Merge "Revert "Audio V4: Split system and vendor Audio.h""
diff --git a/Android.bp b/Android.bp
index fd138b0..4766b71 100644
--- a/Android.bp
+++ b/Android.bp
@@ -3,11 +3,13 @@
cc_library_headers {
name: "libhardware_headers",
header_libs: [
+ "libaudio_system_headers",
"libsystem_headers",
"libcutils_headers",
"libbluetooth-types-header",
],
export_header_lib_headers: [
+ "libaudio_system_headers",
"libsystem_headers",
"libcutils_headers",
"libbluetooth-types-header",
diff --git a/include/hardware/audio.h b/include/hardware/audio.h
new file mode 100644
index 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
diff --git a/include/hardware/audio_alsaops.h b/include/hardware/audio_alsaops.h
new file mode 100644
index 0000000..6a17a35
--- /dev/null
+++ b/include/hardware/audio_alsaops.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/* This file contains shared utility functions to handle the tinyalsa
+ * implementation for Android internal audio, generally in the hardware layer.
+ * Some routines may log a fatal error on failure, as noted.
+ */
+
+#ifndef ANDROID_AUDIO_ALSAOPS_H
+#define ANDROID_AUDIO_ALSAOPS_H
+
+#include <log/log.h>
+
+#include <system/audio.h>
+#include <tinyalsa/asoundlib.h>
+
+__BEGIN_DECLS
+
+/* Converts audio_format to pcm_format.
+ * Parameters:
+ * format the audio_format_t to convert
+ *
+ * Logs a fatal error if format is not a valid convertible audio_format_t.
+ */
+static inline enum pcm_format pcm_format_from_audio_format(audio_format_t format)
+{
+ switch (format) {
+#if HAVE_BIG_ENDIAN
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return PCM_FORMAT_S16_BE;
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ return PCM_FORMAT_S24_3BE;
+ case AUDIO_FORMAT_PCM_32_BIT:
+ return PCM_FORMAT_S32_BE;
+ case AUDIO_FORMAT_PCM_8_24_BIT:
+ return PCM_FORMAT_S24_BE;
+#else
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return PCM_FORMAT_S16_LE;
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ return PCM_FORMAT_S24_3LE;
+ case AUDIO_FORMAT_PCM_32_BIT:
+ return PCM_FORMAT_S32_LE;
+ case AUDIO_FORMAT_PCM_8_24_BIT:
+ return PCM_FORMAT_S24_LE;
+#endif
+ case AUDIO_FORMAT_PCM_FLOAT: /* there is no equivalent for float */
+ default:
+ LOG_ALWAYS_FATAL("pcm_format_from_audio_format: invalid audio format %#x", format);
+ return 0;
+ }
+}
+
+/* Converts pcm_format to audio_format.
+ * Parameters:
+ * format the pcm_format to convert
+ *
+ * Logs a fatal error if format is not a valid convertible pcm_format.
+ */
+static inline audio_format_t audio_format_from_pcm_format(enum pcm_format format)
+{
+ switch (format) {
+#if HAVE_BIG_ENDIAN
+ case PCM_FORMAT_S16_BE:
+ return AUDIO_FORMAT_PCM_16_BIT;
+ case PCM_FORMAT_S24_3BE:
+ return AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ case PCM_FORMAT_S24_BE:
+ return AUDIO_FORMAT_PCM_8_24_BIT;
+ case PCM_FORMAT_S32_BE:
+ return AUDIO_FORMAT_PCM_32_BIT;
+#else
+ case PCM_FORMAT_S16_LE:
+ return AUDIO_FORMAT_PCM_16_BIT;
+ case PCM_FORMAT_S24_3LE:
+ return AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ case PCM_FORMAT_S24_LE:
+ return AUDIO_FORMAT_PCM_8_24_BIT;
+ case PCM_FORMAT_S32_LE:
+ return AUDIO_FORMAT_PCM_32_BIT;
+#endif
+ default:
+ LOG_ALWAYS_FATAL("audio_format_from_pcm_format: invalid pcm format %#x", format);
+ return 0;
+ }
+}
+
+__END_DECLS
+
+#endif /* ANDROID_AUDIO_ALSAOPS_H */
diff --git a/include/hardware/audio_effect.h b/include/hardware/audio_effect.h
new file mode 100644
index 0000000..3366e17
--- /dev/null
+++ b/include/hardware/audio_effect.h
@@ -0,0 +1,310 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#ifndef ANDROID_AUDIO_EFFECT_H
+#define ANDROID_AUDIO_EFFECT_H
+
+#include <errno.h>
+#include <stdint.h>
+#include <strings.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <cutils/bitops.h>
+
+#include <system/audio_effect.h>
+
+
+__BEGIN_DECLS
+
+
+/////////////////////////////////////////////////
+// Common Definitions
+/////////////////////////////////////////////////
+
+#define EFFECT_MAKE_API_VERSION(M, m) (((M)<<16) | ((m) & 0xFFFF))
+#define EFFECT_API_VERSION_MAJOR(v) ((v)>>16)
+#define EFFECT_API_VERSION_MINOR(v) ((m) & 0xFFFF)
+
+
+/////////////////////////////////////////////////
+// Effect control interface
+/////////////////////////////////////////////////
+
+// Effect control interface version 2.0
+#define EFFECT_CONTROL_API_VERSION EFFECT_MAKE_API_VERSION(2,0)
+
+// Effect control interface structure: effect_interface_s
+// The effect control interface is exposed by each effect engine implementation. It consists of
+// a set of functions controlling the configuration, activation and process of the engine.
