diff --git a/audio/README b/audio/README index f4b85553ed..2b814506b4 100644 --- a/audio/README +++ b/audio/README @@ -7,18 +7,15 @@ audio |-- common <== code common to audio core and effect API | |-- 2.0 | | |-- default <== code that wraps the legacy API -| | |-- legacy <== legacy API compatible with 2.0 | | `-- vts <== vts of 2.0 core and effect API common code | |-- 4.0 | | |-- default -| | |-- legacy | | `-- vts | |-- ... <== The future versions should continue this structure | | |-- default | | `-- vts | `-- all_versions <== code common to all version of both core and effect API | |-- default -| | |-- legacy <== legacy API compatible with all versions | `-- vts <== vts of core and effect API common version independent code | |-- core <== code relative to the core API @@ -38,17 +35,13 @@ audio `-- effect <== idem for the effect API |-- 2.0 | |-- default - | |-- legacy <== legacy effect API compatible with 2.0 | `-- vts |-- 4.0 | |-- default - | |-- legacy | `-- vts |-- ... | |-- default - | |-- default | `-- vts `-- all_versions |-- default - |-- legacy `-- vts diff --git a/audio/common/2.0/default/Android.bp b/audio/common/2.0/default/Android.bp index 123f8b3656..ac66479c93 100644 --- a/audio/common/2.0/default/Android.bp +++ b/audio/common/2.0/default/Android.bp @@ -16,7 +16,10 @@ cc_library_shared { name: "android.hardware.audio.common@2.0-util", defaults: ["hidl_defaults"], - vendor: true, + vendor_available: true, + vndk: { + enabled: true, + }, srcs: [ "HidlUtils.cpp", ], @@ -38,7 +41,7 @@ cc_library_shared { ], header_libs: [ - "android.hardware.audio.common.legacy@2.0", + "libaudio_system_headers", "libhardware_headers", ], } diff --git a/audio/common/2.0/legacy/Android.bp b/audio/common/2.0/legacy/Android.bp deleted file mode 100644 index 2888c96e3c..0000000000 --- a/audio/common/2.0/legacy/Android.bp +++ /dev/null @@ -1,15 +0,0 @@ -cc_library_headers { - name: "android.hardware.audio.common.legacy@2.0", - vendor: true, - header_libs: [ - "libhardware_headers", - "android.hardware.audio.common.legacy@all-versions", - ], - export_header_lib_headers: [ - "libhardware_headers", - "android.hardware.audio.common.legacy@all-versions", - ], - - export_include_dirs: ["include"], -} - diff --git a/audio/common/2.0/legacy/OWNERS b/audio/common/2.0/legacy/OWNERS deleted file mode 100644 index 6fdc97ca29..0000000000 --- a/audio/common/2.0/legacy/OWNERS +++ /dev/null @@ -1,3 +0,0 @@ -elaurent@google.com -krocard@google.com -mnaganov@google.com diff --git a/audio/common/2.0/legacy/include/hardware/audio.h b/audio/common/2.0/legacy/include/hardware/audio.h deleted file mode 100644 index 1ad3e0e04e..0000000000 --- a/audio/common/2.0/legacy/include/hardware/audio.h +++ /dev/null @@ -1,709 +0,0 @@ -/* - * 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 -#include -#include -#include -#include - -#include - -#include -#include -#include - -__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_CURRENT AUDIO_DEVICE_API_VERSION_3_0 -/* 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" - -/** - * 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 int (*stream_callback_t)(stream_callback_event_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; - -/** - * 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); -}; -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); -}; -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 diff --git a/audio/common/2.0/legacy/include/system/audio-base.h b/audio/common/2.0/legacy/include/system/audio-base.h deleted file mode 100644 index 53e524b8f2..0000000000 --- a/audio/common/2.0/legacy/include/system/audio-base.h +++ /dev/null @@ -1,434 +0,0 @@ -// This file is autogenerated by hidl-gen. Do not edit manually. -// Source: android.hardware.audio.common@2.0 -// Root: android.hardware:hardware/interfaces - -#ifndef HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_ -#define HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_ - -#ifdef __cplusplus -extern "C" { -#endif - -enum { - AUDIO_IO_HANDLE_NONE = 0, - AUDIO_MODULE_HANDLE_NONE = 0, - AUDIO_PORT_HANDLE_NONE = 0, - AUDIO_PATCH_HANDLE_NONE = 0, -}; - -typedef enum { - AUDIO_STREAM_DEFAULT = -1, // (-1) - AUDIO_STREAM_MIN = 0, - AUDIO_STREAM_VOICE_CALL = 0, - AUDIO_STREAM_SYSTEM = 1, - AUDIO_STREAM_RING = 2, - AUDIO_STREAM_MUSIC = 3, - AUDIO_STREAM_ALARM = 4, - AUDIO_STREAM_NOTIFICATION = 5, - AUDIO_STREAM_BLUETOOTH_SCO = 6, - AUDIO_STREAM_ENFORCED_AUDIBLE = 7, - AUDIO_STREAM_DTMF = 8, - AUDIO_STREAM_TTS = 9, - AUDIO_STREAM_ACCESSIBILITY = 10, - AUDIO_STREAM_REROUTING = 11, - AUDIO_STREAM_PATCH = 12, - AUDIO_STREAM_PUBLIC_CNT = 11, // (ACCESSIBILITY + 1) - AUDIO_STREAM_FOR_POLICY_CNT = 12, // PATCH - AUDIO_STREAM_CNT = 13, // (PATCH + 1) -} audio_stream_type_t; - -typedef enum { - AUDIO_SOURCE_DEFAULT = 0, - AUDIO_SOURCE_MIC = 1, - AUDIO_SOURCE_VOICE_UPLINK = 2, - AUDIO_SOURCE_VOICE_DOWNLINK = 3, - AUDIO_SOURCE_VOICE_CALL = 4, - AUDIO_SOURCE_CAMCORDER = 5, - AUDIO_SOURCE_VOICE_RECOGNITION = 6, - AUDIO_SOURCE_VOICE_COMMUNICATION = 7, - AUDIO_SOURCE_REMOTE_SUBMIX = 8, - AUDIO_SOURCE_UNPROCESSED = 9, - AUDIO_SOURCE_CNT = 10, - AUDIO_SOURCE_MAX = 9, // (CNT - 1) - AUDIO_SOURCE_FM_TUNER = 1998, - AUDIO_SOURCE_HOTWORD = 1999, -} audio_source_t; - -typedef enum { - AUDIO_SESSION_OUTPUT_STAGE = -1, // (-1) - AUDIO_SESSION_OUTPUT_MIX = 0, - AUDIO_SESSION_ALLOCATE = 0, - AUDIO_SESSION_NONE = 0, -} audio_session_t; - -typedef enum { - AUDIO_FORMAT_INVALID = 4294967295u, // 0xFFFFFFFFUL - AUDIO_FORMAT_DEFAULT = 0u, // 0 - AUDIO_FORMAT_PCM = 0u, // 0x00000000UL - AUDIO_FORMAT_MP3 = 16777216u, // 0x01000000UL - AUDIO_FORMAT_AMR_NB = 33554432u, // 0x02000000UL - AUDIO_FORMAT_AMR_WB = 50331648u, // 0x03000000UL - AUDIO_FORMAT_AAC = 67108864u, // 0x04000000UL - AUDIO_FORMAT_HE_AAC_V1 = 83886080u, // 0x05000000UL - AUDIO_FORMAT_HE_AAC_V2 = 100663296u, // 0x06000000UL - AUDIO_FORMAT_VORBIS = 117440512u, // 0x07000000UL - AUDIO_FORMAT_OPUS = 134217728u, // 0x08000000UL - AUDIO_FORMAT_AC3 = 150994944u, // 0x09000000UL - AUDIO_FORMAT_E_AC3 = 167772160u, // 0x0A000000UL - AUDIO_FORMAT_DTS = 184549376u, // 0x0B000000UL - AUDIO_FORMAT_DTS_HD = 201326592u, // 0x0C000000UL - AUDIO_FORMAT_IEC61937 = 218103808u, // 0x0D000000UL - AUDIO_FORMAT_DOLBY_TRUEHD = 234881024u, // 0x0E000000UL - AUDIO_FORMAT_EVRC = 268435456u, // 0x10000000UL - AUDIO_FORMAT_EVRCB = 285212672u, // 0x11000000UL - AUDIO_FORMAT_EVRCWB = 301989888u, // 0x12000000UL - AUDIO_FORMAT_EVRCNW = 318767104u, // 0x13000000UL - AUDIO_FORMAT_AAC_ADIF = 335544320u, // 0x14000000UL - AUDIO_FORMAT_WMA = 352321536u, // 0x15000000UL - AUDIO_FORMAT_WMA_PRO = 369098752u, // 0x16000000UL - AUDIO_FORMAT_AMR_WB_PLUS = 385875968u, // 0x17000000UL - AUDIO_FORMAT_MP2 = 402653184u, // 0x18000000UL - AUDIO_FORMAT_QCELP = 419430400u, // 0x19000000UL - AUDIO_FORMAT_DSD = 436207616u, // 0x1A000000UL - AUDIO_FORMAT_FLAC = 452984832u, // 0x1B000000UL - AUDIO_FORMAT_ALAC = 469762048u, // 0x1C000000UL - AUDIO_FORMAT_APE = 486539264u, // 0x1D000000UL - AUDIO_FORMAT_AAC_ADTS = 503316480u, // 0x1E000000UL - AUDIO_FORMAT_SBC = 520093696u, // 0x1F000000UL - AUDIO_FORMAT_APTX = 536870912u, // 0x20000000UL - AUDIO_FORMAT_APTX_HD = 553648128u, // 0x21000000UL - AUDIO_FORMAT_AC4 = 570425344u, // 0x22000000UL - AUDIO_FORMAT_LDAC = 587202560u, // 0x23000000UL - AUDIO_FORMAT_MAIN_MASK = 4278190080u, // 0xFF000000UL - AUDIO_FORMAT_SUB_MASK = 16777215u, // 0x00FFFFFFUL - AUDIO_FORMAT_PCM_SUB_16_BIT = 1u, // 0x1 - AUDIO_FORMAT_PCM_SUB_8_BIT = 2u, // 0x2 - AUDIO_FORMAT_PCM_SUB_32_BIT = 3u, // 0x3 - AUDIO_FORMAT_PCM_SUB_8_24_BIT = 4u, // 0x4 - AUDIO_FORMAT_PCM_SUB_FLOAT = 5u, // 0x5 - AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED = 6u, // 0x6 - AUDIO_FORMAT_MP3_SUB_NONE = 0u, // 0x0 - AUDIO_FORMAT_AMR_SUB_NONE = 0u, // 0x0 - AUDIO_FORMAT_AAC_SUB_MAIN = 1u, // 0x1 - AUDIO_FORMAT_AAC_SUB_LC = 2u, // 0x2 - AUDIO_FORMAT_AAC_SUB_SSR = 4u, // 0x4 - AUDIO_FORMAT_AAC_SUB_LTP = 8u, // 0x8 - AUDIO_FORMAT_AAC_SUB_HE_V1 = 16u, // 0x10 - AUDIO_FORMAT_AAC_SUB_SCALABLE = 32u, // 0x20 - AUDIO_FORMAT_AAC_SUB_ERLC = 64u, // 0x40 - AUDIO_FORMAT_AAC_SUB_LD = 128u, // 0x80 - AUDIO_FORMAT_AAC_SUB_HE_V2 = 256u, // 0x100 - AUDIO_FORMAT_AAC_SUB_ELD = 512u, // 0x200 - AUDIO_FORMAT_VORBIS_SUB_NONE = 0u, // 0x0 - AUDIO_FORMAT_PCM_16_BIT = 1u, // (PCM | PCM_SUB_16_BIT) - AUDIO_FORMAT_PCM_8_BIT = 2u, // (PCM | PCM_SUB_8_BIT) - AUDIO_FORMAT_PCM_32_BIT = 3u, // (PCM | PCM_SUB_32_BIT) - AUDIO_FORMAT_PCM_8_24_BIT = 4u, // (PCM | PCM_SUB_8_24_BIT) - AUDIO_FORMAT_PCM_FLOAT = 5u, // (PCM | PCM_SUB_FLOAT) - AUDIO_FORMAT_PCM_24_BIT_PACKED = 6u, // (PCM | PCM_SUB_24_BIT_PACKED) - AUDIO_FORMAT_AAC_MAIN = 67108865u, // (AAC | AAC_SUB_MAIN) - AUDIO_FORMAT_AAC_LC = 67108866u, // (AAC | AAC_SUB_LC) - AUDIO_FORMAT_AAC_SSR = 67108868u, // (AAC | AAC_SUB_SSR) - AUDIO_FORMAT_AAC_LTP = 67108872u, // (AAC | AAC_SUB_LTP) - AUDIO_FORMAT_AAC_HE_V1 = 67108880u, // (AAC | AAC_SUB_HE_V1) - AUDIO_FORMAT_AAC_SCALABLE = 67108896u, // (AAC | AAC_SUB_SCALABLE) - AUDIO_FORMAT_AAC_ERLC = 67108928u, // (AAC | AAC_SUB_ERLC) - AUDIO_FORMAT_AAC_LD = 67108992u, // (AAC | AAC_SUB_LD) - AUDIO_FORMAT_AAC_HE_V2 = 67109120u, // (AAC | AAC_SUB_HE_V2) - AUDIO_FORMAT_AAC_ELD = 67109376u, // (AAC | AAC_SUB_ELD) - AUDIO_FORMAT_AAC_ADTS_MAIN = 503316481u, // (AAC_ADTS | AAC_SUB_MAIN) - AUDIO_FORMAT_AAC_ADTS_LC = 503316482u, // (AAC_ADTS | AAC_SUB_LC) - AUDIO_FORMAT_AAC_ADTS_SSR = 503316484u, // (AAC_ADTS | AAC_SUB_SSR) - AUDIO_FORMAT_AAC_ADTS_LTP = 503316488u, // (AAC_ADTS | AAC_SUB_LTP) - AUDIO_FORMAT_AAC_ADTS_HE_V1 = 503316496u, // (AAC_ADTS | AAC_SUB_HE_V1) - AUDIO_FORMAT_AAC_ADTS_SCALABLE = 503316512u, // (AAC_ADTS | AAC_SUB_SCALABLE) - AUDIO_FORMAT_AAC_ADTS_ERLC = 503316544u, // (AAC_ADTS | AAC_SUB_ERLC) - AUDIO_FORMAT_AAC_ADTS_LD = 503316608u, // (AAC_ADTS | AAC_SUB_LD) - AUDIO_FORMAT_AAC_ADTS_HE_V2 = 503316736u, // (AAC_ADTS | AAC_SUB_HE_V2) - AUDIO_FORMAT_AAC_ADTS_ELD = 503316992u, // (AAC_ADTS | AAC_SUB_ELD) -} audio_format_t; - -enum { - FCC_2 = 2, - FCC_8 = 8, -}; - -enum { - AUDIO_CHANNEL_REPRESENTATION_POSITION = 0u, // 0 - AUDIO_CHANNEL_REPRESENTATION_INDEX = 2u, // 2 - AUDIO_CHANNEL_NONE = 0u, // 0x0 - AUDIO_CHANNEL_INVALID = 3221225472u, // 0xC0000000 - AUDIO_CHANNEL_OUT_FRONT_LEFT = 1u, // 0x1 - AUDIO_CHANNEL_OUT_FRONT_RIGHT = 2u, // 0x2 - AUDIO_CHANNEL_OUT_FRONT_CENTER = 4u, // 0x4 - AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 8u, // 0x8 - AUDIO_CHANNEL_OUT_BACK_LEFT = 16u, // 0x10 - AUDIO_CHANNEL_OUT_BACK_RIGHT = 32u, // 0x20 - AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 64u, // 0x40 - AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 128u, // 0x80 - AUDIO_CHANNEL_OUT_BACK_CENTER = 256u, // 0x100 - AUDIO_CHANNEL_OUT_SIDE_LEFT = 512u, // 0x200 - AUDIO_CHANNEL_OUT_SIDE_RIGHT = 1024u, // 0x400 - AUDIO_CHANNEL_OUT_TOP_CENTER = 2048u, // 0x800 - AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 4096u, // 0x1000 - AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 8192u, // 0x2000 - AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 16384u, // 0x4000 - AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 32768u, // 0x8000 - AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 65536u, // 0x10000 - AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 131072u, // 0x20000 - AUDIO_CHANNEL_OUT_MONO = 1u, // OUT_FRONT_LEFT - AUDIO_CHANNEL_OUT_STEREO = 3u, // (OUT_FRONT_LEFT | OUT_FRONT_RIGHT) - AUDIO_CHANNEL_OUT_2POINT1 = 11u, // ((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_LOW_FREQUENCY) - AUDIO_CHANNEL_OUT_QUAD = - 51u, // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_BACK_LEFT) | OUT_BACK_RIGHT) - AUDIO_CHANNEL_OUT_QUAD_BACK = 51u, // OUT_QUAD - AUDIO_CHANNEL_OUT_QUAD_SIDE = - 1539u, // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_SIDE_LEFT) | OUT_SIDE_RIGHT) - AUDIO_CHANNEL_OUT_SURROUND = - 263u, // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) | OUT_BACK_CENTER) - AUDIO_CHANNEL_OUT_PENTA = 55u, // (OUT_QUAD | OUT_FRONT_CENTER) - AUDIO_CHANNEL_OUT_5POINT1 = 63u, // (((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) - // | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | OUT_BACK_RIGHT) - AUDIO_CHANNEL_OUT_5POINT1_BACK = 63u, // OUT_5POINT1 - AUDIO_CHANNEL_OUT_5POINT1_SIDE = 1551u, // (((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | - // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | - // OUT_SIDE_LEFT) | OUT_SIDE_RIGHT) - AUDIO_CHANNEL_OUT_6POINT1 = 319u, // ((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | - // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | - // OUT_BACK_RIGHT) | OUT_BACK_CENTER) - AUDIO_CHANNEL_OUT_7POINT1 = 1599u, // (((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | - // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | - // OUT_BACK_RIGHT) | OUT_SIDE_LEFT) | OUT_SIDE_RIGHT) - AUDIO_CHANNEL_OUT_ALL = - 262143u, // (((((((((((((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) | - // OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | OUT_BACK_RIGHT) | - // OUT_FRONT_LEFT_OF_CENTER) | OUT_FRONT_RIGHT_OF_CENTER) | OUT_BACK_CENTER) | - // OUT_SIDE_LEFT) | OUT_SIDE_RIGHT) | OUT_TOP_CENTER) | OUT_TOP_FRONT_LEFT) | - // OUT_TOP_FRONT_CENTER) | OUT_TOP_FRONT_RIGHT) | OUT_TOP_BACK_LEFT) | - // OUT_TOP_BACK_CENTER) | OUT_TOP_BACK_RIGHT) - AUDIO_CHANNEL_IN_LEFT = 4u, // 0x4 - AUDIO_CHANNEL_IN_RIGHT = 8u, // 0x8 - AUDIO_CHANNEL_IN_FRONT = 16u, // 0x10 - AUDIO_CHANNEL_IN_BACK = 32u, // 0x20 - AUDIO_CHANNEL_IN_LEFT_PROCESSED = 64u, // 0x40 - AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 128u, // 0x80 - AUDIO_CHANNEL_IN_FRONT_PROCESSED = 256u, // 0x100 - AUDIO_CHANNEL_IN_BACK_PROCESSED = 512u, // 0x200 - AUDIO_CHANNEL_IN_PRESSURE = 1024u, // 0x400 - AUDIO_CHANNEL_IN_X_AXIS = 2048u, // 0x800 - AUDIO_CHANNEL_IN_Y_AXIS = 4096u, // 0x1000 - AUDIO_CHANNEL_IN_Z_AXIS = 8192u, // 0x2000 - AUDIO_CHANNEL_IN_VOICE_UPLINK = 16384u, // 0x4000 - AUDIO_CHANNEL_IN_VOICE_DNLINK = 32768u, // 0x8000 - AUDIO_CHANNEL_IN_MONO = 16u, // IN_FRONT - AUDIO_CHANNEL_IN_STEREO = 12u, // (IN_LEFT | IN_RIGHT) - AUDIO_CHANNEL_IN_FRONT_BACK = 48u, // (IN_FRONT | IN_BACK) - AUDIO_CHANNEL_IN_6 = 252u, // (((((IN_LEFT | IN_RIGHT) | IN_FRONT) | IN_BACK) | - // IN_LEFT_PROCESSED) | IN_RIGHT_PROCESSED) - AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO = 16400u, // (IN_VOICE_UPLINK | IN_MONO) - AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO = 32784u, // (IN_VOICE_DNLINK | IN_MONO) - AUDIO_CHANNEL_IN_VOICE_CALL_MONO = 49168u, // (IN_VOICE_UPLINK_MONO | IN_VOICE_DNLINK_MONO) - AUDIO_CHANNEL_IN_ALL = - 65532u, // (((((((((((((IN_LEFT | IN_RIGHT) | IN_FRONT) | IN_BACK) | IN_LEFT_PROCESSED) | - // IN_RIGHT_PROCESSED) | IN_FRONT_PROCESSED) | IN_BACK_PROCESSED) | IN_PRESSURE) | - // IN_X_AXIS) | IN_Y_AXIS) | IN_Z_AXIS) | IN_VOICE_UPLINK) | IN_VOICE_DNLINK) - AUDIO_CHANNEL_COUNT_MAX = 30u, // 30 - AUDIO_CHANNEL_INDEX_HDR = 2147483648u, // (REPRESENTATION_INDEX << COUNT_MAX) - AUDIO_CHANNEL_INDEX_MASK_1 = 2147483649u, // (INDEX_HDR | ((1 << 1) - 1)) - AUDIO_CHANNEL_INDEX_MASK_2 = 2147483651u, // (INDEX_HDR | ((1 << 2) - 1)) - AUDIO_CHANNEL_INDEX_MASK_3 = 2147483655u, // (INDEX_HDR | ((1 << 3) - 1)) - AUDIO_CHANNEL_INDEX_MASK_4 = 2147483663u, // (INDEX_HDR | ((1 << 4) - 1)) - AUDIO_CHANNEL_INDEX_MASK_5 = 2147483679u, // (INDEX_HDR | ((1 << 5) - 1)) - AUDIO_CHANNEL_INDEX_MASK_6 = 2147483711u, // (INDEX_HDR | ((1 << 6) - 1)) - AUDIO_CHANNEL_INDEX_MASK_7 = 2147483775u, // (INDEX_HDR | ((1 << 7) - 1)) - AUDIO_CHANNEL_INDEX_MASK_8 = 2147483903u, // (INDEX_HDR | ((1 << 8) - 1)) -}; - -enum { - AUDIO_INTERLEAVE_LEFT = 0, - AUDIO_INTERLEAVE_RIGHT = 1, -}; - -typedef enum { - AUDIO_MODE_INVALID = -2, // (-2) - AUDIO_MODE_CURRENT = -1, // (-1) - AUDIO_MODE_NORMAL = 0, - AUDIO_MODE_RINGTONE = 1, - AUDIO_MODE_IN_CALL = 2, - AUDIO_MODE_IN_COMMUNICATION = 3, - AUDIO_MODE_CNT = 4, - AUDIO_MODE_MAX = 3, // (CNT - 1) -} audio_mode_t; - -enum { - AUDIO_DEVICE_NONE = 0u, // 0x0 - AUDIO_DEVICE_BIT_IN = 2147483648u, // 0x80000000 - AUDIO_DEVICE_BIT_DEFAULT = 1073741824u, // 0x40000000 - AUDIO_DEVICE_OUT_EARPIECE = 1u, // 0x1 - AUDIO_DEVICE_OUT_SPEAKER = 2u, // 0x2 - AUDIO_DEVICE_OUT_WIRED_HEADSET = 4u, // 0x4 - AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 8u, // 0x8 - AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 16u, // 0x10 - AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 32u, // 0x20 - AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 64u, // 0x40 - AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 128u, // 0x80 - AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 256u, // 0x100 - AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 512u, // 0x200 - AUDIO_DEVICE_OUT_AUX_DIGITAL = 1024u, // 0x400 - AUDIO_DEVICE_OUT_HDMI = 1024u, // OUT_AUX_DIGITAL - AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 2048u, // 0x800 - AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 4096u, // 0x1000 - AUDIO_DEVICE_OUT_USB_ACCESSORY = 8192u, // 0x2000 - AUDIO_DEVICE_OUT_USB_DEVICE = 16384u, // 0x4000 - AUDIO_DEVICE_OUT_REMOTE_SUBMIX = 32768u, // 0x8000 - AUDIO_DEVICE_OUT_TELEPHONY_TX = 65536u, // 0x10000 - AUDIO_DEVICE_OUT_LINE = 131072u, // 0x20000 - AUDIO_DEVICE_OUT_HDMI_ARC = 262144u, // 0x40000 - AUDIO_DEVICE_OUT_SPDIF = 524288u, // 0x80000 - AUDIO_DEVICE_OUT_FM = 1048576u, // 0x100000 - AUDIO_DEVICE_OUT_AUX_LINE = 2097152u, // 0x200000 - AUDIO_DEVICE_OUT_SPEAKER_SAFE = 4194304u, // 0x400000 - AUDIO_DEVICE_OUT_IP = 8388608u, // 0x800000 - AUDIO_DEVICE_OUT_BUS = 16777216u, // 0x1000000 - AUDIO_DEVICE_OUT_PROXY = 33554432u, // 0x2000000 - AUDIO_DEVICE_OUT_USB_HEADSET = 67108864u, // 0x4000000 - AUDIO_DEVICE_OUT_DEFAULT = 1073741824u, // BIT_DEFAULT - AUDIO_DEVICE_OUT_ALL = - 1207959551u, // (((((((((((((((((((((((((((OUT_EARPIECE | OUT_SPEAKER) | OUT_WIRED_HEADSET) - // | OUT_WIRED_HEADPHONE) | OUT_BLUETOOTH_SCO) | OUT_BLUETOOTH_SCO_HEADSET) | - // OUT_BLUETOOTH_SCO_CARKIT) | OUT_BLUETOOTH_A2DP) | - // OUT_BLUETOOTH_A2DP_HEADPHONES) | OUT_BLUETOOTH_A2DP_SPEAKER) | OUT_HDMI) | - // OUT_ANLG_DOCK_HEADSET) | OUT_DGTL_DOCK_HEADSET) | OUT_USB_ACCESSORY) | - // OUT_USB_DEVICE) | OUT_REMOTE_SUBMIX) | OUT_TELEPHONY_TX) | OUT_LINE) | - // OUT_HDMI_ARC) | OUT_SPDIF) | OUT_FM) | OUT_AUX_LINE) | OUT_SPEAKER_SAFE) | - // OUT_IP) | OUT_BUS) | OUT_PROXY) | OUT_USB_HEADSET) | OUT_DEFAULT) - AUDIO_DEVICE_OUT_ALL_A2DP = 896u, // ((OUT_BLUETOOTH_A2DP | OUT_BLUETOOTH_A2DP_HEADPHONES) | - // OUT_BLUETOOTH_A2DP_SPEAKER) - AUDIO_DEVICE_OUT_ALL_SCO = - 112u, // ((OUT_BLUETOOTH_SCO | OUT_BLUETOOTH_SCO_HEADSET) | OUT_BLUETOOTH_SCO_CARKIT) - AUDIO_DEVICE_OUT_ALL_USB = - 67133440u, // ((OUT_USB_ACCESSORY | OUT_USB_DEVICE) | OUT_USB_HEADSET) - AUDIO_DEVICE_IN_COMMUNICATION = 2147483649u, // (BIT_IN | 0x1) - AUDIO_DEVICE_IN_AMBIENT = 2147483650u, // (BIT_IN | 0x2) - AUDIO_DEVICE_IN_BUILTIN_MIC = 2147483652u, // (BIT_IN | 0x4) - AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = 2147483656u, // (BIT_IN | 0x8) - AUDIO_DEVICE_IN_WIRED_HEADSET = 2147483664u, // (BIT_IN | 0x10) - AUDIO_DEVICE_IN_AUX_DIGITAL = 2147483680u, // (BIT_IN | 0x20) - AUDIO_DEVICE_IN_HDMI = 2147483680u, // IN_AUX_DIGITAL - AUDIO_DEVICE_IN_VOICE_CALL = 2147483712u, // (BIT_IN | 0x40) - AUDIO_DEVICE_IN_TELEPHONY_RX = 2147483712u, // IN_VOICE_CALL - AUDIO_DEVICE_IN_BACK_MIC = 2147483776u, // (BIT_IN | 0x80) - AUDIO_DEVICE_IN_REMOTE_SUBMIX = 2147483904u, // (BIT_IN | 0x100) - AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET = 2147484160u, // (BIT_IN | 0x200) - AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET = 2147484672u, // (BIT_IN | 0x400) - AUDIO_DEVICE_IN_USB_ACCESSORY = 2147485696u, // (BIT_IN | 0x800) - AUDIO_DEVICE_IN_USB_DEVICE = 2147487744u, // (BIT_IN | 0x1000) - AUDIO_DEVICE_IN_FM_TUNER = 2147491840u, // (BIT_IN | 0x2000) - AUDIO_DEVICE_IN_TV_TUNER = 2147500032u, // (BIT_IN | 0x4000) - AUDIO_DEVICE_IN_LINE = 2147516416u, // (BIT_IN | 0x8000) - AUDIO_DEVICE_IN_SPDIF = 2147549184u, // (BIT_IN | 0x10000) - AUDIO_DEVICE_IN_BLUETOOTH_A2DP = 2147614720u, // (BIT_IN | 0x20000) - AUDIO_DEVICE_IN_LOOPBACK = 2147745792u, // (BIT_IN | 0x40000) - AUDIO_DEVICE_IN_IP = 2148007936u, // (BIT_IN | 0x80000) - AUDIO_DEVICE_IN_BUS = 2148532224u, // (BIT_IN | 0x100000) - AUDIO_DEVICE_IN_PROXY = 2164260864u, // (BIT_IN | 0x1000000) - AUDIO_DEVICE_IN_USB_HEADSET = 2181038080u, // (BIT_IN | 0x2000000) - AUDIO_DEVICE_IN_DEFAULT = 3221225472u, // (BIT_IN | BIT_DEFAULT) - AUDIO_DEVICE_IN_ALL = - 3273654271u, // (((((((((((((((((((((((IN_COMMUNICATION | IN_AMBIENT) | IN_BUILTIN_MIC) | - // IN_BLUETOOTH_SCO_HEADSET) | IN_WIRED_HEADSET) | IN_HDMI) | IN_TELEPHONY_RX) - // | IN_BACK_MIC) | IN_REMOTE_SUBMIX) | IN_ANLG_DOCK_HEADSET) | - // IN_DGTL_DOCK_HEADSET) | IN_USB_ACCESSORY) | IN_USB_DEVICE) | IN_FM_TUNER) | - // IN_TV_TUNER) | IN_LINE) | IN_SPDIF) | IN_BLUETOOTH_A2DP) | IN_LOOPBACK) | - // IN_IP) | IN_BUS) | IN_PROXY) | IN_USB_HEADSET) | IN_DEFAULT) - AUDIO_DEVICE_IN_ALL_SCO = 2147483656u, // IN_BLUETOOTH_SCO_HEADSET - AUDIO_DEVICE_IN_ALL_USB = 2181044224u, // ((IN_USB_ACCESSORY | IN_USB_DEVICE) | IN_USB_HEADSET) -}; - -typedef enum { - AUDIO_OUTPUT_FLAG_NONE = 0, // 0x0 - AUDIO_OUTPUT_FLAG_DIRECT = 1, // 0x1 - AUDIO_OUTPUT_FLAG_PRIMARY = 2, // 0x2 - AUDIO_OUTPUT_FLAG_FAST = 4, // 0x4 - AUDIO_OUTPUT_FLAG_DEEP_BUFFER = 8, // 0x8 - AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD = 16, // 0x10 - AUDIO_OUTPUT_FLAG_NON_BLOCKING = 32, // 0x20 - AUDIO_OUTPUT_FLAG_HW_AV_SYNC = 64, // 