1 /* 2 * Copyright (C) 2015 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 DRAGON_AUDIO_HW_H 18 #define DRAGON_AUDIO_HW_H 19 20 #include <cutils/list.h> 21 #include <hardware/audio.h> 22 23 #include <tinyalsa/asoundlib.h> 24 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */ 25 #include <audio_utils/resampler.h> 26 #include <audio_route/audio_route.h> 27 28 #define PREPROCESSING_ENABLED 29 30 #define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib/hw/sound_trigger.primary.dragon.so" 31 32 /* Retry for delay in FW loading*/ 33 #define RETRY_NUMBER 10 34 #define RETRY_US 500000 35 36 #ifdef PREPROCESSING_ENABLED 37 #define MAX_PREPROCESSORS 3 38 struct effect_info_s { 39 effect_handle_t effect_itfe; 40 size_t num_channel_configs; 41 channel_config_t *channel_configs; 42 }; 43 #endif 44 45 #define TTY_MODE_OFF 1 46 #define TTY_MODE_FULL 2 47 #define TTY_MODE_VCO 4 48 #define TTY_MODE_HCO 8 49 50 #define DUALMIC_CONFIG_NONE 0 51 #define DUALMIC_CONFIG_1 1 52 53 /* Sound devices specific to the platform 54 * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound 55 * devices to enable corresponding mixer paths 56 */ 57 enum { 58 SND_DEVICE_NONE = 0, 59 60 /* Playback devices */ 61 SND_DEVICE_MIN, 62 SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN, 63 SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN, 64 SND_DEVICE_OUT_SPEAKER, 65 SND_DEVICE_OUT_HEADPHONES, 66 SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES, 67 SND_DEVICE_OUT_VOICE_HANDSET, 68 SND_DEVICE_OUT_VOICE_SPEAKER, 69 SND_DEVICE_OUT_VOICE_HEADPHONES, 70 SND_DEVICE_OUT_HDMI, 71 SND_DEVICE_OUT_SPEAKER_AND_HDMI, 72 SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES, 73 SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES, 74 SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET, 75 SND_DEVICE_OUT_END, 76 77 /* 78 * Note: IN_BEGIN should be same as OUT_END because total number of devices 79 * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices. 80 */ 81 /* Capture devices */ 82 SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END, 83 SND_DEVICE_IN_HANDSET_MIC = SND_DEVICE_IN_BEGIN, 84 SND_DEVICE_IN_SPEAKER_MIC, 85 SND_DEVICE_IN_HEADSET_MIC, 86 SND_DEVICE_IN_HANDSET_MIC_AEC, 87 SND_DEVICE_IN_SPEAKER_MIC_AEC, 88 SND_DEVICE_IN_HEADSET_MIC_AEC, 89 SND_DEVICE_IN_VOICE_SPEAKER_MIC, 90 SND_DEVICE_IN_VOICE_HEADSET_MIC, 91 SND_DEVICE_IN_HDMI_MIC, 92 SND_DEVICE_IN_CAMCORDER_MIC, 93 SND_DEVICE_IN_VOICE_DMIC_1, 94 SND_DEVICE_IN_VOICE_SPEAKER_DMIC_1, 95 SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC, 96 SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC, 97 SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC, 98 SND_DEVICE_IN_VOICE_REC_HEADSET_MIC, 99 SND_DEVICE_IN_VOICE_REC_MIC, 100 SND_DEVICE_IN_VOICE_REC_DMIC_1, 101 SND_DEVICE_IN_VOICE_REC_DMIC_NS_1, 102 SND_DEVICE_IN_LOOPBACK_AEC, 103 SND_DEVICE_IN_END, 104 105 SND_DEVICE_MAX = SND_DEVICE_IN_END, 106 107 }; 108 109 110 #define MIXER_CARD 0 111 #define SOUND_CARD 0 112 113 /* 114 * tinyAlsa library interprets period size as number of frames 115 * one frame = channel_count * sizeof (pcm sample) 116 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes 117 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes 118 * We should take care of returning proper size when AudioFlinger queries for 119 * the buffer size of an input/output stream 120 */ 121 #define PLAYBACK_PERIOD_SIZE 512 122 #define PLAYBACK_PERIOD_COUNT 2 123 #define PLAYBACK_DEFAULT_CHANNEL_COUNT 4 124 #define PLAYBACK_DEFAULT_SAMPLING_RATE 