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Diffstat (limited to 'include/hardware/audio.h')
| l---------[-rw-r--r--] | include/hardware/audio.h | 1089 |
1 files changed, 1 insertions, 1088 deletions
diff --git a/include/hardware/audio.h b/include/hardware/audio.h index daaa16f1..7e25631f 100644..120000 --- a/include/hardware/audio.h +++ b/include/hardware/audio.h @@ -1,1088 +1 @@ -/* - * Copyright (C) 2011 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - - -#ifndef ANDROID_AUDIO_HAL_INTERFACE_H -#define ANDROID_AUDIO_HAL_INTERFACE_H - -#include <stdint.h> -#include <strings.h> -#include <sys/cdefs.h> -#include <sys/types.h> -#include <time.h> - -#include <cutils/bitops.h> - -#include <hardware/hardware.h> -#include <system/audio.h> -#include <hardware/audio_effect.h> - -__BEGIN_DECLS - -/** - * The id of this module - */ -#define AUDIO_HARDWARE_MODULE_ID "audio" - -/** - * Name of the audio devices to open - */ -#define AUDIO_HARDWARE_INTERFACE "audio_hw_if" - - -/* Use version 0.1 to be compatible with first generation of audio hw module with version_major - * hardcoded to 1. No audio module API change. - */ -#define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1) -#define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1 - -/* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0 - * will be considered of first generation API. - */ -#define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0) -#define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) -#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0) -#define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0) -#define AUDIO_DEVICE_API_VERSION_3_1 HARDWARE_DEVICE_API_VERSION(3, 1) -#define AUDIO_DEVICE_API_VERSION_3_2 HARDWARE_DEVICE_API_VERSION(3, 2) -#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_2 -/* Minimal audio HAL version supported by the audio framework */ -#define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0 - -/**************************************/ - -/** - * standard audio parameters that the HAL may need to handle - */ - -/** - * audio device parameters - */ - -/* TTY mode selection */ -#define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode" -#define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off" -#define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco" -#define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco" -#define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full" - -/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */ -#define AUDIO_PARAMETER_KEY_HAC "HACSetting" -#define AUDIO_PARAMETER_VALUE_HAC_ON "ON" -#define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF" - -/* A2DP sink address set by framework */ -#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address" - -/* A2DP source address set by framework */ -#define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address" - -/* Bluetooth SCO wideband */ -#define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs" - -/* BT SCO headset name for debug */ -#define AUDIO_PARAMETER_KEY_BT_SCO_HEADSET_NAME "bt_headset_name" - -/* BT SCO HFP control */ -#define AUDIO_PARAMETER_KEY_HFP_ENABLE "hfp_enable" -#define AUDIO_PARAMETER_KEY_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate" -#define AUDIO_PARAMETER_KEY_HFP_VOLUME "hfp_volume" - -/* Set screen orientation */ -#define AUDIO_PARAMETER_KEY_ROTATION "rotation" - -/** - * audio stream parameters - */ - -/* Enable AANC */ -#define AUDIO_PARAMETER_KEY_AANC "aanc_enabled" - -/**************************************/ - -/* common audio stream parameters and operations */ -struct audio_stream { - - /** - * Return the sampling rate in Hz - eg. 44100. - */ - uint32_t (*get_sample_rate)(const struct audio_stream *stream); - - /* currently unused - use set_parameters with key - * AUDIO_PARAMETER_STREAM_SAMPLING_RATE - */ - int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate); - - /** - * Return size of input/output buffer in bytes for this stream - eg. 4800. - * It should be a multiple of the frame size. See also get_input_buffer_size. - */ - size_t (*get_buffer_size)(const struct audio_stream *stream); - - /** - * Return the channel mask - - * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO - */ - audio_channel_mask_t (*get_channels)(const struct audio_stream *stream); - - /** - * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT - */ - audio_format_t (*get_format)(const struct audio_stream *stream); - - /* currently unused - use set_parameters with key - * AUDIO_PARAMETER_STREAM_FORMAT - */ - int (*set_format)(struct audio_stream *stream, audio_format_t format); - - /** - * Put the audio hardware input/output into standby mode. - * Driver should exit from standby mode at the next I/O operation. - * Returns 0 on success and <0 on failure. - */ - int (*standby)(struct audio_stream *stream); - - /** dump the state of the audio input/output device */ - int (*dump)(const struct audio_stream *stream, int fd); - - /** Return the set of device(s) which this stream is connected to */ - audio_devices_t (*get_device)(const struct audio_stream *stream); - - /** - * Currently unused - set_device() corresponds to set_parameters() with key - * AUDIO_PARAMETER_STREAM_ROUTING for both input and output. - * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by - * input streams only. - */ - int (*set_device)(struct audio_stream *stream, audio_devices_t device); - - /** - * set/get audio stream parameters. The function accepts a list of - * parameter key value pairs in the form: key1=value1;key2=value2;... - * - * Some keys are reserved for standard parameters (See AudioParameter class) - * - * If the implementation does not accept a parameter change while - * the output is active but the parameter is acceptable otherwise, it must - * return -ENOSYS. - * - * The audio flinger will put the stream in standby and then change the - * parameter value. - */ - int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs); - - /* - * Returns a pointer to a heap allocated string. The caller is responsible - * for freeing the memory for it using free(). - */ - char * (*get_parameters)(const struct audio_stream *stream, - const char *keys); - int (*add_audio_effect)(const struct audio_stream *stream, - effect_handle_t effect); - int (*remove_audio_effect)(const struct audio_stream *stream, - effect_handle_t effect); -}; -typedef struct audio_stream audio_stream_t; - -/* type of asynchronous write callback events. Mutually exclusive */ -typedef enum { - STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */ - STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */ - STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */ -} stream_callback_event_t; - -typedef enum { - STREAM_EVENT_CBK_TYPE_CODEC_FORMAT_CHANGED, /* codec format of the stream changed */ -} stream_event_callback_type_t; - -typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie); - -typedef int (*stream_event_callback_t)(stream_event_callback_type_t event, - void *param, void *cookie); - -/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */ -typedef enum { - AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */ - AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data - from the current track has been played to - give time for gapless track switch */ -} audio_drain_type_t; - -typedef struct source_metadata { - size_t track_count; - /** Array of metadata of each track connected to this source. */ - struct playback_track_metadata* tracks; -} source_metadata_t; - -typedef struct sink_metadata { - size_t track_count; - /** Array of metadata of each track connected to this sink. */ - struct record_track_metadata* tracks; -} sink_metadata_t; - -/* HAL version 3.2 and higher only. */ -typedef struct source_metadata_v7 { - size_t track_count; - /** Array of metadata of each track connected to this source. */ - struct playback_track_metadata_v7* tracks; -} source_metadata_v7_t; - -/* HAL version 3.2 and higher only. */ -typedef struct sink_metadata_v7 { - size_t track_count; - /** Array of metadata of each track connected to this sink. */ - struct record_track_metadata_v7* tracks; -} sink_metadata_v7_t; - -/** output stream callback method to indicate changes in supported latency modes */ -typedef void (*stream_latency_mode_callback_t)( - audio_latency_mode_t *modes, size_t num_modes, void *cookie); - -/** - * audio_stream_out is the abstraction interface for the audio output hardware. - * - * It provides information about various properties of the audio output - * hardware driver. - */ -struct audio_stream_out { - /** - * Common methods of the audio stream out. This *must* be the first member of audio_stream_out - * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts - * where it's known the audio_stream references an audio_stream_out. - */ - struct audio_stream common; - - /** - * Return the audio hardware driver estimated latency in milliseconds. - */ - uint32_t (*get_latency)(const struct audio_stream_out *stream); - - /** - * Use this method in situations where audio mixing is done in the - * hardware. This method serves as a direct interface with hardware, - * allowing you to directly set the volume as apposed to via the framework. - * This method might produce multiple PCM outputs or hardware accelerated - * codecs, such as MP3 or AAC. - */ - int (*set_volume)(struct audio_stream_out *stream, float left, float right); - - /** - * Write audio buffer to driver. Returns number of bytes written, or a - * negative status_t. If at least one frame was written successfully prior to the error, - * it is suggested that the driver return that successful (short) byte count - * and then return an error in the subsequent call. - * - * If set_callback() has previously been called to enable non-blocking mode - * the write() is not allowed to block. It must write only the number of - * bytes that currently fit in the driver/hardware buffer and then return - * this byte count. If this is less than the requested write size the - * callback function must be called when more space is available in the - * driver/hardware buffer. - */ - ssize_t (*write)(struct audio_stream_out *stream, const void* buffer, - size_t bytes); - - /* return the number of audio frames written by the audio dsp to DAC since - * the output has exited standby - */ - int (*get_render_position)(const struct audio_stream_out *stream, - uint32_t *dsp_frames); - - /** - * get the local time at which the next write to the audio driver will be presented. - * The units are microseconds, where the epoch is decided by the local audio HAL. - */ - int (*get_next_write_timestamp)(const struct audio_stream_out *stream, - int64_t *timestamp); - - /** - * set the callback function for notifying completion of non-blocking - * write and drain. - * Calling this function implies that all future write() and drain() - * must be non-blocking and use the callback to signal completion. - */ - int (*set_callback)(struct audio_stream_out *stream, - stream_callback_t callback, void *cookie); - - /** - * Notifies to the audio driver to stop playback however the queued buffers are - * retained by the hardware. Useful for implementing pause/resume. Empty implementation - * if not supported however should be implemented for hardware with non-trivial - * latency. In the pause state audio hardware could still be using power. User may - * consider calling suspend after a timeout. - * - * Implementation of this function is mandatory for offloaded playback. - */ - int (*pause)(struct audio_stream_out* stream); - - /** - * Notifies to the audio driver to resume playback following a pause. - * Returns error if called without matching pause. - * - * Implementation of this function is mandatory for offloaded playback. - */ - int (*resume)(struct audio_stream_out* stream); - - /** - * Requests notification when data buffered by the driver/hardware has - * been played. If set_callback() has previously been called to enable - * non-blocking mode, the drain() must not block, instead it should return - * quickly and completion of the drain is notified through the callback. - * If set_callback() has not been called, the drain() must block until - * completion. - * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written - * data has been played. - * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all - * data for the current track has played to allow time for the framework - * to perform a gapless track switch. - * - * Drain must return immediately on stop() and flush() call - * - * Implementation of this function is mandatory for offloaded playback. - */ - int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type ); - - /** - * Notifies to the audio driver to flush the queued data. Stream must already - * be paused before calling flush(). - * - * Implementation of this function is mandatory for offloaded playback. - */ - int (*flush)(struct audio_stream_out* stream); - - /** - * Return a recent count of the number of audio frames presented to an external observer. - * This excludes frames which have been written but are still in the pipeline. - * The count is not reset to zero when output enters standby. - * Also returns the value of CLOCK_MONOTONIC as of this presentation count. - * The returned count is expected to be 'recent', - * but does not need to be the most recent possible value. - * However, the associated time should correspond to whatever count is returned. - * Example: assume that N+M frames have been presented, where M is a 'small' number. - * Then it is permissible to return N instead of N+M, - * and the timestamp should correspond to N rather than N+M. - * The terms 'recent' and 'small' are not defined. - * They reflect the quality of the implementation. - * - * 3.0 and higher only. - */ - int (*get_presentation_position)(const struct audio_stream_out *stream, - uint64_t *frames, struct timespec *timestamp); - - /** - * Called by the framework to start a stream operating in mmap mode. - * create_mmap_buffer must be called before calling start() - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \return 0 in case of success. - * -ENOSYS if called out of sequence or on non mmap stream - */ - int (*start)(const struct audio_stream_out* stream); - - /** - * Called by the framework to stop a stream operating in mmap mode. - * Must be called after start() - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \return 0 in case of success. - * -ENOSYS if called out of