audio_hw.cpp revision 02c2f7126c78334ffb20d95f68284f4ec5bdae02
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 <stdlib.h> 24#include <sys/param.h> 25#include <sys/time.h> 26#include <sys/limits.h> 27 28#include <cutils/log.h> 29#include <cutils/properties.h> 30#include <cutils/str_parms.h> 31 32#include <hardware/audio.h> 33#include <hardware/hardware.h> 34#include <system/audio.h> 35 36#include <media/AudioParameter.h> 37#include <media/AudioBufferProvider.h> 38#include <media/nbaio/MonoPipe.h> 39#include <media/nbaio/MonoPipeReader.h> 40 41#include <utils/String8.h> 42 43#define LOG_STREAMS_TO_FILES 0 44#if LOG_STREAMS_TO_FILES 45#include <fcntl.h> 46#include <stdio.h> 47#include <sys/stat.h> 48#endif // LOG_STREAMS_TO_FILES 49 50extern "C" { 51 52namespace android { 53 54// Set to 1 to enable extremely verbose logging in this module. 55#define SUBMIX_VERBOSE_LOGGING 0 56#if SUBMIX_VERBOSE_LOGGING 57#define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__) 58#define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__) 59#else 60#define SUBMIX_ALOGV(...) 61#define SUBMIX_ALOGE(...) 62#endif // SUBMIX_VERBOSE_LOGGING 63 64// NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe(). 65#define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*8) 66// Value used to divide the MonoPipe() buffer into segments that are written to the source and 67// read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer 68// the minimum latency is the MonoPipe buffer size divided by this value. 69#define DEFAULT_PIPE_PERIOD_COUNT 4 70// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to 71// the duration of a record buffer at the current record sample rate (of the device, not of 72// the recording itself). Here we have: 73// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms 74#define MAX_READ_ATTEMPTS 3 75#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty 76#define DEFAULT_SAMPLE_RATE_HZ 48000 // default sample rate 77// See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h. 78#define DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT 79// A legacy user of this device does not close the input stream when it shuts down, which 80// results in the application opening a new input stream before closing the old input stream 81// handle it was previously using. Setting this value to 1 allows multiple clients to open 82// multiple input streams from this device. If this option is enabled, each input stream returned 83// is *the same stream* which means that readers will race to read data from these streams. 84#define ENABLE_LEGACY_INPUT_OPEN 1 85// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled. 86#define ENABLE_CHANNEL_CONVERSION 1 87// Whether resampling is enabled. 88#define ENABLE_RESAMPLING 1 89#if LOG_STREAMS_TO_FILES 90// Folder to save stream log files to. 91#define LOG_STREAM_FOLDER "/data/misc/media" 92// Log filenames for input and output streams. 93#define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw" 94#define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw" 95// File permissions for stream log files. 96#define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH) 97#endif // LOG_STREAMS_TO_FILES 98 99// Common limits macros. 100#ifndef min 101#define min(a, b) ((a) < (b) ? (a) : (b)) 102#endif // min 103#ifndef max 104#define max(a, b) ((a) > (b) ? (a) : (b)) 105#endif // max 106 107// Set *result_variable_ptr to true if value_to_find is present in the array array_to_search, 108// otherwise set *result_variable_ptr to false. 109#define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \ 110 { \ 111 size_t i; \ 112 *(result_variable_ptr) = false; \ 113 for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \ 114 if ((value_to_find) == (array_to_search)[i]) { \ 115 *(result_variable_ptr) = true; \ 116 break; \ 117 } \ 118 } \ 119 } 120 121// Configuration of the submix pipe. 122struct submix_config { 123 // Channel mask field in this data structure is set to either input_channel_mask or 124 // output_channel_mask depending upon the last stream to be opened on this device. 125 struct audio_config common; 126 // Input stream and output stream channel masks. This is required since input and output 127 // channel bitfields are not equivalent. 128 audio_channel_mask_t input_channel_mask; 129 audio_channel_mask_t output_channel_mask; 130#if ENABLE_RESAMPLING 131 // Input stream and output stream sample rates. 132 uint32_t input_sample_rate; 133 uint32_t output_sample_rate; 134#endif // ENABLE_RESAMPLING 135 size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe. 136 size_t buffer_size_frames; // Size of the audio pipe in frames. 137 // Maximum number of frames buffered by the input and output streams. 138 size_t buffer_period_size_frames; 139}; 140 141struct submix_audio_device { 142 struct audio_hw_device device; 143 bool input_standby; 144 bool output_standby; 145 submix_config config; 146 // Pipe variables: they handle the ring buffer that "pipes" audio: 147 // - from the submix virtual audio output == what needs to be played 148 // remotely, seen as an output for AudioFlinger 149 // - to the virtual audio source == what is captured by the component 150 // which "records" the submix / virtual audio source, and handles it as needed. 151 // A usecase example is one where the component capturing the audio is then sending it over 152 // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a 153 // TV with Wifi Display capabilities), or to a wireless audio player. 154 sp<MonoPipe> rsxSink; 155 sp<MonoPipeReader> rsxSource; 156#if ENABLE_RESAMPLING 157 // Buffer used as temporary storage for resampled data prior to returning data to the output 158 // stream. 159 int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; 160#endif // ENABLE_RESAMPLING 161 162 // Pointers to the current input and output stream instances. rsxSink and rsxSource are 163 // destroyed if both and input and output streams are destroyed. 164 struct submix_stream_out *output; 165 struct submix_stream_in *input; 166 167 // Device lock, also used to protect access to submix_audio_device from the input and output 168 // streams. 169 pthread_mutex_t lock; 170}; 171 172struct submix_stream_out { 173 struct audio_stream_out stream; 174 struct submix_audio_device *dev; 175#if LOG_STREAMS_TO_FILES 176 int log_fd; 177#endif // LOG_STREAMS_TO_FILES 178}; 179 180struct submix_stream_in { 181 struct audio_stream_in stream; 182 struct submix_audio_device *dev; 183 bool output_standby; // output standby state as seen from record thread 184 185 // wall clock when recording starts 186 struct timespec record_start_time; 187 // how many frames have been requested to be read 188 int64_t read_counter_frames; 189 190#if ENABLE_LEGACY_INPUT_OPEN 191 // Number of references to this input stream. 192 volatile int32_t ref_count; 193#endif // ENABLE_LEGACY_INPUT_OPEN 194#if LOG_STREAMS_TO_FILES 195 int log_fd; 196#endif // LOG_STREAMS_TO_FILES 197}; 198 199// Determine whether the specified sample rate is supported by the submix module. 