Merge changes from topic "audio-hal-v4-tmp"

* changes:
  Audio V4: Split system and vendor Audio.h
  Audio V4: Use the framework audio.h instead of the HAL's
gugelfrei
Kevin Rocard 7 years ago committed by Android (Google) Code Review
commit 6a784e71ca

@ -6,7 +6,7 @@ cc_test {
static_libs: ["libsndfile"],
shared_libs: [
"libaaudio",
"libaudioutils",
"libaudioutils_system",
],
header_libs: ["libaaudio_example_utils"],
}

@ -11,6 +11,6 @@ LOCAL_C_INCLUDES := \
LOCAL_SRC_FILES:= ../src/loopback.cpp
LOCAL_CFLAGS := -Wall -Werror
LOCAL_STATIC_LIBRARIES := libsndfile
LOCAL_SHARED_LIBRARIES := libaaudio libaudioutils
LOCAL_SHARED_LIBRARIES := libaaudio libaudioutils_system
LOCAL_MODULE := aaudio_loopback
include $(BUILD_EXECUTABLE)

@ -17,7 +17,7 @@
#define LOG_TAG "AAudioStreamParameters"
#include <utils/Log.h>
#include <hardware/audio.h>
#include <system/audio.h>
#include "AAudioStreamParameters.h"

@ -23,7 +23,7 @@
#include <sys/types.h>
#include <utils/Errors.h>
#include <hardware/audio.h>
#include <system/audio.h>
#include "aaudio/AAudio.h"

@ -45,7 +45,7 @@ cc_library_shared {
"libutils",
"libbinder",
"libdl",
"libaudioutils",
"libaudioutils_system",
"libaudiomanager",
"libmedia_helper",
"libmediametrics",

@ -6,7 +6,6 @@ cc_library_shared {
"DevicesFactoryHalHybrid.cpp",
"DevicesFactoryHalLocal.cpp",
"StreamHalLocal.cpp",
"ConversionHelperHidl.cpp",
"DeviceHalHidl.cpp",
"DevicesFactoryHalHidl.cpp",
@ -14,6 +13,7 @@ cc_library_shared {
"EffectHalHidl.cpp",
"EffectsFactoryHalHidl.cpp",
"StreamHalHidl.cpp",
"HidlUtils.cpp",
],
export_include_dirs: ["."],
@ -24,7 +24,7 @@ cc_library_shared {
],
shared_libs: [
"libaudiohal_deathhandler",
"libaudioutils",
"libaudioutils_system",
"libcutils",
"liblog",
"libutils",
@ -37,7 +37,6 @@ cc_library_shared {
"libhidltransport",
"android.hardware.audio@2.0",
"android.hardware.audio.common@2.0",
"android.hardware.audio.common@2.0-util",
"android.hardware.audio.effect@2.0",
"android.hidl.allocator@1.0",
"android.hidl.memory@1.0",
@ -45,7 +44,8 @@ cc_library_shared {
"libmediautils",
],
header_libs: [
"libaudiohal_headers"
"libaudiohal_headers",
"libaudiohal_util",
],
export_shared_lib_headers: [

@ -18,7 +18,6 @@
//#define LOG_NDEBUG 0
#include <hwbinder/IPCThreadState.h>
#include <media/EffectsFactoryApi.h>
#include <utils/Log.h>
#include "ConversionHelperHidl.h"

@ -0,0 +1,21 @@
/*
* 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.
*/
#include "HidlUtils.h"
#define AUDIO_HAL_VERSION V2_0
#include <util/HidlUtils.impl.h>
#undef AUDIO_HAL_VERSION

@ -0,0 +1,26 @@
/*
* 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_hardware_audio_V2_0_Hidl_Utils_H_
#define android_hardware_audio_V2_0_Hidl_Utils_H_
#include <android/hardware/audio/common/2.0/types.h>
#define AUDIO_HAL_VERSION V2_0
#include <util/HidlUtils.h>
#undef AUDIO_HAL_VERSION
#endif // android_hardware_audio_V2_0_Hidl_Utils_H_

