audio_hw.cpp revision 7d973adff4c9b344b530dd7c585f789d02c605da
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// Compare an audio_config with input channel mask and an audio_config with output channel mask 311// returning false if they do *not* match, true otherwise. 312static bool audio_config_compare(const audio_config * const input_config, 313 const audio_config * const output_config) 314{ 315#if !ENABLE_CHANNEL_CONVERSION 316 const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask); 317 const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask); 318 if (input_channels != output_channels) { 319 ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d", 320 input_channels, output_channels); 321 return false; 322 } 323#endif // !ENABLE_CHANNEL_CONVERSION 324#if ENABLE_RESAMPLING 325 if (input_config->sample_rate != output_config->sample_rate && 326 audio_channel_count_from_in_mask(input_config->channel_mask) != 1) { 327#else 328 if (input_config->sample_rate != output_config->sample_rate) { 329#endif // ENABLE_RESAMPLING 330 ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", 331 input_config->sample_rate, output_config->sample_rate); 332 return false; 333 } 334 if (input_config->format != output_config->format) { 335 ALOGE("audio_config_compare() format mismatch %x vs. %x", 336 input_config->format, output_config->format); 337 return false; 338 } 339 // This purposely ignores offload_info as it's not required for the submix device. 340 return true; 341} 342 343// If one doesn't exist, create a pipe for the submix audio device rsxadev of size 344// buffer_size_frames and optionally associate "in" or "out" with the submix audio device. 345static void submix_audio_device_create_pipe(struct submix_audio_device * const rsxadev, 346 const struct audio_config * const config, 347 const size_t buffer_size_frames, 348 const uint32_t buffer_period_count, 349 struct submix_stream_in * const in, 350 struct submix_stream_out * const out) 351{ 352 ALOG_ASSERT(in || out); 353 ALOGV("submix_audio_device_create_pipe()"); 354 pthread_mutex_lock(&rsxadev->lock); 355 // Save a reference to the specified input or output stream and the associated channel 356 // mask. 357 if (in) { 358 rsxadev->input = in; 359 rsxadev->config.input_channel_mask = config->channel_mask; 360#if ENABLE_RESAMPLING 361 rsxadev->config.input_sample_rate = config->sample_rate; 362 // If the output isn't configured yet, set the output sample rate to the maximum supported 363 // sample rate such that the smallest possible input buffer is created. 364 if (!rsxadev->output) { 365 rsxadev->config.output_sample_rate = 48000; 366 } 367#endif // ENABLE_RESAMPLING 368 } 369 if (out) { 370 rsxadev->output = out; 371 rsxadev->config.output_channel_mask = config->channel_mask; 372#if ENABLE_RESAMPLING 373 rsxadev->config.output_sample_rate = config->sample_rate; 374#endif // ENABLE_RESAMPLING 375 } 376 // If a pipe isn't associated with the device, create one. 377 if (rsxadev->rsxSink == NULL || rsxadev->rsxSource == NULL) { 378 struct submix_config * const device_config = &rsxadev->config; 379 uint32_t channel_count; 380 if (out) 381 channel_count = audio_channel_count_from_out_mask(config->channel_mask); 382 else 383 channel_count = audio_channel_count_from_in_mask(config->channel_mask); 384#if ENABLE_CHANNEL_CONVERSION 385 // If channel conversion is enabled, allocate enough space for the maximum number of 386 // possible channels stored in the pipe for the situation when the number of channels in 387 // the output stream don't match the number in the input stream. 388 const uint32_t pipe_channel_count = max(channel_count, 2); 389#else 390 const uint32_t pipe_channel_count = channel_count; 391#endif // ENABLE_CHANNEL_CONVERSION 392 const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, 393 config->format); 394 const NBAIO_Format offers[1] = {format}; 395 size_t numCounterOffers = 0; 396 // Create a MonoPipe with optional blocking set to true. 397 MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); 398 // Negotiation between the source and sink cannot fail as the device open operation 399 // creates both ends of the pipe using the same audio format. 400 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 401 ALOG_ASSERT(index == 0); 402 MonoPipeReader* source = new MonoPipeReader(sink); 403 numCounterOffers = 0; 404 index = source->negotiate(offers, 1, NULL, numCounterOffers); 405 ALOG_ASSERT(index == 0); 406 ALOGV("submix_audio_device_create_pipe(): created pipe"); 407 408 // Save references to the source and sink. 409 ALOG_ASSERT(rsxadev->rsxSink == NULL); 410 ALOG_ASSERT(rsxadev->rsxSource == NULL); 411 rsxadev->rsxSink = sink; 412 rsxadev->rsxSource = source; 413 // Store the sanitized audio format in the device so that it's possible to determine 414 // the format of the pipe source when opening the input device. 415 memcpy(&device_config->common, config, sizeof(device_config->common)); 416 device_config->buffer_size_frames = sink->maxFrames(); 417 device_config->buffer_period_size_frames = device_config->buffer_size_frames / 418 buffer_period_count; 419 if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); 420 if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); 421#if ENABLE_CHANNEL_CONVERSION 422 // Calculate the pipe frame size based upon the number of channels. 