hal/audio_hw.h - LeafOS-Project/android_hardware_qcom_audio - Gitiles

 /*
  * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
  * Not a contribution.
  *
  * Copyright (C) 2013 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 QCOM_AUDIO_HW_H
 #define QCOM_AUDIO_HW_H

 #include <cutils/list.h>
 #include <hardware/audio.h>
 #include <tinyalsa/asoundlib.h>
 #include <tinycompress/tinycompress.h>

 #include <audio_route/audio_route.h>
 #include "voice.h"

 #define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
 #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"

 #define BT_SCO_SAMPLE_RATE "bt-sco-samplerate"
 #define BT_SCO_WB_SAMPLE_RATE "bt-sco-wb-samplerate"

 /* Flags used to initialize acdb_settings variable that goes to ACDB library */
 #define NONE_FLAG            0x00000000
 #define ANC_FLAG	     0x00000001
 #define DMIC_FLAG            0x00000002
 #define QMIC_FLAG            0x00000004
 #define TTY_MODE_OFF         0x00000010
 #define TTY_MODE_FULL        0x00000020
 #define TTY_MODE_VCO         0x00000040
 #define TTY_MODE_HCO         0x00000080
 #define TTY_MODE_CLEAR       0xFFFFFF0F
 #define FLUENCE_MODE_CLEAR   0xFFFFFFF0

 #define ACDB_DEV_TYPE_OUT 1
 #define ACDB_DEV_TYPE_IN 2

 #define MAX_SUPPORTED_CHANNEL_MASKS 2
 #define DEFAULT_HDMI_OUT_CHANNELS   2

 typedef int snd_device_t;

 /* These are the supported use cases by the hardware.
  * Each usecase is mapped to a specific PCM device.
  * Refer to pcm_device_table[].
  */
 typedef enum {
     USECASE_INVALID = -1,
     /* Playback usecases */
     USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
     USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
     USECASE_AUDIO_PLAYBACK_MULTI_CH,
     USECASE_AUDIO_PLAYBACK_OFFLOAD,

     /* FM usecase */
     USECASE_AUDIO_PLAYBACK_FM,

     /* HFP Use case*/
     USECASE_AUDIO_HFP_SCO,
     USECASE_AUDIO_HFP_SCO_WB,

     /* Capture usecases */
     USECASE_AUDIO_RECORD,
     USECASE_AUDIO_RECORD_COMPRESS,
     USECASE_AUDIO_RECORD_LOW_LATENCY,
     USECASE_AUDIO_RECORD_FM_VIRTUAL,

     /* Voice usecase */
     USECASE_VOICE_CALL,

     /* Voice extension usecases */
     USECASE_VOICE2_CALL,
     USECASE_VOLTE_CALL,
     USECASE_QCHAT_CALL,
     USECASE_VOWLAN_CALL,
     USECASE_COMPRESS_VOIP_CALL,

     USECASE_INCALL_REC_UPLINK,
     USECASE_INCALL_REC_DOWNLINK,
     USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
     USECASE_INCALL_REC_UPLINK_COMPRESS,
     USECASE_INCALL_REC_DOWNLINK_COMPRESS,
     USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS,

     USECASE_INCALL_MUSIC_UPLINK,
     USECASE_INCALL_MUSIC_UPLINK2,

     USECASE_AUDIO_SPKR_CALIB_RX,
     USECASE_AUDIO_SPKR_CALIB_TX,
     AUDIO_USECASE_MAX
 } audio_usecase_t;

 #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))

 /*
  * tinyAlsa library interprets period size as number of frames
  * one frame = channel_count * sizeof (pcm sample)
  * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
  * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
  * We should take care of returning proper size when AudioFlinger queries for
  * the buffer size of an input/output stream
  */

 enum {
     OFFLOAD_CMD_EXIT,               /* exit compress offload thread loop*/
     OFFLOAD_CMD_DRAIN,              /* send a full drain request to DSP */
     OFFLOAD_CMD_PARTIAL_DRAIN,      /* send a partial drain request to DSP */
     OFFLOAD_CMD_WAIT_FOR_BUFFER,    /* wait for buffer released by DSP */
 };