0x40 - AUDIO_OUTPUT_FLAG_TTS = 128, // 0x80 - AUDIO_OUTPUT_FLAG_RAW = 256, // 0x100 - AUDIO_OUTPUT_FLAG_SYNC = 512, // 0x200 - AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO = 1024, // 0x400 - AUDIO_OUTPUT_FLAG_DIRECT_PCM = 8192, // 0x2000 - AUDIO_OUTPUT_FLAG_MMAP_NOIRQ = 16384, // 0x4000 - AUDIO_OUTPUT_FLAG_VOIP_RX = 32768, // 0x8000 -} audio_output_flags_t; - -typedef enum { - AUDIO_INPUT_FLAG_NONE = 0, // 0x0 - AUDIO_INPUT_FLAG_FAST = 1, // 0x1 - AUDIO_INPUT_FLAG_HW_HOTWORD = 2, // 0x2 - AUDIO_INPUT_FLAG_RAW = 4, // 0x4 - AUDIO_INPUT_FLAG_SYNC = 8, // 0x8 - AUDIO_INPUT_FLAG_MMAP_NOIRQ = 16, // 0x10 - AUDIO_INPUT_FLAG_VOIP_TX = 32, // 0x20 -} audio_input_flags_t; - -typedef enum { - AUDIO_USAGE_UNKNOWN = 0, - AUDIO_USAGE_MEDIA = 1, - AUDIO_USAGE_VOICE_COMMUNICATION = 2, - AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3, - AUDIO_USAGE_ALARM = 4, - AUDIO_USAGE_NOTIFICATION = 5, - AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE = 6, - AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST = 7, - AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT = 8, - AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED = 9, - AUDIO_USAGE_NOTIFICATION_EVENT = 10, - AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11, - AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12, - AUDIO_USAGE_ASSISTANCE_SONIFICATION = 13, - AUDIO_USAGE_GAME = 14, - AUDIO_USAGE_VIRTUAL_SOURCE = 15, - AUDIO_USAGE_ASSISTANT = 16, - AUDIO_USAGE_CNT = 17, - AUDIO_USAGE_MAX = 16, // (CNT - 1) -} audio_usage_t; - -enum { - AUDIO_GAIN_MODE_JOINT = 1u, // 0x1 - AUDIO_GAIN_MODE_CHANNELS = 2u, // 0x2 - AUDIO_GAIN_MODE_RAMP = 4u, // 0x4 -}; - -typedef enum { - AUDIO_PORT_ROLE_NONE = 0, - AUDIO_PORT_ROLE_SOURCE = 1, - AUDIO_PORT_ROLE_SINK = 2, -} audio_port_role_t; - -typedef enum { - AUDIO_PORT_TYPE_NONE = 0, - AUDIO_PORT_TYPE_DEVICE = 1, - AUDIO_PORT_TYPE_MIX = 2, - AUDIO_PORT_TYPE_SESSION = 3, -} audio_port_type_t; - -enum { - AUDIO_PORT_CONFIG_SAMPLE_RATE = 1u, // 0x1 - AUDIO_PORT_CONFIG_CHANNEL_MASK = 2u, // 0x2 - AUDIO_PORT_CONFIG_FORMAT = 4u, // 0x4 - AUDIO_PORT_CONFIG_GAIN = 8u, // 0x8 - AUDIO_PORT_CONFIG_ALL = 15u, // (((SAMPLE_RATE | CHANNEL_MASK) | FORMAT) | GAIN) -}; - -typedef enum { - AUDIO_LATENCY_LOW = 0, - AUDIO_LATENCY_NORMAL = 1, -} audio_mix_latency_class_t; - -#ifdef __cplusplus -} -#endif - -#endif // HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_ diff --git a/audio/common/2.0/legacy/include/system/audio_effect-base.h b/audio/common/2.0/legacy/include/system/audio_effect-base.h deleted file mode 100644 index cd17f5529a..0000000000 --- a/audio/common/2.0/legacy/include/system/audio_effect-base.h +++ /dev/null @@ -1,101 +0,0 @@ -// This file is autogenerated by hidl-gen. Do not edit manually. -// Source: android.hardware.audio.effect@2.0 -// Root: android.hardware:hardware/interfaces - -#ifndef HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_ -#define HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_ - -#ifdef __cplusplus -extern "C" { -#endif - -enum { - EFFECT_FLAG_TYPE_SHIFT = 0, - EFFECT_FLAG_TYPE_SIZE = 3, - EFFECT_FLAG_TYPE_MASK = 7, // (((1 << TYPE_SIZE) - 1) << TYPE_SHIFT) - EFFECT_FLAG_TYPE_INSERT = 0, // (0 << TYPE_SHIFT) - EFFECT_FLAG_TYPE_AUXILIARY = 1, // (1 << TYPE_SHIFT) - EFFECT_FLAG_TYPE_REPLACE = 2, // (2 << TYPE_SHIFT) - EFFECT_FLAG_TYPE_PRE_PROC = 3, // (3 << TYPE_SHIFT) - EFFECT_FLAG_TYPE_POST_PROC = 4, // (4 << TYPE_SHIFT) - EFFECT_FLAG_INSERT_SHIFT = 3, // (TYPE_SHIFT + TYPE_SIZE) - EFFECT_FLAG_INSERT_SIZE = 3, - EFFECT_FLAG_INSERT_MASK = 56, // (((1 << INSERT_SIZE) - 1) << INSERT_SHIFT) - EFFECT_FLAG_INSERT_ANY = 0, // (0 << INSERT_SHIFT) - EFFECT_FLAG_INSERT_FIRST = 8, // (1 << INSERT_SHIFT) - EFFECT_FLAG_INSERT_LAST = 16, // (2 << INSERT_SHIFT) - EFFECT_FLAG_INSERT_EXCLUSIVE = 24, // (3 << INSERT_SHIFT) - EFFECT_FLAG_VOLUME_SHIFT = 6, // (INSERT_SHIFT + INSERT_SIZE) - EFFECT_FLAG_VOLUME_SIZE = 3, - EFFECT_FLAG_VOLUME_MASK = 448, // (((1 << VOLUME_SIZE) - 1) << VOLUME_SHIFT) - EFFECT_FLAG_VOLUME_CTRL = 64, // (1 << VOLUME_SHIFT) - EFFECT_FLAG_VOLUME_IND = 128, // (2 << VOLUME_SHIFT) - EFFECT_FLAG_VOLUME_NONE = 0, // (0 << VOLUME_SHIFT) - EFFECT_FLAG_DEVICE_SHIFT = 9, // (VOLUME_SHIFT + VOLUME_SIZE) - EFFECT_FLAG_DEVICE_SIZE = 3, - EFFECT_FLAG_DEVICE_MASK = 3584, // (((1 << DEVICE_SIZE) - 1) << DEVICE_SHIFT) - EFFECT_FLAG_DEVICE_IND = 512, // (1 << DEVICE_SHIFT) - EFFECT_FLAG_DEVICE_NONE = 0, // (0 << DEVICE_SHIFT) - EFFECT_FLAG_INPUT_SHIFT = 12, // (DEVICE_SHIFT + DEVICE_SIZE) - EFFECT_FLAG_INPUT_SIZE = 2, - EFFECT_FLAG_INPUT_MASK = 12288, // (((1 << INPUT_SIZE) - 1) << INPUT_SHIFT) - EFFECT_FLAG_INPUT_DIRECT = 4096, // (1 << INPUT_SHIFT) - EFFECT_FLAG_INPUT_PROVIDER = 8192, // (2 << INPUT_SHIFT) - EFFECT_FLAG_INPUT_BOTH = 12288, // (3 << INPUT_SHIFT) - EFFECT_FLAG_OUTPUT_SHIFT = 14, // (INPUT_SHIFT + INPUT_SIZE) - EFFECT_FLAG_OUTPUT_SIZE = 2, - EFFECT_FLAG_OUTPUT_MASK = 49152, // (((1 << OUTPUT_SIZE) - 1) << OUTPUT_SHIFT) - EFFECT_FLAG_OUTPUT_DIRECT = 16384, // (1 << OUTPUT_SHIFT) - EFFECT_FLAG_OUTPUT_PROVIDER = 32768, // (2 << OUTPUT_SHIFT) - EFFECT_FLAG_OUTPUT_BOTH = 49152, // (3 << OUTPUT_SHIFT) - EFFECT_FLAG_HW_ACC_SHIFT = 16, // (OUTPUT_SHIFT + OUTPUT_SIZE) - EFFECT_FLAG_HW_ACC_SIZE = 2, - EFFECT_FLAG_HW_ACC_MASK = 196608, // (((1 << HW_ACC_SIZE) - 1) << HW_ACC_SHIFT) - EFFECT_FLAG_HW_ACC_SIMPLE = 65536, // (1 << HW_ACC_SHIFT) - EFFECT_FLAG_HW_ACC_TUNNEL = 131072, // (2 << HW_ACC_SHIFT) - EFFECT_FLAG_AUDIO_MODE_SHIFT = 18, // (HW_ACC_SHIFT + HW_ACC_SIZE) - EFFECT_FLAG_AUDIO_MODE_SIZE = 2, - EFFECT_FLAG_AUDIO_MODE_MASK = 786432, // (((1 << AUDIO_MODE_SIZE) - 1) << AUDIO_MODE_SHIFT) - EFFECT_FLAG_AUDIO_MODE_IND = 262144, // (1 << AUDIO_MODE_SHIFT) - EFFECT_FLAG_AUDIO_MODE_NONE = 0, // (0 << AUDIO_MODE_SHIFT) - EFFECT_FLAG_AUDIO_SOURCE_SHIFT = 20, // (AUDIO_MODE_SHIFT + AUDIO_MODE_SIZE) - EFFECT_FLAG_AUDIO_SOURCE_SIZE = 2, - EFFECT_FLAG_AUDIO_SOURCE_MASK = - 3145728, // (((1 << AUDIO_SOURCE_SIZE) - 1) << AUDIO_SOURCE_SHIFT) - EFFECT_FLAG_AUDIO_SOURCE_IND = 1048576, // (1 << AUDIO_SOURCE_SHIFT) - EFFECT_FLAG_AUDIO_SOURCE_NONE = 0, // (0 << AUDIO_SOURCE_SHIFT) - EFFECT_FLAG_OFFLOAD_SHIFT = 22, // (AUDIO_SOURCE_SHIFT + AUDIO_SOURCE_SIZE) - EFFECT_FLAG_OFFLOAD_SIZE = 1, - EFFECT_FLAG_OFFLOAD_MASK = 4194304, // (((1 << OFFLOAD_SIZE) - 1) << OFFLOAD_SHIFT) - EFFECT_FLAG_OFFLOAD_SUPPORTED = 4194304, // (1 << OFFLOAD_SHIFT) - EFFECT_FLAG_NO_PROCESS_SHIFT = 23, // (OFFLOAD_SHIFT + OFFLOAD_SIZE) - EFFECT_FLAG_NO_PROCESS_SIZE = 1, - EFFECT_FLAG_NO_PROCESS_MASK = 8388608, // (((1 << NO_PROCESS_SIZE) - 1) << NO_PROCESS_SHIFT) - EFFECT_FLAG_NO_PROCESS = 8388608, // (1 << NO_PROCESS_SHIFT) -}; - -typedef enum { - EFFECT_BUFFER_ACCESS_WRITE = 0, - EFFECT_BUFFER_ACCESS_READ = 1, - EFFECT_BUFFER_ACCESS_ACCUMULATE = 2, -} effect_buffer_access_e; - -enum { - EFFECT_CONFIG_BUFFER = 1, // 0x0001 - EFFECT_CONFIG_SMP_RATE = 2, // 0x0002 - EFFECT_CONFIG_CHANNELS = 4, // 0x0004 - EFFECT_CONFIG_FORMAT = 8, // 0x0008 - EFFECT_CONFIG_ACC_MODE = 16, // 0x0010 - EFFECT_CONFIG_ALL = 31, // ((((BUFFER | SMP_RATE) | CHANNELS) | FORMAT) | ACC_MODE) -}; - -typedef enum { - EFFECT_FEATURE_AUX_CHANNELS = 0, - EFFECT_FEATURE_CNT = 1, -} effect_feature_e; - -#ifdef __cplusplus -} -#endif - -#endif // HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_ diff --git a/audio/common/all-versions/default/Android.bp b/audio/common/all-versions/default/Android.bp index 9b82f05bb1..8f6b74c96e 100644 --- a/audio/common/all-versions/default/Android.bp +++ b/audio/common/all-versions/default/Android.bp @@ -16,7 +16,10 @@ cc_library_shared { name: "android.hardware.audio.common-util", defaults: ["hidl_defaults"], - vendor: true, + vendor_available: true, + vndk: { + enabled: true, + }, srcs: [ "EffectMap.cpp", ], @@ -30,7 +33,7 @@ cc_library_shared { ], header_libs: [ - "android.hardware.audio.common.legacy@2.0", + "libaudio_system_headers", "libhardware_headers", ], } diff --git a/audio/common/all-versions/legacy/Android.bp b/audio/common/all-versions/legacy/Android.bp deleted file mode 100644 index 2fb01ddc63..0000000000 --- a/audio/common/all-versions/legacy/Android.bp +++ /dev/null @@ -1,8 +0,0 @@ -cc_library_headers { - name: "android.hardware.audio.common.legacy@all-versions", - vendor: true, - export_include_dirs: ["include"], - header_libs: ["libcutils_headers"], - export_header_lib_headers: ["libcutils_headers"], -} - diff --git a/audio/common/all-versions/legacy/OWNERS b/audio/common/all-versions/legacy/OWNERS deleted file mode 100644 index 6fdc97ca29..0000000000 --- a/audio/common/all-versions/legacy/OWNERS +++ /dev/null @@ -1,3 +0,0 @@ -elaurent@google.com -krocard@google.com -mnaganov@google.com diff --git a/audio/common/all-versions/legacy/include/hardware/audio.h b/audio/common/all-versions/legacy/include/hardware/audio.h deleted file mode 100644 index 1ad3e0e04e..0000000000 --- a/audio/common/all-versions/legacy/include/hardware/audio.h +++ /dev/null @@ -1,709 +0,0 @@ -/* - * 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 -#include -#include -#include -#include - -#include - -#include -#include -#include - -__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_CURRENT AUDIO_DEVICE_API_VERSION_3_0 -/* 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" - -/** - * 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 int (*stream_callback_t)(stream_callback_event_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; - -/** - * 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); -}; -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); -}; -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 diff --git a/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h b/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h deleted file mode 100644 index aa166549ea..0000000000 --- a/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h +++ /dev/null @@ -1,101 +0,0 @@ -/* - * Copyright (C) 2014 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. - */ - -/* This file contains shared utility functions to handle the tinyalsa - * implementation for Android internal audio, generally in the hardware layer. - * Some routines may log a fatal error on failure, as noted. - */ - -#ifndef ANDROID_AUDIO_ALSAOPS_H -#define ANDROID_AUDIO_ALSAOPS_H - -#include - -#include -#include - -__BEGIN_DECLS - -/* Converts audio_format to pcm_format. - * Parameters: - * format the audio_format_t to convert - * - * Logs a fatal error if format is not a valid convertible audio_format_t. - */ -static inline enum pcm_format pcm_format_from_audio_format(audio_format_t format) { - switch (format) { -#if HAVE_BIG_ENDIAN - case AUDIO_FORMAT_PCM_16_BIT: - return PCM_FORMAT_S16_BE; - case AUDIO_FORMAT_PCM_24_BIT_PACKED: - return PCM_FORMAT_S24_3BE; - case AUDIO_FORMAT_PCM_32_BIT: - return PCM_FORMAT_S32_BE; - case AUDIO_FORMAT_PCM_8_24_BIT: - return PCM_FORMAT_S24_BE; -#else - case AUDIO_FORMAT_PCM_16_BIT: - return PCM_FORMAT_S16_LE; - case AUDIO_FORMAT_PCM_24_BIT_PACKED: - return PCM_FORMAT_S24_3LE; - case AUDIO_FORMAT_PCM_32_BIT: - return PCM_FORMAT_S32_LE; - case AUDIO_FORMAT_PCM_8_24_BIT: - return PCM_FORMAT_S24_LE; -#endif - case AUDIO_FORMAT_PCM_FLOAT: /* there is no equivalent for float */ - default: - LOG_ALWAYS_FATAL("pcm_format_from_audio_format: invalid audio format %#x", format); - return 0; - } -} - -/* Converts pcm_format to audio_format. - * Parameters: - * format the pcm_format to convert - * - * Logs a fatal error if format is not a valid convertible pcm_format. - */ -static inline audio_format_t audio_format_from_pcm_format(enum pcm_format format) { - switch (format) { -#if HAVE_BIG_ENDIAN - case PCM_FORMAT_S16_BE: - return AUDIO_FORMAT_PCM_16_BIT; - case PCM_FORMAT_S24_3BE: - return AUDIO_FORMAT_PCM_24_BIT_PACKED; - case PCM_FORMAT_S24_BE: - return AUDIO_FORMAT_PCM_8_24_BIT; - case PCM_FORMAT_S32_BE: - return AUDIO_FORMAT_PCM_32_BIT; -#else - case PCM_FORMAT_S16_LE: - return AUDIO_FORMAT_PCM_16_BIT; - case PCM_FORMAT_S24_3LE: - return AUDIO_FORMAT_PCM_24_BIT_PACKED; - case PCM_FORMAT_S24_LE: - return AUDIO_FORMAT_PCM_8_24_BIT; - case PCM_FORMAT_S32_LE: - return AUDIO_FORMAT_PCM_32_BIT; -#endif - default: - LOG_ALWAYS_FATAL("audio_format_from_pcm_format: invalid pcm format %#x", format); - return 0; - } -} - -__END_DECLS - -#endif /* ANDROID_AUDIO_ALSAOPS_H */ diff --git a/audio/common/all-versions/legacy/include/hardware/audio_effect.h b/audio/common/all-versions/legacy/include/hardware/audio_effect.h deleted file mode 100644 index b91c60a201..0000000000 --- a/audio/common/all-versions/legacy/include/hardware/audio_effect.h +++ /dev/null @@ -1,295 +0,0 @@ -/* - * 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_EFFECT_H -#define ANDROID_AUDIO_EFFECT_H - -#include -#include -#include -#include -#include - -#include - -#include - -__BEGIN_DECLS - -///////////////////////////////////////////////// -// Common Definitions -///////////////////////////////////////////////// - -#define EFFECT_MAKE_API_VERSION(M, m) (((M) << 16) | ((m)&0xFFFF)) -#define EFFECT_API_VERSION_MAJOR(v) ((v) >> 16) -#define EFFECT_API_VERSION_MINOR(v) ((m)&0xFFFF) - -///////////////////////////////////////////////// -// Effect control interface -///////////////////////////////////////////////// - -// Effect control interface version 2.0 -#define EFFECT_CONTROL_API_VERSION EFFECT_MAKE_API_VERSION(2, 0) - -// Effect control interface structure: effect_interface_s -// The effect control interface is exposed by each effect engine implementation. It consists of -// a set of functions controlling the configuration, activation and process of the engine. -// The functions are grouped in a structure of type effect_interface_s. -// -// Effect control interface handle: effect_handle_t -// The effect_handle_t serves two purposes regarding the implementation of the effect engine: -// - 1 it is the address of a pointer to an effect_interface_s structure where the functions -// of the effect control API for a particular effect are located. -// - 2 it is the address of the context of a particular effect instance. -// A typical implementation in the effect library would define a structure as follows: -// struct effect_module_s { -// const struct effect_interface_s *itfe; -// effect_config_t config; -// effect_context_t context; -// } -// The implementation of EffectCreate() function would then allocate a structure of this -// type and return its address as effect_handle_t -typedef struct effect_interface_s** effect_handle_t; - -// Effect control interface definition -struct effect_interface_s { - //////////////////////////////////////////////////////////////////////////////// - // - // Function: process - // - // Description: Effect process function. Takes input samples as specified - // (count and location) in input buffer descriptor and output processed - // samples as specified in output buffer descriptor. If the buffer descriptor - // is not specified the function must use either the buffer or the - // buffer provider function installed by the EFFECT_CMD_SET_CONFIG command. - // The effect framework will call the process() function after the EFFECT_CMD_ENABLE - // command is received and until the EFFECT_CMD_DISABLE is received. When the engine - // receives the EFFECT_CMD_DISABLE command it should turn off the effect gracefully - // and when done indicate that it is OK to stop calling the process() function by - // returning the -ENODATA status. - // - // NOTE: the process() function implementation should be "real-time safe" that is - // it should not perform blocking calls: malloc/free, sleep, read/write/open/close, - // pthread_cond_wait/pthread_mutex_lock... - // - // Input: - // self: handle to the effect interface this function - // is called on. - // inBuffer: buffer descriptor indicating where to read samples to process. - // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command. - // - // outBuffer: buffer descriptor indicating where to write processed samples. - // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command. - // - // Output: - // returned value: 0 successful operation - // -ENODATA the engine has finished the disable phase and the framework - // can stop calling process() - // -EINVAL invalid interface handle or - // invalid input/output buffer description - //////////////////////////////////////////////////////////////////////////////// - int32_t (*process)(effect_handle_t self, audio_buffer_t* inBuffer, audio_buffer_t* outBuffer); - //////////////////////////////////////////////////////////////////////////////// - // - // Function: command - // - // Description: Send a command and receive a response to/from effect engine. - // - // Input: - // self: handle to the effect interface this function - // is called on. - // cmdCode: command code: the command can be a standardized command defined in - // effect_command_e (see below) or a proprietary command. - // cmdSize: size of command in bytes - // pCmdData: pointer to command data - // pReplyData: pointer to reply data - // - // Input/Output: - // replySize: maximum size of reply data as input - // actual size of reply data as output - // - // Output: - // returned value: 0 successful operation - // -EINVAL invalid interface handle or - // invalid command/reply size or format according to - // command code - // The return code should be restricted to indicate problems related to this API - // specification. Status related to the execution of a particular command should be - // indicated as part of the reply field. - // - // *pReplyData updated with command response - // - //////////////////////////////////////////////////////////////////////////////// - int32_t (*command)(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize, void* pCmdData, - uint32_t* replySize, void* pReplyData); - //////////////////////////////////////////////////////////////////////////////// - // - // Function: get_descriptor - // - // Description: Returns the effect descriptor - // - // Input: - // self: handle to the effect interface this function - // is called on. - // - // Input/Output: - // pDescriptor: address where to return the effect descriptor. - // - // Output: - // returned value: 0 successful operation. - // -EINVAL invalid interface handle or invalid pDescriptor - // *pDescriptor: updated with the effect descriptor. - // - //////////////////////////////////////////////////////////////////////////////// - int32_t (*get_descriptor)(effect_handle_t self, effect_descriptor_t* pDescriptor); - //////////////////////////////////////////////////////////////////////////////// - // - // Function: process_reverse - // - // Description: Process reverse stream function. This function is used to pass - // a reference stream to the effect engine. If the engine does not need a reference - // stream, this function pointer can be set to NULL. - // This function would typically implemented by an Echo Canceler. - // - // Input: - // self: handle to the effect interface this function - // is called on. - // inBuffer: buffer descriptor indicating where to read samples to process. - // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command. - // - // outBuffer: buffer descriptor indicating where to write processed samples. - // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command. - // If the buffer and buffer provider in the configuration received by - // EFFECT_CMD_SET_CONFIG_REVERSE are also NULL, do not return modified reverse - // stream data - // - // Output: - // returned value: 0 successful operation - // -ENODATA the engine has finished the disable phase and the framework - // can stop calling process_reverse() - // -EINVAL invalid interface handle or - // invalid input/output buffer description - //////////////////////////////////////////////////////////////////////////////// - int32_t (*process_reverse)(effect_handle_t self, audio_buffer_t* inBuffer, - audio_buffer_t* outBuffer); -}; - -///////////////////////////////////////////////// -// Effect library interface -///////////////////////////////////////////////// - -// Effect library interface version 3.0 -// Note that EffectsFactory.c only checks the major version component, so changes to the minor -// number can only be used for fully backwards compatible changes -#define EFFECT_LIBRARY_API_VERSION EFFECT_MAKE_API_VERSION(3, 0) - -#define AUDIO_EFFECT_LIBRARY_TAG ((('A') << 24) | (('E') << 16) | (('L') << 8) | ('T')) - -// Every effect library must have a data structure named AUDIO_EFFECT_LIBRARY_INFO_SYM -// and the fields of this data structure must begin with audio_effect_library_t - -typedef struct audio_effect_library_s { - // tag must be initialized to AUDIO_EFFECT_LIBRARY_TAG - uint32_t tag; - // Version of the effect library API : 0xMMMMmmmm MMMM: Major, mmmm: minor - uint32_t version; - // Name of this library - const char* name; - // Author/owner/implementor of the library - const char* implementor; - - //////////////////////////////////////////////////////////////////////////////// - // - // Function: create_effect - // - // Description: Creates an effect engine of the specified implementation uuid and - // returns an effect control interface on this engine. The function will allocate the - // resources for an instance of the requested effect engine and return - // a handle on the effect control interface. - // - // Input: - // uuid: pointer to the effect uuid. - // sessionId: audio