@ -19,7 +19,7 @@ cc_library_shared {
],
header_libs: [
"libaudiohal_headers"
"libaudiohal_headers",
]
}
@ -50,4 +50,11 @@ cc_library_headers {
name: "libaudiohal_headers",
export_include_dirs: ["include"],
header_libs: [
"libaudiohal_legacy_headers",
],
export_header_lib_headers: [
"libaudiohal_legacy_headers",
],
}

@ -0,0 +1,12 @@
cc_library_headers {
name: "libaudiohal_legacy_headers",
export_include_dirs: ["include"],
header_libs: [
"libaudio_system_headers",
],
export_header_lib_headers: [
"libaudio_system_headers",
],
}

@ -0,0 +1,746 @@
/*
* 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.
*/
// TODO: Remove parts unused by the system
#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/hardware.h>
#include <system/audio.h>
#include <hardware/audio_effect.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

@ -0,0 +1,310 @@
/*
* 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

@ -0,0 +1,13 @@
cc_library_headers {
name: "libaudiohal_util",
export_include_dirs: ["include"],
header_libs: [
"libaudio_system_headers",
],
export_header_lib_headers: [
"libaudio_system_headers",
],
}

@ -0,0 +1,73 @@
/*
* 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 AUDIO_HAL_VERSION
#error "AUDIO_HAL_VERSION must be set before including this file."
#endif
#include <memory>
#include <system/audio.h>
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioConfig;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioGain;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioGainConfig;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioOffloadInfo;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioPort;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioPortConfig;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::Uuid;
using ::android::hardware::hidl_vec;
namespace android {
namespace hardware {
namespace audio {
namespace common {
namespace AUDIO_HAL_VERSION {
class HidlUtils {
public:
static void audioConfigFromHal(const audio_config_t& halConfig, AudioConfig* config);
static void audioConfigToHal(const AudioConfig& config, audio_config_t* halConfig);
static void audioGainConfigFromHal(const struct audio_gain_config& halConfig,
AudioGainConfig* config);
static void audioGainConfigToHal(const AudioGainConfig& config,
struct audio_gain_config* halConfig);
static void audioGainFromHal(const struct audio_gain& halGain, AudioGain* gain);
static void audioGainToHal(const AudioGain& gain, struct audio_gain* halGain);
static void audioOffloadInfoFromHal(const audio_offload_info_t& halOffload,
AudioOffloadInfo* offload);
static void audioOffloadInfoToHal(const AudioOffloadInfo& offload,
audio_offload_info_t* halOffload);
static void audioPortConfigFromHal(const struct audio_port_config& halConfig,
AudioPortConfig* config);
static void audioPortConfigToHal(const AudioPortConfig& config,
struct audio_port_config* halConfig);
static void audioPortConfigsFromHal(unsigned int numHalConfigs,
const struct audio_port_config* halConfigs,
hidl_vec<AudioPortConfig>* configs);
static std::unique_ptr<audio_port_config[]> audioPortConfigsToHal(
const hidl_vec<AudioPortConfig>& configs);
static void audioPortFromHal(const struct audio_port& halPort, AudioPort* port);
static void audioPortToHal(const AudioPort& port, struct audio_port* halPort);
static void uuidFromHal(const audio_uuid_t& halUuid, Uuid* uuid);
static void uuidToHal(const Uuid& uuid, audio_uuid_t* halUuid);
};
} // namespace AUDIO_HAL_VERSION
} // namespace common
} // namespace audio
} // namespace hardware
} // namespace android

