Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2013, The Linux Foundation. All rights reserved. |
| 3 | * Not a Contribution. |
| 4 | * |
| 5 | * Copyright (C) 2013 The Android Open Source Project |
| 6 | * |
| 7 | * Licensed under the Apache License, Version 2.0 (the "License"); |
| 8 | * you may not use this file except in compliance with the License. |
| 9 | * You may obtain a copy of the License at |
| 10 | * |
| 11 | * http://www.apache.org/licenses/LICENSE-2.0 |
| 12 | * |
| 13 | * Unless required by applicable law or agreed to in writing, software |
| 14 | * distributed under the License is distributed on an "AS IS" BASIS, |
| 15 | * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| 16 | * See the License for the specific language governing permissions and |
| 17 | * limitations under the License. |
| 18 | */ |
| 19 | |
| 20 | #define LOG_TAG "audio_hw_primary" |
| 21 | /*#define LOG_NDEBUG 0*/ |
| 22 | /*#define VERY_VERY_VERBOSE_LOGGING*/ |
| 23 | #ifdef VERY_VERY_VERBOSE_LOGGING |
| 24 | #define ALOGVV ALOGV |
| 25 | #else |
| 26 | #define ALOGVV(a...) do { } while(0) |
| 27 | #endif |
| 28 | |
| 29 | #include <errno.h> |
| 30 | #include <pthread.h> |
| 31 | #include <stdint.h> |
| 32 | #include <sys/time.h> |
| 33 | #include <stdlib.h> |
| 34 | #include <math.h> |
| 35 | #include <dlfcn.h> |
| 36 | #include <sys/resource.h> |
| 37 | #include <sys/prctl.h> |
| 38 | |
| 39 | #include <cutils/log.h> |
| 40 | #include <cutils/str_parms.h> |
| 41 | #include <cutils/properties.h> |
| 42 | #include <cutils/atomic.h> |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 43 | |
| 44 | #include <hardware/audio_effect.h> |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 45 | #include <audio_effects/effect_aec.h> |
| 46 | #include <audio_effects/effect_ns.h> |
| 47 | #include "audio_hw.h" |
| 48 | #include "platform_api.h" |
| 49 | #include <platform.h> |
| 50 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 51 | struct pcm_config pcm_config_audio_capture = { |
| 52 | .channels = 2, |
| 53 | .period_count = AUDIO_CAPTURE_PERIOD_COUNT, |
| 54 | .format = PCM_FORMAT_S16_LE, |
| 55 | }; |
| 56 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 57 | static struct audio_device *adev = NULL; |
| 58 | static pthread_mutex_t adev_init_lock; |
| 59 | static unsigned int audio_device_ref_count; |
| 60 | |
| 61 | static int set_voice_volume_l(struct audio_device *adev, float volume); |
| 62 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 63 | int enable_audio_route(struct audio_device *adev, |
| 64 | struct audio_usecase *usecase, |
| 65 | bool update_mixer) |
| 66 | { |
| 67 | snd_device_t snd_device; |
| 68 | char mixer_path[MIXER_PATH_MAX_LENGTH]; |
| 69 | |
| 70 | if (usecase == NULL) |
| 71 | return -EINVAL; |
| 72 | |
| 73 | ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); |
| 74 | |
| 75 | if (usecase->type == PCM_CAPTURE) |
| 76 | snd_device = usecase->in_snd_device; |
| 77 | else |
| 78 | snd_device = usecase->out_snd_device; |
| 79 | |
| 80 | strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path)); |
| 81 | platform_add_backend_name(mixer_path, snd_device); |
| 82 | ALOGV("%s: apply mixer path: %s", __func__, mixer_path); |
| 83 | audio_route_apply_path(adev->audio_route, mixer_path); |
| 84 | if (update_mixer) |
| 85 | audio_route_update_mixer(adev->audio_route); |
| 86 | |
| 87 | ALOGV("%s: exit", __func__); |
| 88 | return 0; |
| 89 | } |
| 90 | |
| 91 | int disable_audio_route(struct audio_device *adev, |
| 92 | struct audio_usecase *usecase, |
| 93 | bool update_mixer) |
| 94 | { |
| 95 | snd_device_t snd_device; |
| 96 | char mixer_path[MIXER_PATH_MAX_LENGTH]; |
| 97 | |
| 98 | if (usecase == NULL) |
| 99 | return -EINVAL; |
| 100 | |
| 101 | ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); |
| 102 | if (usecase->type == PCM_CAPTURE) |
| 103 | snd_device = usecase->in_snd_device; |
| 104 | else |
| 105 | snd_device = usecase->out_snd_device; |
| 106 | strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path)); |
| 107 | platform_add_backend_name(mixer_path, snd_device); |
| 108 | ALOGV("%s: reset mixer path: %s", __func__, mixer_path); |
| 109 | audio_route_reset_path(adev->audio_route, mixer_path); |
| 110 | if (update_mixer) |
| 111 | audio_route_update_mixer(adev->audio_route); |
| 112 | |
| 113 | ALOGV("%s: exit", __func__); |
| 114 | return 0; |
| 115 | } |
| 116 | |
| 117 | int enable_snd_device(struct audio_device *adev, |
| 118 | snd_device_t snd_device, |
| 119 | bool update_mixer) |
| 120 | { |
| 121 | char device_name[DEVICE_NAME_MAX_SIZE] = {0}; |
| 122 | |
| 123 | if (snd_device < SND_DEVICE_MIN || |
| 124 | snd_device >= SND_DEVICE_MAX) { |
| 125 | ALOGE("%s: Invalid sound device %d", __func__, snd_device); |
| 126 | return -EINVAL; |
| 127 | } |
| 128 | |
| 129 | adev->snd_dev_ref_cnt[snd_device]++; |
| 130 | |
| 131 | if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) { |
| 132 | ALOGE("%s: Invalid sound device returned", __func__); |
| 133 | return -EINVAL; |
| 134 | } |
| 135 | if (adev->snd_dev_ref_cnt[snd_device] > 1) { |
| 136 | ALOGV("%s: snd_device(%d: %s) is already active", |
| 137 | __func__, snd_device, device_name); |
| 138 | return 0; |
| 139 | } |
| 140 | |
| 141 | { |
| 142 | ALOGV("%s: snd_device(%d: %s)", __func__, |
| 143 | snd_device, device_name); |
| 144 | if (platform_send_audio_calibration(adev->platform, snd_device) < 0) { |
| 145 | adev->snd_dev_ref_cnt[snd_device]--; |
| 146 | return -EINVAL; |
| 147 | } |
| 148 | audio_route_apply_path(adev->audio_route, device_name); |
| 149 | } |
| 150 | if (update_mixer) |
| 151 | audio_route_update_mixer(adev->audio_route); |
| 152 | |
| 153 | return 0; |
