/*
* 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 <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_CURRENT AUDIO_DEVICE_API_VERSION_1_0
/**
* List of known audio HAL modules. This is the base name of the audio HAL
* library composed of the "audio." prefix, one of the base names below and
* a suffix specific to the device.
* e.g: audio.primary.goldfish.so or audio.a2dp.default.so
*/
#define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary"
#define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp"
#define AUDIO_HARDWARE_MODULE_ID_USB "usb"
/**************************************/
/**
* standard audio parameters that the HAL may need to handle
*/
/**
* audio device parameters
*/
/* BT SCO Noise Reduction + Echo Cancellation parameters */
#define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
#define AUDIO_PARAMETER_VALUE_ON "on"
#define AUDIO_PARAMETER_VALUE_OFF "off"
/* 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"
/* A2DP sink address set by framework */
#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
/* Screen state */
#define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state"
/**
* audio stream parameters
*/
#define AUDIO_PARAMETER_STREAM_ROUTING "routing" // audio_devices_t
#define AUDIO_PARAMETER_STREAM_FORMAT "format" // audio_format_t
#define AUDIO_PARAMETER_STREAM_CHANNELS "channels" // audio_channel_mask_t
#define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" // size_t
#define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" // audio_source_t
#define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" // uint32_t
/* Query supported formats. The response is a '|' separated list of strings from
* audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
#define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
/* Query supported channel masks. The response is a '|' separated list of strings from
* audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
#define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
/* Query supported sampling rates. The response is a '|' separated list of integer values e.g:
* "sup_sampling_rates=44100|48000" */
#define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
/**************************************/
/* common audio stream configuration parameters */
struct audio_config {
uint32_t sample_rate;
audio_channel_mask_t channel_mask;
audio_format_t format;
};
typedef struct audio_config audio_config_t;
/* 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;
/**
* 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 {
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.
*/
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);
};
typedef struct audio_stream_out audio_stream_out_t;
struct 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);
};
typedef struct audio_stream_in audio_stream_in_t;
/**
* return the frame size (number of bytes per sample).
*/
static inline size_t audio_stream_frame_size(struct audio_stream *s)
{
size_t chan_samp_sz;
switch (s->get_format(s)) {
case AUDIO_FORMAT_PCM_16_BIT:
chan_samp_sz = sizeof(int16_t);
break;
case AUDIO_FORMAT_PCM_8_BIT:
default:
chan_samp_sz = sizeof(int8_t);
break;
}
return popcount(s->get_channels(s)) * chan_samp_sz;
}
/**********************************************************************/
/**
* 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 {
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
*/
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 should 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 */
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);
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);
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);
};
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,
(struct hw_device_t**)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