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Revert "Audio V4: Split system and vendor Audio.h"

Browse source 
This reverts commit fc9e212f01.

Reason for revert: Breaks multiple devices

Change-Id: I816671fd92246f85c97d00819858a74e36e2929d
This commit is contained in:
Kevin Rocard 2018-01-24 06:04:27 +00:00
parent fc9e212f01
commit c6ec9486d8
9 changed files with 1989 additions and 15 deletions
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2
Android.bp
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@ -3,11 +3,13 @@
cc_library_headers {
name: "libhardware_headers",
header_libs: [
"libaudio_system_headers",
"libsystem_headers",
"libcutils_headers",
"libbluetooth-types-header",
],
export_header_lib_headers: [
"libaudio_system_headers",
"libsystem_headers",
"libcutils_headers",
"libbluetooth-types-header",

745
include/hardware/audio.h Normal file
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@ -0,0 +1,745 @@
/*
* 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/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

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/*
* 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 */

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/*
* 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

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/*
* 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

24
modules/audio/Android.bp
View file

@ -23,10 +23,7 @@ cc_library_shared {
relative_install_path: "hw",
proprietary: true,
srcs: ["audio_hw.c"],
header_libs: [
"libhardware_headers",
"android.hardware.audio.common.legacy@2.0",
],
header_libs: ["libhardware_headers"],
shared_libs: [
"liblog",
],
@ -44,10 +41,21 @@ cc_library_shared {
relative_install_path: "hw",
proprietary: true,
srcs: ["audio_hw.c"],
header_libs: [
"libhardware_headers",
"android.hardware.audio.common.legacy@2.0",
],
header_libs: ["libhardware_headers"],
shared_libs: [
"liblog",
],
cflags: ["-Wall", "-Werror", "-Wno-unused-parameter"],
}
// The stub audio policy HAL module that can be used as a skeleton for
// new implementations.
cc_library_shared {
name: "audio_policy.stub",
relative_install_path: "hw",
proprietary: true,
srcs: ["audio_policy.c"],
header_libs: ["libhardware_headers"],
shared_libs: [
"liblog",
],

