From 58c68737d8535ba922dbcd893479980a7cfba130 Mon Sep 17 00:00:00 2001 From: Mister Oyster Date: Tue, 31 Oct 2017 16:53:53 +0100 Subject: [PATCH] include: add @danielhk's audio headers with its' legacy_audio_stream_in, thanks to him --- include/hardware/audio.h | 754 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 754 insertions(+) create mode 100644 include/hardware/audio.h diff --git a/include/hardware/audio.h b/include/hardware/audio.h new file mode 100644 index 0000000..f0e4890 --- /dev/null +++ b/include/hardware/audio.h @@ -0,0 +1,754 @@ +/* + * Copyright (C) 2017 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; + +struct legacy_audio_stream_in { + struct audio_stream common; + int (*set_gain)(struct audio_stream_in *stream, float gain); + ssize_t (*read)(struct audio_stream_in *stream, void* buffer, + size_t bytes); + uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream); +}; +typedef struct legacy_audio_stream_in legacy_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