423 device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / 424 channel_count; 425#endif // ENABLE_CHANNEL_CONVERSION 426 SUBMIX_ALOGV("submix_audio_device_create_pipe(): pipe frame size %zd, pipe size %zd, " 427 "period size %zd", device_config->pipe_frame_size, 428 device_config->buffer_size_frames, device_config->buffer_period_size_frames); 429 } 430 pthread_mutex_unlock(&rsxadev->lock); 431} 432 433// Release references to the sink and source. Input and output threads may maintain references 434// to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use 435// before they shutdown. 436static void submix_audio_device_release_pipe(struct submix_audio_device * const rsxadev) 437{ 438 ALOGV("submix_audio_device_release_pipe()"); 439 rsxadev->rsxSink.clear(); 440 rsxadev->rsxSource.clear(); 441} 442 443// Remove references to the specified input and output streams. When the device no longer 444// references input and output streams destroy the associated pipe. 445static void submix_audio_device_destroy_pipe(struct submix_audio_device * const rsxadev, 446 const struct submix_stream_in * const in, 447 const struct submix_stream_out * const out) 448{ 449 MonoPipe* sink; 450 pthread_mutex_lock(&rsxadev->lock); 451 ALOGV("submix_audio_device_destroy_pipe()"); 452 ALOG_ASSERT(in == NULL || rsxadev->input == in); 453 ALOG_ASSERT(out == NULL || rsxadev->output == out); 454 if (in != NULL) { 455#if ENABLE_LEGACY_INPUT_OPEN 456 const_cast<struct submix_stream_in*>(in)->ref_count--; 457 if (in->ref_count == 0) { 458 rsxadev->input = NULL; 459 } 460 ALOGV("submix_audio_device_destroy_pipe(): input ref_count %d", in->ref_count); 461#else 462 rsxadev->input = NULL; 463#endif // ENABLE_LEGACY_INPUT_OPEN 464 } 465 if (out != NULL) rsxadev->output = NULL; 466 if (rsxadev->input != NULL && rsxadev->output != NULL) { 467 submix_audio_device_release_pipe(rsxadev); 468 ALOGV("submix_audio_device_destroy_pipe(): pipe destroyed"); 469 } 470 pthread_mutex_unlock(&rsxadev->lock); 471} 472 473// Sanitize the user specified audio config for a submix input / output stream. 474static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) 475{ 476 config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : 477 get_supported_channel_out_mask(config->channel_mask); 478 config->sample_rate = get_supported_sample_rate(config->sample_rate); 479 config->format = DEFAULT_FORMAT; 480} 481 482// Verify a submix input or output stream can be opened. 483static bool submix_open_validate(const struct submix_audio_device * const rsxadev, 484 pthread_mutex_t * const lock, 485 const struct audio_config * const config, 486 const bool opening_input) 487{ 488 bool input_open; 489 bool output_open; 490 audio_config pipe_config; 491 492 // Query the device for the current audio config and whether input and output streams are open. 493 pthread_mutex_lock(lock); 494 output_open = rsxadev->output != NULL; 495 input_open = rsxadev->input != NULL; 496 memcpy(&pipe_config, &rsxadev->config.common, sizeof(pipe_config)); 497 pthread_mutex_unlock(lock); 498 499 // If the stream is already open, don't open it again. 500 if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { 501 ALOGE("submix_open_validate(): %s stream already open.", opening_input ? "Input" : 502 "Output"); 503 return false; 504 } 505 506 SUBMIX_ALOGV("submix_open_validate(): sample rate=%d format=%x " 507 "%s_channel_mask=%x", config->sample_rate, config->format, 508 opening_input ? "in" : "out", config->channel_mask); 509 510 // If either stream is open, verify the existing audio config the pipe matches the user 511 // specified config. 512 if (input_open || output_open) { 513 const audio_config * const input_config = opening_input ? config : &pipe_config; 514 const audio_config * const output_config = opening_input ? &pipe_config : config; 515 // Get the channel mask of the open device. 516 pipe_config.channel_mask = 517 opening_input ? rsxadev->config.output_channel_mask : 518 rsxadev->config.input_channel_mask; 519 if (!audio_config_compare(input_config, output_config)) { 520 ALOGE("submix_open_validate(): Unsupported format."); 521 return false; 522 } 523 } 524 return true; 525} 526 527// Calculate the maximum size of the pipe buffer in frames for the specified stream. 528static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, 529 const struct submix_config *config, 530 const size_t pipe_frames, 531 const size_t stream_frame_size) 532{ 533 const size_t pipe_frame_size = config->pipe_frame_size; 534 const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); 535 return (pipe_frames * config->pipe_frame_size) / max_frame_size; 536} 537 538/* audio HAL functions */ 539 540static uint32_t out_get_sample_rate(const struct audio_stream *stream) 541{ 542 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 543 const_cast<struct audio_stream *>(stream)); 544#if ENABLE_RESAMPLING 545 const uint32_t out_rate = out->dev->config.output_sample_rate; 546#else 547 const uint32_t out_rate = out->dev->config.common.sample_rate; 548#endif // ENABLE_RESAMPLING 549 SUBMIX_ALOGV("out_get_sample_rate() returns %u", out_rate); 550 return out_rate; 551} 552 553static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 554{ 555 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 556#if ENABLE_RESAMPLING 557 // The sample rate of the stream can't be changed once it's set since this would change the 558 // output buffer size and hence break playback to the shared pipe. 559 if (rate != out->dev->config.output_sample_rate) { 560 ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " 561 "%u to %u", out->dev->config.output_sample_rate, rate); 562 return -ENOSYS; 563 } 564#endif // ENABLE_RESAMPLING 565 if (!sample_rate_supported(rate)) { 566 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 567 return -ENOSYS; 568 } 569 SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); 570 out->dev->config.common.sample_rate = rate; 571 return 0; 572} 573 574static size_t out_get_buffer_size(const struct audio_stream *stream) 575{ 576 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 577 const_cast<struct audio_stream *>(stream)); 578 const struct submix_config * const config = &out->dev->config; 579 const size_t stream_frame_size = 580 audio_stream_out_frame_size((const struct audio_stream_out *)stream); 581 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 582 stream, config, config->buffer_period_size_frames, stream_frame_size); 583 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 584 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", 585 buffer_size_bytes, buffer_size_frames); 586 return buffer_size_bytes; 587} 588 589static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 590{ 591 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 592 const_cast<struct audio_stream *>(stream)); 593 uint32_t channel_mask = out->dev->config.output_channel_mask; 594 SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); 595 return channel_mask; 596} 597 598static audio_format_t out_get_format(const struct audio_stream *stream) 599{ 600 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 601 const_cast<struct audio_stream *>(stream)); 602 const audio_format_t format = out->dev->config.common.format; 603 SUBMIX_ALOGV("out_get_format() returns %x", format); 604 return format; 605} 606 607static int out_set_format(struct audio_stream *stream, audio_format_t format) 608{ 609 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 610 if (format != out->dev->config.common.format) { 611 ALOGE("out_set_format(format=%x) format unsupported", format); 612 return -ENOSYS; 613 } 614 SUBMIX_ALOGV("out_set_format(format=%x)", format); 615 return 0; 616} 617 618static int out_standby(struct audio_stream *stream) 619{ 620 struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_out(stream)->dev; 621 ALOGI("out_standby()"); 622 623 pthread_mutex_lock(&rsxadev->lock); 624 625 rsxadev->output_standby = true; 626 627 pthread_mutex_unlock(&rsxadev->lock); 628 629 return 0; 630} 631 632static int out_dump(const struct audio_stream *stream, int fd) 633{ 634 (void)stream; 635 (void)fd; 636 return 0; 637} 638 639static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 640{ 641 int exiting = -1; 642 AudioParameter parms = AudioParameter(String8(kvpairs)); 643 SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs); 644 645 // FIXME this is using hard-coded strings but in the future, this functionality will be 646 // converted to use audio HAL extensions required to support tunneling 647 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { 648 struct submix_audio_device * const rsxadev = 649 audio_stream_get_submix_stream_out(stream)->dev; 650 pthread_mutex_lock(&rsxadev->lock); 651 { // using the sink 652 sp<MonoPipe> sink = rsxadev->rsxSink; 653 if (sink == NULL) { 654 pthread_mutex_unlock(&rsxadev->lock); 655 return 0; 656 } 657 658 ALOGI("out_set_parameters(): shutdown"); 659 sink->shutdown(true); 660 } // done using the sink 661 pthread_mutex_unlock(&rsxadev->lock); 662 } 663 return 0; 664} 665 666static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 667{ 668 (void)stream; 669 (void)keys; 670 return strdup(""); 671} 672 673static uint32_t out_get_latency(const struct audio_stream_out *stream) 674{ 675 const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( 676 const_cast<struct audio_stream_out *>(stream)); 677 const struct submix_config * const config = &out->dev->config; 678 const size_t stream_frame_size = 679 audio_stream_out_frame_size(stream); 680 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 681 &stream->common, config, config->buffer_size_frames, stream_frame_size); 682 const uint32_t sample_rate = out_get_sample_rate(&stream->common); 683 const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate; 684 SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", 685 latency_ms, buffer_size_frames, sample_rate); 686 return latency_ms; 687} 688 689static int out_set_volume(struct audio_stream_out *stream, float left, 690 float right) 691{ 692 (void)stream; 693 (void)left; 694 (void)right; 695 return -ENOSYS; 696} 697 698static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 699 size_t bytes) 700{ 701 SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); 702 ssize_t written_frames = 0; 703 const size_t frame_size = audio_stream_out_frame_size(stream); 704 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 705 struct submix_audio_device * const rsxadev = out->dev; 706 const size_t frames = bytes / frame_size; 707 708 pthread_mutex_lock(&rsxadev->lock); 709 710 rsxadev->output_standby = false; 711 712 sp<MonoPipe> sink = rsxadev->rsxSink; 713 if (sink != NULL) { 714 if (sink->isShutdown()) { 715 sink.clear(); 716 pthread_mutex_unlock(&rsxadev->lock); 717 SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); 718 // the pipe has already been shutdown, this buffer will be lost but we must 719 // simulate timing so we don't drain the output faster than realtime 720 usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); 721 return bytes; 722 } 723 } else { 724 pthread_mutex_unlock(&rsxadev->lock); 725 ALOGE("out_write without a pipe!"); 726 ALOG_ASSERT("out_write without a pipe!"); 727 return 0; 728 } 729 730 // If the write to the sink would block when no input stream is present, flush enough frames 731 // from the pipe to make space to write the most recent data. 732 { 733 const size_t availableToWrite = sink->availableToWrite(); 