session to which this effect instance will be attached. - // All effects created with the same session ID are connected in series and process - // the same signal stream. Knowing that two effects are part of the same effect - // chain can help the library implement some kind of optimizations. - // ioId: identifies the output or input stream this effect is directed to in - // audio HAL. - // For future use especially with tunneled HW accelerated effects - // - // Input/Output: - // pHandle: address where to return the effect interface handle. - // - // Output: - // returned value: 0 successful operation. - // -ENODEV library failed to initialize - // -EINVAL invalid pEffectUuid or pHandle - // -ENOENT no effect with this uuid found - // *pHandle: updated with the effect interface handle. - // - //////////////////////////////////////////////////////////////////////////////// - int32_t (*create_effect)(const effect_uuid_t* uuid, int32_t sessionId, int32_t ioId, - effect_handle_t* pHandle); - - //////////////////////////////////////////////////////////////////////////////// - // - // Function: release_effect - // - // Description: Releases the effect engine whose handle is given as argument. - // All resources allocated to this particular instance of the effect are - // released. - // - // Input: - // handle: handle on the effect interface to be released. - // - // Output: - // returned value: 0 successful operation. - // -ENODEV library failed to initialize - // -EINVAL invalid interface handle - // - //////////////////////////////////////////////////////////////////////////////// - int32_t (*release_effect)(effect_handle_t handle); - - //////////////////////////////////////////////////////////////////////////////// - // - // Function: get_descriptor - // - // Description: Returns the descriptor of the effect engine which implementation UUID is - // given as argument. - // - // Input/Output: - // uuid: pointer to the effect uuid. - // pDescriptor: address where to return the effect descriptor. - // - // Output: - // returned value: 0 successful operation. - // -ENODEV library failed to initialize - // -EINVAL invalid pDescriptor or uuid - // *pDescriptor: updated with the effect descriptor. - // - //////////////////////////////////////////////////////////////////////////////// - int32_t (*get_descriptor)(const effect_uuid_t* uuid, effect_descriptor_t* pDescriptor); -} audio_effect_library_t; - -// Name of the hal_module_info -#define AUDIO_EFFECT_LIBRARY_INFO_SYM AELI - -// Name of the hal_module_info as a string -#define AUDIO_EFFECT_LIBRARY_INFO_SYM_AS_STR "AELI" - -__END_DECLS - -#endif // ANDROID_AUDIO_EFFECT_H diff --git a/audio/common/all-versions/legacy/include/hardware/audio_policy.h b/audio/common/all-versions/legacy/include/hardware/audio_policy.h deleted file mode 100644 index 8cc79dfca9..0000000000 --- a/audio/common/all-versions/legacy/include/hardware/audio_policy.h +++ /dev/null @@ -1,391 +0,0 @@ -/* - * 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_POLICY_INTERFACE_H -#define ANDROID_AUDIO_POLICY_INTERFACE_H - -#include -#include -#include - -#include - -#include -#include - -__BEGIN_DECLS - -/** - * The id of this module - */ -#define AUDIO_POLICY_HARDWARE_MODULE_ID "audio_policy" - -/** - * Name of the audio devices to open - */ -#define AUDIO_POLICY_INTERFACE "policy" - -/* ---------------------------------------------------------------------------- */ - -/* - * The audio_policy and audio_policy_service_ops structs define the - * communication interfaces between the platform specific audio policy manager - * and Android generic audio policy manager. - * The platform specific audio policy manager must implement methods of the - * audio_policy struct. - * This implementation makes use of the audio_policy_service_ops to control - * the activity and configuration of audio input and output streams. - * - * The platform specific audio policy manager is in charge of the audio - * routing and volume control policies for a given platform. - * The main roles of this module are: - * - keep track of current system state (removable device connections, phone - * state, user requests...). - * System state changes and user actions are notified to audio policy - * manager with methods of the audio_policy. - * - * - process get_output() queries received when AudioTrack objects are - * created: Those queries return a handler on an output that has been - * selected, configured and opened by the audio policy manager and that - * must be used by the AudioTrack when registering to the AudioFlinger - * with the createTrack() method. - * When the AudioTrack object is released, a release_output() query - * is received and the audio policy manager can decide to close or - * reconfigure the output depending on other streams using this output and - * current system state. - * - * - similarly process get_input() and release_input() queries received from - * AudioRecord objects and configure audio inputs. - * - process volume control requests: the stream volume is converted from - * an index value (received from UI) to a float value applicable to each - * output as a function of platform specific settings and current output - * route (destination device). It also make sure that streams are not - * muted if not allowed (e.g. camera shutter sound in some countries). - */ - -/* XXX: this should be defined OUTSIDE of frameworks/base */ -struct effect_descriptor_s; - -struct audio_policy { - /* - * configuration functions - */ - - /* indicate a change in device connection status */ - int (*set_device_connection_state)(struct audio_policy* pol, audio_devices_t device, - audio_policy_dev_state_t state, const char* device_address); - - /* retrieve a device connection status */ - audio_policy_dev_state_t (*get_device_connection_state)(const struct audio_policy* pol, - audio_devices_t device, - const char* device_address); - - /* indicate a change in phone state. Valid phones states are defined - * by audio_mode_t */ - void (*set_phone_state)(struct audio_policy* pol, audio_mode_t state); - - /* deprecated, never called (was "indicate a change in ringer mode") */ - void (*set_ringer_mode)(struct audio_policy* pol, uint32_t mode, uint32_t mask); - - /* force using a specific device category for the specified usage */ - void (*set_force_use)(struct audio_policy* pol, audio_policy_force_use_t usage, - audio_policy_forced_cfg_t config); - - /* retrieve current device category forced for a given usage */ - audio_policy_forced_cfg_t (*get_force_use)(const struct audio_policy* pol, - audio_policy_force_use_t usage); - - /* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE - * can still be muted. */ - void (*set_can_mute_enforced_audible)(struct audio_policy* pol, bool can_mute); - - /* check proper initialization */ - int (*init_check)(const struct audio_policy* pol); - - /* - * Audio routing query functions - */ - - /* request an output appropriate for playback of the supplied stream type and - * parameters */ - audio_io_handle_t (*get_output)(struct audio_policy* pol, audio_stream_type_t stream, - uint32_t samplingRate, audio_format_t format, - audio_channel_mask_t channelMask, audio_output_flags_t flags, - const audio_offload_info_t* offloadInfo); - - /* indicates to the audio policy manager that the output starts being used - * by corresponding stream. */ - int (*start_output)(struct audio_policy* pol, audio_io_handle_t output, - audio_stream_type_t stream, audio_session_t session); - - /* indicates to the audio policy manager that the output stops being used - * by corresponding stream. */ - int (*stop_output)(struct audio_policy* pol, audio_io_handle_t output, - audio_stream_type_t stream, audio_session_t session); - - /* releases the output. */ - void (*release_output)(struct audio_policy* pol, audio_io_handle_t output); - - /* request an input appropriate for record from the supplied device with - * supplied parameters. */ - audio_io_handle_t (*get_input)(struct audio_policy* pol, audio_source_t inputSource, - uint32_t samplingRate, audio_format_t format, - audio_channel_mask_t channelMask, - audio_in_acoustics_t acoustics); - - /* indicates to the audio policy manager that the input starts being used */ - int (*start_input)(struct audio_policy* pol, audio_io_handle_t input); - - /* indicates to the audio policy manager that the input stops being used. */ - int (*stop_input)(struct audio_policy* pol, audio_io_handle_t input); - - /* releases the input. */ - void (*release_input)(struct audio_policy* pol, audio_io_handle_t input); - - /* - * volume control functions - */ - - /* initialises stream volume conversion parameters by specifying volume - * index range. The index range for each stream is defined by AudioService. */ - void (*init_stream_volume)(struct audio_policy* pol, audio_stream_type_t stream, int index_min, - int index_max); - - /* sets the new stream volume at a level corresponding to the supplied - * index. The index is within the range specified by init_stream_volume() */ - int (*set_stream_volume_index)(struct audio_policy* pol, audio_stream_type_t stream, int index); - - /* retrieve current volume index for the specified stream */ - int (*get_stream_volume_index)(const struct audio_policy* pol, audio_stream_type_t stream, - int* index); - - /* sets the new stream volume at a level corresponding to the supplied - * index for the specified device. - * The index is within the range specified by init_stream_volume() */ - int (*set_stream_volume_index_for_device)(struct audio_policy* pol, audio_stream_type_t stream, - int index, audio_devices_t device); - - /* retrieve current volume index for the specified stream for the specified device */ - int (*get_stream_volume_index_for_device)(const struct audio_policy* pol, - audio_stream_type_t stream, int* index, - audio_devices_t device); - - /* return the strategy corresponding to a given stream type */ - uint32_t (*get_strategy_for_stream)(const struct audio_policy* pol, audio_stream_type_t stream); - - /* return the enabled output devices for the given stream type */ - audio_devices_t (*get_devices_for_stream)(const struct audio_policy* pol, - audio_stream_type_t stream); - - /* Audio effect management */ - audio_io_handle_t (*get_output_for_effect)(struct audio_policy* pol, - const struct effect_descriptor_s* desc); - - int (*register_effect)(struct audio_policy* pol, const struct effect_descriptor_s* desc, - audio_io_handle_t output, uint32_t strategy, audio_session_t session, - int id); - - int (*unregister_effect)(struct audio_policy* pol, int id); - - int (*set_effect_enabled)(struct audio_policy* pol, int id, bool enabled); - - bool (*is_stream_active)(const struct audio_policy* pol, audio_stream_type_t stream, - uint32_t in_past_ms); - - bool (*is_stream_active_remotely)(const struct audio_policy* pol, audio_stream_type_t stream, - uint32_t in_past_ms); - - bool (*is_source_active)(const struct audio_policy* pol, audio_source_t source); - - /* dump state */ - int (*dump)(const struct audio_policy* pol, int fd); - - /* check if offload is possible for given sample rate, bitrate, duration, ... */ - bool (*is_offload_supported)(const struct audio_policy* pol, const audio_offload_info_t* info); -}; - -struct audio_policy_service_ops { - /* - * Audio output Control functions - */ - - /* Opens an audio output with the requested parameters. - * - * The parameter values can indicate to use the default values in case the - * audio policy manager has no specific requirements for the output being - * opened. - * - * When the function returns, the parameter values reflect the actual - * values used by the audio hardware output stream. - * - * The audio policy manager can check if the proposed parameters are - * suitable or not and act accordingly. - */ - audio_io_handle_t (*open_output)(void* service, audio_devices_t* pDevices, - uint32_t* pSamplingRate, audio_format_t* pFormat, - audio_channel_mask_t* pChannelMask, uint32_t* pLatencyMs, - audio_output_flags_t flags); - - /* creates a special output that is duplicated to the two outputs passed as - * arguments. The duplication is performed by - * a special mixer thread in the AudioFlinger. - */ - audio_io_handle_t (*open_duplicate_output)(void* service, audio_io_handle_t output1, - audio_io_handle_t output2); - - /* closes the output stream */ - int (*close_output)(void* service, audio_io_handle_t output); - - /* suspends the output. - * - * When an output is suspended, the corresponding audio hardware output - * stream is placed in standby and the AudioTracks attached to the mixer - * thread are still processed but the output mix is discarded. - */ - int (*suspend_output)(void* service, audio_io_handle_t output); - - /* restores a suspended output. */ - int (*restore_output)(void* service, audio_io_handle_t output); - - /* */ - /* Audio input Control functions */ - /* */ - - /* opens an audio input - * deprecated - new implementations should use open_input_on_module, - * and the acoustics parameter is ignored - */ - audio_io_handle_t (*open_input)(void* service, audio_devices_t* pDevices, - uint32_t* pSamplingRate, audio_format_t* pFormat, - audio_channel_mask_t* pChannelMask, - audio_in_acoustics_t acoustics); - - /* closes an audio input */ - int (*close_input)(void* service, audio_io_handle_t input); - - /* */ - /* misc control functions */ - /* */ - - /* set a stream volume for a particular output. - * - * For the same user setting, a given stream type can have different - * volumes for each output (destination device) it is attached to. - */ - int (*set_stream_volume)(void* service, audio_stream_type_t stream, float volume, - audio_io_handle_t output, int delay_ms); - - /* invalidate a stream type, causing a reroute to an unspecified new output */ - int (*invalidate_stream)(void* service, audio_stream_type_t stream); - - /* function enabling to send proprietary informations directly from audio - * policy manager to audio hardware interface. */ - void (*set_parameters)(void* service, audio_io_handle_t io_handle, const char* kv_pairs, - int delay_ms); - - /* function enabling to receive proprietary informations directly from - * audio hardware interface to audio policy manager. - * - * Returns a pointer to a heap allocated string. The caller is responsible - * for freeing the memory for it using free(). - */ - - char* (*get_parameters)(void* service, audio_io_handle_t io_handle, const char* keys); - - /* request the playback of a tone on the specified stream. - * used for instance to replace notification sounds when playing over a - * telephony device during a phone call. - */ - int (*start_tone)(void* service, audio_policy_tone_t tone, audio_stream_type_t stream); - - int (*stop_tone)(void* service); - - /* set down link audio volume. */ - int (*set_voice_volume)(void* service, float volume, int delay_ms); - - /* move effect to the specified output */ - int (*move_effects)(void* service, audio_session_t session, audio_io_handle_t src_output, - audio_io_handle_t dst_output); - - /* loads an audio hw module. - * - * The module name passed is the base name of the HW module library, e.g "primary" or "a2dp". - * The function returns a handle on the module that will be used to specify a particular - * module when calling open_output_on_module() or open_input_on_module() - */ - audio_module_handle_t (*load_hw_module)(void* service, const char* name); - - /* Opens an audio output on a particular HW module. - * - * Same as open_output() but specifying a specific HW module on which the output must be opened. - */ - audio_io_handle_t (*open_output_on_module)(void* service, audio_module_handle_t module, - audio_devices_t* pDevices, uint32_t* pSamplingRate, - audio_format_t* pFormat, - audio_channel_mask_t* pChannelMask, - uint32_t* pLatencyMs, audio_output_flags_t flags, - const audio_offload_info_t* offloadInfo); - - /* Opens an audio input on a particular HW module. - * - * Same as open_input() but specifying a specific HW module on which the input must be opened. - * Also removed deprecated acoustics parameter - */ - audio_io_handle_t (*open_input_on_module)(void* service, audio_module_handle_t module, - audio_devices_t* pDevices, uint32_t* pSamplingRate, - audio_format_t* pFormat, - audio_channel_mask_t* pChannelMask); -}; - -/**********************************************************************/ - -/** - * 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. - */ -typedef struct audio_policy_module { struct hw_module_t common; } audio_policy_module_t; - -struct audio_policy_device { - /** - * Common methods of the audio policy device. This *must* be the first member of - * audio_policy_device as users of this structure will cast a hw_device_t to - * audio_policy_device pointer in contexts where it's known the hw_device_t references an - * audio_policy_device. - */ - struct hw_device_t common; - - int (*create_audio_policy)(const struct audio_policy_device* device, - struct audio_policy_service_ops* aps_ops, void* service, - struct audio_policy** ap); - - int (*destroy_audio_policy)(const struct audio_policy_device* device, struct audio_policy* ap); -}; - -/** convenience API for opening and closing a supported device */ - -static inline int audio_policy_dev_open(const hw_module_t* module, - struct audio_policy_device** device) { - return module->methods->open(module, AUDIO_POLICY_INTERFACE, (hw_device_t**)device); -} - -static inline int audio_policy_dev_close(struct audio_policy_device* device) { - return device->common.close(&device->common); -} - -__END_DECLS - -#endif // ANDROID_AUDIO_POLICY_INTERFACE_H diff --git a/audio/common/all-versions/legacy/include/system/audio.h b/audio/common/all-versions/legacy/include/system/audio.h deleted file mode 100644 index 7afa6c4061..0000000000 --- a/audio/common/all-versions/legacy/include/system/audio.h +++ /dev/null @@ -1,1038 +0,0 @@ -/* - * 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_CORE_H -#define ANDROID_AUDIO_CORE_H - -#include -#include -#include -#include -#include - -#include - -#include "system/audio-base.h" - -__BEGIN_DECLS - -/* The enums were moved here mostly from - * frameworks/base/include/media/AudioSystem.h - */ - -/* represents an invalid uid for tracks; the calling or client uid is often substituted. */ -#define AUDIO_UID_INVALID ((uid_t)-1) - -/* device address used to refer to the standard remote submix */ -#define AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS "0" - -/* AudioFlinger and AudioPolicy services use I/O handles to identify audio sources and sinks */ -typedef int audio_io_handle_t; - -/* Do not change these values without updating their counterparts - * in frameworks/base/media/java/android/media/AudioAttributes.java - */ -typedef enum { - AUDIO_CONTENT_TYPE_UNKNOWN = 0, - AUDIO_CONTENT_TYPE_SPEECH = 1, - AUDIO_CONTENT_TYPE_MUSIC = 2, - AUDIO_CONTENT_TYPE_MOVIE = 3, - AUDIO_CONTENT_TYPE_SONIFICATION = 4, - - AUDIO_CONTENT_TYPE_CNT, - AUDIO_CONTENT_TYPE_MAX = AUDIO_CONTENT_TYPE_CNT - 1, -} audio_content_type_t; - -typedef uint32_t audio_flags_mask_t; - -/* Do not change these values without updating their counterparts - * in frameworks/base/media/java/android/media/AudioAttributes.java - */ -enum { - AUDIO_FLAG_NONE = 0x0, - AUDIO_FLAG_AUDIBILITY_ENFORCED = 0x1, - AUDIO_FLAG_SECURE = 0x2, - AUDIO_FLAG_SCO = 0x4, - AUDIO_FLAG_BEACON = 0x8, - AUDIO_FLAG_HW_AV_SYNC = 0x10, - AUDIO_FLAG_HW_HOTWORD = 0x20, - AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY = 0x40, - AUDIO_FLAG_BYPASS_MUTE = 0x80, - AUDIO_FLAG_LOW_LATENCY = 0x100, - AUDIO_FLAG_DEEP_BUFFER = 0x200, -}; - -/* Audio attributes */ -#define AUDIO_ATTRIBUTES_TAGS_MAX_SIZE 256 -typedef struct { - audio_content_type_t content_type; - audio_usage_t usage; - audio_source_t source; - audio_flags_mask_t flags; - char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */ -} __attribute__((packed)) audio_attributes_t; // sent through Binder; - -/* a unique ID allocated by AudioFlinger for use as an audio_io_handle_t, audio_session_t, - * effect ID (int), audio_module_handle_t, and audio_patch_handle_t. - * Audio port IDs (audio_port_handle_t) are allocated by AudioPolicy - * in a different namespace than AudioFlinger unique IDs. - */ -typedef int audio_unique_id_t; - -/* Possible uses for an audio_unique_id_t */ -typedef enum { - AUDIO_UNIQUE_ID_USE_UNSPECIFIED = 0, - AUDIO_UNIQUE_ID_USE_SESSION = 1, // for allocated sessions, not special AUDIO_SESSION_* - AUDIO_UNIQUE_ID_USE_MODULE = 2, - AUDIO_UNIQUE_ID_USE_EFFECT = 3, - AUDIO_UNIQUE_ID_USE_PATCH = 4, - AUDIO_UNIQUE_ID_USE_OUTPUT = 5, - AUDIO_UNIQUE_ID_USE_INPUT = 6, - AUDIO_UNIQUE_ID_USE_PLAYER = 7, - AUDIO_UNIQUE_ID_USE_MAX = 8, // must be a power-of-two - AUDIO_UNIQUE_ID_USE_MASK = AUDIO_UNIQUE_ID_USE_MAX - 1 -} audio_unique_id_use_t; - -/* Return the use of an audio_unique_id_t */ -static inline audio_unique_id_use_t audio_unique_id_get_use(audio_unique_id_t id) { - return (audio_unique_id_use_t)(id & AUDIO_UNIQUE_ID_USE_MASK); -} - -/* Reserved audio_unique_id_t values. FIXME: not a complete list. */ -#define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE - -/* A channel mask per se only defines the presence or absence of a channel, not the order. - * But see AUDIO_INTERLEAVE_* below for the platform convention of order. - * - * audio_channel_mask_t is an opaque type and its internal layout should