@ -0,0 +1,346 @@
/*
* 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 AUDIO_HAL_VERSION
#error "AUDIO_HAL_VERSION must be set before including this file."
#endif
#include <string.h>
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioChannelMask;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioDevice;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioFormat;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioGainMode;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioMixLatencyClass;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioPortConfigMask;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioPortRole;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioPortType;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioSource;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioStreamType;
using ::android::hardware::audio::common::AUDIO_HAL_VERSION::AudioUsage;
namespace android {
namespace hardware {
namespace audio {
namespace common {
namespace AUDIO_HAL_VERSION {
void HidlUtils::audioConfigFromHal(const audio_config_t& halConfig, AudioConfig* config) {
config->sampleRateHz = halConfig.sample_rate;
config->channelMask = AudioChannelMask(halConfig.channel_mask);
config->format = AudioFormat(halConfig.format);
audioOffloadInfoFromHal(halConfig.offload_info, &config->offloadInfo);
config->frameCount = halConfig.frame_count;
}
void HidlUtils::audioConfigToHal(const AudioConfig& config, audio_config_t* halConfig) {
memset(halConfig, 0, sizeof(audio_config_t));
halConfig->sample_rate = config.sampleRateHz;
halConfig->channel_mask = static_cast<audio_channel_mask_t>(config.channelMask);
halConfig->format = static_cast<audio_format_t>(config.format);
audioOffloadInfoToHal(config.offloadInfo, &halConfig->offload_info);
halConfig->frame_count = config.frameCount;
}
void HidlUtils::audioGainConfigFromHal(const struct audio_gain_config& halConfig,
AudioGainConfig* config) {
config->index = halConfig.index;
config->mode = AudioGainMode(halConfig.mode);
config->channelMask = AudioChannelMask(halConfig.channel_mask);
for (size_t i = 0; i < sizeof(audio_channel_mask_t) * 8; ++i) {
config->values[i] = halConfig.values[i];
}
config->rampDurationMs = halConfig.ramp_duration_ms;
}
void HidlUtils::audioGainConfigToHal(const AudioGainConfig& config,
struct audio_gain_config* halConfig) {
halConfig->index = config.index;
halConfig->mode = static_cast<audio_gain_mode_t>(config.mode);
halConfig->channel_mask = static_cast<audio_channel_mask_t>(config.channelMask);
memset(halConfig->values, 0, sizeof(halConfig->values));
for (size_t i = 0; i < sizeof(audio_channel_mask_t) * 8; ++i) {
halConfig->values[i] = config.values[i];
}
halConfig->ramp_duration_ms = config.rampDurationMs;
}
void HidlUtils::audioGainFromHal(const struct audio_gain& halGain, AudioGain* gain) {
gain->mode = AudioGainMode(halGain.mode);
gain->channelMask = AudioChannelMask(halGain.channel_mask);
gain->minValue = halGain.min_value;
gain->maxValue = halGain.max_value;
gain->defaultValue = halGain.default_value;
gain->stepValue = halGain.step_value;
gain->minRampMs = halGain.min_ramp_ms;
gain->maxRampMs = halGain.max_ramp_ms;
}
void HidlUtils::audioGainToHal(const AudioGain& gain, struct audio_gain* halGain) {
halGain->mode = static_cast<audio_gain_mode_t>(gain.mode);
halGain->channel_mask = static_cast<audio_channel_mask_t>(gain.channelMask);
halGain->min_value = gain.minValue;
halGain->max_value = gain.maxValue;
halGain->default_value = gain.defaultValue;
halGain->step_value = gain.stepValue;
halGain->min_ramp_ms = gain.minRampMs;
halGain->max_ramp_ms = gain.maxRampMs;
}
void HidlUtils::audioOffloadInfoFromHal(const audio_offload_info_t& halOffload,
AudioOffloadInfo* offload) {
offload->sampleRateHz = halOffload.sample_rate;
offload->channelMask = AudioChannelMask(halOffload.channel_mask);
offload->format = AudioFormat(halOffload.format);
offload->streamType = AudioStreamType(halOffload.stream_type);
offload->bitRatePerSecond = halOffload.bit_rate;
offload->durationMicroseconds = halOffload.duration_us;
offload->hasVideo = halOffload.has_video;
offload->isStreaming = halOffload.is_streaming;
offload->bitWidth = halOffload.bit_width;
offload->bufferSize = halOffload.offload_buffer_size;
offload->usage = static_cast<AudioUsage>(halOffload.usage);
}