| 154 | } |
| 155 | |
| 156 | int disable_snd_device(struct audio_device *adev, |
| 157 | snd_device_t snd_device, |
| 158 | bool update_mixer) |
| 159 | { |
| 160 | char device_name[DEVICE_NAME_MAX_SIZE] = {0}; |
| 161 | |
| 162 | if (snd_device < SND_DEVICE_MIN || |
| 163 | snd_device >= SND_DEVICE_MAX) { |
| 164 | ALOGE("%s: Invalid sound device %d", __func__, snd_device); |
| 165 | return -EINVAL; |
| 166 | } |
| 167 | if (adev->snd_dev_ref_cnt[snd_device] <= 0) { |
| 168 | ALOGE("%s: device ref cnt is already 0", __func__); |
| 169 | return -EINVAL; |
| 170 | } |
| 171 | |
| 172 | adev->snd_dev_ref_cnt[snd_device]--; |
| 173 | |
| 174 | if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) { |
| 175 | ALOGE("%s: Invalid sound device returned", __func__); |
| 176 | return -EINVAL; |
| 177 | } |
| 178 | |
| 179 | if (adev->snd_dev_ref_cnt[snd_device] == 0) { |
| 180 | ALOGV("%s: snd_device(%d: %s)", __func__, |
| 181 | snd_device, device_name); |
| 182 | audio_route_reset_path(adev->audio_route, device_name); |
| 183 | |
| 184 | if (update_mixer) |
| 185 | audio_route_update_mixer(adev->audio_route); |
| 186 | } |
| 187 | |
| 188 | return 0; |
| 189 | } |
| 190 | |
| 191 | static void check_usecases_codec_backend(struct audio_device *adev, |
| 192 | struct audio_usecase *uc_info, |
| 193 | snd_device_t snd_device) |
| 194 | { |
| 195 | struct listnode *node; |
| 196 | struct audio_usecase *usecase; |
| 197 | bool switch_device[AUDIO_USECASE_MAX]; |
| 198 | int i, num_uc_to_switch = 0; |
| 199 | |
| 200 | /* |
| 201 | * This function is to make sure that all the usecases that are active on |
| 202 | * the hardware codec backend are always routed to any one device that is |
| 203 | * handled by the hardware codec. |
| 204 | * For example, if low-latency and deep-buffer usecases are currently active |
| 205 | * on speaker and out_set_parameters(headset) is received on low-latency |
| 206 | * output, then we have to make sure deep-buffer is also switched to headset, |
| 207 | * because of the limitation that both the devices cannot be enabled |
| 208 | * at the same time as they share the same backend. |
| 209 | */ |
| 210 | /* Disable all the usecases on the shared backend other than the |
| 211 | specified usecase */ |
| 212 | for (i = 0; i < AUDIO_USECASE_MAX; i++) |
| 213 | switch_device[i] = false; |
| 214 | |
| 215 | list_for_each(node, &adev->usecase_list) { |
| 216 | usecase = node_to_item(node, struct audio_usecase, list); |
| 217 | if (usecase->type == PCM_PLAYBACK && |
| 218 | usecase != uc_info && |
| 219 | usecase->out_snd_device != snd_device && |
| 220 | usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) { |
| 221 | ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", |
| 222 | __func__, use_case_table[usecase->id], |
| 223 | platform_get_snd_device_name(usecase->out_snd_device)); |
| 224 | disable_audio_route(adev, usecase, false); |
| 225 | switch_device[usecase->id] = true; |
| 226 | num_uc_to_switch++; |
| 227 | } |
| 228 | } |
| 229 | |
| 230 | if (num_uc_to_switch) { |
| 231 | /* Make sure all the streams are de-routed before disabling the device */ |
| 232 | audio_route_update_mixer(adev->audio_route); |
| 233 | |
| 234 | list_for_each(node, &adev->usecase_list) { |
| 235 | usecase = node_to_item(node, struct audio_usecase, list); |
| 236 | if (switch_device[usecase->id]) { |
| 237 | disable_snd_device(adev, usecase->out_snd_device, false); |
| 238 | } |
| 239 | } |
| 240 | |
| 241 | list_for_each(node, &adev->usecase_list) { |
| 242 | usecase = node_to_item(node, struct audio_usecase, list); |
| 243 | if (switch_device[usecase->id]) { |
| 244 | enable_snd_device(adev, snd_device, false); |
| 245 | } |
| 246 | } |
| 247 | /* Make sure new snd device is enabled before re-routing the streams */ |
| 248 | audio_route_update_mixer(adev->audio_route); |
| 249 | |
| 250 | /* Re-route all the usecases on the shared backend other than the |
| 251 | specified usecase to new snd devices */ |
| 252 | list_for_each(node, &adev->usecase_list) { |
| 253 | usecase = node_to_item(node, struct audio_usecase, list); |
| 254 | /* Update the out_snd_device only before enabling the audio route */ |
| 255 | if (switch_device[usecase->id] ) { |
| 256 | usecase->out_snd_device = snd_device; |
| 257 | enable_audio_route(adev, usecase, false); |
| 258 | } |
| 259 | } |
| 260 | |
| 261 | audio_route_update_mixer(adev->audio_route); |
| 262 | } |
| 263 | } |
| 264 | |
| 265 | static void check_and_route_capture_usecases(struct audio_device *adev, |
| 266 | struct audio_usecase *uc_info, |
| 267 | snd_device_t snd_device) |
| 268 | { |
| 269 | struct listnode *node; |
| 270 | struct audio_usecase *usecase; |
| 271 | bool switch_device[AUDIO_USECASE_MAX]; |
| 272 | int i, num_uc_to_switch = 0; |
| 273 | |
| 274 | /* |
| 275 | * This function is to make sure that all the active capture usecases |
| 276 | * are always routed to the same input sound device. |
| 277 | * For example, if audio-record and voice-call usecases are currently |
| 278 | * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece) |
| 279 | * is received for voice call then we have to make sure that audio-record |
| 280 | * usecase is also switched to earpiece i.e. voice-dmic-ef, |
| 281 | * because of the limitation that two devices cannot be enabled |
| 282 | * at the same time if they share the same backend. |
| 283 | */ |
| 284 | for (i = 0; i < AUDIO_USECASE_MAX; i++) |
| 285 | switch_device[i] = false; |
| 286 | |
| 287 | list_for_each(node, &adev->usecase_list) { |
| 288 | usecase = node_to_item(node, struct audio_usecase, list); |
| 289 | if (usecase->type == PCM_CAPTURE && |
| 290 | usecase != uc_info && |