 enum {
     OFFLOAD_STATE_IDLE,
     OFFLOAD_STATE_PLAYING,
     OFFLOAD_STATE_PAUSED,
 };

 struct offload_cmd {
     struct listnode node;
     int cmd;
     int data[];
 };

 struct stream_out {
     struct audio_stream_out stream;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
     pthread_cond_t  cond;
     struct pcm_config config;
     struct compr_config compr_config;
     struct pcm *pcm;
     struct compress *compr;
     int standby;
     int pcm_device_id;
     unsigned int sample_rate;
     audio_channel_mask_t channel_mask;
     audio_format_t format;
     audio_devices_t devices;
     audio_output_flags_t flags;
     audio_usecase_t usecase;
     /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
     audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
     bool muted;
     uint64_t written; /* total frames written, not cleared when entering standby */
     audio_io_handle_t handle;

     int non_blocking;
     int playback_started;
     int offload_state;
     pthread_cond_t offload_cond;
     pthread_t offload_thread;
     struct listnode offload_cmd_list;
     bool offload_thread_blocked;

     stream_callback_t offload_callback;
     void *offload_cookie;
     struct compr_gapless_mdata gapless_mdata;
     int send_new_metadata;

     struct audio_device *dev;
 };

 struct stream_in {
     struct audio_stream_in stream;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
     struct pcm_config config;
     struct pcm *pcm;
     int standby;
     int source;
     int pcm_device_id;
     int device;
     audio_channel_mask_t channel_mask;
     audio_usecase_t usecase;
     bool enable_aec;
     bool enable_ns;
     audio_format_t format;

     struct audio_device *dev;
 };

 typedef enum {
     PCM_PLAYBACK,
     PCM_CAPTURE,
     VOICE_CALL,
     VOIP_CALL,
     PCM_HFP_CALL
 } usecase_type_t;

 union stream_ptr {
     struct stream_in *in;
     struct stream_out *out;
 };

 struct audio_usecase {
     struct listnode list;
     audio_usecase_t id;
     usecase_type_t  type;
     audio_devices_t devices;
     snd_device_t out_snd_device;
     snd_device_t in_snd_device;
     union stream_ptr stream;
 };

 struct audio_device {
     struct audio_hw_device device;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
     struct mixer *mixer;
     audio_mode_t mode;
     audio_devices_t out_device;
     struct stream_in *active_input;
     struct stream_out *primary_output;
     bool bluetooth_nrec;
     bool screen_off;
     int *snd_dev_ref_cnt;
     struct listnode usecase_list;
     struct audio_route *audio_route;
     int acdb_settings;
     bool speaker_lr_swap;
     struct voice voice;
     unsigned int cur_hdmi_channels;
     unsigned int cur_wfd_channels;

     int snd_card;
     void *platform;

     void *visualizer_lib;
     int (*visualizer_start_output)(audio_io_handle_t, int);
     int (*visualizer_stop_output)(audio_io_handle_t, int);
     void *offload_effects_lib;
     int (*offload_effects_start_output)(audio_io_handle_t, int);
     int (*offload_effects_stop_output)(audio_io_handle_t, int);
 };

 int select_devices(struct audio_device *adev,
                           audio_usecase_t uc_id);
 int disable_audio_route(struct audio_device *adev,
                                struct audio_usecase *usecase,
                                bool update_mixer);
 int disable_snd_device(struct audio_device *adev,
                               snd_device_t snd_device,
                               bool update_mixer);
 int enable_snd_device(struct audio_device *adev,
                              snd_device_t snd_device,
                              bool update_mixer);
 int enable_audio_route(struct audio_device *adev,
                               struct audio_usecase *usecase,
                               bool update_mixer);
 struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
                                                    audio_usecase_t uc_id);

 #define LITERAL_TO_STRING(x) #x
 #define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
             __FILE__ ":" LITERAL_TO_STRING(__LINE__)\
             " ASSERT_FATAL(" #condition ") failed.")