void HidlUtils::audioOffloadInfoToHal(const AudioOffloadInfo& offload,
audio_offload_info_t* halOffload) {
*halOffload = AUDIO_INFO_INITIALIZER;
halOffload->sample_rate = offload.sampleRateHz;
halOffload->channel_mask = static_cast<audio_channel_mask_t>(offload.channelMask);
halOffload->format = static_cast<audio_format_t>(offload.format);
halOffload->stream_type = static_cast<audio_stream_type_t>(offload.streamType);
halOffload->bit_rate = offload.bitRatePerSecond;
halOffload->duration_us = offload.durationMicroseconds;
halOffload->has_video = offload.hasVideo;
halOffload->is_streaming = offload.isStreaming;
halOffload->bit_width = offload.bitWidth;
halOffload->offload_buffer_size = offload.bufferSize;
halOffload->usage = static_cast<audio_usage_t>(offload.usage);
}
void HidlUtils::audioPortConfigFromHal(const struct audio_port_config& halConfig,
AudioPortConfig* config) {
config->id = halConfig.id;
config->role = AudioPortRole(halConfig.role);
config->type = AudioPortType(halConfig.type);
config->configMask = AudioPortConfigMask(halConfig.config_mask);
config->sampleRateHz = halConfig.sample_rate;
config->channelMask = AudioChannelMask(halConfig.channel_mask);
config->format = AudioFormat(halConfig.format);
audioGainConfigFromHal(halConfig.gain, &config->gain);
switch (halConfig.type) {
case AUDIO_PORT_TYPE_NONE:
break;
case AUDIO_PORT_TYPE_DEVICE: {
config->ext.device.hwModule = halConfig.ext.device.hw_module;
config->ext.device.type = AudioDevice(halConfig.ext.device.type);
memcpy(config->ext.device.address.data(), halConfig.ext.device.address,
AUDIO_DEVICE_MAX_ADDRESS_LEN);
break;
}
case AUDIO_PORT_TYPE_MIX: {
config->ext.mix.hwModule = halConfig.ext.mix.hw_module;
config->ext.mix.ioHandle = halConfig.ext.mix.handle;
if (halConfig.role == AUDIO_PORT_ROLE_SOURCE) {
config->ext.mix.useCase.source = AudioSource(halConfig.ext.mix.usecase.source);
} else if (halConfig.role == AUDIO_PORT_ROLE_SINK) {
config->ext.mix.useCase.stream = AudioStreamType(halConfig.ext.mix.usecase.stream);
}
break;
}
case AUDIO_PORT_TYPE_SESSION: {
config->ext.session.session = halConfig.ext.session.session;
break;
}
}
}
void HidlUtils::audioPortConfigToHal(const AudioPortConfig& config,
struct audio_port_config* halConfig) {
memset(halConfig, 0, sizeof(audio_port_config));
halConfig->id = config.id;
halConfig->role = static_cast<audio_port_role_t>(config.role);
halConfig->type = static_cast<audio_port_type_t>(config.type);
halConfig->config_mask = static_cast<unsigned int>(config.configMask);
halConfig->sample_rate = config.sampleRateHz;
halConfig->channel_mask = static_cast<audio_channel_mask_t>(config.channelMask);
halConfig->format = static_cast<audio_format_t>(config.format);
audioGainConfigToHal(config.gain, &halConfig->gain);
switch (config.type) {
case AudioPortType::NONE:
break;
case AudioPortType::DEVICE: {
halConfig->ext.device.hw_module = config.ext.device.hwModule;
halConfig->ext.device.type = static_cast<audio_devices_t>(config.ext.device.type);
memcpy(halConfig->ext.device.address, config.ext.device.address.data(),
AUDIO_DEVICE_MAX_ADDRESS_LEN);
break;
}
case AudioPortType::MIX: {
halConfig->ext.mix.hw_module = config.ext.mix.hwModule;
halConfig->ext.mix.handle = config.ext.mix.ioHandle;
if (config.role == AudioPortRole::SOURCE) {
halConfig->ext.mix.usecase.source =
static_cast<audio_source_t>(config.ext.mix.useCase.source);
} else if (config.role == AudioPortRole::SINK) {
halConfig->ext.mix.usecase.stream =
static_cast<audio_stream_type_t>(config.ext.mix.useCase.stream);
}
break;
}
case AudioPortType::SESSION: {
halConfig->ext.session.session =
static_cast<audio_session_t>(config.ext.session.session);
break;
}
}
}
void HidlUtils::audioPortConfigsFromHal(unsigned int numHalConfigs,
const struct audio_port_config* halConfigs,
hidl_vec<AudioPortConfig>* configs) {
configs->resize(numHalConfigs);
for (unsigned int i = 0; i < numHalConfigs; ++i) {
audioPortConfigFromHal(halConfigs[i], &(*configs)[i]);
}
}
std::unique_ptr<audio_port_config[]> HidlUtils::audioPortConfigsToHal(
const hidl_vec<AudioPortConfig>& configs) {
std::unique_ptr<audio_port_config[]> halConfigs(new audio_port_config[configs.size()]);
for (size_t i = 0; i < configs.size(); ++i) {
audioPortConfigToHal(configs[i], &halConfigs[i]);
}
return halConfigs;
}
