- // Description: Process reverse stream function. This function is used to pass
- // a reference stream to the effect engine. If the engine does not need a reference
- // stream, this function pointer can be set to NULL.
- // This function would typically implemented by an Echo Canceler.
- //
- // Input:
- // self: handle to the effect interface this function
- // is called on.
- // inBuffer: buffer descriptor indicating where to read samples to process.
- // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command.
- //
- // outBuffer: buffer descriptor indicating where to write processed samples.
- // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command.
- // If the buffer and buffer provider in the configuration received by
- // EFFECT_CMD_SET_CONFIG_REVERSE are also NULL, do not return modified reverse
- // stream data
- //
- // Output:
- // returned value: 0 successful operation
- // -ENODATA the engine has finished the disable phase and the framework
- // can stop calling process_reverse()
- // -EINVAL invalid interface handle or
- // invalid input/output buffer description
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*process_reverse)(effect_handle_t self, audio_buffer_t* inBuffer,
- audio_buffer_t* outBuffer);
-};
-
-/////////////////////////////////////////////////
-// Effect library interface
-/////////////////////////////////////////////////
-
-// Effect library interface version 3.0
-// Note that EffectsFactory.c only checks the major version component, so changes to the minor
-// number can only be used for fully backwards compatible changes
-#define EFFECT_LIBRARY_API_VERSION EFFECT_MAKE_API_VERSION(3, 0)
-
-#define AUDIO_EFFECT_LIBRARY_TAG ((('A') << 24) | (('E') << 16) | (('L') << 8) | ('T'))
-
-// Every effect library must have a data structure named AUDIO_EFFECT_LIBRARY_INFO_SYM
-// and the fields of this data structure must begin with audio_effect_library_t
-
-typedef struct audio_effect_library_s {
- // tag must be initialized to AUDIO_EFFECT_LIBRARY_TAG
- uint32_t tag;
- // Version of the effect library API : 0xMMMMmmmm MMMM: Major, mmmm: minor
- uint32_t version;
- // Name of this library
- const char* name;
- // Author/owner/implementor of the library
- const char* implementor;
-
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: create_effect
- //
- // Description: Creates an effect engine of the specified implementation uuid and
- // returns an effect control interface on this engine. The function will allocate the
- // resources for an instance of the requested effect engine and return
- // a handle on the effect control interface.
- //
- // Input:
- // uuid: pointer to the effect uuid.
- // sessionId: 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. Knowing that two effects are part of the same effect
- // chain can help the library implement some kind of optimizations.
- // ioId: identifies the output or input stream this effect is directed to in
- // audio HAL.
- // For future use especially with tunneled HW accelerated effects
- //
- // Input/Output:
- // pHandle: address where to return the effect interface handle.
- //
- // Output:
- // returned value: 0 successful operation.
- // -ENODEV library failed to initialize
- // -EINVAL invalid pEffectUuid or pHandle
- // -ENOENT no effect with this uuid found
- // *pHandle: updated with the effect interface handle.
- //
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*create_effect)(const effect_uuid_t* uuid, int32_t sessionId, int32_t ioId,
- effect_handle_t* pHandle);
-
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: release_effect
- //
- // Description: Releases the effect engine whose handle is given as argument.
- // All resources allocated to this particular instance of the effect are
- // released.
- //
- // Input:
- // handle: handle on the effect interface to be released.
- //
- // Output:
- // returned value: 0 successful operation.
- // -ENODEV library failed to initialize
- // -EINVAL invalid interface handle
- //
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*release_effect)(effect_handle_t handle);
-
- ////////////////////////////////////////////////////////////////////////////////
- //
- // Function: get_descriptor
- //
- // Description: Returns the descriptor of the effect engine which implementation UUID is
- // given as argument.
- //
- // Input/Output:
- // uuid: pointer to the effect uuid.
- // pDescriptor: address where to return the effect descriptor.
- //
- // Output:
- // returned value: 0 successful operation.
- // -ENODEV library failed to initialize
- // -EINVAL invalid pDescriptor or uuid
- // *pDescriptor: updated with the effect descriptor.
- //
- ////////////////////////////////////////////////////////////////////////////////
- int32_t (*get_descriptor)(const effect_uuid_t* uuid, effect_descriptor_t* pDescriptor);
-} audio_effect_library_t;
-
-// Name of the hal_module_info
-#define AUDIO_EFFECT_LIBRARY_INFO_SYM AELI
-
-// Name of the hal_module_info as a string
-#define AUDIO_EFFECT_LIBRARY_INFO_SYM_AS_STR "AELI"
-
-__END_DECLS
-
-#endif // ANDROID_AUDIO_EFFECT_H
diff --git a/audio/common/all-versions/legacy/include/hardware/audio_policy.h b/audio/common/all-versions/legacy/include/hardware/audio_policy.h
deleted file mode 100644
index 8cc79df..0000000
--- a/audio/common/all-versions/legacy/include/hardware/audio_policy.h
+++ /dev/null
@@ -1,391 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-#ifndef ANDROID_AUDIO_POLICY_INTERFACE_H
-#define ANDROID_AUDIO_POLICY_INTERFACE_H
-
-#include <stdint.h>
-#include <sys/cdefs.h>
-#include <sys/types.h>
-
-#include <hardware/hardware.h>
-
-#include <system/audio.h>
-#include <system/audio_policy.h>
-
-__BEGIN_DECLS
-
-/**
- * The id of this module
- */
-#define AUDIO_POLICY_HARDWARE_MODULE_ID "audio_policy"
-
-/**
- * Name of the audio devices to open
- */
-#define AUDIO_POLICY_INTERFACE "policy"
-
-/* ---------------------------------------------------------------------------- */
-
-/*
- * The audio_policy and audio_policy_service_ops structs define the
- * communication interfaces between the platform specific audio policy manager
- * and Android generic audio policy manager.
- * The platform specific audio policy manager must implement methods of the
- * audio_policy struct.
- * This implementation makes use of the audio_policy_service_ops to control
- * the activity and configuration of audio input and output streams.
- *
- * The platform specific audio policy manager is in charge of the audio
- * routing and volume control policies for a given platform.
- * The main roles of this module are:
- * - keep track of current system state (removable device connections, phone
- * state, user requests...).
- * System state changes and user actions are notified to audio policy
- * manager with methods of the audio_policy.
- *
- * - process get_output() queries received when AudioTrack objects are
- * created: Those queries return a handler on an output that has been
- * selected, configured and opened by the audio policy manager and that
- * must be used by the AudioTrack when registering to the AudioFlinger
- * with the createTrack() method.
- * When the AudioTrack object is released, a release_output() query
- * is received and the audio policy manager can decide to close or
- * reconfigure the output depending on other streams using this output and
- * current system state.
- *
- * - similarly process get_input() and release_input() queries received from
- * AudioRecord objects and configure audio inputs.
- * - process volume control requests: the stream volume is converted from
- * an index value (received from UI) to a float value applicable to each
- * output as a function of platform specific settings and current output
- * route (destination device). It also make sure that streams are not
- * muted if not allowed (e.g. camera shutter sound in some countries).
- */
-
-/* XXX: this should be defined OUTSIDE of frameworks/base */
-struct effect_descriptor_s;
-
-struct audio_policy {
- /*
- * configuration functions
- */
-
- /* indicate a change in device connection status */
- int (*set_device_connection_state)(struct audio_policy* pol, audio_devices_t device,
- audio_policy_dev_state_t state, const char* device_address);
-
- /* retrieve a device connection status */
- audio_policy_dev_state_t (*get_device_connection_state)(const struct audio_policy* pol,
- audio_devices_t device,
- const char* device_address);
-
- /* indicate a change in phone state. Valid phones states are defined
- * by audio_mode_t */
- void (*set_phone_state)(struct audio_policy* pol, audio_mode_t state);
-
- /* deprecated, never called (was "indicate a change in ringer mode") */
- void (*set_ringer_mode)(struct audio_policy* pol, uint32_t mode, uint32_t mask);
-
- /* force using a specific device category for the specified usage */
- void (*set_force_use)(struct audio_policy* pol, audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config);
-
- /* retrieve current device category forced for a given usage */
- audio_policy_forced_cfg_t (*get_force_use)(const struct audio_policy* pol,
- audio_policy_force_use_t usage);
-
- /* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
- * can still be muted. */
- void (*set_can_mute_enforced_audible)(struct audio_policy* pol, bool can_mute);
-
- /* check proper initialization */
- int (*init_check)(const struct audio_policy* pol);
-
- /*
- * Audio routing query functions
- */
-
- /* request an output appropriate for playback of the supplied stream type and
- * parameters */
- audio_io_handle_t (*get_output)(struct audio_policy* pol, audio_stream_type_t stream,
- uint32_t samplingRate, audio_format_t format,
- audio_channel_mask_t channelMask, audio_output_flags_t flags,
- const audio_offload_info_t* offloadInfo);
-
- /* indicates to the audio policy manager that the output starts being used
- * by corresponding stream. */
- int (*start_output)(struct audio_policy* pol, audio_io_handle_t output,
- audio_stream_type_t stream, audio_session_t session);
-
- /* indicates to the audio policy manager that the output stops being used
- * by corresponding stream. */
- int (*stop_output)(struct audio_policy* pol, audio_io_handle_t output,
- audio_stream_type_t stream, audio_session_t session);
-
- /* releases the output. */
- void (*release_output)(struct audio_policy* pol, audio_io_handle_t output);
-
- /* request an input appropriate for record from the supplied device with
- * supplied parameters. */
- audio_io_handle_t (*get_input)(struct audio_policy* pol, audio_source_t inputSource,
- uint32_t samplingRate, audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_in_acoustics_t acoustics);
-
- /* indicates to the audio policy manager that the input starts being used */
- int (*start_input)(struct audio_policy* pol, audio_io_handle_t input);
-
- /* indicates to the audio policy manager that the input stops being used. */
- int (*stop_input)(struct audio_policy* pol, audio_io_handle_t input);
-
- /* releases the input. */
- void (*release_input)(struct audio_policy* pol, audio_io_handle_t input);
-
- /*
- * volume control functions
- */
-
- /* initialises stream volume conversion parameters by specifying volume
- * index range. The index range for each stream is defined by AudioService. */
- void (*init_stream_volume)(struct audio_policy* pol, audio_stream_type_t stream, int index_min,
- int index_max);
-
- /* sets the new stream volume at a level corresponding to the supplied
- * index. The index is within the range specified by init_stream_volume() */
- int (*set_stream_volume_index)(struct audio_policy* pol, audio_stream_type_t stream, int index);
-
- /* retrieve current volume index for the specified stream */
- int (*get_stream_volume_index)(const struct audio_policy* pol, audio_stream_type_t stream,
- int* index);
-
- /* sets the new stream volume at a level corresponding to the supplied
- * index for the specified device.
- * The index is within the range specified by init_stream_volume() */
- int (*set_stream_volume_index_for_device)(struct audio_policy* pol, audio_stream_type_t stream,
- int index, audio_devices_t device);
-
- /* retrieve current volume index for the specified stream for the specified device */
- int (*get_stream_volume_index_for_device)(const struct audio_policy* pol,
- audio_stream_type_t stream, int* index,
- audio_devices_t device);
-
- /* return the strategy corresponding to a given stream type */
- uint32_t (*get_strategy_for_stream)(const struct audio_policy* pol, audio_stream_type_t stream);
-
- /* return the enabled output devices for the given stream type */
- audio_devices_t (*get_devices_for_stream)(const struct audio_policy* pol,
- audio_stream_type_t stream);
-
- /* Audio effect management */
- audio_io_handle_t (*get_output_for_effect)(struct audio_policy* pol,
- const struct effect_descriptor_s* desc);
-
- int (*register_effect)(struct audio_policy* pol, const struct effect_descriptor_s* desc,
- audio_io_handle_t output, uint32_t strategy, audio_session_t session,
- int id);
-
- int (*unregister_effect)(struct audio_policy* pol, int id);
-
- int (*set_effect_enabled)(struct audio_policy* pol, int id, bool enabled);
-
- bool (*is_stream_active)(const struct audio_policy* pol, audio_stream_type_t stream,
- uint32_t in_past_ms);
-
- bool (*is_stream_active_remotely)(const struct audio_policy* pol, audio_stream_type_t stream,
- uint32_t in_past_ms);
-
- bool (*is_source_active)(const struct audio_policy* pol, audio_source_t source);
-
- /* dump state */
- int (*dump)(const struct audio_policy* pol, int fd);
-
- /* check if offload is possible for given sample rate, bitrate, duration, ... */
- bool (*is_offload_supported)(const struct audio_policy* pol, const audio_offload_info_t* info);
-};
-
-struct audio_policy_service_ops {
- /*
- * Audio output Control functions
- */
-
- /* Opens an audio output with the requested parameters.
- *
- * The parameter values can indicate to use the default values in case the
- * audio policy manager has no specific requirements for the output being
- * opened.
- *
- * When the function returns, the parameter values reflect the actual
- * values used by the audio hardware output stream.
- *
- * The audio policy manager can check if the proposed parameters are
- * suitable or not and act accordingly.
- */
- audio_io_handle_t (*open_output)(void* service, audio_devices_t* pDevices,
- uint32_t* pSamplingRate, audio_format_t* pFormat,
- audio_channel_mask_t* pChannelMask, uint32_t* pLatencyMs,
- audio_output_flags_t flags);
-
- /* creates a special output that is duplicated to the two outputs passed as
- * arguments. The duplication is performed by
- * a special mixer thread in the AudioFlinger.
- */
- audio_io_handle_t (*open_duplicate_output)(void* service, audio_io_handle_t output1,
- audio_io_handle_t output2);
-
- /* closes the output stream */
- int (*close_output)(void* service, audio_io_handle_t output);
-
- /* suspends the output.
- *
- * When an output is suspended, the corresponding audio hardware output
- * stream is placed in standby and the AudioTracks attached to the mixer
- * thread are still processed but the output mix is discarded.
- */
- int (*suspend_output)(void* service, audio_io_handle_t output);
-
- /* restores a suspended output. */
- int (*restore_output)(void* service, audio_io_handle_t output);
-
- /* */
- /* Audio input Control functions */
- /* */
-
- /* opens an audio input
- * deprecated - new implementations should use open_input_on_module,
- * and the acoustics parameter is ignored
- */
- audio_io_handle_t (*open_input)(void* service, audio_devices_t* pDevices,
- uint32_t* pSamplingRate, audio_format_t* pFormat,
- audio_channel_mask_t* pChannelMask,
- audio_in_acoustics_t acoustics);
-
- /* closes an audio input */
- int (*close_input)(void* service, audio_io_handle_t input);
-
- /* */
- /* misc control functions */
- /* */
-
- /* set a stream volume for a particular output.
- *
- * For the same user setting, a given stream type can have different
- * volumes for each output (destination device) it is attached to.
- */
- int (*set_stream_volume)(void* service, audio_stream_type_t stream, float volume,
- audio_io_handle_t output, int delay_ms);
-
- /* invalidate a stream type, causing a reroute to an unspecified new output */
- int (*invalidate_stream)(void* service, audio_stream_type_t stream);
-
- /* function enabling to send proprietary informations directly from audio
- * policy manager to audio hardware interface. */
- void (*set_parameters)(void* service, audio_io_handle_t io_handle, const char* kv_pairs,
- int delay_ms);
-
- /* function enabling to receive proprietary informations directly from
- * audio hardware interface to audio policy manager.
- *
- * Returns a pointer to a heap allocated string. The caller is responsible
- * for freeing the memory for it using free().
- */
-
- char* (*get_parameters)(void* service, audio_io_handle_t io_handle, const char* keys);
-
- /* request the playback of a tone on the specified stream.
- * used for instance to replace notification sounds when playing over a
- * telephony device during a phone call.