+// The functions are grouped in a structure of type effect_interface_s.
+//
+// Effect control interface handle: effect_handle_t
+// The effect_handle_t serves two purposes regarding the implementation of the effect engine:
+// - 1 it is the address of a pointer to an effect_interface_s structure where the functions
+// of the effect control API for a particular effect are located.
+// - 2 it is the address of the context of a particular effect instance.
+// A typical implementation in the effect library would define a structure as follows:
+// struct effect_module_s {
+// const struct effect_interface_s *itfe;
+// effect_config_t config;
+// effect_context_t context;
+// }
+// The implementation of EffectCreate() function would then allocate a structure of this
+// type and return its address as effect_handle_t
+typedef struct effect_interface_s **effect_handle_t;
+
+// Effect control interface definition
+struct effect_interface_s {
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: process
+ //
+ // Description: Effect process function. Takes input samples as specified
+ // (count and location) in input buffer descriptor and output processed
+ // samples as specified in output buffer descriptor. If the buffer descriptor
+ // is not specified the function must use either the buffer or the
+ // buffer provider function installed by the EFFECT_CMD_SET_CONFIG command.
+ // The effect framework will call the process() function after the EFFECT_CMD_ENABLE
+ // command is received and until the EFFECT_CMD_DISABLE is received. When the engine
+ // receives the EFFECT_CMD_DISABLE command it should turn off the effect gracefully
+ // and when done indicate that it is OK to stop calling the process() function by
+ // returning the -ENODATA status.
+ //
+ // NOTE: the process() function implementation should be "real-time safe" that is
+ // it should not perform blocking calls: malloc/free, sleep, read/write/open/close,
+ // pthread_cond_wait/pthread_mutex_lock...
+ //
+ // Input:
+ // self: handle to the effect interface this function
+ // is called on.
+ // inBuffer: buffer descriptor indicating where to read samples to process.
+ // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command.
+ //
+ // outBuffer: buffer descriptor indicating where to write processed samples.
+ // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command.
+ //
+ // Output:
+ // returned value: 0 successful operation
+ // -ENODATA the engine has finished the disable phase and the framework
+ // can stop calling process()
+ // -EINVAL invalid interface handle or
+ // invalid input/output buffer description
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*process)(effect_handle_t self,
+ audio_buffer_t *inBuffer,
+ audio_buffer_t *outBuffer);
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: command
+ //
+ // Description: Send a command and receive a response to/from effect engine.
+ //
+ // Input:
+ // self: handle to the effect interface this function
+ // is called on.
+ // cmdCode: command code: the command can be a standardized command defined in
+ // effect_command_e (see below) or a proprietary command.
+ // cmdSize: size of command in bytes
+ // pCmdData: pointer to command data
+ // pReplyData: pointer to reply data
+ //
+ // Input/Output:
+ // replySize: maximum size of reply data as input
+ // actual size of reply data as output
+ //
+ // Output:
+ // returned value: 0 successful operation
+ // -EINVAL invalid interface handle or
+ // invalid command/reply size or format according to
+ // command code
+ // The return code should be restricted to indicate problems related to this API
+ // specification. Status related to the execution of a particular command should be
+ // indicated as part of the reply field.
+ //
+ // *pReplyData updated with command response
+ //
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*command)(effect_handle_t self,
+ uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t *replySize,
+ void *pReplyData);
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: get_descriptor
+ //
+ // Description: Returns the effect descriptor
+ //
+ // Input:
+ // self: handle to the effect interface this function
+ // is called on.
+ //
+ // Input/Output:
+ // pDescriptor: address where to return the effect descriptor.
+ //
+ // Output:
+ // returned value: 0 successful operation.
+ // -EINVAL invalid interface handle or invalid pDescriptor
+ // *pDescriptor: updated with the effect descriptor.
+ //
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*get_descriptor)(effect_handle_t self,
+ effect_descriptor_t *pDescriptor);
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: process_reverse
+ //
+ // Description: Process reverse stream function. This function is used to pass
+ // a reference stream to the effect engine. If the engine does not need a reference
+ // stream, this function pointer can be set to NULL.
+ // This function would typically implemented by an Echo Canceler.
