audio_hw.cpp revision eec87706d21fc0ac4ad10ede86943770b2533564
1/* 2 * Copyright (C) 2012 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 "r_submix" 18#define LOG_NDEBUG 0 19 20#include <errno.h> 21#include <pthread.h> 22#include <stdint.h> 23#include <sys/time.h> 24#include <stdlib.h> 25 26#include <cutils/log.h> 27#include <cutils/str_parms.h> 28#include <cutils/properties.h> 29 30#include <hardware/hardware.h> 31#include <system/audio.h> 32#include <hardware/audio.h> 33 34//#include <media/nbaio/Pipe.h> 35//#include <media/nbaio/PipeReader.h> 36#include <media/nbaio/MonoPipe.h> 37#include <media/nbaio/MonoPipeReader.h> 38#include <media/AudioBufferProvider.h> 39 40extern "C" { 41 42namespace android { 43 44#define MAX_PIPE_DEPTH_IN_FRAMES (1024*8) 45// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to 46// the duration of a record buffer at the current record sample rate (of the device, not of 47// the recording itself). Here we have: 48// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms 49#define MAX_READ_ATTEMPTS 3 50#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty 51#define DEFAULT_RATE_HZ 48000 // default sample rate 52 53struct submix_config { 54 audio_format_t format; 55 audio_channel_mask_t channel_mask; 56 unsigned int rate; // sample rate for the device 57 unsigned int period_size; // size of the audio pipe is period_size * period_count in frames 58 unsigned int period_count; 59}; 60 61struct submix_audio_device { 62 struct audio_hw_device device; 63 bool output_standby; 64 bool input_standby; 65 submix_config config; 66 // Pipe variables: they handle the ring buffer that "pipes" audio: 67 // - from the submix virtual audio output == what needs to be played 68 // remotely, seen as an output for AudioFlinger 69 // - to the virtual audio source == what is captured by the component 70 // which "records" the submix / virtual audio source, and handles it as needed. 71 // A usecase example is one where the component capturing the audio is then sending it over 72 // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a 73 // TV with Wifi Display capabilities), or to a wireless audio player. 74 sp<MonoPipe> rsxSink; 75 sp<MonoPipeReader> rsxSource; 76 77 // device lock, also used to protect access to the audio pipe 78 pthread_mutex_t lock; 79}; 80 81struct submix_stream_out { 82 struct audio_stream_out stream; 83 struct submix_audio_device *dev; 84}; 85 86struct submix_stream_in { 87 struct audio_stream_in stream; 88 struct submix_audio_device *dev; 89 bool output_standby; // output standby state as seen from record thread 90 91 // wall clock when recording starts 92 struct timespec record_start_time; 93 // how many frames have been requested to be read 94 int64_t read_counter_frames; 95}; 96 97 98/* audio HAL functions */ 99 100static uint32_t out_get_sample_rate(const struct audio_stream *stream) 101{ 102 const struct submix_stream_out *out = 103 reinterpret_cast<const struct submix_stream_out *>(stream); 104 uint32_t out_rate = out->dev->config.rate; 105 //ALOGV("out_get_sample_rate() returns %u", out_rate); 106 return out_rate; 107} 108 109static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 110{ 111 if ((rate != 44100) && (rate != 48000)) { 112 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 113 return -ENOSYS; 114 } 115 struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); 116 //ALOGV("out_set_sample_rate(rate=%u)", rate); 117 out->dev->config.rate = rate; 118 return 0; 119} 120 121static size_t out_get_buffer_size(const struct audio_stream *stream) 122{ 123 const struct submix_stream_out *out = 124 reinterpret_cast<const struct submix_stream_out *>(stream); 125 const struct submix_config& config_out = out->dev->config; 126 size_t buffer_size = config_out.period_size * popcount(config_out.channel_mask) 127 * sizeof(int16_t); // only PCM 16bit 128 //ALOGV("out_get_buffer_size() returns %u, period size=%u", 129 // buffer_size, config_out.period_size); 130 return buffer_size; 131} 132 133static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 134{ 135 const struct submix_stream_out *out = 136 reinterpret_cast<const struct submix_stream_out *>(stream); 137 uint32_t channels = out->dev->config.channel_mask; 138 //ALOGV("out_get_channels() returns %08x", channels); 139 return channels; 140} 141 142static audio_format_t out_get_format(const struct audio_stream *stream) 143{ 144 return AUDIO_FORMAT_PCM_16_BIT; 145} 146 147static int out_set_format(struct audio_stream *stream, audio_format_t format) 148{ 149 if (format != AUDIO_FORMAT_PCM_16_BIT) { 150 return -ENOSYS; 151 } else { 152 return 0; 153 } 154} 155 156static int out_standby(struct audio_stream *stream) 157{ 158 ALOGI("out_standby()"); 159 160 const struct submix_stream_out *out = reinterpret_cast<const struct submix_stream_out *>(stream); 161 162 pthread_mutex_lock(&out->dev->lock); 163 164 out->dev->output_standby = true; 165 166 pthread_mutex_unlock(&out->dev->lock); 167 168 return 0; 169} 170 171static int out_dump(const struct audio_stream *stream, int fd) 172{ 173 return 0; 174} 175 176static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 177{ 178 return 0; 179} 180 181static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 182{ 183 return strdup(""); 184} 185 186static uint32_t out_get_latency(const struct audio_stream_out *stream) 187{ 188 const struct submix_stream_out *out = 189 reinterpret_cast<const struct submix_stream_out *>(stream); 190 const struct submix_config * config_out = &(out->dev->config); 191 uint32_t latency = (MAX_PIPE_DEPTH_IN_FRAMES * 1000) / config_out->rate; 192 ALOGV("out_get_latency() returns %u", latency); 193 return latency; 194} 195 196static int out_set_volume(struct audio_stream_out *stream, float left, 197 float right) 198{ 199 return -ENOSYS; 200} 201 202static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 203 size_t bytes) 204{ 205 //ALOGV("out_write(bytes=%d)", bytes); 206 ssize_t written_frames = 0; 207 struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); 208 209 pthread_mutex_lock(&out->dev->lock); 210 211 out->dev->output_standby = false; 212 213 MonoPipe* sink = out->dev->rsxSink.get(); 214 if (sink != NULL) { 215 out->dev->rsxSink->incStrong(buffer); 216 } else { 217 pthread_mutex_unlock(&out->dev->lock); 218 ALOGE("out_write without a pipe!"); 219 ALOG_ASSERT("out_write without a pipe!"); 220 return 0; 221 } 222 223 pthread_mutex_unlock(&out->dev->lock); 224 225 const size_t frame_size = audio_stream_frame_size(&stream->common); 226 const size_t frames = bytes / frame_size; 227 written_frames = sink->write(buffer, frames); 228 if (written_frames < 0) { 229 if (written_frames == (ssize_t)NEGOTIATE) { 230 ALOGE("out_write() write to pipe returned NEGOTIATE"); 231 232 pthread_mutex_lock(&out->dev->lock); 233 out->dev->rsxSink->decStrong(buffer); 234 pthread_mutex_unlock(&out->dev->lock); 235 236 written_frames = 0; 237 return 0; 238 } else { 239 // write() returned UNDERRUN or WOULD_BLOCK, retry 240 ALOGE("out_write() write to pipe returned unexpected %16lx", written_frames); 241 written_frames = sink->write(buffer, frames); 242 } 243 } 244 245 pthread_mutex_lock(&out->dev->lock); 246 247 out->dev->rsxSink->decStrong(buffer); 248 249 pthread_mutex_unlock(&out->dev->lock); 250 251 if (written_frames < 0) { 252 ALOGE("out_write() failed writing to pipe with %16lx", written_frames); 253 return 0; 254 } else { 255 ALOGV("out_write() wrote %lu bytes)", written_frames * frame_size); 256 return written_frames * frame_size; 257 } 258} 259 260static int out_get_render_position(const struct audio_stream_out *stream, 261 uint32_t *dsp_frames) 262{ 263 return -EINVAL; 264} 265 266static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 267{ 268 return 0; 269} 270 271static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 272{ 273 return 0; 274} 275 