1 /* 2 * Copyright (C) 2013 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #ifndef NVIDIA_AUDIO_HW_H 18 #define NVIDIA_AUDIO_HW_H 19 20 #include <cutils/list.h> 21 #include <hardware/audio.h> 22 23 #include <tinyalsa/asoundlib.h> 24 #include <tinycompress/tinycompress.h> 25 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */ 26 #include <audio_utils/resampler.h> 27 #include <audio_route/audio_route.h> 28 29 /* Retry for delay in FW loading*/ 30 #define RETRY_NUMBER 10 31 #define RETRY_US 500000 32 33 #ifdef __LP64__ 34 #define OFFLOAD_FX_LIBRARY_PATH "/system/lib64/soundfx/libnvvisualizer.so" 35 #else 36 #define OFFLOAD_FX_LIBRARY_PATH "/system/lib/soundfx/libnvvisualizer.so" 37 #endif 38 39 #define HTC_ACOUSTIC_LIBRARY_PATH "/vendor/lib/libhtcacoustic.so" 40 41 #ifdef PREPROCESSING_ENABLED 42 #include <audio_utils/echo_reference.h> 43 #define MAX_PREPROCESSORS 3 44 struct effect_info_s { 45 effect_handle_t effect_itfe; 46 size_t num_channel_configs; 47 channel_config_t *channel_configs; 48 }; 49 #endif 50 51 #ifdef __LP64__ 52 #define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib64/hw/sound_trigger.primary.flounder.so" 53 #else 54 #define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib/hw/sound_trigger.primary.flounder.so" 55 #endif 56 57 #define TTY_MODE_OFF 1 58 #define TTY_MODE_FULL 2 59 #define TTY_MODE_VCO 4 60 #define TTY_MODE_HCO 8 61 62 #define DUALMIC_CONFIG_NONE 0 63 #define DUALMIC_CONFIG_1 1 64 65 /* Sound devices specific to the platform 66 * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound 67 * devices to enable corresponding mixer paths 68 */ 69 enum { 70 SND_DEVICE_NONE = 0, 71 72 /* Playback devices */ 73 SND_DEVICE_MIN, 74 SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN, 75 SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN, 76 SND_DEVICE_OUT_SPEAKER, 77 SND_DEVICE_OUT_HEADPHONES, 78 SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES, 79 SND_DEVICE_OUT_VOICE_HANDSET, 80 SND_DEVICE_OUT_VOICE_SPEAKER, 81 SND_DEVICE_OUT_VOICE_HEADPHONES, 82 SND_DEVICE_OUT_HDMI, 83 SND_DEVICE_OUT_SPEAKER_AND_HDMI, 84 SND_DEVICE_OUT_BT_SCO, 85 SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES, 86 SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES, 87 SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET, 88 SND_DEVICE_OUT_END, 89 90 /* 91 * Note: IN_BEGIN should be same as OUT_END because total number of devices 92 * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices. 93 */ 94 /* Capture devices */ 95 SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END, 96 SND_DEVICE_IN_HANDSET_MIC = SND_DEVICE_IN_BEGIN, 97 SND_DEVICE_IN_SPEAKER_MIC, 98 SND_DEVICE_IN_HEADSET_MIC, 99 SND_DEVICE_IN_HANDSET_MIC_AEC, 100 SND_DEVICE_IN_SPEAKER_MIC_AEC, 101 SND_DEVICE_IN_HEADSET_MIC_AEC, 102 SND_DEVICE_IN_VOICE_SPEAKER_MIC, 103 SND_DEVICE_IN_VOICE_HEADSET_MIC, 104 SND_DEVICE_IN_HDMI_MIC, 105 SND_DEVICE_IN_BT_SCO_MIC, 106 SND_DEVICE_IN_CAMCORDER_MIC, 107 SND_DEVICE_IN_VOICE_DMIC_1, 108 SND_DEVICE_IN_VOICE_SPEAKER_DMIC_1, 109 SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC, 110 SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC, 111 SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC, 112 SND_DEVICE_IN_VOICE_REC_HEADSET_MIC, 113 SND_DEVICE_IN_VOICE_REC_MIC, 114 SND_DEVICE_IN_VOICE_REC_DMIC_1, 115 SND_DEVICE_IN_VOICE_REC_DMIC_NS_1, 116 SND_DEVICE_IN_LOOPBACK_AEC, 117 SND_DEVICE_IN_END, 118 119 SND_DEVICE_MAX = SND_DEVICE_IN_END, 120 121 }; 122 123 124 #define MIXER_CARD 0 125 #define SOUND_CARD 0 126 127 /* 128 * tinyAlsa library interprets period