| 291 | usecase->in_snd_device != snd_device) { |
| 292 | ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", |
| 293 | __func__, use_case_table[usecase->id], |
| 294 | platform_get_snd_device_name(usecase->in_snd_device)); |
| 295 | disable_audio_route(adev, usecase, false); |
| 296 | switch_device[usecase->id] = true; |
| 297 | num_uc_to_switch++; |
| 298 | } |
| 299 | } |
| 300 | |
| 301 | if (num_uc_to_switch) { |
| 302 | /* Make sure all the streams are de-routed before disabling the device */ |
| 303 | audio_route_update_mixer(adev->audio_route); |
| 304 | |
| 305 | list_for_each(node, &adev->usecase_list) { |
| 306 | usecase = node_to_item(node, struct audio_usecase, list); |
| 307 | if (switch_device[usecase->id]) { |
| 308 | disable_snd_device(adev, usecase->in_snd_device, false); |
| 309 | enable_snd_device(adev, snd_device, false); |
| 310 | } |
| 311 | } |
| 312 | |
| 313 | /* Make sure new snd device is enabled before re-routing the streams */ |
| 314 | audio_route_update_mixer(adev->audio_route); |
| 315 | |
| 316 | /* Re-route all the usecases on the shared backend other than the |
| 317 | specified usecase to new snd devices */ |
| 318 | list_for_each(node, &adev->usecase_list) { |
| 319 | usecase = node_to_item(node, struct audio_usecase, list); |
| 320 | /* Update the in_snd_device only before enabling the audio route */ |
| 321 | if (switch_device[usecase->id] ) { |
| 322 | usecase->in_snd_device = snd_device; |
| 323 | enable_audio_route(adev, usecase, false); |
| 324 | } |
| 325 | } |
| 326 | |
| 327 | audio_route_update_mixer(adev->audio_route); |
| 328 | } |
| 329 | } |
| 330 | |
| 331 | static int disable_all_usecases_of_type(struct audio_device *adev, |
| 332 | usecase_type_t usecase_type, |
| 333 | bool update_mixer) |
| 334 | { |
| 335 | struct audio_usecase *usecase; |
| 336 | struct listnode *node; |
| 337 | int ret = 0; |
| 338 | |
| 339 | list_for_each(node, &adev->usecase_list) { |
| 340 | usecase = node_to_item(node, struct audio_usecase, list); |
| 341 | if (usecase->type == usecase_type) { |
| 342 | ALOGV("%s: usecase id %d", __func__, usecase->id); |
| 343 | ret = disable_audio_route(adev, usecase, update_mixer); |
| 344 | if (ret) { |
| 345 | ALOGE("%s: Failed to disable usecase id %d", |
| 346 | __func__, usecase->id); |
| 347 | } |
| 348 | } |
| 349 | } |
| 350 | |
| 351 | return ret; |
| 352 | } |
| 353 | |
| 354 | static int enable_all_usecases_of_type(struct audio_device *adev, |
| 355 | usecase_type_t usecase_type, |
| 356 | bool update_mixer) |
| 357 | { |
| 358 | struct audio_usecase *usecase; |
| 359 | struct listnode *node; |
| 360 | int ret = 0; |
| 361 | |
| 362 | list_for_each(node, &adev->usecase_list) { |
| 363 | usecase = node_to_item(node, struct audio_usecase, list); |
| 364 | if (usecase->type == usecase_type) { |
| 365 | ALOGV("%s: usecase id %d", __func__, usecase->id); |
| 366 | ret = enable_audio_route(adev, usecase, update_mixer); |
| 367 | if (ret) { |
| 368 | ALOGE("%s: Failed to enable usecase id %d", |
| 369 | __func__, usecase->id); |
| 370 | } |
| 371 | } |
| 372 | } |
| 373 | |
| 374 | return ret; |
| 375 | } |
| 376 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 377 | static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev) |
| 378 | { |
| 379 | struct audio_usecase *usecase; |
| 380 | struct listnode *node; |
| 381 | |
| 382 | list_for_each(node, &adev->usecase_list) { |
| 383 | usecase = node_to_item(node, struct audio_usecase, list); |
| 384 | if (usecase->type == VOICE_CALL) { |
| 385 | ALOGV("%s: usecase id %d", __func__, usecase->id); |
| 386 | return usecase->id; |
| 387 | } |
| 388 | } |
| 389 | return USECASE_INVALID; |
| 390 | } |
| 391 | |
| 392 | struct audio_usecase *get_usecase_from_list(struct audio_device *adev, |
| 393 | audio_usecase_t uc_id) |
| 394 | { |
| 395 | struct audio_usecase *usecase; |
| 396 | struct listnode *node; |
| 397 | |
| 398 | list_for_each(node, &adev->usecase_list) { |
| 399 | usecase = node_to_item(node, struct audio_usecase, list); |
| 400 | if (usecase->id == uc_id) |
| 401 | return usecase; |
| 402 | } |
| 403 | return NULL; |
| 404 | } |
| 405 | |
| 406 | int select_devices(struct audio_device *adev, audio_usecase_t uc_id) |
| 407 | { |
| 408 | snd_device_t out_snd_device = SND_DEVICE_NONE; |
| 409 | snd_device_t in_snd_device = SND_DEVICE_NONE; |
| 410 | struct audio_usecase *usecase = NULL; |
| 411 | struct audio_usecase *vc_usecase = NULL; |
| 412 | struct audio_usecase *voip_usecase = NULL; |
| 413 | struct listnode *node; |
| 414 | int status = 0; |
| 415 | |
| 416 | usecase = get_usecase_from_list(adev, uc_id); |
| 417 | if (usecase == NULL) { |
| 418 | ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id); |
| 419 | return -EINVAL; |
| 420 | } |
| 421 | |
| 422 | if ((usecase->type == VOICE_CALL) || |
| 423 | (usecase->type == VOIP_CALL)) { |
| 424 | out_snd_device = platform_get_output_snd_device(adev->platform, |
| 425 | usecase->stream.out->devices); |
| 426 | in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices); |
| 427 | usecase->devices = usecase->stream.out->devices; |
| 428 | } else { |
| 429 | /* |
| 430 | * If the voice call is active, use the sound devices of voice call usecase |
| 431 | * so that it would not result any device switch. All the usecases will |
| 432 | * be switched to new device when select_devices() is called for voice call |
| 433 | * usecase. This is to avoid switching devices for voice call when |
| 434 | * check_usecases_codec_backend() is called below. |
| 435 | */ |
| 436 | if (usecase->type == PCM_PLAYBACK) { |
| 437 | usecase->devices = usecase->stream.out->devices; |
| 438 | in_snd_device = SND_DEVICE_NONE; |