276static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 277 int64_t *timestamp) 278{ 279 return -EINVAL; 280} 281 282/** audio_stream_in implementation **/ 283static uint32_t in_get_sample_rate(const struct audio_stream *stream) 284{ 285 const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); 286 //ALOGV("in_get_sample_rate() returns %u", in->dev->config.rate); 287 return in->dev->config.rate; 288} 289 290static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 291{ 292 return -ENOSYS; 293} 294 295static size_t in_get_buffer_size(const struct audio_stream *stream) 296{ 297 const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); 298 ALOGV("in_get_buffer_size() returns %u", 299 in->dev->config.period_size * audio_stream_frame_size(stream)); 300 return in->dev->config.period_size * audio_stream_frame_size(stream); 301} 302 303static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 304{ 305 return AUDIO_CHANNEL_IN_STEREO; 306} 307 308static audio_format_t in_get_format(const struct audio_stream *stream) 309{ 310 return AUDIO_FORMAT_PCM_16_BIT; 311} 312 313static int in_set_format(struct audio_stream *stream, audio_format_t format) 314{ 315 if (format != AUDIO_FORMAT_PCM_16_BIT) { 316 return -ENOSYS; 317 } else { 318 return 0; 319 } 320} 321 322static int in_standby(struct audio_stream *stream) 323{ 324 ALOGI("in_standby()"); 325 const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); 326 327 pthread_mutex_lock(&in->dev->lock); 328 329 in->dev->input_standby = true; 330 331 pthread_mutex_unlock(&in->dev->lock); 332 333 return 0; 334} 335 336static int in_dump(const struct audio_stream *stream, int fd) 337{ 338 return 0; 339} 340 341static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 342{ 343 return 0; 344} 345 346static char * in_get_parameters(const struct audio_stream *stream, 347 const char *keys) 348{ 349 return strdup(""); 350} 351 352static int in_set_gain(struct audio_stream_in *stream, float gain) 353{ 354 return 0; 355} 356 357static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 358 size_t bytes) 359{ 360 //ALOGV("in_read bytes=%u", bytes); 361 ssize_t frames_read = -1977; 362 struct submix_stream_in *in = reinterpret_cast<struct submix_stream_in *>(stream); 363 const size_t frame_size = audio_stream_frame_size(&stream->common); 364 const size_t frames_to_read = bytes / frame_size; 365 366 pthread_mutex_lock(&in->dev->lock); 367 368 const bool output_standby_transition = (in->output_standby != in->dev->output_standby); 369 in->output_standby = in->dev->output_standby; 370 371 if (in->dev->input_standby || output_standby_transition) { 372 in->dev->input_standby = false; 373 // keep track of when we exit input standby (== first read == start "real recording") 374 // or when we start recording silence, and reset projected time 375 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 376 if (rc == 0) { 377 in->read_counter_frames = 0; 378 } 379 } 380 381 in->read_counter_frames += frames_to_read; 382 383 MonoPipeReader* source = in->dev->rsxSource.get(); 384 if (source != NULL) { 385 in->dev->rsxSource->incStrong(in); 386 } else { 387 ALOGE("no audio pipe yet we're trying to read!"); 388 pthread_mutex_unlock(&in->dev->lock); 389 usleep((bytes / frame_size) * 1000000 / in_get_sample_rate(&stream->common)); 390 memset(buffer, 0, bytes); 391 return bytes; 392 } 393 394 pthread_mutex_unlock(&in->dev->lock); 395 396 // read the data from the pipe (it's non blocking) 397 size_t remaining_frames = frames_to_read; 398 int attempts = 0; 399 char* buff = (char*)buffer; 400 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 401 attempts++; 402 frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS); 403 if (frames_read > 0) { 404 remaining_frames -= frames_read; 405 buff += frames_read * frame_size; 406 //ALOGV(" in_read (att=%d) got %ld frames, remaining=%u", 407 // attempts, frames_read, remaining_frames); 408 } else { 409 //ALOGE(" in_read