 /*
  * NOTE: when multiple mutexes have to be acquired, always take the
  * stream_in or stream_out mutex first, followed by the audio_device mutex.
  */

 #endif // QCOM_AUDIO_HW_H
	/*
	* Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
	* Not a contribution.
	*
	* Copyright (C) 2013 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 QCOM_AUDIO_HW_H
	#define QCOM_AUDIO_HW_H

	#include <cutils/list.h>
	#include <hardware/audio.h>
	#include <tinyalsa/asoundlib.h>
	#include <tinycompress/tinycompress.h>

	#include <audio_route/audio_route.h>
	#include "voice.h"

	#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
	#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"

	#define BT_SCO_SAMPLE_RATE "bt-sco-samplerate"
	#define BT_SCO_WB_SAMPLE_RATE "bt-sco-wb-samplerate"

	/* Flags used to initialize acdb_settings variable that goes to ACDB library */
	#define NONE_FLAG 0x00000000
	#define ANC_FLAG 0x00000001
	#define DMIC_FLAG 0x00000002
	#define QMIC_FLAG 0x00000004
	#define TTY_MODE_OFF 0x00000010
	#define TTY_MODE_FULL 0x00000020
	#define TTY_MODE_VCO 0x00000040
	#define TTY_MODE_HCO 0x00000080
	#define TTY_MODE_CLEAR 0xFFFFFF0F
	#define FLUENCE_MODE_CLEAR 0xFFFFFFF0

	#define ACDB_DEV_TYPE_OUT 1
	#define ACDB_DEV_TYPE_IN 2

	#define MAX_SUPPORTED_CHANNEL_MASKS 2
	#define DEFAULT_HDMI_OUT_CHANNELS 2

	typedef int snd_device_t;

	/* These are the supported use cases by the hardware.
	* Each usecase is mapped to a specific PCM device.
	* Refer to pcm_device_table[].
	*/
	typedef enum {
	USECASE_INVALID = -1,
	/* Playback usecases */
	USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
	USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
	USECASE_AUDIO_PLAYBACK_MULTI_CH,
	USECASE_AUDIO_PLAYBACK_OFFLOAD,

	/* FM usecase */
	USECASE_AUDIO_PLAYBACK_FM,

	/* HFP Use case*/
	USECASE_AUDIO_HFP_SCO,
	USECASE_AUDIO_HFP_SCO_WB,

	/* Capture usecases */
	USECASE_AUDIO_RECORD,
	USECASE_AUDIO_RECORD_COMPRESS,
	USECASE_AUDIO_RECORD_LOW_LATENCY,
	USECASE_AUDIO_RECORD_FM_VIRTUAL,

	/* Voice usecase */
	USECASE_VOICE_CALL,

	/* Voice extension usecases */
	USECASE_VOICE2_CALL,
	USECASE_VOLTE_CALL,
	USECASE_QCHAT_CALL,
	USECASE_VOWLAN_CALL,
	USECASE_COMPRESS_VOIP_CALL,

	USECASE_INCALL_REC_UPLINK,
	USECASE_INCALL_REC_DOWNLINK,
	USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
	USECASE_INCALL_REC_UPLINK_COMPRESS,
	USECASE_INCALL_REC_DOWNLINK_COMPRESS,
	USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS,

	USECASE_INCALL_MUSIC_UPLINK,
	USECASE_INCALL_MUSIC_UPLINK2,

	USECASE_AUDIO_SPKR_CALIB_RX,
	USECASE_AUDIO_SPKR_CALIB_TX,
	AUDIO_USECASE_MAX
	} audio_usecase_t;

	#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))

	/*
	* tinyAlsa library interprets period size as number of frames
	* one frame = channel_count * sizeof (pcm sample)
	* so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
	* DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
	* We should take care of returning proper size when AudioFlinger queries for
	* the buffer size of an input/output stream
	*/

	enum {
	OFFLOAD_CMD_EXIT, /* exit compress offload thread loop*/
	OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */
	OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */
	OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */
	};

	enum {
	OFFLOAD_STATE_IDLE,
	OFFLOAD_STATE_PLAYING,
	OFFLOAD_STATE_PAUSED,
	};

	struct offload_cmd {
	struct listnode node;
	int cmd;
	int data[];
	};

