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platform_hardware_libhardware/include/hardware/audio.h
Kuowei Li 96374df243 audio: add implementation for TV related API 
1. AudioDescriptionMixLevel getter/setter.
2. DualMonoMode getter/setter.
3. PlaybackRate getter/setter.

Test: TIS player with instrumented HAL
Test: atest AudioTrackTest#testTunerConfiguration
Test: atest AudioTrackTest#testDualMonoMode
Test: atest AudioTrackTest#testAudioDescriptionMixLevel
Test: atest AudioManagerTest#testGetAdditionalOutputDeviceDelay
Bug: 173482792
Merged-In: Id89dd151797723dbb58a44db48ff05bff3f20e04
Change-Id: Id89dd151797723dbb58a44db48ff05bff3f20e04
2021-01-27 19:24:54 -08:00

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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_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_3_1 HARDWARE_DEVICE_API_VERSION(3, 1)
#define AUDIO_DEVICE_API_VERSION_3_2 HARDWARE_DEVICE_API_VERSION(3, 2)
#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_2
/* 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"
/* BT SCO headset name for debug */
#define AUDIO_PARAMETER_KEY_BT_SCO_HEADSET_NAME "bt_headset_name"
/* BT SCO HFP control */
#define AUDIO_PARAMETER_KEY_HFP_ENABLE "hfp_enable"
#define AUDIO_PARAMETER_KEY_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
#define AUDIO_PARAMETER_KEY_HFP_VOLUME "hfp_volume"
/* Set screen orientation */
#define AUDIO_PARAMETER_KEY_ROTATION "rotation"
/**
* 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 enum {
STREAM_EVENT_CBK_TYPE_CODEC_FORMAT_CHANGED, /* codec format of the stream changed */
} stream_event_callback_type_t;
typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);
typedef int (*stream_event_callback_t)(stream_event_callback_type_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;
typedef struct source_metadata {
size_t track_count;
/** Array of metadata of each track connected to this source. */
struct playback_track_metadata* tracks;
} source_metadata_t;
typedef struct sink_metadata {
size_t track_count;
/** Array of metadata of each track connected to this sink. */
struct record_track_metadata* tracks;
} sink_metadata_t;
/* HAL version 3.2 and higher only. */
typedef struct source_metadata_v7 {
size_t track_count;
/** Array of metadata of each track connected to this source. */
struct playback_track_metadata_v7* tracks;
} source_metadata_v7_t;
/* HAL version 3.2 and higher only. */
typedef struct sink_metadata_v7 {
size_t track_count;
/** Array of metadata of each track connected to this sink. */
struct record_track_metadata_v7* tracks;
} sink_metadata_v7_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);
/**
* Called when the metadata of the stream's source has been changed.
* @param source_metadata Description of the audio that is played by the clients.
*/
void (*update_source_metadata)(struct audio_stream_out *stream,
const struct source_metadata* source_metadata);
/**
* Set the callback function for notifying events for an output stream.
*/
int (*set_event_callback)(struct audio_stream_out *stream,
stream_event_callback_t callback,
void *cookie);
/**
* Called when the metadata of the stream's source has been changed.
* HAL version 3.2 and higher only.
* @param source_metadata Description of the audio that is played by the clients.
*/
void (*update_source_metadata_v7)(struct audio_stream_out *stream,
const struct source_metadata_v7* source_metadata);
/**
* Returns the Dual Mono mode presentation setting.
*
* \param[in] stream the stream object.
* \param[out] mode current setting of Dual Mono mode.
*
* \return 0 if the position is successfully returned.
* -EINVAL if the arguments are invalid
* -ENOSYS if the function is not available
*/
int (*get_dual_mono_mode)(struct audio_stream_out *stream, audio_dual_mono_mode_t *mode);
/**
* Sets the Dual Mono mode presentation on the output device.
*
* \param[in] stream the stream object.
* \param[in] mode selected Dual Mono mode.
*
* \return 0 in case of success.
* -EINVAL if the arguments are invalid
* -ENOSYS if the function is not available
*/
int (*set_dual_mono_mode)(struct audio_stream_out *stream, const audio_dual_mono_mode_t mode);
/**
* Returns the Audio Description Mix level in dB.
*
* \param[in] stream the stream object.
* \param[out] leveldB the current Audio Description Mix Level in dB.
*
* \return 0 in case of success.
* -EINVAL if the arguments are invalid
* -ENOSYS if the function is not available
*/
int (*get_audio_description_mix_level)(struct audio_stream_out *stream, float *leveldB);
/**
* Sets the Audio Description Mix level in dB.
*
* \param[in] stream the stream object.
* \param[in] leveldB Audio Description Mix Level in dB.
*
* \return 0 in case of success.
* -EINVAL if the arguments are invalid
* -ENOSYS if the function is not available
*/
int (*set_audio_description_mix_level)(struct audio_stream_out *stream, const float leveldB);
/**
* Retrieves current playback rate parameters.
