745 lines
29 KiB
C
745 lines
29 KiB
C
/*
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* Copyright (C) 2011 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_HAL_INTERFACE_H
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#define ANDROID_AUDIO_HAL_INTERFACE_H
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#include <stdint.h>
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#include <strings.h>
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#include <sys/cdefs.h>
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#include <sys/types.h>
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#include <time.h>
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#include <cutils/bitops.h>
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#include <hardware/hardware.h>
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#include <system/audio.h>
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#include <hardware/audio_effect.h>
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__BEGIN_DECLS
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/**
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* The id of this module
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*/
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#define AUDIO_HARDWARE_MODULE_ID "audio"
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/**
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* Name of the audio devices to open
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*/
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#define AUDIO_HARDWARE_INTERFACE "audio_hw_if"
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/* Use version 0.1 to be compatible with first generation of audio hw module with version_major
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* hardcoded to 1. No audio module API change.
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*/
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#define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1)
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#define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1
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/* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0
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* will be considered of first generation API.
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*/
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#define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0)
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#define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0)
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#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0)
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#define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0)
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#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_0
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/* Minimal audio HAL version supported by the audio framework */
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#define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0
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/**************************************/
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/**
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* standard audio parameters that the HAL may need to handle
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*/
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/**
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* audio device parameters
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*/
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/* TTY mode selection */
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#define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode"
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#define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off"
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#define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco"
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#define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco"
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#define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full"
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/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */
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#define AUDIO_PARAMETER_KEY_HAC "HACSetting"
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#define AUDIO_PARAMETER_VALUE_HAC_ON "ON"
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#define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF"
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/* A2DP sink address set by framework */
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#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
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/* A2DP source address set by framework */
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#define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"
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/* Bluetooth SCO wideband */
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#define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs"
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/**
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* audio stream parameters
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*/
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/* Enable AANC */
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#define AUDIO_PARAMETER_KEY_AANC "aanc_enabled"
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/**************************************/
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/* common audio stream parameters and operations */
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struct audio_stream {
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/**
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* Return the sampling rate in Hz - eg. 44100.
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*/
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uint32_t (*get_sample_rate)(const struct audio_stream *stream);
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/* currently unused - use set_parameters with key
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* AUDIO_PARAMETER_STREAM_SAMPLING_RATE
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*/
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int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate);
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/**
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* Return size of input/output buffer in bytes for this stream - eg. 4800.
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* It should be a multiple of the frame size. See also get_input_buffer_size.
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*/
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size_t (*get_buffer_size)(const struct audio_stream *stream);
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/**
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* Return the channel mask -
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* e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
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*/
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audio_channel_mask_t (*get_channels)(const struct audio_stream *stream);
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/**
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* Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
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*/
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audio_format_t (*get_format)(const struct audio_stream *stream);
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/* currently unused - use set_parameters with key
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* AUDIO_PARAMETER_STREAM_FORMAT
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*/
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int (*set_format)(struct audio_stream *stream, audio_format_t format);
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/**
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* Put the audio hardware input/output into standby mode.
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* Driver should exit from standby mode at the next I/O operation.
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* Returns 0 on success and <0 on failure.
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*/
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int (*standby)(struct audio_stream *stream);
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/** dump the state of the audio input/output device */
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int (*dump)(const struct audio_stream *stream, int fd);
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/** Return the set of device(s) which this stream is connected to */
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audio_devices_t (*get_device)(const struct audio_stream *stream);
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/**
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* Currently unused - set_device() corresponds to set_parameters() with key
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* AUDIO_PARAMETER_STREAM_ROUTING for both input and output.
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* AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by
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* input streams only.
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*/
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int (*set_device)(struct audio_stream *stream, audio_devices_t device);
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/**
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* set/get audio stream parameters. The function accepts a list of
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* parameter key value pairs in the form: key1=value1;key2=value2;...
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*
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* Some keys are reserved for standard parameters (See AudioParameter class)
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*
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* If the implementation does not accept a parameter change while
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* the output is active but the parameter is acceptable otherwise, it must
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* return -ENOSYS.
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*
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* The audio flinger will put the stream in standby and then change the
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* parameter value.
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*/
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int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs);
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/*
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* Returns a pointer to a heap allocated string. The caller is responsible
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* for freeing the memory for it using free().
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*/
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char * (*get_parameters)(const struct audio_stream *stream,
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const char *keys);
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int (*add_audio_effect)(const struct audio_stream *stream,
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effect_handle_t effect);
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int (*remove_audio_effect)(const struct audio_stream *stream,
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effect_handle_t effect);
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};
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typedef struct audio_stream audio_stream_t;
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/* type of asynchronous write callback events. Mutually exclusive */
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typedef enum {
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STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
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STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */
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STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */
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} stream_callback_event_t;
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typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);
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/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
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typedef enum {
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AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
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AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
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from the current track has been played to
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give time for gapless track switch */
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} audio_drain_type_t;
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/**
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* audio_stream_out is the abstraction interface for the audio output hardware.