| 439 | if (out_snd_device == SND_DEVICE_NONE) { |
| 440 | out_snd_device = platform_get_output_snd_device(adev->platform, |
| 441 | usecase->stream.out->devices); |
| 442 | if (usecase->stream.out == adev->primary_output && |
| 443 | adev->active_input && |
| 444 | adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { |
| 445 | select_devices(adev, adev->active_input->usecase); |
| 446 | } |
| 447 | } |
| 448 | } else if (usecase->type == PCM_CAPTURE) { |
| 449 | usecase->devices = usecase->stream.in->device; |
| 450 | out_snd_device = SND_DEVICE_NONE; |
| 451 | if (in_snd_device == SND_DEVICE_NONE) { |
| 452 | if (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION && |
| 453 | adev->primary_output && !adev->primary_output->standby) { |
| 454 | in_snd_device = platform_get_input_snd_device(adev->platform, |
| 455 | adev->primary_output->devices); |
| 456 | } else { |
| 457 | in_snd_device = platform_get_input_snd_device(adev->platform, |
| 458 | AUDIO_DEVICE_NONE); |
| 459 | } |
| 460 | } |
| 461 | } |
| 462 | } |
| 463 | |
| 464 | if (out_snd_device == usecase->out_snd_device && |
| 465 | in_snd_device == usecase->in_snd_device) { |
| 466 | return 0; |
| 467 | } |
| 468 | |
| 469 | ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__, |
| 470 | out_snd_device, platform_get_snd_device_name(out_snd_device), |
| 471 | in_snd_device, platform_get_snd_device_name(in_snd_device)); |
| 472 | |
| 473 | /* |
| 474 | * Limitation: While in call, to do a device switch we need to disable |
| 475 | * and enable both RX and TX devices though one of them is same as current |
| 476 | * device. |
| 477 | */ |
| 478 | if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) { |
| 479 | status = platform_switch_voice_call_device_pre(adev->platform); |
| 480 | disable_all_usecases_of_type(adev, VOICE_CALL, true); |
| 481 | } |
| 482 | |
| 483 | /* Disable current sound devices */ |
| 484 | if (usecase->out_snd_device != SND_DEVICE_NONE) { |
| 485 | disable_audio_route(adev, usecase, true); |
| 486 | disable_snd_device(adev, usecase->out_snd_device, false); |
| 487 | } |
| 488 | |
| 489 | if (usecase->in_snd_device != SND_DEVICE_NONE) { |
| 490 | disable_audio_route(adev, usecase, true); |
| 491 | disable_snd_device(adev, usecase->in_snd_device, false); |
| 492 | } |
| 493 | |
| 494 | /* Enable new sound devices */ |
| 495 | if (out_snd_device != SND_DEVICE_NONE) { |
| 496 | if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) |
| 497 | check_usecases_codec_backend(adev, usecase, out_snd_device); |
| 498 | enable_snd_device(adev, out_snd_device, false); |
| 499 | } |
| 500 | |
| 501 | if (in_snd_device != SND_DEVICE_NONE) { |
| 502 | check_and_route_capture_usecases(adev, usecase, in_snd_device); |
| 503 | enable_snd_device(adev, in_snd_device, false); |
| 504 | } |
| 505 | |
| 506 | if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) |
| 507 | status = platform_switch_voice_call_device_post(adev->platform, |
| 508 | out_snd_device, |
| 509 | in_snd_device); |
| 510 | |
| 511 | audio_route_update_mixer(adev->audio_route); |
| 512 | |
| 513 | usecase->in_snd_device = in_snd_device; |
| 514 | usecase->out_snd_device = out_snd_device; |
| 515 | |
| 516 | if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) |
| 517 | enable_all_usecases_of_type(adev, usecase->type, true); |
| 518 | else |
| 519 | enable_audio_route(adev, usecase, true); |
| 520 | |
| 521 | /* Applicable only on the targets that has external modem. |
| 522 | * Enable device command should be sent to modem only after |
| 523 | * enabling voice call mixer controls |
| 524 | */ |
| 525 | if (usecase->type == VOICE_CALL) |
| 526 | status = platform_switch_voice_call_usecase_route_post(adev->platform, |
| 527 | out_snd_device, |
| 528 | in_snd_device); |
| 529 | |
| 530 | return status; |
| 531 | } |
| 532 | |
| 533 | static int stop_input_stream(struct stream_in *in) |
| 534 | { |
| 535 | int i, ret = 0; |
| 536 | struct audio_usecase *uc_info; |
| 537 | struct audio_device *adev = in->dev; |
| 538 | |
| 539 | adev->active_input = NULL; |
| 540 | |
| 541 | ALOGV("%s: enter: usecase(%d: %s)", __func__, |
| 542 | in->usecase, use_case_table[in->usecase]); |
| 543 | uc_info = get_usecase_from_list(adev, in->usecase); |
| 544 | if (uc_info == NULL) { |
| 545 | ALOGE("%s: Could not find the usecase (%d) in the list", |
| 546 | __func__, in->usecase); |
| 547 | return -EINVAL; |
| 548 | } |
| 549 | |
| 550 | /* 1. Disable stream specific mixer controls */ |
| 551 | disable_audio_route(adev, uc_info, true); |
| 552 | |
| 553 | /* 2. Disable the tx device */ |
| 554 | disable_snd_device(adev, uc_info->in_snd_device, true); |
| 555 | |
| 556 | list_remove(&uc_info->list); |
| 557 | free(uc_info); |
| 558 | |
| 559 | ALOGV("%s: exit: status(%d)", __func__, ret); |
| 560 | return ret; |
| 561 | } |
| 562 | |
| 563 | int start_input_stream(struct stream_in *in) |
| 564 | { |
| 565 | /* 1. Enable output device and stream routing controls */ |
| 566 | int ret = 0; |
| 567 | struct audio_usecase *uc_info; |
| 568 | struct audio_device *adev = in->dev; |
| 569 | |
| 570 | in->usecase = platform_update_usecase_from_source(in->source,in->usecase); |
| 571 | ALOGV("%s: enter: usecase(%d)", __func__, in->usecase); |
| 572 | |
| 573 | in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE); |
| 574 | if (in->pcm_device_id < 0) { |
| 575 | ALOGE("%s: Could not find PCM device id for the usecase(%d)", |
| 576 | __func__, in->usecase); |
| 577 | ret = -EINVAL; |
| 578 | goto error_config; |
| 579 | } |
| 580 | |
| 581 | adev->active_input = in; |
| 582 | uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); |
| 583 | uc_info->id = in->usecase; |
| 584 | uc_info->type = PCM_CAPTURE; |
