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/compiler.h> 29#include <cutils/log.h> 30#include <cutils/properties.h> 31#include <cutils/str_parms.h> 32 33#include <hardware/audio.h> 34#include <hardware/hardware.h> 35#include <system/audio.h> 36 37#include <media/AudioParameter.h> 38#include <media/AudioBufferProvider.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41 42#include <utils/String8.h> 43 44#define LOG_STREAMS_TO_FILES 0 45#if LOG_STREAMS_TO_FILES 46#include <fcntl.h> 47#include <stdio.h> 48#include <sys/stat.h> 49#endif // LOG_STREAMS_TO_FILES 50 51extern "C" { 52 53namespace android { 54 55// Set to 1 to enable extremely verbose logging in this module. 56#define SUBMIX_VERBOSE_LOGGING 0 57#if SUBMIX_VERBOSE_LOGGING 58#define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__) 59#define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__) 60#else 61#define SUBMIX_ALOGV(...) 62#define SUBMIX_ALOGE(...) 63#endif // SUBMIX_VERBOSE_LOGGING 64 65// NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe(). 66#define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*4) 67// Value used to divide the MonoPipe() buffer into segments that are written to the source and 68// read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer 69// the minimum latency is the MonoPipe buffer size divided by this value. 70#define DEFAULT_PIPE_PERIOD_COUNT 4 71// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to 72// the duration of a record buffer at the current record sample rate (of the device, not of 73// the recording itself). Here we have: 74// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms 75#define MAX_READ_ATTEMPTS 3 76#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty 77#define DEFAULT_SAMPLE_RATE_HZ 48000 // default sample rate 78// See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h. 79#define DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT 80// A legacy user of this device does not close the input stream when it shuts down, which 81// results in the application opening a new input stream before closing the old input stream 82// handle it was previously using. Setting this value to 1 allows multiple clients to open 83// multiple input streams from this device. If this option is enabled, each input stream returned 84// is *the same stream* which means that readers will race to read data from these streams. 85#define ENABLE_LEGACY_INPUT_OPEN 1 86// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled. 87#define ENABLE_CHANNEL_CONVERSION 1 88// Whether resampling is enabled. 89#define ENABLE_RESAMPLING 1 90#if LOG_STREAMS_TO_FILES 91// Folder to save stream log files to. 92#define LOG_STREAM_FOLDER "/data/misc/audioserver" 93// Log filenames for input and output streams. 94#define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw" 95#define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw" 96// File permissions for stream log files. 97#define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH) 98#endif // LOG_STREAMS_TO_FILES 99// limit for number of read error log entries to avoid spamming the logs 100#define MAX_READ_ERROR_LOGS 5 101 102// Common limits macros. 103#ifndef min 104#define min(a, b) ((a) < (b) ? (a) : (b)) 105#endif // min 106#ifndef max 107#define max(a, b) ((a) > (b) ? (a) : (b)) 108#endif // max 109 110// Set *result_variable_ptr to true if value_to_find is present in the array array_to_search, 111// otherwise set *result_variable_ptr to false. 112#define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \ 113 { \ 114 size_t i; \ 115 *(result_variable_ptr) = false; \ 116 for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \ 117 if ((value_to_find) == (array_to_search)[i]) { \ 118 *(result_variable_ptr) = true; \ 119 break; \ 120 } \ 121 } \ 122 } 123 124// Configuration of the submix pipe. 125struct submix_config { 126 // Channel mask field in this data structure is set to either input_channel_mask or 127 // output_channel_mask depending upon the last stream to be opened on this device. 128 struct audio_config common; 129 // Input stream and output stream channel masks. This is required since input and output 130 // channel bitfields are not equivalent. 131 audio_channel_mask_t input_channel_mask; 132 audio_channel_mask_t output_channel_mask; 133#if ENABLE_RESAMPLING 134 // Input stream and output stream sample rates. 135 uint32_t input_sample_rate; 136 uint32_t output_sample_rate; 137#endif // ENABLE_RESAMPLING 138 size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe. 139 size_t buffer_size_frames; // Size of the audio pipe in frames. 140 // Maximum number of frames buffered by the input and output streams. 141 size_t buffer_period_size_frames; 142}; 143 144#define MAX_ROUTES 10 145typedef struct route_config { 146 struct submix_config config; 147 char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; 148 // Pipe variables: they handle the ring buffer that "pipes" audio: 149 // - from the submix virtual audio output == what needs to be played 150 // remotely, seen as an output for AudioFlinger 151 // - to the virtual audio source == what is captured by the component 152 // which "records" the submix / virtual audio source, and handles it as needed. 153 // A usecase example is one where the component capturing the audio is then sending it over 154 // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a 155 // TV with Wifi Display capabilities), or to a wireless audio player. 156 sp<MonoPipe> rsxSink; 157 sp<MonoPipeReader> rsxSource; 158 // Pointers to the current input and output stream instances. rsxSink and rsxSource are 159 // destroyed if both and input and output streams are destroyed. 160 struct submix_stream_out *output; 161 struct submix_stream_in *input; 162#if ENABLE_RESAMPLING 163 // Buffer used as temporary storage for resampled data prior to returning data to the output 164 // stream. 