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*
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* It provides information about various properties of the audio output
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* hardware driver.
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*/
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struct audio_stream_out {
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/**
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* Common methods of the audio stream out. This *must* be the first member of audio_stream_out
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* as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
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* where it's known the audio_stream references an audio_stream_out.
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*/
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struct audio_stream common;
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/**
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* Return the audio hardware driver estimated latency in milliseconds.
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*/
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uint32_t (*get_latency)(const struct audio_stream_out *stream);
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/**
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* Use this method in situations where audio mixing is done in the
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* hardware. This method serves as a direct interface with hardware,
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* allowing you to directly set the volume as apposed to via the framework.
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* This method might produce multiple PCM outputs or hardware accelerated
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* codecs, such as MP3 or AAC.
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*/
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int (*set_volume)(struct audio_stream_out *stream, float left, float right);
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/**
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* Write audio buffer to driver. Returns number of bytes written, or a
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* negative status_t. If at least one frame was written successfully prior to the error,
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* it is suggested that the driver return that successful (short) byte count
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* and then return an error in the subsequent call.
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*
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* If set_callback() has previously been called to enable non-blocking mode
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* the write() is not allowed to block. It must write only the number of
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* bytes that currently fit in the driver/hardware buffer and then return
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* this byte count. If this is less than the requested write size the
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* callback function must be called when more space is available in the
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* driver/hardware buffer.
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*/
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ssize_t (*write)(struct audio_stream_out *stream, const void* buffer,
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size_t bytes);
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/* return the number of audio frames written by the audio dsp to DAC since
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* the output has exited standby
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*/
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int (*get_render_position)(const struct audio_stream_out *stream,
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uint32_t *dsp_frames);
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/**
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* get the local time at which the next write to the audio driver will be presented.
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* The units are microseconds, where the epoch is decided by the local audio HAL.
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*/
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int (*get_next_write_timestamp)(const struct audio_stream_out *stream,
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int64_t *timestamp);
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/**
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* set the callback function for notifying completion of non-blocking
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* write and drain.
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* Calling this function implies that all future write() and drain()
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* must be non-blocking and use the callback to signal completion.
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*/
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int (*set_callback)(struct audio_stream_out *stream,
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stream_callback_t callback, void *cookie);
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/**
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* Notifies to the audio driver to stop playback however the queued buffers are
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* retained by the hardware. Useful for implementing pause/resume. Empty implementation
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* if not supported however should be implemented for hardware with non-trivial
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* latency. In the pause state audio hardware could still be using power. User may
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* consider calling suspend after a timeout.
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*
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* Implementation of this function is mandatory for offloaded playback.
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*/
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int (*pause)(struct audio_stream_out* stream);
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/**
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* Notifies to the audio driver to resume playback following a pause.
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* Returns error if called without matching pause.
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*
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* Implementation of this function is mandatory for offloaded playback.
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*/
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int (*resume)(struct audio_stream_out* stream);
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/**
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* Requests notification when data buffered by the driver/hardware has
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* been played. If set_callback() has previously been called to enable
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* non-blocking mode, the drain() must not block, instead it should return
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* quickly and completion of the drain is notified through the callback.
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* If set_callback() has not been called, the drain() must block until
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* completion.
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* If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
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* data has been played.
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* If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
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* data for the current track has played to allow time for the framework
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* to perform a gapless track switch.
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*
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* Drain must return immediately on stop() and flush() call
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*
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* Implementation of this function is mandatory for offloaded playback.
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*/
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int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type );
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/**
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* Notifies to the audio driver to flush the queued data. Stream must already
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* be paused before calling flush().
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*
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* Implementation of this function is mandatory for offloaded playback.
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*/
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int (*flush)(struct audio_stream_out* stream);
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/**
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* Return a recent count of the number of audio frames presented to an external observer.
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* This excludes frames which have been written but are still in the pipeline.
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* The count is not reset to zero when output enters standby.
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* Also returns the value of CLOCK_MONOTONIC as of this presentation count.
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* The returned count is expected to be 'recent',
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* but does not need to be the most recent possible value.
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* However, the associated time should correspond to whatever count is returned.
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* Example: assume that N+M frames have been presented, where M is a 'small' number.
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* Then it is permissible to return N instead of N+M,
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* and the timestamp should correspond to N rather than N+M.
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* The terms 'recent' and 'small' are not defined.
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* They reflect the quality of the implementation.
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*
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* 3.0 and higher only.