734 sp<MonoPipeReader> source = rsxadev->rsxSource; 735 if (rsxadev->input == NULL && availableToWrite < frames) { 736 static uint8_t flush_buffer[64]; 737 const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; 738 size_t frames_to_flush_from_source = frames - availableToWrite; 739 SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", 740 frames_to_flush_from_source); 741 while (frames_to_flush_from_source) { 742 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); 743 frames_to_flush_from_source -= flush_size; 744 source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); 745 } 746 } 747 } 748 749 pthread_mutex_unlock(&rsxadev->lock); 750 751 written_frames = sink->write(buffer, frames); 752 753#if LOG_STREAMS_TO_FILES 754 if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); 755#endif // LOG_STREAMS_TO_FILES 756 757 if (written_frames < 0) { 758 if (written_frames == (ssize_t)NEGOTIATE) { 759 ALOGE("out_write() write to pipe returned NEGOTIATE"); 760 761 pthread_mutex_lock(&rsxadev->lock); 762 sink.clear(); 763 pthread_mutex_unlock(&rsxadev->lock); 764 765 written_frames = 0; 766 return 0; 767 } else { 768 // write() returned UNDERRUN or WOULD_BLOCK, retry 769 ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); 770 written_frames = sink->write(buffer, frames); 771 } 772 } 773 774 pthread_mutex_lock(&rsxadev->lock); 775 sink.clear(); 776 pthread_mutex_unlock(&rsxadev->lock); 777 778 if (written_frames < 0) { 779 ALOGE("out_write() failed writing to pipe with %zd", written_frames); 780 return 0; 781 } 782 const ssize_t written_bytes = written_frames * frame_size; 783 SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); 784 return written_bytes; 785} 786 787static int out_get_render_position(const struct audio_stream_out *stream, 788 uint32_t *dsp_frames) 789{ 790 (void)stream; 791 (void)dsp_frames; 792 return -EINVAL; 793} 794 795static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 796{ 797 (void)stream; 798 (void)effect; 799 return 0; 800} 801 802static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 803{ 804 (void)stream; 805 (void)effect; 806 return 0; 807} 808 809static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 810 int64_t *timestamp) 811{ 812 (void)stream; 813 (void)timestamp; 814 return -EINVAL; 815} 816 817/** audio_stream_in implementation **/ 818static uint32_t in_get_sample_rate(const struct audio_stream *stream) 819{ 820 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 821 const_cast<struct audio_stream*>(stream)); 822#if ENABLE_RESAMPLING 823 const uint32_t rate = in->dev->config.input_sample_rate; 824#else 825 const uint32_t rate = in->dev->config.common.sample_rate; 826#endif // ENABLE_RESAMPLING 827 SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); 828 return rate; 829} 830 831static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 832{ 833 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 834#if ENABLE_RESAMPLING 835 // The sample rate of the stream can't be changed once it's set since this would change the 836 // input buffer size and hence break recording from the shared pipe. 837 if (rate != in->dev->config.input_sample_rate) { 838 ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " 839 "%u to %u", in->dev->config.input_sample_rate, rate); 840 return -ENOSYS; 841 } 842#endif // ENABLE_RESAMPLING 843 if (!sample_rate_supported(rate)) { 844 ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); 845 return -ENOSYS; 846 } 847 in->dev->config.common.sample_rate = rate; 848 SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); 849 return 0; 850} 851 852static size_t in_get_buffer_size(const struct audio_stream *stream) 853{ 854 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 855 const_cast<struct audio_stream*>(stream)); 856 const struct submix_config * const config = &in->dev->config; 857 const size_t stream_frame_size = 858 audio_stream_in_frame_size((const struct audio_stream_in *)stream); 859 size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 860 stream, config, config->buffer_period_size_frames, stream_frame_size); 861#if ENABLE_RESAMPLING 862 // Scale the size of the buffer based upon the maximum number of frames that could be returned 863 // given the ratio of output to input sample rate. 864 buffer_size_frames = (size_t)(((float)buffer_size_frames * 865 (float)config->input_sample_rate) / 866 (float)config->output_sample_rate); 867#endif // ENABLE_RESAMPLING 868 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 869 SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, 870 buffer_size_frames); 871 return buffer_size_bytes; 872} 873 874static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 875{ 876 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 877 const_cast<struct audio_stream*>(stream)); 878 const audio_channel_mask_t channel_mask = in->dev->config.input_channel_mask; 879 SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); 880 return channel_mask; 881} 882 883static audio_format_t in_get_format(const struct audio_stream *stream) 884{ 885 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 886 const_cast<struct audio_stream*>(stream)); 887 const audio_format_t format = in->dev->config.common.format; 888 SUBMIX_ALOGV("in_get_format() returns %x", format); 889 return format; 890} 891 892static int in_set_format(struct audio_stream *stream, audio_format_t format) 893{ 894 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 895 if (format != in->dev->config.common.format) { 896 ALOGE("in_set_format(format=%x) format unsupported", format); 897 return -ENOSYS; 898 } 