165 int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; 166#endif // ENABLE_RESAMPLING 167} route_config_t; 168 169struct submix_audio_device { 170 struct audio_hw_device device; 171 route_config_t routes[MAX_ROUTES]; 172 // Device lock, also used to protect access to submix_audio_device from the input and output 173 // streams. 174 pthread_mutex_t lock; 175}; 176 177struct submix_stream_out { 178 struct audio_stream_out stream; 179 struct submix_audio_device *dev; 180 int route_handle; 181 bool output_standby; 182 uint64_t frames_written; 183 uint64_t frames_written_since_standby; 184#if LOG_STREAMS_TO_FILES 185 int log_fd; 186#endif // LOG_STREAMS_TO_FILES 187}; 188 189struct submix_stream_in { 190 struct audio_stream_in stream; 191 struct submix_audio_device *dev; 192 int route_handle; 193 bool input_standby; 194 bool output_standby_rec_thr; // output standby state as seen from record thread 195 // wall clock when recording starts 196 struct timespec record_start_time; 197 // how many frames have been requested to be read 198 uint64_t read_counter_frames; 199 200#if ENABLE_LEGACY_INPUT_OPEN 201 // Number of references to this input stream. 202 volatile int32_t ref_count; 203#endif // ENABLE_LEGACY_INPUT_OPEN 204#if LOG_STREAMS_TO_FILES 205 int log_fd; 206#endif // LOG_STREAMS_TO_FILES 207 208 volatile int16_t read_error_count; 209}; 210 211// Determine whether the specified sample rate is supported by the submix module. 212static bool sample_rate_supported(const uint32_t sample_rate) 213{ 214 // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp. 215 static const unsigned int supported_sample_rates[] = { 216 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 217 }; 218 bool return_value; 219 SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value); 220 return return_value; 221} 222 223// Determine whether the specified sample rate is supported, if it is return the specified sample 224// rate, otherwise return the default sample rate for the submix module. 225static uint32_t get_supported_sample_rate(uint32_t sample_rate) 226{ 227 return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ; 228} 229 230// Determine whether the specified channel in mask is supported by the submix module. 231static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask) 232{ 233 // Set of channel in masks supported by Format_from_SR_C() 234 // frameworks/av/media/libnbaio/NAIO.cpp. 235 static const audio_channel_mask_t supported_channel_in_masks[] = { 236 AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, 237 }; 238 bool return_value; 239 SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value); 240 return return_value; 241} 242 243// Determine whether the specified channel in mask is supported, if it is return the specified 244// channel in mask, otherwise return the default channel in mask for the submix module. 245static audio_channel_mask_t get_supported_channel_in_mask( 246 const audio_channel_mask_t channel_in_mask) 247{ 248 return channel_in_mask_supported(channel_in_mask) ? channel_in_mask : 249 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO); 250} 251 252// Determine whether the specified channel out mask is supported by the submix module. 253static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask) 254{ 255 // Set of channel out masks supported by Format_from_SR_C() 256 // frameworks/av/media/libnbaio/NAIO.cpp. 257 static const audio_channel_mask_t supported_channel_out_masks[] = { 258 AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO, 259 }; 260 bool return_value; 261 SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value); 262 return return_value; 263} 264 265// Determine whether the specified channel out mask is supported, if it is return the specified 266// channel out mask, otherwise return the default channel out mask for the submix module. 267static audio_channel_mask_t get_supported_channel_out_mask( 268 const audio_channel_mask_t channel_out_mask) 269{ 270 return channel_out_mask_supported(channel_out_mask) ? channel_out_mask : 271 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO); 272} 273 274// Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the 275// structure. 276static struct submix_stream_out * audio_stream_out_get_submix_stream_out( 277 struct audio_stream_out * const stream) 278{ 279 ALOG_ASSERT(stream); 280 return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) - 281 offsetof(struct submix_stream_out, stream)); 282} 283 284// Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure. 285static struct submix_stream_out * audio_stream_get_submix_stream_out( 286 struct audio_stream * const stream) 287{ 288 ALOG_ASSERT(stream); 289 return audio_stream_out_get_submix_stream_out( 290 reinterpret_cast<struct audio_stream_out *>(stream)); 291} 292 293// Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the 294// structure. 295static struct submix_stream_in * audio_stream_in_get_submix_stream_in( 296 struct audio_stream_in * const stream) 297{ 298 ALOG_ASSERT(stream); 299 return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) - 300 offsetof(struct submix_stream_in, stream)); 301} 302 303// Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure. 304static struct submix_stream_in * audio_stream_get_submix_stream_in( 305 struct audio_stream * const stream) 306{ 307 ALOG_ASSERT(stream); 308 return audio_stream_in_get_submix_stream_in( 309 reinterpret_cast<struct audio_stream_in *>(stream)); 310} 311 312// Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within 313// the structure. 314static struct submix_audio_device * audio_hw_device_get_submix_audio_device( 315 struct audio_hw_device *device) 316{ 317 ALOG_ASSERT(device); 318 return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) - 319 offsetof(struct submix_audio_device, device)); 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 = audio_channel_count_from_in_mask(input_config->channel_mask); 329 const uint32_t output_channels = audio_channel_count_from_out_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 audio_channel_count_from_in_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. 