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/MonoPipe.h> 35#include <media/nbaio/MonoPipeReader.h> 36#include <media/AudioBufferProvider.h> 37 38#include <utils/String8.h> 39#include <media/AudioParameter.h> 40 41extern "C" { 42 43namespace android { 44 45#define MAX_PIPE_DEPTH_IN_FRAMES (1024*8) 46// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to 47// the duration of a record buffer at the current record sample rate (of the device, not of 48// the recording itself). Here we have: 49// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms 50#define MAX_READ_ATTEMPTS 3 51#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty 52#define DEFAULT_RATE_HZ 48000 // default sample rate 53 54struct submix_config { 55 audio_format_t format; 56 audio_channel_mask_t channel_mask; 57 unsigned int rate; // sample rate for the device 58 unsigned int period_size; // size of the audio pipe is period_size * period_count in frames 59 unsigned int period_count; 60}; 61 62struct submix_audio_device { 63 struct audio_hw_device device; 64 bool output_standby; 65 bool input_standby; 66 submix_config config; 67 // Pipe variables: they handle the ring buffer that "pipes" audio: 68 // - from the submix virtual audio output == what needs to be played 69 // remotely, seen as an output for AudioFlinger 70 // - to the virtual audio source == what is captured by the component 71 // which "records" the submix / virtual audio source, and handles it as needed. 72 // A usecase example is one where the component capturing the audio is then sending it over 73 // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a 74 // TV with Wifi Display capabilities), or to a wireless audio player. 75 sp<MonoPipe> rsxSink; 76 sp<MonoPipeReader> rsxSource; 77 78 // device lock, also used to protect access to the audio pipe 79 pthread_mutex_t lock; 80}; 81 82struct submix_stream_out { 83 struct audio_stream_out stream; 84 struct submix_audio_device *dev; 85}; 86 87struct submix_stream_in { 88 struct audio_stream_in stream; 89 struct submix_audio_device *dev; 90 bool output_standby; // output standby state as seen from record thread 91 92 // wall clock when recording starts 93 struct timespec record_start_time; 94 // how many frames have been requested to be read 95 int64_t read_counter_frames; 96}; 97 98 99/* audio HAL functions */ 100 101static uint32_t out_get_sample_rate(const struct audio_stream *stream) 102{ 103 const struct submix_stream_out *out = 104 reinterpret_cast<const struct submix_stream_out *>(stream); 105 uint32_t out_rate = out->dev->config.rate; 106 //ALOGV("out_get_sample_rate() returns %u", out_rate); 107 return out_rate; 108} 109 110static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 111{ 112 if ((rate != 44100) && (rate != 48000)) { 113 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 114 return -ENOSYS; 115 } 116 struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); 117 //ALOGV("out_set_sample_rate(rate=%u)", rate); 118 out->dev->config.rate = rate; 119 return 0; 120} 121 122static size_t out_get_buffer_size(const struct audio_stream *stream) 123{ 124 const struct submix_stream_out *out = 125 reinterpret_cast<const struct submix_stream_out *>(stream); 126 const struct submix_config& config_out = out->dev->config; 127 size_t buffer_size = config_out.period_size * popcount(config_out.channel_mask) 128 * sizeof(int16_t); // only PCM 16bit 129 //ALOGV("out_get_buffer_size() returns %u, period size=%u", 130 // buffer_size, config_out.period_size); 131 return buffer_size; 132} 133 134static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 135{ 136 const struct submix_stream_out *out = 137 reinterpret_cast<const struct submix_stream_out *>(stream); 138 uint32_t channels = out->dev->config.channel_mask; 139 //ALOGV("out_get_channels() returns %08x", channels); 140 return channels; 141} 142 143static audio_format_t out_get_format(const struct audio_stream *stream) 144{ 145 return AUDIO_FORMAT_PCM_16_BIT; 146} 147 148static int out_set_format(struct audio_stream *stream, audio_format_t