| 585 | uc_info->stream.in = in; |
| 586 | uc_info->devices = in->device; |
| 587 | uc_info->in_snd_device = SND_DEVICE_NONE; |
| 588 | uc_info->out_snd_device = SND_DEVICE_NONE; |
| 589 | |
| 590 | list_add_tail(&adev->usecase_list, &uc_info->list); |
| 591 | select_devices(adev, in->usecase); |
| 592 | |
| 593 | ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d", |
| 594 | __func__, SOUND_CARD, in->pcm_device_id, in->config.channels); |
| 595 | in->pcm = pcm_open(SOUND_CARD, in->pcm_device_id, |
| 596 | PCM_IN, &in->config); |
| 597 | if (in->pcm && !pcm_is_ready(in->pcm)) { |
| 598 | ALOGE("%s: %s", __func__, pcm_get_error(in->pcm)); |
| 599 | pcm_close(in->pcm); |
| 600 | in->pcm = NULL; |
| 601 | ret = -EIO; |
| 602 | goto error_open; |
| 603 | } |
| 604 | ALOGV("%s: exit", __func__); |
| 605 | return ret; |
| 606 | |
| 607 | error_open: |
| 608 | stop_input_stream(in); |
| 609 | |
| 610 | error_config: |
| 611 | adev->active_input = NULL; |
| 612 | ALOGD("%s: exit: status(%d)", __func__, ret); |
| 613 | |
| 614 | return ret; |
| 615 | } |
| 616 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 617 | static int check_input_parameters(uint32_t sample_rate, |
| 618 | audio_format_t format, |
| 619 | int channel_count) |
| 620 | { |
| 621 | int ret = 0; |
| 622 | |
| 623 | if ((format != AUDIO_FORMAT_PCM_16_BIT)) ret = -EINVAL; |
| 624 | |
| 625 | switch (channel_count) { |
| 626 | case 1: |
| 627 | case 2: |
| 628 | case 6: |
| 629 | break; |
| 630 | default: |
| 631 | ret = -EINVAL; |
| 632 | } |
| 633 | |
| 634 | switch (sample_rate) { |
| 635 | case 8000: |
| 636 | case 11025: |
| 637 | case 12000: |
| 638 | case 16000: |
| 639 | case 22050: |
| 640 | case 24000: |
| 641 | case 32000: |
| 642 | case 44100: |
| 643 | case 48000: |
| 644 | break; |
| 645 | default: |
| 646 | ret = -EINVAL; |
| 647 | } |
| 648 | |
| 649 | return ret; |
| 650 | } |
| 651 | |
| 652 | static size_t get_input_buffer_size(uint32_t sample_rate, |
| 653 | audio_format_t format, |
| 654 | int channel_count) |
| 655 | { |
| 656 | size_t size = 0; |
| 657 | |
| 658 | if (check_input_parameters(sample_rate, format, channel_count) != 0) |
| 659 | return 0; |
| 660 | |
| 661 | size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000; |
| 662 | /* ToDo: should use frame_size computed based on the format and |
| 663 | channel_count here. */ |
| 664 | size *= sizeof(short) * channel_count; |
| 665 | |
| 666 | /* make sure the size is multiple of 64 */ |
| 667 | size += 0x3f; |
| 668 | size &= ~0x3f; |
| 669 | |
| 670 | return size; |
| 671 | } |
| 672 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 673 | /** audio_stream_in implementation **/ |
| 674 | static uint32_t in_get_sample_rate(const struct audio_stream *stream) |
| 675 | { |
| 676 | struct stream_in *in = (struct stream_in *)stream; |
| 677 | |
| 678 | return in->config.rate; |
| 679 | } |
| 680 | |
| 681 | static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) |
| 682 | { |
| 683 | return -ENOSYS; |
| 684 | } |
| 685 | |
| 686 | static size_t in_get_buffer_size(const struct audio_stream *stream) |
| 687 | { |
| 688 | struct stream_in *in = (struct stream_in *)stream; |
| 689 | |
| 690 | return in->config.period_size * audio_stream_frame_size(stream); |
| 691 | } |
| 692 | |
| 693 | static uint32_t in_get_channels(const struct audio_stream *stream) |
| 694 | { |
| 695 | struct stream_in *in = (struct stream_in *)stream; |
| 696 | |
| 697 | return in->channel_mask; |
| 698 | } |
| 699 | |
| 700 | static audio_format_t in_get_format(const struct audio_stream *stream) |
| 701 | { |
| 702 | struct stream_in *in = (struct stream_in *)stream; |
| 703 | |
| 704 | return in->format; |
| 705 | } |
| 706 | |
| 707 | static int in_set_format(struct audio_stream *stream, audio_format_t format) |
| 708 | { |
| 709 | return -ENOSYS; |
| 710 | } |
| 711 | |
| 712 | static int in_standby(struct audio_stream *stream) |
| 713 | { |
| 714 | struct stream_in *in = (struct stream_in *)stream; |
| 715 | struct audio_device *adev = in->dev; |
| 716 | int status = 0; |
| 717 | ALOGV("%s: enter", __func__); |
| 718 | |
| 719 | if (in->usecase == USECASE_COMPRESS_VOIP_CALL) { |
| 720 | /* Ignore standby in case of voip call because the voip input |
| 721 | * stream is closed in adev_close_input_stream() |
| 722 | */ |
| 723 | ALOGV("%s: Ignore Standby in VOIP call", __func__); |
| 724 | return status; |
| 725 | } |
| 726 | |
| 727 | pthread_mutex_lock(&in->lock); |
| 728 | if (!in->standby) { |
| 729 | in->standby = true; |
| 730 | if (in->pcm) { |
| 731 | pcm_close(in->pcm); |
| 732 | in->pcm = NULL; |
| 733 | } |
| 734 | pthread_mutex_lock(&adev->lock); |
| 735 | status = stop_input_stream(in); |
| 736 | pthread_mutex_unlock(&adev->lock); |
| 737 | } |
| 738 | pthread_mutex_unlock(&in->lock); |
| 739 | ALOGV("%s: exit: status(%d)", __func__, status); |
| 740 | return status; |
| 741 | } |
| 742 | |
| 743 | static int in_dump(const struct audio_stream *stream, int fd) |
| 744 | { |
| 745 | return 0; |
| 746 | } |
| 747 | |
| 748 | static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) |
| 749 | { |
| 750 | struct stream_in *in = (struct stream_in *)stream; |
| 751 | struct audio_device *adev = in->dev; |
| 752 | struct str_parms *parms; |
| 753 | char *str; |
| 754 | char value[32]; |
| 755 | int ret, val = 0; |
| 756 | |
| 757 | ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs); |
| 758 | parms = str_parms_create_str(kvpairs); |
| 759 | |
| 760 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value)); |
| 761 | |
| 762 | pthread_mutex_lock(&in->lock); |
| 763 | pthread_mutex_lock(&adev->lock); |
| 764 | if (ret >= 0) { |
| 765 | val = atoi(value); |