void HidlUtils::audioPortFromHal(const struct audio_port& halPort, AudioPort* port) {
port->id = halPort.id;
port->role = AudioPortRole(halPort.role);
port->type = AudioPortType(halPort.type);
port->name.setToExternal(halPort.name, strlen(halPort.name));
port->sampleRates.resize(halPort.num_sample_rates);
for (size_t i = 0; i < halPort.num_sample_rates; ++i) {
port->sampleRates[i] = halPort.sample_rates[i];
}
port->channelMasks.resize(halPort.num_channel_masks);
for (size_t i = 0; i < halPort.num_channel_masks; ++i) {
port->channelMasks[i] = AudioChannelMask(halPort.channel_masks[i]);
}
port->formats.resize(halPort.num_formats);
for (size_t i = 0; i < halPort.num_formats; ++i) {
port->formats[i] = AudioFormat(halPort.formats[i]);
}
port->gains.resize(halPort.num_gains);
for (size_t i = 0; i < halPort.num_gains; ++i) {
audioGainFromHal(halPort.gains[i], &port->gains[i]);
}
audioPortConfigFromHal(halPort.active_config, &port->activeConfig);
switch (halPort.type) {
case AUDIO_PORT_TYPE_NONE:
break;
case AUDIO_PORT_TYPE_DEVICE: {
port->ext.device.hwModule = halPort.ext.device.hw_module;
port->ext.device.type = AudioDevice(halPort.ext.device.type);
memcpy(port->ext.device.address.data(), halPort.ext.device.address,
AUDIO_DEVICE_MAX_ADDRESS_LEN);
break;
}
case AUDIO_PORT_TYPE_MIX: {
port->ext.mix.hwModule = halPort.ext.mix.hw_module;
port->ext.mix.ioHandle = halPort.ext.mix.handle;
port->ext.mix.latencyClass = AudioMixLatencyClass(halPort.ext.mix.latency_class);
break;
}
case AUDIO_PORT_TYPE_SESSION: {
port->ext.session.session = halPort.ext.session.session;
break;
}
}
}
void HidlUtils::audioPortToHal(const AudioPort& port, struct audio_port* halPort) {
memset(halPort, 0, sizeof(audio_port));
halPort->id = port.id;
halPort->role = static_cast<audio_port_role_t>(port.role);
halPort->type = static_cast<audio_port_type_t>(port.type);
memcpy(halPort->name, port.name.c_str(),
std::min(port.name.size(), static_cast<size_t>(AUDIO_PORT_MAX_NAME_LEN)));
halPort->num_sample_rates =
std::min(port.sampleRates.size(), static_cast<size_t>(AUDIO_PORT_MAX_SAMPLING_RATES));
for (size_t i = 0; i < halPort->num_sample_rates; ++i) {
halPort->sample_rates[i] = port.sampleRates[i];
}
halPort->num_channel_masks =
std::min(port.channelMasks.size(), static_cast<size_t>(AUDIO_PORT_MAX_CHANNEL_MASKS));
for (size_t i = 0; i < halPort->num_channel_masks; ++i) {
halPort->channel_masks[i] = static_cast<audio_channel_mask_t>(port.channelMasks[i]);
}
halPort->num_formats =
std::min(port.formats.size(), static_cast<size_t>(AUDIO_PORT_MAX_FORMATS));
for (size_t i = 0; i < halPort->num_formats; ++i) {
halPort->formats[i] = static_cast<audio_format_t>(port.formats[i]);
}
halPort->num_gains = std::min(port.gains.size(), static_cast<size_t>(AUDIO_PORT_MAX_GAINS));
for (size_t i = 0; i < halPort->num_gains; ++i) {
audioGainToHal(port.gains[i], &halPort->gains[i]);
}
audioPortConfigToHal(port.activeConfig, &halPort->active_config);
switch (port.type) {
case AudioPortType::NONE:
break;
case AudioPortType::DEVICE: {
halPort->ext.device.hw_module = port.ext.device.hwModule;
halPort->ext.device.type = static_cast<audio_devices_t>(port.ext.device.type);
memcpy(halPort->ext.device.address, port.ext.device.address.data(),
AUDIO_DEVICE_MAX_ADDRESS_LEN);
break;
}
case AudioPortType::MIX: {
halPort->ext.mix.hw_module = port.ext.mix.hwModule;
halPort->ext.mix.handle = port.ext.mix.ioHandle;
halPort->ext.mix.latency_class =
static_cast<audio_mix_latency_class_t>(port.ext.mix.latencyClass);
break;
}
case AudioPortType::SESSION: {
halPort->ext.session.session = static_cast<audio_session_t>(port.ext.session.session);
break;
}
}
}
void HidlUtils::uuidFromHal(const audio_uuid_t& halUuid, Uuid* uuid) {
uuid->timeLow = halUuid.timeLow;
uuid->timeMid = halUuid.timeMid;
uuid->versionAndTimeHigh = halUuid.timeHiAndVersion;
uuid->variantAndClockSeqHigh = halUuid.clockSeq;
memcpy(uuid->node.data(), halUuid.node, uuid->node.size());
}
void HidlUtils::uuidToHal(const Uuid& uuid, audio_uuid_t* halUuid) {
halUuid->timeLow = uuid.timeLow;
halUuid->timeMid = uuid.timeMid;
halUuid->timeHiAndVersion = uuid.versionAndTimeHigh;
halUuid->clockSeq = uuid.variantAndClockSeqHigh;
memcpy(halUuid->node, uuid.node.data(), uuid.node.size());
}
} // namespace AUDIO_HAL_VERSION
} // namespace common
} // namespace audio
} // namespace hardware
} // namespace android