format) 149{ 150 if (format != AUDIO_FORMAT_PCM_16_BIT) { 151 return -ENOSYS; 152 } else { 153 return 0; 154 } 155} 156 157static int out_standby(struct audio_stream *stream) 158{ 159 ALOGI("out_standby()"); 160 161 const struct submix_stream_out *out = reinterpret_cast<const struct submix_stream_out *>(stream); 162 163 pthread_mutex_lock(&out->dev->lock); 164 165 out->dev->output_standby = true; 166 167 pthread_mutex_unlock(&out->dev->lock); 168 169 return 0; 170} 171 172static int out_dump(const struct audio_stream *stream, int fd) 173{ 174 return 0; 175} 176 177static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 178{ 179 int exiting = -1; 180 AudioParameter parms = AudioParameter(String8(kvpairs)); 181 // FIXME this is using hard-coded strings but in the future, this functionality will be 182 // converted to use audio HAL extensions required to support tunneling 183 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { 184 const struct submix_stream_out *out = 185 reinterpret_cast<const struct submix_stream_out *>(stream); 186 187 pthread_mutex_lock(&out->dev->lock); 188 189 MonoPipe* sink = out->dev->rsxSink.get(); 190 if (sink != NULL) { 191 sink->incStrong(out); 192 } else { 193 pthread_mutex_unlock(&out->dev->lock); 194 return 0; 195 } 196 197 ALOGI("shutdown"); 198 sink->shutdown(true); 199 200 sink->decStrong(out); 201 202 pthread_mutex_unlock(&out->dev->lock); 203 } 204 205 return 0; 206} 207 208static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 209{ 210 return strdup(""); 211} 212 213static uint32_t out_get_latency(const struct audio_stream_out *stream) 214{ 215 const struct submix_stream_out *out = 216 reinterpret_cast<const struct submix_stream_out *>(stream); 217 const struct submix_config * config_out = &(out->dev->config); 218 uint32_t latency = (MAX_PIPE_DEPTH_IN_FRAMES * 1000) / config_out->rate; 219 ALOGV("out_get_latency() returns %u", latency); 220 return latency; 221} 222 223static int out_set_volume(struct audio_stream_out *stream, float left, 224 float right) 225{ 226 return -ENOSYS; 227} 228 229static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 230 size_t bytes) 231{ 232 //ALOGV("out_write(bytes=%d)", bytes); 233 ssize_t written_frames = 0; 234 struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream); 235 236 const size_t frame_size = audio_stream_frame_size(&stream->common); 237 const size_t frames = bytes / frame_size; 238 239 pthread_mutex_lock(&out->dev->lock); 240 241 out->dev->output_standby = false; 242 243 MonoPipe* sink = out->dev->rsxSink.get(); 244 if (sink != NULL) { 245 if (sink->isShutdown()) { 246 pthread_mutex_unlock(&out->dev->lock); 247 // the pipe has already been shutdown, this buffer will be lost but we must 248 // simulate timing so we don't drain the output faster than realtime 249 usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); 250 return bytes; 251 } 252 sink->incStrong(buffer); 253 } else { 254 pthread_mutex_unlock(&out->dev->lock); 255 ALOGE("out_write without a pipe!"); 256 ALOG_ASSERT("out_write without a pipe!"); 257 return 0; 258 } 259 260 pthread_mutex_unlock(&out->dev->lock); 261 262 written_frames = sink->write(buffer, frames); 263 if (written_frames < 0) { 264 if (written_frames == (ssize_t)NEGOTIATE) { 265 ALOGE("out_write() write to pipe returned NEGOTIATE"); 266 267 pthread_mutex_lock(&out->dev->lock); 268 sink->decStrong(buffer); 269 pthread_mutex_unlock(&out->dev->lock); 270 271 written_frames = 0; 272 return 0; 273 } else { 274 // write() returned UNDERRUN or WOULD_BLOCK, retry 275 ALOGE("out_write() write to pipe returned unexpected %16lx", written_frames); 276 written_frames = sink->write(buffer, frames); 277 } 278 } 279 280 pthread_mutex_lock(&out->dev->lock); 281 282 sink->decStrong(buffer); 283 284 pthread_mutex_unlock(&out->dev->lock); 285 286 if (written_frames < 0) { 287 ALOGE("out_write() failed writing to pipe with %16lx", written_frames); 288 return 0; 289 } else { 290 ALOGV("out_write() wrote %lu bytes)", written_frames * frame_size); 291 return written_frames * frame_size; 292 } 293} 294 295static int out_get_render_position(const struct audio_stream_out *stream, 296 uint32_t *dsp_frames) 297{ 298 return -EINVAL; 299} 300 301static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 302{ 303 return 0; 304} 305 306static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 307{ 308 return 0; 309} 310 311static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 312 int64_t *timestamp) 313{ 314 return -EINVAL; 315} 316 317/** audio_stream_in implementation **/ 318static uint32_t in_get_sample_rate(const struct audio_stream *stream) 319{ 320 const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); 321 //ALOGV("in_get_sample_rate() returns %u", in->dev->config.rate); 322 return in->dev->config.rate; 323} 324 325static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 326{ 327 return -ENOSYS; 328} 329 330static size_t in_get_buffer_size(const struct audio_stream *stream) 331{ 332 const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); 333 ALOGV("in_get_buffer_size() returns %u", 334 in->dev->config.period_size * audio_stream_frame_size(stream)); 335 return in->dev->config.period_size * audio_stream_frame_size(stream); 336} 337 338static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 339{ 340 return AUDIO_CHANNEL_IN_STEREO; 341} 342 343static audio_format_t in_get_format(const struct audio_stream *stream) 344{ 345 return AUDIO_FORMAT_PCM_16_BIT; 346} 347 348static int in_set_format(struct audio_stream *stream, audio_format_t format) 349{ 350 if (format != AUDIO_FORMAT_PCM_16_BIT) { 351 return -ENOSYS; 352 } else { 353 return 0; 354 } 355} 356 357static int in_standby(struct audio_stream *stream) 358{ 359 ALOGI("in_standby()"); 360 const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream); 361 362 pthread_mutex_lock(&in->dev->lock); 363 364 in->dev->input_standby = true; 365 366 pthread_mutex_unlock(&in->dev->lock); 367 368 return 0; 369} 370 371static int in_dump(const struct audio_stream *stream, int fd) 372{ 373 return 0; 374} 375 376static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 377{ 378 return 0; 379} 380 381static char * in_get_parameters(const struct audio_stream *stream, 382 const char *keys) 383{ 384 return strdup(""); 385} 386 387static int in_set_gain(struct audio_stream_in *stream, float gain) 388{ 389 return 0; 390} 391 392static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 393 size_t bytes) 394{ 395 //ALOGV("in_read bytes=%u", bytes); 396 ssize_t frames_read = -1977; 397 struct submix_stream_in *in = reinterpret_cast<struct submix_stream_in *>(stream); 398 const size_t frame_size = audio_stream_frame_size(&stream->common); 399 const size_t frames_to_read = bytes / frame_size; 400 401 pthread_mutex_lock(&in->dev->lock); 402 403 const bool output_standby_transition = (in->output_standby != in->dev->output_standby); 404 in->output_standby = in->dev->output_standby; 405 406 if (in->dev->input_standby || output_standby_transition) { 407 in->dev->input_standby = false; 408 // keep track of when we exit input standby (== first read == start "real recording") 409 // or when we start recording silence, and reset projected time 410 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 411 if (rc == 0) { 412 in->read_counter_frames = 0; 413 } 414 } 415 416 in->read_counter_frames += frames_to_read; 417 418 MonoPipeReader* source = in->dev->rsxSource.get(); 419 if (source != NULL) { 420 source->incStrong(buffer); 421 } else { 422 ALOGE("no audio pipe yet we're trying to read!"); 423 pthread_mutex_unlock(&in->dev->lock); 424 usleep((bytes / frame_size) * 1000000 / in_get_sample_rate(&stream->common)); 425 memset(buffer, 0, bytes); 426 return bytes; 427 } 428 429 pthread_mutex_unlock(&in->dev->lock); 430 431 // read the data from the pipe (it's non blocking) 432 size_t remaining_frames = frames_to_read; 433 int attempts = 0; 434 char* buff = (char*)buffer; 435 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 436 attempts++; 437 frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS); 438 if (frames_read > 0) { 439 remaining_frames -= frames_read; 440 buff += frames_read * frame_size; 441 //ALOGV(" in_read (att=%d) got %ld frames, remaining=%u", 442 // attempts, frames_read, remaining_frames); 443 } else { 444 //ALOGE(" in_read read returned %ld", frames_read); 445 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 446 } 447 } 448 449 // done using the source 450 pthread_mutex_lock(&in->dev->lock); 451 452 source->decStrong(buffer); 453 454 pthread_mutex_unlock(&in->dev->lock); 455 456 if (remaining_frames > 0) { 457 ALOGV(" remaining_frames = %d", remaining_frames); 458 memset(((char*)buffer)+ bytes - (remaining_frames * frame_size), 0, 459 remaining_frames * frame_size); 460 } 461 462 // compute how much we need to sleep after reading the data by comparing the wall clock with 463 // the projected time at which we should return. 464 struct timespec time_after_read;// wall clock after reading from the pipe 465 struct timespec record_duration;// observed record duration 466 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 467 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 468 if (rc == 0) { 469 // for how long have we been recording? 470 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 471 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 472 if (record_duration.tv_nsec < 0) { 473 record_duration.tv_sec--; 474 record_duration.tv_nsec += 1000000000; 475 } 476 477 // read_counter_frames contains the number of frames that have been read since the beginning 478 // of recording (including this call): it's converted to usec and compared to how long we've 479 // been recording for, which gives us how long we must wait to sync the projected recording 480 // time, and the observed recording time 481 long projected_vs_observed_offset_us = 482 ((int64_t)(in->read_counter_frames 483 - (record_duration.tv_sec*sample_rate))) 484 * 1000000 / sample_rate 485 - (record_duration.tv_nsec / 1000); 486 487 ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 488 record_duration.tv_sec, record_duration.tv_nsec/1000000, 489 projected_vs_observed_offset_us); 490 if (projected_vs_observed_offset_us > 0) { 491 usleep(projected_vs_observed_offset_us); 492 } 493 } 494 495 496 ALOGV("in_read returns %d", bytes); 497 return bytes; 498 499} 500 501static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 502{ 503 return 0; 504} 505 506static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 507{ 508 return 0; 509} 510 511static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 512{ 513 return 0; 514} 515 516static int adev_open_output_stream(struct audio_hw_device *dev, 517 audio_io_handle_t handle, 518 audio_devices_t devices, 519 audio_output_flags_t flags, 520 struct audio_config *config, 521 struct audio_stream_out **stream_out) 522{ 523 ALOGV("adev_open_output_stream()"); 524 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 525 struct submix_stream_out *out; 526 int ret; 527 528 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 529 if (!out) { 530 ret = -ENOMEM; 531 goto err_open; 532 } 533 534 pthread_mutex_lock(&rsxadev->lock); 535 536 out->stream.common.get_sample_rate = out_get_sample_rate; 537 out->stream.common.set_sample_rate = out_set_sample_rate; 538 out->stream.common.get_buffer_size = out_get_buffer_size; 539 out->stream.common.get_channels = out_get_channels; 540 out->stream.common.get_format = out_get_format; 541 out->stream.common.set_format = out_set_format; 542 out->stream.common.standby = out_standby; 543 out->stream.common.dump = out_dump; 544 out->stream.common.set_parameters = out_set_parameters; 545 out->stream.common.get_parameters = out_get_parameters; 546 out->stream.common.add_audio_effect = out_add_audio_effect; 547 out->stream.common.remove_audio_effect = out_remove_audio_effect; 548 out->stream.get_latency = out_get_latency; 549 out->stream.set_volume = out_set_volume; 550 out->stream.write = out_write; 551 out->stream.get_render_position = out_get_render_position; 552 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 553 554 