1 /* 2 * Copyright (C) 2011 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 #define LOG_TAG "audio_hw_default" 18 //#define LOG_NDEBUG 0 19 20 #include <errno.h> 21 #include <malloc.h> 22 #include <pthread.h> 23 #include <stdint.h> 24 #include <stdlib.h> 25 #include <string.h> 26 #include <time.h> 27 #include <unistd.h> 28 29 #include <log/log.h> 30 31 #include <hardware/audio.h> 32 #include <hardware/hardware.h> 33 #include <system/audio.h> 34 35 #define STUB_DEFAULT_SAMPLE_RATE 48000 36 #define STUB_DEFAULT_AUDIO_FORMAT AUDIO_FORMAT_PCM_16_BIT 37 38 #define STUB_INPUT_BUFFER_MILLISECONDS 20 39 #define STUB_INPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_IN_STEREO 40 41 #define STUB_OUTPUT_BUFFER_MILLISECONDS 10 42 #define STUB_OUTPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_OUT_STEREO 43 44 struct stub_audio_device { 45 struct audio_hw_device device; 46 }; 47 48 struct stub_stream_out { 49 struct audio_stream_out stream; 50 int64_t last_write_time_us; 51 uint32_t sample_rate; 52 audio_channel_mask_t channel_mask; 53 audio_format_t format; 54 size_t frame_count; 55 }; 56 57 struct stub_stream_in { 58 struct audio_stream_in stream; 59 int64_t last_read_time_us; 60 uint32_t sample_rate; 61 audio_channel_mask_t channel_mask; 62 audio_format_t format; 63 size_t frame_count; 64 }; 65 66 static uint32_t out_get_sample_rate(const struct audio_stream *stream) 67 { 68 const struct stub_stream_out *out = (const struct stub_stream_out *)stream; 69 70 ALOGV("out_get_sample_rate: %u", out->sample_rate); 71 return out->sample_rate; 72 } 73 74 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 75 { 76 struct stub_stream_out *out = (struct stub_stream_out *)stream; 77 78 ALOGV("out_set_sample_rate: %d", rate); 79 out->sample_rate = rate; 80 return 0; 81 } 82 83 static size_t out_get_buffer_size(const struct audio_stream *stream) 84 { 85 const struct stub_stream_out *out = (const struct stub_stream_out *)stream; 86 size_t buffer_size = out->frame_count * 87 audio_stream_out_frame_size(&out->stream); 88 89 ALOGV("out_get_buffer_size: %zu", buffer_size); 90 return buffer_size; 91 } 92 93 static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 94 { 95 const struct stub_stream_out *out = (const struct stub_stream_out *)stream; 96 97 ALOGV("out_get_channels: %x", out->channel_mask); 98 return out->channel_mask; 99 } 100 101 static audio_format_t out_get_format(const struct audio_stream *stream) 102 { 103 const struct stub_stream_out *out = (const struct stub_stream_out *)stream; 104 105 ALOGV("out_get_format: %d", out->format); 106 return out->format; 107 } 108 109 static int out_set_format(struct audio_stream *stream, audio_format_t format) 110 { 111 struct stub_stream_out *out = (struct stub_stream_out *)stream; 112 113 ALOGV("out_set_format: %d", format); 114 out->format = format; 115 return 0; 116 } 117 118 static int out_standby(struct audio_stream *stream) 119 { 120 ALOGV("out_standby"); 121 // out->last_write_time_us = 0; unnecessary as a stale write time has same effect 122 return 0; 123 } 124 125 static int out_dump(const struct audio_stream *stream, int fd) 126 { 127 ALOGV("out_dump"); 128 return 0; 129 } 130 131 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 132 { 133 ALOGV("out_set_parameters"); 134 return 0; 135 } 136 137 static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 138 { 139 ALOGV("out_get_parameters"); 140 return strdup(""); 141 } 142 143 static uint32_t out_get_latency(const struct audio_stream_out *stream) 144 { 145 ALOGV("out_get_latency"); 146 return STUB_OUTPUT_BUFFER_MILLISECONDS; 147 } 148 149 static int out_set_volume(struct audio_stream_out *stream, float left, 150 float right) 151 { 152 ALOGV("out_set_volume: Left:%f Right:%f", left, right); 153 return 0; 154 } 155 