| 766 | /* no audio source uses val == 0 */ |
| 767 | if ((in->source != val) && (val != 0)) { |
| 768 | in->source = val; |
| 769 | } |
| 770 | } |
| 771 | |
| 772 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); |
| 773 | if (ret >= 0) { |
| 774 | val = atoi(value); |
| 775 | if ((in->device != val) && (val != 0)) { |
| 776 | in->device = val; |
| 777 | /* If recording is in progress, change the tx device to new device */ |
| 778 | if (!in->standby) |
| 779 | ret = select_devices(adev, in->usecase); |
| 780 | } |
| 781 | } |
| 782 | |
| 783 | pthread_mutex_unlock(&adev->lock); |
| 784 | pthread_mutex_unlock(&in->lock); |
| 785 | |
| 786 | str_parms_destroy(parms); |
| 787 | ALOGV("%s: exit: status(%d)", __func__, ret); |
| 788 | return ret; |
| 789 | } |
| 790 | |
| 791 | static char* in_get_parameters(const struct audio_stream *stream, |
| 792 | const char *keys) |
| 793 | { |
| 794 | struct stream_in *in = (struct stream_in *)stream; |
| 795 | struct str_parms *query = str_parms_create_str(keys); |
| 796 | char *str; |
| 797 | char value[256]; |
| 798 | struct str_parms *reply = str_parms_create(); |
| 799 | ALOGV("%s: enter: keys - %s", __func__, keys); |
| 800 | |
| 801 | str = str_parms_to_str(reply); |
| 802 | str_parms_destroy(query); |
| 803 | str_parms_destroy(reply); |
| 804 | |
| 805 | ALOGV("%s: exit: returns - %s", __func__, str); |
| 806 | return str; |
| 807 | } |
| 808 | |
| 809 | static int in_set_gain(struct audio_stream_in *stream, float gain) |
| 810 | { |
| 811 | return 0; |
| 812 | } |
| 813 | |
| 814 | static ssize_t in_read(struct audio_stream_in *stream, void *buffer, |
| 815 | size_t bytes) |
| 816 | { |
| 817 | struct stream_in *in = (struct stream_in *)stream; |
| 818 | struct audio_device *adev = in->dev; |
| 819 | int i, ret = -1; |
| 820 | |
| 821 | pthread_mutex_lock(&in->lock); |
| 822 | if (in->standby) { |
| 823 | pthread_mutex_lock(&adev->lock); |
| 824 | ret = start_input_stream(in); |
| 825 | pthread_mutex_unlock(&adev->lock); |
| 826 | if (ret != 0) { |
| 827 | goto exit; |
| 828 | } |
| 829 | in->standby = 0; |
| 830 | } |
| 831 | |
| 832 | if (in->pcm) { |
| 833 | ret = pcm_read(in->pcm, buffer, bytes); |
| 834 | } |
| 835 | |
| 836 | exit: |
| 837 | pthread_mutex_unlock(&in->lock); |
| 838 | |
| 839 | if (ret != 0) { |
| 840 | in_standby(&in->stream.common); |
| 841 | ALOGV("%s: read failed - sleeping for buffer duration", __func__); |
| 842 | usleep(bytes * 1000000 / audio_stream_frame_size(&in->stream.common) / |
| 843 | in_get_sample_rate(&in->stream.common)); |
| 844 | } |
| 845 | return bytes; |
| 846 | } |
| 847 | |
| 848 | static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) |
| 849 | { |
| 850 | return 0; |
| 851 | } |
| 852 | |
| 853 | static int add_remove_audio_effect(const struct audio_stream *stream, |
| 854 | effect_handle_t effect, |
| 855 | bool enable) |
| 856 | { |
| 857 | struct stream_in *in = (struct stream_in *)stream; |
| 858 | int status = 0; |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 859 | |
| 860 | return 0; |
| 861 | } |
| 862 | |
| 863 | static int in_add_audio_effect(const struct audio_stream *stream, |
| 864 | effect_handle_t effect) |
| 865 | { |
| 866 | ALOGV("%s: effect %p", __func__, effect); |
| 867 | return add_remove_audio_effect(stream, effect, true); |
| 868 | } |
| 869 | |
| 870 | static int in_remove_audio_effect(const struct audio_stream *stream, |
| 871 | effect_handle_t effect) |
| 872 | { |
| 873 | ALOGV("%s: effect %p", __func__, effect); |
| 874 | return add_remove_audio_effect(stream, effect, false); |
| 875 | } |
| 876 | |
Dhananjay Kumar | daf6ebb | 2013-10-07 11:38:46 -0700 | [diff] [blame] | 877 | static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) |
| 878 | { |
| 879 | struct audio_device *adev = (struct audio_device *)dev; |
| 880 | struct str_parms *parms; |
| 881 | char *str; |
| 882 | char value[32]; |
| 883 | int val; |
| 884 | int ret; |
| 885 | |
| 886 | ALOGD("%s: enter: %s", __func__, kvpairs); |
| 887 | |
| 888 | pthread_mutex_lock(&adev->lock); |
| 889 | parms = str_parms_create_str(kvpairs); |
| 890 | |
| 891 | platform_set_parameters(adev->platform, parms); |
| 892 | |
| 893 | ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value)); |
| 894 | if (ret >= 0) { |
| 895 | /* When set to false, HAL should disable EC and NS |
| 896 | * But it is currently not supported. |
| 897 | */ |
| 898 | if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) |
| 899 | adev->bluetooth_nrec = true; |
| 900 | else |
| 901 | adev->bluetooth_nrec = false; |
| 902 | } |
| 903 | |
| 904 | ret = str_parms_get_str(parms, "screen_state", value, sizeof(value)); |
| 905 | if (ret >= 0) { |
| 906 | if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) |
| 907 | adev->screen_off = false; |
| 908 | else |
| 909 | adev->screen_off = true; |
| 910 | } |
| 911 | |
| 912 | ret = str_parms_get_int(parms, "rotation", &val); |
| 913 | if (ret >= 0) { |
| 914 | bool reverse_speakers = false; |
| 915 | switch(val) { |
| 916 | // FIXME: note that the code below assumes that the speakers are in the correct placement |
| 917 | // relative to the user when the device is rotated 90deg from its default rotation. This |
| 918 | // assumption is device-specific, not platform-specific like this code. |
| 919 | case 270: |
| 920 | reverse_speakers = true; |
| 921 | break; |
| 922 | case 0: |
| 923 | case 90: |
| 924 | case 180: |
| 925 | break; |
| 926 | default: |
| 927 | ALOGE("%s: unexpected rotation of %d", __func__, val); |
| 928 | } |
| 929 | if (adev->speaker_lr_swap != reverse_speakers) { |
| 930 | adev->speaker_lr_swap = reverse_speakers; |
| 931 | // only update the selected device if there is active pcm playback |
