/* * Copyright (C) 2013-2016 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/str_parms.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" // dlopen() does not go through default library path search if there is a "/" in the library name. #ifdef __LP64__ #define VISUALIZER_LIBRARY_PATH "/system/lib64/soundfx/libqcomvisualizer.so" #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib64/soundfx/libqcompostprocbundle.so" #else #define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so" #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so" #endif #define ADM_LIBRARY_PATH "libadm.so" /* Flags used to initialize acdb_settings variable that goes to ACDB library */ #define DMIC_FLAG 0x00000002 #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 ACDB_DEV_TYPE_OUT 1 #define ACDB_DEV_TYPE_IN 2 #define MAX_SUPPORTED_CHANNEL_MASKS 2 #define DEFAULT_HDMI_OUT_CHANNELS 2 #define ERROR_LOG_ENTRIES 16 typedef enum card_status_t { CARD_STATUS_OFFLINE, CARD_STATUS_ONLINE } card_status_t; /* These are the supported use cases by the hardware. * Each usecase is mapped to a specific PCM device. * Refer to pcm_device_table[]. */ 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, USECASE_AUDIO_PLAYBACK_TTS, USECASE_AUDIO_PLAYBACK_ULL, /* HFP Use case*/ USECASE_AUDIO_HFP_SCO, USECASE_AUDIO_HFP_SCO_WB, /* Capture usecases */ USECASE_AUDIO_RECORD, USECASE_AUDIO_RECORD_LOW_LATENCY, /* Voice extension usecases * * Following usecase are specific to voice session names created by * MODEM and APPS on 8992/8994/8084/8974 platforms. */ USECASE_VOICE_CALL, /* Usecase setup for voice session on first subscription for DSDS/DSDA */ USECASE_VOICE2_CALL, /* Usecase setup for voice session on second subscription for DSDS/DSDA */ USECASE_VOLTE_CALL, /* Usecase setup for VoLTE session on first subscription */ USECASE_QCHAT_CALL, /* Usecase setup for QCHAT session */ USECASE_VOWLAN_CALL, /* Usecase setup for VoWLAN session */ /* * Following usecase are specific to voice session names created by * MODEM and APPS on 8996 platforms. */ USECASE_VOICEMMODE1_CALL, /* Usecase setup for Voice/VoLTE/VoWLAN sessions on first * subscription for DSDS/DSDA */ USECASE_VOICEMMODE2_CALL, /* Usecase setup for voice/VoLTE/VoWLAN sessions on second * subscription for DSDS/DSDA */ USECASE_INCALL_REC_UPLINK, USECASE_INCALL_REC_DOWNLINK, USECASE_INCALL_REC_UPLINK_AND_DOWNLINK, USECASE_AUDIO_SPKR_CALIB_RX, USECASE_AUDIO_SPKR_CALIB_TX, USECASE_AUDIO_PLAYBACK_AFE_PROXY, USECASE_AUDIO_RECORD_AFE_PROXY, USECASE_AUDIO_DSM_FEEDBACK, AUDIO_USECASE_MAX }; const char * const use_case_table[AUDIO_USECASE_MAX]; #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 */ OFFLOAD_CMD_ERROR, /* offload playback hit some error */ }; enum { OFFLOAD_STATE_IDLE, OFFLOAD_STATE_PLAYING, OFFLOAD_STATE_PAUSED, }; struct offload_cmd { struct listnode node; int cmd; int data[]; }; enum { ERROR_CODE_STANDBY, ERROR_CODE_WRITE, }; struct error_log_entry { int32_t code; int32_t count; int64_t first_time; int64_t last_time; }; struct error_log { uint32_t errors; uint32_t idx; struct error_log_entry entries[ERROR_LOG_ENTRIES]; }; struct stream_out { struct audio_stream_out stream; pthread_mutex_t lock; /* see note below on mutex acquisition order */ pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */ 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; bool realtime; int af_period_multiplier; bool routing_change; struct audio_device *dev; card_status_t card_status; struct error_log error_log; }; struct stream_in { struct audio_stream_in stream; pthread_mutex_t lock; /* see note below on mutex acquisition order */ pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by capture thread */ struct pcm_config config; struct pcm *pcm; int standby; int source; int pcm_device_id; audio_devices_t device; audio_channel_mask_t channel_mask; audio_usecase_t usecase; bool enable_aec; bool enable_ns; int64_t frames_read; /* total frames read, not cleared when entering standby */ audio_io_handle_t capture_handle; audio_input_flags_t flags; bool is_st_session; bool is_st_session_active; bool realtime; int af_period_multiplier; bool routing_change; struct audio_device *dev; audio_format_t format; card_status_t card_status; }; typedef enum { PCM_PLAYBACK, PCM_CAPTURE, VOICE_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; }; typedef void* (*adm_init_t)(); typedef void (*adm_deinit_t)(void *); typedef void (*adm_register_output_stream_t)(void *, audio_io_handle_t, audio_output_flags_t); typedef void (*adm_register_input_stream_t)(void *, audio_io_handle_t, audio_input_flags_t); typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t); typedef void (*adm_request_focus_t)(void *, audio_io_handle_t); typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t); typedef void (*adm_set_config_t)(void *, audio_io_handle_t, struct pcm *, struct pcm_config *); typedef void (*adm_request_focus_v2_t)(void *, audio_io_handle_t, long); typedef bool (*adm_is_noirq_avail_t)(void *, int, int, int); typedef void (*adm_on_routing_change_t)(void *, audio_io_handle_t); 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; struct stream_in *active_input; struct stream_out *primary_output; struct stream_out *voice_tx_output; struct stream_out *current_call_output; bool bluetooth_nrec; bool screen_off; int *snd_dev_ref_cnt; struct listnode usecase_list; struct audio_route *audio_route; int acdb_settings; struct voice voice; unsigned int cur_hdmi_channels; bool bt_wb_speech_enabled; bool mic_muted; bool enable_voicerx; bool enable_hfp; int snd_card; void *platform; void *extspk; card_status_t card_status; void *visualizer_lib; int (*visualizer_start_output)(audio_io_handle_t, int); int (*visualizer_stop_output)(audio_io_handle_t, int); /* The pcm_params use_case_table is loaded by adev_verify_devices() upon * calling adev_open(). * * If an entry is not NULL, it can be used to determine if extended precision * or other capabilities are present for the device corresponding to that usecase. */ struct pcm_params *use_case_table[AUDIO_USECASE_MAX]; void *offload_effects_lib; int (*offload_effects_start_output)(audio_io_handle_t, int); int (*offload_effects_stop_output)(audio_io_handle_t, int); void *adm_data; void *adm_lib; adm_init_t adm_init; adm_deinit_t adm_deinit; adm_register_input_stream_t adm_register_input_stream; adm_register_output_stream_t adm_register_output_stream; adm_deregister_stream_t adm_deregister_stream; adm_request_focus_t adm_request_focus; adm_abandon_focus_t adm_abandon_focus; adm_set_config_t adm_set_config; adm_request_focus_v2_t adm_request_focus_v2; adm_is_noirq_avail_t adm_is_noirq_avail; adm_on_routing_change_t adm_on_routing_change; /* logging */ snd_device_t last_logged_snd_device[AUDIO_USECASE_MAX][2]; /* [out, in] */ }; int select_devices(struct audio_device *adev, audio_usecase_t uc_id); int disable_audio_route(struct audio_device *adev, struct audio_usecase *usecase); int disable_snd_device(struct audio_device *adev, snd_device_t snd_device); int enable_snd_device(struct audio_device *adev, snd_device_t snd_device); int enable_audio_route(struct audio_device *adev, struct audio_usecase *usecase); 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