- */
- int (*start_tone)(void* service, audio_policy_tone_t tone, audio_stream_type_t stream);
-
- int (*stop_tone)(void* service);
-
- /* set down link audio volume. */
- int (*set_voice_volume)(void* service, float volume, int delay_ms);
-
- /* move effect to the specified output */
- int (*move_effects)(void* service, audio_session_t session, audio_io_handle_t src_output,
- audio_io_handle_t dst_output);
-
- /* loads an audio hw module.
- *
- * The module name passed is the base name of the HW module library, e.g "primary" or "a2dp".
- * The function returns a handle on the module that will be used to specify a particular
- * module when calling open_output_on_module() or open_input_on_module()
- */
- audio_module_handle_t (*load_hw_module)(void* service, const char* name);
-
- /* Opens an audio output on a particular HW module.
- *
- * Same as open_output() but specifying a specific HW module on which the output must be opened.
- */
- audio_io_handle_t (*open_output_on_module)(void* service, audio_module_handle_t module,
- audio_devices_t* pDevices, uint32_t* pSamplingRate,
- audio_format_t* pFormat,
- audio_channel_mask_t* pChannelMask,
- uint32_t* pLatencyMs, audio_output_flags_t flags,
- const audio_offload_info_t* offloadInfo);
-
- /* Opens an audio input on a particular HW module.
- *
- * Same as open_input() but specifying a specific HW module on which the input must be opened.
- * Also removed deprecated acoustics parameter
- */
- audio_io_handle_t (*open_input_on_module)(void* service, audio_module_handle_t module,
- audio_devices_t* pDevices, uint32_t* pSamplingRate,
- audio_format_t* pFormat,
- audio_channel_mask_t* pChannelMask);
-};
-
-/**********************************************************************/
-
-/**
- * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
- * and the fields of this data structure must begin with hw_module_t
- * followed by module specific information.
- */
-typedef struct audio_policy_module { struct hw_module_t common; } audio_policy_module_t;
-
-struct audio_policy_device {
- /**
- * Common methods of the audio policy device. This *must* be the first member of
- * audio_policy_device as users of this structure will cast a hw_device_t to
- * audio_policy_device pointer in contexts where it's known the hw_device_t references an
- * audio_policy_device.
- */
- struct hw_device_t common;
-
- int (*create_audio_policy)(const struct audio_policy_device* device,
- struct audio_policy_service_ops* aps_ops, void* service,
- struct audio_policy** ap);
-
- int (*destroy_audio_policy)(const struct audio_policy_device* device, struct audio_policy* ap);
-};
-
-/** convenience API for opening and closing a supported device */
-
-static inline int audio_policy_dev_open(const hw_module_t* module,
- struct audio_policy_device** device) {
- return module->methods->open(module, AUDIO_POLICY_INTERFACE, (hw_device_t**)device);
-}
-
-static inline int audio_policy_dev_close(struct audio_policy_device* device) {
- return device->common.close(&device->common);
-}
-
-__END_DECLS
-
-#endif // ANDROID_AUDIO_POLICY_INTERFACE_H
diff --git a/audio/common/all-versions/legacy/include/system/audio.h b/audio/common/all-versions/legacy/include/system/audio.h
deleted file mode 100644
index 7afa6c4..0000000
--- a/audio/common/all-versions/legacy/include/system/audio.h
+++ /dev/null
@@ -1,1038 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-#ifndef ANDROID_AUDIO_CORE_H
-#define ANDROID_AUDIO_CORE_H
-
-#include <stdbool.h>
-#include <stdint.h>
-#include <stdio.h>
-#include <sys/cdefs.h>
-#include <sys/types.h>
-
-#include <cutils/bitops.h>
-
-#include "system/audio-base.h"
-
-__BEGIN_DECLS
-
-/* The enums were moved here mostly from
- * frameworks/base/include/media/AudioSystem.h
- */
-
-/* represents an invalid uid for tracks; the calling or client uid is often substituted. */
-#define AUDIO_UID_INVALID ((uid_t)-1)
-
-/* device address used to refer to the standard remote submix */
-#define AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS "0"
-
-/* AudioFlinger and AudioPolicy services use I/O handles to identify audio sources and sinks */
-typedef int audio_io_handle_t;
-
-/* Do not change these values without updating their counterparts
- * in frameworks/base/media/java/android/media/AudioAttributes.java
- */
-typedef enum {
- AUDIO_CONTENT_TYPE_UNKNOWN = 0,
- AUDIO_CONTENT_TYPE_SPEECH = 1,
- AUDIO_CONTENT_TYPE_MUSIC = 2,
- AUDIO_CONTENT_TYPE_MOVIE = 3,
- AUDIO_CONTENT_TYPE_SONIFICATION = 4,
-
- AUDIO_CONTENT_TYPE_CNT,
- AUDIO_CONTENT_TYPE_MAX = AUDIO_CONTENT_TYPE_CNT - 1,
-} audio_content_type_t;
-
-typedef uint32_t audio_flags_mask_t;
-
-/* Do not change these values without updating their counterparts
- * in frameworks/base/media/java/android/media/AudioAttributes.java
- */
-enum {
- AUDIO_FLAG_NONE = 0x0,
- AUDIO_FLAG_AUDIBILITY_ENFORCED = 0x1,
- AUDIO_FLAG_SECURE = 0x2,
- AUDIO_FLAG_SCO = 0x4,
- AUDIO_FLAG_BEACON = 0x8,
- AUDIO_FLAG_HW_AV_SYNC = 0x10,
- AUDIO_FLAG_HW_HOTWORD = 0x20,
- AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY = 0x40,
- AUDIO_FLAG_BYPASS_MUTE = 0x80,
- AUDIO_FLAG_LOW_LATENCY = 0x100,
- AUDIO_FLAG_DEEP_BUFFER = 0x200,
-};
-
-/* Audio attributes */
-#define AUDIO_ATTRIBUTES_TAGS_MAX_SIZE 256
-typedef struct {
- audio_content_type_t content_type;
- audio_usage_t usage;
- audio_source_t source;
- audio_flags_mask_t flags;
- char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */
-} __attribute__((packed)) audio_attributes_t; // sent through Binder;
-
-/* a unique ID allocated by AudioFlinger for use as an audio_io_handle_t, audio_session_t,
- * effect ID (int), audio_module_handle_t, and audio_patch_handle_t.
- * Audio port IDs (audio_port_handle_t) are allocated by AudioPolicy
- * in a different namespace than AudioFlinger unique IDs.
- */
-typedef int audio_unique_id_t;
-
-/* Possible uses for an audio_unique_id_t */
-typedef enum {
- AUDIO_UNIQUE_ID_USE_UNSPECIFIED = 0,
- AUDIO_UNIQUE_ID_USE_SESSION = 1, // for allocated sessions, not special AUDIO_SESSION_*
- AUDIO_UNIQUE_ID_USE_MODULE = 2,
- AUDIO_UNIQUE_ID_USE_EFFECT = 3,
- AUDIO_UNIQUE_ID_USE_PATCH = 4,
- AUDIO_UNIQUE_ID_USE_OUTPUT = 5,
- AUDIO_UNIQUE_ID_USE_INPUT = 6,
- AUDIO_UNIQUE_ID_USE_PLAYER = 7,
- AUDIO_UNIQUE_ID_USE_MAX = 8, // must be a power-of-two
- AUDIO_UNIQUE_ID_USE_MASK = AUDIO_UNIQUE_ID_USE_MAX - 1
-} audio_unique_id_use_t;
-
-/* Return the use of an audio_unique_id_t */
-static inline audio_unique_id_use_t audio_unique_id_get_use(audio_unique_id_t id) {
- return (audio_unique_id_use_t)(id & AUDIO_UNIQUE_ID_USE_MASK);
-}
-
-/* Reserved audio_unique_id_t values. FIXME: not a complete list. */
-#define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE
-
-/* A channel mask per se only defines the presence or absence of a channel, not the order.
- * But see AUDIO_INTERLEAVE_* below for the platform convention of order.
- *
- * audio_channel_mask_t is an opaque type and its internal layout should not
- * be assumed as it may change in the future.
- * Instead, always use the functions declared in this header to examine.
- *
- * These are the current representations:
- *
- * AUDIO_CHANNEL_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 AUDIO_CHANNEL_OUT_* or AUDIO_CHANNEL_IN_* apply to the bits portion.
- * It is not permitted for no bits to be set.
- *
- * AUDIO_CHANNEL_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.
- */
-typedef uint32_t audio_channel_mask_t;
-
-/* log(2) of maximum number of representations, not part of public API */
-#define AUDIO_CHANNEL_REPRESENTATION_LOG2 2
-
-/* The return value is undefined if the channel mask is invalid. */
-static inline uint32_t audio_channel_mask_get_bits(audio_channel_mask_t channel) {
- return channel & ((1 << AUDIO_CHANNEL_COUNT_MAX) - 1);
-}
-
-typedef uint32_t audio_channel_representation_t;
-
-/* The return value is undefined if the channel mask is invalid. */
-static inline audio_channel_representation_t audio_channel_mask_get_representation(
- audio_channel_mask_t channel) {
- // The right shift should be sufficient, but also "and" for safety in case mask is not 32 bits
- return (audio_channel_representation_t)((channel >> AUDIO_CHANNEL_COUNT_MAX) &
- ((1 << AUDIO_CHANNEL_REPRESENTATION_LOG2) - 1));
-}
-
-/* Returns true if the channel mask is valid,
- * or returns false for AUDIO_CHANNEL_NONE, AUDIO_CHANNEL_INVALID, and other invalid values.
- * This function is unable to determine whether a channel mask for position assignment
- * is invalid because an output mask has an invalid output bit set,
- * or because an input mask has an invalid input bit set.
- * All other APIs that take a channel mask assume that it is valid.
- */
-static inline bool audio_channel_mask_is_valid(audio_channel_mask_t channel) {
- uint32_t bits = audio_channel_mask_get_bits(channel);
- audio_channel_representation_t representation = audio_channel_mask_get_representation(channel);
- switch (representation) {
- case AUDIO_CHANNEL_REPRESENTATION_POSITION:
- case AUDIO_CHANNEL_REPRESENTATION_INDEX:
- break;
- default:
- bits = 0;
- break;
- }
- return bits != 0;
-}
-
-/* Not part of public API */
-static inline audio_channel_mask_t audio_channel_mask_from_representation_and_bits(
- audio_channel_representation_t representation, uint32_t bits) {
- return (audio_channel_mask_t)((representation << AUDIO_CHANNEL_COUNT_MAX) | bits);
-}
-
-/* This enum is deprecated */
-typedef enum {
- AUDIO_IN_ACOUSTICS_NONE = 0,
- AUDIO_IN_ACOUSTICS_AGC_ENABLE = 0x0001,
- AUDIO_IN_ACOUSTICS_AGC_DISABLE = 0,
- AUDIO_IN_ACOUSTICS_NS_ENABLE = 0x0002,
- AUDIO_IN_ACOUSTICS_NS_DISABLE = 0,
- AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE = 0x0004,
- AUDIO_IN_ACOUSTICS_TX_DISABLE = 0,
-} audio_in_acoustics_t;
-
-typedef uint32_t audio_devices_t;
-/**
- * Stub audio output device. Used in policy configuration file on platforms without audio outputs.
- * This alias value to AUDIO_DEVICE_OUT_DEFAULT is only used in the audio policy context.
- */
-#define AUDIO_DEVICE_OUT_STUB AUDIO_DEVICE_OUT_DEFAULT
-/**
- * Stub audio input device. Used in policy configuration file on platforms without audio inputs.
- * This alias value to AUDIO_DEVICE_IN_DEFAULT is only used in the audio policy context.
- */
-#define AUDIO_DEVICE_IN_STUB AUDIO_DEVICE_IN_DEFAULT
-
-/* Additional information about compressed streams offloaded to
- * hardware playback
- * The version and size fields must be initialized by the caller by using
- * one of the constants defined here.
- * Must be aligned to transmit as raw memory through Binder.
- */
-typedef struct {
- uint16_t version; // version of the info structure
- uint16_t size; // total size of the structure including version and size
- uint32_t sample_rate; // sample rate in Hz
- audio_channel_mask_t channel_mask; // channel mask
- audio_format_t format; // audio format
- audio_stream_type_t stream_type; // stream type
- uint32_t bit_rate; // bit rate in bits per second
- int64_t duration_us; // duration in microseconds, -1 if unknown
- bool has_video; // true if stream is tied to a video stream
- bool is_streaming; // true if streaming, false if local playback
- uint32_t bit_width;
- uint32_t offload_buffer_size; // offload fragment size
- audio_usage_t usage;
-} __attribute__((aligned(8))) audio_offload_info_t;
-
-#define AUDIO_MAKE_OFFLOAD_INFO_VERSION(maj, min) ((((maj)&0xff) << 8) | ((min)&0xff))
-
-#define AUDIO_OFFLOAD_INFO_VERSION_0_1 AUDIO_MAKE_OFFLOAD_INFO_VERSION(0, 1)
-#define AUDIO_OFFLOAD_INFO_VERSION_CURRENT AUDIO_OFFLOAD_INFO_VERSION_0_1
-
-static const audio_offload_info_t AUDIO_INFO_INITIALIZER = {
- /* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT,
- /* .size = */ sizeof(audio_offload_info_t),
- /* .sample_rate = */ 0,
- /* .channel_mask = */ 0,
- /* .format = */ AUDIO_FORMAT_DEFAULT,
- /* .stream_type = */ AUDIO_STREAM_VOICE_CALL,
- /* .bit_rate = */ 0,
- /* .duration_us = */ 0,
- /* .has_video = */ false,
- /* .is_streaming = */ false,
- /* .bit_width = */ 16,
- /* .offload_buffer_size = */ 0,
- /* .usage = */ AUDIO_USAGE_UNKNOWN};
-
-/* common audio stream configuration parameters
- * You should memset() the entire structure to zero before use to
- * ensure forward compatibility
- * Must be aligned to transmit as raw memory through Binder.
- */
-struct __attribute__((aligned(8))) audio_config {
- uint32_t sample_rate;
- audio_channel_mask_t channel_mask;
- audio_format_t format;
- audio_offload_info_t offload_info;
- uint32_t frame_count;
-};
-typedef struct audio_config audio_config_t;
-
-static const audio_config_t AUDIO_CONFIG_INITIALIZER = {
- /* .sample_rate = */ 0,
- /* .channel_mask = */ AUDIO_CHANNEL_NONE,
- /* .format = */ AUDIO_FORMAT_DEFAULT,
- /* .offload_info = */
- {/* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT,
- /* .size = */ sizeof(audio_offload_info_t),
- /* .sample_rate = */ 0,
- /* .channel_mask = */ 0,
- /* .format = */ AUDIO_FORMAT_DEFAULT,
- /* .stream_type = */ AUDIO_STREAM_VOICE_CALL,
- /* .bit_rate = */ 0,
- /* .duration_us = */ 0,
- /* .has_video = */ false,
- /* .is_streaming = */ false,
- /* .bit_width = */ 16,
- /* .offload_buffer_size = */ 0,
- /* .usage = */ AUDIO_USAGE_UNKNOWN},
- /* .frame_count = */ 0,
-};
-
-struct audio_config_base {
- uint32_t sample_rate;
- audio_channel_mask_t channel_mask;
- audio_format_t format;
-};
-
-typedef struct audio_config_base audio_config_base_t;
-
-static const audio_config_base_t AUDIO_CONFIG_BASE_INITIALIZER = {
- /* .sample_rate = */ 0,
- /* .channel_mask = */ AUDIO_CHANNEL_NONE,
- /* .format = */ AUDIO_FORMAT_DEFAULT};
-
-/* audio hw module handle functions or structures referencing a module */
-typedef int audio_module_handle_t;
-
-/******************************
- * Volume control
- *****************************/
-
-/** 3 dB headroom are allowed on float samples (3db = 10^(3/20) = 1.412538).