200static bool sample_rate_supported(const uint32_t sample_rate) 201{ 202 // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp. 203 static const unsigned int supported_sample_rates[] = { 204 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 205 }; 206 bool return_value; 207 SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value); 208 return return_value; 209} 210 211// Determine whether the specified sample rate is supported, if it is return the specified sample 212// rate, otherwise return the default sample rate for the submix module. 213static uint32_t get_supported_sample_rate(uint32_t sample_rate) 214{ 215 return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ; 216} 217 218// Determine whether the specified channel in mask is supported by the submix module. 219static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask) 220{ 221 // Set of channel in masks supported by Format_from_SR_C() 222 // frameworks/av/media/libnbaio/NAIO.cpp. 223 static const audio_channel_mask_t supported_channel_in_masks[] = { 224 AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, 225 }; 226 bool return_value; 227 SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value); 228 return return_value; 229} 230 231// Determine whether the specified channel in mask is supported, if it is return the specified 232// channel in mask, otherwise return the default channel in mask for the submix module. 233static audio_channel_mask_t get_supported_channel_in_mask( 234 const audio_channel_mask_t channel_in_mask) 235{ 236 return channel_in_mask_supported(channel_in_mask) ? channel_in_mask : 237 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO); 238} 239 240// Determine whether the specified channel out mask is supported by the submix module. 241static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask) 242{ 243 // Set of channel out masks supported by Format_from_SR_C() 244 // frameworks/av/media/libnbaio/NAIO.cpp. 245 static const audio_channel_mask_t supported_channel_out_masks[] = { 246 AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO, 247 }; 248 bool return_value; 249 SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value); 250 return return_value; 251} 252 253// Determine whether the specified channel out mask is supported, if it is return the specified 254// channel out mask, otherwise return the default channel out mask for the submix module. 255static audio_channel_mask_t get_supported_channel_out_mask( 256 const audio_channel_mask_t channel_out_mask) 257{ 258 return channel_out_mask_supported(channel_out_mask) ? channel_out_mask : 259 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO); 260} 261 262// Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the 263// structure. 264static struct submix_stream_out * audio_stream_out_get_submix_stream_out( 265 struct audio_stream_out * const stream) 266{ 267 ALOG_ASSERT(stream); 268 return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) - 269 offsetof(struct submix_stream_out, stream)); 270} 271 272// Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure. 273static struct submix_stream_out * audio_stream_get_submix_stream_out( 274 struct audio_stream * const stream) 275{ 276 ALOG_ASSERT(stream); 277 return audio_stream_out_get_submix_stream_out( 278 reinterpret_cast<struct audio_stream_out *>(stream)); 279} 280 281// Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the 282// structure. 283static struct submix_stream_in * audio_stream_in_get_submix_stream_in( 284 struct audio_stream_in * const stream) 285{ 286 ALOG_ASSERT(stream); 287 return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) - 288 offsetof(struct submix_stream_in, stream)); 289} 290 291// Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure. 292static struct submix_stream_in * audio_stream_get_submix_stream_in( 293 struct audio_stream * const stream) 294{ 295 ALOG_ASSERT(stream); 296 return audio_stream_in_get_submix_stream_in( 297 reinterpret_cast<struct audio_stream_in *>(stream)); 298} 299 300// Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within 301// the structure. 302static struct submix_audio_device * audio_hw_device_get_submix_audio_device( 303 struct audio_hw_device *device) 304{ 305 ALOG_ASSERT(device); 306 return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) - 307 offsetof(struct submix_audio_device, device)); 308} 309 310// Get the number of channels referenced by the specified channel_mask. The channel_mask can 311// reference either input or output channels. 312uint32_t get_channel_count_from_mask(const audio_channel_mask_t channel_mask) { 313 if (audio_is_input_channel(channel_mask)) { 314 return popcount(channel_mask & AUDIO_CHANNEL_IN_ALL); 315 } else if (audio_is_output_channel(channel_mask)) { 316 return popcount(channel_mask & AUDIO_CHANNEL_OUT_ALL); 317 } 318 ALOGE("get_channel_count(): No channels specified in channel mask %x", channel_mask); 319 return 0; 320} 321 322// Compare an audio_config with input channel mask and an audio_config with output channel mask 323// returning false if they do *not* match, true otherwise. 324static bool audio_config_compare(const audio_config * const input_config, 325 const audio_config * const output_config) 326{ 327#if !ENABLE_CHANNEL_CONVERSION 328 const uint32_t input_channels = get_channel_count_from_mask(input_config->channel_mask); 329 const uint32_t output_channels = get_channel_count_from_mask(output_config->channel_mask); 330 if (input_channels != output_channels) { 331 ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d", 332 input_channels, output_channels); 333 return false; 334 } 335#endif // !ENABLE_CHANNEL_CONVERSION 336#if ENABLE_RESAMPLING 337 if (input_config->sample_rate != output_config->sample_rate && 338 get_channel_count_from_mask(input_config->channel_mask) != 1) { 339#else 340 if (input_config->sample_rate != output_config->sample_rate) { 341#endif // ENABLE_RESAMPLING 342 ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", 343 input_config->sample_rate, output_config->sample_rate); 344 return false; 345 } 346 if (input_config->format != output_config->format) { 347 ALOGE("audio_config_compare() format mismatch %x vs. %x", 348 input_config->format, output_config->format); 349 return false; 350 } 351 // This purposely ignores offload_info as it's not required for the submix device. 352 return true; 353} 354 355// If one doesn't exist, create a pipe for the submix audio device rsxadev of size 356// buffer_size_frames and optionally associate "in" or "out" with the submix audio device. 