85e08e2625
Changed audio device API version to 2.0 because of new enums for audio input and output devices. Removed implementations of get_supported_devices() in stub and usb audio modules. Change-Id: I09345d38929d931e5015e36d18259f5a5f950298
453 lines
16 KiB
C
453 lines
16 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 <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_CURRENT AUDIO_DEVICE_API_VERSION_2_0
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/**
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* List of known audio HAL modules. This is the base name of the audio HAL
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* library composed of the "audio." prefix, one of the base names below and
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* a suffix specific to the device.
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* e.g: audio.primary.goldfish.so or audio.a2dp.default.so
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*/
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#define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary"
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#define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp"
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#define AUDIO_HARDWARE_MODULE_ID_USB "usb"
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#define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix"
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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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/* BT SCO Noise Reduction + Echo Cancellation parameters */
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#define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
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#define AUDIO_PARAMETER_VALUE_ON "on"
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#define AUDIO_PARAMETER_VALUE_OFF "off"
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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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/* A2DP sink address set by framework */
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#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
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/* Screen state */
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#define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state"
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/**
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* audio stream parameters
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*/
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#define AUDIO_PARAMETER_STREAM_ROUTING "routing" // audio_devices_t
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#define AUDIO_PARAMETER_STREAM_FORMAT "format" // audio_format_t
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#define AUDIO_PARAMETER_STREAM_CHANNELS "channels" // audio_channel_mask_t
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#define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" // size_t
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#define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" // audio_source_t
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#define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" // uint32_t
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/* Query supported formats. The response is a '|' separated list of strings from
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* audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
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#define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
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/* Query supported channel masks. The response is a '|' separated list of strings from
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* audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
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#define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
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/* Query supported sampling rates. The response is a '|' separated list of integer values e.g:
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* "sup_sampling_rates=44100|48000" */
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#define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
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/**************************************/
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/* common audio stream configuration parameters */
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struct audio_config {
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uint32_t sample_rate;
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audio_channel_mask_t channel_mask;
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audio_format_t format;
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};
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typedef struct audio_config audio_config_t;
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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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/**
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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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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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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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typedef struct audio_stream_out audio_stream_out_t;
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struct audio_stream_in {
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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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typedef struct audio_stream_in audio_stream_in_t;
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/**
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* return the frame size (number of bytes per sample).
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*/
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static inline size_t audio_stream_frame_size(const struct audio_stream *s)
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{
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size_t chan_samp_sz;
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switch (s->get_format(s)) {
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case AUDIO_FORMAT_PCM_16_BIT:
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chan_samp_sz = sizeof(int16_t);
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break;
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case AUDIO_FORMAT_PCM_8_BIT:
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default:
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chan_samp_sz = sizeof(int8_t);
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break;
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}
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return popcount(s->get_channels(s)) * chan_samp_sz;
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}
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/**********************************************************************/
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/**
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* Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
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* and the fields of this data structure must begin with hw_module_t
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* followed by module specific information.
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*/
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struct audio_module {
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struct hw_module_t common;
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};
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struct audio_hw_device {
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struct hw_device_t common;
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/**
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* used by audio flinger to enumerate what devices are supported by
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* each audio_hw_device implementation.
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*
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* Return value is a bitmask of 1 or more values of audio_devices_t
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*
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* NOTE: audio HAL implementations starting with
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* AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
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* All supported devices should be listed in audio_policy.conf
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* file and the audio policy manager must choose the appropriate
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* audio module based on information in this file.
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*/
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uint32_t (*get_supported_devices)(const struct audio_hw_device *dev);
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/**
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* check to see if the audio hardware interface has been initialized.
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* returns 0 on success, -ENODEV on failure.
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*/
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int (*init_check)(const struct audio_hw_device *dev);
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/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
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int (*set_voice_volume)(struct audio_hw_device *dev, float volume);
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/**
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* set the audio volume for all audio activities other than voice call.
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* Range between 0.0 and 1.0. If any value other than 0 is returned,
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* the software mixer will emulate this capability.
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*/
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int (*set_master_volume)(struct audio_hw_device *dev, float volume);
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/**
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* Get the current master volume value for the HAL, if the HAL supports
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* master volume control. AudioFlinger will query this value from the
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* primary audio HAL when the service starts and use the value for setting
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* the initial master volume across all HALs. HALs which do not support
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* this method may leave it set to NULL.
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*/
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int (*get_master_volume)(struct audio_hw_device *dev, float *volume);
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/**
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* set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
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* is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
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* playing, and AUDIO_MODE_IN_CALL when a call is in progress.
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*/
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int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode);
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/* mic mute */
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int (*set_mic_mute)(struct audio_hw_device *dev, bool state);
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int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state);
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/* set/get global audio parameters */
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int (*set_parameters)(struct audio_hw_device *dev, 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_hw_device *dev,
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const char *keys);
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/* Returns audio input buffer size according to parameters passed or
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* 0 if one of the parameters is not supported.
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* See also get_buffer_size which is for a particular stream.
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*/
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size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
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const struct audio_config *config);
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/** This method creates and opens the audio hardware output stream */
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int (*open_output_stream)(struct audio_hw_device *dev,
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audio_io_handle_t handle,
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audio_devices_t devices,
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audio_output_flags_t flags,
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struct audio_config *config,
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struct audio_stream_out **stream_out);
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void (*close_output_stream)(struct audio_hw_device *dev,
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struct audio_stream_out* stream_out);
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/** This method creates and opens the audio hardware input stream */
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int (*open_input_stream)(struct audio_hw_device *dev,
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audio_io_handle_t handle,
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audio_devices_t devices,
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struct audio_config *config,
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struct audio_stream_in **stream_in);
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void (*close_input_stream)(struct audio_hw_device *dev,
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struct audio_stream_in *stream_in);
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/** This method dumps the state of the audio hardware */
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int (*dump)(const struct audio_hw_device *dev, int fd);
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/**
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* set the audio mute status for all audio activities. If any value other
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* than 0 is returned, the software mixer will emulate this capability.
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*/
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int (*set_master_mute)(struct audio_hw_device *dev, bool mute);
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/**
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* Get the current master mute status for the HAL, if the HAL supports
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* master mute control. AudioFlinger will query this value from the primary
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* audio HAL when the service starts and use the value for setting the
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* initial master mute across all HALs. HALs which do not support this
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* method may leave it set to NULL.
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*/
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int (*get_master_mute)(struct audio_hw_device *dev, bool *mute);
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};
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typedef struct audio_hw_device audio_hw_device_t;
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/** convenience API for opening and closing a supported device */
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static inline int audio_hw_device_open(const struct hw_module_t* module,
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struct audio_hw_device** device)
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{
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return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
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(struct hw_device_t**)device);
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}
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static inline int audio_hw_device_close(struct audio_hw_device* device)
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{
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return device->common.close(&device->common);
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}
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__END_DECLS
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#endif // ANDROID_AUDIO_INTERFACE_H
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