sequence or on non mmap stream - */ - int (*stop)(const struct audio_stream_out* stream); - - /** - * Called by the framework to retrieve information on the mmap buffer used for audio - * samples transfer. - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \param[in] min_size_frames minimum buffer size requested. The actual buffer - * size returned in struct audio_mmap_buffer_info can be larger. - * \param[out] info address at which the mmap buffer information should be returned. - * - * \return 0 if the buffer was allocated. - * -ENODEV in case of initialization error - * -EINVAL if the requested buffer size is too large - * -ENOSYS if called out of sequence (e.g. buffer already allocated) - */ - int (*create_mmap_buffer)(const struct audio_stream_out *stream, - int32_t min_size_frames, - struct audio_mmap_buffer_info *info); - - /** - * Called by the framework to read current read/write position in the mmap buffer - * with associated time stamp. - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \param[out] position address at which the mmap read/write position should be returned. - * - * \return 0 if the position is successfully returned. - * -ENODATA if the position cannot be retrieved - * -ENOSYS if called before create_mmap_buffer() - */ - int (*get_mmap_position)(const struct audio_stream_out *stream, - struct audio_mmap_position *position); - - /** - * Called when the metadata of the stream's source has been changed. - * @param source_metadata Description of the audio that is played by the clients. - */ - void (*update_source_metadata)(struct audio_stream_out *stream, - const struct source_metadata* source_metadata); - - /** - * Set the callback function for notifying events for an output stream. - */ - int (*set_event_callback)(struct audio_stream_out *stream, - stream_event_callback_t callback, - void *cookie); - - /** - * Called when the metadata of the stream's source has been changed. - * HAL version 3.2 and higher only. - * @param source_metadata Description of the audio that is played by the clients. - */ - void (*update_source_metadata_v7)(struct audio_stream_out *stream, - const struct source_metadata_v7* source_metadata); - - /** - * Returns the Dual Mono mode presentation setting. - * - * \param[in] stream the stream object. - * \param[out] mode current setting of Dual Mono mode. - * - * \return 0 if the position is successfully returned. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*get_dual_mono_mode)(struct audio_stream_out *stream, audio_dual_mono_mode_t *mode); - - /** - * Sets the Dual Mono mode presentation on the output device. - * - * \param[in] stream the stream object. - * \param[in] mode selected Dual Mono mode. - * - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*set_dual_mono_mode)(struct audio_stream_out *stream, const audio_dual_mono_mode_t mode); - - /** - * Returns the Audio Description Mix level in dB. - * - * \param[in] stream the stream object. - * \param[out] leveldB the current Audio Description Mix Level in dB. - * - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*get_audio_description_mix_level)(struct audio_stream_out *stream, float *leveldB); - - /** - * Sets the Audio Description Mix level in dB. - * - * \param[in] stream the stream object. - * \param[in] leveldB Audio Description Mix Level in dB. - * - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*set_audio_description_mix_level)(struct audio_stream_out *stream, const float leveldB); - - /** - * Retrieves current playback rate parameters. - * - * \param[in] stream the stream object. - * \param[out] playbackRate current playback parameters. - * - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*get_playback_rate_parameters)(struct audio_stream_out *stream, - audio_playback_rate_t *playbackRate); - - /** - * Sets the playback rate parameters that control playback behavior. - * - * \param[in] stream the stream object. - * \param[in] playbackRate playback parameters. - * - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*set_playback_rate_parameters)(struct audio_stream_out *stream, - const audio_playback_rate_t *playbackRate); - - /** - * Indicates the requested latency mode for this output stream. - * - * The requested mode can be one of the modes returned by - * get_recommended_latency_modes(). - * - * Support for this method is optional but mandated on specific spatial audio - * streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can be routed - * to a BT classic sink. - * - * \param[in] stream the stream object. - * \param[in] mode the requested latency mode. - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*set_latency_mode)(struct audio_stream_out *stream, audio_latency_mode_t mode); - - /** - * Indicates which latency modes are currently supported on this output stream. - * If the transport protocol (e.g