357static void submix_audio_device_create_pipe(struct submix_audio_device * const rsxadev, 358 const struct audio_config * const config, 359 const size_t buffer_size_frames, 360 const uint32_t buffer_period_count, 361 struct submix_stream_in * const in, 362 struct submix_stream_out * const out) 363{ 364 ALOG_ASSERT(in || out); 365 ALOGV("submix_audio_device_create_pipe()"); 366 pthread_mutex_lock(&rsxadev->lock); 367 // Save a reference to the specified input or output stream and the associated channel 368 // mask. 369 if (in) { 370 rsxadev->input = in; 371 rsxadev->config.input_channel_mask = config->channel_mask; 372#if ENABLE_RESAMPLING 373 rsxadev->config.input_sample_rate = config->sample_rate; 374 // If the output isn't configured yet, set the output sample rate to the maximum supported 375 // sample rate such that the smallest possible input buffer is created. 376 if (!rsxadev->output) { 377 rsxadev->config.output_sample_rate = 48000; 378 } 379#endif // ENABLE_RESAMPLING 380 } 381 if (out) { 382 rsxadev->output = out; 383 rsxadev->config.output_channel_mask = config->channel_mask; 384#if ENABLE_RESAMPLING 385 rsxadev->config.output_sample_rate = config->sample_rate; 386#endif // ENABLE_RESAMPLING 387 } 388 // If a pipe isn't associated with the device, create one. 389 if (rsxadev->rsxSink == NULL || rsxadev->rsxSource == NULL) { 390 struct submix_config * const device_config = &rsxadev->config; 391 const NBAIO_Format format = Format_from_SR_C(config->sample_rate, 392 get_channel_count_from_mask(config->channel_mask), config->format); 393 const NBAIO_Format offers[1] = {format}; 394 size_t numCounterOffers = 0; 395 // Create a MonoPipe with optional blocking set to true. 396 MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); 397 // Negotiation between the source and sink cannot fail as the device open operation 398 // creates both ends of the pipe using the same audio format. 399 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 400 ALOG_ASSERT(index == 0); 401 MonoPipeReader* source = new MonoPipeReader(sink); 402 numCounterOffers = 0; 403 index = source->negotiate(offers, 1, NULL, numCounterOffers); 404 ALOG_ASSERT(index == 0); 405 ALOGV("submix_audio_device_create_pipe(): created pipe"); 406 407 // Save references to the source and sink. 408 ALOG_ASSERT(rsxadev->rsxSink == NULL); 409 ALOG_ASSERT(rsxadev->rsxSource == NULL); 410 rsxadev->rsxSink = sink; 411 rsxadev->rsxSource = source; 412 // Store the sanitized audio format in the device so that it's possible to determine 413 // the format of the pipe source when opening the input device. 414 memcpy(&device_config->common, config, sizeof(device_config->common)); 415 device_config->buffer_size_frames = sink->maxFrames(); 416 device_config->buffer_period_size_frames = device_config->buffer_size_frames / 417 buffer_period_count; 418 if (in) device_config->pipe_frame_size = audio_stream_frame_size(&in->stream.common); 419 if (out) device_config->pipe_frame_size = audio_stream_frame_size(&out->stream.common); 420 SUBMIX_ALOGV("submix_audio_device_create_pipe(): pipe frame size %zd, pipe size %zd, " 421 "period size %zd", device_config->pipe_frame_size, 422 device_config->buffer_size_frames, device_config->buffer_period_size_frames); 423 } 424 pthread_mutex_unlock(&rsxadev->lock); 425} 426 427// Release references to the sink and source. Input and output threads may maintain references 428// to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use 429// before they shutdown. 430static void submix_audio_device_release_pipe(struct submix_audio_device * const rsxadev) 431{ 432 ALOGV("submix_audio_device_release_pipe()"); 433 rsxadev->rsxSink.clear(); 434 rsxadev->rsxSource.clear(); 435} 436 437// Remove references to the specified input and output streams. When the device no longer 438// references input and output streams destroy the associated pipe. 439static void submix_audio_device_destroy_pipe(struct submix_audio_device * const rsxadev, 440 const struct submix_stream_in * const in, 441 const struct submix_stream_out * const out) 442{ 443 MonoPipe* sink; 444 pthread_mutex_lock(&rsxadev->lock); 445 ALOGV("submix_audio_device_destroy_pipe()"); 446 ALOG_ASSERT(in == NULL || rsxadev->input == in); 447 ALOG_ASSERT(out == NULL || rsxadev->output == out); 448 if (in != NULL) { 449#if ENABLE_LEGACY_INPUT_OPEN 450 const_cast<struct submix_stream_in*>(in)->ref_count--; 451 if (in->ref_count == 0) { 452 rsxadev->input = NULL; 453 } 454 ALOGV("submix_audio_device_destroy_pipe(): input ref_count %d", in->ref_count); 455#else 456 rsxadev->input = NULL; 457#endif // ENABLE_LEGACY_INPUT_OPEN 458 } 459 if (out != NULL) rsxadev->output = NULL; 460 if (rsxadev->input != NULL && rsxadev->output != NULL) { 461 submix_audio_device_release_pipe(rsxadev); 462 ALOGV("submix_audio_device_destroy_pipe(): pipe destroyed"); 463 } 464 pthread_mutex_unlock(&rsxadev->lock); 465} 466 467// Sanitize the user specified audio config for a submix input / output stream. 468static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) 469{ 470 config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : 471 get_supported_channel_out_mask(config->channel_mask); 472 config->sample_rate = get_supported_sample_rate(config->sample_rate); 473 config->format = DEFAULT_FORMAT; 474} 475 476// Verify a submix input or output stream can be opened. 477static bool submix_open_validate(const struct submix_audio_device * const rsxadev, 478 pthread_mutex_t * const lock, 479 const struct audio_config * const config, 480 const bool opening_input) 481{ 482 bool input_open; 483 bool output_open; 484 audio_config pipe_config; 485 486 // Query the device for the current audio config and whether input and output streams are open. 487 pthread_mutex_lock(lock); 488 output_open = rsxadev->output != NULL; 489 input_open = rsxadev->input != NULL; 490 memcpy(&pipe_config, &rsxadev->config.common, sizeof(pipe_config)); 491 pthread_mutex_unlock(lock); 492 493 // If the stream is already open, don't open it again. 494 if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { 495 ALOGE("submix_open_validate(): %s stream already open.", opening_input ? "Input" : 496 "Output"); 497 return false; 498 } 499 500 SUBMIX_ALOGV("submix_open_validate(): sample rate=%d format=%x " 501 "%s_channel_mask=%x", config->sample_rate, config->format, 502 opening_input ? "in" : "out", config->channel_mask); 503 504 // If either stream is open, verify the existing audio config the pipe matches the user 505 // specified config. 506 if (input_open || output_open) { 507 const audio_config * const input_config = opening_input ? config : &pipe_config; 508 const audio_config * const output_config = opening_input ? &pipe_config : config; 509 // Get the channel mask of the open device. 510 pipe_config.channel_mask = 511 opening_input ? rsxadev->config.output_channel_mask : 512 rsxadev->config.input_channel_mask; 513 if (!audio_config_compare(input_config, output_config)) { 514 ALOGE("submix_open_validate(): Unsupported format."); 515 return false; 516 } 517 } 518 return true; 519} 520 521// Calculate the maximum size of the pipe buffer in frames for the specified stream. 