| /* |
| * Copyright (c) 2013-2016, 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 <stdlib.h> |
| #include <cutils/list.h> |
| #include <hardware/audio.h> |
| #include <tinyalsa/asoundlib.h> |
| #include <tinycompress/tinycompress.h> |
| |
| #include <audio_route/audio_route.h> |
| #include "audio_defs.h" |
| #include "voice.h" |
| #include "audio_hw_extn_api.h" |
| |
| #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" |
| |
| /* 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 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, |
| |
| 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[]; |
| }; |
| |
| struct stream_app_type_cfg { |
| int sample_rate; |
| uint32_t bit_width; |
| int app_type; |
| }; |
| |
| 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; |
| |
| stream_callback_t offload_callback; |
| void *offload_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; |
| }; |
| |
| 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; |
| int sample_rate; |
| 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 |
| } 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; |
| 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; |
| unsigned int pcm_record_uc_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; |
| }; |
| |
| 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); |
| |
| 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 |