Bluetooth A2DP) used by this output stream to reach - * the output device supports variable latency modes, the HAL indicates which - * modes are currently supported. - * The framework can then call setLatencyMode() with one of the supported modes to select - * the desired operation mode. - * - * Support for this method is optional but mandated on specific spatial audio - * streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can be routed - * to a BT classic sink. - * - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - * \param[in] stream the stream object. - * \param[out] modes the supported latency modes. - * \param[in/out] num_modes as input the maximum number of modes to return, - * as output the actual number of modes returned. - */ - int (*get_recommended_latency_modes)(struct audio_stream_out *stream, - audio_latency_mode_t *modes, size_t *num_modes); - - /** - * Set the callback interface for notifying changes in supported latency modes. - * - * Calling this method with a null pointer will result in clearing a previously set callback. - * - * Support for this method is optional but mandated on specific spatial audio - * streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can be routed - * to a BT classic sink. - * - * \param[in] stream the stream object. - * \param[in] callback the registered callback or null to unregister. - * \param[in] cookie the context to pass when calling the callback. - * \return 0 in case of success. - * -EINVAL if the arguments are invalid - * -ENOSYS if the function is not available - */ - int (*set_latency_mode_callback)(struct audio_stream_out *stream, - stream_latency_mode_callback_t callback, void *cookie); -}; - -typedef struct audio_stream_out audio_stream_out_t; - -struct audio_stream_in { - /** - * Common methods of the audio stream in. This *must* be the first member of audio_stream_in - * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts - * where it's known the audio_stream references an audio_stream_in. - */ - struct audio_stream common; - - /** set the input gain for the audio driver. This method is for - * for future use */ - int (*set_gain)(struct audio_stream_in *stream, float gain); - - /** Read audio buffer in from audio driver. Returns number of bytes read, or a - * negative status_t. If at least one frame was read prior to the error, - * read should return that byte count and then return an error in the subsequent call. - */ - ssize_t (*read)(struct audio_stream_in *stream, void* buffer, - size_t bytes); - - /** - * Return the amount of input frames lost in the audio driver since the - * last call of this function. - * Audio driver is expected to reset the value to 0 and restart counting - * upon returning the current value by this function call. - * Such loss typically occurs when the user space process is blocked - * longer than the capacity of audio driver buffers. - * - * Unit: the number of input audio frames - */ - uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream); - - /** - * Return a recent count of the number of audio frames received and - * the clock time associated with that frame count. - * - * frames is the total frame count received. This should be as early in - * the capture pipeline as possible. In general, - * frames should be non-negative and should not go "backwards". - * - * time is the clock MONOTONIC time when frames was measured. In general, - * time should be a positive quantity and should not go "backwards". - * - * The status returned is 0 on success, -ENOSYS if the device is not - * ready/available, or -EINVAL if the arguments are null or otherwise invalid. - */ - int (*get_capture_position)(const struct audio_stream_in *stream, - int64_t *frames, int64_t *time); - - /** - * Called by the framework to start a stream operating in mmap mode. - * create_mmap_buffer must be called before calling start() - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \return 0 in case off success. - * -ENOSYS if called out of sequence or on non mmap stream - */ - int (*start)(const struct audio_stream_in* stream); - - /** - * Called by the framework to stop a stream operating in mmap mode. - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \return 0 in case of success. - * -ENOSYS if called out of sequence or on non mmap stream - */ - int (*stop)(const struct audio_stream_in* stream); - - /** - * Called by the framework to retrieve information on the mmap buffer used for audio - * samples transfer. - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \param[in] min_size_frames minimum buffer size requested. The actual buffer - * size returned in struct audio_mmap_buffer_info can be larger. - * \param[out] info address at which the mmap buffer information should be returned. - * - * \return 0 if the buffer was allocated. - * -ENODEV in case of initialization error - * -EINVAL if the requested buffer size is too large - * -ENOSYS if called out of sequence (e.g. buffer already allocated) - */ - int (*create_mmap_buffer)(const struct audio_stream_in *stream, - int32_t min_size_frames, - struct audio_mmap_buffer_info *info); - - /** - * Called by the framework to read current read/write position in the mmap buffer - * with associated time stamp. - * - * \note Function only implemented by streams operating in mmap mode. - * - * \param[in] stream the stream object. - * \param[out] position address at which the mmap read/write position should be returned. - * - * \return 0 if the position is successfully returned. - * -ENODATA if the position cannot be retreived - * -ENOSYS if called before mmap_read_position() - */ - int (*get_mmap_position)(const struct audio_stream_in *stream, - struct audio_mmap_position *position); - - /** - * Called by the framework to read active microphones - * - * \param[in] stream the stream object. - * \param[out] mic_array Pointer to first element on array with microphone info - * \param[out] mic_count When called, this holds the value of the max number of elements - * allowed in the mic_array. The actual number of elements written - * is returned here. - * if mic_count is passed as zero, mic_array will not be populated, - * and mic_count will return the actual number of active microphones. - * - * \return 0 if the microphone array is successfully filled. - * -ENOSYS if there is an error filling the data - */ - int (*get_active_microphones)(const struct audio_stream_in *stream, - struct audio_microphone_characteristic_t *mic_array, - size_t *mic_count); - - /** - * Called by the framework to instruct the HAL to optimize the capture stream in the - * specified direction. - * - * \param[in] stream the stream object. - * \param[in] direction The direction constant (from audio-base.h) - * MIC_DIRECTION_UNSPECIFIED Don't do any directionality processing of the - * activated microphone(s). - * MIC_DIRECTION_FRONT Optimize capture for audio coming from the screen-side - * of the device. - * MIC_DIRECTION_BACK Optimize capture for audio coming from the side of the - * device opposite the screen. - * MIC_DIRECTION_EXTERNAL Optimize capture for audio coming from an off-device - * microphone. - * \return OK if the call is successful, an error code otherwise. - */ - int (*set_microphone_direction)(const struct audio_stream_in *stream, - audio_microphone_direction_t direction); - - /** - * Called by the framework to specify to the HAL the desired zoom factor for the selected - * microphone(s). - * - * \param[in] stream the stream object. - * \param[in] zoom the zoom factor. - * \return OK if the call is successful, an error code otherwise. - */ - int (*set_microphone_field_dimension)(const struct audio_stream_in *stream, - float zoom); - - /** - * Called when the metadata of the stream's sink has been changed. - * @param sink_metadata Description of the audio that is recorded by the clients. - */ - void (*update_sink_metadata)(struct audio_stream_in *stream, - const struct sink_metadata* sink_metadata); - - /** - * Called when the metadata of the stream's sink has been changed. - * HAL version 3.2 and higher only. - * @param sink_metadata Description of the audio that is recorded by the clients. - */ - void (*update_sink_metadata_v7)(struct audio_stream_in *stream, - const struct sink_metadata_v7* sink_metadata); -}; -typedef struct audio_stream_in audio_stream_in_t; - -/** - * return the frame size (number of bytes per sample). - * - * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead. - */ -__attribute__((__deprecated__)) -static inline size_t audio_stream_frame_size(const struct audio_stream *s) -{ - size_t chan_samp_sz; - audio_format_t format = s->get_format(s); - - if (audio_has_proportional_frames(format)) { - chan_samp_sz = audio_bytes_per_sample(format); - return popcount(s->get_channels(s)) * chan_samp_sz; - } - - return sizeof(int8_t); -} - -/** - * return the frame size (number of bytes per sample) of an output stream. - */ -static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s) -{ - size_t chan_samp_sz; - audio_format_t format = s->common.get_format(&s->common); - - if (audio_has_proportional_frames(format)) { - chan_samp_sz = audio_bytes_per_sample(format); - return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz; - } - - return sizeof(int8_t); -} - -/** - * return the frame size (number of bytes per sample) of an input stream. - */ -static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s) -{ - size_t chan_samp_sz; - audio_format_t format = s->common.get_format(&s->common); - - if (audio_has_proportional_frames(format)) { - chan_samp_sz = audio_bytes_per_sample(format); - return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz; - } - - return