- * See: https://developer.android.com/reference/android/media/AudioTrack.html#write(float[], int,
- * int, int)
- */
-#define FLOAT_NOMINAL_RANGE_HEADROOM 1.412538
-
-/* If the audio hardware supports gain control on some audio paths,
- * the platform can expose them in the audio_policy.conf file. The audio HAL
- * will then implement gain control functions that will use the following data
- * structures. */
-
-typedef uint32_t audio_gain_mode_t;
-
-/* An audio_gain struct is a representation of a gain stage.
- * A gain stage is always attached to an audio port. */
-struct audio_gain {
- audio_gain_mode_t mode; /* e.g. AUDIO_GAIN_MODE_JOINT */
- audio_channel_mask_t channel_mask; /* channels which gain an be controlled.
- N/A if AUDIO_GAIN_MODE_CHANNELS is not supported */
- int min_value; /* minimum gain value in millibels */
- int max_value; /* maximum gain value in millibels */
- int default_value; /* default gain value in millibels */
- unsigned int step_value; /* gain step in millibels */
- unsigned int min_ramp_ms; /* minimum ramp duration in ms */
- unsigned int max_ramp_ms; /* maximum ramp duration in ms */
-};
-
-/* The gain configuration structure is used to get or set the gain values of a
- * given port */
-struct audio_gain_config {
- int index; /* index of the corresponding audio_gain in the
- audio_port gains[] table */
- audio_gain_mode_t mode; /* mode requested for this command */
- audio_channel_mask_t channel_mask; /* channels which gain value follows.
- N/A in joint mode */
-
- // note this "8" is not FCC_8, so it won't need to be changed for > 8 channels
- int values[sizeof(audio_channel_mask_t) * 8]; /* 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) */
- unsigned int ramp_duration_ms; /* 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. */
-
-/* Each port has a unique ID or handle allocated by policy manager */
-typedef int audio_port_handle_t;
-
-/* the maximum length for the human-readable device name */
-#define AUDIO_PORT_MAX_NAME_LEN 128
-
-/* maximum audio device address length */
-#define AUDIO_DEVICE_MAX_ADDRESS_LEN 32
-
-/* extension for audio port configuration structure when the audio port is a
- * hardware device */
-struct audio_port_config_device_ext {
- audio_module_handle_t hw_module; /* module the device is attached to */
- audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
- char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; /* device address. "" if N/A */
-};
-
-/* extension for audio port configuration structure when the audio port is a
- * sub mix */
-struct audio_port_config_mix_ext {
- audio_module_handle_t hw_module; /* module the stream is attached to */
- audio_io_handle_t handle; /* I/O handle of the input/output stream */
- union {
- // TODO: change use case for output streams: use strategy and mixer attributes
- audio_stream_type_t stream;
- audio_source_t source;
- } usecase;
-};
-
-/* extension for audio port configuration structure when the audio port is an
- * audio session */
-struct audio_port_config_session_ext {
- audio_session_t session; /* audio session */
-};
-
-/* audio port configuration structure used to specify a particular configuration of
- * an audio port */
-struct audio_port_config {
- audio_port_handle_t id; /* port unique ID */
- audio_port_role_t role; /* sink or source */
- audio_port_type_t type; /* device, mix ... */
- unsigned int config_mask; /* e.g AUDIO_PORT_CONFIG_ALL */
- unsigned int sample_rate; /* sampling rate in Hz */
- audio_channel_mask_t channel_mask; /* channel mask if applicable */
- audio_format_t format; /* format if applicable */
- struct audio_gain_config gain; /* gain to apply if applicable */
- union {
- struct audio_port_config_device_ext device; /* device specific info */
- struct audio_port_config_mix_ext mix; /* mix specific info */
- struct audio_port_config_session_ext session; /* session specific info */
- } ext;
-};
-
-/* max number of sampling rates in audio port */
-#define AUDIO_PORT_MAX_SAMPLING_RATES 32
-/* max number of channel masks in audio port */
-#define AUDIO_PORT_MAX_CHANNEL_MASKS 32
-/* max number of audio formats in audio port */
-#define AUDIO_PORT_MAX_FORMATS 32
-/* max number of gain controls in audio port */
-#define AUDIO_PORT_MAX_GAINS 16
-
-/* extension for audio port structure when the audio port is a hardware device */
-struct audio_port_device_ext {
- audio_module_handle_t hw_module; /* module the device is attached to */
- audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
- char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
-};
-
-/* extension for audio port structure when the audio port is a sub mix */
-struct audio_port_mix_ext {
- audio_module_handle_t hw_module; /* module the stream is attached to */
- audio_io_handle_t handle; /* I/O handle of the input.output stream */
- audio_mix_latency_class_t latency_class; /* latency class */
- // other attributes: routing strategies
-};
-
-/* extension for audio port structure when the audio port is an audio session */
-struct audio_port_session_ext {
- audio_session_t session; /* audio session */
-};
-
-struct audio_port {
- audio_port_handle_t id; /* port unique ID */
- audio_port_role_t role; /* sink or source */
- audio_port_type_t type; /* device, mix ... */
- char name[AUDIO_PORT_MAX_NAME_LEN];
- unsigned int num_sample_rates; /* number of sampling rates in following array */
- unsigned int sample_rates[AUDIO_PORT_MAX_SAMPLING_RATES];
- unsigned int num_channel_masks; /* number of channel masks in following array */
- audio_channel_mask_t channel_masks[AUDIO_PORT_MAX_CHANNEL_MASKS];
- unsigned int num_formats; /* number of formats in following array */
- audio_format_t formats[AUDIO_PORT_MAX_FORMATS];
- unsigned int num_gains; /* number of gains in following array */
- struct audio_gain gains[AUDIO_PORT_MAX_GAINS];
- struct audio_port_config active_config; /* current audio port configuration */
- union {
- struct audio_port_device_ext device;
- struct audio_port_mix_ext mix;
- struct audio_port_session_ext session;
- } ext;
-};
-
-/* An audio patch represents a connection between one or more source ports and
- * one or more sink ports. Patches are connected and disconnected by audio policy manager or by
- * applications via framework APIs.
- * 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 int audio_patch_handle_t;
-
-#define AUDIO_PATCH_PORTS_MAX 16
-
-struct audio_patch {
- audio_patch_handle_t id; /* patch unique ID */
- unsigned int num_sources; /* number of sources in following array */
- struct audio_port_config sources[AUDIO_PATCH_PORTS_MAX];
- unsigned int num_sinks; /* number of sinks in following array */
- struct audio_port_config sinks[AUDIO_PATCH_PORTS_MAX];
-};
-
-/* a HW synchronization source returned by the audio HAL */
-typedef uint32_t audio_hw_sync_t;
-
-/* an invalid HW synchronization source indicating an error */
-#define AUDIO_HW_SYNC_INVALID 0
-
-/**
- * Mmap buffer descriptor returned by audio_stream->create_mmap_buffer().
- * note\ Used by streams opened in mmap mode.
- */
-struct audio_mmap_buffer_info {
- void* shared_memory_address; /**< base address of mmap memory buffer.
- For use by local process only */
- int32_t shared_memory_fd; /**< FD for mmap memory buffer */
- int32_t buffer_size_frames; /**< total buffer size in frames */
- int32_t burst_size_frames; /**< transfer size granularity in frames */
-};
-
-/**
- * Mmap buffer read/write position returned by audio_stream->get_mmap_position().
- * note\ Used by streams opened in mmap mode.
- */
-struct audio_mmap_position {
- int64_t time_nanoseconds; /**< timestamp in ns, CLOCK_MONOTONIC */
- int32_t position_frames; /**< increasing 32 bit frame count reset when stream->stop()
- is called */
-};
-
-static inline bool audio_is_output_device(audio_devices_t device) {
- if (((device & AUDIO_DEVICE_BIT_IN) == 0) && (popcount(device) == 1) &&
- ((device & ~AUDIO_DEVICE_OUT_ALL) == 0))
- return true;
- else
- return false;
-}
-
-static inline bool audio_is_input_device(audio_devices_t device) {
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_ALL) == 0)) return true;
- }
- return false;
-}
-
-static inline bool audio_is_output_devices(audio_devices_t device) {
- return (device & AUDIO_DEVICE_BIT_IN) == 0;
-}
-
-static inline bool audio_is_a2dp_in_device(audio_devices_t device) {
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if ((popcount(device) == 1) && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP)) return true;
- }
- return false;
-}
-
-static inline bool audio_is_a2dp_out_device(audio_devices_t device) {
- if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_A2DP))
- return true;
- else
- return false;
-}
-
-// Deprecated - use audio_is_a2dp_out_device() instead
-static inline bool audio_is_a2dp_device(audio_devices_t device) {
- return audio_is_a2dp_out_device(device);
-}
-
-static inline bool audio_is_bluetooth_sco_device(audio_devices_t device) {
- if ((device & AUDIO_DEVICE_BIT_IN) == 0) {
- if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL_SCO) == 0)) return true;
- } else {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) == 0))
- return true;
- }
-
- return false;
-}
-
-static inline bool audio_is_usb_out_device(audio_devices_t device) {
- return ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_USB));
-}
-
-static inline bool audio_is_usb_in_device(audio_devices_t device) {
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if (popcount(device) == 1 && (device & AUDIO_DEVICE_IN_ALL_USB) != 0) return true;
- }
- return false;
-}
-
-/* OBSOLETE - use audio_is_usb_out_device() instead. */
-static inline bool audio_is_usb_device(audio_devices_t device) {
- return audio_is_usb_out_device(device);
-}
-
-static inline bool audio_is_remote_submix_device(audio_devices_t device) {
- if ((audio_is_output_devices(device) &&
- (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX) == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) ||
- (!audio_is_output_devices(device) &&
- (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) == AUDIO_DEVICE_IN_REMOTE_SUBMIX))
- return true;
- else
- return false;
-}
-
-/* Returns true if:
- * representation is valid, and
- * there is at least one channel bit set which _could_ correspond to an input channel, and
- * there are no channel bits set which could _not_ correspond to an input channel.
- * Otherwise returns false.
- */
-static inline bool audio_is_input_channel(audio_channel_mask_t channel) {
- uint32_t bits = audio_channel_mask_get_bits(channel);
- switch (audio_channel_mask_get_representation(channel)) {
- case AUDIO_CHANNEL_REPRESENTATION_POSITION:
- if (bits & ~AUDIO_CHANNEL_IN_ALL) {
- bits = 0;
- }
- // fall through
- case AUDIO_CHANNEL_REPRESENTATION_INDEX:
- return bits != 0;
- default:
- return false;
- }
-}
-
-/* Returns true if:
- * representation is valid, and
- * there is at least one channel bit set which _could_ correspond to an output channel, and
- * there are no channel bits set which could _not_ correspond to an output channel.
- * Otherwise returns false.
- */
-static inline bool audio_is_output_channel(audio_channel_mask_t channel) {
- uint32_t bits = audio_channel_mask_get_bits(channel);
- switch (audio_channel_mask_get_representation(channel)) {
- case AUDIO_CHANNEL_REPRESENTATION_POSITION:
- if (bits & ~AUDIO_CHANNEL_OUT_ALL) {
- bits = 0;
- }
- // fall through
- case AUDIO_CHANNEL_REPRESENTATION_INDEX:
- return bits != 0;
- default:
- return false;
- }
-}
-
-/* Returns the number of channels from an input channel mask,
- * used in the context of audio input or recording.
- * If a channel bit is set which could _not_ correspond to an input channel,
- * it is excluded from the count.
- * Returns zero if the representation is invalid.
- */
-static inline uint32_t audio_channel_count_from_in_mask(audio_channel_mask_t channel) {
- uint32_t bits = audio_channel_mask_get_bits(channel);
- switch (audio_channel_mask_get_representation(channel)) {
- case AUDIO_CHANNEL_REPRESENTATION_POSITION:
- // TODO: We can now merge with from_out_mask and remove anding
- bits &= AUDIO_CHANNEL_IN_ALL;
- // fall through
- case AUDIO_CHANNEL_REPRESENTATION_INDEX:
- return popcount(bits);
- default:
- return 0;
- }
-}
-
-/* Returns the number of channels from an output channel mask,
- * used in the context of audio output or playback.
- * If a channel bit is set which could _not_ correspond to an output channel,
- * it is excluded from the count.
- * Returns zero if the representation is invalid.
- */
-static inline uint32_t audio_channel_count_from_out_mask(audio_channel_mask_t channel) {
- uint32_t bits = audio_channel_mask_get_bits(channel);
- switch (audio_channel_mask_get_representation(channel)) {
- case AUDIO_CHANNEL_REPRESENTATION_POSITION:
- // TODO: We can now merge with from_in_mask and remove anding
- bits &= AUDIO_CHANNEL_OUT_ALL;
- // fall through
- case AUDIO_CHANNEL_REPRESENTATION_INDEX:
- return popcount(bits);
- default:
- return 0;
- }
-}
-
-/* Derive a channel mask for index assignment from a channel count.
- * Returns the matching channel mask,
- * or AUDIO_CHANNEL_NONE if the channel count is zero,
- * or AUDIO_CHANNEL_INVALID if the channel count exceeds AUDIO_CHANNEL_COUNT_MAX.
- */
-static inline audio_channel_mask_t audio_channel_mask_for_index_assignment_from_count(
- uint32_t channel_count) {
- if (channel_count == 0) {
- return AUDIO_CHANNEL_NONE;
- }
- if (channel_count > AUDIO_CHANNEL_COUNT_MAX) {
- return AUDIO_CHANNEL_INVALID;
- }
- uint32_t bits = (1 << channel_count) - 1;
- return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_INDEX,
- bits);
-}
-
-/* Derive an output channel mask for position assignment from a channel count.
- * This is to be used when the content channel mask is unknown. The 1, 2, 4, 5, 6, 7 and 8 channel
- * cases are mapped to the standard game/home-theater layouts, but note that 4 is mapped to quad,
- * and not stereo + FC + mono surround. A channel count of 3 is arbitrarily mapped to stereo + FC
- * for continuity with stereo.
- * Returns the matching channel mask,
- * or AUDIO_CHANNEL_NONE if the channel count is zero,
- * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
- * configurations for which a default output channel mask is defined.
- */
-static inline audio_channel_mask_t audio_channel_out_mask_from_count(uint32_t channel_count) {
- uint32_t bits;
- switch (channel_count) {
- case 0:
- return AUDIO_CHANNEL_NONE;
- case 1:
- bits = AUDIO_CHANNEL_OUT_MONO;
- break;
- case 2:
- bits = AUDIO_CHANNEL_OUT_STEREO;
- break;
- case 3:
- bits = AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_FRONT_CENTER;
- break;
- case 4: // 4.0
- bits = AUDIO_CHANNEL_OUT_QUAD;
- break;
- case 5: // 5.0
- bits = AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER;
- break;
- case 6: // 5.1
- bits = AUDIO_CHANNEL_OUT_5POINT1;
- break;
- case 7: // 6.1
- bits = AUDIO_CHANNEL_OUT_5POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER;
- break;
- case 8:
- bits = AUDIO_CHANNEL_OUT_7POINT1;
- break;
- // FIXME FCC_8
- default:
- return AUDIO_CHANNEL_INVALID;
- }
- return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_POSITION,
- bits);
-}
-
-/* Derive a default input channel mask from a channel count.
- * Assumes a position mask for mono and stereo, or an index mask for channel counts > 2.
- * Returns the matching channel mask,
- * or AUDIO_CHANNEL_NONE if the channel count is zero,
- * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
- * configurations for which a default input channel mask is defined.
