Revert "Audio V4: Split system and vendor Audio.h"
This reverts commit fc9e212f018ce41430ec8dcebc3cae3c346474bb.
Reason for revert: Breaks multiple devices
Change-Id: I816671fd92246f85c97d00819858a74e36e2929d
diff --git a/include/hardware/audio.h b/include/hardware/audio.h
new file mode 100644
index 0000000..2d6eb30
--- /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