522static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, 523 const struct submix_config *config, 524 const size_t pipe_frames) 525{ 526 const size_t stream_frame_size = audio_stream_frame_size(stream); 527 const size_t pipe_frame_size = config->pipe_frame_size; 528 const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); 529 return (pipe_frames * config->pipe_frame_size) / max_frame_size; 530} 531 532/* audio HAL functions */ 533 534static uint32_t out_get_sample_rate(const struct audio_stream *stream) 535{ 536 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 537 const_cast<struct audio_stream *>(stream)); 538#if ENABLE_RESAMPLING 539 const uint32_t out_rate = out->dev->config.output_sample_rate; 540#else 541 const uint32_t out_rate = out->dev->config.common.sample_rate; 542#endif // ENABLE_RESAMPLING 543 SUBMIX_ALOGV("out_get_sample_rate() returns %u", out_rate); 544 return out_rate; 545} 546 547static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 548{ 549 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 550#if ENABLE_RESAMPLING 551 // The sample rate of the stream can't be changed once it's set since this would change the 552 // output buffer size and hence break playback to the shared pipe. 553 if (rate != out->dev->config.output_sample_rate) { 554 ALOGE("out_set_sample_rate(rate=%u) resampling enabled can't change sample rate from " 555 "%u to %u", out->dev->config.output_sample_rate, rate); 556 return -ENOSYS; 557 } 558#endif // ENABLE_RESAMPLING 559 if (!sample_rate_supported(rate)) { 560 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 561 return -ENOSYS; 562 } 563 SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); 564 out->dev->config.common.sample_rate = rate; 565 return 0; 566} 567 568static size_t out_get_buffer_size(const struct audio_stream *stream) 569{ 570 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 571 const_cast<struct audio_stream *>(stream)); 572 const struct submix_config * const config = &out->dev->config; 573 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 574 stream, config, config->buffer_period_size_frames); 575 const size_t buffer_size_bytes = buffer_size_frames * audio_stream_frame_size(stream); 576 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", 577 buffer_size_bytes, buffer_size_frames); 578 return buffer_size_bytes; 579} 580 581static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 582{ 583 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 584 const_cast<struct audio_stream *>(stream)); 585 uint32_t channel_mask = out->dev->config.output_channel_mask; 586 SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); 587 return channel_mask; 588} 589 590static audio_format_t out_get_format(const struct audio_stream *stream) 591{ 592 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 593 const_cast<struct audio_stream *>(stream)); 594 const audio_format_t format = out->dev->config.common.format; 595 SUBMIX_ALOGV("out_get_format() returns %x", format); 596 return format; 597} 598 599static int out_set_format(struct audio_stream *stream, audio_format_t format) 600{ 601 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 602 if (format != out->dev->config.common.format) { 603 ALOGE("out_set_format(format=%x) format unsupported", format); 604 return -ENOSYS; 605 } 606 SUBMIX_ALOGV("out_set_format(format=%x)", format); 607 return 0; 608} 609 610static int out_standby(struct audio_stream *stream) 611{ 612 struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_out(stream)->dev; 613 ALOGI("out_standby()"); 614 615 pthread_mutex_lock(&rsxadev->lock); 616 617 rsxadev->output_standby = true; 618 619 pthread_mutex_unlock(&rsxadev->lock); 620 621 return 0; 622} 623 624static int out_dump(const struct audio_stream *stream, int fd) 625{ 626 (void)stream; 627 (void)fd; 628 return 0; 629} 630 631static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 632{ 633 int exiting = -1; 634 AudioParameter parms = AudioParameter(String8(kvpairs)); 635 SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs); 636 637 // FIXME this is using hard-coded strings but in the future, this functionality will be 638 // converted to use audio HAL extensions required to support tunneling 639 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { 640 struct submix_audio_device * const rsxadev = 641 audio_stream_get_submix_stream_out(stream)->dev; 642 pthread_mutex_lock(&rsxadev->lock); 643 { // using the sink 644 sp<MonoPipe> sink = rsxadev->rsxSink; 645 if (sink == NULL) { 646 pthread_mutex_unlock(&rsxadev->lock); 647 return 0; 648 } 649 650 ALOGI("out_set_parameters(): shutdown"); 651 sink->shutdown(true); 652 } // done using the sink 653 pthread_mutex_unlock(&rsxadev->lock); 654 } 655 return 0; 656} 657 658static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 659{ 660 (void)stream; 661 (void)keys; 662 return strdup(""); 663} 664 665static uint32_t out_get_latency(const struct audio_stream_out *stream) 666{ 667 const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( 668 const_cast<struct audio_stream_out *>(stream)); 669 const struct submix_config * const config = &out->dev->config; 670 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 671 &stream->common, config, config->buffer_size_frames); 672#if ENABLE_RESAMPLING 673 // Sample rate conversion occurs when data is read from the input so data in the buffer is 674 // at output_sample_rate Hz. 675 const uint32_t latency_ms = (buffer_size_frames * 1000) / config->output_sample_rate; 676#else 677 const uint32_t latency_ms = (buffer_size_frames * 1000) / config->common.sample_rate; 678#endif // ENABLE_RESAMPLING 679 SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", 680 latency_ms, buffer_size_frames, config->common.sample_rate); 681 return latency_ms; 682} 683 684static int out_set_volume(struct audio_stream_out *stream, float left, 685 float right) 686{ 687 (void)stream; 688 (void)left; 689 (void)right; 690 return -ENOSYS; 691} 692 693static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 694 size_t bytes) 695{ 696 SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); 697 ssize_t written_frames = 0; 698 const size_t frame_size = audio_stream_frame_size(&stream->common); 699 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 700 struct submix_audio_device * const rsxadev = out->dev; 701 const size_t frames = bytes / frame_size; 702 703 pthread_mutex_lock(&rsxadev->lock); 704 705 rsxadev->output_standby = false; 706 707 sp<MonoPipe> sink = rsxadev->rsxSink; 708 if (sink != NULL) { 709 if (sink->isShutdown()) { 710 sink.clear(); 711 pthread_mutex_unlock(&rsxadev->lock); 712 SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); 713 // the pipe has already been shutdown, this buffer will be lost but we must 714 // simulate timing so we don't drain the output faster than realtime 715 usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); 716 return bytes; 717 } 718 } else { 719 pthread_mutex_unlock(&rsxadev->lock); 720 ALOGE("out_write without a pipe!"); 721 ALOG_ASSERT("out_write without a