/* * Copyright (c) 2013-2017, 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. * * This file was modified by DTS, Inc. The portions of the * code modified by DTS, Inc are copyrighted and * licensed separately, as follows: * * (C) 2014 DTS, Inc. * * 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 #include #include #include #include #include #include "audio_defs.h" #include "voice.h" #include "audio_hw_extn_api.h" #if LINUX_ENABLED #define VISUALIZER_LIBRARY_PATH "/usr/lib/libqcomvisualizer.so" #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/usr/lib/libqcompostprocbundle.so" #define ADM_LIBRARY_PATH "/usr/lib/libadm.so" #else #define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so" #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so" #define ADM_LIBRARY_PATH "/system/vendor/lib/libadm.so" #endif /* 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 14 #define MAX_SUPPORTED_FORMATS 15 #define MAX_SUPPORTED_SAMPLE_RATES 7 #define DEFAULT_HDMI_OUT_CHANNELS 2 #define DEFAULT_HDMI_OUT_SAMPLE_RATE 48000 #define DEFAULT_HDMI_OUT_FORMAT AUDIO_FORMAT_PCM_16_BIT #define SND_CARD_STATE_OFFLINE 0 #define SND_CARD_STATE_ONLINE 1 #define STREAM_DIRECTION_IN 0 #define STREAM_DIRECTION_OUT 1 #define MAX_PERF_LOCK_OPTS 20 #define MAX_STREAM_PROFILE_STR_LEN 32 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_OFFLOAD2, USECASE_AUDIO_PLAYBACK_OFFLOAD3, USECASE_AUDIO_PLAYBACK_OFFLOAD4, USECASE_AUDIO_PLAYBACK_OFFLOAD5, USECASE_AUDIO_PLAYBACK_OFFLOAD6, USECASE_AUDIO_PLAYBACK_OFFLOAD7, USECASE_AUDIO_PLAYBACK_OFFLOAD8, USECASE_AUDIO_PLAYBACK_OFFLOAD9, USECASE_AUDIO_PLAYBACK_ULL, /* 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_COMPRESS2, USECASE_AUDIO_RECORD_COMPRESS3, USECASE_AUDIO_RECORD_COMPRESS4, 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_VOICEMMODE1_CALL, USECASE_VOICEMMODE2_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, USECASE_AUDIO_PLAYBACK_AFE_PROXY, USECASE_AUDIO_RECORD_AFE_PROXY, USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE, USECASE_AUDIO_TRANSCODE_LOOPBACK, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM1, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM2, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM3, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM4, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM5, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM6, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM7, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM8, 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[]; }; typedef enum render_mode { RENDER_MODE_AUDIO_NO_TIMESTAMP = 0, RENDER_MODE_AUDIO_MASTER, RENDER_MODE_AUDIO_STC_MASTER, } render_mode_t; struct stream_app_type_cfg { int sample_rate; uint32_t bit_width; int app_type; }; struct stream_config { unsigned int sample_rate; audio_channel_mask_t channel_mask; audio_format_t format; audio_devices_t devices; unsigned int bit_width; }; struct stream_inout { 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 stream_config in_config; struct stream_config out_config; struct stream_app_type_cfg out_app_type_cfg; char profile[MAX_STREAM_PROFILE_STR_LEN]; struct audio_device *dev; void *adsp_hdlr_stream_handle; void *ip_hdlr_handle; stream_callback_t client_callback; void *client_cookie; }; 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; char profile[MAX_STREAM_PROFILE_STR_LEN]; 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]; audio_format_t supported_formats[MAX_SUPPORTED_FORMATS+1]; uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES+1]; bool muted; uint64_t written; /* total frames written, not cleared when entering standby */ audio_io_handle_t handle; struct stream_app_type_cfg app_type_cfg; 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; void *adsp_hdlr_stream_handle; void *ip_hdlr_handle; stream_callback_t client_callback; void *client_cookie; struct compr_gapless_mdata gapless_mdata; int send_new_metadata; bool send_next_track_params; bool is_compr_metadata_avail; unsigned int bit_width; uint32_t hal_fragment_size; audio_format_t hal_ip_format; audio_format_t hal_op_format; void *convert_buffer; bool realtime; int