+ //
+ // Input:
+ // self: handle to the effect interface this function
+ // is called on.
+ // inBuffer: buffer descriptor indicating where to read samples to process.
+ // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command.
+ //
+ // outBuffer: buffer descriptor indicating where to write processed samples.
+ // If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command.
+ // If the buffer and buffer provider in the configuration received by
+ // EFFECT_CMD_SET_CONFIG_REVERSE are also NULL, do not return modified reverse
+ // stream data
+ //
+ // Output:
+ // returned value: 0 successful operation
+ // -ENODATA the engine has finished the disable phase and the framework
+ // can stop calling process_reverse()
+ // -EINVAL invalid interface handle or
+ // invalid input/output buffer description
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*process_reverse)(effect_handle_t self,
+ audio_buffer_t *inBuffer,
+ audio_buffer_t *outBuffer);
+};
+
+/////////////////////////////////////////////////
+// Effect library interface
+/////////////////////////////////////////////////
+
+// Effect library interface version 3.0
+// Note that EffectsFactory.c only checks the major version component, so changes to the minor
+// number can only be used for fully backwards compatible changes
+#define EFFECT_LIBRARY_API_VERSION EFFECT_MAKE_API_VERSION(3,0)
+
+#define AUDIO_EFFECT_LIBRARY_TAG ((('A') << 24) | (('E') << 16) | (('L') << 8) | ('T'))
+
+// Every effect library must have a data structure named AUDIO_EFFECT_LIBRARY_INFO_SYM
+// and the fields of this data structure must begin with audio_effect_library_t
+
+typedef struct audio_effect_library_s {
+ // tag must be initialized to AUDIO_EFFECT_LIBRARY_TAG
+ uint32_t tag;
+ // Version of the effect library API : 0xMMMMmmmm MMMM: Major, mmmm: minor
+ uint32_t version;
+ // Name of this library
+ const char *name;
+ // Author/owner/implementor of the library
+ const char *implementor;
+
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: create_effect
+ //
+ // Description: Creates an effect engine of the specified implementation uuid and
+ // returns an effect control interface on this engine. The function will allocate the
+ // resources for an instance of the requested effect engine and return
+ // a handle on the effect control interface.
+ //
+ // Input:
+ // uuid: pointer to the effect uuid.
+ // sessionId: audio session to which this effect instance will be attached.
+ // All effects created with the same session ID are connected in series and process
+ // the same signal stream. Knowing that two effects are part of the same effect
+ // chain can help the library implement some kind of optimizations.
+ // ioId: identifies the output or input stream this effect is directed to in
+ // audio HAL.
+ // For future use especially with tunneled HW accelerated effects
+ //
+ // Input/Output:
+ // pHandle: address where to return the effect interface handle.
+ //
+ // Output:
+ // returned value: 0 successful operation.
+ // -ENODEV library failed to initialize
+ // -EINVAL invalid pEffectUuid or pHandle
+ // -ENOENT no effect with this uuid found
+ // *pHandle: updated with the effect interface handle.
+ //
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*create_effect)(const effect_uuid_t *uuid,
+ int32_t sessionId,
+ int32_t ioId,
+ effect_handle_t *pHandle);
+
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: release_effect
+ //
+ // Description: Releases the effect engine whose handle is given as argument.
+ // All resources allocated to this particular instance of the effect are
+ // released.
+ //
+ // Input:
+ // handle: handle on the effect interface to be released.
+ //
+ // Output:
+ // returned value: 0 successful operation.
+ // -ENODEV library failed to initialize
+ // -EINVAL invalid interface handle
+ //
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*release_effect)(effect_handle_t handle);
+
+ ////////////////////////////////////////////////////////////////////////////////
+ //
+ // Function: get_descriptor
+ //
+ // Description: Returns the descriptor of the effect engine which implementation UUID is
+ // given as argument.
+ //
+ // Input/Output:
+ // uuid: pointer to the effect uuid.
+ // pDescriptor: address where to return the effect descriptor.
+ //
+ // Output:
+ // returned value: 0 successful operation.
+ // -ENODEV library failed to initialize
+ // -EINVAL invalid pDescriptor or uuid
+ // *pDescriptor: updated with the effect descriptor.