size as number of frames 129 * one frame = channel_count * sizeof (pcm sample) 130 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes 131 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes 132 * We should take care of returning proper size when AudioFlinger queries for 133 * the buffer size of an input/output stream 134 */ 135 #define PLAYBACK_PERIOD_SIZE 256 136 #define PLAYBACK_PERIOD_COUNT 2 137 #define PLAYBACK_DEFAULT_CHANNEL_COUNT 2 138 #define PLAYBACK_DEFAULT_SAMPLING_RATE 48000 139 #define PLAYBACK_START_THRESHOLD(size, count) (((size) * (count)) - 1) 140 #define PLAYBACK_STOP_THRESHOLD(size, count) ((size) * ((count) + 2)) 141 #define PLAYBACK_AVAILABLE_MIN 1 142 143 144 #define SCO_PERIOD_SIZE 168 145 #define SCO_PERIOD_COUNT 2 146 #define SCO_DEFAULT_CHANNEL_COUNT 2 147 #define SCO_DEFAULT_SAMPLING_RATE 8000 148 #define SCO_START_THRESHOLD 335 149 #define SCO_STOP_THRESHOLD 336 150 #define SCO_AVAILABLE_MIN 1 151 152 #define PLAYBACK_HDMI_MULTI_PERIOD_SIZE 1024 153 #define PLAYBACK_HDMI_MULTI_PERIOD_COUNT 4 154 #define PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6 155 #define PLAYBACK_HDMI_MULTI_PERIOD_BYTES \ 156 (PLAYBACK_HDMI_MULTI_PERIOD_SIZE * PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2) 157 #define PLAYBACK_HDMI_MULTI_START_THRESHOLD 4095 158 #define PLAYBACK_HDMI_MULTI_STOP_THRESHOLD 4096 159 #define PLAYBACK_HDMI_MULTI_AVAILABLE_MIN 1 160 161 #define PLAYBACK_HDMI_DEFAULT_CHANNEL_COUNT 2 162 163 #define CAPTURE_PERIOD_SIZE 1024 164 #define CAPTURE_PERIOD_SIZE_LOW_LATENCY 256 165 #define CAPTURE_PERIOD_COUNT 2 166 #define CAPTURE_PERIOD_COUNT_LOW_LATENCY 2 167 #define CAPTURE_DEFAULT_CHANNEL_COUNT 2 168 #define CAPTURE_DEFAULT_SAMPLING_RATE 48000 169 #define CAPTURE_START_THRESHOLD 1 170 171 #define COMPRESS_CARD 0 172 #define COMPRESS_DEVICE 5 173 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024) 174 #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4 175 /* ToDo: Check and update a proper value in msec */ 176 #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96 177 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x10000 //NV suggested value 178 179 #define DEEP_BUFFER_OUTPUT_SAMPLING_RATE 48000 180 #define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 480 181 #define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8 182 183 #define MAX_SUPPORTED_CHANNEL_MASKS 2 184 185 typedef int snd_device_t; 186 187 /* These are the supported use cases by the hardware. 188 * Each usecase is mapped to a specific PCM device. 189 * Refer to pcm_device_table[]. 190 */ 191 typedef enum { 192 USECASE_INVALID = -1, 193 /* Playback usecases */ 194 USECASE_AUDIO_PLAYBACK = 0, 195 USECASE_AUDIO_PLAYBACK_MULTI_CH, 196 USECASE_AUDIO_PLAYBACK_OFFLOAD, 197 USECASE_AUDIO_PLAYBACK_DEEP_BUFFER, 198 199 /* Capture usecases */ 200 USECASE_AUDIO_CAPTURE, 201 USECASE_AUDIO_CAPTURE_HOTWORD, 202 203 USECASE_VOICE_CALL, 204 AUDIO_USECASE_MAX 205 } audio_usecase_t; 206 207 #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) 208 209 /* 210 * tinyAlsa library interprets period size as number of frames 211 * one frame = channel_count * sizeof (pcm sample) 212 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes 213 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes 214 * We should take care of returning proper size when AudioFlinger queries for 215 * the buffer size of an input/output stream 216 */ 217 218 enum { 219 OFFLOAD_CMD_EXIT, /* exit compress offload thread loop*/ 220 OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */ 221 OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */ 222 OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */ 223 }; 224 225 enum { 226 OFFLOAD_STATE_IDLE, 227 OFFLOAD_STATE_PLAYING, 228 OFFLOAD_STATE_PAUSED, 229 OFFLOAD_STATE_PAUSED_FLUSHED, 230 }; 231 232 typedef enum { 233 PCM_PLAYBACK = 0x1, 234 PCM_CAPTURE = 0x2, 235 VOICE_CALL = 0x4, 236 PCM_HOTWORD_STREAMING = 0x8, 237 PCM_CAPTURE_LOW_LATENCY = 0x10, 238 } usecase_type_t; 239 240 struct offload_cmd { 241 struct listnode node; 242 int cmd; 243 int data[]; 244 }; 245 246 struct pcm_device_profile { 247 struct pcm_config config; 248 int card; 249 int id; 250 usecase_type_t type; 251 audio_devices_t devices; 252 }; 253 254 struct pcm_device { 255 struct listnode stream_list_node; 256 struct pcm_device_profile* pcm_profile; 257 struct pcm* pcm; 258 int status; 259 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */ 260 struct resampler_itfe* resampler; 261 int16_t* res_buffer; 262 size_t res_byte_count; 263 int sound_trigger_handle; 264 }; 265 266 struct stream_out { 267 struct audio_stream_out stream; 268 pthread_mutex_t lock; /* see note below on mutex acquisition order */ 269 pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */ 270 pthread_cond_t cond; 271 struct pcm_config config; 272 struct listnode pcm_dev_list; 273 struct compr_config compr_config; 274 struct compress* compr; 275 int standby; 276 unsigned int sample_rate; 277 audio_channel_mask_t channel_mask; 278 audio_format_t format; 279 audio_devices_t devices; 280 audio_output_flags_t flags; 281 audio_usecase_t usecase; 282 /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */ 283 audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1]; 284 bool muted; 285 /* total frames written, not cleared when entering standby */ 286 uint64_t written; 287 audio_io_handle_t handle; 288 289 int non_blocking; 290 int offload_state; 291 pthread_cond_t offload_cond; 292 pthread_t offload_thread; 293 struct listnode offload_cmd_list; 294 bool offload_thread_blocked; 295 296 stream_callback_t offload_callback; 297 void* offload_cookie; 298 struct compr_gapless_mdata gapless_mdata; 299 int send_new_metadata; 300 301 struct audio_device* dev; 302 303 #ifdef PREPROCESSING_ENABLED 304 struct echo_reference_itfe *echo_reference; 305 // echo_reference_generation indicates if the echo reference used by the output stream is 306 // in sync with the one known by the audio_device. When different from the generation stored 307 // in the audio_device the output stream must release the echo reference. 308 // always modified with audio device and stream mutex locked. 309 int32_t echo_reference_generation; 310 #endif 311 312 bool is_fastmixer_affinity_set; 313 }; 314 315 struct stream_in { 316 struct audio_stream_in stream; 317 pthread_mutex_t lock; /* see note below on mutex acquisition order */ 318 pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by 319 capture thread */ 320 struct pcm_config config; 321 struct listnode pcm_dev_list; 322 int standby; 323 audio_source_t source; 324 audio_devices_t devices; 325 uint32_t main_channels; 326 audio_usecase_t usecase; 327 usecase_type_t usecase_type; 328 bool enable_aec; 329 audio_input_flags_t input_flags; 330 331 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */ 332 unsigned int requested_rate; 333 struct resampler_itfe* resampler; 334 struct resampler_buffer_provider buf_provider; 335 int read_status; 336 int16_t* read_buf; 337 size_t read_buf_size; 338 size_t read_buf_frames; 339 340 int16_t *proc_buf_in; 341 int16_t *proc_buf_out; 342 size_t proc_buf_size; 343 size_t proc_buf_frames; 344 