899 SUBMIX_ALOGV("in_set_format(format=%x)", format); 900 return 0; 901} 902 903static int in_standby(struct audio_stream *stream) 904{ 905 struct submix_audio_device * const rsxadev = audio_stream_get_submix_stream_in(stream)->dev; 906 ALOGI("in_standby()"); 907 908 pthread_mutex_lock(&rsxadev->lock); 909 910 rsxadev->input_standby = true; 911 912 pthread_mutex_unlock(&rsxadev->lock); 913 914 return 0; 915} 916 917static int in_dump(const struct audio_stream *stream, int fd) 918{ 919 (void)stream; 920 (void)fd; 921 return 0; 922} 923 924static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 925{ 926 (void)stream; 927 (void)kvpairs; 928 return 0; 929} 930 931static char * in_get_parameters(const struct audio_stream *stream, 932 const char *keys) 933{ 934 (void)stream; 935 (void)keys; 936 return strdup(""); 937} 938 939static int in_set_gain(struct audio_stream_in *stream, float gain) 940{ 941 (void)stream; 942 (void)gain; 943 return 0; 944} 945 946static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 947 size_t bytes) 948{ 949 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 950 struct submix_audio_device * const rsxadev = in->dev; 951 struct audio_config *format; 952 const size_t frame_size = audio_stream_in_frame_size(stream); 953 const size_t frames_to_read = bytes / frame_size; 954 955 SUBMIX_ALOGV("in_read bytes=%zu", bytes); 956 pthread_mutex_lock(&rsxadev->lock); 957 958 const bool output_standby_transition = (in->output_standby != in->dev->output_standby); 959 in->output_standby = rsxadev->output_standby; 960 961 if (rsxadev->input_standby || output_standby_transition) { 962 rsxadev->input_standby = false; 963 // keep track of when we exit input standby (== first read == start "real recording") 964 // or when we start recording silence, and reset projected time 965 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 966 if (rc == 0) { 967 in->read_counter_frames = 0; 968 } 969 } 970 971 in->read_counter_frames += frames_to_read; 972 size_t remaining_frames = frames_to_read; 973 974 { 975 // about to read from audio source 976 sp<MonoPipeReader> source = rsxadev->rsxSource; 977 if (source == NULL) { 978 ALOGE("no audio pipe yet we're trying to read!"); 979 pthread_mutex_unlock(&rsxadev->lock); 980 usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); 981 memset(buffer, 0, bytes); 982 return bytes; 983 } 984 985 pthread_mutex_unlock(&rsxadev->lock); 986 987 // read the data from the pipe (it's non blocking) 988 int attempts = 0; 989 char* buff = (char*)buffer; 990#if ENABLE_CHANNEL_CONVERSION 991 // Determine whether channel conversion is required. 992 const uint32_t input_channels = audio_channel_count_from_in_mask( 993 rsxadev->config.input_channel_mask); 994 const uint32_t output_channels = audio_channel_count_from_out_mask( 995 rsxadev->config.output_channel_mask); 996 if (input_channels != output_channels) { 997 SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " 998 "input channels", output_channels, input_channels); 999 // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. 1000 ALOG_ASSERT(rsxadev->config.common.format == AUDIO_FORMAT_PCM_16_BIT); 1001 ALOG_ASSERT((input_channels == 1 && output_channels == 2) || 1002 (input_channels == 2 && output_channels == 1)); 1003 } 1004#endif // ENABLE_CHANNEL_CONVERSION 1005 1006#if ENABLE_RESAMPLING 1007 const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); 1008 const uint32_t output_sample_rate = rsxadev->config.output_sample_rate; 1009 const size_t resampler_buffer_size_frames = 1010 sizeof(rsxadev->resampler_buffer) / sizeof(rsxadev->resampler_buffer[0]); 1011 float resampler_ratio = 1.0f; 1012 // Determine whether resampling is required. 1013 if (input_sample_rate != output_sample_rate) { 1014 resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; 1015 // Only support 16-bit PCM mono resampling. 1016 // NOTE: Resampling is performed after the channel conversion step. 1017 ALOG_ASSERT(rsxadev->config.common.format == AUDIO_FORMAT_PCM_16_BIT); 1018 ALOG_ASSERT(audio_channel_count_from_in_mask(rsxadev->config.input_channel_mask) == 1); 1019 } 1020#endif // ENABLE_RESAMPLING 1021 1022 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 1023 ssize_t frames_read = -1977; 1024 size_t read_frames = remaining_frames; 1025#if ENABLE_RESAMPLING 1026 char* const saved_buff = buff; 1027 if (resampler_ratio != 1.0f) { 1028 // Calculate the number of frames from the pipe that need to be read to generate 1029 // the data for the input stream read. 1030 const size_t frames_required_for_resampler = (size_t)( 1031 (float)read_frames * (float)resampler_ratio); 1032 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); 1033 // Read into the resampler buffer. 1034 buff = (char*)rsxadev->resampler_buffer; 1035 } 1036#endif // ENABLE_RESAMPLING 1037#if ENABLE_CHANNEL_CONVERSION 1038 if (output_channels == 1 && input_channels == 2) { 1039 // Need to read half the requested frames since the converted output 1040 // data will take twice the space (mono->stereo). 1041 read_frames /= 2; 1042 } 1043#endif // ENABLE_CHANNEL_CONVERSION 1044 1045 SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); 1046 1047 frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS); 1048 1049 SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); 1050 1051#if ENABLE_CHANNEL_CONVERSION 1052 // Perform in-place channel conversion. 1053 // NOTE: In the following "input stream" refers to the data returned by this function 1054 // and "output stream" refers to the data read from the pipe. 1055 if (input_channels != output_channels && frames_read > 0) { 1056 int16_t *data = (int16_t*)buff; 1057 if (output_channels == 2 && input_channels == 1) { 1058 // Offset into the output stream data in samples. 