- */
-static inline audio_channel_mask_t audio_channel_in_mask_from_count(uint32_t channel_count) {
- uint32_t bits;
- switch (channel_count) {
- case 0:
- return AUDIO_CHANNEL_NONE;
- case 1:
- bits = AUDIO_CHANNEL_IN_MONO;
- break;
- case 2:
- bits = AUDIO_CHANNEL_IN_STEREO;
- break;
- case 3:
- case 4:
- case 5:
- case 6:
- case 7:
- case 8:
- // FIXME FCC_8
- return audio_channel_mask_for_index_assignment_from_count(channel_count);
- default:
- return AUDIO_CHANNEL_INVALID;
- }
- return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_POSITION,
- bits);
-}
-
-static inline bool audio_is_valid_format(audio_format_t format) {
- switch (format & AUDIO_FORMAT_MAIN_MASK) {
- case AUDIO_FORMAT_PCM:
- switch (format) {
- case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_8_BIT:
- case AUDIO_FORMAT_PCM_32_BIT:
- case AUDIO_FORMAT_PCM_8_24_BIT:
- case AUDIO_FORMAT_PCM_FLOAT:
- case AUDIO_FORMAT_PCM_24_BIT_PACKED:
- return true;
- default:
- return false;
- }
- /* not reached */
- case AUDIO_FORMAT_MP3:
- case AUDIO_FORMAT_AMR_NB:
- case AUDIO_FORMAT_AMR_WB:
- case AUDIO_FORMAT_AAC:
- case AUDIO_FORMAT_AAC_ADTS:
- case AUDIO_FORMAT_HE_AAC_V1:
- case AUDIO_FORMAT_HE_AAC_V2:
- case AUDIO_FORMAT_VORBIS:
- case AUDIO_FORMAT_OPUS:
- case AUDIO_FORMAT_AC3:
- case AUDIO_FORMAT_E_AC3:
- case AUDIO_FORMAT_DTS:
- case AUDIO_FORMAT_DTS_HD:
- case AUDIO_FORMAT_IEC61937:
- case AUDIO_FORMAT_DOLBY_TRUEHD:
- case AUDIO_FORMAT_QCELP:
- case AUDIO_FORMAT_EVRC:
- case AUDIO_FORMAT_EVRCB:
- case AUDIO_FORMAT_EVRCWB:
- case AUDIO_FORMAT_AAC_ADIF:
- case AUDIO_FORMAT_AMR_WB_PLUS:
- case AUDIO_FORMAT_MP2:
- case AUDIO_FORMAT_EVRCNW:
- case AUDIO_FORMAT_FLAC:
- case AUDIO_FORMAT_ALAC:
- case AUDIO_FORMAT_APE:
- case AUDIO_FORMAT_WMA:
- case AUDIO_FORMAT_WMA_PRO:
- case AUDIO_FORMAT_DSD:
- case AUDIO_FORMAT_AC4:
- case AUDIO_FORMAT_LDAC:
- return true;
- default:
- return false;
- }
-}
-
-/**
- * Extract the primary format, eg. PCM, AC3, etc.
- */
-static inline audio_format_t audio_get_main_format(audio_format_t format) {
- return (audio_format_t)(format & AUDIO_FORMAT_MAIN_MASK);
-}
-
-/**
- * Is the data plain PCM samples that can be scaled and mixed?
- */
-static inline bool audio_is_linear_pcm(audio_format_t format) {
- return (audio_get_main_format(format) == AUDIO_FORMAT_PCM);
-}
-
-/**
- * For this format, is the number of PCM audio frames directly proportional
- * to the number of data bytes?
- *
- * In other words, is the format transported as PCM audio samples,
- * but not necessarily scalable or mixable.
- * This returns true for real PCM, but also for AUDIO_FORMAT_IEC61937,
- * which is transported as 16 bit PCM audio, but where the encoded data
- * cannot be mixed or scaled.
- */
-static inline bool audio_has_proportional_frames(audio_format_t format) {
- audio_format_t mainFormat = audio_get_main_format(format);
- return (mainFormat == AUDIO_FORMAT_PCM || mainFormat == AUDIO_FORMAT_IEC61937);
-}
-
-static inline size_t audio_bytes_per_sample(audio_format_t format) {
- size_t size = 0;
-
- switch (format) {
- case AUDIO_FORMAT_PCM_32_BIT:
- case AUDIO_FORMAT_PCM_8_24_BIT:
- size = sizeof(int32_t);
- break;
- case AUDIO_FORMAT_PCM_24_BIT_PACKED:
- size = sizeof(uint8_t) * 3;
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_IEC61937:
- size = sizeof(int16_t);
- break;
- case AUDIO_FORMAT_PCM_8_BIT:
- size = sizeof(uint8_t);
- break;
- case AUDIO_FORMAT_PCM_FLOAT:
- size = sizeof(float);
- break;
- default:
- break;
- }
- return size;
-}
-
-/* converts device address to string sent to audio HAL via set_parameters */
-static inline char* audio_device_address_to_parameter(audio_devices_t device, const char* address) {
- const size_t kSize = AUDIO_DEVICE_MAX_ADDRESS_LEN + sizeof("a2dp_sink_address=");
- char param[kSize];
-
- if (device & AUDIO_DEVICE_OUT_ALL_A2DP)
- snprintf(param, kSize, "%s=%s", "a2dp_sink_address", address);
- else if (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
- snprintf(param, kSize, "%s=%s", "mix", address);
- else
- snprintf(param, kSize, "%s", address);
-
- return strdup(param);
-}
-
-static inline bool audio_device_is_digital(audio_devices_t device) {
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- // input
- return (~AUDIO_DEVICE_BIT_IN & device &
- (AUDIO_DEVICE_IN_ALL_USB | AUDIO_DEVICE_IN_HDMI | AUDIO_DEVICE_IN_SPDIF |
- AUDIO_DEVICE_IN_IP | AUDIO_DEVICE_IN_BUS)) != 0;
- } else {
- // output
- return (device &
- (AUDIO_DEVICE_OUT_ALL_USB | AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_HDMI_ARC |
- AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_IP | AUDIO_DEVICE_OUT_BUS)) != 0;
- }
-}
-
-// Unique effect ID (can be generated from the following site:
-// http://www.itu.int/ITU-T/asn1/uuid.html)
-// This struct is used for effects identification and in soundtrigger.
-typedef struct audio_uuid_s {
- uint32_t timeLow;
- uint16_t timeMid;
- uint16_t timeHiAndVersion;
- uint16_t clockSeq;
- uint8_t node[6];
-} audio_uuid_t;
-
-__END_DECLS
-
-/**
- * List of known audio HAL modules. This is the base name of the audio HAL
- * library composed of the "audio." prefix, one of the base names below and
- * a suffix specific to the device.
- * e.g: audio.primary.goldfish.so or audio.a2dp.default.so
- *
- * The same module names are used in audio policy configuration files.
- */
-
-#define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary"
-#define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp"
-#define AUDIO_HARDWARE_MODULE_ID_USB "usb"
-#define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix"
-#define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload"
-#define AUDIO_HARDWARE_MODULE_ID_STUB "stub"
-
-/**
- * Multi-Stream Decoder (MSD) HAL service name. MSD HAL is used to mix
- * encoded streams together with PCM streams, producing re-encoded
- * streams or PCM streams.
- *
- * The service must register itself using this name, and audioserver
- * tries to instantiate a device factory using this name as well.
- * Note that the HIDL implementation library file name *must* have the
- * suffix "msd" in order to be picked up by HIDL that is:
- *
- * android.hardware.audio@x.x-implmsd.so
- */
-#define AUDIO_HAL_SERVICE_NAME_MSD "msd"
-
-/**
- * Parameter definitions.
- * Note that in the framework code it's recommended to use AudioParameter.h
- * instead of these preprocessor defines, and for sure avoid just copying
- * the constant values.
- */
-
-#define AUDIO_PARAMETER_VALUE_ON "on"
-#define AUDIO_PARAMETER_VALUE_OFF "off"
-
-/**
- * audio device parameters
- */
-
-/* BT SCO Noise Reduction + Echo Cancellation parameters */
-#define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
-
-/* Get a new HW synchronization source identifier.
- * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs
- * or no HW sync is available. */
-#define AUDIO_PARAMETER_HW_AV_SYNC "hw_av_sync"
-
-/* Screen state */
-#define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state"
-
-/**
- * audio stream parameters
- */
-
-#define AUDIO_PARAMETER_STREAM_ROUTING "routing" /* audio_devices_t */
-#define AUDIO_PARAMETER_STREAM_FORMAT "format" /* audio_format_t */
-#define AUDIO_PARAMETER_STREAM_CHANNELS "channels" /* audio_channel_mask_t */
-#define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" /* size_t */
-#define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" /* audio_source_t */
-#define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */
-
-#define AUDIO_PARAMETER_DEVICE_CONNECT "connect" /* audio_devices_t */
-#define AUDIO_PARAMETER_DEVICE_DISCONNECT "disconnect" /* audio_devices_t */
-
-/* Enable mono audio playback if 1, else should be 0. */
-#define AUDIO_PARAMETER_MONO_OUTPUT "mono_output"
-
-/* Set the HW synchronization source for an output stream. */
-#define AUDIO_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync"
-
-/* Query supported formats. The response is a '|' separated list of strings from
- * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
-#define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
-/* Query supported channel masks. The response is a '|' separated list of strings from
- * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
-#define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
-/* Query supported sampling rates. The response is a '|' separated list of integer values e.g:
- * "sup_sampling_rates=44100|48000" */
-#define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
-
-#define AUDIO_PARAMETER_VALUE_LIST_SEPARATOR "|"
-
-/**
- * audio codec parameters
- */
-
-#define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param"
-#define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample"
-#define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate"
-#define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate"
-#define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id"
-#define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align"
-#define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate"
-#define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option"
-#define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels"
-#define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling"
-#define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples"
-#define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples"
-
-// FIXME: a temporary declaration for the incall music flag, will be removed when
-// declared in types.hal for audio HAL V4.0 and auto imported to audio-base.h
-#define AUDIO_OUTPUT_FLAG_INCALL_MUSIC 0x10000
-
-#endif // ANDROID_AUDIO_CORE_H
diff --git a/audio/common/all-versions/legacy/include/system/audio_effect.h b/audio/common/all-versions/legacy/include/system/audio_effect.h
deleted file mode 100644
index f99f604..0000000
--- a/audio/common/all-versions/legacy/include/system/audio_effect.h
+++ /dev/null
@@ -1,528 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_AUDIO_EFFECT_CORE_H
-#define ANDROID_AUDIO_EFFECT_CORE_H
-
-#include "system/audio.h"
-#include "system/audio_effect-base.h"
-
-__BEGIN_DECLS
-
-/////////////////////////////////////////////////
-// Common Definitions
-/////////////////////////////////////////////////
-
-//
-//--- Effect descriptor structure effect_descriptor_t
-//
-
-// This format is used for both "type" and "uuid" fields of the effect descriptor structure.
-// - When used for effect type and the engine is implementing and effect corresponding to a standard
-// OpenSL ES interface, this ID must be the one defined in OpenSLES_IID.h for that interface.
-// - When used as uuid, it should be a unique UUID for this particular implementation.
-typedef audio_uuid_t effect_uuid_t;
-
-// Maximum length of character strings in structures defines by this API.
-#define EFFECT_STRING_LEN_MAX 64
-
-// NULL UUID definition (matches SL_IID_NULL_)
-#define EFFECT_UUID_INITIALIZER \
- { \
- 0xec7178ec, 0xe5e1, 0x4432, 0xa3f4, { 0x46, 0x57, 0xe6, 0x79, 0x52, 0x10 } \
- }
-static const effect_uuid_t EFFECT_UUID_NULL_ = EFFECT_UUID_INITIALIZER;
-static const effect_uuid_t* const EFFECT_UUID_NULL = &EFFECT_UUID_NULL_;
-static const char* const EFFECT_UUID_NULL_STR = "ec7178ec-e5e1-4432-a3f4-4657e6795210";
-
-// The effect descriptor contains necessary information to facilitate the enumeration of the effect
-// engines present in a library.
-typedef struct effect_descriptor_s {
- effect_uuid_t type; // UUID of to the OpenSL ES interface implemented by this effect
- effect_uuid_t uuid; // UUID for this particular implementation
- uint32_t apiVersion; // Version of the effect control API implemented
- uint32_t flags; // effect engine capabilities/requirements flags (see below)
- uint16_t cpuLoad; // CPU load indication (see below)
- uint16_t memoryUsage; // Data Memory usage (see below)
- char name[EFFECT_STRING_LEN_MAX]; // human readable effect name
- char implementor[EFFECT_STRING_LEN_MAX]; // human readable effect implementor name
-} effect_descriptor_t;
-
-/////////////////////////////////////////////////
-// Effect control interface
-/////////////////////////////////////////////////
-
-//
-//--- Standardized command codes for command() function
-//
-enum effect_command_e {
- EFFECT_CMD_INIT, // initialize effect engine
- EFFECT_CMD_SET_CONFIG, // configure effect engine (see effect_config_t)
- EFFECT_CMD_RESET, // reset effect engine
- EFFECT_CMD_ENABLE, // enable effect process
- EFFECT_CMD_DISABLE, // disable effect process
- EFFECT_CMD_SET_PARAM, // set parameter immediately (see effect_param_t)
- EFFECT_CMD_SET_PARAM_DEFERRED, // set parameter deferred
- EFFECT_CMD_SET_PARAM_COMMIT, // commit previous set parameter deferred
- EFFECT_CMD_GET_PARAM, // get parameter
- EFFECT_CMD_SET_DEVICE, // set audio device (see audio.h, audio_devices_t)
- EFFECT_CMD_SET_VOLUME, // set volume
- EFFECT_CMD_SET_AUDIO_MODE, // set the audio mode (normal, ring, ...)
- EFFECT_CMD_SET_CONFIG_REVERSE, // configure effect engine reverse stream(see effect_config_t)
- EFFECT_CMD_SET_INPUT_DEVICE, // set capture device (see audio.h, audio_devices_t)
- EFFECT_CMD_GET_CONFIG, // read effect engine configuration
- EFFECT_CMD_GET_CONFIG_REVERSE, // read configure effect engine reverse stream configuration
- EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS, // get all supported configurations for a feature.
- EFFECT_CMD_GET_FEATURE_CONFIG, // get current feature configuration
- EFFECT_CMD_SET_FEATURE_CONFIG, // set current feature configuration
- EFFECT_CMD_SET_AUDIO_SOURCE, // set the audio source (see audio.h, audio_source_t)
- EFFECT_CMD_OFFLOAD, // set if effect thread is an offload one,
- // send the ioHandle of the effect thread
- EFFECT_CMD_FIRST_PROPRIETARY = 0x10000 // first proprietary command code
-};
-
-//==================================================================================================
-// command: EFFECT_CMD_INIT
-//--------------------------------------------------------------------------------------------------
-// description:
-// Initialize effect engine: All configurations return to default
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_SET_CONFIG
-//--------------------------------------------------------------------------------------------------
-// description:
-// Apply new audio parameters configurations for input and output buffers
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(effect_config_t)
-// data: effect_config_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_RESET
-//--------------------------------------------------------------------------------------------------
-// description:
-// Reset the effect engine. Keep configuration but resets state and buffer content
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 0
-// data: N/A
-//==================================================================================================
-// command: EFFECT_CMD_ENABLE
-//--------------------------------------------------------------------------------------------------
-// description:
-// Enable the process. Called by the framework before the first call to process()
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_DISABLE
-//--------------------------------------------------------------------------------------------------
-// description:
-// Disable the process. Called by the framework after the last call to process()
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_SET_PARAM
-//--------------------------------------------------------------------------------------------------
-// description:
-// Set a parameter and apply it immediately
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(effect_param_t) + size of param and value
-// data: effect_param_t + param + value. See effect_param_t definition below for value offset
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_SET_PARAM_DEFERRED
-//--------------------------------------------------------------------------------------------------
-// description:
-// Set a parameter but apply it only when receiving EFFECT_CMD_SET_PARAM_COMMIT command
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(effect_param_t) + size of param and value
-// data: effect_param_t + param + value. See effect_param_t definition below for value offset
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 0
-// data: N/A
-//==================================================================================================
-// command: EFFECT_CMD_SET_PARAM_COMMIT
-//--------------------------------------------------------------------------------------------------
-// description:
-// Apply all previously received EFFECT_CMD_SET_PARAM_DEFERRED commands
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_GET_PARAM
-//--------------------------------------------------------------------------------------------------
-// description:
-// Get a parameter value
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(effect_param_t) + size of param
-// data: effect_param_t + param
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(effect_param_t) + size of param and value
-// data: effect_param_t + param + value. See effect_param_t definition below for value offset
-//==================================================================================================
-// command: EFFECT_CMD_SET_DEVICE
-//--------------------------------------------------------------------------------------------------
-// description:
-// Set the rendering device the audio output path is connected to. See audio.h, audio_devices_t
-// for device values.