357// Must be called with lock held on the submix_audio_device 358static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev, 359 const struct audio_config * const config, 360 const size_t buffer_size_frames, 361 const uint32_t buffer_period_count, 362 struct submix_stream_in * const in, 363 struct submix_stream_out * const out, 364 const char *address, 365 int route_idx) 366{ 367 ALOG_ASSERT(in || out); 368 ALOG_ASSERT(route_idx > -1); 369 ALOG_ASSERT(route_idx < MAX_ROUTES); 370 ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx); 371 372 // Save a reference to the specified input or output stream and the associated channel 373 // mask. 374 if (in) { 375 in->route_handle = route_idx; 376 rsxadev->routes[route_idx].input = in; 377 rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask; 378#if ENABLE_RESAMPLING 379 rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate; 380 // If the output isn't configured yet, set the output sample rate to the maximum supported 381 // sample rate such that the smallest possible input buffer is created, and put a default 382 // value for channel count 383 if (!rsxadev->routes[route_idx].output) { 384 rsxadev->routes[route_idx].config.output_sample_rate = 48000; 385 rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO; 386 } 387#endif // ENABLE_RESAMPLING 388 } 389 if (out) { 390 out->route_handle = route_idx; 391 rsxadev->routes[route_idx].output = out; 392 rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask; 393#if ENABLE_RESAMPLING 394 rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate; 395#endif // ENABLE_RESAMPLING 396 } 397 // Save the address 398 strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN); 399 ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx); 400 // If a pipe isn't associated with the device, create one. 401 if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL) 402 { 403 struct submix_config * const device_config = &rsxadev->routes[route_idx].config; 404 uint32_t channel_count; 405 if (out) 406 channel_count = audio_channel_count_from_out_mask(config->channel_mask); 407 else 408 channel_count = audio_channel_count_from_in_mask(config->channel_mask); 409#if ENABLE_CHANNEL_CONVERSION 410 // If channel conversion is enabled, allocate enough space for the maximum number of 411 // possible channels stored in the pipe for the situation when the number of channels in 412 // the output stream don't match the number in the input stream. 413 const uint32_t pipe_channel_count = max(channel_count, 2); 414#else 415 const uint32_t pipe_channel_count = channel_count; 416#endif // ENABLE_CHANNEL_CONVERSION 417 const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, 418 config->format); 419 const NBAIO_Format offers[1] = {format}; 420 size_t numCounterOffers = 0; 421 // Create a MonoPipe with optional blocking set to true. 422 MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); 423 // Negotiation between the source and sink cannot fail as the device open operation 424 // creates both ends of the pipe using the same audio format. 425 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 426 ALOG_ASSERT(index == 0); 427 MonoPipeReader* source = new MonoPipeReader(sink); 428 numCounterOffers = 0; 429 index = source->negotiate(offers, 1, NULL, numCounterOffers); 430 ALOG_ASSERT(index == 0); 431 ALOGV("submix_audio_device_create_pipe_l(): created pipe"); 432 433 // Save references to the source and sink. 434 ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL); 435 ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL); 436 rsxadev->routes[route_idx].rsxSink = sink; 437 rsxadev->routes[route_idx].rsxSource = source; 438 // Store the sanitized audio format in the device so that it's possible to determine 439 // the format of the pipe source when opening the input device. 440 memcpy(&device_config->common, config, sizeof(device_config->common)); 441 device_config->buffer_size_frames = sink->maxFrames(); 442 device_config->buffer_period_size_frames = device_config->buffer_size_frames / 443 buffer_period_count; 444 if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); 445 if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); 446#if ENABLE_CHANNEL_CONVERSION 447 // Calculate the pipe frame size based upon the number of channels. 448 device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / 449 channel_count; 450#endif // ENABLE_CHANNEL_CONVERSION 451 SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, " 452 "period size %zd", device_config->pipe_frame_size, 453 device_config->buffer_size_frames, device_config->buffer_period_size_frames); 454 } 455} 456 457// Release references to the sink and source. Input and output threads may maintain references 458// to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use 459// before they shutdown. 460// Must be called with lock held on the submix_audio_device 461static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev, 462 int route_idx) 463{ 464 ALOG_ASSERT(route_idx > -1); 465 ALOG_ASSERT(route_idx < MAX_ROUTES); 466 ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx, 467 rsxadev->routes[route_idx].address); 468 if (rsxadev->routes[route_idx].rsxSink != 0) { 469 rsxadev->routes[route_idx].rsxSink.clear(); 470 rsxadev->routes[route_idx].rsxSink = 0; 471 } 472 if (rsxadev->routes[route_idx].rsxSource != 0) { 473 rsxadev->routes[route_idx].rsxSource.clear(); 474 rsxadev->routes[route_idx].rsxSource = 0; 475 } 476 memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); 477#ifdef ENABLE_RESAMPLING 478 memset(rsxadev->routes[route_idx].resampler_buffer, 0, 479 sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); 480#endif 481} 482 483// Remove references to the specified input and output streams. When the device no longer 484// references input and output streams destroy the associated pipe. 485// Must be called with lock held on the