sizeof(int8_t); -} - -/**********************************************************************/ - -/** - * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM - * and the fields of this data structure must begin with hw_module_t - * followed by module specific information. - */ -struct audio_module { - struct hw_module_t common; -}; - -struct audio_hw_device { - /** - * Common methods of the audio device. This *must* be the first member of audio_hw_device - * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts - * where it's known the hw_device_t references an audio_hw_device. - */ - struct hw_device_t common; - - /** - * used by audio flinger to enumerate what devices are supported by - * each audio_hw_device implementation. - * - * Return value is a bitmask of 1 or more values of audio_devices_t - * - * NOTE: audio HAL implementations starting with - * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function. - * All supported devices should be listed in audio_policy.conf - * file and the audio policy manager must choose the appropriate - * audio module based on information in this file. - */ - uint32_t (*get_supported_devices)(const struct audio_hw_device *dev); - - /** - * check to see if the audio hardware interface has been initialized. - * returns 0 on success, -ENODEV on failure. - */ - int (*init_check)(const struct audio_hw_device *dev); - - /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ - int (*set_voice_volume)(struct audio_hw_device *dev, float volume); - - /** - * set the audio volume for all audio activities other than voice call. - * Range between 0.0 and 1.0. If any value other than 0 is returned, - * the software mixer will emulate this capability. - */ - int (*set_master_volume)(struct audio_hw_device *dev, float volume); - - /** - * Get the current master volume value for the HAL, if the HAL supports - * master volume control. AudioFlinger will query this value from the - * primary audio HAL when the service starts and use the value for setting - * the initial master volume across all HALs. HALs which do not support - * this method may leave it set to NULL. - */ - int (*get_master_volume)(struct audio_hw_device *dev, float *volume); - - /** - * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode - * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is - * playing, and AUDIO_MODE_IN_CALL when a call is in progress. - */ - int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode); - - /* mic mute */ - int (*set_mic_mute)(struct audio_hw_device *dev, bool state); - int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state); - - /* set/get global audio parameters */ - int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs); - - /* - * Returns a pointer to a heap allocated string. The caller is responsible - * for freeing the memory for it using free(). - */ - char * (*get_parameters)(const struct audio_hw_device *dev, - const char *keys); - - /* Returns audio input buffer size according to parameters passed or - * 0 if one of the parameters is not supported. - * See also get_buffer_size which is for a particular stream. - */ - size_t (*get_input_buffer_size)(const struct audio_hw_device *dev, - const struct audio_config *config); - - /** This method creates and opens the audio hardware output stream. - * The "address" parameter qualifies the "devices" audio device type if needed. - * The format format depends on the device type: - * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC" - * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y" - * - Other devices may use a number or any other string. - */ - - int (*open_output_stream)(struct audio_hw_device *dev, - audio_io_handle_t handle, - audio_devices_t devices, - audio_output_flags_t flags, - struct audio_config *config, - struct audio_stream_out **stream_out, - const char *address); - - void (*close_output_stream)(struct audio_hw_device *dev, - struct audio_stream_out* stream_out); - - /** This method creates and opens the audio hardware input stream */ - int (*open_input_stream)(struct audio_hw_device *dev, - audio_io_handle_t handle, - audio_devices_t devices, - struct audio_config *config, - struct audio_stream_in **stream_in, - audio_input_flags_t flags, - const char *address, - audio_source_t source); - - void (*close_input_stream)(struct audio_hw_device *dev, - struct audio_stream_in *stream_in); - - /** - * Called by the framework to read available microphones characteristics. - * - * \param[in] dev the hw_device object. - * \param[out] mic_array Pointer to first element on array with microphone info - * \param[out] mic_count When called, this holds the value of the max number of elements - * allowed in the mic_array. The actual number of elements written - * is returned here. - * if mic_count is passed as zero, mic_array will not be populated, - * and mic_count