*
* \param[in] stream the stream object.
* \param[out] playbackRate current playback parameters.
*
* \return 0 in case of success.
* -EINVAL if the arguments are invalid
* -ENOSYS if the function is not available
*/
int (*get_playback_rate_parameters)(struct audio_stream_out *stream,
audio_playback_rate_t *playbackRate);
/**
* Sets the playback rate parameters that control playback behavior.
*
* \param[in] stream the stream object.
* \param[in] playbackRate playback parameters.
*
* \return 0 in case of success.
* -EINVAL if the arguments are invalid
* -ENOSYS if the function is not available
*/
int (*set_playback_rate_parameters)(struct audio_stream_out *stream,
const audio_playback_rate_t *playbackRate);
};
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);
/**
* Called by the framework to read active microphones
*
* \param[in] stream the stream object.
* \param[out] mic_array Pointer to first element on array with microphone info
* \param[out] mic_count When called, this holds the value of the max number of elements
* allowed in the mic_array. The actual number of elements written
* is returned here.
* if mic_count is passed as zero, mic_array will not be populated,
* and mic_count will return the actual number of active microphones.
*
* \return 0 if the microphone array is successfully filled.
* -ENOSYS if there is an error filling the data
*/
int (*get_active_microphones)(const struct audio_stream_in *stream,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count);
/**
* Called by the framework to instruct the HAL to optimize the capture stream in the
* specified direction.
*
* \param[in] stream the stream object.
* \param[in] direction The direction constant (from audio-base.h)
* MIC_DIRECTION_UNSPECIFIED Don't do any directionality processing of the
* activated microphone(s).
* MIC_DIRECTION_FRONT Optimize capture for audio coming from the screen-side
* of the device.
* MIC_DIRECTION_BACK Optimize capture for audio coming from the side of the
* device opposite the screen.
* MIC_DIRECTION_EXTERNAL Optimize capture for audio coming from an off-device
* microphone.
* \return OK if the call is successful, an error code otherwise.
*/
int (*set_microphone_direction)(const struct audio_stream_in *stream,
audio_microphone_direction_t direction);
/**
* Called by the framework to specify to the HAL the desired zoom factor for the selected
* microphone(s).
*
* \param[in] stream the stream object.
* \param[in] zoom the zoom factor.
* \return OK if the call is successful, an error code otherwise.
*/
int (*set_microphone_field_dimension)(const struct audio_stream_in *stream,
float zoom);
/**
* Called when the metadata of the stream's sink has been changed.
* @param sink_metadata Description of the audio that is recorded by the clients.
*/
void (*update_sink_metadata)(struct audio_stream_in *stream,
const struct sink_metadata* sink_metadata);
/**
* Called when the metadata of the stream's sink has been changed.
* HAL version 3.2 and higher only.
* @param sink_metadata Description of the audio that is recorded by the clients.
*/
void (*update_sink_metadata_v7)(struct audio_stream_in *stream,
const struct sink_metadata_v7* sink_metadata);
};
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);
/**
* Called by the framework to read available microphones characteristics.
*
* \param[in] dev the hw_device object.
* \param[out] mic_array Pointer to first element on array with microphone info
* \param[out] mic_count When called, this holds the value of the max number of elements
* allowed in the mic_array. The actual number of elements written
* is returned here.
* if mic_count is passed as zero, mic_array will not be populated,
* and mic_count will return the actual number of microphones in the
* system.
*
* \return 0 if the microphone array is successfully filled.
* -ENOSYS if there is an error filling the data
*/
int (*get_microphones)(const struct audio_hw_device *dev,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count);
/** 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);
/**
* Applies an audio effect to an audio device.
*
* @param dev the audio HAL device context.
* @param device identifies the sink or source device the effect must be applied to.
* "device" is the audio_port_handle_t indicated for the device when
* the audio patch connecting that device was created.
* @param effect effect interface handle corresponding to the effect being added.
* @return retval operation completion status.
*/
int (*add_device_effect)(struct audio_hw_device *dev,
audio_port_handle_t device, effect_handle_t effect);
/**
* Stops applying an audio effect to an audio device.
*
* @param dev the audio HAL device context.
* @param device identifies the sink or source device this effect was applied to.
* "device" is the audio_port_handle_t indicated for the device when
* the audio patch is created.
* @param effect effect interface handle corresponding to the effect being removed.
* @return retval operation completion status.
*/
int (*remove_device_effect)(struct audio_hw_device *dev,
audio_port_handle_t device, effect_handle_t effect);
/**
* 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. The possible attributes
* are saved as audio profiles, which contains audio format and the supported
* sampling rates and channel masks.
*/
int (*get_audio_port_v7)(struct audio_hw_device *dev,
struct audio_port_v7 *port);
};
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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