-// The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this
-// command when the device changes
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(uint32_t)
-// data: uint32_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 0
-// data: N/A
-//==================================================================================================
-// command: EFFECT_CMD_SET_VOLUME
-//--------------------------------------------------------------------------------------------------
-// description:
-// Set and get volume. Used by audio framework to delegate volume control to effect engine.
-// The effect implementation must set EFFECT_FLAG_VOLUME_IND or EFFECT_FLAG_VOLUME_CTRL flag in
-// its descriptor to receive this command before every call to process() function
-// If EFFECT_FLAG_VOLUME_CTRL flag is set in the effect descriptor, 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.
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: n * sizeof(uint32_t)
-// data: volume for each channel defined in effect_config_t for output buffer expressed in
-// 8.24 fixed point format
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: n * sizeof(uint32_t) / 0
-// data: - if EFFECT_FLAG_VOLUME_CTRL is set in effect descriptor:
-// volume for each channel defined in effect_config_t for output buffer expressed in
-// 8.24 fixed point format
-// - if EFFECT_FLAG_VOLUME_CTRL is not set in effect descriptor:
-// N/A
-// It is legal to receive a null pointer as pReplyData in which case the effect framework has
-// delegated volume control to another effect
-//==================================================================================================
-// command: EFFECT_CMD_SET_AUDIO_MODE
-//--------------------------------------------------------------------------------------------------
-// description:
-// 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.
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(uint32_t)
-// data: audio_mode_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 0
-// data: N/A
-//==================================================================================================
-// command: EFFECT_CMD_SET_CONFIG_REVERSE
-//--------------------------------------------------------------------------------------------------
-// description:
-// 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.
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(effect_config_t)
-// data: effect_config_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(int)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_SET_INPUT_DEVICE
-//--------------------------------------------------------------------------------------------------
-// description:
-// Set the capture device the audio input path is connected to. See audio.h, audio_devices_t
-// for device values.
-// The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this
-// command when the device changes
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(uint32_t)
-// data: uint32_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 0
-// data: N/A
-//==================================================================================================
-// command: EFFECT_CMD_GET_CONFIG
-//--------------------------------------------------------------------------------------------------
-// description:
-// Read audio parameters configurations for input and output buffers
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(effect_config_t)
-// data: effect_config_t
-//==================================================================================================
-// command: EFFECT_CMD_GET_CONFIG_REVERSE
-//--------------------------------------------------------------------------------------------------
-// description:
-// Read audio parameters configurations for input and output buffers of reverse stream
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 0
-// data: N/A
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(effect_config_t)
-// data: effect_config_t
-//==================================================================================================
-// command: EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS
-//--------------------------------------------------------------------------------------------------
-// description:
-// Queries for supported configurations for a particular feature (e.g. get the supported
-// combinations of main and auxiliary channels for a noise suppressor).
-// The command parameter is the feature identifier (See effect_feature_e for a list of defined
-// features) followed by the maximum number of configuration descriptor to return.
-// The reply is composed of:
-// - status (uint32_t):
-// - 0 if feature is supported
-// - -ENOSYS if the feature is not supported,
-// - -ENOMEM if the feature is supported but the total number of supported configurations
-// exceeds the maximum number indicated by the caller.
-// - total number of supported configurations (uint32_t)
-// - an array of configuration descriptors.
-// The actual number of descriptors returned must not exceed the maximum number indicated by
-// the caller.
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: 2 x sizeof(uint32_t)
-// data: effect_feature_e + maximum number of configurations to return
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 2 x sizeof(uint32_t) + n x sizeof (<config descriptor>)
-// data: status + total number of configurations supported + array of n config descriptors
-//==================================================================================================
-// command: EFFECT_CMD_GET_FEATURE_CONFIG
-//--------------------------------------------------------------------------------------------------
-// description:
-// Retrieves current configuration for a given feature.
-// The reply status is:
-// - 0 if feature is supported
-// - -ENOSYS if the feature is not supported,
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(uint32_t)
-// data: effect_feature_e
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(uint32_t) + sizeof (<config descriptor>)
-// data: status + config descriptor
-//==================================================================================================
-// command: EFFECT_CMD_SET_FEATURE_CONFIG
-//--------------------------------------------------------------------------------------------------
-// description:
-// Sets current configuration for a given feature.
-// The reply status is:
-// - 0 if feature is supported
-// - -ENOSYS if the feature is not supported,
-// - -EINVAL if the configuration is invalid
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(uint32_t) + sizeof (<config descriptor>)
-// data: effect_feature_e + config descriptor
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(uint32_t)
-// data: status
-//==================================================================================================
-// command: EFFECT_CMD_SET_AUDIO_SOURCE
-//--------------------------------------------------------------------------------------------------
-// description:
-// Set the audio source the capture path is configured for (Camcorder, voice recognition...).
-// See audio.h, audio_source_t for values.
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(uint32_t)
-// data: uint32_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: 0
-// data: N/A
-//==================================================================================================
-// command: EFFECT_CMD_OFFLOAD
-//--------------------------------------------------------------------------------------------------
-// description:
-// 1.indicate if the playback thread the effect is attached to is offloaded or not
-// 2.update the io handle of the playback thread the effect is attached to
-//--------------------------------------------------------------------------------------------------
-// command format:
-// size: sizeof(effect_offload_param_t)
-// data: effect_offload_param_t
-//--------------------------------------------------------------------------------------------------
-// reply format:
-// size: sizeof(uint32_t)
-// data: uint32_t
-//--------------------------------------------------------------------------------------------------
-// command: EFFECT_CMD_FIRST_PROPRIETARY
-//--------------------------------------------------------------------------------------------------
-// description:
-// All proprietary effect commands must use command codes above this value. The size and format of
-// command and response fields is free in this case
-//==================================================================================================
-
-// Audio buffer descriptor used by process(), bufferProvider() functions and buffer_config_t
-// structure. 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: left, right
-// 5 point 1: front left, front right, front center, low frequency, back left, back right
-// 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
-typedef struct audio_buffer_s {
- size_t frameCount; // number of frames in buffer
- union {
- void* raw; // raw pointer to start of buffer
- float* f32; // pointer to float 32 bit data at start of buffer
- int32_t* s32; // pointer to signed 32 bit data at start of buffer
- int16_t* s16; // pointer to signed 16 bit data at start of buffer
- uint8_t* u8; // pointer to unsigned 8 bit data at start of buffer
- };
-} audio_buffer_t;
-
-// The buffer_provider_s structure contains functions that can be used
-// by the effect engine process() function to query and release input
-// or output audio buffer.
-// The getBuffer() function is called to retrieve a buffer where data
-// should read from or written to by process() function.
-// The releaseBuffer() function MUST be called when the buffer retrieved
-// with getBuffer() is not needed anymore.
-// The process function should use the buffer provider mechanism to retrieve
-// input or output buffer if the inBuffer or outBuffer passed as argument is NULL
-// and the buffer configuration (buffer_config_t) given by the EFFECT_CMD_SET_CONFIG
-// command did not specify an audio buffer.
-
-typedef int32_t (*buffer_function_t)(void* cookie, audio_buffer_t* buffer);
-
-typedef struct buffer_provider_s {
- buffer_function_t getBuffer; // retrieve next buffer
- buffer_function_t releaseBuffer; // release used buffer
- void* cookie; // for use by client of buffer provider functions
-} buffer_provider_t;
-
-// The buffer_config_s structure specifies the input or output audio format
-// to be used by the effect engine.
-typedef struct buffer_config_s {
- audio_buffer_t buffer; // buffer for use by process() function if not passed explicitly
- uint32_t samplingRate; // sampling rate
- uint32_t channels; // channel mask (see audio_channel_mask_t in audio.h)
- buffer_provider_t bufferProvider; // buffer provider
- uint8_t format; // Audio format (see audio_format_t in audio.h)
- uint8_t accessMode; // read/write or accumulate in buffer (effect_buffer_access_e)
- uint16_t mask; // indicates which of the above fields is valid
-} buffer_config_t;
-
-// EFFECT_FEATURE_AUX_CHANNELS feature configuration descriptor. Describe a combination
-// of main and auxiliary channels supported
-typedef struct channel_config_s {
- audio_channel_mask_t main_channels; // channel mask for main channels
- audio_channel_mask_t aux_channels; // channel mask for auxiliary channels
-} channel_config_t;
-
-// effect_config_s structure is used to configure audio parameters and buffers for effect engine
-// input and output.
-typedef struct effect_config_s {
- buffer_config_t inputCfg;
- buffer_config_t outputCfg;
-} effect_config_t;
-
-// effect_param_s structure describes the format of the pCmdData argument of EFFECT_CMD_SET_PARAM
-// command and pCmdData and pReplyData of EFFECT_CMD_GET_PARAM command.
-// psize and vsize represent the actual size of parameter and value.
-//
-// NOTE: the start of value field inside the data field is always on a 32 bit boundary:
-//
-// +-----------+
-// | status | sizeof(int)
-// +-----------+
-// | psize | sizeof(int)
-// +-----------+
-// | vsize | sizeof(int)
-// +-----------+
-// | | | |
-// ~ parameter ~ > psize |
-// | | | > ((psize - 1)/sizeof(int) + 1) * sizeof(int)
-// +-----------+ |
-// | padding | |
-// +-----------+
-// | | |
-// ~ value ~ > vsize
-// | | |
-// +-----------+
-
-typedef struct effect_param_s {
- int32_t status; // Transaction status (unused for command, used for reply)
- uint32_t psize; // Parameter size
- uint32_t vsize; // Value size
- char data[]; // Start of Parameter + Value data
-} effect_param_t;
-
-// Maximum effect_param_t size
-#define EFFECT_PARAM_SIZE_MAX 65536
-
-// structure used by EFFECT_CMD_OFFLOAD command
-typedef struct effect_offload_param_s {
- bool isOffload; // true if the playback thread the effect is attached to is offloaded
- int ioHandle; // io handle of the playback thread the effect is attached to
-} effect_offload_param_t;
-
-__END_DECLS
-
-#endif // ANDROID_AUDIO_EFFECT_CORE_H
diff --git a/audio/common/all-versions/util/Android.bp b/audio/common/all-versions/util/Android.bp
index 7132667..5d33a3a 100644
--- a/audio/common/all-versions/util/Android.bp
+++ b/audio/common/all-versions/util/Android.bp
@@ -1,7 +1,10 @@
cc_library_headers {
name: "android.hardware.audio.common.util@all-versions",
defaults: ["hidl_defaults"],
- vendor: true,
+ vendor_available: true,
+ vndk: {
+ enabled: true,
+ },
export_include_dirs: ["include"],
}
diff --git a/audio/core/2.0/default/Android.bp b/audio/core/2.0/default/Android.bp
index 87e6a9a..9847886 100644
--- a/audio/core/2.0/default/Android.bp
+++ b/audio/core/2.0/default/Android.bp
@@ -37,13 +37,13 @@
"android.hardware.audio.common.util@all-versions",
"android.hardware.audio.core@all-versions-impl",
"libaudioclient_headers",
- "android.hardware.audio.common.legacy@2.0",
+ "libaudio_system_headers",
"libhardware_headers",
"libmedia_headers",
],
whole_static_libs: [
- "libmedia_helper@2.0",
+ "libmedia_helper",
],
}
diff --git a/audio/core/all-versions/default/Android.bp b/audio/core/all-versions/default/Android.bp
index a02a6bb..214b8d5 100644
--- a/audio/core/all-versions/default/Android.bp
+++ b/audio/core/all-versions/default/Android.bp
@@ -22,7 +22,7 @@
header_libs: [
"libaudioclient_headers",
- "android.hardware.audio.common.legacy@2.0",
+ "libaudio_system_headers",
"libhardware_headers",
"libmedia_headers",
"android.hardware.audio.common.util@all-versions",
diff --git a/audio/effect/2.0/default/Android.bp b/audio/effect/2.0/default/Android.bp
index d32a9d9..db00988 100644
--- a/audio/effect/2.0/default/Android.bp
+++ b/audio/effect/2.0/default/Android.bp
@@ -41,9 +41,9 @@
header_libs: [
"android.hardware.audio.common.util@all-versions",
"android.hardware.audio.effect@all-versions-impl",
- "android.hardware.audio.common.legacy@2.0",
- "android.hardware.audio.effect.legacy@2.0",
+ "libaudio_system_headers",
"libaudioclient_headers",
+ "libeffects_headers",
"libhardware_headers",
"libmedia_headers",
],
diff --git a/audio/effect/2.0/legacy/Android.bp b/audio/effect/2.0/legacy/Android.bp
deleted file mode 100644
index 68de70e..0000000
--- a/audio/effect/2.0/legacy/Android.bp
+++ /dev/null
@@ -1,12 +0,0 @@
-cc_library_headers {
- name: "android.hardware.audio.effect.legacy@2.0",
- vendor: true,
- header_libs: [
- "android.hardware.audio.common.legacy@2.0",
- "android.hardware.audio.effect.legacy@all-versions",
- ],
- export_header_lib_headers: [
- "android.hardware.audio.common.legacy@2.0",
- "android.hardware.audio.effect.legacy@all-versions",
- ],
-}
diff --git a/audio/effect/2.0/legacy/OWNERS b/audio/effect/2.0/legacy/OWNERS
deleted file mode 100644
index 6fdc97c..0000000
--- a/audio/effect/2.0/legacy/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-elaurent@google.com
-krocard@google.com
-mnaganov@google.com
diff --git a/audio/effect/all-versions/default/Android.bp b/audio/effect/all-versions/default/Android.bp
index 47d74a8..ed2a093 100644
--- a/audio/effect/all-versions/default/Android.bp
+++ b/audio/effect/all-versions/default/Android.bp
@@ -9,6 +9,7 @@
shared_libs: [
"libbase",
"libcutils",
+ "libeffects",
"libfmq",
"libhidlbase",
"libhidlmemory",
@@ -20,7 +21,9 @@
],
header_libs: [
+ "libaudio_system_headers",
"libaudioclient_headers",
+ "libeffects_headers",
"libhardware_headers",
"libmedia_headers",
"android.hardware.audio.common.util@all-versions",
diff --git a/audio/effect/all-versions/legacy/Android.bp b/audio/effect/all-versions/legacy/Android.bp
deleted file mode 100644
index bcf81b3..0000000
--- a/audio/effect/all-versions/legacy/Android.bp
+++ /dev/null
@@ -1,11 +0,0 @@
-cc_library_headers {
- name: "android.hardware.audio.effect.legacy@all-versions",
- vendor: true,
- export_include_dirs: ["include"],
- header_libs: [
- "android.hardware.audio.common.legacy@all-versions",
- ],
- export_header_lib_headers: [
- "android.hardware.audio.common.legacy@all-versions",
- ],
-}
diff --git a/audio/effect/all-versions/legacy/OWNERS b/audio/effect/all-versions/legacy/OWNERS
deleted file mode 100644
index 6fdc97c..0000000
--- a/audio/effect/all-versions/legacy/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-elaurent@google.com
-krocard@google.com
-mnaganov@google.com
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_aec.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_aec.h
deleted file mode 100644
index f48749a..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_aec.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_AEC_H_
-#define ANDROID_EFFECT_AEC_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_aec.h>
-
-#endif /*ANDROID_EFFECT_AEC_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_agc.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_agc.h
deleted file mode 100644
index 466ea96..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_agc.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_AGC_H_
-#define ANDROID_EFFECT_AGC_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_agc.h>
-
-#endif /*ANDROID_EFFECT_AGC_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_bassboost.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_bassboost.h
deleted file mode 100644
index 157452e..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_bassboost.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_BASSBOOST_H_
-#define ANDROID_EFFECT_BASSBOOST_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_bassboost.h>
-
-#endif /*ANDROID_EFFECT_BASSBOOST_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_downmix.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_downmix.h
deleted file mode 100644
index 26b849b..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_downmix.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2012 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_DOWNMIX_H_
-#define ANDROID_EFFECT_DOWNMIX_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_downmix.h>
-
-#endif /*ANDROID_EFFECT_DOWNMIX_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_environmentalreverb.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_environmentalreverb.h
deleted file mode 100644
index dd474c2..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_environmentalreverb.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_ENVIRONMENTALREVERB_H_
-#define ANDROID_EFFECT_ENVIRONMENTALREVERB_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_environmentalreverb.h>
-
-#endif /*ANDROID_EFFECT_ENVIRONMENTALREVERB_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_equalizer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_equalizer.h
deleted file mode 100644
index 3059ec2..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_equalizer.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_EQUALIZER_H_
-#define ANDROID_EFFECT_EQUALIZER_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_equalizer.h>
-
-#endif /*ANDROID_EFFECT_EQUALIZER_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_loudnessenhancer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_loudnessenhancer.h
deleted file mode 100644
index f37ba45..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_loudnessenhancer.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2013 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_LOUDNESS_ENHANCER_H_
-#define ANDROID_EFFECT_LOUDNESS_ENHANCER_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_loudnessenhancer.h>
-
-#endif /*ANDROID_EFFECT_LOUDNESS_ENHANCER_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_ns.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_ns.h
deleted file mode 100644
index 3bd8a41..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_ns.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_NS_H_
-#define ANDROID_EFFECT_NS_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_ns.h>
-
-#endif /*ANDROID_EFFECT_NS_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_presetreverb.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_presetreverb.h
deleted file mode 100644
index eac1f5f..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_presetreverb.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_PRESETREVERB_H_
-#define ANDROID_EFFECT_PRESETREVERB_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_presetreverb.h>
-
-#endif /*ANDROID_EFFECT_PRESETREVERB_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_virtualizer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_virtualizer.h
deleted file mode 100644
index aeecfa5..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_virtualizer.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_VIRTUALIZER_H_
-#define ANDROID_EFFECT_VIRTUALIZER_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_virtualizer.h>
-
-#endif /*ANDROID_EFFECT_VIRTUALIZER_H_*/
diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_visualizer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_visualizer.h
deleted file mode 100644
index 47217e7..0000000
--- a/audio/effect/all-versions/legacy/include/audio_effects/effect_visualizer.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-/*
- * USAGE NOTE: Only include this header when _implementing_ a particular
- * effect. When access to UUID and properties is enough, include the
- * corresponding header from system/audio_effects/, which doesn't include
- * hardware/audio_effect.h.