not - * be assumed as it may change in the future. - * Instead, always use the functions declared in this header to examine. - * - * These are the current representations: - * - * AUDIO_CHANNEL_REPRESENTATION_POSITION - * is a channel mask representation for position assignment. - * Each low-order bit corresponds to the spatial position of a transducer (output), - * or interpretation of channel (input). - * The user of a channel mask needs to know the context of whether it is for output or input. - * The constants AUDIO_CHANNEL_OUT_* or AUDIO_CHANNEL_IN_* apply to the bits portion. - * It is not permitted for no bits to be set. - * - * AUDIO_CHANNEL_REPRESENTATION_INDEX - * is a channel mask representation for index assignment. - * Each low-order bit corresponds to a selected channel. - * There is no platform interpretation of the various bits. - * There is no concept of output or input. - * It is not permitted for no bits to be set. - * - * All other representations are reserved for future use. - * - * Warning: current representation distinguishes between input and output, but this will not the be - * case in future revisions of the platform. Wherever there is an ambiguity between input and output - * that is currently resolved by checking the channel mask, the implementer should look for ways to - * fix it with additional information outside of the mask. - */ -typedef uint32_t audio_channel_mask_t; - -/* log(2) of maximum number of representations, not part of public API */ -#define AUDIO_CHANNEL_REPRESENTATION_LOG2 2 - -/* The return value is undefined if the channel mask is invalid. */ -static inline uint32_t audio_channel_mask_get_bits(audio_channel_mask_t channel) { - return channel & ((1 << AUDIO_CHANNEL_COUNT_MAX) - 1); -} - -typedef uint32_t audio_channel_representation_t; - -/* The return value is undefined if the channel mask is invalid. */ -static inline audio_channel_representation_t audio_channel_mask_get_representation( - audio_channel_mask_t channel) { - // The right shift should be sufficient, but also "and" for safety in case mask is not 32 bits - return (audio_channel_representation_t)((channel >> AUDIO_CHANNEL_COUNT_MAX) & - ((1 << AUDIO_CHANNEL_REPRESENTATION_LOG2) - 1)); -} - -/* Returns true if the channel mask is valid, - * or returns false for AUDIO_CHANNEL_NONE, AUDIO_CHANNEL_INVALID, and other invalid values. - * This function is unable to determine whether a channel mask for position assignment - * is invalid because an output mask has an invalid output bit set, - * or because an input mask has an invalid input bit set. - * All other APIs that take a channel mask assume that it is valid. - */ -static inline bool audio_channel_mask_is_valid(audio_channel_mask_t channel) { - uint32_t bits = audio_channel_mask_get_bits(channel); - audio_channel_representation_t representation = audio_channel_mask_get_representation(channel); - switch (representation) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - break; - default: - bits = 0; - break; - } - return bits != 0; -} - -/* Not part of public API */ -static inline audio_channel_mask_t audio_channel_mask_from_representation_and_bits( - audio_channel_representation_t representation, uint32_t bits) { - return (audio_channel_mask_t)((representation << AUDIO_CHANNEL_COUNT_MAX) | bits); -} - -/* This enum is deprecated */ -typedef enum { - AUDIO_IN_ACOUSTICS_NONE = 0, - AUDIO_IN_ACOUSTICS_AGC_ENABLE = 0x0001, - AUDIO_IN_ACOUSTICS_AGC_DISABLE = 0, - AUDIO_IN_ACOUSTICS_NS_ENABLE = 0x0002, - AUDIO_IN_ACOUSTICS_NS_DISABLE = 0, - AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE = 0x0004, - AUDIO_IN_ACOUSTICS_TX_DISABLE = 0, -} audio_in_acoustics_t; - -typedef uint32_t audio_devices_t; -/** - * Stub audio output device. Used in policy configuration file on platforms without audio outputs. - * This alias value to AUDIO_DEVICE_OUT_DEFAULT is only used in the audio policy context. - */ -#define AUDIO_DEVICE_OUT_STUB AUDIO_DEVICE_OUT_DEFAULT -/** - * Stub audio input device. Used in policy configuration file on platforms without audio inputs. - * This alias value to AUDIO_DEVICE_IN_DEFAULT is only used in the audio policy context. - */ -#define AUDIO_DEVICE_IN_STUB AUDIO_DEVICE_IN_DEFAULT - -/* Additional information about compressed streams offloaded to - * hardware playback - * The version and size fields must be initialized by the caller by using - * one of the constants defined here. - * Must be aligned to transmit as raw memory through Binder. - */ -typedef struct { - uint16_t version; // version of the info structure - uint16_t size; // total size of the structure including version and size - uint32_t sample_rate; // sample rate in Hz - audio_channel_mask_t channel_mask; // channel mask - audio_format_t format; // audio format - audio_stream_type_t stream_type; // stream type - uint32_t bit_rate; // bit rate in bits per second - int64_t duration_us; // duration in microseconds, -1 if unknown - bool has_video; // true if stream is tied to a video stream - bool is_streaming; // true if streaming, false if local playback - uint32_t bit_width; - uint32_t offload_buffer_size; // offload fragment size - audio_usage_t usage; -} __attribute__((aligned(8))) audio_offload_info_t; - -#define AUDIO_MAKE_OFFLOAD_INFO_VERSION(maj, min) ((((maj)&0xff) << 8) | ((min)&0xff)) - -#define AUDIO_OFFLOAD_INFO_VERSION_0_1 AUDIO_MAKE_OFFLOAD_INFO_VERSION(0, 1) -#define AUDIO_OFFLOAD_INFO_VERSION_CURRENT AUDIO_OFFLOAD_INFO_VERSION_0_1 - -static const audio_offload_info_t AUDIO_INFO_INITIALIZER = { - /* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT, - /* .size = */ sizeof(audio_offload_info_t), - /* .sample_rate = */ 0, - /* .channel_mask = */ 0, - /* .format = */ AUDIO_FORMAT_DEFAULT, - /* .stream_type = */ AUDIO_STREAM_VOICE_CALL, - /* .bit_rate = */ 0, - /* .duration_us = */ 0, - /* .has_video = */ false, - /* .is_streaming = */ false, - /* .bit_width = */ 16, - /* .offload_buffer_size = */ 0, - /* .usage = */ AUDIO_USAGE_UNKNOWN}; - -/* common audio stream configuration parameters - * You should memset() the entire structure to zero before use to - * ensure forward compatibility - * Must be aligned to transmit as raw memory through Binder. - */ -struct __attribute__((aligned(8))) audio_config { - uint32_t sample_rate; - audio_channel_mask_t channel_mask; - audio_format_t format; - audio_offload_info_t offload_info; - uint32_t frame_count; -}; -typedef struct audio_config audio_config_t; - -static const audio_config_t AUDIO_CONFIG_INITIALIZER = { - /* .sample_rate = */ 0, - /* .channel_mask = */ AUDIO_CHANNEL_NONE, - /* .format = */ AUDIO_FORMAT_DEFAULT, - /* .offload_info = */ - {/* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT, - /* .size = */ sizeof(audio_offload_info_t), - /* .sample_rate = */ 0, - /* .channel_mask = */ 0, - /* .format = */ AUDIO_FORMAT_DEFAULT, - /* .stream_type = */ AUDIO_STREAM_VOICE_CALL, - /* .bit_rate = */ 0, - /* .duration_us = */ 0, - /* .has_video = */ false, - /* .is_streaming = */ false, - /* .bit_width = */ 16, - /* .offload_buffer_size = */ 0, - /* .usage = */ AUDIO_USAGE_UNKNOWN}, - /* .frame_count = */ 0, -}; - -struct audio_config_base { - uint32_t sample_rate; - audio_channel_mask_t channel_mask; - audio_format_t format; -}; - -typedef struct audio_config_base audio_config_base_t; - -static const audio_config_base_t AUDIO_CONFIG_BASE_INITIALIZER = { - /* .sample_rate = */ 0, - /* .channel_mask = */ AUDIO_CHANNEL_NONE, - /* .format = */ AUDIO_FORMAT_DEFAULT}; - -/* audio hw module handle functions or structures referencing a module */ -typedef int audio_module_handle_t; - -/****************************** - * Volume control - *****************************/ - -/** 3 dB headroom are allowed on float samples (3db = 10^(3/20) = 1.412538). - * See: https://developer.android.com/reference/android/media/AudioTrack.html#write(float[], int, - * int, int) - */ -#define FLOAT_NOMINAL_RANGE_HEADROOM 1.412538 - -/* If the audio hardware supports gain control on some audio paths, - * the platform can expose them in the audio_policy.conf file. The audio HAL - * will then implement gain control functions that will use the following data - * structures. */ - -typedef uint32_t audio_gain_mode_t; - -/* An audio_gain struct is a representation of a gain stage. - * A gain stage is always attached to an audio port. */ -struct audio_gain { - audio_gain_mode_t mode; /* e.g. AUDIO_GAIN_MODE_JOINT */ - audio_channel_mask_t channel_mask; /* channels which gain an be controlled. - N/A if AUDIO_GAIN_MODE_CHANNELS is not supported */ - int min_value; /* minimum gain value in millibels */ - int max_value; /* maximum gain value in millibels */ - int default_value; /* default gain value in millibels */ - unsigned int step_value; /* gain step in millibels */ - unsigned int min_ramp_ms; /* minimum ramp duration in ms */ - unsigned int max_ramp_ms; /* maximum ramp duration in ms */ -}; - -/* The gain configuration structure is used to get or set the gain values of a - * given port */ -struct audio_gain_config { - int index; /* index of the corresponding audio_gain in the - audio_port gains[] table */ - audio_gain_mode_t mode; /* mode requested for this command */ - audio_channel_mask_t channel_mask; /* channels which gain value follows. - N/A in joint mode */ - - // note this "8" is not FCC_8, so it won't need to be changed for > 8 channels - int values[sizeof(audio_channel_mask_t) * 8]; /* gain values in millibels - for each channel ordered from LSb to MSb in - channel mask. The number of values is 1 in joint - mode or popcount(channel_mask) */ - unsigned int ramp_duration_ms; /* ramp duration in ms */ -}; - -/****************************** - * Routing control - *****************************/ - -/* Types defined here are used to describe an audio source or sink at internal - * framework interfaces (audio policy, patch panel) or at the audio HAL. - * Sink and sources are grouped in a concept of “audio port” representing an - * audio end point at the edge of the system managed by the module exposing - * the interface. */ - -/* Each port has a unique ID or handle allocated by policy manager */ -typedef int audio_port_handle_t; - -/* the maximum length for the human-readable device name */ -#define AUDIO_PORT_MAX_NAME_LEN 128 - -/* maximum audio device address length */ -#define AUDIO_DEVICE_MAX_ADDRESS_LEN 32 - -/* extension for audio port configuration structure when the audio port is a - * hardware device */ -struct audio_port_config_device_ext { - audio_module_handle_t hw_module; /* module the device is attached to */ - audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */ - char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; /* device address. "" if N/A */ -}; - -/* extension for audio port configuration structure when the audio port is a - * sub mix */ -struct audio_port_config_mix_ext { - audio_module_handle_t hw_module; /* module the stream is attached to */ - audio_io_handle_t handle; /* I/O handle of the input/output stream */ - union { - // TODO: change use case for output streams: use strategy and mixer attributes - audio_stream_type_t stream; - audio_source_t source; - } usecase; -}; - -/* extension for audio port configuration structure when the audio port is an - * audio session */ -struct audio_port_config_session_ext { - audio_session_t session; /* audio session */ -}; - -/* audio port configuration structure used to specify a particular configuration of - * an audio port */ -struct audio_port_config { - audio_port_handle_t id; /* port unique ID */ - audio_port_role_t role; /* sink or source */ - audio_port_type_t type; /* device, mix ... */ - unsigned int config_mask; /* e.g AUDIO_PORT_CONFIG_ALL */ - unsigned int sample_rate; /* sampling rate in Hz */ - audio_channel_mask_t channel_mask; /* channel mask if applicable */ - audio_format_t format; /* format if applicable */ - struct audio_gain_config gain; /* gain to apply if applicable */ - union { - struct audio_port_config_device_ext device; /* device specific info */ - struct audio_port_config_mix_ext mix; /* mix specific info */ - struct audio_port_config_session_ext session; /* session specific info */ - } ext; -}; - -/* max number of sampling rates in audio port */ -#define AUDIO_PORT_MAX_SAMPLING_RATES 32 -/* max number of channel masks in audio port */ -#define AUDIO_PORT_MAX_CHANNEL_MASKS 32 -/* max number of audio formats in audio port */ -#define AUDIO_PORT_MAX_FORMATS 32 -/* max number of gain controls in audio port */ -#define AUDIO_PORT_MAX_GAINS 16 - -/* extension for audio port structure when the audio port is a hardware device */ -struct audio_port_device_ext { - audio_module_handle_t hw_module; /* module the device is attached to */ - audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */ - char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; -}; - -/* extension for audio port structure when the audio port is a sub mix */ -struct audio_port_mix_ext { - audio_module_handle_t hw_module; /* module the stream is attached to */ - audio_io_handle_t handle; /* I/O handle of the input.output stream */ - audio_mix_latency_class_t latency_class; /* latency class */ - // other attributes: routing strategies -}; - -/* extension for audio port structure when the audio port is an audio session */ -struct audio_port_session_ext { - audio_session_t session; /* audio session */ -}; - -struct audio_port { - audio_port_handle_t id; /* port unique ID */ - audio_port_role_t role; /* sink or source */ - audio_port_type_t type; /* device, mix ... */ - char name[AUDIO_PORT_MAX_NAME_LEN]; - unsigned int num_sample_rates; /* number of sampling rates in following array */ - unsigned int sample_rates[AUDIO_PORT_MAX_SAMPLING_RATES]; - unsigned int num_channel_masks; /* number of channel masks in following array */ - audio_channel_mask_t channel_masks[AUDIO_PORT_MAX_CHANNEL_MASKS]; - unsigned int num_formats; /* number of formats in following array */ - audio_format_t formats[AUDIO_PORT_MAX_FORMATS]; - unsigned int num_gains; /* number of gains in following array */ - struct audio_gain gains[AUDIO_PORT_MAX_GAINS]; - struct audio_port_config active_config; /* current audio port configuration */ - union { - struct audio_port_device_ext device; - struct audio_port_mix_ext mix; - struct audio_port_session_ext session; - } ext; -}; - -/* An audio patch represents a connection between one or more source ports and - * one or more sink ports. Patches are connected and disconnected by audio policy manager or by - * applications via framework APIs. - * Each patch is identified by a handle at the interface used to create that patch. For instance, - * when a patch is created by the audio HAL, the HAL allocates and returns a handle. - * This handle is unique to a given audio HAL hardware module. - * But the same patch receives another system wide unique handle allocated by the framework. - * This unique handle is used for all transactions inside the framework. - */ -typedef int audio_patch_handle_t; - -#define AUDIO_PATCH_PORTS_MAX 16 - -struct audio_patch { - audio_patch_handle_t id; /* patch unique ID */ - unsigned int num_sources; /* number of sources in following array */ - struct audio_port_config sources[AUDIO_PATCH_PORTS_MAX]; - unsigned int num_sinks; /* number of sinks in following array */ - struct audio_port_config sinks[AUDIO_PATCH_PORTS_MAX]; -}; - -/* a HW synchronization source returned by the audio HAL */ -typedef uint32_t audio_hw_sync_t; - -/* an invalid HW synchronization source indicating an error */ -#define AUDIO_HW_SYNC_INVALID 0 - -/** - * Mmap buffer descriptor returned by audio_stream->create_mmap_buffer(). - * note\ Used by streams opened in mmap mode. - */ -struct audio_mmap_buffer_info { - void* shared_memory_address; /**< base address of mmap memory buffer. - For use by local process only */ - int32_t shared_memory_fd; /**< FD for mmap memory buffer */ - int32_t buffer_size_frames; /**< total buffer size in frames */ - int32_t burst_size_frames; /**< transfer size granularity in frames */ -}; - -/** - * Mmap buffer read/write position returned by audio_stream->get_mmap_position(). - * note\ Used by streams opened in mmap mode. - */ -struct audio_mmap_position { - int64_t time_nanoseconds; /**< timestamp in ns, CLOCK_MONOTONIC */ - int32_t position_frames; /**< increasing 32 bit frame count reset when stream->stop() - is called */ -}; - -static inline bool audio_is_output_device(audio_devices_t device) { - if (((device & AUDIO_DEVICE_BIT_IN) == 0) && (popcount(device) == 1) && - ((device & ~AUDIO_DEVICE_OUT_ALL) == 0)) - return true; - else - return false; -} - -static inline bool audio_is_input_device(audio_devices_t device) { - if ((device & AUDIO_DEVICE_BIT_IN) != 0) { - device &= ~AUDIO_DEVICE_BIT_IN; - if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_ALL) == 0)) return true; - } - return false; -} - -static inline bool audio_is_output_devices(audio_devices_t device) { - return (device & AUDIO_DEVICE_BIT_IN) == 0; -} - -static inline bool audio_is_a2dp_in_device(audio_devices_t device) { - if ((device & AUDIO_DEVICE_BIT_IN) != 0) { - device &= ~AUDIO_DEVICE_BIT_IN; - if ((popcount(device) == 1) && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP)) return true; - } - return false; -} - -static inline bool audio_is_a2dp_out_device(audio_devices_t device) { - if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_A2DP)) - return true; - else - return false; -} - -// Deprecated - use audio_is_a2dp_out_device() instead -static inline bool audio_is_a2dp_device(audio_devices_t device) { - return audio_is_a2dp_out_device(device); -} - -static inline bool audio_is_bluetooth_sco_device(audio_devices_t device) { - if ((device & AUDIO_DEVICE_BIT_IN) == 0) { - if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL_SCO) == 0)) return true; - } else { - device &= ~AUDIO_DEVICE_BIT_IN; - if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) == 0)) - return true; - } - - return false; -} - -static inline bool audio_is_usb_out_device(audio_devices_t device) { - return ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_USB)); -} - -static inline bool audio_is_usb_in_device(audio_devices_t device) { - if ((device & AUDIO_DEVICE_BIT_IN) != 0) { - device &= ~AUDIO_DEVICE_BIT_IN; - if (popcount(device) == 1 && (device & AUDIO_DEVICE_IN_ALL_USB) != 0) return true; - } - return false; -} - -/* OBSOLETE - use audio_is_usb_out_device() instead. */ -static inline bool audio_is_usb_device(audio_devices_t device) { - return audio_is_usb_out_device(device); -} - -static inline bool audio_is_remote_submix_device(audio_devices_t device) { - if ((audio_is_output_devices(device) && - (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX) == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) || - (!audio_is_output_devices(device) && - (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) == AUDIO_DEVICE_IN_REMOTE_SUBMIX)) - return true; - else - return false; -} - -/* Returns true if: - * representation is valid, and - * there is at least one channel bit set which _could_ correspond to an input channel, and - * there are no channel bits set which could _not_ correspond to an input channel. - * Otherwise returns false. - */ -static inline bool audio_is_input_channel(audio_channel_mask_t channel) { - uint32_t bits = audio_channel_mask_get_bits(channel); - switch (audio_channel_mask_get_representation(channel)) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - if (bits & ~AUDIO_CHANNEL_IN_ALL) { - bits = 0; - } - // fall through - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - return bits != 0; - default: - return false; - } -} - -/* Returns true if: - * representation is valid, and - * there is at least one channel bit set which _could_ correspond to an output channel, and - * there are no channel bits set which could _not_ correspond to an output channel. - * Otherwise returns false. - */ -static inline bool audio_is_output_channel(audio_channel_mask_t channel) { - uint32_t bits = audio_channel_mask_get_bits(channel); - switch (audio_channel_mask_get_representation(channel)) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - if (bits & ~AUDIO_CHANNEL_OUT_ALL) { - bits = 0; - } - // fall through - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - return bits != 0; - default: - return false; - } -} - -/* Returns the number of channels from an input channel mask, - * used in the context of audio input or recording. - * If a channel bit is set which could _not_ correspond to an input channel, - * it is excluded from the count. - * Returns zero if the representation is invalid. - */ -static inline uint32_t audio_channel_count_from_in_mask(audio_channel_mask_t channel) { - uint32_t bits = audio_channel_mask_get_bits(channel); - switch (audio_channel_mask_get_representation(channel)) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - // TODO: We can now merge with from_out_mask and remove anding - bits &= AUDIO_CHANNEL_IN_ALL; - // fall through - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - return popcount(bits); - default: - return 0; - } -} - -/* Returns the number of channels from an output channel mask, - * used in the context of audio output or playback. - * If a channel bit is set which could _not_ correspond to an output channel, - * it is excluded from the count. - * Returns zero if the representation is invalid. - */ -static inline uint32_t audio_channel_count_from_out_mask(audio_channel_mask_t channel) { - uint32_t bits = audio_channel_mask_get_bits(channel); - switch (audio_channel_mask_get_representation(channel)) { - case AUDIO_CHANNEL_REPRESENTATION_POSITION: - // TODO: We can now merge with from_in_mask and remove anding - bits &= AUDIO_CHANNEL_OUT_ALL; - // fall through - case AUDIO_CHANNEL_REPRESENTATION_INDEX: - return popcount(bits); - default: - return 0; - } -} - -/* Derive a channel mask for index assignment from a channel count. - * Returns the matching channel mask, - * or AUDIO_CHANNEL_NONE if the channel count is zero, - * or AUDIO_CHANNEL_INVALID if the channel count exceeds AUDIO_CHANNEL_COUNT_MAX. - */ -static inline audio_channel_mask_t audio_channel_mask_for_index_assignment_from_count( - uint32_t channel_count) { - if (channel_count == 0) { - return AUDIO_CHANNEL_NONE; - } - if (channel_count > AUDIO_CHANNEL_COUNT_MAX) { - return AUDIO_CHANNEL_INVALID; - } - uint32_t bits = (1 << channel_count) - 1; - return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_INDEX, - bits); -} - -/* Derive an output channel mask for position assignment from a channel count. - * This is to be used when the content channel mask is unknown. The 1, 2, 4, 5, 6, 7 and 8 channel - * cases are mapped to the standard game/home-theater layouts, but note that 4 is mapped to quad, - * and not stereo + FC + mono surround. A channel count of 3 is arbitrarily mapped to stereo + FC - * for continuity with stereo. - * Returns the matching channel mask, - * or AUDIO_CHANNEL_NONE if the channel count is zero, - * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the - * configurations for which a default output channel mask is defined. - */ -static inline audio_channel_mask_t audio_channel_out_mask_from_count(uint32_t channel_count) { - uint32_t bits; - switch (channel_count) { - case 0: - return AUDIO_CHANNEL_NONE; - case 1: - bits = AUDIO_CHANNEL_OUT_MONO; - break; - case 2: - bits = AUDIO_CHANNEL_OUT_STEREO; - break; - case 3: - bits = AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_FRONT_CENTER; - break; - case 4: // 4.0 - bits = AUDIO_CHANNEL_OUT_QUAD; - break; - case 5: // 5.0 - bits = AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER; - break; - case 6: // 5.1 - bits = AUDIO_CHANNEL_OUT_5POINT1; - break; - case 7: // 6.1 - bits = AUDIO_CHANNEL_OUT_5POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER; - break; - case 8: - bits = AUDIO_CHANNEL_OUT_7POINT1; - break; - // FIXME FCC_8 - default: - return AUDIO_CHANNEL_INVALID; - } - return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_POSITION, - bits); -} - -/* Derive a default input channel mask from a channel count. - * Assumes a position mask for mono and stereo, or an index mask for channel counts > 2. - * Returns the matching channel mask, - * or AUDIO_CHANNEL_NONE if the channel count is zero, - * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the - * configurations for which a default input channel mask is defined. - */ -static inline audio_channel_mask_t audio_channel_in_mask_from_count(uint32_t channel_count) { - uint32_t bits; - switch (channel_count) { - case 0: - return AUDIO_CHANNEL_NONE; - case 1: - bits = AUDIO_CHANNEL_IN_MONO; - break; - case 2: - bits = AUDIO_CHANNEL_IN_STEREO; - break; - case 3: - case 4: - case 5: - case 6: - case 7: - case 8: - // FIXME FCC_8 - return audio_channel_mask_for_index_assignment_from_count(channel_count); - default: - return AUDIO_CHANNEL_INVALID; - } - return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_POSITION, - bits); -} - -static inline bool audio_is_valid_format(audio_format_t format) { - switch (format & AUDIO_FORMAT_MAIN_MASK) { - case AUDIO_FORMAT_PCM: - switch (format) { - case AUDIO_FORMAT_PCM_16_BIT: - case AUDIO_FORMAT_PCM_8_BIT: - case AUDIO_FORMAT_PCM_32_BIT: - case AUDIO_FORMAT_PCM_8_24_BIT: - case AUDIO_FORMAT_PCM_FLOAT: - case AUDIO_FORMAT_PCM_24_BIT_PACKED: - return true; - default: - return false; - } - /* not reached */ - case AUDIO_FORMAT_MP3: - case AUDIO_FORMAT_AMR_NB: - case AUDIO_FORMAT_AMR_WB: - case AUDIO_FORMAT_AAC: - case AUDIO_FORMAT_AAC_ADTS: - case AUDIO_FORMAT_HE_AAC_V1: - case AUDIO_FORMAT_HE_AAC_V2: - case AUDIO_FORMAT_VORBIS: - case AUDIO_FORMAT_OPUS: - case AUDIO_FORMAT_AC3: - case AUDIO_FORMAT_E_AC3: - case AUDIO_FORMAT_DTS: - case AUDIO_FORMAT_DTS_HD: - case AUDIO_FORMAT_IEC61937: - case AUDIO_FORMAT_DOLBY_TRUEHD: - case AUDIO_FORMAT_QCELP: - case AUDIO_FORMAT_EVRC: - case AUDIO_FORMAT_EVRCB: - case AUDIO_FORMAT_EVRCWB: - case AUDIO_FORMAT_AAC_ADIF: - case AUDIO_FORMAT_AMR_WB_PLUS: - case AUDIO_FORMAT_MP2: - case AUDIO_FORMAT_EVRCNW: - case AUDIO_FORMAT_FLAC: - case AUDIO_FORMAT_ALAC: - case AUDIO_FORMAT_APE: - case AUDIO_FORMAT_WMA: - case AUDIO_FORMAT_WMA_PRO: - case AUDIO_FORMAT_DSD: - case AUDIO_FORMAT_AC4: - case AUDIO_FORMAT_LDAC: - return true; - default: - return false; - } -} - -/** - * Extract the primary format, eg. PCM, AC3, etc. - */ -static inline audio_format_t audio_get_main_format(audio_format_t format) { - return (audio_format_t)(format & AUDIO_FORMAT_MAIN_MASK); -} - -/** - * Is the data plain PCM samples that can be scaled and mixed? - */ -static inline bool audio_is_linear_pcm(audio_format_t format) { - return (audio_get_main_format(format) == AUDIO_FORMAT_PCM); -} - -/** - * For this format, is the number of PCM audio frames directly proportional - * to the number of data bytes? - * - * In other words, is the format transported as PCM audio samples, - * but not necessarily scalable or mixable. - * This returns true for real PCM, but also for AUDIO_FORMAT_IEC61937, - * which is transported as 16 bit PCM audio, but where the encoded data - * cannot be mixed or scaled. - */ -static inline bool audio_has_proportional_frames(audio_format_t format) { - audio_format_t mainFormat = audio_get_main_format(format); - return (mainFormat == AUDIO_FORMAT_PCM || mainFormat == AUDIO_FORMAT_IEC61937); -} - -static inline size_t audio_bytes_per_sample(audio_format_t format) { - size_t size = 0; - - switch (format) { - case AUDIO_FORMAT_PCM_32_BIT: - case AUDIO_FORMAT_PCM_8_24_BIT: - size = sizeof(int32_t); - break; - case AUDIO_FORMAT_PCM_24_BIT_PACKED: - size = sizeof(uint8_t) * 3; - break; - case AUDIO_FORMAT_PCM_16_BIT: - case AUDIO_FORMAT_IEC61937: - size = sizeof(int16_t); - break; - case AUDIO_FORMAT_PCM_8_BIT: - size = sizeof(uint8_t); - break; - case AUDIO_FORMAT_PCM_FLOAT: - size = sizeof(float); - break; - default: - break; - } - return size; -} - -/* converts device address to string sent to audio HAL via set_parameters */ -static inline char* audio_device_address_to_parameter(audio_devices_t device, const char* address) { - const size_t kSize = AUDIO_DEVICE_MAX_ADDRESS_LEN + sizeof("a2dp_sink_address="); - char param[kSize]; - - if (device & AUDIO_DEVICE_OUT_ALL_A2DP) - snprintf(param, kSize, "%s=%s", "a2dp_sink_address", address); - else if (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX) - snprintf(param, kSize, "%s=%s", "mix", address); - else - snprintf(param, kSize, "%s", address); - - return strdup(param); -} - -static inline bool audio_device_is_digital(audio_devices_t device) { - if ((device & AUDIO_DEVICE_BIT_IN) != 0) { - // input - return (~AUDIO_DEVICE_BIT_IN & device & - (AUDIO_DEVICE_IN_ALL_USB | AUDIO_DEVICE_IN_HDMI | AUDIO_DEVICE_IN_SPDIF | - AUDIO_DEVICE_IN_IP | AUDIO_DEVICE_IN_BUS)) != 0; - } else { - // output - return (device & - (AUDIO_DEVICE_OUT_ALL_USB | AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_HDMI_ARC | - AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_IP | AUDIO_DEVICE_OUT_BUS)) != 0; - } -} - -// Unique effect ID (can be generated from the following site: -// http://www.itu.int/ITU-T/asn1/uuid.html) -// This struct is used for effects identification and in soundtrigger. -typedef struct audio_uuid_s { - uint32_t timeLow; - uint16_t timeMid; - uint16_t timeHiAndVersion; - uint16_t clockSeq; - uint8_t node[6]; -} audio_uuid_t; - -__END_DECLS - -/** - * List of known audio HAL modules. This is the base name of the audio HAL - * library composed of the "audio." prefix, one of the base names below and - * a suffix specific to the device. - * e.g: audio.primary.goldfish.so or audio.a2dp.default.so - * - * The same module names are used in audio policy configuration files. - */ - -#define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary" -#define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp" -#define AUDIO_HARDWARE_MODULE_ID_USB "usb" -#define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix" -#define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload" -#define AUDIO_HARDWARE_MODULE_ID_STUB "stub" - -/** - * Multi-Stream Decoder (MSD) HAL service name. MSD HAL is used to mix - * encoded streams together with PCM streams, producing re-encoded - * streams or PCM streams. - * - * The service must register itself using this name, and audioserver - * tries to instantiate a device factory using this name as well. - * Note that the HIDL implementation library file name *must* have the - * suffix "msd" in order to be picked up by HIDL that is: - * - * android.hardware.audio@x.x-implmsd.so - */ -#define AUDIO_HAL_SERVICE_NAME_MSD "msd" - -/** - * Parameter definitions. - * Note that in the framework code it's recommended to use AudioParameter.h - * instead of these preprocessor defines, and for sure avoid just copying - * the constant values. - */ - -#define AUDIO_PARAMETER_VALUE_ON "on" -#define AUDIO_PARAMETER_VALUE_OFF "off" - -/** - * audio device parameters - */ - -/* BT SCO Noise Reduction + Echo Cancellation parameters */ -#define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec" - -/* Get a new HW synchronization source identifier. - * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs - * or no HW sync is available. */ -#define AUDIO_PARAMETER_HW_AV_SYNC "hw_av_sync" - -/* Screen state */ -#define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state" - -/** - * audio stream parameters - */ - -#define AUDIO_PARAMETER_STREAM_ROUTING "routing" /* audio_devices_t */ -#define AUDIO_PARAMETER_STREAM_FORMAT "format" /* audio_format_t */ -#define AUDIO_PARAMETER_STREAM_CHANNELS "channels" /* audio_channel_mask_t */ -#define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" /* size_t */ -#define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" /* audio_source_t */ -#define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */ - -#define AUDIO_PARAMETER_DEVICE_CONNECT "connect" /* audio_devices_t */ -#define AUDIO_PARAMETER_DEVICE_DISCONNECT "disconnect" /* audio_devices_t */ - -/* Enable mono audio playback if 1, else should be 0. */ -#define AUDIO_PARAMETER_MONO_OUTPUT "mono_output" - -/* Set the HW synchronization source for an output stream. */ -#define AUDIO_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync" - -/* Query supported formats. The response is a '|' separated list of strings from - * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */ -#define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats" -/* Query supported channel masks. The response is a '|' separated list of strings from - * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */ -#define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels" -/* Query supported sampling rates. The response is a '|' separated list of integer values e.g: - * "sup_sampling_rates=44100|48000" */ -#define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates" - -#define AUDIO_PARAMETER_VALUE_LIST_SEPARATOR "|" - -/** - * audio codec parameters - */ - -#define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param" -#define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample" -#define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate" -#define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate" -#define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id" -#define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align" -#define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate" -#define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option" -#define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels" -#define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling" -#define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples" -#define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples" - -// FIXME: a temporary declaration for the incall music flag, will be removed when -// declared in types.hal for audio HAL V4.0 and auto imported to audio-base.h -#define AUDIO_OUTPUT_FLAG_INCALL_MUSIC 0x10000 - -#endif // ANDROID_AUDIO_CORE_H diff --git a/audio/common/all-versions/legacy/include/system/audio_effect.h b/audio/common/all-versions/legacy/include/system/audio_effect.h deleted file mode 100644 index f99f604fb3..0000000000 --- a/audio/common/all-versions/legacy/include/system/audio_effect.h +++ /dev/null @@ -1,528 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_CORE_H -#define ANDROID_AUDIO_EFFECT_CORE_H - -#include "system/audio.h" -#include "system/audio_effect-base.h" - -__BEGIN_DECLS - -///////////////////////////////////////////////// -// Common Definitions -///////////////////////////////////////////////// - -// -//--- Effect descriptor structure effect_descriptor_t -// - -// This format is used for both "type" and "uuid" fields of the effect descriptor structure. -// - When used for effect type and the engine is implementing and effect corresponding to a standard -// OpenSL ES interface, this ID must be the one defined in OpenSLES_IID.h for that interface. -// - When used as uuid, it should be a unique UUID for this particular implementation. -typedef audio_uuid_t effect_uuid_t; - -// Maximum length of character strings in structures defines by this API. -#define EFFECT_STRING_LEN_MAX 64 - -// NULL UUID definition (matches SL_IID_NULL_) -#define EFFECT_UUID_INITIALIZER \ - { \ - 0xec7178ec, 0xe5e1, 0x4432, 0xa3f4, { 0x46, 0x57, 0xe6, 0x79, 0x52, 0x10 } \ - } -static const effect_uuid_t EFFECT_UUID_NULL_ = EFFECT_UUID_INITIALIZER; -static const effect_uuid_t* const EFFECT_UUID_NULL = &EFFECT_UUID_NULL_; -static const char* const EFFECT_UUID_NULL_STR = "ec7178ec-e5e1-4432-a3f4-4657e6795210"; - -// The effect descriptor contains necessary information to facilitate the enumeration of the effect -// engines present in a library. -typedef struct effect_descriptor_s { - effect_uuid_t type; // UUID of to the OpenSL ES interface implemented by this effect - effect_uuid_t uuid; // UUID for this particular implementation - uint32_t apiVersion; // Version of the effect control API implemented - uint32_t flags; // effect engine capabilities/requirements flags (see below) - uint16_t cpuLoad; // CPU load indication (see below) - uint16_t memoryUsage; // Data Memory usage (see below) - char name[EFFECT_STRING_LEN_MAX]; // human readable effect name - char implementor[EFFECT_STRING_LEN_MAX]; // human readable effect implementor name -} effect_descriptor_t; - -///////////////////////////////////////////////// -// Effect control interface -///////////////////////////////////////////////// - -// -//--- Standardized command codes for command() function -// -enum effect_command_e { - EFFECT_CMD_INIT, // initialize effect engine - EFFECT_CMD_SET_CONFIG, // configure effect engine (see effect_config_t) - EFFECT_CMD_RESET, // reset effect engine - EFFECT_CMD_ENABLE, // enable effect process - EFFECT_CMD_DISABLE, // disable effect process - EFFECT_CMD_SET_PARAM, // set parameter immediately (see effect_param_t) - EFFECT_CMD_SET_PARAM_DEFERRED, // set parameter deferred - EFFECT_CMD_SET_PARAM_COMMIT, // commit previous set parameter deferred - EFFECT_CMD_GET_PARAM, // get parameter - EFFECT_CMD_SET_DEVICE, // set audio device (see audio.h, audio_devices_t) - EFFECT_CMD_SET_VOLUME, // set volume - EFFECT_CMD_SET_AUDIO_MODE, // set the audio mode (normal, ring, ...) - EFFECT_CMD_SET_CONFIG_REVERSE, // configure effect engine reverse stream(see effect_config_t) - EFFECT_CMD_SET_INPUT_DEVICE, // set capture device (see audio.h, audio_devices_t) - EFFECT_CMD_GET_CONFIG, // read effect engine configuration - EFFECT_CMD_GET_CONFIG_REVERSE, // read configure effect engine reverse stream configuration - EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS, // get all supported configurations for a feature. - EFFECT_CMD_GET_FEATURE_CONFIG, // get current feature configuration - EFFECT_CMD_SET_FEATURE_CONFIG, // set current feature configuration - EFFECT_CMD_SET_AUDIO_SOURCE, // set the audio source (see audio.h, audio_source_t) - EFFECT_CMD_OFFLOAD, // set if effect thread is an offload one, - // send the ioHandle of the effect thread - EFFECT_CMD_FIRST_PROPRIETARY = 0x10000 // first proprietary command code -}; - -//================================================================================================== -// command: EFFECT_CMD_INIT -//-------------------------------------------------------------------------------------------------- -// description: -// Initialize effect engine: All configurations return to default -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_SET_CONFIG -//-------------------------------------------------------------------------------------------------- -// description: -// Apply new audio parameters configurations for input and output buffers -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(effect_config_t) -// data: effect_config_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_RESET -//-------------------------------------------------------------------------------------------------- -// description: -// Reset the effect engine. Keep configuration but resets state and buffer content -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 0 -// data: N/A -//================================================================================================== -// command: EFFECT_CMD_ENABLE -//-------------------------------------------------------------------------------------------------- -// description: -// Enable the process. Called by the framework before the first call to process() -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_DISABLE -//-------------------------------------------------------------------------------------------------- -// description: -// Disable the process. Called by the framework after the last call to process() -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_SET_PARAM -//-------------------------------------------------------------------------------------------------- -// description: -// Set a parameter and apply it immediately -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(effect_param_t) + size of param and value -// data: effect_param_t + param + value. See effect_param_t definition below for value offset -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_SET_PARAM_DEFERRED -//-------------------------------------------------------------------------------------------------- -// description: -// Set a parameter but apply it only when receiving EFFECT_CMD_SET_PARAM_COMMIT command -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(effect_param_t) + size of param and value -// data: effect_param_t + param + value. See effect_param_t definition below for value offset -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 0 -// data: N/A -//================================================================================================== -// command: EFFECT_CMD_SET_PARAM_COMMIT -//-------------------------------------------------------------------------------------------------- -// description: -// Apply all previously received EFFECT_CMD_SET_PARAM_DEFERRED commands -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_GET_PARAM -//-------------------------------------------------------------------------------------------------- -// description: -// Get a parameter value -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(effect_param_t) + size of param -// data: effect_param_t + param -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(effect_param_t) + size of param and value -// data: effect_param_t + param + value. See effect_param_t definition below for value offset -//================================================================================================== -// command: EFFECT_CMD_SET_DEVICE -//-------------------------------------------------------------------------------------------------- -// description: -// Set the rendering device the audio output path is connected to. See audio.h, audio_devices_t -// for device values. -// The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this -// command when the device changes -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(uint32_t) -// data: uint32_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 0 -// data: N/A -//================================================================================================== -// command: EFFECT_CMD_SET_VOLUME -//-------------------------------------------------------------------------------------------------- -// description: -// Set and get volume. Used by audio framework to delegate volume control to effect engine. -// The effect implementation must set EFFECT_FLAG_VOLUME_IND or EFFECT_FLAG_VOLUME_CTRL flag in -// its descriptor to receive this command before every call to process() function -// If EFFECT_FLAG_VOLUME_CTRL