@ -20,7 +20,7 @@ LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
LOCAL_SHARED_LIBRARIES := \
libaudiohal \
libaudioutils \
libaudioutils_system \
libcutils \
liblog \
libnbaio \

@ -8,7 +8,7 @@ LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
LOCAL_SHARED_LIBRARIES := \
libaudioutils \
libaudioutils_system \
libaudioprocessing \
libcutils \
liblog \
@ -44,7 +44,7 @@ LOCAL_STATIC_LIBRARIES := \
LOCAL_SHARED_LIBRARIES := \
libaudioprocessing \
libaudioutils \
libaudioutils_system \
libcutils \
liblog \
libutils \
@ -73,7 +73,7 @@ LOCAL_STATIC_LIBRARIES := \
LOCAL_SHARED_LIBRARIES := \
libaudioprocessing \
libaudioutils \
libaudioutils_system \
libcutils \
liblog \
libutils \

@ -1,7 +1,6 @@
// Effect configuration
cc_library_shared {
name: "libeffectsconfig",
vendor_available: true,
srcs: ["src/EffectsConfig.cpp"],
@ -20,3 +19,29 @@ cc_library_shared {
export_include_dirs: ["include"],
}
cc_library_shared {
name: "libeffectsconfig@2.0",
vendor: true,
srcs: ["src/EffectsConfig.cpp"],
cflags: [
"-Wall",
"-Werror",
],
shared_libs: [
"liblog",
"libtinyxml2",
],
export_include_dirs: ["include"],
header_libs: [
"android.hardware.audio.common.legacy@2.0",
],
export_header_lib_headers: [
"android.hardware.audio.common.legacy@2.0",
],
}

@ -17,12 +17,14 @@ LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-effects) \
$(call include-path-for, audio-utils)
#-DBUILD_FLOAT
LOCAL_CFLAGS += -fvisibility=hidden
LOCAL_CFLAGS += -Wall -Werror
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += \
libhardware_headers \
android.hardware.audio.effect.legacy@2.0 \
include $(BUILD_SHARED_LIBRARY)