156 static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 157 size_t bytes) 158 { 159 ALOGV("out_write: bytes: %zu", bytes); 160 161 /* XXX: fake timing for audio output */ 162 struct stub_stream_out *out = (struct stub_stream_out *)stream; 163 struct timespec t = { .tv_sec = 0, .tv_nsec = 0 }; 164 clock_gettime(CLOCK_MONOTONIC, &t); 165 const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000; 166 const int64_t elapsed_time_since_last_write = now - out->last_write_time_us; 167 int64_t sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) / 168 out_get_sample_rate(&stream->common) - elapsed_time_since_last_write; 169 if (sleep_time > 0) { 170 usleep(sleep_time); 171 } else { 172 // we don't sleep when we exit standby (this is typical for a real alsa buffer). 173 sleep_time = 0; 174 } 175 out->last_write_time_us = now + sleep_time; 176 // last_write_time_us is an approximation of when the (simulated) alsa 177 // buffer is believed completely full. The usleep above waits for more space 178 // in the buffer, but by the end of the sleep the buffer is considered 179 // topped-off. 180 // 181 // On the subsequent out_write(), we measure the elapsed time spent in 182 // the mixer. This is subtracted from the sleep estimate based on frames, 183 // thereby accounting for drain in the alsa buffer during mixing. 184 // This is a crude approximation; we don't handle underruns precisely. 185 return bytes; 186 } 187 188 static int out_get_render_position(const struct audio_stream_out *stream, 189 uint32_t *dsp_frames) 190 { 191 *dsp_frames = 0; 192 ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames); 193 return -EINVAL; 194 } 195 196 static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 197 { 198 ALOGV("out_add_audio_effect: %p", effect); 199 return 0; 200 } 201 202 static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 203 { 204 ALOGV("out_remove_audio_effect: %p", effect); 205 return 0; 206 } 207 208 static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 209 int64_t *timestamp) 210 { 211 *timestamp = 0; 212 ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp)); 213 return -EINVAL; 214 } 215 216 /** audio_stream_in implementation **/ 217 static uint32_t in_get_sample_rate(const struct audio_stream *stream) 218 { 219 const struct stub_stream_in *in = (const struct stub_stream_in *)stream; 220 221 ALOGV("in_get_sample_rate: %u", in->sample_rate); 222 return in->sample_rate; 223 } 224 225 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 226 { 227 struct stub_stream_in *in = (struct stub_stream_in *)stream; 228 229 ALOGV("in_set_sample_rate: %u", rate); 230 in->sample_rate = rate; 231 return 0; 232 } 233 234 static size_t in_get_buffer_size(const struct audio_stream *stream) 235 { 236 const struct stub_stream_in *in = (const struct stub_stream_in *)stream; 237 size_t buffer_size = in->frame_count * 238 audio_stream_in_frame_size(&in->stream); 239 240 ALOGV("in_get_buffer_size: %zu", buffer_size); 241 return buffer_size; 242 } 243 244 static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 245 { 246 const struct stub_stream_in *in = (const struct stub_stream_in *)stream; 247 248 ALOGV("in_get_channels: %x", in->channel_mask); 249 return in->channel_mask; 250 } 251 252 static audio_format_t in_get_format(const struct audio_stream *stream) 253 { 254 const struct stub_stream_in *in = (const struct stub_stream_in *)stream; 255 256 ALOGV("in_get_format: %d", in->format); 257 return in->format; 258 } 259 260 static int in_set_format(struct audio_stream *stream, audio_format_t format) 261 { 262 struct stub_stream_in *in = (struct stub_stream_in *)stream; 263 264 ALOGV("in_set_format: %d", format); 265 in->format = format; 266 return 0; 267 } 268 269 static int in_standby(struct audio_stream *stream) 