1059 ssize_t output_stream_offset = 0; 1060 for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; 1061 input_stream_frame++, output_stream_offset += 2) { 1062 // Average the content from both channels. 1063 data[input_stream_frame] = ((int32_t)data[output_stream_offset] + 1064 (int32_t)data[output_stream_offset + 1]) / 2; 1065 } 1066 } else if (output_channels == 1 && input_channels == 2) { 1067 // Offset into the input stream data in samples. 1068 ssize_t input_stream_offset = (frames_read - 1) * 2; 1069 for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; 1070 output_stream_frame--, input_stream_offset -= 2) { 1071 const short sample = data[output_stream_frame]; 1072 data[input_stream_offset] = sample; 1073 data[input_stream_offset + 1] = sample; 1074 } 1075 } 1076 } 1077#endif // ENABLE_CHANNEL_CONVERSION 1078 1079#if ENABLE_RESAMPLING 1080 if (resampler_ratio != 1.0f) { 1081 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); 1082 const int16_t * const data = (int16_t*)buff; 1083 int16_t * const resampled_buffer = (int16_t*)saved_buff; 1084 // Resample with *no* filtering - if the data from the ouptut stream was really 1085 // sampled at a different rate this will result in very nasty aliasing. 1086 const float output_stream_frames = (float)frames_read; 1087 size_t input_stream_frame = 0; 1088 for (float output_stream_frame = 0.0f; 1089 output_stream_frame < output_stream_frames && 1090 input_stream_frame < remaining_frames; 1091 output_stream_frame += resampler_ratio, input_stream_frame++) { 1092 resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; 1093 } 1094 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); 1095 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); 1096 frames_read = input_stream_frame; 1097 buff = saved_buff; 1098 } 1099#endif // ENABLE_RESAMPLING 1100 1101 if (frames_read > 0) { 1102#if LOG_STREAMS_TO_FILES 1103 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); 1104#endif // LOG_STREAMS_TO_FILES 1105 1106 remaining_frames -= frames_read; 1107 buff += frames_read * frame_size; 1108 SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", 1109 attempts, frames_read, remaining_frames); 1110 } else { 1111 attempts++; 1112 SUBMIX_ALOGE(" in_read read returned %zd", frames_read); 1113 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 1114 } 1115 } 1116 // done using the source 1117 pthread_mutex_lock(&rsxadev->lock); 1118 source.clear(); 1119 pthread_mutex_unlock(&rsxadev->lock); 1120 } 1121 1122 if (remaining_frames > 0) { 1123 const size_t remaining_bytes = remaining_frames * frame_size; 1124 SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames); 1125 memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); 1126 } 1127 1128 // compute how much we need to sleep after reading the data by comparing the wall clock with 1129 // the projected time at which we should return. 1130 struct timespec time_after_read;// wall clock after reading from the pipe 1131 struct timespec record_duration;// observed record duration 1132 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 1133 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 1134 if (rc == 0) { 1135 // for how long have we been recording? 1136 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 1137 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 1138 if (record_duration.tv_nsec < 0) { 1139 record_duration.tv_sec--; 1140 record_duration.tv_nsec += 1000000000; 1141 } 1142 1143 // read_counter_frames contains the number of frames that have been read since the 1144 // beginning of recording (including this call): it's converted to usec and compared to 1145 // how long we've been recording for, which gives us how long we must wait to sync the 1146 // projected recording time, and the observed recording time. 1147 long projected_vs_observed_offset_us = 1148 ((int64_t)(in->read_counter_frames 1149 - (record_duration.tv_sec*sample_rate))) 1150 * 1000000 / sample_rate 1151 - (record_duration.tv_nsec / 1000); 1152 1153 SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 1154 record_duration.tv_sec, record_duration.tv_nsec/1000000, 1155 projected_vs_observed_offset_us); 1156 if (projected_vs_observed_offset_us > 0) { 1157 usleep(projected_vs_observed_offset_us); 1158 } 1159 } 1160 1161 SUBMIX_ALOGV("in_read returns %zu", bytes); 1162 return bytes; 1163 1164} 1165 1166static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 1167{ 1168 (void)stream; 1169 return 0; 1170} 1171 1172static int in_add_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 in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1180{ 1181 (void)stream; 1182 (void)effect; 1183 return 0; 1184} 1185 1186static int adev_open_output_stream(struct audio_hw_device *dev, 1187 audio_io_handle_t handle, 1188 audio_devices_t devices, 1189 audio_output_flags_t flags, 1190 struct audio_config *config, 1191 struct audio_stream_out **stream_out) 1192{ 1193 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1194 ALOGV("adev_open_output_stream()"); 1195 struct submix_stream_out *out; 1196 bool force_pipe_creation = false; 1197 (void)handle; 1198 (void)devices; 1199 (void)flags; 1200 1201 *stream_out = NULL; 1202 1203 // Make sure it's possible to open the device given the current audio config. 