+ //
+ ////////////////////////////////////////////////////////////////////////////////
+ int32_t (*get_descriptor)(const effect_uuid_t *uuid,
+ effect_descriptor_t *pDescriptor);
+} audio_effect_library_t;
+
+// Name of the hal_module_info
+#define AUDIO_EFFECT_LIBRARY_INFO_SYM AELI
+
+// Name of the hal_module_info as a string
+#define AUDIO_EFFECT_LIBRARY_INFO_SYM_AS_STR "AELI"
+
+__END_DECLS
+
+#endif // ANDROID_AUDIO_EFFECT_H
diff --git a/include/hardware/audio_policy.h b/include/hardware/audio_policy.h
new file mode 100644
index 0000000..bacb1e5
--- /dev/null
+++ b/include/hardware/audio_policy.h
@@ -0,0 +1,457 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#ifndef ANDROID_AUDIO_POLICY_INTERFACE_H
+#define ANDROID_AUDIO_POLICY_INTERFACE_H
+
+#include <stdint.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <hardware/hardware.h>
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+
+__BEGIN_DECLS
+
+/**
+ * The id of this module
+ */
+#define AUDIO_POLICY_HARDWARE_MODULE_ID "audio_policy"
+
+/**
+ * Name of the audio devices to open
+ */
+#define AUDIO_POLICY_INTERFACE "policy"
+
+/* ---------------------------------------------------------------------------- */
+
+/*
+ * The audio_policy and audio_policy_service_ops structs define the
+ * communication interfaces between the platform specific audio policy manager
+ * and Android generic audio policy manager.
+ * The platform specific audio policy manager must implement methods of the
+ * audio_policy struct.
+ * This implementation makes use of the audio_policy_service_ops to control
+ * the activity and configuration of audio input and output streams.
+ *
+ * The platform specific audio policy manager is in charge of the audio
+ * routing and volume control policies for a given platform.
+ * The main roles of this module are:
+ * - keep track of current system state (removable device connections, phone
+ * state, user requests...).
+ * System state changes and user actions are notified to audio policy
+ * manager with methods of the audio_policy.
+ *
+ * - process get_output() queries received when AudioTrack objects are
+ * created: Those queries return a handler on an output that has been
+ * selected, configured and opened by the audio policy manager and that
+ * must be used by the AudioTrack when registering to the AudioFlinger
+ * with the createTrack() method.
+ * When the AudioTrack object is released, a release_output() query
+ * is received and the audio policy manager can decide to close or
+ * reconfigure the output depending on other streams using this output and
+ * current system state.
+ *
+ * - similarly process get_input() and release_input() queries received from
+ * AudioRecord objects and configure audio inputs.
+ * - process volume control requests: the stream volume is converted from
+ * an index value (received from UI) to a float value applicable to each
+ * output as a function of platform specific settings and current output
+ * route (destination device). It also make sure that streams are not
+ * muted if not allowed (e.g. camera shutter sound in some countries).
+ */
+
+/* XXX: this should be defined OUTSIDE of frameworks/base */
+struct effect_descriptor_s;
+
+struct audio_policy {
+ /*
+ * configuration functions
+ */
+
+ /* indicate a change in device connection status */
+ int (*set_device_connection_state)(struct audio_policy *pol,
+ audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address);
+
+ /* retrieve a device connection status */
+ audio_policy_dev_state_t (*get_device_connection_state)(
+ const struct audio_policy *pol,
+ audio_devices_t device,
+ const char *device_address);
+
+ /* indicate a change in phone state. Valid phones states are defined
+ * by audio_mode_t */
+ void (*set_phone_state)(struct audio_policy *pol, audio_mode_t state);
+
+ /* deprecated, never called (was "indicate a change in ringer mode") */
+ void (*set_ringer_mode)(struct audio_policy *pol, uint32_t mode,
+ uint32_t mask);
+
+ /* force using a specific device category for the specified usage */
+ void (*set_force_use)(struct audio_policy *pol,
+ audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
+
+ /* retrieve current device category forced for a given usage */
+ audio_policy_forced_cfg_t (*get_force_use)(const struct audio_policy *pol,
+ audio_policy_force_use_t usage);
+
+ /* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
+ * can still be muted. */
+ void (*set_can_mute_enforced_audible)(struct audio_policy *pol,
+ bool can_mute);
+
+ /* check proper initialization */
+ int (*init_check)(const struct audio_policy *pol);
+
+ /*
+ * Audio routing query functions
+ */
+
+ /* request an output appropriate for playback of the supplied stream type and
+ * parameters */
+ audio_io_handle_t (*get_output)(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+
+ /* indicates to the audio policy manager that the output starts being used
+ * by corresponding stream. */
+ int (*start_output)(struct audio_policy *pol,
+ audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+
+ /* indicates to the audio policy manager that the output stops being used
+ * by corresponding stream. */
+ int (*stop_output)(struct audio_policy *pol,
+ audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+
+ /* releases the output. */
+ void (*release_output)(struct audio_policy *pol, audio_io_handle_t output);
+
+ /* request an input appropriate for record from the supplied device with
+ * supplied parameters. */
+ audio_io_handle_t (*get_input)(struct audio_policy *pol, audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_in_acoustics_t acoustics);
+
+ /* indicates to the audio policy manager that the input starts being used */
+ int (*start_input)(struct audio_policy *pol, audio_io_handle_t input);
+
+ /* indicates to the audio policy manager that the input stops being used. */
+ int (*stop_input)(struct audio_policy *pol, audio_io_handle_t input);
+
+ /* releases the input. */
+ void (*release_input)(struct audio_policy *pol, audio_io_handle_t input);
+
+ /*
+ * volume control functions
+ */
+
+ /* initialises stream volume conversion parameters by specifying volume
+ * index range. The index range for each stream is defined by AudioService. */
+ void (*init_stream_volume)(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int index_min,
+ int index_max);
+
+ /* sets the new stream volume at a level corresponding to the supplied
+ * index. The index is within the range specified by init_stream_volume() */
+ int (*set_stream_volume_index)(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int index);
+
+ /* retrieve current volume index for the specified stream */
+ int (*get_stream_volume_index)(const struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int *index);
+
+ /* sets the new stream volume at a level corresponding to the supplied
+ * index for the specified device.