read returned %ld", frames_read); 410 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 411 } 412 } 413 414 // done using the source 415 pthread_mutex_lock(&in->dev->lock); 416 417 in->dev->rsxSource->decStrong(in); 418 419 pthread_mutex_unlock(&in->dev->lock); 420 421 if (remaining_frames > 0) { 422 ALOGV(" remaining_frames = %d", remaining_frames); 423 memset(((char*)buffer)+ bytes - (remaining_frames * frame_size), 0, 424 remaining_frames * frame_size); 425 } 426 427 // compute how much we need to sleep after reading the data by comparing the wall clock with 428 // the projected time at which we should return. 429 struct timespec time_after_read;// wall clock after reading from the pipe 430 struct timespec record_duration;// observed record duration 431 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 432 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 433 if (rc == 0) { 434 // for how long have we been recording? 435 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 436 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 437 if (record_duration.tv_nsec < 0) { 438 record_duration.tv_sec--; 439 record_duration.tv_nsec += 1000000000; 440 } 441 442 // read_counter_frames contains the number of frames that have been read since the beginning 443 // of recording (including this call): it's converted to usec and compared to how long we've 444 // been recording for, which gives us how long we must wait to sync the projected recording 445 // time, and the observed recording time 446 long projected_vs_observed_offset_us = 447 ((int64_t)(in->read_counter_frames 448 - (record_duration.tv_sec*sample_rate))) 449 * 1000000 / sample_rate 450 - (record_duration.tv_nsec / 1000); 451 452 ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 453 record_duration.tv_sec, record_duration.tv_nsec/1000000, 454 projected_vs_observed_offset_us); 455 if (projected_vs_observed_offset_us > 0) { 456 usleep(projected_vs_observed_offset_us); 457 } 458 } 459 460 461 ALOGV("in_read returns %d", bytes); 462 return bytes; 463 464} 465 466static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 467{ 468 return 0; 469} 470 471static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 472{ 473 return 0; 474} 475 476static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 477{ 478 return 0; 479} 480 481static int adev_open_output_stream(struct audio_hw_device *dev, 482 audio_io_handle_t handle, 483 audio_devices_t devices, 484 audio_output_flags_t flags, 485 struct audio_config *config, 486 struct audio_stream_out **stream_out) 487{ 488 ALOGV("adev_open_output_stream()"); 489 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 490 struct submix_stream_out *out; 491 int ret; 492 493 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 494 if (!out) { 495 ret = -ENOMEM; 496 goto err_open; 497 } 498 499 pthread_mutex_lock(&rsxadev->lock); 500 501 out->stream.common.get_sample_rate = out_get_sample_rate; 502 out->stream.common.set_sample_rate = out_set_sample_rate; 503 out->stream.common.get_buffer_size = out_get_buffer_size; 504 out->stream.common.get_channels = out_get_channels; 505 out->stream.common.get_format = out_get_format; 506 out->stream.common.set_format = out_set_format; 507 out->stream.common.standby = out_standby; 508 out->stream.common.dump = out_dump; 509 out->stream.common.set_parameters = out_set_parameters; 510 out->stream.common.get_parameters = out_get_parameters; 511 out->stream.common.add_audio_effect = out_add_audio_effect; 512 out->stream.common.remove_audio_effect = out_remove_audio_effect; 513 out->stream.get_latency = out_get_latency; 514 out->stream.set_volume = out_set_volume; 515 out->stream.write = out_write; 516 out->stream.get_render_position = out_get_render_position; 517 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 518 519 config->channel_mask = AUDIO_CHANNEL_OUT_STEREO; 520 rsxadev->config.channel_mask = config->channel_mask; 521 522 