config->channel_mask = AUDIO_CHANNEL_OUT_STEREO; 555 rsxadev->config.channel_mask = config->channel_mask; 556 557 if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) { 558 config->sample_rate = DEFAULT_RATE_HZ; 559 } 560 rsxadev->config.rate = config->sample_rate; 561 562 config->format = AUDIO_FORMAT_PCM_16_BIT; 563 rsxadev->config.format = config->format; 564 565 rsxadev->config.period_size = 1024; 566 rsxadev->config.period_count = 4; 567 out->dev = rsxadev; 568 569 *stream_out = &out->stream; 570 571 // initialize pipe 572 { 573 ALOGV(" initializing pipe"); 574 const NBAIO_Format format = 575 config->sample_rate == 48000 ? Format_SR48_C2_I16 : Format_SR44_1_C2_I16; 576 const NBAIO_Format offers[1] = {format}; 577 size_t numCounterOffers = 0; 578 // creating a MonoPipe with optional blocking set to true. 579 MonoPipe* sink = new MonoPipe(MAX_PIPE_DEPTH_IN_FRAMES, format, true/*writeCanBlock*/); 580 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 581 ALOG_ASSERT(index == 0); 582 MonoPipeReader* source = new MonoPipeReader(sink); 583 numCounterOffers = 0; 584 index = source->negotiate(offers, 1, NULL, numCounterOffers); 585 ALOG_ASSERT(index == 0); 586 rsxadev->rsxSink = sink; 587 rsxadev->rsxSource = source; 588 } 589 590 pthread_mutex_unlock(&rsxadev->lock); 591 592 return 0; 593 594err_open: 595 *stream_out = NULL; 596 return ret; 597} 598 599static void adev_close_output_stream(struct audio_hw_device *dev, 600 struct audio_stream_out *stream) 601{ 602 ALOGV("adev_close_output_stream()"); 603 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 604 605 pthread_mutex_lock(&rsxadev->lock); 606 607 rsxadev->rsxSink.clear(); 608 rsxadev->rsxSource.clear(); 609 free(stream); 610 611 pthread_mutex_unlock(&rsxadev->lock); 612} 613 614static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 615{ 616 return -ENOSYS; 617} 618 619static char * adev_get_parameters(const struct audio_hw_device *dev, 620 const char *keys) 621{ 622 return strdup("");; 623} 624 625static int adev_init_check(const struct audio_hw_device *dev) 626{ 627 ALOGI("adev_init_check()"); 628 return 0; 629} 630 631static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 632{ 633 return -ENOSYS; 634} 635 636static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 637{ 638 return -ENOSYS; 639} 640 641static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 642{ 643 return -ENOSYS; 644} 645 646static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 647{ 648 return -ENOSYS; 649} 650 651static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 652{ 653 return -ENOSYS; 654} 655 656static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 657{ 658 return 0; 659} 660 661static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 662{ 663 return -ENOSYS; 664} 665 666static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 667{ 668 return -ENOSYS; 669} 670 671static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 672 const struct audio_config *config) 673{ 674 //### TODO correlate this with pipe parameters 675 return 4096; 676} 677 678static int adev_open_input_stream(struct audio_hw_device *dev, 679 audio_io_handle_t handle, 680 audio_devices_t devices, 681 struct audio_config *config, 682 struct audio_stream_in **stream_in) 683{ 684 ALOGI("adev_open_input_stream()"); 685 686 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 687 struct submix_stream_in *in; 688 int ret; 689 690 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 691 if (!in) { 692 ret = -ENOMEM; 693 goto err_open; 694 } 695 696 pthread_mutex_lock(&rsxadev->lock); 697 698 in->stream.common.get_sample_rate = in_get_sample_rate; 699 in->stream.common.set_sample_rate = in_set_sample_rate; 700 in->stream.common.get_buffer_size = in_get_buffer_size; 701 in->stream.common.get_channels = in_get_channels; 702 in->stream.common.get_format = in_get_format; 703 in->stream.common.set_format = in_set_format; 704 in->stream.common.standby = in_standby; 705 in->stream.common.dump = in_dump; 706 in->stream.common.set_parameters = in_set_parameters; 