+ * The index is within the range specified by init_stream_volume() */
+ int (*set_stream_volume_index_for_device)(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+
+ /* retrieve current volume index for the specified stream for the specified device */
+ int (*get_stream_volume_index_for_device)(const struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ /* return the strategy corresponding to a given stream type */
+ uint32_t (*get_strategy_for_stream)(const struct audio_policy *pol,
+ audio_stream_type_t stream);
+
+ /* return the enabled output devices for the given stream type */
+ audio_devices_t (*get_devices_for_stream)(const struct audio_policy *pol,
+ audio_stream_type_t stream);
+
+ /* Audio effect management */
+ audio_io_handle_t (*get_output_for_effect)(struct audio_policy *pol,
+ const struct effect_descriptor_s *desc);
+
+ int (*register_effect)(struct audio_policy *pol,
+ const struct effect_descriptor_s *desc,
+ audio_io_handle_t output,
+ uint32_t strategy,
+ audio_session_t session,
+ int id);
+
+ int (*unregister_effect)(struct audio_policy *pol, int id);
+
+ int (*set_effect_enabled)(struct audio_policy *pol, int id, bool enabled);
+
+ bool (*is_stream_active)(const struct audio_policy *pol,
+ audio_stream_type_t stream,
+ uint32_t in_past_ms);
+
+ bool (*is_stream_active_remotely)(const struct audio_policy *pol,
+ audio_stream_type_t stream,
+ uint32_t in_past_ms);
+
+ bool (*is_source_active)(const struct audio_policy *pol,
+ audio_source_t source);
+
+ /* dump state */
+ int (*dump)(const struct audio_policy *pol, int fd);
+
+ /* check if offload is possible for given sample rate, bitrate, duration, ... */
+ bool (*is_offload_supported)(const struct audio_policy *pol,
+ const audio_offload_info_t *info);
+};
+
+
+struct audio_policy_service_ops {
+ /*
+ * Audio output Control functions
+ */
+
+ /* Opens an audio output with the requested parameters.
+ *
+ * The parameter values can indicate to use the default values in case the
+ * audio policy manager has no specific requirements for the output being
+ * opened.
+ *
+ * When the function returns, the parameter values reflect the actual
+ * values used by the audio hardware output stream.
+ *
+ * The audio policy manager can check if the proposed parameters are
+ * suitable or not and act accordingly.
+ */
+ audio_io_handle_t (*open_output)(void *service,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags);
+
+ /* creates a special output that is duplicated to the two outputs passed as
+ * arguments. The duplication is performed by
+ * a special mixer thread in the AudioFlinger.
+ */
+ audio_io_handle_t (*open_duplicate_output)(void *service,
+ audio_io_handle_t output1,
+ audio_io_handle_t output2);
+
+ /* closes the output stream */
+ int (*close_output)(void *service, audio_io_handle_t output);
+
+ /* suspends the output.
+ *
+ * When an output is suspended, the corresponding audio hardware output
+ * stream is placed in standby and the AudioTracks attached to the mixer
+ * thread are still processed but the output mix is discarded.
+ */
+ int (*suspend_output)(void *service, audio_io_handle_t output);
+
+ /* restores a suspended output. */
+ int (*restore_output)(void *service, audio_io_handle_t output);
+
+ /* */
+ /* Audio input Control functions */
+ /* */
+
+ /* opens an audio input
+ * deprecated - new implementations should use open_input_on_module,
+ * and the acoustics parameter is ignored
+ */
+ audio_io_handle_t (*open_input)(void *service,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ audio_in_acoustics_t acoustics);
+
+ /* closes an audio input */
+ int (*close_input)(void *service, audio_io_handle_t input);
+
+ /* */
+ /* misc control functions */
+ /* */
+
+ /* set a stream volume for a particular output.
+ *
+ * For the same user setting, a given stream type can have different
+ * volumes for each output (destination device) it is attached to.