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*/
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int (*get_presentation_position)(const struct audio_stream_out *stream,
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uint64_t *frames, struct timespec *timestamp);
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/**
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* Called by the framework to start a stream operating in mmap mode.
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* create_mmap_buffer must be called before calling start()
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \return 0 in case of success.
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* -ENOSYS if called out of sequence or on non mmap stream
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*/
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int (*start)(const struct audio_stream_out* stream);
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/**
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* Called by the framework to stop a stream operating in mmap mode.
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* Must be called after start()
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \return 0 in case of success.
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* -ENOSYS if called out of sequence or on non mmap stream
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*/
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int (*stop)(const struct audio_stream_out* stream);
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/**
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* Called by the framework to retrieve information on the mmap buffer used for audio
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* samples transfer.
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \param[in] min_size_frames minimum buffer size requested. The actual buffer
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* size returned in struct audio_mmap_buffer_info can be larger.
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* \param[out] info address at which the mmap buffer information should be returned.
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*
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* \return 0 if the buffer was allocated.
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* -ENODEV in case of initialization error
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* -EINVAL if the requested buffer size is too large
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* -ENOSYS if called out of sequence (e.g. buffer already allocated)
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*/
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int (*create_mmap_buffer)(const struct audio_stream_out *stream,
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int32_t min_size_frames,
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struct audio_mmap_buffer_info *info);
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/**
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* Called by the framework to read current read/write position in the mmap buffer
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* with associated time stamp.
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \param[out] position address at which the mmap read/write position should be returned.
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*
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* \return 0 if the position is successfully returned.
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* -ENODATA if the position cannot be retrieved
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* -ENOSYS if called before create_mmap_buffer()
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*/
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int (*get_mmap_position)(const struct audio_stream_out *stream,
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struct audio_mmap_position *position);
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};
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typedef struct audio_stream_out audio_stream_out_t;
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struct audio_stream_in {
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/**
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* Common methods of the audio stream in. This *must* be the first member of audio_stream_in
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* as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
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* where it's known the audio_stream references an audio_stream_in.
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*/
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struct audio_stream common;
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/** set the input gain for the audio driver. This method is for
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* for future use */
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int (*set_gain)(struct audio_stream_in *stream, float gain);
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/** Read audio buffer in from audio driver. Returns number of bytes read, or a
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* negative status_t. If at least one frame was read prior to the error,
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* read should return that byte count and then return an error in the subsequent call.
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*/
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ssize_t (*read)(struct audio_stream_in *stream, void* buffer,
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size_t bytes);
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/**
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* Return the amount of input frames lost in the audio driver since the
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* last call of this function.
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* Audio driver is expected to reset the value to 0 and restart counting
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* upon returning the current value by this function call.
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* Such loss typically occurs when the user space process is blocked
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* longer than the capacity of audio driver buffers.
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*
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* Unit: the number of input audio frames
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*/
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uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream);
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/**
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* Return a recent count of the number of audio frames received and
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* the clock time associated with that frame count.
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*
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* frames is the total frame count received. This should be as early in
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* the capture pipeline as possible. In general,
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* frames should be non-negative and should not go "backwards".
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*
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* time is the clock MONOTONIC time when frames was measured. In general,
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* time should be a positive quantity and should not go "backwards".
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*
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* The status returned is 0 on success, -ENOSYS if the device is not
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* ready/available, or -EINVAL if the arguments are null or otherwise invalid.
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*/
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int (*get_capture_position)(const struct audio_stream_in *stream,
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int64_t *frames, int64_t *time);
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/**
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* Called by the framework to start a stream operating in mmap mode.
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* create_mmap_buffer must be called before calling start()
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \return 0 in case off success.
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* -ENOSYS if called out of sequence or on non mmap stream
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*/
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int (*start)(const struct audio_stream_in* stream);
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/**
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* Called by the framework to stop a stream operating in mmap mode.
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \return 0 in case of success.
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* -ENOSYS if called out of sequence or on non mmap stream
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*/
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int (*stop)(const struct audio_stream_in* stream);
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/**
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* Called by the framework to retrieve information on the mmap buffer used for audio
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* samples transfer.
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*
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* \note Function only implemented by streams operating in mmap mode.
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*
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* \param[in] stream the stream object.
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* \param[in] min_size_frames minimum buffer size requested. The actual buffer
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* size returned in struct audio_mmap_buffer_info can be larger.
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* \param[out] info address at which the mmap buffer information should be returned.
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*
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* \return 0 if the buffer was allocated.
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* -ENODEV in case of initialization error
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* -EINVAL if the requested buffer size is too large
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* -ENOSYS if called out of sequence (e.g. buffer already allocated)
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|
*/
|
|
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
|