submix_audio_device 486static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev, 487 const struct submix_stream_in * const in, 488 const struct submix_stream_out * const out) 489{ 490 MonoPipe* sink; 491 ALOGV("submix_audio_device_destroy_pipe_l()"); 492 int route_idx = -1; 493 if (in != NULL) { 494#if ENABLE_LEGACY_INPUT_OPEN 495 const_cast<struct submix_stream_in*>(in)->ref_count--; 496 route_idx = in->route_handle; 497 ALOG_ASSERT(rsxadev->routes[route_idx].input == in); 498 if (in->ref_count == 0) { 499 rsxadev->routes[route_idx].input = NULL; 500 } 501 ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count); 502#else 503 rsxadev->input = NULL; 504#endif // ENABLE_LEGACY_INPUT_OPEN 505 } 506 if (out != NULL) { 507 route_idx = out->route_handle; 508 ALOG_ASSERT(rsxadev->routes[route_idx].output == out); 509 rsxadev->routes[route_idx].output = NULL; 510 } 511 if (route_idx != -1 && 512 rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) { 513 submix_audio_device_release_pipe_l(rsxadev, route_idx); 514 ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed"); 515 } 516} 517 518// Sanitize the user specified audio config for a submix input / output stream. 519static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) 520{ 521 config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : 522 get_supported_channel_out_mask(config->channel_mask); 523 config->sample_rate = get_supported_sample_rate(config->sample_rate); 524 config->format = DEFAULT_FORMAT; 525} 526 527// Verify a submix input or output stream can be opened. 528// Must be called with lock held on the submix_audio_device 529static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev, 530 int route_idx, 531 const struct audio_config * const config, 532 const bool opening_input) 533{ 534 bool input_open; 535 bool output_open; 536 audio_config pipe_config; 537 538 // Query the device for the current audio config and whether input and output streams are open. 539 output_open = rsxadev->routes[route_idx].output != NULL; 540 input_open = rsxadev->routes[route_idx].input != NULL; 541 memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config)); 542 543 // If the stream is already open, don't open it again. 544 if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { 545 ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" : 546 "Output"); 547 return false; 548 } 549 550 SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x " 551 "%s_channel_mask=%x", config->sample_rate, config->format, 552 opening_input ? "in" : "out", config->channel_mask); 553 554 // If either stream is open, verify the existing audio config the pipe matches the user 555 // specified config. 556 if (input_open || output_open) { 557 const audio_config * const input_config = opening_input ? config : &pipe_config; 558 const audio_config * const output_config = opening_input ? &pipe_config : config; 559 // Get the channel mask of the open device. 560 pipe_config.channel_mask = 561 opening_input ? rsxadev->routes[route_idx].config.output_channel_mask : 562 rsxadev->routes[route_idx].config.input_channel_mask; 563 if (!audio_config_compare(input_config, output_config)) { 564 ALOGE("submix_open_validate_l(): Unsupported format."); 565 return false; 566 } 567 } 568 return true; 569} 570 571// Must be called with lock held on the submix_audio_device 572static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev, 573 const char* address, /*in*/ 574 int *idx /*out*/) 575{ 576 // Do we already have a route for this address 577 int route_idx = -1; 578 int route_empty_idx = -1; // index of an empty route slot that can be used if needed 579 for (int i=0 ; i < MAX_ROUTES ; i++) { 580 if (strcmp(rsxadev->routes[i].address, "") == 0) { 581 route_empty_idx = i; 582 } 583 if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { 584 route_idx = i; 585 break; 586 } 587 } 588 589 if ((route_idx == -1) && (route_empty_idx == -1)) { 590 ALOGE("Cannot create new route for address %s, max number of routes reached", address); 591 return -ENOMEM; 592 } 593 if (route_idx == -1) { 594 route_idx = route_empty_idx; 595 } 596 *idx = route_idx; 597 return OK; 598} 599 600 601// Calculate the maximum size of the pipe buffer in frames for the specified stream. 602static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, 603 const struct submix_config *config, 604 const size_t pipe_frames, 605 const size_t stream_frame_size) 606{ 607 const size_t pipe_frame_size = config->pipe_frame_size; 608 const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); 609 return (pipe_frames * config->pipe_frame_size) / max_frame_size; 610} 611 612/* audio HAL functions */ 613 614static uint32_t out_get_sample_rate(const struct audio_stream *stream) 615{ 616 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 617 const_cast<struct audio_stream *>(stream)); 618#if ENABLE_RESAMPLING 619 const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; 620#else 621 const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate; 622#endif // ENABLE_RESAMPLING 623 SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", 624 out_rate, out->dev->routes[out->route_handle].address); 625 return out_rate; 626} 627 628static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 629{ 630 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 631#if ENABLE_RESAMPLING 632 // The sample rate of the stream can't be changed once it's set since this would change the 633 // output buffer size and hence break playback to the shared pipe. 634 if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) { 635 ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " 636 "%u to %u for addr %s", 637 out->dev->routes[out->route_handle].config.output_sample_rate, rate, 638 out->dev->routes[out->route_handle].address); 639 return -ENOSYS; 640 } 641#endif // ENABLE_RESAMPLING 642 if (!sample_rate_supported(rate)) { 643 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 644 return -ENOSYS; 645 } 646 SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); 647 out->dev->routes[out->route_handle].config.common.sample_rate = rate; 648 return 0; 649} 650 651static size_t out_get_buffer_size(const struct audio_stream *stream) 652{ 653 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 654 const_cast<struct audio_stream *>(stream)); 655 const struct submix_config * const config = &out->dev->routes[out->route_handle].config; 656 const size_t stream_frame_size = 657 audio_stream_out_frame_size((const struct audio_stream_out *)stream); 658 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 659 stream, config, config->buffer_period_size_frames, stream_frame_size); 660 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 661 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", 662 buffer_size_bytes, buffer_size_frames); 663 return buffer_size_bytes; 664} 665 666static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 667{ 668 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 669 const_cast<struct audio_stream *>(stream)); 670 uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask; 671 SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); 672 return channel_mask; 673} 674 675static audio_format_t out_get_format(const struct audio_stream *stream) 676{ 677 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 678 const_cast<struct audio_stream *>(stream)); 679 const audio_format_t format = out->dev->routes[out->route_handle].config.common.format; 680 SUBMIX_ALOGV("out_get_format() returns %x", format); 681 return format; 682} 683 684static int out_set_format(struct audio_stream *stream, audio_format_t format) 685{ 686 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 687 if (format != out->dev->routes[out->route_handle].config.common.format) { 688 ALOGE("out_set_format(format=%x) format unsupported", format); 689 return -ENOSYS; 690 } 691 SUBMIX_ALOGV("out_set_format(format=%x)", format); 692 return 0; 693} 694 695static int out_standby(struct audio_stream *stream) 696{ 697 ALOGI("out_standby()"); 698 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 699 struct submix_audio_device * const rsxadev = out->dev; 700 701 pthread_mutex_lock(&rsxadev->lock); 702 703 out->output_standby = true; 704 out->frames_written_since_standby = 0; 705 706 pthread_mutex_unlock(&rsxadev->lock); 707 708 return 0; 709} 710 711static int out_dump(const struct audio_stream *stream, int fd) 712{ 713 (void)stream; 714 (void)fd; 715 return 0; 716} 717 718static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 719{ 720 int exiting = -1; 721 AudioParameter parms = AudioParameter(String8(kvpairs)); 722 SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs); 723 724 // FIXME this is using hard-coded strings but in the future, this functionality will be 725 // converted to use audio HAL extensions required to support tunneling 726 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { 727 struct submix_audio_device * const rsxadev = 728 audio_stream_get_submix_stream_out(stream)->dev; 729 pthread_mutex_lock(&rsxadev->lock); 730 { // using the sink 731 sp<MonoPipe> sink = 732 rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle] 733 .rsxSink; 734 if (sink == NULL) { 735 pthread_mutex_unlock(&rsxadev->lock); 736 return 0; 737 } 738 739 ALOGD("out_set_parameters(): shutting down MonoPipe sink"); 740 sink->shutdown(true); 741 } // done using the sink 742 pthread_mutex_unlock(&rsxadev->lock); 743 } 744 return 0; 745} 746 747static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 748{ 749 (void)stream; 750 (void)keys; 751 return strdup(""); 752} 753 754static uint32_t out_get_latency(const struct audio_stream_out *stream) 755{ 756 const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( 757 const_cast<struct audio_stream_out *>(stream)); 758 const struct submix_config * const config = &out->dev->routes[out->route_handle].config; 759 const size_t stream_frame_size = 760 audio_stream_out_frame_size(stream); 761 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 762 &stream->common, config, config->buffer_size_frames, stream_frame_size); 763 const uint32_t sample_rate = out_get_sample_rate(&stream->common); 764 const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate; 765 SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", 766 latency_ms, buffer_size_frames, sample_rate); 767 return latency_ms; 768} 769 770static int out_set_volume(struct audio_stream_out *stream, float left, 771 float right) 772{ 773 (void)stream; 774 (void)left; 775 (void)right; 776 return -ENOSYS; 777} 778 779static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 780 size_t bytes) 781{ 782 SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); 783 ssize_t written_frames = 0; 784 const size_t frame_size = audio_stream_out_frame_size(stream); 785 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 786 struct submix_audio_device * const rsxadev = out->dev; 787 const size_t frames = bytes / frame_size; 788 789 pthread_mutex_lock(&rsxadev->lock); 790 791 out->output_standby = false; 792 793 sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink; 794 if (sink != NULL) { 795 if (sink->isShutdown()) { 796 sink.clear(); 797 pthread_mutex_unlock(&rsxadev->lock); 798 SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); 799 // the pipe has already been shutdown, this buffer will be lost but we must 800 // simulate timing so we don't drain the output faster than realtime 801 usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); 802 return bytes; 803 } 804 } else { 805 pthread_mutex_unlock(&rsxadev->lock); 806 ALOGE("out_write without a pipe!"); 807 ALOG_ASSERT("out_write without a pipe!"); 808 return 0; 809 } 810 811 // If the write to the sink would block when no input stream is present, flush enough frames 812 // from the pipe to make space to write the most recent data. 813 { 814 const size_t availableToWrite = sink->availableToWrite(); 815 sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource; 