if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) { 523 config->sample_rate = DEFAULT_RATE_HZ; 524 } 525 rsxadev->config.rate = config->sample_rate; 526 527 config->format = AUDIO_FORMAT_PCM_16_BIT; 528 rsxadev->config.format = config->format; 529 530 rsxadev->config.period_size = 1024; 531 rsxadev->config.period_count = 4; 532 out->dev = rsxadev; 533 534 *stream_out = &out->stream; 535 536 // initialize pipe 537 { 538 ALOGV(" initializing pipe"); 539 const NBAIO_Format format = 540 config->sample_rate == 48000 ? Format_SR48_C2_I16 : Format_SR44_1_C2_I16; 541 const NBAIO_Format offers[1] = {format}; 542 size_t numCounterOffers = 0; 543 // creating a MonoPipe with optional blocking set to true. 544 MonoPipe* sink = new MonoPipe(MAX_PIPE_DEPTH_IN_FRAMES, format, true/*writeCanBlock*/); 545 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 546 ALOG_ASSERT(index == 0); 547 MonoPipeReader* source = new MonoPipeReader(sink); 548 numCounterOffers = 0; 549 index = source->negotiate(offers, 1, NULL, numCounterOffers); 550 ALOG_ASSERT(index == 0); 551 rsxadev->rsxSink = sink; 552 rsxadev->rsxSource = source; 553 } 554 555 pthread_mutex_unlock(&rsxadev->lock); 556 557 return 0; 558 559err_open: 560 *stream_out = NULL; 561 return ret; 562} 563 564static void adev_close_output_stream(struct audio_hw_device *dev, 565 struct audio_stream_out *stream) 566{ 567 ALOGV("adev_close_output_stream()"); 568 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 569 570 pthread_mutex_lock(&rsxadev->lock); 571 572 rsxadev->rsxSink.clear(); 573 rsxadev->rsxSource.clear(); 574 free(stream); 575 576 pthread_mutex_unlock(&rsxadev->lock); 577} 578 579static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 580{ 581 return -ENOSYS; 582} 583 584static char * adev_get_parameters(const struct audio_hw_device *dev, 585 const char *keys) 586{ 587 return strdup("");; 588} 589 590static int adev_init_check(const struct audio_hw_device *dev) 591{ 592 ALOGI("adev_init_check()"); 593 return 0; 594} 595 596static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 597{ 598 return -ENOSYS; 599} 600 601static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 602{ 603 return -ENOSYS; 604} 605 606static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 607{ 608 return -ENOSYS; 609} 610 611static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 612{ 613 return -ENOSYS; 614} 615 616static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 617{ 618 return -ENOSYS; 619} 620 621static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 622{ 623 return 0; 624} 625 626static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 627{ 628 return -ENOSYS; 629} 630 631static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 632{ 633 return -ENOSYS; 634} 635 636static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 637 const struct audio_config *config) 638{ 639 //### TODO correlate this with pipe parameters 640 return 4096; 641} 642 643static int adev_open_input_stream(struct audio_hw_device *dev, 644 audio_io_handle_t handle, 645 audio_devices_t devices, 646 struct audio_config *config, 647 struct audio_stream_in **stream_in) 648{ 649 ALOGI("adev_open_input_stream()"); 650 651 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 652 struct submix_stream_in *in; 653 int ret; 654 655 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 656 if (!in) { 657 ret = -ENOMEM; 658 goto err_open; 659 } 660 661 pthread_mutex_lock(&rsxadev->lock); 662 663 in->stream.common.get_sample_rate = in_get_sample_rate; 664 in->stream.common.set_sample_rate = in_set_sample_rate; 665 in->stream.common.get_buffer_size = in_get_buffer_size; 666 in->stream.common.get_channels = in_get_channels; 667 in->stream.common.get_format = in_get_format; 668 in->stream.common.set_format = in_set_format; 669 in->stream.common.standby = in_standby; 670 in->stream.common.dump = in_dump; 671 in->stream.common.set_parameters = in_set_parameters; 672 in->stream.common.get_parameters = in_get_parameters; 673 in->stream.common.add_audio_effect = in_add_audio_effect; 674 in->stream.common.remove_audio_effect = in_remove_audio_effect; 675 in->stream.set_gain = in_set_gain; 676 in->stream.read = in_read; 677 in->stream.get_input_frames_lost = in_get_input_frames_lost; 678 679 config->channel_mask = AUDIO_CHANNEL_IN_STEREO; 680 rsxadev->config.channel_mask = config->channel_mask; 681 682 if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) { 683 config->sample_rate = DEFAULT_RATE_HZ; 684 } 685 rsxadev->config.rate = config->sample_rate; 686 687 config->format = AUDIO_FORMAT_PCM_16_BIT; 688 rsxadev->config.format = config->format; 689 690 rsxadev->config.period_size = 1024; 691 rsxadev->config.period_count = 4; 692 693 *stream_in = &in->stream; 694 695 in->dev = rsxadev; 696 697 in->read_counter_frames = 0; 698 in->output_standby = rsxadev->output_standby; 699 700 pthread_mutex_unlock(&rsxadev->lock); 701 702 return 0; 703 704err_open: 705 *stream_in = NULL; 706 return ret; 707} 708 709static void adev_close_input_stream(struct audio_hw_device *dev, 710 struct audio_stream_in *stream) 711{ 712 ALOGV("adev_close_input_stream()"); 713 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 714 715 pthread_mutex_lock(&rsxadev->lock); 716 717 free(stream); 718 719 pthread_mutex_unlock(&rsxadev->lock); 720} 721 722static int adev_dump(const audio_hw_device_t *device, int fd) 723{ 724 return 0; 725} 726 727static int adev_close(hw_device_t *device) 728{ 729 ALOGI("adev_close()"); 730 free(device); 731 return 0; 732} 733 734static int adev_open(const hw_module_t* module, const char* name, 735 hw_device_t** device) 736{ 737 ALOGI("adev_open(name=%s)", name); 738 struct submix_audio_device *rsxadev; 739 740 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 741 return -EINVAL; 742 743 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 744 if (!rsxadev) 745 return -ENOMEM; 746 747 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 748 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 749 rsxadev->device.common.module = (struct hw_module_t *) module; 750 rsxadev->device.common.close = adev_close; 751 752 rsxadev->device.init_check = adev_init_check; 753 rsxadev->device.set_voice_volume = adev_set_voice_volume; 754 rsxadev->device.set_master_volume = adev_set_master_volume; 755 rsxadev->device.get_master_volume = adev_get_master_volume; 756 rsxadev->device.set_master_mute = adev_set_master_mute; 757 rsxadev->device.get_master_mute = adev_get_master_mute; 758 rsxadev->device.set_mode = adev_set_mode; 759 rsxadev->device.set_mic_mute = adev_set_mic_mute; 760 rsxadev->device.get_mic_mute = adev_get_mic_mute; 761 rsxadev->device.set_parameters = adev_set_parameters; 762 rsxadev->device.get_parameters = adev_get_parameters; 763 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 764 rsxadev->device.open_output_stream = adev_open_output_stream; 765 rsxadev->device.close_output_stream = adev_close_output_stream; 766 rsxadev->device.open_input_stream = adev_open_input_stream; 767 rsxadev->device.close_input_stream = adev_close_input_stream; 768 rsxadev->device.dump = adev_dump; 769 770 rsxadev->input_standby = true; 771 rsxadev->output_standby = true; 772 773 *device = &rsxadev->device.common; 774 775 return 0; 776} 777 778static struct hw_module_methods_t hal_module_methods = { 779 /* open */ adev_open, 780}; 781 782struct audio_module HAL_MODULE_INFO_SYM = { 783 /* common */ { 784 /* tag */ HARDWARE_MODULE_TAG, 785 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 786 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 787 /* id */ AUDIO_HARDWARE_MODULE_ID, 788 /* name */ "Wifi Display audio HAL", 789 /* author */ "The Android Open Source Project", 790 /* methods */ &hal_module_methods, 791 /* dso */ NULL, 792 /* reserved */ { 0 }, 793 }, 794}; 795 796} //namespace android 797 798} //extern "C" 799