flag is set in the effect descriptor, the effect engine must return -// the volume that should be applied before the effect is processed. The overall volume (the volume -// actually applied by the effect engine multiplied by the returned value) should match the value -// indicated in the command. -//-------------------------------------------------------------------------------------------------- -// command format: -// size: n * sizeof(uint32_t) -// data: volume for each channel defined in effect_config_t for output buffer expressed in -// 8.24 fixed point format -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: n * sizeof(uint32_t) / 0 -// data: - if EFFECT_FLAG_VOLUME_CTRL is set in effect descriptor: -// volume for each channel defined in effect_config_t for output buffer expressed in -// 8.24 fixed point format -// - if EFFECT_FLAG_VOLUME_CTRL is not set in effect descriptor: -// N/A -// It is legal to receive a null pointer as pReplyData in which case the effect framework has -// delegated volume control to another effect -//================================================================================================== -// command: EFFECT_CMD_SET_AUDIO_MODE -//-------------------------------------------------------------------------------------------------- -// description: -// Set the audio mode. The effect implementation must set EFFECT_FLAG_AUDIO_MODE_IND flag in its -// descriptor to receive this command when the audio mode changes. -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(uint32_t) -// data: audio_mode_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 0 -// data: N/A -//================================================================================================== -// command: EFFECT_CMD_SET_CONFIG_REVERSE -//-------------------------------------------------------------------------------------------------- -// description: -// Apply new audio parameters configurations for input and output buffers of reverse stream. -// An example of reverse stream is the echo reference supplied to an Acoustic Echo Canceler. -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(effect_config_t) -// data: effect_config_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(int) -// data: status -//================================================================================================== -// command: EFFECT_CMD_SET_INPUT_DEVICE -//-------------------------------------------------------------------------------------------------- -// description: -// Set the capture device the audio input path is connected to. See audio.h, audio_devices_t -// for device values. -// The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this -// command when the device changes -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(uint32_t) -// data: uint32_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 0 -// data: N/A -//================================================================================================== -// command: EFFECT_CMD_GET_CONFIG -//-------------------------------------------------------------------------------------------------- -// description: -// Read audio parameters configurations for input and output buffers -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(effect_config_t) -// data: effect_config_t -//================================================================================================== -// command: EFFECT_CMD_GET_CONFIG_REVERSE -//-------------------------------------------------------------------------------------------------- -// description: -// Read audio parameters configurations for input and output buffers of reverse stream -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 0 -// data: N/A -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(effect_config_t) -// data: effect_config_t -//================================================================================================== -// command: EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS -//-------------------------------------------------------------------------------------------------- -// description: -// Queries for supported configurations for a particular feature (e.g. get the supported -// combinations of main and auxiliary channels for a noise suppressor). -// The command parameter is the feature identifier (See effect_feature_e for a list of defined -// features) followed by the maximum number of configuration descriptor to return. -// The reply is composed of: -// - status (uint32_t): -// - 0 if feature is supported -// - -ENOSYS if the feature is not supported, -// - -ENOMEM if the feature is supported but the total number of supported configurations -// exceeds the maximum number indicated by the caller. -// - total number of supported configurations (uint32_t) -// - an array of configuration descriptors. -// The actual number of descriptors returned must not exceed the maximum number indicated by -// the caller. -//-------------------------------------------------------------------------------------------------- -// command format: -// size: 2 x sizeof(uint32_t) -// data: effect_feature_e + maximum number of configurations to return -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 2 x sizeof(uint32_t) + n x sizeof () -// data: status + total number of configurations supported + array of n config descriptors -//================================================================================================== -// command: EFFECT_CMD_GET_FEATURE_CONFIG -//-------------------------------------------------------------------------------------------------- -// description: -// Retrieves current configuration for a given feature. -// The reply status is: -// - 0 if feature is supported -// - -ENOSYS if the feature is not supported, -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(uint32_t) -// data: effect_feature_e -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(uint32_t) + sizeof () -// data: status + config descriptor -//================================================================================================== -// command: EFFECT_CMD_SET_FEATURE_CONFIG -//-------------------------------------------------------------------------------------------------- -// description: -// Sets current configuration for a given feature. -// The reply status is: -// - 0 if feature is supported -// - -ENOSYS if the feature is not supported, -// - -EINVAL if the configuration is invalid -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(uint32_t) + sizeof () -// data: effect_feature_e + config descriptor -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(uint32_t) -// data: status -//================================================================================================== -// command: EFFECT_CMD_SET_AUDIO_SOURCE -//-------------------------------------------------------------------------------------------------- -// description: -// Set the audio source the capture path is configured for (Camcorder, voice recognition...). -// See audio.h, audio_source_t for values. -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(uint32_t) -// data: uint32_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: 0 -// data: N/A -//================================================================================================== -// command: EFFECT_CMD_OFFLOAD -//-------------------------------------------------------------------------------------------------- -// description: -// 1.indicate if the playback thread the effect is attached to is offloaded or not -// 2.update the io handle of the playback thread the effect is attached to -//-------------------------------------------------------------------------------------------------- -// command format: -// size: sizeof(effect_offload_param_t) -// data: effect_offload_param_t -//-------------------------------------------------------------------------------------------------- -// reply format: -// size: sizeof(uint32_t) -// data: uint32_t -//-------------------------------------------------------------------------------------------------- -// command: EFFECT_CMD_FIRST_PROPRIETARY -//-------------------------------------------------------------------------------------------------- -// description: -// All proprietary effect commands must use command codes above this value. The size and format of -// command and response fields is free in this case -//================================================================================================== - -// Audio buffer descriptor used by process(), bufferProvider() functions and buffer_config_t -// structure. Multi-channel audio is always interleaved. The channel order is from LSB to MSB with -// regard to the channel mask definition in audio.h, audio_channel_mask_t e.g : -// Stereo: left, right -// 5 point 1: front left, front right, front center, low frequency, back left, back right -// The buffer size is expressed in frame count, a frame being composed of samples for all -// channels at a given time. Frame size for unspecified format (AUDIO_FORMAT_OTHER) is 8 bit by -// definition -typedef struct audio_buffer_s { - size_t frameCount; // number of frames in buffer - union { - void* raw; // raw pointer to start of buffer - float* f32; // pointer to float 32 bit data at start of buffer - int32_t* s32; // pointer to signed 32 bit data at start of buffer - int16_t* s16; // pointer to signed 16 bit data at start of buffer - uint8_t* u8; // pointer to unsigned 8 bit data at start of buffer - }; -} audio_buffer_t; - -// The buffer_provider_s structure contains functions that can be used -// by the effect engine process() function to query and release input -// or output audio buffer. -// The getBuffer() function is called to retrieve a buffer where data -// should read from or written to by process() function. -// The releaseBuffer() function MUST be called when the buffer retrieved -// with getBuffer() is not needed anymore. -// The process function should use the buffer provider mechanism to retrieve -// input or output buffer if the inBuffer or outBuffer passed as argument is NULL -// and the buffer configuration (buffer_config_t) given by the EFFECT_CMD_SET_CONFIG -// command did not specify an audio buffer. - -typedef int32_t (*buffer_function_t)(void* cookie, audio_buffer_t* buffer); - -typedef struct buffer_provider_s { - buffer_function_t getBuffer; // retrieve next buffer - buffer_function_t releaseBuffer; // release used buffer - void* cookie; // for use by client of buffer provider functions -} buffer_provider_t; - -// The buffer_config_s structure specifies the input or output audio format -// to be used by the effect engine. -typedef struct buffer_config_s { - audio_buffer_t buffer; // buffer for use by process() function if not passed explicitly - uint32_t samplingRate; // sampling rate - uint32_t channels; // channel mask (see audio_channel_mask_t in audio.h) - buffer_provider_t bufferProvider; // buffer provider - uint8_t format; // Audio format (see audio_format_t in audio.h) - uint8_t accessMode; // read/write or accumulate in buffer (effect_buffer_access_e) - uint16_t mask; // indicates which of the above fields is valid -} buffer_config_t; - -// EFFECT_FEATURE_AUX_CHANNELS feature configuration descriptor. Describe a combination -// of main and auxiliary channels supported -typedef struct channel_config_s { - audio_channel_mask_t main_channels; // channel mask for main channels - audio_channel_mask_t aux_channels; // channel mask for auxiliary channels -} channel_config_t; - -// effect_config_s structure is used to configure audio parameters and buffers for effect engine -// input and output. -typedef struct effect_config_s { - buffer_config_t inputCfg; - buffer_config_t outputCfg; -} effect_config_t; - -// effect_param_s structure describes the format of the pCmdData argument of EFFECT_CMD_SET_PARAM -// command and pCmdData and pReplyData of EFFECT_CMD_GET_PARAM command. -// psize and vsize represent the actual size of parameter and value. -// -// NOTE: the start of value field inside the data field is always on a 32 bit boundary: -// -// +-----------+ -// | status | sizeof(int) -// +-----------+ -// | psize | sizeof(int) -// +-----------+ -// | vsize | sizeof(int) -// +-----------+ -// | | | | -// ~ parameter ~ > psize | -// | | | > ((psize - 1)/sizeof(int) + 1) * sizeof(int) -// +-----------+ | -// | padding | | -// +-----------+ -// | | | -// ~ value ~ > vsize -// | | | -// +-----------+ - -typedef struct effect_param_s { - int32_t status; // Transaction status (unused for command, used for reply) - uint32_t psize; // Parameter size - uint32_t vsize; // Value size - char data[]; // Start of Parameter + Value data -} effect_param_t; - -// Maximum effect_param_t size -#define EFFECT_PARAM_SIZE_MAX 65536 - -// structure used by EFFECT_CMD_OFFLOAD command -typedef struct effect_offload_param_s { - bool isOffload; // true if the playback thread the effect is attached to is offloaded - int ioHandle; // io handle of the playback thread the effect is attached to -} effect_offload_param_t; - -__END_DECLS - -#endif // ANDROID_AUDIO_EFFECT_CORE_H diff --git a/audio/common/all-versions/util/Android.bp b/audio/common/all-versions/util/Android.bp index 71326670c0..5d33a3a189 100644 --- a/audio/common/all-versions/util/Android.bp +++ b/audio/common/all-versions/util/Android.bp @@ -1,7 +1,10 @@ cc_library_headers { name: "android.hardware.audio.common.util@all-versions", defaults: ["hidl_defaults"], - vendor: true, + vendor_available: true, + vndk: { + enabled: true, + }, export_include_dirs: ["include"], } diff --git a/audio/core/2.0/default/Android.bp b/audio/core/2.0/default/Android.bp index 87e6a9a735..98478860c2 100644 --- a/audio/core/2.0/default/Android.bp +++ b/audio/core/2.0/default/Android.bp @@ -37,13 +37,13 @@ cc_library_shared { "android.hardware.audio.common.util@all-versions", "android.hardware.audio.core@all-versions-impl", "libaudioclient_headers", - "android.hardware.audio.common.legacy@2.0", + "libaudio_system_headers", "libhardware_headers", "libmedia_headers", ], whole_static_libs: [ - "libmedia_helper@2.0", + "libmedia_helper", ], } diff --git a/audio/core/all-versions/default/Android.bp b/audio/core/all-versions/default/Android.bp index a02a6bb7bf..214b8d5b26 100644 --- a/audio/core/all-versions/default/Android.bp +++ b/audio/core/all-versions/default/Android.bp @@ -22,7 +22,7 @@ cc_library_headers { header_libs: [ "libaudioclient_headers", - "android.hardware.audio.common.legacy@2.0", + "libaudio_system_headers", "libhardware_headers", "libmedia_headers", "android.hardware.audio.common.util@all-versions", diff --git a/audio/effect/2.0/default/Android.bp b/audio/effect/2.0/default/Android.bp index d32a9d9240..db0098849c 100644 --- a/audio/effect/2.0/default/Android.bp +++ b/audio/effect/2.0/default/Android.bp @@ -41,9 +41,9 @@ cc_library_shared { header_libs: [ "android.hardware.audio.common.util@all-versions", "android.hardware.audio.effect@all-versions-impl", - "android.hardware.audio.common.legacy@2.0", - "android.hardware.audio.effect.legacy@2.0", + "libaudio_system_headers", "libaudioclient_headers", + "libeffects_headers", "libhardware_headers", "libmedia_headers", ], diff --git a/audio/effect/2.0/legacy/Android.bp b/audio/effect/2.0/legacy/Android.bp deleted file mode 100644 index 68de70e7a7..0000000000 --- a/audio/effect/2.0/legacy/Android.bp +++ /dev/null @@ -1,12 +0,0 @@ -cc_library_headers { - name: "android.hardware.audio.effect.legacy@2.0", - vendor: true, - header_libs: [ - "android.hardware.audio.common.legacy@2.0", - "android.hardware.audio.effect.legacy@all-versions", - ], - export_header_lib_headers: [ - "android.hardware.audio.common.legacy@2.0", - "android.hardware.audio.effect.legacy@all-versions", - ], -} diff --git a/audio/effect/2.0/legacy/OWNERS b/audio/effect/2.0/legacy/OWNERS deleted file mode 100644 index 6fdc97ca29..0000000000 --- a/audio/effect/2.0/legacy/OWNERS +++ /dev/null @@ -1,3 +0,0 @@ -elaurent@google.com -krocard@google.com -mnaganov@google.com diff --git a/audio/effect/all-versions/default/Android.bp b/audio/effect/all-versions/default/Android.bp index 47d74a817f..ed2a093050 100644 --- a/audio/effect/all-versions/default/Android.bp +++ b/audio/effect/all-versions/default/Android.bp @@ -9,6 +9,7 @@ cc_library_headers { shared_libs: [ "libbase", "libcutils", + "libeffects", "libfmq", "libhidlbase", "libhidlmemory", @@ -20,7 +21,9 @@ cc_library_headers { ], header_libs: [ + "libaudio_system_headers", "libaudioclient_headers", + "libeffects_headers", "libhardware_headers", "libmedia_headers", "android.hardware.audio.common.util@all-versions", diff --git a/audio/effect/all-versions/legacy/Android.bp b/audio/effect/all-versions/legacy/Android.bp deleted file mode 100644 index bcf81b3da0..0000000000 --- a/audio/effect/all-versions/legacy/Android.bp +++ /dev/null @@ -1,11 +0,0 @@ -cc_library_headers { - name: "android.hardware.audio.effect.legacy@all-versions", - vendor: true, - export_include_dirs: ["include"], - header_libs: [ - "android.hardware.audio.common.legacy@all-versions", - ], - export_header_lib_headers: [ - "android.hardware.audio.common.legacy@all-versions", - ], -} diff --git a/audio/effect/all-versions/legacy/OWNERS b/audio/effect/all-versions/legacy/OWNERS deleted file mode 100644 index 6fdc97ca29..0000000000 --- a/audio/effect/all-versions/legacy/OWNERS +++ /dev/null @@ -1,3 +0,0 @@ -elaurent@google.com -krocard@google.com -mnaganov@google.com diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_aec.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_aec.h deleted file mode 100644 index f48749a4ea..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_aec.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_AEC_H_ -#define ANDROID_EFFECT_AEC_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_AEC_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_agc.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_agc.h deleted file mode 100644 index 466ea96fda..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_agc.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_AGC_H_ -#define ANDROID_EFFECT_AGC_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_AGC_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_bassboost.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_bassboost.h deleted file mode 100644 index 157452e2f7..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_bassboost.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_BASSBOOST_H_ -#define ANDROID_EFFECT_BASSBOOST_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_BASSBOOST_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_downmix.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_downmix.h deleted file mode 100644 index 26b849bff1..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_downmix.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (C) 2012 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_DOWNMIX_H_ -#define ANDROID_EFFECT_DOWNMIX_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_DOWNMIX_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_environmentalreverb.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_environmentalreverb.h deleted file mode 100644 index dd474c25bc..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_environmentalreverb.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_ENVIRONMENTALREVERB_H_ -#define ANDROID_EFFECT_ENVIRONMENTALREVERB_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_ENVIRONMENTALREVERB_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_equalizer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_equalizer.h deleted file mode 100644 index 3059ec20e5..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_equalizer.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_EQUALIZER_H_ -#define ANDROID_EFFECT_EQUALIZER_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_EQUALIZER_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_loudnessenhancer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_loudnessenhancer.h deleted file mode 100644 index f37ba458d2..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_loudnessenhancer.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * Copyright (C) 2013 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_LOUDNESS_ENHANCER_H_ -#define ANDROID_EFFECT_LOUDNESS_ENHANCER_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_LOUDNESS_ENHANCER_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_ns.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_ns.h deleted file mode 100644 index 3bd8a41b6d..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_ns.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_NS_H_ -#define ANDROID_EFFECT_NS_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_NS_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_presetreverb.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_presetreverb.h deleted file mode 100644 index eac1f5f205..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_presetreverb.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_PRESETREVERB_H_ -#define ANDROID_EFFECT_PRESETREVERB_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_PRESETREVERB_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_virtualizer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_virtualizer.h deleted file mode 100644 index aeecfa5195..