816 if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) { 817 static uint8_t flush_buffer[64]; 818 const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; 819 size_t frames_to_flush_from_source = frames - availableToWrite; 820 SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", 821 frames_to_flush_from_source); 822 while (frames_to_flush_from_source) { 823 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); 824 frames_to_flush_from_source -= flush_size; 825 // read does not block 826 source->read(flush_buffer, flush_size); 827 } 828 } 829 } 830 831 pthread_mutex_unlock(&rsxadev->lock); 832 833 written_frames = sink->write(buffer, frames); 834 835#if LOG_STREAMS_TO_FILES 836 if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); 837#endif // LOG_STREAMS_TO_FILES 838 839 if (written_frames < 0) { 840 if (written_frames == (ssize_t)NEGOTIATE) { 841 ALOGE("out_write() write to pipe returned NEGOTIATE"); 842 843 pthread_mutex_lock(&rsxadev->lock); 844 sink.clear(); 845 pthread_mutex_unlock(&rsxadev->lock); 846 847 written_frames = 0; 848 return 0; 849 } else { 850 // write() returned UNDERRUN or WOULD_BLOCK, retry 851 ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); 852 written_frames = sink->write(buffer, frames); 853 } 854 } 855 856 pthread_mutex_lock(&rsxadev->lock); 857 sink.clear(); 858 if (written_frames > 0) { 859 out->frames_written_since_standby += written_frames; 860 out->frames_written += written_frames; 861 } 862 pthread_mutex_unlock(&rsxadev->lock); 863 864 if (written_frames < 0) { 865 ALOGE("out_write() failed writing to pipe with %zd", written_frames); 866 return 0; 867 } 868 const ssize_t written_bytes = written_frames * frame_size; 869 SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); 870 return written_bytes; 871} 872 873static int out_get_presentation_position(const struct audio_stream_out *stream, 874 uint64_t *frames, struct timespec *timestamp) 875{ 876 if (stream == NULL || frames == NULL || timestamp == NULL) { 877 return -EINVAL; 878 } 879 880 const submix_stream_out *out = audio_stream_out_get_submix_stream_out( 881 const_cast<struct audio_stream_out *>(stream)); 882 struct submix_audio_device * const rsxadev = out->dev; 883 884 int ret = -EWOULDBLOCK; 885 pthread_mutex_lock(&rsxadev->lock); 886 const ssize_t frames_in_pipe = 887 rsxadev->routes[out->route_handle].rsxSource->availableToRead(); 888 if (CC_UNLIKELY(frames_in_pipe < 0)) { 889 *frames = out->frames_written; 890 ret = 0; 891 } else if (out->frames_written >= (uint64_t)frames_in_pipe) { 892 *frames = out->frames_written - frames_in_pipe; 893 ret = 0; 894 } 895 pthread_mutex_unlock(&rsxadev->lock); 896 897 if (ret == 0) { 898 clock_gettime(CLOCK_MONOTONIC, timestamp); 899 } 900 901 SUBMIX_ALOGV("out_get_presentation_position() got frames=%llu timestamp sec=%llu", 902 frames ? *frames : -1, timestamp ? timestamp->tv_sec : -1); 903 904 return ret; 905} 906 907static int out_get_render_position(const struct audio_stream_out *stream, 908 uint32_t *dsp_frames) 909{ 910 if (stream == NULL || dsp_frames == NULL) { 911 return -EINVAL; 912 } 913 914 const submix_stream_out *out = audio_stream_out_get_submix_stream_out( 915 const_cast<struct audio_stream_out *>(stream)); 916 struct submix_audio_device * const rsxadev = out->dev; 917 918 pthread_mutex_lock(&rsxadev->lock); 919 const ssize_t frames_in_pipe = 920 rsxadev->routes[out->route_handle].rsxSource->availableToRead(); 921 if (CC_UNLIKELY(frames_in_pipe < 0)) { 922 *dsp_frames = (uint32_t)out->frames_written_since_standby; 923 } else { 924 *dsp_frames = out->frames_written_since_standby > (uint64_t) frames_in_pipe ? 925 (uint32_t)(out->frames_written_since_standby - frames_in_pipe) : 0; 926 } 927 pthread_mutex_unlock(&rsxadev->lock); 928 929 return 0; 930} 931 932static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 933{ 934 (void)stream; 935 (void)effect; 936 return 0; 937} 938 939static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 940{ 941 (void)stream; 942 (void)effect; 943 return 0; 944} 945 946static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 947 int64_t *timestamp) 948{ 949 (void)stream; 950 (void)timestamp; 951 return -EINVAL; 952} 953 954/** audio_stream_in implementation **/ 955static uint32_t in_get_sample_rate(const struct audio_stream *stream) 956{ 957 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 958 const_cast<struct audio_stream*>(stream)); 959#if ENABLE_RESAMPLING 960 const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; 961#else 962 const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate; 963#endif // ENABLE_RESAMPLING 964 SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); 965 return rate; 966} 967 968static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 969{ 970 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 971#if ENABLE_RESAMPLING 972 // The sample rate of the stream can't be changed once it's set since this would change the 973 // input buffer size and hence break recording from the shared pipe. 974 if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) { 975 ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " 976 "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); 977 return -ENOSYS; 978 } 979#endif // ENABLE_RESAMPLING 980 if (!sample_rate_supported(rate)) { 981 ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); 982 return -ENOSYS; 983 } 984 in->dev->routes[in->route_handle].config.common.sample_rate = rate; 985 SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); 986 return 0; 987} 988 989static size_t in_get_buffer_size(const struct audio_stream *stream) 990{ 991 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 992 const_cast<struct audio_stream*>(stream)); 993 const struct submix_config * const config = &in->dev->routes[in->route_handle].config; 994 const size_t stream_frame_size = 995 audio_stream_in_frame_size((const struct audio_stream_in *)stream); 996 size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 