707 in->stream.common.get_parameters = in_get_parameters; 708 in->stream.common.add_audio_effect = in_add_audio_effect; 709 in->stream.common.remove_audio_effect = in_remove_audio_effect; 710 in->stream.set_gain = in_set_gain; 711 in->stream.read = in_read; 712 in->stream.get_input_frames_lost = in_get_input_frames_lost; 713 714 config->channel_mask = AUDIO_CHANNEL_IN_STEREO; 715 rsxadev->config.channel_mask = config->channel_mask; 716 717 if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) { 718 config->sample_rate = DEFAULT_RATE_HZ; 719 } 720 rsxadev->config.rate = config->sample_rate; 721 722 config->format = AUDIO_FORMAT_PCM_16_BIT; 723 rsxadev->config.format = config->format; 724 725 rsxadev->config.period_size = 1024; 726 rsxadev->config.period_count = 4; 727 728 *stream_in = &in->stream; 729 730 in->dev = rsxadev; 731 732 in->read_counter_frames = 0; 733 in->output_standby = rsxadev->output_standby; 734 735 pthread_mutex_unlock(&rsxadev->lock); 736 737 return 0; 738 739err_open: 740 *stream_in = NULL; 741 return ret; 742} 743 744static void adev_close_input_stream(struct audio_hw_device *dev, 745 struct audio_stream_in *stream) 746{ 747 ALOGV("adev_close_input_stream()"); 748 struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev; 749 750 pthread_mutex_lock(&rsxadev->lock); 751 752 MonoPipe* sink = rsxadev->rsxSink.get(); 753 if (sink != NULL) { 754 ALOGI("shutdown"); 755 sink->shutdown(true); 756 } 757 758 free(stream); 759 760 pthread_mutex_unlock(&rsxadev->lock); 761} 762 763static int adev_dump(const audio_hw_device_t *device, int fd) 764{ 765 return 0; 766} 767 768static int adev_close(hw_device_t *device) 769{ 770 ALOGI("adev_close()"); 771 free(device); 772 return 0; 773} 774 775static int adev_open(const hw_module_t* module, const char* name, 776 hw_device_t** device) 777{ 778 ALOGI("adev_open(name=%s)", name); 779 struct submix_audio_device *rsxadev; 780 781 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 782 return -EINVAL; 783 784 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 785 if (!rsxadev) 786 return -ENOMEM; 787 788 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 789 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 790 rsxadev->device.common.module = (struct hw_module_t *) module; 791 rsxadev->device.common.close = adev_close; 792 793 rsxadev->device.init_check = adev_init_check; 794 rsxadev->device.set_voice_volume = adev_set_voice_volume; 795 rsxadev->device.set_master_volume = adev_set_master_volume; 796 rsxadev->device.get_master_volume = adev_get_master_volume; 797 rsxadev->device.set_master_mute = adev_set_master_mute; 798 rsxadev->device.get_master_mute = adev_get_master_mute; 799 rsxadev->device.set_mode = adev_set_mode; 800 rsxadev->device.set_mic_mute = adev_set_mic_mute; 801 rsxadev->device.get_mic_mute = adev_get_mic_mute; 802 rsxadev->device.set_parameters = adev_set_parameters; 803 rsxadev->device.get_parameters = adev_get_parameters; 804 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 805 rsxadev->device.open_output_stream = adev_open_output_stream; 806 rsxadev->device.close_output_stream = adev_close_output_stream; 807 rsxadev->device.open_input_stream = adev_open_input_stream; 808 rsxadev->device.close_input_stream = adev_close_input_stream; 809 rsxadev->device.dump = adev_dump; 810 811 rsxadev->input_standby = true; 812 rsxadev->output_standby = true; 813 814 *device = &rsxadev->device.common; 815 816 return 0; 817} 818 819static struct hw_module_methods_t hal_module_methods = { 820 /* open */ adev_open, 821}; 822 823struct audio_module HAL_MODULE_INFO_SYM = { 824 /* common */ { 825 /* tag */ HARDWARE_MODULE_TAG, 826 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 827 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 828 /* id */ AUDIO_HARDWARE_MODULE_ID, 829 /* name */ "Wifi Display audio HAL", 830 /* author */ "The Android Open Source Project", 831 /* methods */ &hal_module_methods, 832 /* dso */ NULL, 833 /* reserved */ { 0 }, 834 }, 835}; 836 837} //namespace android 838 839} //extern "C" 840