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_virtualizer.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_VIRTUALIZER_H_ -#define ANDROID_EFFECT_VIRTUALIZER_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_VIRTUALIZER_H_*/ diff --git a/audio/effect/all-versions/legacy/include/audio_effects/effect_visualizer.h b/audio/effect/all-versions/legacy/include/audio_effects/effect_visualizer.h deleted file mode 100644 index 47217e7776..0000000000 --- a/audio/effect/all-versions/legacy/include/audio_effects/effect_visualizer.h +++ /dev/null @@ -1,33 +0,0 @@ -/* - * 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. - */ - -/* - * USAGE NOTE: Only include this header when _implementing_ a particular - * effect. When access to UUID and properties is enough, include the - * corresponding header from system/audio_effects/, which doesn't include - * hardware/audio_effect.h. - * - * Only code that immediately calls into HAL or implements an effect - * can import hardware/audio_effect.h. - */ - -#ifndef ANDROID_EFFECT_VISUALIZER_H_ -#define ANDROID_EFFECT_VISUALIZER_H_ - -#include -#include - -#endif /*ANDROID_EFFECT_VISUALIZER_H_*/ diff --git a/audio/effect/all-versions/legacy/include/media/EffectsFactoryApi.h b/audio/effect/all-versions/legacy/include/media/EffectsFactoryApi.h deleted file mode 100644 index e08fd0b3bc..0000000000 --- a/audio/effect/all-versions/legacy/include/media/EffectsFactoryApi.h +++ /dev/null @@ -1,188 +0,0 @@ -/* - * Copyright (C) 2010 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_EFFECTSFACTORYAPI_H_ -#define ANDROID_EFFECTSFACTORYAPI_H_ - -#include -#include -#include -#include -#include - -#if __cplusplus -extern "C" { -#endif - -///////////////////////////////////////////////// -// Effect factory interface -///////////////////////////////////////////////// - -//////////////////////////////////////////////////////////////////////////////// -// -// Function: EffectQueryNumberEffects -// -// Description: Returns the number of different effects in all loaded libraries. -// Each effect must have a different effect uuid (see -// effect_descriptor_t). This function together with EffectQueryEffect() -// is used to enumerate all effects present in all loaded libraries. -// Each time EffectQueryNumberEffects() is called, the factory must -// reset the index of the effect descriptor returned by next call to -// EffectQueryEffect() to restart enumeration from the beginning. -// -// Input/Output: -// pNumEffects: address where the number of effects should be returned. -// -// Output: -// returned value: 0 successful operation. -// -ENODEV factory failed to initialize -// -EINVAL invalid pNumEffects -// *pNumEffects: updated with number of effects in factory -// -//////////////////////////////////////////////////////////////////////////////// -ANDROID_API -int EffectQueryNumberEffects(uint32_t* pNumEffects); - -//////////////////////////////////////////////////////////////////////////////// -// -// Function: EffectQueryEffect -// -// Description: Returns a descriptor of the next available effect. -// See effect_descriptor_t for a details on effect descriptor. -// This function together with EffectQueryNumberEffects() is used to enumerate all -// effects present in all loaded libraries. The enumeration sequence is: -// EffectQueryNumberEffects(&num_effects); -// for (i = 0; i < num_effects; i++) -// EffectQueryEffect(i,...); -// -// Input/Output: -// pDescriptor: address where to return the effect descriptor. -// -// Output: -// returned value: 0 successful operation. -// -ENOENT no more effect available -// -ENODEV factory failed to initialize -// -EINVAL invalid pDescriptor -// -ENOSYS effect list has changed since last execution of -// EffectQueryNumberEffects() -// *pDescriptor: updated with the effect descriptor. -// -//////////////////////////////////////////////////////////////////////////////// -ANDROID_API -int EffectQueryEffect(uint32_t index, effect_descriptor_t* pDescriptor); - -//////////////////////////////////////////////////////////////////////////////// -// -// Function: EffectCreate -// -// Description: Creates an effect engine of the specified type and returns an -// effect control interface on this engine. The function will allocate the -// resources for an instance of the requested effect engine and return -// a handle on the effect control interface. -// -// Input: -// pEffectUuid: pointer to the effect uuid. -// sessionId: audio session to which this effect instance will be attached. All effects -// created with the same session ID are connected in series and process the same signal -// stream. Knowing that two effects are part of the same effect chain can help the -// library implement some kind of optimizations. -// ioId: identifies the output or input stream this effect is directed to at audio HAL. -// For future use especially with tunneled HW accelerated effects -// -// Input/Output: -// pHandle: address where to return the effect handle. -// -// Output: -// returned value: 0 successful operation. -// -ENODEV factory failed to initialize -// -EINVAL invalid pEffectUuid or pHandle -// -ENOENT no effect with this uuid found -// *pHandle: updated with the effect handle. -// -//////////////////////////////////////////////////////////////////////////////// -ANDROID_API -int EffectCreate(const effect_uuid_t* pEffectUuid, int32_t sessionId, int32_t ioId, - effect_handle_t* pHandle); - -//////////////////////////////////////////////////////////////////////////////// -// -// Function: EffectRelease -// -// Description: Releases the effect engine whose handle is given as argument. -// All resources allocated to this particular instance of the effect are -// released. -// -// Input: -// handle: handle on the effect interface to be released. -// -// Output: -// returned value: 0 successful operation. -// -ENODEV factory failed to initialize -// -EINVAL invalid interface handle -// -//////////////////////////////////////////////////////////////////////////////// -ANDROID_API -int EffectRelease(effect_handle_t handle); - -//////////////////////////////////////////////////////////////////////////////// -// -// Function: EffectGetDescriptor -// -// Description: Returns the descriptor of the effect which uuid is pointed -// to by first argument. -// -// Input: -// pEffectUuid: pointer to the effect uuid. -// -// Input/Output: -// pDescriptor: address where to return the effect descriptor. -// -// Output: -// returned value: 0 successful operation. -// -ENODEV factory failed to initialize -// -EINVAL invalid pEffectUuid or pDescriptor -// -ENOENT no effect with this uuid found -// *pDescriptor: updated with the effect descriptor. -// -//////////////////////////////////////////////////////////////////////////////// -ANDROID_API -int EffectGetDescriptor(const effect_uuid_t* pEffectUuid, effect_descriptor_t* pDescriptor); - -//////////////////////////////////////////////////////////////////////////////// -// -// Function: EffectIsNullUuid -// -// Description: Helper function to compare effect uuid to EFFECT_UUID_NULL -// -// Input: -// pEffectUuid: pointer to effect uuid to compare to EFFECT_UUID_NULL. -// -// Output: -// returned value: 0 if uuid is different from EFFECT_UUID_NULL. -// 1 if uuid is equal to EFFECT_UUID_NULL. -// -//////////////////////////////////////////////////////////////////////////////// -ANDROID_API -int EffectIsNullUuid(const effect_uuid_t* pEffectUuid); - -ANDROID_API -int EffectDumpEffects(int fd); - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECTSFACTORYAPI_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/audio_effects_conf.h b/audio/effect/all-versions/legacy/include/system/audio_effects/audio_effects_conf.h deleted file mode 100644 index b68a6c2dac..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/audio_effects_conf.h +++ /dev/null @@ -1,67 +0,0 @@ -/* - * 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_EFFECTS_CONF_H -#define ANDROID_AUDIO_EFFECTS_CONF_H - -///////////////////////////////////////////////// -// Definitions for effects configuration file (audio_effects.conf) -///////////////////////////////////////////////// - -#define AUDIO_EFFECT_DEFAULT_CONFIG_FILE "/system/etc/audio_effects.conf" -#define AUDIO_EFFECT_VENDOR_CONFIG_FILE "/vendor/etc/audio_effects.conf" -#define LIBRARIES_TAG "libraries" -#define PATH_TAG "path" - -#define EFFECTS_TAG "effects" -#define LIBRARY_TAG "library" -#define UUID_TAG "uuid" - -#define PREPROCESSING_TAG "pre_processing" -#define OUTPUT_SESSION_PROCESSING_TAG "output_session_processing" - -#define PARAM_TAG "param" -#define VALUE_TAG "value" -#define INT_TAG "int" -#define SHORT_TAG "short" -#define FLOAT_TAG "float" -#define BOOL_TAG "bool" -#define STRING_TAG "string" - -// audio_source_t -#define MIC_SRC_TAG "mic" // AUDIO_SOURCE_MIC -#define VOICE_UL_SRC_TAG "voice_uplink" // AUDIO_SOURCE_VOICE_UPLINK -#define VOICE_DL_SRC_TAG "voice_downlink" // AUDIO_SOURCE_VOICE_DOWNLINK -#define VOICE_CALL_SRC_TAG "voice_call" // AUDIO_SOURCE_VOICE_CALL -#define CAMCORDER_SRC_TAG "camcorder" // AUDIO_SOURCE_CAMCORDER -#define VOICE_REC_SRC_TAG "voice_recognition" // AUDIO_SOURCE_VOICE_RECOGNITION -#define VOICE_COMM_SRC_TAG "voice_communication" // AUDIO_SOURCE_VOICE_COMMUNICATION -#define UNPROCESSED_SRC_TAG "unprocessed" // AUDIO_SOURCE_UNPROCESSED - -// audio_stream_type_t -#define AUDIO_STREAM_DEFAULT_TAG "default" -#define AUDIO_STREAM_VOICE_CALL_TAG "voice_call" -#define AUDIO_STREAM_SYSTEM_TAG "system" -#define AUDIO_STREAM_RING_TAG "ring" -#define AUDIO_STREAM_MUSIC_TAG "music" -#define AUDIO_STREAM_ALARM_TAG "alarm" -#define AUDIO_STREAM_NOTIFICATION_TAG "notification" -#define AUDIO_STREAM_BLUETOOTH_SCO_TAG "bluetooth_sco" -#define AUDIO_STREAM_ENFORCED_AUDIBLE_TAG "enforced_audible" -#define AUDIO_STREAM_DTMF_TAG "dtmf" -#define AUDIO_STREAM_TTS_TAG "tts" - -#endif // ANDROID_AUDIO_EFFECTS_CONF_H diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_aec.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_aec.h deleted file mode 100644 index 9785055eab..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_aec.h +++ /dev/null @@ -1,44 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_AEC_CORE_H_ -#define ANDROID_EFFECT_AEC_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -// The AEC type UUID is not defined by OpenSL ES and has been generated from -// http://www.itu.int/ITU-T/asn1/uuid.html -static const effect_uuid_t FX_IID_AEC_ = { - 0x7b491460, 0x8d4d, 0x11e0, 0xbd61, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const FX_IID_AEC = &FX_IID_AEC_; - -typedef enum { - AEC_PARAM_ECHO_DELAY, // echo delay in microseconds - AEC_PARAM_PROPERTIES -} t_aec_params; - -// t_equalizer_settings groups all current aec settings for backup and restore. -typedef struct s_aec_settings { uint32_t echoDelay; } t_aec_settings; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_AEC_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_agc.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_agc.h deleted file mode 100644 index 319bcd4ab8..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_agc.h +++ /dev/null @@ -1,50 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_AGC_CORE_H_ -#define ANDROID_EFFECT_AGC_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -// The AGC type UUID is not defined by OpenSL ES and has been generated from -// http://www.itu.int/ITU-T/asn1/uuid.html -static const effect_uuid_t FX_IID_AGC_ = { - 0x0a8abfe0, 0x654c, 0x11e0, 0xba26, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const FX_IID_AGC = &FX_IID_AGC_; - -typedef enum { - AGC_PARAM_TARGET_LEVEL, // target output level in millibel - AGC_PARAM_COMP_GAIN, // gain in the compression range in millibel - AGC_PARAM_LIMITER_ENA, // enable or disable limiter (boolean) - AGC_PARAM_PROPERTIES -} t_agc_params; - -// t_agc_settings groups all current agc settings for backup and restore. -typedef struct s_agc_settings { - int16_t targetLevel; - int16_t compGain; - bool limiterEnabled; -} t_agc_settings; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_AGC_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_bassboost.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_bassboost.h deleted file mode 100644 index 7828d66f5c..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_bassboost.h +++ /dev/null @@ -1,39 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_BASSBOOST_CORE_H_ -#define ANDROID_EFFECT_BASSBOOST_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_BASSBOOST_ = { - 0x0634f220, 0xddd4, 0x11db, 0xa0fc, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const SL_IID_BASSBOOST = &SL_IID_BASSBOOST_; -#endif // OPENSL_ES_H_ - -/* enumerated parameter settings for BassBoost effect */ -typedef enum { BASSBOOST_PARAM_STRENGTH_SUPPORTED, BASSBOOST_PARAM_STRENGTH } t_bassboost_params; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_BASSBOOST_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_downmix.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_downmix.h deleted file mode 100644 index 9f02e41454..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_downmix.h +++ /dev/null @@ -1,50 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_DOWNMIX_CORE_H_ -#define ANDROID_EFFECT_DOWNMIX_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -#define EFFECT_UIID_DOWNMIX__ \ - { \ - 0x381e49cc, 0xa858, 0x4aa2, 0x87f6, { 0xe8, 0x38, 0x8e, 0x76, 0x01, 0xb2 } \ - } -static const effect_uuid_t EFFECT_UIID_DOWNMIX_ = EFFECT_UIID_DOWNMIX__; -const effect_uuid_t* const EFFECT_UIID_DOWNMIX = &EFFECT_UIID_DOWNMIX_; - -/* enumerated parameter settings for downmix effect */ -typedef enum { DOWNMIX_PARAM_TYPE } downmix_params_t; - -typedef enum { - DOWNMIX_TYPE_INVALID = -1, - // throw away the extra channels - DOWNMIX_TYPE_STRIP = 0, - // mix the extra channels with FL/FR - DOWNMIX_TYPE_FOLD = 1, - DOWNMIX_TYPE_CNT, - DOWNMIX_TYPE_LAST = DOWNMIX_TYPE_CNT - 1 -} downmix_type_t; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_DOWNMIX_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_environmentalreverb.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_environmentalreverb.h deleted file mode 100644 index 8caee32064..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_environmentalreverb.h +++ /dev/null @@ -1,67 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_ENVIRONMENTALREVERB_CORE_H_ -#define ANDROID_EFFECT_ENVIRONMENTALREVERB_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_ENVIRONMENTALREVERB_ = { - 0xc2e5d5f0, 0x94bd, 0x4763, 0x9cac, {0x4e, 0x23, 0x4d, 0x6, 0x83, 0x9e}}; -const effect_uuid_t* const SL_IID_ENVIRONMENTALREVERB = &SL_IID_ENVIRONMENTALREVERB_; -#endif // OPENSL_ES_H_ - -/* enumerated parameter settings for environmental reverb effect */ -typedef enum { - // Parameters below are as defined in OpenSL ES specification for environmental reverb interface - REVERB_PARAM_ROOM_LEVEL, // in millibels, range -6000 to 0 - REVERB_PARAM_ROOM_HF_LEVEL, // in millibels, range -4000 to 0 - REVERB_PARAM_DECAY_TIME, // in milliseconds, range 100 to 20000 - REVERB_PARAM_DECAY_HF_RATIO, // in permilles, range 100 to 1000 - REVERB_PARAM_REFLECTIONS_LEVEL, // in millibels, range -6000 to 0 - REVERB_PARAM_REFLECTIONS_DELAY, // in milliseconds, range 0 to 65 - REVERB_PARAM_REVERB_LEVEL, // in millibels, range -6000 to 0 - REVERB_PARAM_REVERB_DELAY, // in milliseconds, range 0 to 65 - REVERB_PARAM_DIFFUSION, // in permilles, range 0 to 1000 - REVERB_PARAM_DENSITY, // in permilles, range 0 to 1000 - REVERB_PARAM_PROPERTIES, - REVERB_PARAM_BYPASS -} t_env_reverb_params; - -// t_reverb_settings is equal to SLEnvironmentalReverbSettings defined in OpenSL ES specification. -typedef struct s_reverb_settings { - int16_t roomLevel; - int16_t roomHFLevel; - uint32_t decayTime; - int16_t decayHFRatio; - int16_t reflectionsLevel; - uint32_t reflectionsDelay; - int16_t reverbLevel; - uint32_t reverbDelay; - int16_t diffusion; - int16_t density; -} __attribute__((packed)) t_reverb_settings; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_ENVIRONMENTALREVERB_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_equalizer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_equalizer.h deleted file mode 100644 index 83fddcfe76..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_equalizer.h +++ /dev/null @@ -1,59 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_EQUALIZER_CORE_H_ -#define ANDROID_EFFECT_EQUALIZER_CORE_H_ - -#include - -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_EQUALIZER_ = { - 0x0bed4300, 0xddd6, 0x11db, 0x8f34, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const SL_IID_EQUALIZER = &SL_IID_EQUALIZER_; -#endif // OPENSL_ES_H_ - -#if __cplusplus -extern "C" { -#endif - -/* enumerated parameters for Equalizer effect */ -typedef enum { - EQ_PARAM_NUM_BANDS, // Gets the number of frequency bands that the equalizer - // supports. - EQ_PARAM_LEVEL_RANGE, // Returns the minimum and maximum band levels supported. - EQ_PARAM_BAND_LEVEL, // Gets/Sets the gain set for the given equalizer band. - EQ_PARAM_CENTER_FREQ, // Gets the center frequency of the given band. - EQ_PARAM_BAND_FREQ_RANGE, // Gets the frequency range of the given frequency band. - EQ_PARAM_GET_BAND, // Gets the band that has the most effect on the given - // frequency. - EQ_PARAM_CUR_PRESET, // Gets/Sets the current preset. - EQ_PARAM_GET_NUM_OF_PRESETS, // Gets the total number of presets the equalizer supports. - EQ_PARAM_GET_PRESET_NAME, // Gets the preset name based on the index. - EQ_PARAM_PROPERTIES // Gets/Sets all parameters at a time. -} t_equalizer_params; - -// t_equalizer_settings groups all current equalizer setting for backup and restore. -typedef struct s_equalizer_settings { - uint16_t curPreset; - uint16_t numBands; - uint16_t bandLevels[]; -} t_equalizer_settings; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_EQUALIZER_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_loudnessenhancer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_loudnessenhancer.h deleted file mode 100644 index 5c780132d5..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_loudnessenhancer.h +++ /dev/null @@ -1,43 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_LOUDNESS_ENHANCER_CORE_H_ -#define ANDROID_EFFECT_LOUDNESS_ENHANCER_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -// this effect is not defined in OpenSL ES as one of the standard effects -static const effect_uuid_t FX_IID_LOUDNESS_ENHANCER_ = { - 0xfe3199be, 0xaed0, 0x413f, 0x87bb, {0x11, 0x26, 0x0e, 0xb6, 0x3c, 0xf1}}; -const effect_uuid_t* const FX_IID_LOUDNESS_ENHANCER = &FX_IID_LOUDNESS_ENHANCER_; - -#define LOUDNESS_ENHANCER_DEFAULT_TARGET_GAIN_MB 0 // mB - -// enumerated parameters for DRC effect -// to keep in sync with frameworks/base/media/java/android/media/audiofx/LoudnessEnhancer.java -typedef enum { - LOUDNESS_ENHANCER_PARAM_TARGET_GAIN_MB = 0, // target gain expressed in mB -} t_level_monitor_params; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_LOUDNESS_ENHANCER_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_ns.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_ns.h deleted file mode 100644 index 8b9ac76404..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_ns.h +++ /dev/null @@ -1,54 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_NS_CORE_H_ -#define ANDROID_EFFECT_NS_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -// The NS type UUID is not defined by OpenSL ES and has been generated from -// http://www.itu.int/ITU-T/asn1/uuid.html -static const effect_uuid_t FX_IID_NS_ = { - 0x58b4b260, 0x8e06, 0x11e0, 0xaa8e, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const FX_IID_NS = &FX_IID_NS_; - -typedef enum { - NS_PARAM_LEVEL, // noise suppression level (t_ns_level) - NS_PARAM_PROPERTIES, - NS_PARAM_TYPE // noise suppression type (t_ns_type) -} t_ns_params; - -// noise suppression level -typedef enum { NS_LEVEL_LOW, NS_LEVEL_MEDIUM, NS_LEVEL_HIGH } t_ns_level; - -// noise suppression type -typedef enum { NS_TYPE_SINGLE_CHANNEL, NS_TYPE_MULTI_CHANNEL } t_ns_type; - -// s_ns_settings groups all current ns settings for backup and restore. -typedef struct s_ns_settings { - uint32_t level; - uint32_t type; -} t_ns_settings; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_NS_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_presetreverb.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_presetreverb.h deleted file mode 100644 index 6804fed160..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_presetreverb.h +++ /dev/null @@ -1,50 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_PRESETREVERB_CORE_H_ -#define ANDROID_EFFECT_PRESETREVERB_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_PRESETREVERB_ = { - 0x47382d60, 0xddd8, 0x11db, 0xbf3a, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const SL_IID_PRESETREVERB = &SL_IID_PRESETREVERB_; -#endif // OPENSL_ES_H_ - -/* enumerated parameter settings for preset reverb effect */ -typedef enum { REVERB_PARAM_PRESET } t_preset_reverb_params; - -typedef enum { - REVERB_PRESET_NONE, - REVERB_PRESET_SMALLROOM, - REVERB_PRESET_MEDIUMROOM, - REVERB_PRESET_LARGEROOM, - REVERB_PRESET_MEDIUMHALL, - REVERB_PRESET_LARGEHALL, - REVERB_PRESET_PLATE, - REVERB_PRESET_LAST = REVERB_PRESET_PLATE -} t_reverb_presets; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_PRESETREVERB_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_virtualizer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_virtualizer.h deleted file mode 100644 index a6a31ec5b1..