- *
- * Only code that immediately calls into HAL or implements an effect
- * can import hardware/audio_effect.h.
- */
-
-#ifndef ANDROID_EFFECT_VISUALIZER_H_
-#define ANDROID_EFFECT_VISUALIZER_H_
-
-#include <hardware/audio_effect.h>
-#include <system/audio_effects/effect_visualizer.h>
-
-#endif /*ANDROID_EFFECT_VISUALIZER_H_*/
diff --git a/audio/effect/all-versions/legacy/include/media/EffectsFactoryApi.h b/audio/effect/all-versions/legacy/include/media/EffectsFactoryApi.h
deleted file mode 100644
index e08fd0b..0000000
--- a/audio/effect/all-versions/legacy/include/media/EffectsFactoryApi.h
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (C) 2010 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.
- */
-
-#ifndef ANDROID_EFFECTSFACTORYAPI_H_
-#define ANDROID_EFFECTSFACTORYAPI_H_
-
-#include <cutils/compiler.h>
-#include <errno.h>
-#include <hardware/audio_effect.h>
-#include <stdint.h>
-#include <sys/types.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-/////////////////////////////////////////////////
-// Effect factory interface
-/////////////////////////////////////////////////
-
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectQueryNumberEffects
-//
-// Description: Returns the number of different effects in all loaded libraries.
-// Each effect must have a different effect uuid (see
-// effect_descriptor_t). This function together with EffectQueryEffect()
-// is used to enumerate all effects present in all loaded libraries.
-// Each time EffectQueryNumberEffects() is called, the factory must
-// reset the index of the effect descriptor returned by next call to
-// EffectQueryEffect() to restart enumeration from the beginning.
-//
-// Input/Output:
-// pNumEffects: address where the number of effects should be returned.
-//
-// Output:
-// returned value: 0 successful operation.
-// -ENODEV factory failed to initialize
-// -EINVAL invalid pNumEffects
-// *pNumEffects: updated with number of effects in factory
-//
-////////////////////////////////////////////////////////////////////////////////
-ANDROID_API
-int EffectQueryNumberEffects(uint32_t* pNumEffects);
-
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectQueryEffect
-//
-// Description: Returns a descriptor of the next available effect.
-// See effect_descriptor_t for a details on effect descriptor.
-// This function together with EffectQueryNumberEffects() is used to enumerate all
-// effects present in all loaded libraries. The enumeration sequence is:
-// EffectQueryNumberEffects(&num_effects);
-// for (i = 0; i < num_effects; i++)
-// EffectQueryEffect(i,...);
-//
-// Input/Output:
-// pDescriptor: address where to return the effect descriptor.
-//
-// Output:
-// returned value: 0 successful operation.
-// -ENOENT no more effect available
-// -ENODEV factory failed to initialize
-// -EINVAL invalid pDescriptor
-// -ENOSYS effect list has changed since last execution of
-// EffectQueryNumberEffects()
-// *pDescriptor: updated with the effect descriptor.
-//
-////////////////////////////////////////////////////////////////////////////////
-ANDROID_API
-int EffectQueryEffect(uint32_t index, effect_descriptor_t* pDescriptor);
-
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectCreate
-//
-// Description: Creates an effect engine of the specified type and returns an
-// effect control interface on this engine. The function will allocate the
-// resources for an instance of the requested effect engine and return
-// a handle on the effect control interface.
-//
-// Input:
-// pEffectUuid: pointer to the effect uuid.
-// sessionId: 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. Knowing that two effects are part of the same effect chain can help the
-// library implement some kind of optimizations.
-// ioId: identifies the output or input stream this effect is directed to at audio HAL.
-// For future use especially with tunneled HW accelerated effects
-//
-// Input/Output:
-// pHandle: address where to return the effect handle.
-//
-// Output:
-// returned value: 0 successful operation.
-// -ENODEV factory failed to initialize
-// -EINVAL invalid pEffectUuid or pHandle
-// -ENOENT no effect with this uuid found
-// *pHandle: updated with the effect handle.
-//
-////////////////////////////////////////////////////////////////////////////////
-ANDROID_API
-int EffectCreate(const effect_uuid_t* pEffectUuid, int32_t sessionId, int32_t ioId,
- effect_handle_t* pHandle);
-
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectRelease
-//
-// Description: Releases the effect engine whose handle is given as argument.
-// All resources allocated to this particular instance of the effect are
-// released.
-//
-// Input:
-// handle: handle on the effect interface to be released.
-//
-// Output:
-// returned value: 0 successful operation.
-// -ENODEV factory failed to initialize
-// -EINVAL invalid interface handle
-//
-////////////////////////////////////////////////////////////////////////////////
-ANDROID_API
-int EffectRelease(effect_handle_t handle);
-
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectGetDescriptor
-//
-// Description: Returns the descriptor of the effect which uuid is pointed
-// to by first argument.
-//
-// Input:
-// pEffectUuid: pointer to the effect uuid.
-//
-// Input/Output:
-// pDescriptor: address where to return the effect descriptor.
-//
-// Output:
-// returned value: 0 successful operation.
-// -ENODEV factory failed to initialize
-// -EINVAL invalid pEffectUuid or pDescriptor
-// -ENOENT no effect with this uuid found
-// *pDescriptor: updated with the effect descriptor.
-//
-////////////////////////////////////////////////////////////////////////////////
-ANDROID_API
-int EffectGetDescriptor(const effect_uuid_t* pEffectUuid, effect_descriptor_t* pDescriptor);
-
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectIsNullUuid
-//
-// Description: Helper function to compare effect uuid to EFFECT_UUID_NULL
-//
-// Input:
-// pEffectUuid: pointer to effect uuid to compare to EFFECT_UUID_NULL.
-//
-// Output:
-// returned value: 0 if uuid is different from EFFECT_UUID_NULL.
-// 1 if uuid is equal to EFFECT_UUID_NULL.
-//
-////////////////////////////////////////////////////////////////////////////////
-ANDROID_API
-int EffectIsNullUuid(const effect_uuid_t* pEffectUuid);
-
-ANDROID_API
-int EffectDumpEffects(int fd);
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECTSFACTORYAPI_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/audio_effects_conf.h b/audio/effect/all-versions/legacy/include/system/audio_effects/audio_effects_conf.h
deleted file mode 100644
index b68a6c2..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/audio_effects_conf.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (C) 2011 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.
- */
-
-#ifndef ANDROID_AUDIO_EFFECTS_CONF_H
-#define ANDROID_AUDIO_EFFECTS_CONF_H
-
-/////////////////////////////////////////////////
-// Definitions for effects configuration file (audio_effects.conf)
-/////////////////////////////////////////////////
-
-#define AUDIO_EFFECT_DEFAULT_CONFIG_FILE "/system/etc/audio_effects.conf"
-#define AUDIO_EFFECT_VENDOR_CONFIG_FILE "/vendor/etc/audio_effects.conf"
-#define LIBRARIES_TAG "libraries"
-#define PATH_TAG "path"
-
-#define EFFECTS_TAG "effects"
-#define LIBRARY_TAG "library"
-#define UUID_TAG "uuid"
-
-#define PREPROCESSING_TAG "pre_processing"
-#define OUTPUT_SESSION_PROCESSING_TAG "output_session_processing"
-
-#define PARAM_TAG "param"
-#define VALUE_TAG "value"
-#define INT_TAG "int"
-#define SHORT_TAG "short"
-#define FLOAT_TAG "float"
-#define BOOL_TAG "bool"
-#define STRING_TAG "string"
-
-// audio_source_t
-#define MIC_SRC_TAG "mic" // AUDIO_SOURCE_MIC
-#define VOICE_UL_SRC_TAG "voice_uplink" // AUDIO_SOURCE_VOICE_UPLINK
-#define VOICE_DL_SRC_TAG "voice_downlink" // AUDIO_SOURCE_VOICE_DOWNLINK
-#define VOICE_CALL_SRC_TAG "voice_call" // AUDIO_SOURCE_VOICE_CALL
-#define CAMCORDER_SRC_TAG "camcorder" // AUDIO_SOURCE_CAMCORDER
-#define VOICE_REC_SRC_TAG "voice_recognition" // AUDIO_SOURCE_VOICE_RECOGNITION
-#define VOICE_COMM_SRC_TAG "voice_communication" // AUDIO_SOURCE_VOICE_COMMUNICATION
-#define UNPROCESSED_SRC_TAG "unprocessed" // AUDIO_SOURCE_UNPROCESSED
-
-// audio_stream_type_t
-#define AUDIO_STREAM_DEFAULT_TAG "default"
-#define AUDIO_STREAM_VOICE_CALL_TAG "voice_call"
-#define AUDIO_STREAM_SYSTEM_TAG "system"
-#define AUDIO_STREAM_RING_TAG "ring"
-#define AUDIO_STREAM_MUSIC_TAG "music"
-#define AUDIO_STREAM_ALARM_TAG "alarm"
-#define AUDIO_STREAM_NOTIFICATION_TAG "notification"
-#define AUDIO_STREAM_BLUETOOTH_SCO_TAG "bluetooth_sco"
-#define AUDIO_STREAM_ENFORCED_AUDIBLE_TAG "enforced_audible"
-#define AUDIO_STREAM_DTMF_TAG "dtmf"
-#define AUDIO_STREAM_TTS_TAG "tts"
-
-#endif // ANDROID_AUDIO_EFFECTS_CONF_H
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_aec.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_aec.h
deleted file mode 100644
index 9785055..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_aec.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_AEC_CORE_H_
-#define ANDROID_EFFECT_AEC_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-// The AEC type UUID is not defined by OpenSL ES and has been generated from
-// http://www.itu.int/ITU-T/asn1/uuid.html
-static const effect_uuid_t FX_IID_AEC_ = {
- 0x7b491460, 0x8d4d, 0x11e0, 0xbd61, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const FX_IID_AEC = &FX_IID_AEC_;
-
-typedef enum {
- AEC_PARAM_ECHO_DELAY, // echo delay in microseconds
- AEC_PARAM_PROPERTIES
-} t_aec_params;
-
-// t_equalizer_settings groups all current aec settings for backup and restore.
-typedef struct s_aec_settings { uint32_t echoDelay; } t_aec_settings;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_AEC_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_agc.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_agc.h
deleted file mode 100644
index 319bcd4..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_agc.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_AGC_CORE_H_
-#define ANDROID_EFFECT_AGC_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-// The AGC type UUID is not defined by OpenSL ES and has been generated from
-// http://www.itu.int/ITU-T/asn1/uuid.html
-static const effect_uuid_t FX_IID_AGC_ = {
- 0x0a8abfe0, 0x654c, 0x11e0, 0xba26, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const FX_IID_AGC = &FX_IID_AGC_;
-
-typedef enum {
- AGC_PARAM_TARGET_LEVEL, // target output level in millibel
- AGC_PARAM_COMP_GAIN, // gain in the compression range in millibel
- AGC_PARAM_LIMITER_ENA, // enable or disable limiter (boolean)
- AGC_PARAM_PROPERTIES
-} t_agc_params;
-
-// t_agc_settings groups all current agc settings for backup and restore.