345 #ifdef PREPROCESSING_ENABLED 346 struct echo_reference_itfe *echo_reference; 347 int16_t *ref_buf; 348 size_t ref_buf_size; 349 size_t ref_buf_frames; 350 351 #ifdef HW_AEC_LOOPBACK 352 bool hw_echo_reference; 353 int16_t* hw_ref_buf; 354 size_t hw_ref_buf_size; 355 #endif 356 357 int num_preprocessors; 358 struct effect_info_s preprocessors[MAX_PREPROCESSORS]; 359 360 bool aux_channels_changed; 361 uint32_t aux_channels; 362 #endif 363 364 struct audio_device* dev; 365 bool is_fastcapture_affinity_set; 366 }; 367 368 struct mixer_card { 369 struct listnode adev_list_node; 370 struct listnode uc_list_node[AUDIO_USECASE_MAX]; 371 int card; 372 struct mixer* mixer; 373 struct audio_route* audio_route; 374 }; 375 376 struct audio_usecase { 377 struct listnode adev_list_node; 378 audio_usecase_t id; 379 usecase_type_t type; 380 audio_devices_t devices; 381 snd_device_t out_snd_device; 382 snd_device_t in_snd_device; 383 struct audio_stream* stream; 384 struct listnode mixer_list; 385 }; 386 387 388 struct audio_device { 389 struct audio_hw_device device; 390 pthread_mutex_t lock; /* see note below on mutex acquisition order */ 391 struct listnode mixer_list; 392 audio_mode_t mode; 393 struct stream_in* active_input; 394 struct stream_out* primary_output; 395 int in_call; 396 float voice_volume; 397 bool mic_mute; 398 int tty_mode; 399 bool bluetooth_nrec; 400 bool screen_off; 401 int* snd_dev_ref_cnt; 402 struct listnode usecase_list; 403 bool speaker_lr_swap; 404 unsigned int cur_hdmi_channels; 405 int dualmic_config; 406 bool ns_in_voice_rec; 407 408 void* offload_fx_lib; 409 int (*offload_fx_start_output)(audio_io_handle_t); 410 int (*offload_fx_stop_output)(audio_io_handle_t); 411 412 #ifdef PREPROCESSING_ENABLED 413 struct echo_reference_itfe* echo_reference; 414 // echo_reference_generation indicates if the echo reference used by the output stream is 415 // in sync with the one known by the audio_device. 416 // incremented atomically with a memory barrier and audio device mutex locked but WITHOUT 417 // stream mutex locked: the stream will load it atomically with a barrier and re-read it 418 // with audio device mutex if needed 419 volatile int32_t echo_reference_generation; 420 #endif 421 422 void* htc_acoustic_lib; 423 int (*htc_acoustic_init_rt5506)(); 424 int (*htc_acoustic_set_rt5506_amp)(int, int); 425 int (*htc_acoustic_set_amp_mode)(int, int, int, int, bool); 426 int (*htc_acoustic_spk_reverse)(bool); 427 428 void* sound_trigger_lib; 429 int (*sound_trigger_open_for_streaming)(); 430 size_t (*sound_trigger_read_samples)(int, void*, size_t); 431 int (*sound_trigger_close_for_streaming)(int); 432 433 int tfa9895_init; 434 int tfa9895_mode_change; 435 pthread_mutex_t tfa9895_lock; 436 437 int dummybuf_thread_timeout; 438 int dummybuf_thread_cancel; 439 int dummybuf_thread_active; 440 audio_devices_t dummybuf_thread_devices; 441 pthread_mutex_t dummybuf_thread_lock; 442 pthread_t dummybuf_thread; 443 444 pthread_mutex_t lock_inputs; /* see note below on mutex acquisition order */ 445 }; 446 447 /* 448 * NOTE: when multiple mutexes have to be acquired, always take the 449 * lock_inputs, stream_in, stream_out, audio_device, then tfa9895 mutex. 450 * stream_in mutex must always be before stream_out mutex 451 * if both have to be taken (see get_echo_reference(), put_echo_reference()...) 452 * dummybuf_thread mutex is not related to the other mutexes with respect to order. 453 * lock_inputs must be held in order to either close the input stream, or prevent closure. 454 */ 455 456 #endif // NVIDIA_AUDIO_HW_H 457