| 932 | struct audio_usecase *usecase; |
| 933 | struct listnode *node; |
| 934 | list_for_each(node, &adev->usecase_list) { |
| 935 | usecase = node_to_item(node, struct audio_usecase, list); |
| 936 | if (usecase->type == PCM_PLAYBACK) { |
| 937 | select_devices(adev, usecase->id); |
| 938 | break; |
| 939 | } |
| 940 | } |
| 941 | } |
| 942 | } |
| 943 | |
| 944 | str_parms_destroy(parms); |
| 945 | |
| 946 | pthread_mutex_unlock(&adev->lock); |
| 947 | ALOGV("%s: exit with code(%d)", __func__, ret); |
| 948 | return ret; |
| 949 | } |
| 950 | |
| 951 | static char* adev_get_parameters(const struct audio_hw_device *dev, |
| 952 | const char *keys) |
| 953 | { |
| 954 | struct audio_device *adev = (struct audio_device *)dev; |
| 955 | struct str_parms *reply = str_parms_create(); |
| 956 | struct str_parms *query = str_parms_create_str(keys); |
| 957 | char *str; |
| 958 | |
| 959 | pthread_mutex_lock(&adev->lock); |
| 960 | |
| 961 | platform_get_parameters(adev->platform, query, reply); |
| 962 | str = str_parms_to_str(reply); |
| 963 | str_parms_destroy(query); |
| 964 | str_parms_destroy(reply); |
| 965 | |
| 966 | pthread_mutex_unlock(&adev->lock); |
| 967 | ALOGV("%s: exit: returns - %s", __func__, str); |
| 968 | return str; |
| 969 | } |
| 970 | |
| 971 | static int adev_init_check(const struct audio_hw_device *dev) |
| 972 | { |
| 973 | return 0; |
| 974 | } |
| 975 | |
| 976 | static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) |
| 977 | { |
| 978 | int ret = 0; |
| 979 | return ret; |
| 980 | } |
| 981 | |
| 982 | static int adev_set_master_volume(struct audio_hw_device *dev, float volume) |
| 983 | { |
| 984 | return -ENOSYS; |
| 985 | } |
| 986 | |
| 987 | static int adev_get_master_volume(struct audio_hw_device *dev, |
| 988 | float *volume) |
| 989 | { |
| 990 | return -ENOSYS; |
| 991 | } |
| 992 | |
| 993 | static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) |
| 994 | { |
| 995 | return -ENOSYS; |
| 996 | } |
| 997 | |
| 998 | static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) |
| 999 | { |
| 1000 | return -ENOSYS; |
| 1001 | } |
| 1002 | |
| 1003 | static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) |
| 1004 | { |
| 1005 | struct audio_device *adev = (struct audio_device *)dev; |
| 1006 | pthread_mutex_lock(&adev->lock); |
| 1007 | if (adev->mode != mode) { |
| 1008 | ALOGD("%s mode %d\n", __func__, mode); |
| 1009 | adev->mode = mode; |
| 1010 | } |
| 1011 | pthread_mutex_unlock(&adev->lock); |
| 1012 | return 0; |
| 1013 | } |
| 1014 | |
| 1015 | static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) |
| 1016 | { |
| 1017 | int ret = 0; |
| 1018 | |
| 1019 | return ret; |
| 1020 | } |
| 1021 | |
| 1022 | static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) |
| 1023 | { |
| 1024 | return 0; |
| 1025 | } |
| 1026 | |
| 1027 | static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, |
| 1028 | const struct audio_config *config) |
| 1029 | { |
| 1030 | int channel_count = popcount(config->channel_mask); |
| 1031 | |
| 1032 | return get_input_buffer_size(config->sample_rate, config->format, channel_count); |
| 1033 | } |
| 1034 | |
| 1035 | static int adev_open_input_stream(struct audio_hw_device *dev, |
| 1036 | audio_io_handle_t handle, |
| 1037 | audio_devices_t devices, |
| 1038 | struct audio_config *config, |
| 1039 | struct audio_stream_in **stream_in) |
| 1040 | { |
| 1041 | struct audio_device *adev = (struct audio_device *)dev; |
| 1042 | struct stream_in *in; |
| 1043 | int ret = 0, buffer_size, frame_size; |
| 1044 | int channel_count = popcount(config->channel_mask); |
| 1045 | |
| 1046 | ALOGV("%s: enter", __func__); |
| 1047 | *stream_in = NULL; |
| 1048 | if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) |
| 1049 | return -EINVAL; |
| 1050 | |
| 1051 | in = (struct stream_in *)calloc(1, sizeof(struct stream_in)); |
| 1052 | |
| 1053 | in->stream.common.get_sample_rate = in_get_sample_rate; |
| 1054 | in->stream.common.set_sample_rate = in_set_sample_rate; |
| 1055 | in->stream.common.get_buffer_size = in_get_buffer_size; |
| 1056 | in->stream.common.get_channels = in_get_channels; |
| 1057 | in->stream.common.get_format = in_get_format; |
| 1058 | in->stream.common.set_format = in_set_format; |
| 1059 | in->stream.common.standby = in_standby; |
| 1060 | in->stream.common.dump = in_dump; |
| 1061 | in->stream.common.set_parameters = in_set_parameters; |
| 1062 | in->stream.common.get_parameters = in_get_parameters; |
| 1063 | in->stream.common.add_audio_effect = in_add_audio_effect; |
| 1064 | in->stream.common.remove_audio_effect = in_remove_audio_effect; |
| 1065 | in->stream.set_gain = in_set_gain; |
| 1066 | in->stream.read = in_read; |
| 1067 | in->stream.get_input_frames_lost = in_get_input_frames_lost; |
| 1068 | |
| 1069 | in->device = devices; |
| 1070 | in->source = AUDIO_SOURCE_DEFAULT; |
| 1071 | in->dev = adev; |
| 1072 | in->standby = 1; |
| 1073 | in->channel_mask = config->channel_mask; |
| 1074 | |
| 1075 | /* Update config params with the requested sample rate and channels */ |
| 1076 | in->usecase = USECASE_AUDIO_RECORD; |
| 1077 | in->config = pcm_config_audio_capture; |
| 1078 | in->config.rate = config->sample_rate; |
| 1079 | in->format = config->format; |
| 1080 | |
| 1081 | { |
| 1082 | in->config.channels = channel_count; |
| 1083 | frame_size = audio_stream_frame_size((struct audio_stream *)in); |
| 1084 | buffer_size = get_input_buffer_size(config->sample_rate, |
| 1085 | config->format, |
| 1086 | channel_count); |
| 1087 | in->config.period_size = buffer_size / frame_size; |
| 1088 | } |
| 1089 | |
| 1090 | *stream_in = &in->stream; |