-typedef struct s_agc_settings {
- int16_t targetLevel;
- int16_t compGain;
- bool limiterEnabled;
-} t_agc_settings;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_AGC_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_bassboost.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_bassboost.h
deleted file mode 100644
index 7828d66..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_bassboost.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_BASSBOOST_CORE_H_
-#define ANDROID_EFFECT_BASSBOOST_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_BASSBOOST_ = {
- 0x0634f220, 0xddd4, 0x11db, 0xa0fc, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const SL_IID_BASSBOOST = &SL_IID_BASSBOOST_;
-#endif // OPENSL_ES_H_
-
-/* enumerated parameter settings for BassBoost effect */
-typedef enum { BASSBOOST_PARAM_STRENGTH_SUPPORTED, BASSBOOST_PARAM_STRENGTH } t_bassboost_params;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_BASSBOOST_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_downmix.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_downmix.h
deleted file mode 100644
index 9f02e41..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_downmix.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_DOWNMIX_CORE_H_
-#define ANDROID_EFFECT_DOWNMIX_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-#define EFFECT_UIID_DOWNMIX__ \
- { \
- 0x381e49cc, 0xa858, 0x4aa2, 0x87f6, { 0xe8, 0x38, 0x8e, 0x76, 0x01, 0xb2 } \
- }
-static const effect_uuid_t EFFECT_UIID_DOWNMIX_ = EFFECT_UIID_DOWNMIX__;
-const effect_uuid_t* const EFFECT_UIID_DOWNMIX = &EFFECT_UIID_DOWNMIX_;
-
-/* enumerated parameter settings for downmix effect */
-typedef enum { DOWNMIX_PARAM_TYPE } downmix_params_t;
-
-typedef enum {
- DOWNMIX_TYPE_INVALID = -1,
- // throw away the extra channels
- DOWNMIX_TYPE_STRIP = 0,
- // mix the extra channels with FL/FR
- DOWNMIX_TYPE_FOLD = 1,
- DOWNMIX_TYPE_CNT,
- DOWNMIX_TYPE_LAST = DOWNMIX_TYPE_CNT - 1
-} downmix_type_t;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_DOWNMIX_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_environmentalreverb.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_environmentalreverb.h
deleted file mode 100644
index 8caee32..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_environmentalreverb.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_ENVIRONMENTALREVERB_CORE_H_
-#define ANDROID_EFFECT_ENVIRONMENTALREVERB_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_ENVIRONMENTALREVERB_ = {
- 0xc2e5d5f0, 0x94bd, 0x4763, 0x9cac, {0x4e, 0x23, 0x4d, 0x6, 0x83, 0x9e}};
-const effect_uuid_t* const SL_IID_ENVIRONMENTALREVERB = &SL_IID_ENVIRONMENTALREVERB_;
-#endif // OPENSL_ES_H_
-
-/* enumerated parameter settings for environmental reverb effect */
-typedef enum {
- // Parameters below are as defined in OpenSL ES specification for environmental reverb interface
- REVERB_PARAM_ROOM_LEVEL, // in millibels, range -6000 to 0
- REVERB_PARAM_ROOM_HF_LEVEL, // in millibels, range -4000 to 0
- REVERB_PARAM_DECAY_TIME, // in milliseconds, range 100 to 20000
- REVERB_PARAM_DECAY_HF_RATIO, // in permilles, range 100 to 1000
- REVERB_PARAM_REFLECTIONS_LEVEL, // in millibels, range -6000 to 0
- REVERB_PARAM_REFLECTIONS_DELAY, // in milliseconds, range 0 to 65
- REVERB_PARAM_REVERB_LEVEL, // in millibels, range -6000 to 0
- REVERB_PARAM_REVERB_DELAY, // in milliseconds, range 0 to 65
- REVERB_PARAM_DIFFUSION, // in permilles, range 0 to 1000
- REVERB_PARAM_DENSITY, // in permilles, range 0 to 1000
- REVERB_PARAM_PROPERTIES,
- REVERB_PARAM_BYPASS
-} t_env_reverb_params;
-
-// t_reverb_settings is equal to SLEnvironmentalReverbSettings defined in OpenSL ES specification.
-typedef struct s_reverb_settings {
- int16_t roomLevel;
- int16_t roomHFLevel;
- uint32_t decayTime;
- int16_t decayHFRatio;
- int16_t reflectionsLevel;
- uint32_t reflectionsDelay;
- int16_t reverbLevel;
- uint32_t reverbDelay;
- int16_t diffusion;
- int16_t density;
-} __attribute__((packed)) t_reverb_settings;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_ENVIRONMENTALREVERB_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_equalizer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_equalizer.h
deleted file mode 100644
index 83fddcf..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_equalizer.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_EQUALIZER_CORE_H_
-#define ANDROID_EFFECT_EQUALIZER_CORE_H_
-
-#include <system/audio_effect.h>
-
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_EQUALIZER_ = {
- 0x0bed4300, 0xddd6, 0x11db, 0x8f34, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const SL_IID_EQUALIZER = &SL_IID_EQUALIZER_;
-#endif // OPENSL_ES_H_
-
-#if __cplusplus
-extern "C" {
-#endif
-
-/* enumerated parameters for Equalizer effect */
-typedef enum {
- EQ_PARAM_NUM_BANDS, // Gets the number of frequency bands that the equalizer
- // supports.
- EQ_PARAM_LEVEL_RANGE, // Returns the minimum and maximum band levels supported.
- EQ_PARAM_BAND_LEVEL, // Gets/Sets the gain set for the given equalizer band.
- EQ_PARAM_CENTER_FREQ, // Gets the center frequency of the given band.
- EQ_PARAM_BAND_FREQ_RANGE, // Gets the frequency range of the given frequency band.
- EQ_PARAM_GET_BAND, // Gets the band that has the most effect on the given
- // frequency.
- EQ_PARAM_CUR_PRESET, // Gets/Sets the current preset.
- EQ_PARAM_GET_NUM_OF_PRESETS, // Gets the total number of presets the equalizer supports.
- EQ_PARAM_GET_PRESET_NAME, // Gets the preset name based on the index.
- EQ_PARAM_PROPERTIES // Gets/Sets all parameters at a time.
-} t_equalizer_params;
-
-// t_equalizer_settings groups all current equalizer setting for backup and restore.
-typedef struct s_equalizer_settings {
- uint16_t curPreset;
- uint16_t numBands;
- uint16_t bandLevels[];
-} t_equalizer_settings;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_EQUALIZER_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_loudnessenhancer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_loudnessenhancer.h
deleted file mode 100644
index 5c78013..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_loudnessenhancer.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_LOUDNESS_ENHANCER_CORE_H_
-#define ANDROID_EFFECT_LOUDNESS_ENHANCER_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-// this effect is not defined in OpenSL ES as one of the standard effects
-static const effect_uuid_t FX_IID_LOUDNESS_ENHANCER_ = {
- 0xfe3199be, 0xaed0, 0x413f, 0x87bb, {0x11, 0x26, 0x0e, 0xb6, 0x3c, 0xf1}};
-const effect_uuid_t* const FX_IID_LOUDNESS_ENHANCER = &FX_IID_LOUDNESS_ENHANCER_;
-
-#define LOUDNESS_ENHANCER_DEFAULT_TARGET_GAIN_MB 0 // mB
-
-// enumerated parameters for DRC effect
-// to keep in sync with frameworks/base/media/java/android/media/audiofx/LoudnessEnhancer.java
-typedef enum {
- LOUDNESS_ENHANCER_PARAM_TARGET_GAIN_MB = 0, // target gain expressed in mB
-} t_level_monitor_params;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_LOUDNESS_ENHANCER_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_ns.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_ns.h
deleted file mode 100644
index 8b9ac76..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_ns.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_NS_CORE_H_
-#define ANDROID_EFFECT_NS_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-// The NS type UUID is not defined by OpenSL ES and has been generated from
-// http://www.itu.int/ITU-T/asn1/uuid.html
-static const effect_uuid_t FX_IID_NS_ = {
- 0x58b4b260, 0x8e06, 0x11e0, 0xaa8e, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const FX_IID_NS = &FX_IID_NS_;
-
-typedef enum {
- NS_PARAM_LEVEL, // noise suppression level (t_ns_level)
- NS_PARAM_PROPERTIES,
- NS_PARAM_TYPE // noise suppression type (t_ns_type)
-} t_ns_params;
-
-// noise suppression level
-typedef enum { NS_LEVEL_LOW, NS_LEVEL_MEDIUM, NS_LEVEL_HIGH } t_ns_level;
-
-// noise suppression type
-typedef enum { NS_TYPE_SINGLE_CHANNEL, NS_TYPE_MULTI_CHANNEL } t_ns_type;
-
-// s_ns_settings groups all current ns settings for backup and restore.
-typedef struct s_ns_settings {
- uint32_t level;
- uint32_t type;
-} t_ns_settings;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_NS_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_presetreverb.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_presetreverb.h
deleted file mode 100644
index 6804fed..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_presetreverb.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_PRESETREVERB_CORE_H_
-#define ANDROID_EFFECT_PRESETREVERB_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_PRESETREVERB_ = {
- 0x47382d60, 0xddd8, 0x11db, 0xbf3a, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const SL_IID_PRESETREVERB = &SL_IID_PRESETREVERB_;
-#endif // OPENSL_ES_H_
-
-/* enumerated parameter settings for preset reverb effect */
-typedef enum { REVERB_PARAM_PRESET } t_preset_reverb_params;
-
-typedef enum {
- REVERB_PRESET_NONE,
- REVERB_PRESET_SMALLROOM,
- REVERB_PRESET_MEDIUMROOM,
- REVERB_PRESET_LARGEROOM,
- REVERB_PRESET_MEDIUMHALL,
- REVERB_PRESET_LARGEHALL,
- REVERB_PRESET_PLATE,
- REVERB_PRESET_LAST = REVERB_PRESET_PLATE
-} t_reverb_presets;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_PRESETREVERB_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_virtualizer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_virtualizer.h
deleted file mode 100644
index a6a31ec..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_virtualizer.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_VIRTUALIZER_CORE_H_
-#define ANDROID_EFFECT_VIRTUALIZER_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_VIRTUALIZER_ = {
- 0x37cc2c00, 0xdddd, 0x11db, 0x8577, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const SL_IID_VIRTUALIZER = &SL_IID_VIRTUALIZER_;
-#endif // OPENSL_ES_H_
-
-/* enumerated parameter settings for virtualizer effect */
-/* to keep in sync with frameworks/base/media/java/android/media/audiofx/Virtualizer.java */
-typedef enum {
- VIRTUALIZER_PARAM_STRENGTH_SUPPORTED,
- VIRTUALIZER_PARAM_STRENGTH,
- // used with EFFECT_CMD_GET_PARAM
- // format:
- // parameters int32_t VIRTUALIZER_PARAM_VIRTUAL_SPEAKER_ANGLES
- // audio_channel_mask_t input channel mask
- // audio_devices_t audio output device
- // output int32_t* an array of length 3 * the number of channels in the mask
- // where entries are the succession of the channel mask
- // of each speaker (i.e. a single bit is selected in the
- // channel mask) followed by the azimuth and the
- // elevation angles.
- // status int -EINVAL if configuration is not supported or invalid or not forcing
- // 0 if configuration is supported and the mode is forced
- // notes:
- // - all angles are expressed in degrees and are relative to the listener,
- // - for azimuth: 0 is the direction the listener faces, 180 is behind the listener, and
- // -90 is to her/his left,
- // - for elevation: 0 is the horizontal plane, +90 is above the listener, -90 is below.
- VIRTUALIZER_PARAM_VIRTUAL_SPEAKER_ANGLES,
- // used with EFFECT_CMD_SET_PARAM
- // format:
- // parameters int32_t VIRTUALIZER_PARAM_FORCE_VIRTUALIZATION_MODE
- // audio_devices_t audio output device
- // status int -EINVAL if the device is not supported or invalid
- // 0 if the device is supported and the mode is forced, or forcing
- // was disabled for the AUDIO_DEVICE_NONE audio device.
- VIRTUALIZER_PARAM_FORCE_VIRTUALIZATION_MODE,
- // used with EFFECT_CMD_GET_PARAM
- // format:
- // parameters int32_t VIRTUALIZER_PARAM_VIRTUALIZATION_MODE
- // output audio_device_t audio device reflecting the current virtualization mode,
- // AUDIO_DEVICE_NONE when not virtualizing
- // status int -EINVAL if an error occurred
- // 0 if the output value is successfully retrieved
- VIRTUALIZER_PARAM_VIRTUALIZATION_MODE
-} t_virtualizer_params;
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_VIRTUALIZER_CORE_H_*/
diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_visualizer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_visualizer.h
deleted file mode 100644
index cc78e15..0000000
--- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_visualizer.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (C) 2016 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.
- */
-
-#ifndef ANDROID_EFFECT_VISUALIZER_CORE_H_
-#define ANDROID_EFFECT_VISUALIZER_CORE_H_
-
-#include <system/audio_effect.h>
-
-#if __cplusplus
-extern "C" {
-#endif
-
-#ifndef OPENSL_ES_H_
-static const effect_uuid_t SL_IID_VISUALIZATION_ = {
- 0xe46b26a0, 0xdddd, 0x11db, 0x8afd, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
-const effect_uuid_t* const SL_IID_VISUALIZATION = &SL_IID_VISUALIZATION_;
-#endif // OPENSL_ES_H_
-
-#define VISUALIZER_CAPTURE_SIZE_MAX 1024 // maximum capture size in samples
-#define VISUALIZER_CAPTURE_SIZE_MIN 128 // minimum capture size in samples
-
-// to keep in sync with frameworks/base/media/java/android/media/audiofx/Visualizer.java
-#define VISUALIZER_SCALING_MODE_NORMALIZED 0
-#define VISUALIZER_SCALING_MODE_AS_PLAYED 1
-
-#define MEASUREMENT_MODE_NONE 0x0
-#define MEASUREMENT_MODE_PEAK_RMS 0x1
-
-#define MEASUREMENT_IDX_PEAK 0
-#define MEASUREMENT_IDX_RMS 1
-#define MEASUREMENT_COUNT 2
-
-/* enumerated parameters for Visualizer effect */
-typedef enum {
- VISUALIZER_PARAM_CAPTURE_SIZE, // Sets the number PCM samples in the capture.
- VISUALIZER_PARAM_SCALING_MODE, // Sets the way the captured data is scaled
- VISUALIZER_PARAM_LATENCY, // Informs the visualizer about the downstream latency
- VISUALIZER_PARAM_MEASUREMENT_MODE, // Sets which measurements are to be made
-} t_visualizer_params;
-
-/* commands */
-typedef enum {
- VISUALIZER_CMD_CAPTURE = EFFECT_CMD_FIRST_PROPRIETARY, // Gets the latest PCM capture.
- VISUALIZER_CMD_MEASURE, // Gets the current measurements
-} t_visualizer_cmds;
-
-// VISUALIZER_CMD_CAPTURE retrieves the latest PCM snapshot captured by the visualizer engine.
-// It returns the number of samples specified by VISUALIZER_PARAM_CAPTURE_SIZE
-// in 8 bit unsigned format (0 = 0x80)
-
-// VISUALIZER_CMD_MEASURE retrieves the lastest measurements as int32_t saved in the
-// MEASUREMENT_IDX_* array index order.
-
-#if __cplusplus
-} // extern "C"
-#endif
-
-#endif /*ANDROID_EFFECT_VISUALIZER_CORE_H_*/
diff --git a/soundtrigger/2.0/default/Android.bp b/soundtrigger/2.0/default/Android.bp
index 21e50e1..cc20f91 100644
--- a/soundtrigger/2.0/default/Android.bp
+++ b/soundtrigger/2.0/default/Android.bp
@@ -16,7 +16,10 @@
cc_library_shared {
name: "android.hardware.soundtrigger@2.0-core",
defaults: ["hidl_defaults"],
- vendor: true,
+ vendor_available: true,
+ vndk: {
+ enabled: true,
+ },
srcs: [
"SoundTriggerHalImpl.cpp",
],
@@ -34,7 +37,7 @@
],
header_libs: [
- "android.hardware.soundtrigger.legacy@2.0",
+ "libaudio_system_headers",
"libhardware_headers",
],
}
diff --git a/soundtrigger/2.0/default/Android.mk b/soundtrigger/2.0/default/Android.mk
index 1b6360b..835a020 100644
--- a/soundtrigger/2.0/default/Android.mk
+++ b/soundtrigger/2.0/default/Android.mk
@@ -32,7 +32,6 @@
android.hardware.soundtrigger@2.0-core
LOCAL_C_INCLUDE_DIRS := $(LOCAL_PATH)
-LOCAL_HEADER_LIBRARIES += android.hardware.soundtrigger.legacy@2.0
ifeq ($(strip $(AUDIOSERVER_MULTILIB)),)
LOCAL_MULTILIB := 32
diff --git a/soundtrigger/2.0/legacy/Android.bp b/soundtrigger/2.0/legacy/Android.bp
deleted file mode 100644
index 9954779..0000000
--- a/soundtrigger/2.0/legacy/Android.bp
+++ /dev/null
@@ -1,11 +0,0 @@
-cc_library_headers {
- name: "android.hardware.soundtrigger.legacy@2.0",
- vendor: true,
- export_include_dirs: ["include"],
- header_libs: [
- "android.hardware.audio.common.legacy@2.0",
- ],
- export_header_lib_headers: [
- "android.hardware.audio.common.legacy@2.0",
- ],
-}
diff --git a/soundtrigger/2.0/legacy/OWNERS b/soundtrigger/2.0/legacy/OWNERS
deleted file mode 100644
index 6fdc97c..0000000
--- a/soundtrigger/2.0/legacy/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-elaurent@google.com
-krocard@google.com
-mnaganov@google.com
diff --git a/soundtrigger/2.0/legacy/include/hardware/sound_trigger.h b/soundtrigger/2.0/legacy/include/hardware/sound_trigger.h
deleted file mode 100644
index 57b405e..0000000
--- a/soundtrigger/2.0/legacy/include/hardware/sound_trigger.h
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- * Copyright (C) 2014 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.