will return the actual number of microphones in the - * system. - * - * \return 0 if the microphone array is successfully filled. - * -ENOSYS if there is an error filling the data - */ - int (*get_microphones)(const struct audio_hw_device *dev, - struct audio_microphone_characteristic_t *mic_array, - size_t *mic_count); - - /** This method dumps the state of the audio hardware */ - int (*dump)(const struct audio_hw_device *dev, int fd); - - /** - * set the audio mute status for all audio activities. If any value other - * than 0 is returned, the software mixer will emulate this capability. - */ - int (*set_master_mute)(struct audio_hw_device *dev, bool mute); - - /** - * Get the current master mute status for the HAL, if the HAL supports - * master mute control. AudioFlinger will query this value from the primary - * audio HAL when the service starts and use the value for setting the - * initial master mute across all HALs. HALs which do not support this - * method may leave it set to NULL. - */ - int (*get_master_mute)(struct audio_hw_device *dev, bool *mute); - - /** - * Routing control - */ - - /* Creates an audio patch between several source and sink ports. - * The handle is allocated by the HAL and should be unique for this - * audio HAL module. */ - int (*create_audio_patch)(struct audio_hw_device *dev, - unsigned int num_sources, - const struct audio_port_config *sources, - unsigned int num_sinks, - const struct audio_port_config *sinks, - audio_patch_handle_t *handle); - - /* Release an audio patch */ - int (*release_audio_patch)(struct audio_hw_device *dev, - audio_patch_handle_t handle); - - /* Fills the list of supported attributes for a given audio port. - * As input, "port" contains the information (type, role, address etc...) - * needed by the HAL to identify the port. - * As output, "port" contains possible attributes (sampling rates, formats, - * channel masks, gain controllers...) for this port. - */ - int (*get_audio_port)(struct audio_hw_device *dev, - struct audio_port *port); - - /* Set audio port configuration */ - int (*set_audio_port_config)(struct audio_hw_device *dev, - const struct audio_port_config *config); - - /** - * Applies an audio effect to an audio device. - * - * @param dev the audio HAL device context. - * @param device identifies the sink or source device the effect must be applied to. - * "device" is the audio_port_handle_t indicated for the device when - * the audio patch connecting that device was created. - * @param effect effect interface handle corresponding to the effect being added. - * @return retval operation completion status. - */ - int (*add_device_effect)(struct audio_hw_device *dev, - audio_port_handle_t device, effect_handle_t effect); - - /** - * Stops applying an audio effect to an audio device. - * - * @param dev the audio HAL device context. - * @param device identifies the sink or source device this effect was applied to. - * "device" is the audio_port_handle_t indicated for the device when - * the audio patch is created. - * @param effect effect interface handle corresponding to the effect being removed. - * @return retval operation completion status. - */ - int (*remove_device_effect)(struct audio_hw_device *dev, - audio_port_handle_t device, effect_handle_t effect); - - /** - * Fills the list of supported attributes for a given audio port. - * As input, "port" contains the information (type, role, address etc...) - * needed by the HAL to identify the port. - * As output, "port" contains possible attributes (sampling rates, formats, - * channel masks, gain controllers...) for this port. The possible attributes - * are saved as audio profiles, which contains audio format and the supported - * sampling rates and channel masks. - */ - int (*get_audio_port_v7)(struct audio_hw_device *dev, - struct audio_port_v7 *port); - - /** - * Called when the state of the connection of an external device has been changed. - * The "port" parameter is only used as input and besides identifying the device - * port, also may contain additional information such as extra audio descriptors. - * - * HAL version 3.2 and higher only. If the HAL does not implement this method, - * it must leave the function entry as null, or return -ENOSYS. In this case - * the framework will use 'set_parameters', which can only pass the device address. - * - * @param dev the audio HAL device context. - * @param port device port identification and extra information. - * @param connected whether the external device is connected. - * @return retval operation completion status. - */ - int (*set_device_connected_state_v7)(struct audio_hw_device *dev, - struct audio_port_v7 *port, - bool connected); -}; -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 +../../include_all/hardware/audio.h
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