270 { 271 struct stub_stream_in *in = (struct stub_stream_in *)stream; 272 in->last_read_time_us = 0; 273 return 0; 274 } 275 276 static int in_dump(const struct audio_stream *stream, int fd) 277 { 278 return 0; 279 } 280 281 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 282 { 283 return 0; 284 } 285 286 static char * in_get_parameters(const struct audio_stream *stream, 287 const char *keys) 288 { 289 return strdup(""); 290 } 291 292 static int in_set_gain(struct audio_stream_in *stream, float gain) 293 { 294 return 0; 295 } 296 297 static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 298 size_t bytes) 299 { 300 ALOGV("in_read: bytes %zu", bytes); 301 302 /* XXX: fake timing for audio input */ 303 struct stub_stream_in *in = (struct stub_stream_in *)stream; 304 struct timespec t = { .tv_sec = 0, .tv_nsec = 0 }; 305 clock_gettime(CLOCK_MONOTONIC, &t); 306 const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000; 307 308 // we do a full sleep when exiting standby. 309 const bool standby = in->last_read_time_us == 0; 310 const int64_t elapsed_time_since_last_read = standby ? 311 0 : now - in->last_read_time_us; 312 int64_t sleep_time = bytes * 1000000LL / audio_stream_in_frame_size(stream) / 313 in_get_sample_rate(&stream->common) - elapsed_time_since_last_read; 314 if (sleep_time > 0) { 315 usleep(sleep_time); 316 } else { 317 sleep_time = 0; 318 } 319 in->last_read_time_us = now + sleep_time; 320 // last_read_time_us is an approximation of when the (simulated) alsa 321 // buffer is drained by the read, and is empty. 322 // 323 // On the subsequent in_read(), we measure the elapsed time spent in 324 // the recording thread. This is subtracted from the sleep estimate based on frames, 325 // thereby accounting for fill in the alsa buffer during the interim. 326 memset(buffer, 0, bytes); 327 return bytes; 328 } 329 330 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 331 { 332 return 0; 333 } 334 335 static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 336 { 337 return 0; 338 } 339 340 static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 341 { 342 return 0; 343 } 344 345 static size_t samples_per_milliseconds(size_t milliseconds, 346 uint32_t sample_rate, 347 size_t channel_count) 348 { 349 return milliseconds * sample_rate * channel_count / 1000; 350 } 351 352 static int adev_open_output_stream(struct audio_hw_device *dev, 353 audio_io_handle_t handle, 354 audio_devices_t devices, 355 audio_output_flags_t flags, 356 struct audio_config *config, 357 struct audio_stream_out **stream_out, 358 const char *address __unused) 359 { 360 ALOGV("adev_open_output_stream..."); 361 362 *stream_out = NULL; 363 struct stub_stream_out *out = 364 (struct stub_stream_out *)calloc(1, sizeof(struct stub_stream_out)); 365 if (!out) 366 return -ENOMEM; 367 368 out->stream.common.get_sample_rate = out_get_sample_rate; 369 out->stream.common.set_sample_rate = out_set_sample_rate; 370 out->stream.common.get_buffer_size = out_get_buffer_size; 371 out->stream.common.get_channels = out_get_channels; 372 out->stream.common.get_format = out_get_format; 373 out->stream.common.set_format = out_set_format; 374 out->stream.common.standby = out_standby; 375 out->stream.common.dump = out_dump; 376 out->stream.common.set_parameters = out_set_parameters; 377 out->stream.common.get_parameters = out_get_parameters; 378 out->stream.common.add_audio_effect = out_add_audio_effect; 379 out->stream.common.remove_audio_effect = out_remove_audio_effect; 380 out->stream.get_latency = out_get_latency; 381 out->stream.set_volume = out_set_volume; 382 out->stream.write = out_write; 383 out->stream.get_render_position = out_get_render_position; 384 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 385 out->sample_rate = config->sample_rate; 