- */
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/sound_trigger.h>
-
-#ifndef ANDROID_SOUND_TRIGGER_HAL_H
-#define ANDROID_SOUND_TRIGGER_HAL_H
-
-__BEGIN_DECLS
-
-/**
- * The id of this module
- */
-#define SOUND_TRIGGER_HARDWARE_MODULE_ID "sound_trigger"
-
-/**
- * Name of the audio devices to open
- */
-#define SOUND_TRIGGER_HARDWARE_INTERFACE "sound_trigger_hw_if"
-
-#define SOUND_TRIGGER_MODULE_API_VERSION_1_0 HARDWARE_MODULE_API_VERSION(1, 0)
-#define SOUND_TRIGGER_MODULE_API_VERSION_CURRENT SOUND_TRIGGER_MODULE_API_VERSION_1_0
-
-#define SOUND_TRIGGER_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0)
-#define SOUND_TRIGGER_DEVICE_API_VERSION_1_1 HARDWARE_DEVICE_API_VERSION(1, 1)
-#define SOUND_TRIGGER_DEVICE_API_VERSION_CURRENT SOUND_TRIGGER_DEVICE_API_VERSION_1_1
-
-/**
- * List of known sound trigger HAL modules. This is the base name of the sound_trigger HAL
- * library composed of the "sound_trigger." prefix, one of the base names below and
- * a suffix specific to the device.
- * e.g: sondtrigger.primary.goldfish.so or sound_trigger.primary.default.so
- */
-
-#define SOUND_TRIGGER_HARDWARE_MODULE_ID_PRIMARY "primary"
-
-/**
- * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
- * and the fields of this data structure must begin with hw_module_t
- * followed by module specific information.
- */
-struct sound_trigger_module {
- struct hw_module_t common;
-};
-
-typedef void (*recognition_callback_t)(struct sound_trigger_recognition_event* event, void* cookie);
-typedef void (*sound_model_callback_t)(struct sound_trigger_model_event* event, void* cookie);
-
-struct sound_trigger_hw_device {
- struct hw_device_t common;
-
- /*
- * Retrieve implementation properties.
- */
- int (*get_properties)(const struct sound_trigger_hw_device* dev,
- struct sound_trigger_properties* properties);
-
- /*
- * Load a sound model. Once loaded, recognition of this model can be started and stopped.
- * Only one active recognition per model at a time. The SoundTrigger service will handle
- * concurrent recognition requests by different users/applications on the same model.
- * The implementation returns a unique handle used by other functions (unload_sound_model(),
- * start_recognition(), etc...
- */
- int (*load_sound_model)(const struct sound_trigger_hw_device* dev,
- struct sound_trigger_sound_model* sound_model,
- sound_model_callback_t callback, void* cookie,
- sound_model_handle_t* handle);
-
- /*
- * Unload a sound model. A sound model can be unloaded to make room for a new one to overcome
- * implementation limitations.
- */
- int (*unload_sound_model)(const struct sound_trigger_hw_device* dev,
- sound_model_handle_t handle);
-
- /* Start recognition on a given model. Only one recognition active at a time per model.
- * Once recognition succeeds of fails, the callback is called.
- * TODO: group recognition configuration parameters into one struct and add key phrase options.
- */
- int (*start_recognition)(const struct sound_trigger_hw_device* dev,
- sound_model_handle_t sound_model_handle,
- const struct sound_trigger_recognition_config* config,
- recognition_callback_t callback, void* cookie);
-
- /* Stop recognition on a given model.
- * The implementation does not have to call the callback when stopped via this method.
- */
- int (*stop_recognition)(const struct sound_trigger_hw_device* dev,
- sound_model_handle_t sound_model_handle);
-
- /* Stop recognition on all models.
- * Only supported for device api versions SOUND_TRIGGER_DEVICE_API_VERSION_1_1 or above.
- * If no implementation is provided, stop_recognition will be called for each running model.
- */
- int (*stop_all_recognitions)(const struct sound_trigger_hw_device* dev);
-};
-
-typedef struct sound_trigger_hw_device sound_trigger_hw_device_t;
-
-/** convenience API for opening and closing a supported device */
-
-static inline int sound_trigger_hw_device_open(const struct hw_module_t* module,
- struct sound_trigger_hw_device** device) {
- return module->methods->open(module, SOUND_TRIGGER_HARDWARE_INTERFACE,
- TO_HW_DEVICE_T_OPEN(device));
-}
-
-static inline int sound_trigger_hw_device_close(struct sound_trigger_hw_device* device) {
- return device->common.close(&device->common);
-}
-
-__END_DECLS
-
-#endif // ANDROID_SOUND_TRIGGER_HAL_H
diff --git a/soundtrigger/2.0/legacy/include/system/sound_trigger.h b/soundtrigger/2.0/legacy/include/system/sound_trigger.h
deleted file mode 100644
index 5d00c12..0000000
--- a/soundtrigger/2.0/legacy/include/system/sound_trigger.h
+++ /dev/null
@@ -1,228 +0,0 @@
-/*
- * Copyright (C) 2014 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.
- */
-
-#ifndef ANDROID_SOUND_TRIGGER_H
-#define ANDROID_SOUND_TRIGGER_H
-
-#include <stdbool.h>
-#include <system/audio.h>
-
-#define SOUND_TRIGGER_MAX_STRING_LEN 64 // max length of strings in properties & descriptor structs
-#define SOUND_TRIGGER_MAX_LOCALE_LEN 6 // max length of locale string. e.g en_US
-#define SOUND_TRIGGER_MAX_USERS 10 // max number of concurrent users
-#define SOUND_TRIGGER_MAX_PHRASES 10 // max number of concurrent phrases
-
-typedef enum {
- SOUND_TRIGGER_STATE_NO_INIT = -1, /* The sound trigger service is not initialized */
- SOUND_TRIGGER_STATE_ENABLED = 0, /* The sound trigger service is enabled */
- SOUND_TRIGGER_STATE_DISABLED = 1 /* The sound trigger service is disabled */
-} sound_trigger_service_state_t;
-
-#define RECOGNITION_MODE_VOICE_TRIGGER 0x1 // simple voice trigger
-#define RECOGNITION_MODE_USER_IDENTIFICATION 0x2 // trigger only if one user in model identified
-#define RECOGNITION_MODE_USER_AUTHENTICATION 0x4 // trigger only if one user in mode authenticated
-#define RECOGNITION_MODE_GENERIC_TRIGGER 0x8 // generic sound trigger
-
-#define RECOGNITION_STATUS_SUCCESS 0
-#define RECOGNITION_STATUS_ABORT 1
-#define RECOGNITION_STATUS_FAILURE 2
-
-#define SOUND_MODEL_STATUS_UPDATED 0
-
-typedef enum {
- SOUND_MODEL_TYPE_UNKNOWN = -1, /* use for unspecified sound model type */
- SOUND_MODEL_TYPE_KEYPHRASE = 0, /* use for key phrase sound models */
- SOUND_MODEL_TYPE_GENERIC = 1 /* use for all models other than keyphrase */
-} sound_trigger_sound_model_type_t;
-
-typedef audio_uuid_t sound_trigger_uuid_t;
-
-/*
- * sound trigger implementation descriptor read by the framework via get_properties().
- * Used by SoundTrigger service to report to applications and manage concurrency and policy.
- */
-struct sound_trigger_properties {
- char implementor[SOUND_TRIGGER_MAX_STRING_LEN]; /* implementor name */
- char description[SOUND_TRIGGER_MAX_STRING_LEN]; /* implementation description */
- unsigned int version; /* implementation version */
- sound_trigger_uuid_t uuid; /* unique implementation ID.
- Must change with version each version */
- unsigned int max_sound_models; /* maximum number of concurrent sound models
- loaded */
- unsigned int max_key_phrases; /* maximum number of key phrases */
- unsigned int max_users; /* maximum number of concurrent users detected */
- unsigned int recognition_modes; /* all supported modes.
- e.g RECOGNITION_MODE_VOICE_TRIGGER */
- bool capture_transition; /* supports seamless transition from detection
- to capture */
- unsigned int max_buffer_ms; /* maximum buffering capacity in ms if
- capture_transition is true*/
- bool concurrent_capture; /* supports capture by other use cases while
- detection is active */
- bool trigger_in_event; /* returns the trigger capture in event */
- unsigned int power_consumption_mw; /* Rated power consumption when detection is active
- with TDB silence/sound/speech ratio */
-};
-
-typedef int sound_trigger_module_handle_t;
-
-struct sound_trigger_module_descriptor {
- sound_trigger_module_handle_t handle;
- struct sound_trigger_properties properties;
-};
-
-typedef int sound_model_handle_t;
-
-/*
- * Base sound model descriptor. This struct is the header of a larger block passed to
- * load_sound_model() and containing the binary data of the sound model.
- * Proprietary representation of users in binary data must match information indicated
- * by users field
- */
-struct sound_trigger_sound_model {
- sound_trigger_sound_model_type_t type; /* model type. e.g. SOUND_MODEL_TYPE_KEYPHRASE */
- sound_trigger_uuid_t uuid; /* unique sound model ID. */
- sound_trigger_uuid_t vendor_uuid; /* unique vendor ID. Identifies the engine the
- sound model was build for */
- unsigned int data_size; /* size of opaque model data */
- unsigned int data_offset; /* offset of opaque data start from head of struct
- (e.g sizeof struct sound_trigger_sound_model) */
-};
-
-/* key phrase descriptor */
-struct sound_trigger_phrase {
- unsigned int id; /* keyphrase ID */
- unsigned int recognition_mode; /* recognition modes supported by this key phrase */
- unsigned int num_users; /* number of users in the key phrase */
- unsigned int users[SOUND_TRIGGER_MAX_USERS]; /* users ids: (not uid_t but sound trigger
- specific IDs */
- char locale[SOUND_TRIGGER_MAX_LOCALE_LEN]; /* locale - JAVA Locale style (e.g. en_US) */
- char text[SOUND_TRIGGER_MAX_STRING_LEN]; /* phrase text in UTF-8 format. */
-};
-
-/*
- * Specialized sound model for key phrase detection.
- * Proprietary representation of key phrases in binary data must match information indicated
- * by phrases field
- */
-struct sound_trigger_phrase_sound_model {
- struct sound_trigger_sound_model common;
- unsigned int num_phrases; /* number of key phrases in model */
- struct sound_trigger_phrase phrases[SOUND_TRIGGER_MAX_PHRASES];
-};
-
-/*
- * Generic sound model, used for all cases except key phrase detection.
- */
-struct sound_trigger_generic_sound_model {
- struct sound_trigger_sound_model common;
-};
-
-/*
- * Generic recognition event sent via recognition callback
- * Must be aligned to transmit as raw memory through Binder.
- */
-struct __attribute__((aligned(8))) sound_trigger_recognition_event {
- int status; /* recognition status e.g.
- RECOGNITION_STATUS_SUCCESS */
- sound_trigger_sound_model_type_t type; /* event type, same as sound model type.
- e.g. SOUND_MODEL_TYPE_KEYPHRASE */
- sound_model_handle_t model; /* loaded sound model that triggered the
- event */
- bool capture_available; /* it is possible to capture audio from this
- utterance buffered by the
- implementation */
- int capture_session; /* audio session ID. framework use */
- int capture_delay_ms; /* delay in ms between end of model
- detection and start of audio available
- for capture. A negative value is possible
- (e.g. if key phrase is also available for
- capture */
- int capture_preamble_ms; /* duration in ms of audio captured
- before the start of the trigger.
- 0 if none. */
- bool trigger_in_data; /* the opaque data is the capture of
- the trigger sound */
- audio_config_t audio_config; /* audio format of either the trigger in
- event data or to use for capture of the
- rest of the utterance */
- unsigned int data_size; /* size of opaque event data */
- unsigned int data_offset; /* offset of opaque data start from start of
- this struct (e.g sizeof struct
- sound_trigger_phrase_recognition_event) */
-};
-
-/*
- * Confidence level for each user in struct sound_trigger_phrase_recognition_extra
- */
-struct sound_trigger_confidence_level {
- unsigned int user_id; /* user ID */
- unsigned int level; /* confidence level in percent (0 - 100).
- - min level for recognition configuration
- - detected level for recognition event */
-};
-
-/*
- * Specialized recognition event for key phrase detection
- */
-struct sound_trigger_phrase_recognition_extra {
- unsigned int id; /* keyphrase ID */
- unsigned int recognition_modes; /* recognition modes used for this keyphrase */
- unsigned int confidence_level; /* confidence level for mode RECOGNITION_MODE_VOICE_TRIGGER */
- unsigned int num_levels; /* number of user confidence levels */
- struct sound_trigger_confidence_level levels[SOUND_TRIGGER_MAX_USERS];
-};
-
-struct sound_trigger_phrase_recognition_event {
- struct sound_trigger_recognition_event common;
- unsigned int num_phrases;
- struct sound_trigger_phrase_recognition_extra phrase_extras[SOUND_TRIGGER_MAX_PHRASES];
-};
-
-struct sound_trigger_generic_recognition_event {
- struct sound_trigger_recognition_event common;
-};
-
-/*
- * configuration for sound trigger capture session passed to start_recognition()
- */
-struct sound_trigger_recognition_config {
- audio_io_handle_t capture_handle; /* IO handle that will be used for capture.
- N/A if capture_requested is false */
- audio_devices_t capture_device; /* input device requested for detection capture */
- bool capture_requested; /* capture and buffer audio for this recognition
- instance */
- unsigned int num_phrases; /* number of key phrases recognition extras */
- struct sound_trigger_phrase_recognition_extra phrases[SOUND_TRIGGER_MAX_PHRASES];
- /* configuration for each key phrase */
- unsigned int data_size; /* size of opaque capture configuration data */
- unsigned int data_offset; /* offset of opaque data start from start of this struct
- (e.g sizeof struct sound_trigger_recognition_config) */
-};
-
-/*
- * Event sent via load sound model callback
- */
-struct sound_trigger_model_event {
- int status; /* sound model status e.g. SOUND_MODEL_STATUS_UPDATED */
- sound_model_handle_t model; /* loaded sound model that triggered the event */
- unsigned int data_size; /* size of event data if any. Size of updated sound model if
- status is SOUND_MODEL_STATUS_UPDATED */
- unsigned int data_offset; /* offset of data start from start of this struct
- (e.g sizeof struct sound_trigger_model_event) */
-};
-
-#endif // ANDROID_SOUND_TRIGGER_H
diff --git a/soundtrigger/2.1/default/Android.mk b/soundtrigger/2.1/default/Android.mk
index 04d3f36..5851d63 100644
--- a/soundtrigger/2.1/default/Android.mk
+++ b/soundtrigger/2.1/default/Android.mk
@@ -38,8 +38,6 @@
android.hidl.allocator@1.0 \
android.hidl.memory@1.0
-LOCAL_HEADER_LIBRARIES := android.hardware.soundtrigger.legacy@2.0
-
LOCAL_C_INCLUDE_DIRS := $(LOCAL_PATH)
ifeq ($(strip $(AUDIOSERVER_MULTILIB)),)
diff --git a/tv/input/1.0/default/Android.bp b/tv/input/1.0/default/Android.bp
index c422230..7c140a5 100644
--- a/tv/input/1.0/default/Android.bp
+++ b/tv/input/1.0/default/Android.bp
@@ -16,9 +16,6 @@
"android.hardware.tv.input@1.0",
],
- header_libs: [
- "android.hardware.audio.common.legacy@2.0",
- ],
}
cc_binary {