350
modules/audio/audio_policy.c Normal file
View file

@ -0,0 +1,350 @@
/*
* 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.
*/
#define LOG_TAG "audio_policy_default"
//#define LOG_NDEBUG 0
#include <errno.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <system/audio_policy.h>
#include <hardware/audio_policy.h>
struct default_ap_module {
struct audio_policy_module module;
};
struct default_ap_device {
struct audio_policy_device device;
};
struct default_audio_policy {
struct audio_policy policy;
struct audio_policy_service_ops *aps_ops;
void *service;
};
static int ap_set_device_connection_state(struct audio_policy *pol,
audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address)
{
return -ENOSYS;
}
static audio_policy_dev_state_t ap_get_device_connection_state(
const struct audio_policy *pol,
audio_devices_t device,
const char *device_address)
{
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
static void ap_set_phone_state(struct audio_policy *pol, audio_mode_t state)
{
}
// deprecated, never called
static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
uint32_t mask)
{
}
static void ap_set_force_use(struct audio_policy *pol,
audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config)
{
}
/* retreive current device category forced for a given usage */
static audio_policy_forced_cfg_t ap_get_force_use(
const struct audio_policy *pol,
audio_policy_force_use_t usage)
{
return AUDIO_POLICY_FORCE_NONE;
}
/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
* can still be muted. */
static void ap_set_can_mute_enforced_audible(struct audio_policy *pol,
bool can_mute)
{
}
static int ap_init_check(const struct audio_policy *pol)
{
return 0;
}
static audio_io_handle_t ap_get_output(struct audio_policy *pol,
audio_stream_type_t stream,
uint32_t sampling_rate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *info)
{
return 0;
}
static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output,
audio_stream_type_t stream, int session)
{
return -ENOSYS;
}
static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output,
audio_stream_type_t stream, int session)
{
return -ENOSYS;
}
static void ap_release_output(struct audio_policy *pol,
audio_io_handle_t output)
{
}
static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
uint32_t sampling_rate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_in_acoustics_t acoustics)
{
return 0;
}
static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input)
{
return -ENOSYS;
}
static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input)
{
return -ENOSYS;
}
static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input)
{
}
static void ap_init_stream_volume(struct audio_policy *pol,
audio_stream_type_t stream, int index_min,
int index_max)
{
}
static int ap_set_stream_volume_index(struct audio_policy *pol,
audio_stream_type_t stream,
int index)
{
return -ENOSYS;
}
static int ap_get_stream_volume_index(const struct audio_policy *pol,
audio_stream_type_t stream,
int *index)
{
return -ENOSYS;
}
static int ap_set_stream_volume_index_for_device(struct audio_policy *pol,
audio_stream_type_t stream,
int index,
audio_devices_t device)
{
return -ENOSYS;
}
static int ap_get_stream_volume_index_for_device(const struct audio_policy *pol,
audio_stream_type_t stream,
int *index,
audio_devices_t device)
{
return -ENOSYS;
}
static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol,
audio_stream_type_t stream)
{
return 0;
}
static audio_devices_t ap_get_devices_for_stream(const struct audio_policy *pol,
audio_stream_type_t stream)
{
return 0;
}
static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
const struct effect_descriptor_s *desc)
{
return 0;
}
static int ap_register_effect(struct audio_policy *pol,
const struct effect_descriptor_s *desc,
audio_io_handle_t output,
uint32_t strategy,
int session,
int id)
{
return -ENOSYS;
}
static int ap_unregister_effect(struct audio_policy *pol, int id)
{
return -ENOSYS;
}
static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled)
{
return -ENOSYS;
}
static bool ap_is_stream_active(const struct audio_policy *pol, audio_stream_type_t stream,
uint32_t in_past_ms)
{
return false;
}
static int ap_dump(const struct audio_policy *pol, int fd)
{
return -ENOSYS;
}
static bool ap_is_offload_supported(const struct audio_policy *pol,
const audio_offload_info_t *info)
{
return false;
}
static int create_default_ap(const struct audio_policy_device *device,
struct audio_policy_service_ops *aps_ops,
void *service,
struct audio_policy **ap)
{
struct default_audio_policy *dap;
*ap = NULL;
if (!service || !aps_ops)
return -EINVAL;
dap = (struct default_audio_policy *)calloc(1, sizeof(*dap));
if (!dap)
return -ENOMEM;
dap->policy.set_device_connection_state = ap_set_device_connection_state;
dap->policy.get_device_connection_state = ap_get_device_connection_state;
dap->policy.set_phone_state = ap_set_phone_state;
dap->policy.set_ringer_mode = ap_set_ringer_mode;
dap->policy.set_force_use = ap_set_force_use;
dap->policy.get_force_use = ap_get_force_use;
dap->policy.set_can_mute_enforced_audible =
ap_set_can_mute_enforced_audible;
dap->policy.init_check = ap_init_check;
dap->policy.get_output = ap_get_output;
dap->policy.start_output = ap_start_output;
dap->policy.stop_output = ap_stop_output;
dap->policy.release_output = ap_release_output;
dap->policy.get_input = ap_get_input;
dap->policy.start_input = ap_start_input;
dap->policy.stop_input = ap_stop_input;
dap->policy.release_input = ap_release_input;
dap->policy.init_stream_volume = ap_init_stream_volume;
dap->policy.set_stream_volume_index = ap_set_stream_volume_index;
dap->policy.get_stream_volume_index = ap_get_stream_volume_index;
dap->policy.set_stream_volume_index_for_device = ap_set_stream_volume_index_for_device;
dap->policy.get_stream_volume_index_for_device = ap_get_stream_volume_index_for_device;
dap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
dap->policy.get_devices_for_stream = ap_get_devices_for_stream;
dap->policy.get_output_for_effect = ap_get_output_for_effect;
dap->policy.register_effect = ap_register_effect;
dap->policy.unregister_effect = ap_unregister_effect;
dap->policy.set_effect_enabled = ap_set_effect_enabled;
dap->policy.is_stream_active = ap_is_stream_active;
dap->policy.dump = ap_dump;
dap->policy.is_offload_supported = ap_is_offload_supported;
dap->service = service;
dap->aps_ops = aps_ops;
*ap = &dap->policy;
return 0;
}
static int destroy_default_ap(const struct audio_policy_device *ap_dev,
struct audio_policy *ap)
{
free(ap);
return 0;
}
static int default_ap_dev_close(hw_device_t* device)
{
free(device);
return 0;
}
static int default_ap_dev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct default_ap_device *dev;
*device = NULL;
if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0)
return -EINVAL;
dev = (struct default_ap_device *)calloc(1, sizeof(*dev));
if (!dev)
return -ENOMEM;
dev->device.common.tag = HARDWARE_DEVICE_TAG;
dev->device.common.version = 0;
dev->device.common.module = (hw_module_t *)module;
dev->device.common.close = default_ap_dev_close;
dev->device.create_audio_policy = create_default_ap;
dev->device.destroy_audio_policy = destroy_default_ap;
*device = &dev->device.common;
return 0;
}
static struct hw_module_methods_t default_ap_module_methods = {
.open = default_ap_dev_open,
};
struct default_ap_module HAL_MODULE_INFO_SYM = {
.module = {
.common = {
.tag = HARDWARE_MODULE_TAG,
.version_major = 1,
.version_minor = 0,
.id = AUDIO_POLICY_HARDWARE_MODULE_ID,
.name = "Default audio policy HAL",
.author = "The Android Open Source Project",
.methods = &default_ap_module_methods,
},
},
};

9
modules/audio_remote_submix/Android.bp
View file

@ -17,15 +17,16 @@ cc_library_shared {
relative_install_path: "hw",
vendor: true,
srcs: ["audio_hw.cpp"],
include_dirs: [
"system/media/audio_utils/include",
],
shared_libs: [
"liblog",
"libcutils",
"libutils",
"libnbaio_mono@2.0",
"libaudioutils",
"libnbaio_mono",
],
static_libs: ["libmedia_helper@2.0"],
static_libs: ["libmedia_helper"],
cflags: ["-Wno-unused-parameter"],

4
modules/usbaudio/Android.bp
View file

@ -25,7 +25,5 @@ cc_library_shared {
"libalsautils",
],
cflags: ["-Wno-unused-parameter"],
header_libs: [
"android.hardware.audio.common.legacy@2.0",
],
header_libs: ["libhardware_headers"],
}