386 if (out->sample_rate == 0) 387 out->sample_rate = STUB_DEFAULT_SAMPLE_RATE; 388 out->channel_mask = config->channel_mask; 389 if (out->channel_mask == AUDIO_CHANNEL_NONE) 390 out->channel_mask = STUB_OUTPUT_DEFAULT_CHANNEL_MASK; 391 out->format = config->format; 392 if (out->format == AUDIO_FORMAT_DEFAULT) 393 out->format = STUB_DEFAULT_AUDIO_FORMAT; 394 out->frame_count = samples_per_milliseconds( 395 STUB_OUTPUT_BUFFER_MILLISECONDS, 396 out->sample_rate, 1); 397 398 ALOGV("adev_open_output_stream: sample_rate: %u, channels: %x, format: %d," 399 " frames: %zu", out->sample_rate, out->channel_mask, out->format, 400 out->frame_count); 401 *stream_out = &out->stream; 402 return 0; 403 } 404 405 static void adev_close_output_stream(struct audio_hw_device *dev, 406 struct audio_stream_out *stream) 407 { 408 ALOGV("adev_close_output_stream..."); 409 free(stream); 410 } 411 412 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 413 { 414 ALOGV("adev_set_parameters"); 415 return -ENOSYS; 416 } 417 418 static char * adev_get_parameters(const struct audio_hw_device *dev, 419 const char *keys) 420 { 421 ALOGV("adev_get_parameters"); 422 return strdup(""); 423 } 424 425 static int adev_init_check(const struct audio_hw_device *dev) 426 { 427 ALOGV("adev_init_check"); 428 return 0; 429 } 430 431 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 432 { 433 ALOGV("adev_set_voice_volume: %f", volume); 434 return -ENOSYS; 435 } 436 437 static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 438 { 439 ALOGV("adev_set_master_volume: %f", volume); 440 return -ENOSYS; 441 } 442 443 static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 444 { 445 ALOGV("adev_get_master_volume: %f", *volume); 446 return -ENOSYS; 447 } 448 449 static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 450 { 451 ALOGV("adev_set_master_mute: %d", muted); 452 return -ENOSYS; 453 } 454 455 static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 456 { 457 ALOGV("adev_get_master_mute: %d", *muted); 458 return -ENOSYS; 459 } 460 461 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 462 { 463 ALOGV("adev_set_mode: %d", mode); 464 return 0; 465 } 466 467 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 468 { 469 ALOGV("adev_set_mic_mute: %d",state); 470 return -ENOSYS; 471 } 472 473 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 474 { 475 ALOGV("adev_get_mic_mute"); 476 return -ENOSYS; 477 } 478 479 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 480 const struct audio_config *config) 481 { 482 size_t buffer_size = samples_per_milliseconds( 483 STUB_INPUT_BUFFER_MILLISECONDS, 484 config->sample_rate, 485 audio_channel_count_from_in_mask( 486 config->channel_mask)); 487 488 if (!audio_has_proportional_frames(config->format)) { 489 // Since the audio data is not proportional choose an arbitrary size for 490 // the buffer. 491 buffer_size *= 4; 492 } else { 493 buffer_size *= audio_bytes_per_sample(config->format); 494 } 495 ALOGV("adev_get_input_buffer_size: %zu", buffer_size); 496 return buffer_size; 497 } 498 499 static int adev_open_input_stream(struct audio_hw_device *dev, 500 audio_io_handle_t handle, 501 audio_devices_t devices, 502 struct audio_config *config, 503 struct audio_stream_in **stream_in, 504 audio_input_flags_t flags __unused, 505 const char *address __unused, 506 audio_source_t source __unused) 507 { 508 ALOGV("adev_open_input_stream..."); 509 510 *stream_in = NULL; 511 struct stub_stream_in *in = (struct stub_stream_in *)calloc(1, sizeof(struct stub_stream_in)); 512 if (!in) 513 return -ENOMEM; 514 515 in->stream.common.get_sample_rate = in_get_sample_rate; 516 in->stream.common.set_sample_rate = in_set_sample_rate; 517 in->stream.common.get_buffer_size = in_get_buffer_size; 518 in->stream.common.get_channels = in_get_channels; 519 in->stream.common.get_format = in_get_format; 520 in->stream.common.set_format = in_set_format; 521 in->stream.common.standby = in_standby; 522 in->stream.common.dump = in_dump; 523 in->stream.common.set_parameters = in_set_parameters; 524 in->stream.common.get_parameters = in_get_parameters; 525 in->stream.common.add_audio_effect = in_add_audio_effect; 526 in->stream.common.remove_audio_effect = in_remove_audio_effect; 527 in->stream.set_gain = in_set_gain; 528 in->stream.read = in_read; 529 in->stream.get_input_frames_lost = in_get_input_frames_lost; 530 in->sample_rate = config->sample_rate; 531 if (in->sample_rate == 0) 532 in->sample_rate = STUB_DEFAULT_SAMPLE_RATE; 533 in->channel_mask = config->channel_mask; 534 if (in->channel_mask == AUDIO_CHANNEL_NONE) 535 in->channel_mask = STUB_INPUT_DEFAULT_CHANNEL_MASK; 536 in->format = config->format; 537 if (in->format == AUDIO_FORMAT_DEFAULT) 538 in->format = STUB_DEFAULT_AUDIO_FORMAT; 539 in->frame_count = samples_per_milliseconds( 540 STUB_INPUT_BUFFER_MILLISECONDS, in->sample_rate, 1); 541 542 ALOGV("adev_open_input_stream: sample_rate: %u, channels: %x, format: %d," 543 "frames: %zu", in->sample_rate, in->channel_mask, in->format, 544 in->frame_count); 545 *stream_in = &in->stream; 546 return 0; 547 } 548 549 static void adev_close_input_stream(struct audio_hw_device *dev, 550 struct audio_stream_in *in) 551 { 552 ALOGV("adev_close_input_stream..."); 553 return; 554 } 555 556 static int adev_dump(const audio_hw_device_t *device, int fd) 557 { 558 ALOGV("adev_dump"); 559 return 0; 560 } 561 562 static int adev_close(hw_device_t *device) 563 { 564 ALOGV("adev_close"); 565 free(device); 566 return 0; 567 } 568 569 static int adev_open(const hw_module_t* module, const char* name, 570 hw_device_t** device) 571 { 572 ALOGV("adev_open: %s", name); 573 574 struct stub_audio_device *adev; 575 576 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 577 return -EINVAL; 578 579 adev = calloc(1, sizeof(struct stub_audio_device)); 580 if (!adev) 581 return -ENOMEM; 582 583 adev->device.common.tag = HARDWARE_DEVICE_TAG; 584 adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 585 adev->device.common.module = (struct hw_module_t *) module; 586 adev->device.common.close = adev_close; 587 588 adev->device.init_check = adev_init_check; 589 adev->device.set_voice_volume = adev_set_voice_volume; 590 adev->device.set_master_volume = adev_set_master_volume; 591 adev->device.get_master_volume = adev_get_master_volume; 592 adev->device.set_master_mute = adev_set_master_mute; 593 adev->device.get_master_mute = adev_get_master_mute; 594 adev->device.set_mode = adev_set_mode; 595 adev->device.set_mic_mute = adev_set_mic_mute; 596 adev->device.get_mic_mute = adev_get_mic_mute; 597 adev->device.set_parameters = adev_set_parameters; 598 adev->device.get_parameters = adev_get_parameters; 599 adev->device.get_input_buffer_size = adev_get_input_buffer_size; 600 adev->device.open_output_stream = adev_open_output_stream; 601 adev->device.close_output_stream = adev_close_output_stream; 602 adev->device.open_input_stream = adev_open_input_stream; 603 adev->device.close_input_stream = adev_close_input_stream; 604 adev->device.dump = adev_dump; 605 606 *device = &adev->device.common; 607 608 return 0; 609 } 610 611 static struct hw_module_methods_t hal_module_methods = { 612 .open = adev_open, 613 }; 614 615 struct audio_module HAL_MODULE_INFO_SYM = { 616 .common = { 617 .tag = HARDWARE_MODULE_TAG, 618 .module_api_version = AUDIO_MODULE_API_VERSION_0_1, 619 .hal_api_version = HARDWARE_HAL_API_VERSION, 620 .id = AUDIO_HARDWARE_MODULE_ID, 621 .name = "Default audio HW HAL", 622 .author = "The Android Open Source Project", 623 .methods = &hal_module_methods, 624 }, 625 }; 626