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_virtualizer.h +++ /dev/null @@ -1,77 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_VIRTUALIZER_CORE_H_ -#define ANDROID_EFFECT_VIRTUALIZER_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_VIRTUALIZER_ = { - 0x37cc2c00, 0xdddd, 0x11db, 0x8577, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const SL_IID_VIRTUALIZER = &SL_IID_VIRTUALIZER_; -#endif // OPENSL_ES_H_ - -/* enumerated parameter settings for virtualizer effect */ -/* to keep in sync with frameworks/base/media/java/android/media/audiofx/Virtualizer.java */ -typedef enum { - VIRTUALIZER_PARAM_STRENGTH_SUPPORTED, - VIRTUALIZER_PARAM_STRENGTH, - // used with EFFECT_CMD_GET_PARAM - // format: - // parameters int32_t VIRTUALIZER_PARAM_VIRTUAL_SPEAKER_ANGLES - // audio_channel_mask_t input channel mask - // audio_devices_t audio output device - // output int32_t* an array of length 3 * the number of channels in the mask - // where entries are the succession of the channel mask - // of each speaker (i.e. a single bit is selected in the - // channel mask) followed by the azimuth and the - // elevation angles. - // status int -EINVAL if configuration is not supported or invalid or not forcing - // 0 if configuration is supported and the mode is forced - // notes: - // - all angles are expressed in degrees and are relative to the listener, - // - for azimuth: 0 is the direction the listener faces, 180 is behind the listener, and - // -90 is to her/his left, - // - for elevation: 0 is the horizontal plane, +90 is above the listener, -90 is below. - VIRTUALIZER_PARAM_VIRTUAL_SPEAKER_ANGLES, - // used with EFFECT_CMD_SET_PARAM - // format: - // parameters int32_t VIRTUALIZER_PARAM_FORCE_VIRTUALIZATION_MODE - // audio_devices_t audio output device - // status int -EINVAL if the device is not supported or invalid - // 0 if the device is supported and the mode is forced, or forcing - // was disabled for the AUDIO_DEVICE_NONE audio device. - VIRTUALIZER_PARAM_FORCE_VIRTUALIZATION_MODE, - // used with EFFECT_CMD_GET_PARAM - // format: - // parameters int32_t VIRTUALIZER_PARAM_VIRTUALIZATION_MODE - // output audio_device_t audio device reflecting the current virtualization mode, - // AUDIO_DEVICE_NONE when not virtualizing - // status int -EINVAL if an error occurred - // 0 if the output value is successfully retrieved - VIRTUALIZER_PARAM_VIRTUALIZATION_MODE -} t_virtualizer_params; - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_VIRTUALIZER_CORE_H_*/ diff --git a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_visualizer.h b/audio/effect/all-versions/legacy/include/system/audio_effects/effect_visualizer.h deleted file mode 100644 index cc78e15d2c..0000000000 --- a/audio/effect/all-versions/legacy/include/system/audio_effects/effect_visualizer.h +++ /dev/null @@ -1,71 +0,0 @@ -/* - * Copyright (C) 2016 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_EFFECT_VISUALIZER_CORE_H_ -#define ANDROID_EFFECT_VISUALIZER_CORE_H_ - -#include - -#if __cplusplus -extern "C" { -#endif - -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_VISUALIZATION_ = { - 0xe46b26a0, 0xdddd, 0x11db, 0x8afd, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; -const effect_uuid_t* const SL_IID_VISUALIZATION = &SL_IID_VISUALIZATION_; -#endif // OPENSL_ES_H_ - -#define VISUALIZER_CAPTURE_SIZE_MAX 1024 // maximum capture size in samples -#define VISUALIZER_CAPTURE_SIZE_MIN 128 // minimum capture size in samples - -// to keep in sync with frameworks/base/media/java/android/media/audiofx/Visualizer.java -#define VISUALIZER_SCALING_MODE_NORMALIZED 0 -#define VISUALIZER_SCALING_MODE_AS_PLAYED 1 - -#define MEASUREMENT_MODE_NONE 0x0 -#define MEASUREMENT_MODE_PEAK_RMS 0x1 - -#define MEASUREMENT_IDX_PEAK 0 -#define MEASUREMENT_IDX_RMS 1 -#define MEASUREMENT_COUNT 2 - -/* enumerated parameters for Visualizer effect */ -typedef enum { - VISUALIZER_PARAM_CAPTURE_SIZE, // Sets the number PCM samples in the capture. - VISUALIZER_PARAM_SCALING_MODE, // Sets the way the captured data is scaled - VISUALIZER_PARAM_LATENCY, // Informs the visualizer about the downstream latency - VISUALIZER_PARAM_MEASUREMENT_MODE, // Sets which measurements are to be made -} t_visualizer_params; - -/* commands */ -typedef enum { - VISUALIZER_CMD_CAPTURE = EFFECT_CMD_FIRST_PROPRIETARY, // Gets the latest PCM capture. - VISUALIZER_CMD_MEASURE, // Gets the current measurements -} t_visualizer_cmds; - -// VISUALIZER_CMD_CAPTURE retrieves the latest PCM snapshot captured by the visualizer engine. -// It returns the number of samples specified by VISUALIZER_PARAM_CAPTURE_SIZE -// in 8 bit unsigned format (0 = 0x80) - -// VISUALIZER_CMD_MEASURE retrieves the lastest measurements as int32_t saved in the -// MEASUREMENT_IDX_* array index order. - -#if __cplusplus -} // extern "C" -#endif - -#endif /*ANDROID_EFFECT_VISUALIZER_CORE_H_*/ diff --git a/soundtrigger/2.0/default/Android.bp b/soundtrigger/2.0/default/Android.bp index 21e50e1f59..cc20f91cd5 100644 --- a/soundtrigger/2.0/default/Android.bp +++ b/soundtrigger/2.0/default/Android.bp @@ -16,7 +16,10 @@ cc_library_shared { name: "android.hardware.soundtrigger@2.0-core", defaults: ["hidl_defaults"], - vendor: true, + vendor_available: true, + vndk: { + enabled: true, + }, srcs: [ "SoundTriggerHalImpl.cpp", ], @@ -34,7 +37,7 @@ cc_library_shared { ], header_libs: [ - "android.hardware.soundtrigger.legacy@2.0", + "libaudio_system_headers", "libhardware_headers", ], } diff --git a/soundtrigger/2.0/default/Android.mk b/soundtrigger/2.0/default/Android.mk index 1b6360b2b9..835a020800 100644 --- a/soundtrigger/2.0/default/Android.mk +++ b/soundtrigger/2.0/default/Android.mk @@ -32,7 +32,6 @@ LOCAL_SHARED_LIBRARIES := \ android.hardware.soundtrigger@2.0-core LOCAL_C_INCLUDE_DIRS := $(LOCAL_PATH) -LOCAL_HEADER_LIBRARIES += android.hardware.soundtrigger.legacy@2.0 ifeq ($(strip $(AUDIOSERVER_MULTILIB)),) LOCAL_MULTILIB := 32 diff --git a/soundtrigger/2.0/legacy/Android.bp b/soundtrigger/2.0/legacy/Android.bp deleted file mode 100644 index 9954779655..0000000000 --- a/soundtrigger/2.0/legacy/Android.bp +++ /dev/null @@ -1,11 +0,0 @@ -cc_library_headers { - name: "android.hardware.soundtrigger.legacy@2.0", - vendor: true, - export_include_dirs: ["include"], - header_libs: [ - "android.hardware.audio.common.legacy@2.0", - ], - export_header_lib_headers: [ - "android.hardware.audio.common.legacy@2.0", - ], -} diff --git a/soundtrigger/2.0/legacy/OWNERS b/soundtrigger/2.0/legacy/OWNERS deleted file mode 100644 index 6fdc97ca29..0000000000 --- a/soundtrigger/2.0/legacy/OWNERS +++ /dev/null @@ -1,3 +0,0 @@ -elaurent@google.com -krocard@google.com -mnaganov@google.com diff --git a/soundtrigger/2.0/legacy/include/hardware/sound_trigger.h b/soundtrigger/2.0/legacy/include/hardware/sound_trigger.h deleted file mode 100644 index 57b405e665..0000000000 --- a/soundtrigger/2.0/legacy/include/hardware/sound_trigger.h +++ /dev/null @@ -1,130 +0,0 @@ -/* - * Copyright (C) 2014 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. - */ - -#include -#include -#include - -#ifndef ANDROID_SOUND_TRIGGER_HAL_H -#define ANDROID_SOUND_TRIGGER_HAL_H - -__BEGIN_DECLS - -/** - * The id of this module - */ -#define SOUND_TRIGGER_HARDWARE_MODULE_ID "sound_trigger" - -/** - * Name of the audio devices to open - */ -#define SOUND_TRIGGER_HARDWARE_INTERFACE "sound_trigger_hw_if" - -#define SOUND_TRIGGER_MODULE_API_VERSION_1_0 HARDWARE_MODULE_API_VERSION(1, 0) -#define SOUND_TRIGGER_MODULE_API_VERSION_CURRENT SOUND_TRIGGER_MODULE_API_VERSION_1_0 - -#define SOUND_TRIGGER_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) -#define SOUND_TRIGGER_DEVICE_API_VERSION_1_1 HARDWARE_DEVICE_API_VERSION(1, 1) -#define SOUND_TRIGGER_DEVICE_API_VERSION_CURRENT SOUND_TRIGGER_DEVICE_API_VERSION_1_1 - -/** - * List of known sound trigger HAL modules. This is the base name of the sound_trigger HAL - * library composed of the "sound_trigger." prefix, one of the base names below and - * a suffix specific to the device. - * e.g: sondtrigger.primary.goldfish.so or sound_trigger.primary.default.so - */ - -#define SOUND_TRIGGER_HARDWARE_MODULE_ID_PRIMARY "primary" - -/** - * 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 sound_trigger_module { - struct hw_module_t common; -}; - -typedef void (*recognition_callback_t)(struct sound_trigger_recognition_event* event, void* cookie); -typedef void (*sound_model_callback_t)(struct sound_trigger_model_event* event, void* cookie); - -struct sound_trigger_hw_device { - struct hw_device_t common; - - /* - * Retrieve implementation properties. - */ - int (*get_properties)(const struct sound_trigger_hw_device* dev, - struct sound_trigger_properties* properties); - - /* - * Load a sound model. Once loaded, recognition of this model can be started and stopped. - * Only one active recognition per model at a time. The SoundTrigger service will handle - * concurrent recognition requests by different users/applications on the same model. - * The implementation returns a unique handle used by other functions (unload_sound_model(), - * start_recognition(), etc... - */ - int (*load_sound_model)(const struct sound_trigger_hw_device* dev, - struct sound_trigger_sound_model* sound_model, - sound_model_callback_t callback, void* cookie, - sound_model_handle_t* handle); - - /* - * Unload a sound model. A sound model can be unloaded to make room for a new one to overcome - * implementation limitations. - */ - int (*unload_sound_model)(const struct sound_trigger_hw_device* dev, - sound_model_handle_t handle); - - /* Start recognition on a given model. Only one recognition active at a time per model. - * Once recognition succeeds of fails, the callback is called. - * TODO: group recognition configuration parameters into one struct and add key phrase options. - */ - int (*start_recognition)(const struct sound_trigger_hw_device* dev, - sound_model_handle_t sound_model_handle, - const struct sound_trigger_recognition_config* config, - recognition_callback_t callback, void* cookie); - - /* Stop recognition on a given model. - * The implementation does not have to call the callback when stopped via this method. - */ - int (*stop_recognition)(const struct sound_trigger_hw_device* dev, - sound_model_handle_t sound_model_handle); - - /* Stop recognition on all models. - * Only supported for device api versions SOUND_TRIGGER_DEVICE_API_VERSION_1_1 or above. - * If no implementation is provided, stop_recognition will be called for each running model. - */ - int (*stop_all_recognitions)(const struct sound_trigger_hw_device* dev); -}; - -typedef struct sound_trigger_hw_device sound_trigger_hw_device_t; - -/** convenience API for opening and closing a supported device */ - -static inline int sound_trigger_hw_device_open(const struct hw_module_t* module, - struct sound_trigger_hw_device** device) { - return module->methods->open(module, SOUND_TRIGGER_HARDWARE_INTERFACE, - TO_HW_DEVICE_T_OPEN(device)); -} - -static inline int sound_trigger_hw_device_close(struct sound_trigger_hw_device* device) { - return device->common.close(&device->common); -} - -__END_DECLS - -#endif // ANDROID_SOUND_TRIGGER_HAL_H diff --git a/soundtrigger/2.0/legacy/include/system/sound_trigger.h b/soundtrigger/2.0/legacy/include/system/sound_trigger.h deleted file mode 100644 index 5d00c1240a..0000000000 --- a/soundtrigger/2.0/legacy/include/system/sound_trigger.h +++ /dev/null @@ -1,228 +0,0 @@ -/* - * Copyright (C) 2014 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_SOUND_TRIGGER_H -#define ANDROID_SOUND_TRIGGER_H - -#include -#include - -#define SOUND_TRIGGER_MAX_STRING_LEN 64 // max length of strings in properties & descriptor structs -#define SOUND_TRIGGER_MAX_LOCALE_LEN 6 // max length of locale string. e.g en_US -#define SOUND_TRIGGER_MAX_USERS 10 // max number of concurrent users -#define SOUND_TRIGGER_MAX_PHRASES 10 // max number of concurrent phrases - -typedef enum { - SOUND_TRIGGER_STATE_NO_INIT = -1, /* The sound trigger service is not initialized */ - SOUND_TRIGGER_STATE_ENABLED = 0, /* The sound trigger service is enabled */ - SOUND_TRIGGER_STATE_DISABLED = 1 /* The sound trigger service is disabled */ -} sound_trigger_service_state_t; - -#define RECOGNITION_MODE_VOICE_TRIGGER 0x1 // simple voice trigger -#define RECOGNITION_MODE_USER_IDENTIFICATION 0x2 // trigger only if one user in model identified -#define RECOGNITION_MODE_USER_AUTHENTICATION 0x4 // trigger only if one user in mode authenticated -#define RECOGNITION_MODE_GENERIC_TRIGGER 0x8 // generic sound trigger - -#define RECOGNITION_STATUS_SUCCESS 0 -#define RECOGNITION_STATUS_ABORT 1 -#define RECOGNITION_STATUS_FAILURE 2 - -#define SOUND_MODEL_STATUS_UPDATED 0 - -typedef enum { - SOUND_MODEL_TYPE_UNKNOWN = -1, /* use for unspecified sound model type */ - SOUND_MODEL_TYPE_KEYPHRASE = 0, /* use for key phrase sound models */ - SOUND_MODEL_TYPE_GENERIC = 1 /* use for all models other than keyphrase */ -} sound_trigger_sound_model_type_t; - -typedef audio_uuid_t sound_trigger_uuid_t; - -/* - * sound trigger implementation descriptor read by the framework via get_properties(). - * Used by SoundTrigger service to report to applications and manage concurrency and policy. - */ -struct sound_trigger_properties { - char implementor[SOUND_TRIGGER_MAX_STRING_LEN]; /* implementor name */ - char description[SOUND_TRIGGER_MAX_STRING_LEN]; /* implementation description */ - unsigned int version; /* implementation version */ - sound_trigger_uuid_t uuid; /* unique implementation ID. - Must change with version each version */ - unsigned int max_sound_models; /* maximum number of concurrent sound models - loaded */ - unsigned int max_key_phrases; /* maximum number of key phrases */ - unsigned int max_users; /* maximum number of concurrent users detected */ - unsigned int recognition_modes; /* all supported modes. - e.g RECOGNITION_MODE_VOICE_TRIGGER */ - bool capture_transition; /* supports seamless transition from detection - to capture */ - unsigned int max_buffer_ms; /* maximum buffering capacity in ms if - capture_transition is true*/ - bool concurrent_capture; /* supports capture by other use cases while - detection is active */ - bool trigger_in_event; /* returns the trigger capture in event */ - unsigned int power_consumption_mw; /* Rated power consumption when detection is active - with TDB silence/sound/speech ratio */ -}; - -typedef int sound_trigger_module_handle_t; - -struct sound_trigger_module_descriptor { - sound_trigger_module_handle_t handle; - struct sound_trigger_properties properties; -}; - -typedef int sound_model_handle_t; - -/* - * Base sound model descriptor. This struct is the header of a larger block passed to - * load_sound_model() and containing the binary data of the sound model. - * Proprietary representation of users in binary data must match information indicated - * by users field - */ -struct sound_trigger_sound_model { - sound_trigger_sound_model_type_t type; /* model type. e.g. SOUND_MODEL_TYPE_KEYPHRASE */ - sound_trigger_uuid_t uuid; /* unique sound model ID. */ - sound_trigger_uuid_t vendor_uuid; /* unique vendor ID. Identifies the engine the - sound model was build for */ - unsigned int data_size; /* size of opaque model data */ - unsigned int data_offset; /* offset of opaque data start from head of struct - (e.g sizeof struct sound_trigger_sound_model) */ -}; - -/* key phrase descriptor */ -struct sound_trigger_phrase { - unsigned int id; /* keyphrase ID */ - unsigned int recognition_mode; /* recognition modes supported by this key phrase */ - unsigned int num_users; /* number of users in the key phrase */ - unsigned int users[SOUND_TRIGGER_MAX_USERS]; /* users ids: (not uid_t but sound trigger - specific IDs */ - char locale[SOUND_TRIGGER_MAX_LOCALE_LEN]; /* locale - JAVA Locale style (e.g. en_US) */ - char text[SOUND_TRIGGER_MAX_STRING_LEN]; /* phrase text in UTF-8 format. */ -}; - -/* - * Specialized sound model for key phrase detection. - * Proprietary representation of key phrases in binary data must match information indicated - * by phrases field - */ -struct sound_trigger_phrase_sound_model { - struct sound_trigger_sound_model common; - unsigned int num_phrases; /* number of key phrases in model */ - struct sound_trigger_phrase phrases[SOUND_TRIGGER_MAX_PHRASES]; -}; - -/* - * Generic sound model, used for all cases except key phrase detection. - */ -struct sound_trigger_generic_sound_model { - struct sound_trigger_sound_model common; -}; - -/* - * Generic recognition event sent via recognition callback - * Must be aligned to transmit as raw memory through Binder. - */ -struct __attribute__((aligned(8))) sound_trigger_recognition_event { - int status; /* recognition status e.g. - RECOGNITION_STATUS_SUCCESS */ - sound_trigger_sound_model_type_t type; /* event type, same as sound model type. - e.g. SOUND_MODEL_TYPE_KEYPHRASE */ - sound_model_handle_t model; /* loaded sound model that triggered the - event */ - bool capture_available; /* it is possible to capture audio from this - utterance buffered by the - implementation */ - int capture_session; /* audio session ID. framework use */ - int capture_delay_ms; /* delay in ms between end of model - detection and start of audio available - for capture. A negative value is possible - (e.g. if key phrase is also available for - capture */ - int capture_preamble_ms; /* duration in ms of audio captured - before the start of the trigger. - 0 if none. */ - bool trigger_in_data; /* the opaque data is the capture of - the trigger sound */ - audio_config_t audio_config; /* audio format of either the trigger in - event data or to use for capture of the - rest of the utterance */ - unsigned int data_size; /* size of opaque event data */ - unsigned int data_offset; /* offset of opaque data start from start of - this struct (e.g sizeof struct - sound_trigger_phrase_recognition_event) */ -}; - -/* - * Confidence level for each user in struct sound_trigger_phrase_recognition_extra - */ -struct sound_trigger_confidence_level { - unsigned int user_id; /* user ID */ - unsigned int level; /* confidence level in percent (0 - 100). - - min level for recognition configuration - - detected level for recognition event */ -}; - -/* - * Specialized recognition event for key phrase detection - */ -struct sound_trigger_phrase_recognition_extra { - unsigned int id; /* keyphrase ID */ - unsigned int recognition_modes; /* recognition modes used for this keyphrase */ - unsigned int confidence_level; /* confidence level for mode RECOGNITION_MODE_VOICE_TRIGGER */ - unsigned int num_levels; /* number of user confidence levels */ - struct sound_trigger_confidence_level levels[SOUND_TRIGGER_MAX_USERS]; -}; - -struct sound_trigger_phrase_recognition_event { - struct sound_trigger_recognition_event common; - unsigned int num_phrases; - struct sound_trigger_phrase_recognition_extra phrase_extras[SOUND_TRIGGER_MAX_PHRASES]; -}; - -struct sound_trigger_generic_recognition_event { - struct sound_trigger_recognition_event common; -}; - -/* - * configuration for sound trigger capture session passed to start_recognition() - */ -struct sound_trigger_recognition_config { - audio_io_handle_t capture_handle; /* IO handle that will be used for capture. - N/A if capture_requested is false */ - audio_devices_t capture_device; /* input device requested for detection capture */ - bool capture_requested; /* capture and buffer audio for this recognition - instance */ - unsigned int num_phrases; /* number of key phrases recognition extras */ - struct sound_trigger_phrase_recognition_extra phrases[SOUND_TRIGGER_MAX_PHRASES]; - /* configuration for each key phrase */ - unsigned int data_size; /* size of opaque capture configuration data */ - unsigned int data_offset; /* offset of opaque data start from start of this struct - (e.g sizeof struct sound_trigger_recognition_config) */ -}; - -/* - * Event sent via load sound model callback - */ -struct sound_trigger_model_event { - int status; /* sound model status e.g. SOUND_MODEL_STATUS_UPDATED */ - sound_model_handle_t model; /* loaded sound model that triggered the event */ - unsigned int data_size; /* size of event data if any. Size of updated sound model if - status is SOUND_MODEL_STATUS_UPDATED */ - unsigned int data_offset; /* offset of data start from start of this struct - (e.g sizeof struct sound_trigger_model_event) */ -}; - -#endif // ANDROID_SOUND_TRIGGER_H diff --git a/soundtrigger/2.1/default/Android.mk b/soundtrigger/2.1/default/Android.mk index 04d3f3650a..5851d63eda 100644 --- a/soundtrigger/2.1/default/Android.mk +++ b/soundtrigger/2.1/default/Android.mk @@ -38,8 +38,6 @@ LOCAL_SHARED_LIBRARIES := \ android.hidl.allocator@1.0 \ android.hidl.memory@1.0 -LOCAL_HEADER_LIBRARIES := android.hardware.soundtrigger.legacy@2.0 - LOCAL_C_INCLUDE_DIRS := $(LOCAL_PATH) ifeq ($(strip $(AUDIOSERVER_MULTILIB)),) diff --git a/tv/input/1.0/default/Android.bp b/tv/input/1.0/default/Android.bp index c4222301e4..7c140a5c41 100644 --- a/tv/input/1.0/default/Android.bp +++ b/tv/input/1.0/default/Android.bp @@ -16,9 +16,6 @@ cc_library_shared { "android.hardware.tv.input@1.0", ], - header_libs: [ - "android.hardware.audio.common.legacy@2.0", - ], } cc_binary {