pipe!"); 722 return 0; 723 } 724 725 // If the write to the sink would block when no input stream is present, flush enough frames 726 // from the pipe to make space to write the most recent data. 727 { 728 const size_t availableToWrite = sink->availableToWrite(); 729 sp<MonoPipeReader> source = rsxadev->rsxSource; 730 if (rsxadev->input == NULL && availableToWrite < frames) { 731 static uint8_t flush_buffer[64]; 732 const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; 733 size_t frames_to_flush_from_source = frames - availableToWrite; 734 SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", 735 frames_to_flush_from_source); 736 while (frames_to_flush_from_source) { 737 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); 738 frames_to_flush_from_source -= flush_size; 739 source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); 740 } 741 } 742 } 743 744 pthread_mutex_unlock(&rsxadev->lock); 745 746 written_frames = sink->write(buffer, frames); 747 748#if LOG_STREAMS_TO_FILES 749 if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); 750#endif // LOG_STREAMS_TO_FILES 751 752 if (written_frames < 0) { 753 if (written_frames == (ssize_t)NEGOTIATE) { 754 ALOGE("out_write() write to pipe returned NEGOTIATE"); 755 756 pthread_mutex_lock(&rsxadev->lock); 757 sink.clear(); 758 pthread_mutex_unlock(&rsxadev->lock); 759 760 written_frames = 0; 761 return 0; 762 } else { 763 // write() returned UNDERRUN or WOULD_BLOCK, retry 764 ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); 765 written_frames = sink->write(buffer, frames); 766 } 767 } 768 769 pthread_mutex_lock(&rsxadev->lock); 770 sink.clear(); 771 pthread_mutex_unlock(&rsxadev->lock); 772 773 if (written_frames < 0) { 774 ALOGE("out_write() failed writing to pipe with %zd", written_frames); 775 return 0; 776 } 777 const ssize_t written_bytes = written_frames * frame_size; 778 SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); 779 return written_bytes; 780} 781 782static int out_get_render_position(const struct audio_stream_out *stream, 783 uint32_t *dsp_frames) 784{ 785 (void)stream; 786 (void)dsp_frames; 787 return -EINVAL; 788} 789 790static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 791{ 792 (void)stream; 793 (void)effect; 794 return 0; 795} 796 797static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 798{ 799 (void)stream; 800 (void)effect; 801 return 0; 802} 803 804static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 805 int64_t *timestamp) 806{ 807 (void)stream; 808 (void)timestamp; 809 return -EINVAL; 810} 811 812/** audio_stream_in implementation **/ 813static uint32_t in_get_sample_rate(const struct audio_stream *stream) 814{ 815 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 816 const_cast<struct audio_stream*>(stream)); 817#if ENABLE_RESAMPLING 818 const uint32_t rate = in->dev->config.input_sample_rate; 819#else 820 const uint32_t rate = in->dev->config.common.sample_rate; 821#endif // ENABLE_RESAMPLING 822 SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); 823 return rate; 824} 825 826static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 827{ 828 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 829#if ENABLE_RESAMPLING 830 // The sample rate of the stream can't be changed once it's set since this would change the 831 // input buffer size and hence break recording from the shared pipe. 832 if (rate != in->dev->config.input_sample_rate) { 833 ALOGE("in_set_sample_rate(rate=%u) resampling enabled can't change sample rate from " 834 "%u to %u", in->dev->config.input_sample_rate, rate); 835 return -ENOSYS; 836 } 837#endif // ENABLE_RESAMPLING 838 if (!sample_rate_supported(rate)) { 839 ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); 840 return -ENOSYS; 841 } 842 in->dev->config.common.sample_rate = rate; 843 SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); 844 return 0; 845} 846 847static size_t in_get_buffer_size(const struct audio_stream *stream) 848{ 849 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 850 const_cast<struct audio_stream*>(stream)); 851 const struct submix_config * const config = &in->dev->config; 852 size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 853 stream, config, config->buffer_period_size_frames); 854#if ENABLE_RESAMPLING 855 // Scale the size of the buffer based upon the maximum number of frames that could be returned 856 // given the ratio of output to input sample rate. 857 buffer_size_frames = (size_t)(((float)buffer_size_frames * 858 (float)config->input_sample_rate) / 859 (float)config->output_sample_rate); 860#endif // ENABLE_RESAMPLING 861 const size_t buffer_size_bytes = buffer_size_frames * audio_stream_frame_size(stream); 862 SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, 863 buffer_size_frames); 864 return buffer_size_bytes; 865} 866 867static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 868{ 869 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 870 const_cast<struct audio_stream*>(stream)); 871 const audio_channel_mask_t channel_mask = in->dev->config.input_channel_mask; 872 SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); 873 return channel_mask; 874} 875 876static audio_format_t in_get_format(const struct audio_stream *stream) 877{ 878 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 879 const_cast<struct audio_stream*>(stream)); 880 const audio_format_t format = in->dev->config.common.format; 881 SUBMIX_ALOGV("in_get_format() returns %x", format); 882 return format; 883} 884 885static int in_set_format(struct audio_stream *stream, audio_format_t format) 886{ 887 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 888 if (format != in->dev->config.common.format) { 889 ALOGE("in_set_format(format=%x) format unsupported", format); 890 return -ENOSYS; 891 } 892 SUBMIX_ALOGV("in_set_format(format=%x)", format); 893 return 0; 894} 895 896static int in_standby(struct audio_stream *stream) 897{ 898 struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_in(stream)->dev; 899 ALOGI("in_standby()"); 900 901 pthread_mutex_lock(&rsxadev->lock); 902 903 rsxadev->input_standby = true; 904 905 pthread_mutex_unlock(&rsxadev->lock); 906 907 return 0; 908} 909 910static int in_dump(const struct audio_stream *stream, int fd) 911{ 912 (void)stream; 913 (void)fd; 914 return 0; 915} 916 917static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 918{ 919 (void)stream; 920 (void)kvpairs; 921 return 0; 922} 923 924static char * in_get_parameters(const struct audio_stream *stream, 925 const char *keys) 926{ 927 (void)stream; 928 (void)keys; 929 return strdup(""); 930} 931 932static int in_set_gain(struct audio_stream_in *stream, float gain) 933{ 934 (void)stream; 935 (void)gain; 936 return 0; 937} 938 939static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 940 size_t bytes) 941{ 942 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 943 struct submix_audio_device * const rsxadev = in->dev; 944 struct audio_config *format; 945 const size_t frame_size = audio_stream_frame_size(&stream->common); 946 const size_t frames_to_read = bytes / frame_size; 947 948 SUBMIX_ALOGV("in_read bytes=%zu", bytes); 949 pthread_mutex_lock(&rsxadev->lock); 950 951 const bool output_standby_transition = (in->output_standby != in->dev->output_standby); 952 in->output_standby = rsxadev->output_standby; 953 954 if (rsxadev->input_standby || output_standby_transition) { 955 rsxadev->input_standby = false; 956 // keep track of when we exit input standby (== first read == start "real recording") 957 // or when we start recording silence, and reset projected time 958 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 959 if (rc == 0) { 960 in->read_counter_frames = 0; 961 } 962 } 963 964 in->read_counter_frames += frames_to_read; 965 size_t remaining_frames = frames_to_read; 966 967 { 968 // about to read from audio source 969 sp<MonoPipeReader> source = rsxadev->rsxSource; 970 if (source == NULL) { 971 ALOGE("no audio pipe yet we're trying to read!"); 972 pthread_mutex_unlock(&rsxadev->lock); 973 usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); 974 memset(buffer, 0, bytes); 975 return bytes; 976 } 977 978 pthread_mutex_unlock(&rsxadev->lock); 979 980 // read the data from the pipe (it's non blocking) 981 int attempts = 0; 982 char* buff = (char*)buffer; 983#if ENABLE_CHANNEL_CONVERSION 984 // Determine whether channel conversion is required. 985 const uint32_t input_channels = get_channel_count_from_mask( 986 rsxadev->config.input_channel_mask); 987 const uint32_t output_channels = get_channel_count_from_mask( 988 rsxadev->config.output_channel_mask); 989 if (input_channels != output_channels) { 990 SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " 991 "input channels", output_channels, input_channels); 992 // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. 993 ALOG_ASSERT(rsxadev->config.common.format == AUDIO_FORMAT_PCM_16_BIT); 994 ALOG_ASSERT((input_channels == 1 && output_channels == 2) || 995 (input_channels == 2 && output_channels == 1)); 996 } 997#endif // ENABLE_CHANNEL_CONVERSION 998 999#if ENABLE_RESAMPLING 1000 const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); 1001 const uint32_t output_sample_rate = rsxadev->config.output_sample_rate; 1002 const size_t resampler_buffer_size_frames = 1003 sizeof(rsxadev->resampler_buffer) / sizeof(rsxadev->resampler_buffer[0]); 1004 float resampler_ratio = 1.0f; 1005 // Determine whether resampling is required. 1006 if (input_sample_rate != output_sample_rate) { 1007 resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; 1008 // Only support 16-bit PCM mono resampling. 1009 // NOTE: Resampling is performed after the channel conversion step. 1010 ALOG_ASSERT(rsxadev->config.common.format == AUDIO_FORMAT_PCM_16_BIT); 1011 ALOG_ASSERT(get_channel_count_from_mask(rsxadev->config.input_channel_mask) == 1); 1012 } 1013#endif // ENABLE_RESAMPLING 1014 1015 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 1016 ssize_t frames_read = -1977; 1017 size_t read_frames = remaining_frames; 1018#if ENABLE_RESAMPLING 1019 char* const saved_buff = buff; 1020 if (resampler_ratio != 1.0f) { 1021 // Calculate the number of frames from the pipe that need to be read to generate 1022 // the data for the input stream read. 1023 const size_t frames_required_for_resampler = (size_t)( 1024 (float)read_frames * (float)resampler_ratio); 1025 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); 1026 // Read into the resampler buffer. 1027 buff = (char*)rsxadev->resampler_buffer; 1028 } 1029#endif // ENABLE_RESAMPLING 1030#if ENABLE_CHANNEL_CONVERSION 1031 if (output_channels == 1 && input_channels == 2) { 1032 // Need to read half the requested frames since the converted output 1033 // data will take twice the space (mono->stereo). 1034 read_frames /= 2; 1035 } 1036#endif // ENABLE_CHANNEL_CONVERSION 1037 1038 SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); 1039 1040 frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS); 1041 1042 SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); 1043 1044#if ENABLE_CHANNEL_CONVERSION 1045 // Perform in-place channel conversion. 1046 // NOTE: In the following "input stream" refers to the data returned by this function 1047 // and "output stream" refers to the data read from the pipe. 1048 if (input_channels != output_channels && frames_read > 0) { 1049 int16_t *data = (int16_t*)buff; 1050 if (output_channels == 2 && input_channels == 1) { 1051 // Offset into the output stream data in samples. 1052 ssize_t output_stream_offset = 0; 1053 for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; 1054 input_stream_frame++, output_stream_offset += 2) { 1055 // Average the content from both channels. 1056 data[input_stream_frame] = ((int32_t)data[output_stream_offset] + 1057 (int32_t)data[output_stream_offset + 1]) / 2; 1058 } 1059 } else if (output_channels == 1 && input_channels == 2) { 1060 // Offset into the input stream data in samples. 1061 ssize_t input_stream_offset = (frames_read - 1) * 2; 1062 for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; 1063 output_stream_frame--, input_stream_offset -= 2) { 1064 const short sample = data[output_stream_frame]; 1065 data[input_stream_offset] = sample; 1066 data[input_stream_offset + 1] = sample; 1067 } 1068 } 1069 } 1070#endif // ENABLE_CHANNEL_CONVERSION 1071 1072#if ENABLE_RESAMPLING 1073 if (resampler_ratio != 1.0f) { 1074 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); 1075 const int16_t * const data = (int16_t*)buff; 1076 int16_t * const resampled_buffer = (int16_t*)saved_buff; 1077 // Resample with *no* filtering - if the data from the ouptut stream was really 1078 // sampled at a different rate this will result in very nasty aliasing. 1079 const float output_stream_frames = (float)frames_read; 1080 size_t input_stream_frame = 0; 1081 for (float output_stream_frame = 0.0f; 1082 output_stream_frame < output_stream_frames && 1083 input_stream_frame < remaining_frames; 1084 output_stream_frame += resampler_ratio, input_stream_frame++) { 1085 resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; 1086 } 1087 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); 1088 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); 1089 frames_read = input_stream_frame; 1090 buff = saved_buff; 1091 } 1092#endif // ENABLE_RESAMPLING 1093 1094 if (frames_read > 0) { 1095#if LOG_STREAMS_TO_FILES 1096 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); 1097#endif // LOG_STREAMS_TO_FILES 1098 1099 remaining_frames -= frames_read; 1100 buff += frames_read * frame_size; 1101 SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", 1102 attempts, frames_read, remaining_frames); 1103 } else { 1104 attempts++; 1105 SUBMIX_ALOGE(" in_read read returned %zd", frames_read); 1106 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 1107 } 1108 } 1109 // done using the source 1110 pthread_mutex_lock(&rsxadev->lock); 1111 source.clear(); 1112 pthread_mutex_unlock(&rsxadev->lock); 1113 } 1114 1115 if (remaining_frames > 0) { 1116 const size_t remaining_bytes = remaining_frames * frame_size; 1117 SUBMIX_ALOGV(" remaining_frames = %zu", remaining_frames); 1118 memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); 1119 } 1120 1121 // compute how much we need to sleep after reading the data by comparing the wall clock with 1122 // the projected time at which we should return. 