+ */
+ int (*set_stream_volume)(void *service,
+ audio_stream_type_t stream,
+ float volume,
+ audio_io_handle_t output,
+ int delay_ms);
+
+ /* invalidate a stream type, causing a reroute to an unspecified new output */
+ int (*invalidate_stream)(void *service,
+ audio_stream_type_t stream);
+
+ /* function enabling to send proprietary informations directly from audio
+ * policy manager to audio hardware interface. */
+ void (*set_parameters)(void *service,
+ audio_io_handle_t io_handle,
+ const char *kv_pairs,
+ int delay_ms);
+
+ /* function enabling to receive proprietary informations directly from
+ * audio hardware interface to audio policy manager.
+ *
+ * Returns a pointer to a heap allocated string. The caller is responsible
+ * for freeing the memory for it using free().
+ */
+
+ char * (*get_parameters)(void *service, audio_io_handle_t io_handle,
+ const char *keys);
+
+ /* request the playback of a tone on the specified stream.
+ * used for instance to replace notification sounds when playing over a
+ * telephony device during a phone call.
+ */
+ int (*start_tone)(void *service,
+ audio_policy_tone_t tone,
+ audio_stream_type_t stream);
+
+ int (*stop_tone)(void *service);
+
+ /* set down link audio volume. */
+ int (*set_voice_volume)(void *service,
+ float volume,
+ int delay_ms);
+
+ /* move effect to the specified output */
+ int (*move_effects)(void *service,
+ audio_session_t session,
+ audio_io_handle_t src_output,
+ audio_io_handle_t dst_output);
+
+ /* loads an audio hw module.
+ *
+ * The module name passed is the base name of the HW module library, e.g "primary" or "a2dp".
+ * The function returns a handle on the module that will be used to specify a particular
+ * module when calling open_output_on_module() or open_input_on_module()
+ */
+ audio_module_handle_t (*load_hw_module)(void *service,
+ const char *name);
+
+ /* Opens an audio output on a particular HW module.
+ *
+ * Same as open_output() but specifying a specific HW module on which the output must be opened.
+ */
+ audio_io_handle_t (*open_output_on_module)(void *service,
+ audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+
+ /* Opens an audio input on a particular HW module.
+ *
+ * Same as open_input() but specifying a specific HW module on which the input must be opened.
+ * Also removed deprecated acoustics parameter
+ */
+ audio_io_handle_t (*open_input_on_module)(void *service,
+ audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask);
+
+};
+
+/**********************************************************************/
+
+/**
+ * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
+ * and the fields of this data structure must begin with hw_module_t
+ * followed by module specific information.
+ */
+typedef struct audio_policy_module {
+ struct hw_module_t common;
+} audio_policy_module_t;
+
+struct audio_policy_device {
+ /**
+ * Common methods of the audio policy device. This *must* be the first member of
+ * audio_policy_device as users of this structure will cast a hw_device_t to
+ * audio_policy_device pointer in contexts where it's known the hw_device_t references an
+ * audio_policy_device.
+ */
+ struct hw_device_t common;
+
+ int (*create_audio_policy)(const struct audio_policy_device *device,
+ struct audio_policy_service_ops *aps_ops,
+ void *service,
+ struct audio_policy **ap);
+
+ int (*destroy_audio_policy)(const struct audio_policy_device *device,
+ struct audio_policy *ap);
+};
+
+/** convenience API for opening and closing a supported device */
+
+static inline int audio_policy_dev_open(const hw_module_t* module,
+ struct audio_policy_device** device)
+{
+ return module->methods->open(module, AUDIO_POLICY_INTERFACE,
+ (hw_device_t**)device);
+}
+
+static inline int audio_policy_dev_close(struct audio_policy_device* device)
+{
+ return device->common.close(&device->common);
+}
+
+
+__END_DECLS
+
+#endif // ANDROID_AUDIO_POLICY_INTERFACE_H
diff --git a/modules/audio/Android.bp b/modules/audio/Android.bp
index 7929c66..a7467c2 100644
--- a/modules/audio/Android.bp
+++ b/modules/audio/Android.bp
@@ -23,10 +23,7 @@
relative_install_path: "hw",
proprietary: true,
srcs: ["audio_hw.c"],
- header_libs: [
- "libhardware_headers",
- "android.hardware.audio.common.legacy@2.0",
- ],
+ header_libs: ["libhardware_headers"],
shared_libs: [
"liblog",
],
@@ -44,10 +41,21 @@
relative_install_path: "hw",
proprietary: true,
srcs: ["audio_hw.c"],
- header_libs: [
- "libhardware_headers",
- "android.hardware.audio.common.legacy@2.0",
+ header_libs: ["libhardware_headers"],
+ shared_libs: [
+ "liblog",
],
+ cflags: ["-Wall", "-Werror", "-Wno-unused-parameter"],
+}
+
+// The stub audio policy HAL module that can be used as a skeleton for
+// new implementations.