@ -1,11 +1,3 @@
cc_library_headers {
name: "libeffects_headers",
vendor_available: true,
export_include_dirs: ["include"],
header_libs: ["libhardware_headers"],
export_header_lib_headers: ["libhardware_headers"],
}
// Effect factory library
cc_library_shared {
name: "libeffects",
@ -21,21 +13,20 @@ cc_library_shared {
"libcutils",
"liblog",
"libdl",
"libeffectsconfig",
"libeffectsconfig@2.0",
],
cflags: ["-fvisibility=hidden"],
local_include_dirs:["include/media"],
export_include_dirs: ["."],
header_libs: [
"libaudioeffects",
"libeffects_headers",
"android.hardware.audio.effect.legacy@2.0",
],
export_header_lib_headers: ["libeffects_headers"],
export_header_lib_headers: ["android.hardware.audio.effect.legacy@2.0"],
}
cc_binary {
name: "dumpEffectConfigFile",
name: "dumpEffectConfigFile@2.0",
vendor: true,
srcs: ["test/DumpConfig.cpp"],
@ -47,10 +38,12 @@ cc_binary {
"-Werror",
],
header_libs: [
"android.hardware.audio.effect.legacy@2.0",
],
shared_libs: [
"libeffectsconfig",
"libeffectsconfig@2.0",
"libeffects",
],
local_include_dirs:[".", "include"],
}

@ -1,189 +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 <stdint.h>
#include <sys/types.h>
#include <hardware/audio_effect.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_*/

@ -19,6 +19,6 @@ LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE:= libldnhncr
LOCAL_HEADER_LIBRARIES := \
libaudioeffects
android.hardware.audio.effect.legacy@2.0 \
include $(BUILD_SHARED_LIBRARY)

@ -35,7 +35,10 @@ LOCAL_C_INCLUDES += \
$(call include-path-for, audio-effects) \
$(call include-path-for, audio-utils) \
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += \
libhardware_headers \
android.hardware.audio.effect.legacy@2.0 \
include $(BUILD_SHARED_LIBRARY)
@ -67,10 +70,11 @@ LOCAL_C_INCLUDES += \
$(LOCAL_PATH)/Reverb \
$(LOCAL_PATH)/../lib/Common/lib/ \
$(LOCAL_PATH)/../lib/Reverb/lib/ \
$(call include-path-for, audio-effects) \
$(call include-path-for, audio-utils) \
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += \
libhardware_headers \
android.hardware.audio.effect.legacy@2.0 \
LOCAL_SANITIZE := integer_overflow

@ -32,5 +32,8 @@ LOCAL_CFLAGS += \
LOCAL_CFLAGS += -fvisibility=hidden
LOCAL_CFLAGS += -Wall -Werror
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += \
libhardware_headers \
android.hardware.audio.effect.legacy@2.0
include $(BUILD_SHARED_LIBRARY)

@ -0,0 +1,26 @@
cc_library_shared {
name: "libeffectproxy",
relative_install_path: "soundfx",
vendor: true,
srcs: ["EffectProxy.cpp"],
include_dirs: ["."],
cflags: [
"-fvisibility=hidden",
"-Wall",
"-Werror",
],
shared_libs: [
"liblog",
"libcutils",
"libutils",
"libdl",
"libeffects",
],
header_libs: [
"android.hardware.audio.effect.legacy@2.0",
],
}

@ -1,35 +0,0 @@
# Copyright 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.
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
LOCAL_MODULE:= libeffectproxy
LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE_TAGS := optional
LOCAL_VENDOR_MODULE := true
LOCAL_SRC_FILES := \
EffectProxy.cpp
LOCAL_CFLAGS+= -fvisibility=hidden
LOCAL_CFLAGS += -Wall -Werror
LOCAL_SHARED_LIBRARIES := liblog libcutils libutils libdl libeffects
LOCAL_C_INCLUDES := \
system/media/audio_effects/include \
frameworks/av/media/libeffects/factory
include $(BUILD_SHARED_LIBRARY)