1204 submix_sanitize_config(config, false); 1205 if (!submix_open_validate(rsxadev, &rsxadev->lock, config, false)) { 1206 ALOGE("adev_open_output_stream(): Unable to open output stream."); 1207 return -EINVAL; 1208 } 1209 1210 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 1211 if (!out) return -ENOMEM; 1212 1213 // Initialize the function pointer tables (v-tables). 1214 out->stream.common.get_sample_rate = out_get_sample_rate; 1215 out->stream.common.set_sample_rate = out_set_sample_rate; 1216 out->stream.common.get_buffer_size = out_get_buffer_size; 1217 out->stream.common.get_channels = out_get_channels; 1218 out->stream.common.get_format = out_get_format; 1219 out->stream.common.set_format = out_set_format; 1220 out->stream.common.standby = out_standby; 1221 out->stream.common.dump = out_dump; 1222 out->stream.common.set_parameters = out_set_parameters; 1223 out->stream.common.get_parameters = out_get_parameters; 1224 out->stream.common.add_audio_effect = out_add_audio_effect; 1225 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1226 out->stream.get_latency = out_get_latency; 1227 out->stream.set_volume = out_set_volume; 1228 out->stream.write = out_write; 1229 out->stream.get_render_position = out_get_render_position; 1230 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1231 1232#if ENABLE_RESAMPLING 1233 // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits 1234 // writes correctly. 1235 force_pipe_creation = rsxadev->config.common.sample_rate != config->sample_rate; 1236#endif // ENABLE_RESAMPLING 1237 1238 // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so 1239 // that it's recreated. 1240 pthread_mutex_lock(&rsxadev->lock); 1241 if ((rsxadev->rsxSink != NULL && rsxadev->rsxSink->isShutdown()) || force_pipe_creation) { 1242 submix_audio_device_release_pipe(rsxadev); 1243 } 1244 pthread_mutex_unlock(&rsxadev->lock); 1245 1246 // Store a pointer to the device from the output stream. 1247 out->dev = rsxadev; 1248 // Initialize the pipe. 1249 ALOGV("adev_open_output_stream(): Initializing pipe"); 1250 submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1251 DEFAULT_PIPE_PERIOD_COUNT, NULL, out); 1252#if LOG_STREAMS_TO_FILES 1253 out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1254 LOG_STREAM_FILE_PERMISSIONS); 1255 ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", 1256 strerror(errno)); 1257 ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); 1258#endif // LOG_STREAMS_TO_FILES 1259 // Return the output stream. 1260 *stream_out = &out->stream; 1261 1262 return 0; 1263} 1264 1265static void adev_close_output_stream(struct audio_hw_device *dev, 1266 struct audio_stream_out *stream) 1267{ 1268 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 1269 ALOGV("adev_close_output_stream()"); 1270 submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), NULL, out); 1271#if LOG_STREAMS_TO_FILES 1272 if (out->log_fd >= 0) close(out->log_fd); 1273#endif // LOG_STREAMS_TO_FILES 1274 free(out); 1275} 1276 1277static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1278{ 1279 (void)dev; 1280 (void)kvpairs; 1281 return -ENOSYS; 1282} 1283 1284static char * adev_get_parameters(const struct audio_hw_device *dev, 1285 const char *keys) 1286{ 1287 (void)dev; 1288 (void)keys; 1289 return strdup("");; 1290} 1291 1292static int adev_init_check(const struct audio_hw_device *dev) 1293{ 1294 ALOGI("adev_init_check()"); 1295 (void)dev; 1296 return 0; 1297} 1298 1299static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 1300{ 1301 (void)dev; 1302 (void)volume; 1303 return -ENOSYS; 1304} 1305 1306static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 1307{ 1308 (void)dev; 1309 (void)volume; 1310 return -ENOSYS; 1311} 1312 1313static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 1314{ 1315 (void)dev; 1316 (void)volume; 1317 return -ENOSYS; 1318} 1319 1320static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 1321{ 1322 (void)dev; 1323 (void)muted; 1324 return -ENOSYS; 1325} 1326 1327static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 1328{ 1329 (void)dev; 1330 (void)muted; 1331 return -ENOSYS; 1332} 1333 1334static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 1335{ 1336 (void)dev; 1337 (void)mode; 1338 return 0; 1339} 1340 1341static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 1342{ 1343 (void)dev; 1344 (void)state; 1345 return -ENOSYS; 1346} 1347 1348static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 1349{ 1350 (void)dev; 1351 (void)state; 1352 return -ENOSYS; 1353} 1354 1355static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 1356 const struct audio_config *config) 1357{ 1358 if (audio_is_linear_pcm(config->format)) { 1359 const size_t buffer_period_size_frames = 1360 audio_hw_device_get_submix_audio_device(const_cast<struct audio_hw_device*>(dev))-> 1361 config.buffer_period_size_frames; 1362 const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * 1363 audio_bytes_per_sample(config->format); 1364 const size_t buffer_size = buffer_period_size_frames * frame_size_in_bytes; 1365 SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", 1366 buffer_size, buffer_period_size_frames); 1367 return buffer_size; 1368 } 1369 return 0; 1370} 1371 1372static int adev_open_input_stream(struct audio_hw_device *dev, 1373 audio_io_handle_t handle, 1374 audio_devices_t devices, 1375 struct audio_config *config, 1376 struct audio_stream_in **stream_in, 1377 audio_input_flags_t flags __unused) 1378{ 1379 struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); 1380 struct submix_stream_in *in; 1381 ALOGI("adev_open_input_stream()"); 1382 (void)handle; 1383 (void)devices; 1384 1385 *stream_in = NULL; 1386 1387 // Make sure it's possible to open the device given the current audio config. 