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