+cc_library_shared {
+ name: "audio_policy.stub",
+ relative_install_path: "hw",
+ proprietary: true,
+ srcs: ["audio_policy.c"],
+ header_libs: ["libhardware_headers"],
shared_libs: [
"liblog",
],
diff --git a/modules/audio/audio_policy.c b/modules/audio/audio_policy.c
new file mode 100644
index 0000000..4f9cd5a
--- /dev/null
+++ b/modules/audio/audio_policy.c
@@ -0,0 +1,350 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_policy_default"
+//#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+
+struct default_ap_module {
+ struct audio_policy_module module;
+};
+
+struct default_ap_device {
+ struct audio_policy_device device;
+};
+
+struct default_audio_policy {
+ struct audio_policy policy;
+
+ struct audio_policy_service_ops *aps_ops;
+ void *service;
+};
+
+static int ap_set_device_connection_state(struct audio_policy *pol,
+ audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address)
+{
+ return -ENOSYS;
+}
+
+static audio_policy_dev_state_t ap_get_device_connection_state(
+ const struct audio_policy *pol,
+ audio_devices_t device,
+ const char *device_address)
+{
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+}
+
+static void ap_set_phone_state(struct audio_policy *pol, audio_mode_t state)
+{
+}
+
+// deprecated, never called
+static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
+ uint32_t mask)
+{
+}
+
+static void ap_set_force_use(struct audio_policy *pol,
+ audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+}
+
+ /* retreive current device category forced for a given usage */
+static audio_policy_forced_cfg_t ap_get_force_use(
+ const struct audio_policy *pol,
+ audio_policy_force_use_t usage)
+{
+ return AUDIO_POLICY_FORCE_NONE;
+}
+
+/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
+ * can still be muted. */
+static void ap_set_can_mute_enforced_audible(struct audio_policy *pol,
+ bool can_mute)
+{
+}
+
+static int ap_init_check(const struct audio_policy *pol)
+{
+ return 0;
+}
+
+static audio_io_handle_t ap_get_output(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ uint32_t sampling_rate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *info)
+{
+ return 0;
+}
+
+static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output,
+ audio_stream_type_t stream, int session)
+{
+ return -ENOSYS;
+}
+
+static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output,
+ audio_stream_type_t stream, int session)
+{
+ return -ENOSYS;
+}
+
+static void ap_release_output(struct audio_policy *pol,
+ audio_io_handle_t output)
+{
+}
+
+static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
+ uint32_t sampling_rate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_in_acoustics_t acoustics)
+{
+ return 0;
+}
+
+static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input)
+{
+ return -ENOSYS;
+}
+
+static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input)
+{
+ return -ENOSYS;
+}
+
+static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input)
+{
+}
+
+static void ap_init_stream_volume(struct audio_policy *pol,
+ audio_stream_type_t stream, int index_min,
+ int index_max)
+{
+}
+
+static int ap_set_stream_volume_index(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int index)
+{
+ return -ENOSYS;
+}
+
+static int ap_get_stream_volume_index(const struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int *index)
+{
+ return -ENOSYS;
+}
+
+static int ap_set_stream_volume_index_for_device(struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
+{
+ return -ENOSYS;
+}
+
+static int ap_get_stream_volume_index_for_device(const struct audio_policy *pol,
+ audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
+{
+ return -ENOSYS;
+}
+
+static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol,
+ audio_stream_type_t stream)
+{
+ return 0;
+}
+
+static audio_devices_t ap_get_devices_for_stream(const struct audio_policy *pol,
+ audio_stream_type_t stream)
+{
+ return 0;
+}
+
+static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
+ const struct effect_descriptor_s *desc)
+{
+ return 0;
+}
+
+static int ap_register_effect(struct audio_policy *pol,
+ const struct effect_descriptor_s *desc,
+ audio_io_handle_t output,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ return -ENOSYS;
+}
+
+static int ap_unregister_effect(struct audio_policy *pol, int id)
+{
+ return -ENOSYS;
+}
+
+static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled)
+{
+ return -ENOSYS;
+}
+
+static bool ap_is_stream_active(const struct audio_policy *pol, audio_stream_type_t stream,
+ uint32_t in_past_ms)
+{
+ return false;
+}
+
+static int ap_dump(const struct audio_policy *pol, int fd)