997 stream, config, config->buffer_period_size_frames, stream_frame_size); 998#if ENABLE_RESAMPLING 999 // Scale the size of the buffer based upon the maximum number of frames that could be returned 1000 // given the ratio of output to input sample rate. 1001 buffer_size_frames = (size_t)(((float)buffer_size_frames * 1002 (float)config->input_sample_rate) / 1003 (float)config->output_sample_rate); 1004#endif // ENABLE_RESAMPLING 1005 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 1006 SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, 1007 buffer_size_frames); 1008 return buffer_size_bytes; 1009} 1010 1011static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 1012{ 1013 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 1014 const_cast<struct audio_stream*>(stream)); 1015 const audio_channel_mask_t channel_mask = 1016 in->dev->routes[in->route_handle].config.input_channel_mask; 1017 SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); 1018 return channel_mask; 1019} 1020 1021static audio_format_t in_get_format(const struct audio_stream *stream) 1022{ 1023 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 1024 const_cast<struct audio_stream*>(stream)); 1025 const audio_format_t format = in->dev->routes[in->route_handle].config.common.format; 1026 SUBMIX_ALOGV("in_get_format() returns %x", format); 1027 return format; 1028} 1029 1030static int in_set_format(struct audio_stream *stream, audio_format_t format) 1031{ 1032 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 1033 if (format != in->dev->routes[in->route_handle].config.common.format) { 1034 ALOGE("in_set_format(format=%x) format unsupported", format); 1035 return -ENOSYS; 1036 } 1037 SUBMIX_ALOGV("in_set_format(format=%x)", format); 1038 return 0; 1039} 1040 1041static int in_standby(struct audio_stream *stream) 1042{ 1043 ALOGI("in_standby()"); 1044 struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 1045 struct submix_audio_device * const rsxadev = in->dev; 1046 1047 pthread_mutex_lock(&rsxadev->lock); 1048 1049 in->input_standby = true; 1050 1051 pthread_mutex_unlock(&rsxadev->lock); 1052 1053 return 0; 1054} 1055 1056static int in_dump(const struct audio_stream *stream, int fd) 1057{ 1058 (void)stream; 1059 (void)fd; 1060 return 0; 1061} 1062 1063static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 1064{ 1065 (void)stream; 1066 (void)kvpairs; 1067 return 0; 1068} 1069 1070static char * in_get_parameters(const struct audio_stream *stream, 1071 const char *keys) 1072{ 1073 (void)stream; 1074 (void)keys; 1075 return strdup(""); 1076} 1077 1078static int in_set_gain(struct audio_stream_in *stream, float gain) 1079{ 1080 (void)stream; 1081 (void)gain; 1082 return 0; 1083} 1084 1085static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 1086 size_t bytes) 1087{ 1088 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1089 struct submix_audio_device * const rsxadev = in->dev; 1090 struct audio_config *format; 1091 const size_t frame_size = audio_stream_in_frame_size(stream); 1092 const size_t frames_to_read = bytes / frame_size; 1093 1094 SUBMIX_ALOGV("in_read bytes=%zu", bytes); 1095 pthread_mutex_lock(&rsxadev->lock); 1096 1097 const bool output_standby = rsxadev->routes[in->route_handle].output == NULL 1098 ? true : rsxadev->routes[in->route_handle].output->output_standby; 1099 const bool output_standby_transition = (in->output_standby_rec_thr != output_standby); 1100 in->output_standby_rec_thr = output_standby; 1101 1102 if (in->input_standby || output_standby_transition) { 1103 in->input_standby = false; 1104 // keep track of when we exit input standby (== first read == start "real recording") 1105 // or when we start recording silence, and reset projected time 1106 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 1107 if (rc == 0) { 1108 in->read_counter_frames = 0; 1109 } 1110 } 1111 1112 in->read_counter_frames += frames_to_read; 1113 size_t remaining_frames = frames_to_read; 1114 1115 { 1116 // about to read from audio source 1117 sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource; 1118 if (source == NULL) { 1119 in->read_error_count++;// ok if it rolls over 1120 ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS, 1121 "no audio pipe yet we're trying to read! (not all errors will be logged)"); 1122 pthread_mutex_unlock(&rsxadev->lock); 1123 usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); 1124 memset(buffer, 0, bytes); 1125 return bytes; 1126 } 1127 1128 pthread_mutex_unlock(&rsxadev->lock); 1129 1130 // read the data from the pipe (it's non blocking) 1131 int attempts = 0; 1132 char* buff = (char*)buffer; 1133#if ENABLE_CHANNEL_CONVERSION 1134 // Determine whether channel conversion is required. 1135 const uint32_t input_channels = audio_channel_count_from_in_mask( 1136 rsxadev->routes[in->route_handle].config.input_channel_mask); 1137 const uint32_t output_channels = audio_channel_count_from_out_mask( 1138 rsxadev->routes[in->route_handle].config.output_channel_mask); 1139 if (input_channels != output_channels) { 1140 SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " 1141 "input channels", output_channels, input_channels); 1142 // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. 1143 ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == 1144 AUDIO_FORMAT_PCM_16_BIT); 1145 ALOG_ASSERT((input_channels == 1 && output_channels == 2) || 1146 (input_channels == 2 && output_channels == 1)); 1147 } 1148#endif // ENABLE_CHANNEL_CONVERSION 1149 1150#if ENABLE_RESAMPLING 1151 const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); 1152 const uint32_t output_sample_rate = 1153 rsxadev->routes[in->route_handle].config.output_sample_rate; 1154 const size_t resampler_buffer_size_frames = 1155 sizeof(rsxadev->routes[in->route_handle].resampler_buffer) / 1156 sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]); 1157 float resampler_ratio = 1.0f; 1158 // Determine whether resampling is required. 