1123 struct timespec time_after_read;// wall clock after reading from the pipe 1124 struct timespec record_duration;// observed record duration 1125 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 1126 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 1127 if (rc == 0) { 1128 // for how long have we been recording? 1129 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 1130 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 1131 if (record_duration.tv_nsec < 0) { 1132 record_duration.tv_sec--; 1133 record_duration.tv_nsec += 1000000000; 1134 } 1135 1136 // read_counter_frames contains the number of frames that have been read since the 1137 // beginning of recording (including this call): it's converted to usec and compared to 1138 // how long we've been recording for, which gives us how long we must wait to sync the 1139 // projected recording time, and the observed recording time. 1140 long projected_vs_observed_offset_us = 1141 ((int64_t)(in->read_counter_frames 1142 - (record_duration.tv_sec*sample_rate))) 1143 * 1000000 / sample_rate 1144 - (record_duration.tv_nsec / 1000); 1145 1146 SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 1147 record_duration.tv_sec, record_duration.tv_nsec/1000000, 1148 projected_vs_observed_offset_us); 1149 if (projected_vs_observed_offset_us > 0) { 1150 usleep(projected_vs_observed_offset_us); 1151 } 1152 } 1153 1154 SUBMIX_ALOGV("in_read returns %zu", bytes); 1155 return bytes; 1156 1157} 1158 1159static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 1160{ 1161 (void)stream; 1162 return 0; 1163} 1164 1165static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1166{ 1167 (void)stream; 1168 (void)effect; 1169 return 0; 1170} 1171 1172static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1173{ 1174 (void)stream; 1175 (void)effect; 1176 return 0; 1177} 1178 1179static int adev_open_output_stream(struct audio_hw_device *dev, 1180 audio_io_handle_t handle, 1181 audio_devices_t devices, 1182 audio_output_flags_t flags, 1183 struct audio_config *config, 1184 struct audio_stream_out **stream_out) 1185{ 1186 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1187 ALOGV("adev_open_output_stream()"); 1188 struct submix_stream_out *out; 1189 (void)handle; 1190 (void)devices; 1191 (void)flags; 1192 1193 *stream_out = NULL; 1194 1195 // Make sure it's possible to open the device given the current audio config. 1196 submix_sanitize_config(config, false); 1197 if (!submix_open_validate(rsxadev, &rsxadev->lock, config, false)) { 1198 ALOGE("adev_open_output_stream(): Unable to open output stream."); 1199 return -EINVAL; 1200 } 1201 1202 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 1203 if (!out) return -ENOMEM; 1204 1205 // Initialize the function pointer tables (v-tables). 1206 out->stream.common.get_sample_rate = out_get_sample_rate; 1207 out->stream.common.set_sample_rate = out_set_sample_rate; 1208 out->stream.common.get_buffer_size = out_get_buffer_size; 1209 out->stream.common.get_channels = out_get_channels; 1210 out->stream.common.get_format = out_get_format; 1211 out->stream.common.set_format = out_set_format; 1212 out->stream.common.standby = out_standby; 1213 out->stream.common.dump = out_dump; 1214 out->stream.common.set_parameters = out_set_parameters; 1215 out->stream.common.get_parameters = out_get_parameters; 1216 out->stream.common.add_audio_effect = out_add_audio_effect; 1217 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1218 out->stream.get_latency = out_get_latency; 1219 out->stream.set_volume = out_set_volume; 1220 out->stream.write = out_write; 1221 out->stream.get_render_position = out_get_render_position; 1222 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1223 1224 // If the sink has been shutdown, delete the pipe so that it's recreated. 1225 pthread_mutex_lock(&rsxadev->lock); 1226 if (rsxadev->rsxSink != NULL && rsxadev->rsxSink->isShutdown()) { 1227 submix_audio_device_release_pipe(rsxadev); 1228 } 1229 pthread_mutex_unlock(&rsxadev->lock); 1230 1231 // Store a pointer to the device from the output stream. 1232 out->dev = rsxadev; 1233 // Initialize the pipe. 1234 ALOGV("adev_open_output_stream(): Initializing pipe"); 1235 submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1236 DEFAULT_PIPE_PERIOD_COUNT, NULL, out); 1237#if LOG_STREAMS_TO_FILES 1238 out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1239 LOG_STREAM_FILE_PERMISSIONS); 1240 ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", 1241 strerror(errno)); 1242 ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); 1243#endif // LOG_STREAMS_TO_FILES 1244 // Return the output stream. 1245 *stream_out = &out->stream; 1246 1247 return 0; 1248} 1249 1250static void adev_close_output_stream(struct audio_hw_device *dev, 1251 struct audio_stream_out *stream) 1252{ 1253 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 1254 ALOGV("adev_close_output_stream()"); 1255 submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), NULL, out); 1256#if LOG_STREAMS_TO_FILES 1257 if (out->log_fd >= 0) close(out->log_fd); 1258#endif // LOG_STREAMS_TO_FILES 1259 free(out); 1260} 1261 1262static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1263{ 1264 (void)dev; 1265 (void)kvpairs; 1266 return -ENOSYS; 1267} 1268 1269static char * adev_get_parameters(const struct audio_hw_device *dev, 1270 const char *keys) 1271{ 1272 (void)dev; 1273 (void)keys; 1274 return strdup("");; 1275} 1276 1277static int adev_init_check(const struct audio_hw_device *dev) 1278{ 1279 ALOGI("adev_init_check()"); 1280 (void)dev; 1281 return 0; 1282} 1283 1284static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 1285{ 1286 (void)dev; 1287 (void)volume; 1288 return -ENOSYS; 1289} 1290 1291static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 1292{ 1293 (void)dev; 1294 (void)volume; 1295 return -ENOSYS; 1296} 1297 1298static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 1299{ 1300 (void)dev; 1301 (void)volume; 1302 return -ENOSYS; 1303} 1304 1305static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 1306{ 1307 (void)dev; 1308 (void)muted; 1309 return -ENOSYS; 1310} 1311 1312static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 1313{ 1314 (void)dev; 1315 (void)muted; 1316 return -ENOSYS; 1317} 1318 1319static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 1320{ 1321 (void)dev; 1322 (void)mode; 1323 return 0; 1324} 1325 1326static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 1327{ 1328 (void)dev; 1329 (void)state; 1330 return -ENOSYS; 1331} 1332 1333static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 1334{ 1335 (void)dev; 1336 (void)state; 1337 return -ENOSYS; 1338} 1339 1340static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 1341 const struct audio_config *config) 1342{ 1343 if (audio_is_linear_pcm(config->format)) { 1344 const size_t buffer_period_size_frames = 1345 audio_hw_device_get_submix_audio_device(const_cast<struct audio_hw_device*>(dev))-> 1346 config.buffer_period_size_frames; 1347 const size_t frame_size_in_bytes = get_channel_count_from_mask(config->channel_mask) * 1348 audio_bytes_per_sample(config->format); 1349 const size_t buffer_size = buffer_period_size_frames * frame_size_in_bytes; 1350 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", 1351 buffer_size, buffer_period_size_frames); 1352 return buffer_size; 1353 } 1354 return 0; 1355} 1356 1357static int adev_open_input_stream(struct audio_hw_device *dev, 1358 audio_io_handle_t handle, 1359 audio_devices_t devices, 1360 struct audio_config *config, 1361 struct audio_stream_in **stream_in) 1362{ 1363 struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); 1364 struct submix_stream_in *in; 1365 ALOGI("adev_open_input_stream()"); 1366 (void)handle; 1367 (void)devices; 1368 1369 *stream_in = NULL; 1370 1371 // Make sure it's possible to open the device given the current audio config. 1372 submix_sanitize_config(config, true); 1373 if (!submix_open_validate(rsxadev, &rsxadev->lock, config, true)) { 1374 ALOGE("adev_open_input_stream(): Unable to open input stream."); 1375 return -EINVAL; 1376 } 1377 1378#if ENABLE_LEGACY_INPUT_OPEN 1379 pthread_mutex_lock(&rsxadev->lock); 1380 in = rsxadev->input; 1381 if (in) { 1382 in->ref_count++; 1383 sp<MonoPipe> sink = rsxadev->rsxSink; 1384 ALOG_ASSERT(sink != NULL); 1385 // If the sink has been shutdown, delete the pipe. 1386 if (sink->isShutdown()) submix_audio_device_release_pipe(rsxadev); 1387 } 1388 pthread_mutex_unlock(&rsxadev->lock); 1389#else 1390 in = NULL; 1391#endif // ENABLE_LEGACY_INPUT_OPEN 1392 1393 if (!in) { 1394 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 1395 if (!in) return -ENOMEM; 1396 in->ref_count = 1; 1397 1398 // Initialize the function pointer tables (v-tables). 1399 in->stream.common.get_sample_rate = in_get_sample_rate; 1400 in->stream.common.set_sample_rate = in_set_sample_rate; 1401 in->stream.common.get_buffer_size = in_get_buffer_size; 1402 in->stream.common.get_channels = in_get_channels; 1403 in->stream.common.get_format = in_get_format; 1404 in->stream.common.set_format = in_set_format; 1405 in->stream.common.standby = in_standby; 1406 in->stream.common.dump = in_dump; 1407 in->stream.common.set_parameters = in_set_parameters; 1408 in->stream.common.get_parameters = in_get_parameters; 1409 in->stream.common.add_audio_effect = in_add_audio_effect; 1410 in->stream.common.remove_audio_effect = in_remove_audio_effect; 1411 in->stream.set_gain = in_set_gain; 1412 in->stream.read = in_read; 1413 in->stream.get_input_frames_lost = in_get_input_frames_lost; 1414 } 1415 1416 // Initialize the input stream. 1417 in->read_counter_frames = 0; 1418 in->output_standby = rsxadev->output_standby; 1419 in->dev = rsxadev; 1420 // Initialize the pipe. 1421 submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1422 DEFAULT_PIPE_PERIOD_COUNT, in, NULL); 1423#if LOG_STREAMS_TO_FILES 1424 in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1425 LOG_STREAM_FILE_PERMISSIONS); 1426 ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", 1427 strerror(errno)); 1428 ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); 1429#endif // LOG_STREAMS_TO_FILES 1430 // Return the input stream. 1431 *stream_in = &in->stream; 1432 1433 return 0; 1434} 1435 1436static void adev_close_input_stream(struct audio_hw_device *dev, 1437 struct audio_stream_in *stream) 1438{ 1439 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1440 ALOGV("adev_close_input_stream()"); 1441 submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), in, NULL); 1442#if LOG_STREAMS_TO_FILES 1443 if (in->log_fd >= 0) close(in->log_fd); 1444#endif // LOG_STREAMS_TO_FILES 1445#if ENABLE_LEGACY_INPUT_OPEN 1446 if (in->ref_count == 0) free(in); 1447#else 1448 free(in); 1449#endif // ENABLE_LEGACY_INPUT_OPEN 1450} 1451 1452static int adev_dump(const audio_hw_device_t *device, int fd) 1453{ 1454 (void)device; 1455 (void)fd; 1456 return 0; 1457} 1458 1459static int adev_close(hw_device_t *device) 1460{ 1461 ALOGI("adev_close()"); 1462 free(device); 1463 return 0; 1464} 1465 1466static int adev_open(const hw_module_t* module, const char* name, 1467 hw_device_t** device) 1468{ 1469 ALOGI("adev_open(name=%s)", name); 1470 struct submix_audio_device *rsxadev; 1471 1472 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 1473 return -EINVAL; 1474 1475 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 1476 if (!rsxadev) 1477 return -ENOMEM; 1478 1479 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 1480 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 1481 rsxadev->device.common.module = (struct hw_module_t *) module; 1482 rsxadev->device.common.close = adev_close; 1483 1484 rsxadev->device.init_check = adev_init_check; 1485 rsxadev->device.set_voice_volume = adev_set_voice_volume; 1486 rsxadev->device.set_master_volume = adev_set_master_volume; 1487 rsxadev->device.get_master_volume = adev_get_master_volume; 1488 rsxadev->device.set_master_mute = adev_set_master_mute; 1489 rsxadev->device.get_master_mute = adev_get_master_mute; 1490 rsxadev->device.set_mode = adev_set_mode; 1491 rsxadev->device.set_mic_mute = adev_set_mic_mute; 1492 rsxadev->device.get_mic_mute = adev_get_mic_mute; 1493 rsxadev->device.set_parameters = adev_set_parameters; 1494 rsxadev->device.get_parameters = adev_get_parameters; 1495 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 1496 rsxadev->device.open_output_stream = adev_open_output_stream; 1497 rsxadev->device.close_output_stream = adev_close_output_stream; 1498 rsxadev->device.open_input_stream = adev_open_input_stream; 1499 rsxadev->device.close_input_stream = adev_close_input_stream; 1500 rsxadev->device.dump = adev_dump; 1501 1502 rsxadev->input_standby = true; 1503 rsxadev->output_standby = true; 1504 1505 *device = &rsxadev->device.common; 1506 1507 return 0; 1508} 1509 1510static struct hw_module_methods_t hal_module_methods = { 1511 /* open */ adev_open, 1512}; 1513 1514struct audio_module HAL_MODULE_INFO_SYM = { 1515 /* common */ { 1516 /* tag */ HARDWARE_MODULE_TAG, 1517 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 1518 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 1519 /* id */ AUDIO_HARDWARE_MODULE_ID, 1520 /* name */ "Wifi Display audio HAL", 1521 /* author */ "The Android Open Source Project", 1522 /* methods */ &hal_module_methods, 1523 /* dso */ NULL, 1524 /* reserved */ { 0 }, 1525 }, 1526}; 1527 1528} //namespace android 1529 1530} //extern "C" 1531