1388 submix_sanitize_config(config, true); 1389 if (!submix_open_validate(rsxadev, &rsxadev->lock, config, true)) { 1390 ALOGE("adev_open_input_stream(): Unable to open input stream."); 1391 return -EINVAL; 1392 } 1393 1394#if ENABLE_LEGACY_INPUT_OPEN 1395 pthread_mutex_lock(&rsxadev->lock); 1396 in = rsxadev->input; 1397 if (in) { 1398 in->ref_count++; 1399 sp<MonoPipe> sink = rsxadev->rsxSink; 1400 ALOG_ASSERT(sink != NULL); 1401 // If the sink has been shutdown, delete the pipe. 1402 if (sink->isShutdown()) submix_audio_device_release_pipe(rsxadev); 1403 } 1404 pthread_mutex_unlock(&rsxadev->lock); 1405#else 1406 in = NULL; 1407#endif // ENABLE_LEGACY_INPUT_OPEN 1408 1409 if (!in) { 1410 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 1411 if (!in) return -ENOMEM; 1412 in->ref_count = 1; 1413 1414 // Initialize the function pointer tables (v-tables). 1415 in->stream.common.get_sample_rate = in_get_sample_rate; 1416 in->stream.common.set_sample_rate = in_set_sample_rate; 1417 in->stream.common.get_buffer_size = in_get_buffer_size; 1418 in->stream.common.get_channels = in_get_channels; 1419 in->stream.common.get_format = in_get_format; 1420 in->stream.common.set_format = in_set_format; 1421 in->stream.common.standby = in_standby; 1422 in->stream.common.dump = in_dump; 1423 in->stream.common.set_parameters = in_set_parameters; 1424 in->stream.common.get_parameters = in_get_parameters; 1425 in->stream.common.add_audio_effect = in_add_audio_effect; 1426 in->stream.common.remove_audio_effect = in_remove_audio_effect; 1427 in->stream.set_gain = in_set_gain; 1428 in->stream.read = in_read; 1429 in->stream.get_input_frames_lost = in_get_input_frames_lost; 1430 } 1431 1432 // Initialize the input stream. 1433 in->read_counter_frames = 0; 1434 in->output_standby = rsxadev->output_standby; 1435 in->dev = rsxadev; 1436 // Initialize the pipe. 1437 submix_audio_device_create_pipe(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1438 DEFAULT_PIPE_PERIOD_COUNT, in, NULL); 1439#if LOG_STREAMS_TO_FILES 1440 in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1441 LOG_STREAM_FILE_PERMISSIONS); 1442 ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", 1443 strerror(errno)); 1444 ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); 1445#endif // LOG_STREAMS_TO_FILES 1446 // Return the input stream. 1447 *stream_in = &in->stream; 1448 1449 return 0; 1450} 1451 1452static void adev_close_input_stream(struct audio_hw_device *dev, 1453 struct audio_stream_in *stream) 1454{ 1455 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1456 ALOGV("adev_close_input_stream()"); 1457 submix_audio_device_destroy_pipe(audio_hw_device_get_submix_audio_device(dev), in, NULL); 1458#if LOG_STREAMS_TO_FILES 1459 if (in->log_fd >= 0) close(in->log_fd); 1460#endif // LOG_STREAMS_TO_FILES 1461#if ENABLE_LEGACY_INPUT_OPEN 1462 if (in->ref_count == 0) free(in); 1463#else 1464 free(in); 1465#endif // ENABLE_LEGACY_INPUT_OPEN 1466} 1467 1468static int adev_dump(const audio_hw_device_t *device, int fd) 1469{ 1470 (void)device; 1471 (void)fd; 1472 return 0; 1473} 1474 1475static int adev_close(hw_device_t *device) 1476{ 1477 ALOGI("adev_close()"); 1478 free(device); 1479 return 0; 1480} 1481 1482static int adev_open(const hw_module_t* module, const char* name, 1483 hw_device_t** device) 1484{ 1485 ALOGI("adev_open(name=%s)", name); 1486 struct submix_audio_device *rsxadev; 1487 1488 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 1489 return -EINVAL; 1490 1491 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 1492 if (!rsxadev) 1493 return -ENOMEM; 1494 1495 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 1496 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 1497 rsxadev->device.common.module = (struct hw_module_t *) module; 1498 rsxadev->device.common.close = adev_close; 1499 1500 rsxadev->device.init_check = adev_init_check; 1501 rsxadev->device.set_voice_volume = adev_set_voice_volume; 1502 rsxadev->device.set_master_volume = adev_set_master_volume; 1503 rsxadev->device.get_master_volume = adev_get_master_volume; 1504 rsxadev->device.set_master_mute = adev_set_master_mute; 1505 rsxadev->device.get_master_mute = adev_get_master_mute; 1506 rsxadev->device.set_mode = adev_set_mode; 1507 rsxadev->device.set_mic_mute = adev_set_mic_mute; 1508 rsxadev->device.get_mic_mute = adev_get_mic_mute; 1509 rsxadev->device.set_parameters = adev_set_parameters; 1510 rsxadev->device.get_parameters = adev_get_parameters; 1511 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 1512 rsxadev->device.open_output_stream = adev_open_output_stream; 1513 rsxadev->device.close_output_stream = adev_close_output_stream; 1514 rsxadev->device.open_input_stream = adev_open_input_stream; 1515 rsxadev->device.close_input_stream = adev_close_input_stream; 1516 rsxadev->device.dump = adev_dump; 1517 1518 rsxadev->input_standby = true; 1519 rsxadev->output_standby = true; 1520 1521 *device = &rsxadev->device.common; 1522 1523 return 0; 1524} 1525 1526static struct hw_module_methods_t hal_module_methods = { 1527 /* open */ adev_open, 1528}; 1529 1530struct audio_module HAL_MODULE_INFO_SYM = { 1531 /* common */ { 1532 /* tag */ HARDWARE_MODULE_TAG, 1533 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 1534 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 1535 /* id */ AUDIO_HARDWARE_MODULE_ID, 1536 /* name */ "Wifi Display audio HAL", 1537 /* author */ "The Android Open Source Project", 1538 /* methods */ &hal_module_methods, 1539 /* dso */ NULL, 1540 /* reserved */ { 0 }, 1541 }, 1542}; 1543 1544} //namespace android 1545 1546} //extern "C" 1547