48000 125 #define PLAYBACK_START_THRESHOLD ((PLAYBACK_PERIOD_SIZE * PLAYBACK_PERIOD_COUNT) - 1) 126 #define PLAYBACK_STOP_THRESHOLD (PLAYBACK_PERIOD_SIZE * PLAYBACK_PERIOD_COUNT) 127 #define PLAYBACK_AVAILABLE_MIN 1 128 129 #define PLAYBACK_HDMI_MULTI_PERIOD_SIZE 1024 130 #define PLAYBACK_HDMI_MULTI_PERIOD_COUNT 4 131 #define PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6 132 #define PLAYBACK_HDMI_MULTI_PERIOD_BYTES \ 133 (PLAYBACK_HDMI_MULTI_PERIOD_SIZE * PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2) 134 #define PLAYBACK_HDMI_MULTI_START_THRESHOLD 4095 135 #define PLAYBACK_HDMI_MULTI_STOP_THRESHOLD 4096 136 #define PLAYBACK_HDMI_MULTI_AVAILABLE_MIN 1 137 138 #define PLAYBACK_HDMI_DEFAULT_CHANNEL_COUNT 2 139 140 #define CAPTURE_PERIOD_SIZE 1024 141 #define CAPTURE_PERIOD_SIZE_LOW_LATENCY 512 142 #define CAPTURE_PERIOD_COUNT 2 143 #define CAPTURE_DEFAULT_CHANNEL_COUNT 4 144 #define CAPTURE_DEFAULT_SAMPLING_RATE 48000 145 #define CAPTURE_START_THRESHOLD 1 146 147 #define DEEP_BUFFER_OUTPUT_SAMPLING_RATE 48000 148 #define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 1440 149 #define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8 150 151 #define MAX_SUPPORTED_CHANNEL_MASKS 2 152 153 struct cras_dsp_context; 154 155 typedef int snd_device_t; 156 157 /* These are the supported use cases by the hardware. 158 * Each usecase is mapped to a specific PCM device. 159 * Refer to pcm_device_table[]. 160 */ 161 typedef enum { 162 USECASE_INVALID = -1, 163 /* Playback usecases */ 164 USECASE_AUDIO_PLAYBACK = 0, 165 USECASE_AUDIO_PLAYBACK_MULTI_CH, 166 USECASE_AUDIO_PLAYBACK_DEEP_BUFFER, 167 168 /* Capture usecases */ 169 USECASE_AUDIO_CAPTURE, 170 USECASE_AUDIO_CAPTURE_HOTWORD, 171 172 USECASE_VOICE_CALL, 173 AUDIO_USECASE_MAX 174 } audio_usecase_t; 175 176 #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) 177 178 /* 179 * tinyAlsa library interprets period size as number of frames 180 * one frame = channel_count * sizeof (pcm sample) 181 * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes 182 * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes 183 * We should take care of returning proper size when AudioFlinger queries for 184 * the buffer size of an input/output stream 185 */ 186 187 typedef enum { 188 PCM_PLAYBACK = 0x1, 189 PCM_CAPTURE = 0x2, 190 VOICE_CALL = 0x4, 191 PCM_HOTWORD_STREAMING = 0x8 192 } usecase_type_t; 193 194 struct pcm_device_profile { 195 struct pcm_config config; 196 int card; 197 int device; 198 int id; 199 usecase_type_t type; 200 audio_devices_t devices; 201 const char* dsp_name; 202 }; 203 204 struct pcm_device { 205 struct listnode stream_list_node; 206 struct pcm_device_profile* pcm_profile; 207 struct pcm* pcm; 208 int status; 209 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */ 210 struct resampler_itfe* resampler; 211 int16_t* res_buffer; 212 size_t res_byte_count; 213 struct cras_dsp_context* dsp_context; 214 int sound_trigger_handle; 215 }; 216 217 struct stream_out { 218 struct audio_stream_out stream; 219 pthread_mutex_t lock; /* see note below on mutex acquisition order */ 220 pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */ 221 pthread_cond_t cond; 222 struct pcm_config config; 223 struct listnode pcm_dev_list; 224 int standby; 225 unsigned int sample_rate; 226 audio_channel_mask_t channel_mask; 227 audio_format_t format; 228 audio_devices_t devices; 229 audio_output_flags_t flags; 230 audio_usecase_t usecase; 231 /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */ 232 audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1]; 233 bool muted; 234 /* total frames written, not cleared when entering standby */ 235 uint64_t written; 236 audio_io_handle_t handle; 237 238 int non_blocking; 239 240 int send_new_metadata; 241 242 struct audio_device* dev; 243 void *proc_buf_out; 244 size_t proc_buf_size; 245 }; 246 247 struct stream_in { 248 struct audio_stream_in stream; 249 pthread_mutex_t lock; /* see note below on mutex acquisition order */ 250 pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by 251 capture thread */ 252 struct pcm_config config; 253 struct listnode pcm_dev_list; 254 int standby; 255 audio_source_t source; 256 audio_devices_t devices; 257 uint32_t main_channels; 258 audio_usecase_t usecase; 259 usecase_type_t usecase_type; 260 bool enable_aec; 261 audio_input_flags_t input_flags; 262 263 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */ 264 unsigned int requested_rate; 265 struct resampler_itfe* resampler; 266 struct resampler_buffer_provider buf_provider; 267 int read_status; 268 int16_t* read_buf; 269 size_t read_buf_size; 270 size_t read_buf_frames; 271 272 void *proc_buf_out; 273 size_t proc_buf_size; 274 275 #ifdef PREPROCESSING_ENABLED 276 void *proc_buf_in; 277 size_t proc_buf_frames; 278 279 int num_preprocessors; 280 struct effect_info_s preprocessors[MAX_PREPROCESSORS]; 281 282 bool aux_channels_changed; 283 uint32_t aux_channels; 284 #endif 285 286 struct audio_device* dev; 287 }; 288 289 struct mixer_card { 290 struct listnode adev_list_node; 291 struct listnode uc_list_node[AUDIO_USECASE_MAX]; 292 int card; 293 struct mixer* mixer; 294 struct audio_route* audio_route; 295 }; 296 297 struct audio_usecase { 298 struct listnode adev_list_node; 299 audio_usecase_t id; 300 usecase_type_t type; 301 audio_devices_t devices; 302 snd_device_t out_snd_device; 303 snd_device_t in_snd_device; 304 struct audio_stream* stream; 305 struct listnode mixer_list; 306 }; 307 308 309 struct audio_device { 310 struct audio_hw_device device; 311 pthread_mutex_t lock; /* see note below on mutex acquisition order */ 312 struct listnode mixer_list; 313 audio_mode_t mode; 314 struct stream_in* active_input; 315 struct stream_out* primary_output; 316 int in_call; 317 float voice_volume; 318 bool mic_mute; 319 int tty_mode; 320 bool bluetooth_nrec; 321 bool screen_off; 322 int* snd_dev_ref_cnt; 323 struct listnode usecase_list; 324 bool speaker_lr_swap; 325 unsigned int cur_hdmi_channels; 326 int dualmic_config; 327 bool ns_in_voice_rec; 328 329 void* sound_trigger_lib; 330 int (*sound_trigger_open_for_streaming)(); 331 size_t (*sound_trigger_read_samples)(int, void*, size_t); 332 int (*sound_trigger_close_for_streaming)(int); 333 334 int dummybuf_thread_timeout; 335 int dummybuf_thread_cancel; 336 int dummybuf_thread_active; 337 audio_devices_t dummybuf_thread_devices; 338 pthread_mutex_t dummybuf_thread_lock; 339 pthread_t dummybuf_thread; 340 341 pthread_mutex_t lock_inputs; /* see note below on mutex acquisition order */ 342 }; 343 344 /* 345 * NOTE: when multiple mutexes have to be acquired, always take the 346 * lock_inputs, stream_in, stream_out, and then audio_device. 347 * stream_in mutex must always be before stream_out mutex 348 * if both have to be taken (see get_echo_reference(), put_echo_reference()...) 349 * dummybuf_thread mutex is not related to the other mutexes with respect to order. 350 * lock_inputs must be held in order to either close the input stream, or prevent closure. 351 */ 352 353 #endif // NVIDIA_AUDIO_HW_H 354
