diff options
author | Kevin Rocard <krocard@google.com> | 2018-01-24 06:26:16 +0000 |
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committer | Android (Google) Code Review <android-gerrit@google.com> | 2018-01-24 06:26:16 +0000 |
commit | 52a0dad79c63db6d8588809fcb9e702ada01b9ed (patch) | |
tree | 7357a5da0ac64400c2d308cbb180374139018e06 | |
parent | a03b72961d7c1a2b0ab2ef0ff89ef4377bf6eb63 (diff) | |
parent | c6ec9486d80a58a2f9fdaff2a2660dec125aac28 (diff) | |
download | libhardware-52a0dad79c63db6d8588809fcb9e702ada01b9ed.tar.gz |
Merge "Revert "Audio V4: Split system and vendor Audio.h""
-rw-r--r-- | Android.bp | 2 | ||||
-rw-r--r-- | include/hardware/audio.h | 745 | ||||
-rw-r--r-- | include/hardware/audio_alsaops.h | 103 | ||||
-rw-r--r-- | include/hardware/audio_effect.h | 310 | ||||
-rw-r--r-- | include/hardware/audio_policy.h | 457 | ||||
-rw-r--r-- | modules/audio/Android.bp | 22 | ||||
-rw-r--r-- | modules/audio/audio_policy.c | 350 | ||||
-rw-r--r-- | modules/audio_remote_submix/Android.bp | 9 | ||||
-rw-r--r-- | modules/usbaudio/Android.bp | 4 |
9 files changed, 1988 insertions, 14 deletions
@@ -3,11 +3,13 @@ cc_library_headers { name: "libhardware_headers", header_libs: [ + "libaudio_system_headers", "libsystem_headers", "libcutils_headers", "libbluetooth-types-header", ], export_header_lib_headers: [ + "libaudio_system_headers", "libsystem_headers", "libcutils_headers", "libbluetooth-types-header", diff --git a/include/hardware/audio.h b/include/hardware/audio.h new file mode 100644 index 00000000..2d6eb309 --- /dev/null +++ b/include/hardware/audio.h @@ -0,0 +1,745 @@ +/* + * Copyright (C) 2011 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#ifndef ANDROID_AUDIO_HAL_INTERFACE_H +#define ANDROID_AUDIO_HAL_INTERFACE_H + +#include <stdint.h> +#include <strings.h> +#include <sys/cdefs.h> +#include <sys/types.h> +#include <time.h> + +#include <cutils/bitops.h> + +#include <hardware/hardware.h> +#include <system/audio.h> +#include <hardware/audio_effect.h> + +__BEGIN_DECLS + +/** + * The id of this module + */ +#define AUDIO_HARDWARE_MODULE_ID "audio" + +/** + * Name of the audio devices to open + */ +#define AUDIO_HARDWARE_INTERFACE "audio_hw_if" + + +/* Use version 0.1 to be compatible with first generation of audio hw module with version_major + * hardcoded to 1. No audio module API change. + */ +#define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1) +#define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1 + +/* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0 + * will be considered of first generation API. + */ +#define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0) +#define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) +#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0) +#define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0) +#define AUDIO_DEVICE_API_VERSION_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/include/hardware/audio_alsaops.h b/include/hardware/audio_alsaops.h new file mode 100644 index 00000000..6a17a351 --- /dev/null +++ b/include/hardware/audio_alsaops.h @@ -0,0 +1,103 @@ +/* + * 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 <log/log.h> + +#include <system/audio.h> +#include <tinyalsa/asoundlib.h> + +__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/include/hardware/audio_effect.h b/include/hardware/audio_effect.h new file mode 100644 index 00000000..3366e178 --- /dev/null +++ b/include/hardware/audio_effect.h @@ -0,0 +1,310 @@ +/* + * 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 <errno.h> +#include <stdint.h> +#include <strings.h> +#include <sys/cdefs.h> +#include <sys/types.h> + +#include <cutils/bitops.h> + +#include <system/audio_effect.h> + + +__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/include/hardware/audio_policy.h b/include/hardware/audio_policy.h new file mode 100644 index 00000000..bacb1e55 --- /dev/null +++ b/include/hardware/audio_policy.h @@ -0,0 +1,457 @@ +/* + * 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 <stdint.h> +#include <sys/cdefs.h> +#include <sys/types.h> + +#include <hardware/hardware.h> + +#include <system/audio.h> +#include <system/audio_policy.h> + +__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/modules/audio/Android.bp b/modules/audio/Android.bp index 7929c66c..a7467c29 100644 --- a/modules/audio/Android.bp +++ b/modules/audio/Android.bp @@ -23,10 +23,7 @@ cc_library_shared { relative_install_path: "hw", proprietary: true, srcs: ["audio_hw.c"], - header_libs: [ - "libhardware_headers", - "android.hardware.audio.common.legacy@2.0", - ], + header_libs: ["libhardware_headers"], shared_libs: [ "liblog", ], @@ -44,10 +41,21 @@ cc_library_shared { relative_install_path: "hw", proprietary: true, srcs: ["audio_hw.c"], - header_libs: [ - "libhardware_headers", - "android.hardware.audio.common.legacy@2.0", + header_libs: ["libhardware_headers"], + shared_libs: [ + "liblog", ], + cflags: ["-Wall", "-Werror", "-Wno-unused-parameter"], +} + +// The stub audio policy HAL module that can be used as a skeleton for +// new implementations. +cc_library_shared { + name: "audio_policy.stub", + relative_install_path: "hw", + proprietary: true, + srcs: ["audio_policy.c"], + header_libs: ["libhardware_headers"], shared_libs: [ "liblog", ], diff --git a/modules/audio/audio_policy.c b/modules/audio/audio_policy.c new file mode 100644 index 00000000..4f9cd5a5 --- /dev/null +++ b/modules/audio/audio_policy.c @@ -0,0 +1,350 @@ +/* + * 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. + */ + +#define LOG_TAG "audio_policy_default" +//#define LOG_NDEBUG 0 + +#include <errno.h> +#include <stdint.h> +#include <stdlib.h> +#include <string.h> + +#include <hardware/hardware.h> +#include <system/audio.h> +#include <system/audio_policy.h> +#include <hardware/audio_policy.h> + +struct default_ap_module { + struct audio_policy_module module; +}; + +struct default_ap_device { + struct audio_policy_device device; +}; + +struct default_audio_policy { + struct audio_policy policy; + + struct audio_policy_service_ops *aps_ops; + void *service; +}; + +static int ap_set_device_connection_state(struct audio_policy *pol, + audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address) +{ + return -ENOSYS; +} + +static audio_policy_dev_state_t ap_get_device_connection_state( + const struct audio_policy *pol, + audio_devices_t device, + const char *device_address) +{ + return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; +} + +static void ap_set_phone_state(struct audio_policy *pol, audio_mode_t state) +{ +} + +// deprecated, never called +static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode, + uint32_t mask) +{ +} + +static void ap_set_force_use(struct audio_policy *pol, + audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) +{ +} + + /* retreive current device category forced for a given usage */ +static audio_policy_forced_cfg_t ap_get_force_use( + const struct audio_policy *pol, + audio_policy_force_use_t usage) +{ + return AUDIO_POLICY_FORCE_NONE; +} + +/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE + * can still be muted. */ +static void ap_set_can_mute_enforced_audible(struct audio_policy *pol, + bool can_mute) +{ +} + +static int ap_init_check(const struct audio_policy *pol) +{ + return 0; +} + +static audio_io_handle_t ap_get_output(struct audio_policy *pol, + audio_stream_type_t stream, + uint32_t sampling_rate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *info) +{ + return 0; +} + +static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output, + audio_stream_type_t stream, int session) +{ + return -ENOSYS; +} + +static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output, + audio_stream_type_t stream, int session) +{ + return -ENOSYS; +} + +static void ap_release_output(struct audio_policy *pol, + audio_io_handle_t output) +{ +} + +static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource, + uint32_t sampling_rate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_in_acoustics_t acoustics) +{ + return 0; +} + +static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input) +{ + return -ENOSYS; +} + +static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input) +{ + return -ENOSYS; +} + +static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input) +{ +} + +static void ap_init_stream_volume(struct audio_policy *pol, + audio_stream_type_t stream, int index_min, + int index_max) +{ +} + +static int ap_set_stream_volume_index(struct audio_policy *pol, + audio_stream_type_t stream, + int index) +{ + return -ENOSYS; +} + +static int ap_get_stream_volume_index(const struct audio_policy *pol, + audio_stream_type_t stream, + int *index) +{ + return -ENOSYS; +} + +static int ap_set_stream_volume_index_for_device(struct audio_policy *pol, + audio_stream_type_t stream, + int index, + audio_devices_t device) +{ + return -ENOSYS; +} + +static int ap_get_stream_volume_index_for_device(const struct audio_policy *pol, + audio_stream_type_t stream, + int *index, + audio_devices_t device) +{ + return -ENOSYS; +} + +static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol, + audio_stream_type_t stream) +{ + return 0; +} + +static audio_devices_t ap_get_devices_for_stream(const struct audio_policy *pol, + audio_stream_type_t stream) +{ + return 0; +} + +static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol, + const struct effect_descriptor_s *desc) +{ + return 0; +} + +static int ap_register_effect(struct audio_policy *pol, + const struct effect_descriptor_s *desc, + audio_io_handle_t output, + uint32_t strategy, + int session, + int id) +{ + return -ENOSYS; +} + +static int ap_unregister_effect(struct audio_policy *pol, int id) +{ + return -ENOSYS; +} + +static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled) +{ + return -ENOSYS; +} + +static bool ap_is_stream_active(const struct audio_policy *pol, audio_stream_type_t stream, + uint32_t in_past_ms) +{ + return false; +} + +static int ap_dump(const struct audio_policy *pol, int fd) +{ + return -ENOSYS; +} + +static bool ap_is_offload_supported(const struct audio_policy *pol, + const audio_offload_info_t *info) +{ + return false; +} + +static int create_default_ap(const struct audio_policy_device *device, + struct audio_policy_service_ops *aps_ops, + void *service, + struct audio_policy **ap) +{ + struct default_audio_policy *dap; + + *ap = NULL; + + if (!service || !aps_ops) + return -EINVAL; + + dap = (struct default_audio_policy *)calloc(1, sizeof(*dap)); + if (!dap) + return -ENOMEM; + + dap->policy.set_device_connection_state = ap_set_device_connection_state; + dap->policy.get_device_connection_state = ap_get_device_connection_state; + dap->policy.set_phone_state = ap_set_phone_state; + dap->policy.set_ringer_mode = ap_set_ringer_mode; + dap->policy.set_force_use = ap_set_force_use; + dap->policy.get_force_use = ap_get_force_use; + dap->policy.set_can_mute_enforced_audible = + ap_set_can_mute_enforced_audible; + dap->policy.init_check = ap_init_check; + dap->policy.get_output = ap_get_output; + dap->policy.start_output = ap_start_output; + dap->policy.stop_output = ap_stop_output; + dap->policy.release_output = ap_release_output; + dap->policy.get_input = ap_get_input; + dap->policy.start_input = ap_start_input; + dap->policy.stop_input = ap_stop_input; + dap->policy.release_input = ap_release_input; + dap->policy.init_stream_volume = ap_init_stream_volume; + dap->policy.set_stream_volume_index = ap_set_stream_volume_index; + dap->policy.get_stream_volume_index = ap_get_stream_volume_index; + dap->policy.set_stream_volume_index_for_device = ap_set_stream_volume_index_for_device; + dap->policy.get_stream_volume_index_for_device = ap_get_stream_volume_index_for_device; + dap->policy.get_strategy_for_stream = ap_get_strategy_for_stream; + dap->policy.get_devices_for_stream = ap_get_devices_for_stream; + dap->policy.get_output_for_effect = ap_get_output_for_effect; + dap->policy.register_effect = ap_register_effect; + dap->policy.unregister_effect = ap_unregister_effect; + dap->policy.set_effect_enabled = ap_set_effect_enabled; + dap->policy.is_stream_active = ap_is_stream_active; + dap->policy.dump = ap_dump; + + dap->policy.is_offload_supported = ap_is_offload_supported; + + dap->service = service; + dap->aps_ops = aps_ops; + + *ap = &dap->policy; + return 0; +} + +static int destroy_default_ap(const struct audio_policy_device *ap_dev, + struct audio_policy *ap) +{ + free(ap); + return 0; +} + +static int default_ap_dev_close(hw_device_t* device) +{ + free(device); + return 0; +} + +static int default_ap_dev_open(const hw_module_t* module, const char* name, + hw_device_t** device) +{ + struct default_ap_device *dev; + + *device = NULL; + + if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0) + return -EINVAL; + + dev = (struct default_ap_device *)calloc(1, sizeof(*dev)); + if (!dev) + return -ENOMEM; + + dev->device.common.tag = HARDWARE_DEVICE_TAG; + dev->device.common.version = 0; + dev->device.common.module = (hw_module_t *)module; + dev->device.common.close = default_ap_dev_close; + dev->device.create_audio_policy = create_default_ap; + dev->device.destroy_audio_policy = destroy_default_ap; + + *device = &dev->device.common; + + return 0; +} + +static struct hw_module_methods_t default_ap_module_methods = { + .open = default_ap_dev_open, +}; + +struct default_ap_module HAL_MODULE_INFO_SYM = { + .module = { + .common = { + .tag = HARDWARE_MODULE_TAG, + .version_major = 1, + .version_minor = 0, + .id = AUDIO_POLICY_HARDWARE_MODULE_ID, + .name = "Default audio policy HAL", + .author = "The Android Open Source Project", + .methods = &default_ap_module_methods, + }, + }, +}; diff --git a/modules/audio_remote_submix/Android.bp b/modules/audio_remote_submix/Android.bp index 578949ed..c7d018c0 100644 --- a/modules/audio_remote_submix/Android.bp +++ b/modules/audio_remote_submix/Android.bp @@ -17,15 +17,16 @@ cc_library_shared { relative_install_path: "hw", vendor: true, srcs: ["audio_hw.cpp"], - + include_dirs: [ + "system/media/audio_utils/include", + ], shared_libs: [ "liblog", "libcutils", "libutils", - "libnbaio_mono@2.0", - "libaudioutils", + "libnbaio_mono", ], - static_libs: ["libmedia_helper@2.0"], + static_libs: ["libmedia_helper"], cflags: ["-Wno-unused-parameter"], diff --git a/modules/usbaudio/Android.bp b/modules/usbaudio/Android.bp index ecc74fbd..c7d403f5 100644 --- a/modules/usbaudio/Android.bp +++ b/modules/usbaudio/Android.bp @@ -25,7 +25,5 @@ cc_library_shared { "libalsautils", ], cflags: ["-Wno-unused-parameter"], - header_libs: [ - "android.hardware.audio.common.legacy@2.0", - ], + header_libs: ["libhardware_headers"], } |