@ -22,5 +22,8 @@ LOCAL_C_INCLUDES := \
$(call include-path-for, audio-effects)
LOCAL_HEADER_LIBRARIES += libhardware_headers
LOCAL_HEADER_LIBRARIES += \
libhardware_headers \
android.hardware.audio.effect.legacy@2.0 \
include $(BUILD_SHARED_LIBRARY)

@ -12,12 +12,8 @@ cc_library_headers {
],
}
cc_library {
name: "libmedia_helper",
vendor_available: true,
vndk: {
enabled: true,
},
cc_defaults {
name: "libmedia_helper_defaults",
srcs: ["AudioParameter.cpp", "TypeConverter.cpp"],
cflags: [
"-Werror",
@ -28,10 +24,24 @@ cc_library {
header_libs: [
"libmedia_headers",
"libaudioclient_headers",
"libaudio_system_headers",
],
clang: true,
}
cc_library {
name: "libmedia_helper",
defaults: ["libmedia_helper_defaults"],
header_libs: [
"libaudio_system_headers",
],
}
cc_library {
name: "libmedia_helper@2.0",
vendor: true,
defaults: ["libmedia_helper_defaults"],
header_libs: [
"android.hardware.audio.common.legacy@2.0",
],
}
cc_library_shared {
name: "libmedia_omx",
@ -206,7 +216,7 @@ cc_library {
"libmediaextractor",
"libgui",
"libdl",
"libaudioutils",
"libaudioutils_system",
"libaudioclient",
"libmedia_omx",
],

@ -8,7 +8,6 @@ cc_defaults {
],
header_libs: [
"libaudioclient_headers",
"libaudio_system_headers",
"libmedia_headers",
],
export_header_lib_headers: [
@ -17,7 +16,6 @@ cc_defaults {
],
shared_libs: [
"libaudioutils",
"liblog",
"libutils",
],
@ -29,9 +27,17 @@ cc_defaults {
// link against libnbaio and system modules can't link against libnbaio_mono. The rest of libnbaio
// pulls in too many other dependencies.
cc_library_shared {
name: "libnbaio_mono",
name: "libnbaio_mono@2.0",
vendor: true,
defaults: ["libnbaio_mono_defaults"],
shared_libs: [
"libaudioutils",
],
header_libs: [
"android.hardware.audio.common.legacy@2.0",
],
}
cc_library_shared {
@ -54,13 +60,17 @@ cc_library_shared {
// static_libs: ["libsndfile"],
shared_libs: [
"libaudioutils",
"libaudioutils_system",
"libbinder",
"libcutils",
"liblog",
"libutils",
],
header_libs: [
"libaudio_system_headers",
],
cflags: [
"-Werror",
"-Wall",

@ -9,7 +9,7 @@ cc_library_shared {
],
shared_libs: [
"libaudioutils",
"libaudioutils_system",
"libbinder",
"libcutils",
"liblog",

@ -84,7 +84,7 @@ cc_library_shared {
],
shared_libs: [
"libaudioutils",
"libaudioutils_system",
"libbinder",
"libcamera_client",
"libcutils",

@ -129,7 +129,7 @@ cc_test {
shared_libs: [
"libstagefright_amrnb_common",
"libaudioutils",
"libaudioutils_system",
"liblog",
],

@ -78,7 +78,7 @@ cc_test {
local_include_dirs: ["src"],
shared_libs: ["libaudioutils"],
shared_libs: ["libaudioutils_system"],
sanitize: {
misc_undefined: [

@ -151,5 +151,5 @@ cc_test {
"libsndfile",
],
shared_libs: ["libaudioutils"],
shared_libs: ["libaudioutils_system"],
}

@ -42,7 +42,7 @@ LOCAL_SHARED_LIBRARIES := \
libaudiohal \
libaudioprocessing \
libaudiospdif \
libaudioutils \
libaudioutils_system \
libcutils \
libutils \
liblog \

@ -49,7 +49,7 @@ LOCAL_SHARED_LIBRARIES := \
libcutils \
libutils \
liblog \
libaudioutils \
libaudioutils_system \
libparameter \
libmedia_helper \
libxml2

@ -7,7 +7,7 @@ cc_library_shared {
],
shared_libs: [
"libaudioutils",
"libaudioutils_system",
"libbinder",
"liblog",
"libnbaio",

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