	struct stream_out {
	struct audio_stream_out stream;
	pthread_mutex_t lock; /* see note below on mutex acquisition order */
	pthread_cond_t cond;
	struct pcm_config config;
	struct compr_config compr_config;
	struct pcm *pcm;
	struct compress *compr;
	int standby;
	int pcm_device_id;
	unsigned int sample_rate;
	audio_channel_mask_t channel_mask;
	audio_format_t format;
	audio_devices_t devices;
	audio_output_flags_t flags;
	audio_usecase_t usecase;
	/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
	audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
	bool muted;
	uint64_t written; /* total frames written, not cleared when entering standby */
	audio_io_handle_t handle;

	int non_blocking;
	int playback_started;
	int offload_state;
	pthread_cond_t offload_cond;
	pthread_t offload_thread;
	struct listnode offload_cmd_list;
	bool offload_thread_blocked;

	stream_callback_t offload_callback;
	void *offload_cookie;
	struct compr_gapless_mdata gapless_mdata;
	int send_new_metadata;

	struct audio_device *dev;
	};

	struct stream_in {
	struct audio_stream_in stream;
	pthread_mutex_t lock; /* see note below on mutex acquisition order */
	struct pcm_config config;
	struct pcm *pcm;
	int standby;
	int source;
	int pcm_device_id;
	int device;
	audio_channel_mask_t channel_mask;
	audio_usecase_t usecase;
	bool enable_aec;
	bool enable_ns;
	audio_format_t format;

	struct audio_device *dev;
	};

	typedef enum {
	PCM_PLAYBACK,
	PCM_CAPTURE,
	VOICE_CALL,
	VOIP_CALL,
	PCM_HFP_CALL
	} usecase_type_t;

	union stream_ptr {
	struct stream_in *in;
	struct stream_out *out;
	};

	struct audio_usecase {
	struct listnode list;
	audio_usecase_t id;
	usecase_type_t type;
	audio_devices_t devices;
	snd_device_t out_snd_device;
	snd_device_t in_snd_device;
	union stream_ptr stream;
	};

	struct audio_device {
	struct audio_hw_device device;
	pthread_mutex_t lock; /* see note below on mutex acquisition order */
	struct mixer *mixer;
	audio_mode_t mode;
	audio_devices_t out_device;
	struct stream_in *active_input;
	struct stream_out *primary_output;
	bool bluetooth_nrec;
	bool screen_off;
	int *snd_dev_ref_cnt;
	struct listnode usecase_list;
	struct audio_route *audio_route;
	int acdb_settings;
	bool speaker_lr_swap;
	struct voice voice;
	unsigned int cur_hdmi_channels;
	unsigned int cur_wfd_channels;

	int snd_card;
	void *platform;

	void *visualizer_lib;
	int (*visualizer_start_output)(audio_io_handle_t, int);
	int (*visualizer_stop_output)(audio_io_handle_t, int);
	void *offload_effects_lib;
	int (*offload_effects_start_output)(audio_io_handle_t, int);
	int (*offload_effects_stop_output)(audio_io_handle_t, int);
	};

	int select_devices(struct audio_device *adev,
	audio_usecase_t uc_id);
	int disable_audio_route(struct audio_device *adev,
	struct audio_usecase *usecase,
	bool update_mixer);
	int disable_snd_device(struct audio_device *adev,
	snd_device_t snd_device,
	bool update_mixer);
	int enable_snd_device(struct audio_device *adev,
	snd_device_t snd_device,
	bool update_mixer);
	int enable_audio_route(struct audio_device *adev,
	struct audio_usecase *usecase,
	bool update_mixer);
	struct audio_usecase get_usecase_from_list(struct audio_device adev,
	audio_usecase_t uc_id);

	#define LITERAL_TO_STRING(x) #x
	#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
	__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
	" ASSERT_FATAL(" #condition ") failed.")

	/*
	* NOTE: when multiple mutexes have to be acquired, always take the
	* stream_in or stream_out mutex first, followed by the audio_device mutex.
	*/

	#endif // QCOM_AUDIO_HW_H