+{
+ return -ENOSYS;
+}
+
+static bool ap_is_offload_supported(const struct audio_policy *pol,
+ const audio_offload_info_t *info)
+{
+ return false;
+}
+
+static int create_default_ap(const struct audio_policy_device *device,
+ struct audio_policy_service_ops *aps_ops,
+ void *service,
+ struct audio_policy **ap)
+{
+ struct default_audio_policy *dap;
+
+ *ap = NULL;
+
+ if (!service || !aps_ops)
+ return -EINVAL;
+
+ dap = (struct default_audio_policy *)calloc(1, sizeof(*dap));
+ if (!dap)
+ return -ENOMEM;
+
+ dap->policy.set_device_connection_state = ap_set_device_connection_state;
+ dap->policy.get_device_connection_state = ap_get_device_connection_state;
+ dap->policy.set_phone_state = ap_set_phone_state;
+ dap->policy.set_ringer_mode = ap_set_ringer_mode;
+ dap->policy.set_force_use = ap_set_force_use;
+ dap->policy.get_force_use = ap_get_force_use;
+ dap->policy.set_can_mute_enforced_audible =
+ ap_set_can_mute_enforced_audible;
+ dap->policy.init_check = ap_init_check;
+ dap->policy.get_output = ap_get_output;
+ dap->policy.start_output = ap_start_output;
+ dap->policy.stop_output = ap_stop_output;
+ dap->policy.release_output = ap_release_output;
+ dap->policy.get_input = ap_get_input;
+ dap->policy.start_input = ap_start_input;
+ dap->policy.stop_input = ap_stop_input;
+ dap->policy.release_input = ap_release_input;
+ dap->policy.init_stream_volume = ap_init_stream_volume;
+ dap->policy.set_stream_volume_index = ap_set_stream_volume_index;
+ dap->policy.get_stream_volume_index = ap_get_stream_volume_index;
+ dap->policy.set_stream_volume_index_for_device = ap_set_stream_volume_index_for_device;
+ dap->policy.get_stream_volume_index_for_device = ap_get_stream_volume_index_for_device;
+ dap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
+ dap->policy.get_devices_for_stream = ap_get_devices_for_stream;
+ dap->policy.get_output_for_effect = ap_get_output_for_effect;
+ dap->policy.register_effect = ap_register_effect;
+ dap->policy.unregister_effect = ap_unregister_effect;
+ dap->policy.set_effect_enabled = ap_set_effect_enabled;
+ dap->policy.is_stream_active = ap_is_stream_active;
+ dap->policy.dump = ap_dump;
+
+ dap->policy.is_offload_supported = ap_is_offload_supported;
+
+ dap->service = service;
+ dap->aps_ops = aps_ops;
+
+ *ap = &dap->policy;
+ return 0;
+}
+
+static int destroy_default_ap(const struct audio_policy_device *ap_dev,
+ struct audio_policy *ap)
+{
+ free(ap);
+ return 0;
+}
+
+static int default_ap_dev_close(hw_device_t* device)
+{
+ free(device);
+ return 0;
+}
+
+static int default_ap_dev_open(const hw_module_t* module, const char* name,
+ hw_device_t** device)
+{
+ struct default_ap_device *dev;
+
+ *device = NULL;
+
+ if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0)
+ return -EINVAL;
+
+ dev = (struct default_ap_device *)calloc(1, sizeof(*dev));
+ if (!dev)
+ return -ENOMEM;
+
+ dev->device.common.tag = HARDWARE_DEVICE_TAG;
+ dev->device.common.version = 0;
+ dev->device.common.module = (hw_module_t *)module;
+ dev->device.common.close = default_ap_dev_close;
+ dev->device.create_audio_policy = create_default_ap;
+ dev->device.destroy_audio_policy = destroy_default_ap;
+
+ *device = &dev->device.common;
+
+ return 0;
+}
+
+static struct hw_module_methods_t default_ap_module_methods = {
+ .open = default_ap_dev_open,
+};
+
+struct default_ap_module HAL_MODULE_INFO_SYM = {
+ .module = {
+ .common = {
+ .tag = HARDWARE_MODULE_TAG,
+ .version_major = 1,
+ .version_minor = 0,
+ .id = AUDIO_POLICY_HARDWARE_MODULE_ID,
+ .name = "Default audio policy HAL",
+ .author = "The Android Open Source Project",
+ .methods = &default_ap_module_methods,
+ },
+ },
+};
diff --git a/modules/audio_remote_submix/Android.bp b/modules/audio_remote_submix/Android.bp
index 578949e..c7d018c 100644
--- a/modules/audio_remote_submix/Android.bp
+++ b/modules/audio_remote_submix/Android.bp
@@ -17,15 +17,16 @@
relative_install_path: "hw",
vendor: true,
srcs: ["audio_hw.cpp"],
-
+ include_dirs: [
+ "system/media/audio_utils/include",
+ ],
shared_libs: [
"liblog",
"libcutils",
"libutils",
- "libnbaio_mono@2.0",
- "libaudioutils",
+ "libnbaio_mono",
],
- static_libs: ["libmedia_helper@2.0"],
+ static_libs: ["libmedia_helper"],
cflags: ["-Wno-unused-parameter"],
diff --git a/modules/usbaudio/Android.bp b/modules/usbaudio/Android.bp
index ecc74fb..c7d403f 100644
--- a/modules/usbaudio/Android.bp
+++ b/modules/usbaudio/Android.bp
@@ -25,7 +25,5 @@
"libalsautils",
],
cflags: ["-Wno-unused-parameter"],
- header_libs: [
- "android.hardware.audio.common.legacy@2.0",
- ],
+ header_libs: ["libhardware_headers"],
}