| 1091 | ALOGV("%s: exit", __func__); |
| 1092 | return ret; |
| 1093 | |
| 1094 | err_open: |
| 1095 | free(in); |
| 1096 | *stream_in = NULL; |
| 1097 | return ret; |
| 1098 | } |
| 1099 | |
| 1100 | static void adev_close_input_stream(struct audio_hw_device *dev, |
| 1101 | struct audio_stream_in *stream) |
| 1102 | { |
| 1103 | int ret; |
| 1104 | struct stream_in *in = (struct stream_in *)stream; |
| 1105 | ALOGV("%s", __func__); |
| 1106 | |
| 1107 | in_standby(&stream->common); |
| 1108 | |
| 1109 | free(stream); |
| 1110 | |
| 1111 | return; |
| 1112 | } |
| 1113 | |
| 1114 | static int adev_dump(const audio_hw_device_t *device, int fd) |
| 1115 | { |
| 1116 | return 0; |
| 1117 | } |
| 1118 | |
| 1119 | static int adev_close(hw_device_t *device) |
| 1120 | { |
| 1121 | struct audio_device *adev = (struct audio_device *)device; |
| 1122 | |
| 1123 | if (!adev) |
| 1124 | return 0; |
| 1125 | |
| 1126 | pthread_mutex_lock(&adev_init_lock); |
| 1127 | |
| 1128 | if ((--audio_device_ref_count) == 0) { |
| 1129 | audio_route_free(adev->audio_route); |
| 1130 | free(adev->snd_dev_ref_cnt); |
| 1131 | platform_deinit(adev->platform); |
| 1132 | free(device); |
| 1133 | adev = NULL; |
| 1134 | } |
| 1135 | pthread_mutex_unlock(&adev_init_lock); |
| 1136 | return 0; |
| 1137 | } |
| 1138 | |
| 1139 | static int adev_open(const hw_module_t *module, const char *name, |
| 1140 | hw_device_t **device) |
| 1141 | { |
| 1142 | int i, ret; |
| 1143 | |
| 1144 | ALOGD("%s: enter", __func__); |
| 1145 | if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL; |
| 1146 | |
| 1147 | pthread_mutex_lock(&adev_init_lock); |
| 1148 | if (audio_device_ref_count != 0){ |
| 1149 | *device = &adev->device.common; |
| 1150 | audio_device_ref_count++; |
| 1151 | ALOGD("%s: returning existing instance of adev", __func__); |
| 1152 | ALOGD("%s: exit", __func__); |
| 1153 | pthread_mutex_unlock(&adev_init_lock); |
| 1154 | return 0; |
| 1155 | } |
| 1156 | |
| 1157 | adev = calloc(1, sizeof(struct audio_device)); |
| 1158 | |
| 1159 | adev->device.common.tag = HARDWARE_DEVICE_TAG; |
| 1160 | adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; |
| 1161 | adev->device.common.module = (struct hw_module_t *)module; |
| 1162 | adev->device.common.close = adev_close; |
| 1163 | |
| 1164 | adev->device.init_check = adev_init_check; |
| 1165 | adev->device.set_voice_volume = adev_set_voice_volume; |
| 1166 | adev->device.set_master_volume = adev_set_master_volume; |
| 1167 | adev->device.get_master_volume = adev_get_master_volume; |
| 1168 | adev->device.set_master_mute = adev_set_master_mute; |
| 1169 | adev->device.get_master_mute = adev_get_master_mute; |
| 1170 | adev->device.set_mode = adev_set_mode; |
| 1171 | adev->device.set_mic_mute = adev_set_mic_mute; |
| 1172 | adev->device.get_mic_mute = adev_get_mic_mute; |
| 1173 | adev->device.set_parameters = adev_set_parameters; |
| 1174 | adev->device.get_parameters = adev_get_parameters; |
| 1175 | adev->device.get_input_buffer_size = adev_get_input_buffer_size; |
| 1176 | adev->device.open_output_stream = adev_open_output_stream; |
| 1177 | adev->device.close_output_stream = adev_close_output_stream; |
| 1178 | adev->device.open_input_stream = adev_open_input_stream; |
| 1179 | adev->device.close_input_stream = adev_close_input_stream; |
| 1180 | adev->device.dump = adev_dump; |
| 1181 | |
| 1182 | /* Set the default route before the PCM stream is opened */ |
| 1183 | adev->mode = AUDIO_MODE_NORMAL; |
| 1184 | adev->active_input = NULL; |
| 1185 | adev->primary_output = NULL; |
| 1186 | adev->out_device = AUDIO_DEVICE_NONE; |
| 1187 | adev->bluetooth_nrec = true; |
| 1188 | adev->acdb_settings = TTY_MODE_OFF; |
| 1189 | /* adev->cur_hdmi_channels = 0; by calloc() */ |
| 1190 | adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int)); |
| 1191 | list_init(&adev->usecase_list); |
| 1192 | |
| 1193 | /* Loads platform specific libraries dynamically */ |
| 1194 | adev->platform = platform_init(adev); |
| 1195 | if (!adev->platform) { |
| 1196 | free(adev->snd_dev_ref_cnt); |
| 1197 | free(adev); |
| 1198 | ALOGE("%s: Failed to init platform data, aborting.", __func__); |
| 1199 | *device = NULL; |
| 1200 | return -EINVAL; |
| 1201 | } |
| 1202 | |
| 1203 | if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) { |
| 1204 | adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW); |
| 1205 | if (adev->visualizer_lib == NULL) { |
| 1206 | ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH); |
| 1207 | } else { |
| 1208 | ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH); |
| 1209 | adev->visualizer_start_output = |
| 1210 | (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib, |
| 1211 | "visualizer_hal_start_output"); |
| 1212 | adev->visualizer_stop_output = |
| 1213 | (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib, |
| 1214 | "visualizer_hal_stop_output"); |
| 1215 | } |
| 1216 | } |
| 1217 | *device = &adev->device.common; |
| 1218 | |
| 1219 | audio_device_ref_count++; |
| 1220 | pthread_mutex_unlock(&adev_init_lock); |
| 1221 | |
| 1222 | ALOGV("%s: exit", __func__); |
| 1223 | return 0; |
| 1224 | } |
| 1225 | |
| 1226 | static struct hw_module_methods_t hal_module_methods = { |
| 1227 | .open = adev_open, |
| 1228 | }; |
| 1229 | |
| 1230 | struct audio_module HAL_MODULE_INFO_SYM = { |
| 1231 | .common = { |
| 1232 | .tag = HARDWARE_MODULE_TAG, |
| 1233 | .module_api_version = AUDIO_MODULE_API_VERSION_0_1, |
| 1234 | .hal_api_version = HARDWARE_HAL_API_VERSION, |
| 1235 | .id = AUDIO_HARDWARE_MODULE_ID, |
| 1236 | .name = "MPQ Audio HAL", |
| 1237 | .author = "The Linux Foundation", |
| 1238 | .methods = &hal_module_methods, |
| 1239 | }, |
| 1240 | }; |