af_period_multiplier; struct audio_device *dev; void* qaf_stream_handle; pthread_cond_t qaf_offload_cond; pthread_t qaf_offload_thread; struct listnode qaf_offload_cmd_list; uint32_t platform_latency; render_mode_t render_mode; bool drift_correction_enabled; struct audio_out_channel_map_param channel_map_param; /* input channel map */ audio_offload_info_t info; int started; qahwi_stream_out_t qahwi_out; bool is_iec61937_info_available; bool stream_config_changed; mix_matrix_params_t pan_scale_params; mix_matrix_params_t downmix_params; }; 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 playback 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; audio_format_t format; audio_io_handle_t capture_handle; audio_input_flags_t flags; char profile[MAX_STREAM_PROFILE_STR_LEN]; bool is_st_session; bool is_st_session_active; unsigned int sample_rate; unsigned int bit_width; bool realtime; int af_period_multiplier; struct stream_app_type_cfg app_type_cfg; void *cin_extn; qahwi_stream_in_t qahwi_in; struct audio_device *dev; }; typedef enum { PCM_PLAYBACK, PCM_CAPTURE, VOICE_CALL, VOIP_CALL, PCM_HFP_CALL, TRANSCODE_LOOPBACK } usecase_type_t; union stream_ptr { struct stream_in *in; struct stream_out *out; struct stream_inout *inout; }; 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; struct stream_app_type_cfg out_app_type_cfg; struct stream_app_type_cfg in_app_type_cfg; union stream_ptr stream; }; struct sound_card_status { pthread_mutex_t lock; int state; }; struct stream_format { struct listnode list; audio_format_t format; }; struct stream_sample_rate { struct listnode list; uint32_t sample_rate; }; typedef union { audio_output_flags_t out_flags; audio_input_flags_t in_flags; } audio_io_flags_t; struct streams_io_cfg { struct listnode list; audio_io_flags_t flags; char profile[MAX_STREAM_PROFILE_STR_LEN]; struct listnode format_list; struct listnode sample_rate_list; struct stream_app_type_cfg app_type_cfg; }; 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; audio_devices_t out_device; 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 listnode streams_output_cfg_list; struct listnode streams_input_cfg_list; struct audio_route *audio_route; int acdb_settings; bool speaker_lr_swap; struct voice voice; unsigned int cur_hdmi_channels; audio_format_t cur_hdmi_format; unsigned int cur_hdmi_sample_rate; unsigned int cur_hdmi_bit_width; unsigned int cur_wfd_channels; bool bt_wb_speech_enabled; bool allow_afe_proxy_usage; int snd_card; unsigned int cur_codec_backend_samplerate; unsigned int cur_codec_backend_bit_width; bool is_channel_status_set; void *platform; unsigned int offload_usecases_state; 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, struct mixer *); int (*offload_effects_stop_output)(audio_io_handle_t, int); struct sound_card_status snd_card_status; int (*offload_effects_set_hpx_state)(bool); 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; void (*offload_effects_get_parameters)(struct str_parms *, struct str_parms *); void (*offload_effects_set_parameters)(struct str_parms *); bool multi_offload_enable; int perf_lock_handle; int perf_lock_opts[MAX_PERF_LOCK_OPTS]; int perf_lock_opts_size; bool native_playback_enabled; bool asrc_mode_enabled; qahwi_device_t qahwi_dev; bool vr_audio_mode_enabled; bool bt_sco_on; struct audio_device_config_param *device_cfg_params; unsigned int interactive_usecase_state; }; 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(const struct audio_device *adev, audio_usecase_t uc_id); struct stream_in *get_next_active_input(const struct audio_device *adev); bool is_offload_usecase(audio_usecase_t uc_id); bool audio_is_true_native_stream_active(struct audio_device *adev); bool audio_is_dsd_native_stream_active(struct audio_device *adev); int pcm_ioctl(struct pcm *pcm, int request, ...); int get_snd_card_state(struct audio_device *adev); audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev, usecase_type_t type); int adev_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 __unused); void adev_close_output_stream(struct audio_hw_device *dev __unused, struct audio_stream_out *stream); #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