1159 if (input_sample_rate != output_sample_rate) { 1160 resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; 1161 // Only support 16-bit PCM mono resampling. 1162 // NOTE: Resampling is performed after the channel conversion step. 1163 ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == 1164 AUDIO_FORMAT_PCM_16_BIT); 1165 ALOG_ASSERT(audio_channel_count_from_in_mask( 1166 rsxadev->routes[in->route_handle].config.input_channel_mask) == 1); 1167 } 1168#endif // ENABLE_RESAMPLING 1169 1170 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 1171 ssize_t frames_read = -1977; 1172 size_t read_frames = remaining_frames; 1173#if ENABLE_RESAMPLING 1174 char* const saved_buff = buff; 1175 if (resampler_ratio != 1.0f) { 1176 // Calculate the number of frames from the pipe that need to be read to generate 1177 // the data for the input stream read. 1178 const size_t frames_required_for_resampler = (size_t)( 1179 (float)read_frames * (float)resampler_ratio); 1180 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); 1181 // Read into the resampler buffer. 1182 buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; 1183 } 1184#endif // ENABLE_RESAMPLING 1185#if ENABLE_CHANNEL_CONVERSION 1186 if (output_channels == 1 && input_channels == 2) { 1187 // Need to read half the requested frames since the converted output 1188 // data will take twice the space (mono->stereo). 1189 read_frames /= 2; 1190 } 1191#endif // ENABLE_CHANNEL_CONVERSION 1192 1193 SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); 1194 1195 frames_read = source->read(buff, read_frames); 1196 1197 SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); 1198 1199#if ENABLE_CHANNEL_CONVERSION 1200 // Perform in-place channel conversion. 1201 // NOTE: In the following "input stream" refers to the data returned by this function 1202 // and "output stream" refers to the data read from the pipe. 1203 if (input_channels != output_channels && frames_read > 0) { 1204 int16_t *data = (int16_t*)buff; 1205 if (output_channels == 2 && input_channels == 1) { 1206 // Offset into the output stream data in samples. 1207 ssize_t output_stream_offset = 0; 1208 for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; 1209 input_stream_frame++, output_stream_offset += 2) { 1210 // Average the content from both channels. 1211 data[input_stream_frame] = ((int32_t)data[output_stream_offset] + 1212 (int32_t)data[output_stream_offset + 1]) / 2; 1213 } 1214 } else if (output_channels == 1 && input_channels == 2) { 1215 // Offset into the input stream data in samples. 1216 ssize_t input_stream_offset = (frames_read - 1) * 2; 1217 for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; 1218 output_stream_frame--, input_stream_offset -= 2) { 1219 const short sample = data[output_stream_frame]; 1220 data[input_stream_offset] = sample; 1221 data[input_stream_offset + 1] = sample; 1222 } 1223 } 1224 } 1225#endif // ENABLE_CHANNEL_CONVERSION 1226 1227#if ENABLE_RESAMPLING 1228 if (resampler_ratio != 1.0f) { 1229 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); 1230 const int16_t * const data = (int16_t*)buff; 1231 int16_t * const resampled_buffer = (int16_t*)saved_buff; 1232 // Resample with *no* filtering - if the data from the ouptut stream was really 1233 // sampled at a different rate this will result in very nasty aliasing. 1234 const float output_stream_frames = (float)frames_read; 1235 size_t input_stream_frame = 0; 1236 for (float output_stream_frame = 0.0f; 1237 output_stream_frame < output_stream_frames && 1238 input_stream_frame < remaining_frames; 1239 output_stream_frame += resampler_ratio, input_stream_frame++) { 1240 resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; 1241 } 1242 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); 1243 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); 1244 frames_read = input_stream_frame; 1245 buff = saved_buff; 1246 } 1247#endif // ENABLE_RESAMPLING 1248 1249 if (frames_read > 0) { 1250#if LOG_STREAMS_TO_FILES 1251 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); 1252#endif // LOG_STREAMS_TO_FILES 1253 1254 remaining_frames -= frames_read; 1255 buff += frames_read * frame_size; 1256 SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", 1257 attempts, frames_read, remaining_frames); 1258 } else { 1259 attempts++; 1260 SUBMIX_ALOGE(" in_read read returned %zd", frames_read); 1261 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 1262 } 1263 } 1264 // done using the source 1265 pthread_mutex_lock(&rsxadev->lock); 1266 source.clear(); 1267 pthread_mutex_unlock(&rsxadev->lock); 1268 } 1269 1270 if (remaining_frames > 0) { 1271 const size_t remaining_bytes = remaining_frames * frame_size; 1272 SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames); 1273 memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); 1274 } 1275 1276 // compute how much we need to sleep after reading the data by comparing the wall clock with 1277 // the projected time at which we should return. 1278 struct timespec time_after_read;// wall clock after reading from the pipe 1279 struct timespec record_duration;// observed record duration 1280 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 1281 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 1282 if (rc == 0) { 1283 // for how long have we been recording? 1284 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 1285 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 1286 if (record_duration.tv_nsec < 0) { 1287 record_duration.tv_sec--; 1288 record_duration.tv_nsec += 1000000000; 1289 } 1290 1291 // read_counter_frames contains the number of frames that have been read since the 1292 // beginning of recording (including this call): it's converted to usec and compared to 1293 // how long we've been recording for, which gives us how long we must wait to sync the 1294 // projected recording time, and the observed recording time. 1295 long projected_vs_observed_offset_us = 1296 ((int64_t)(in->read_counter_frames 1297 - (record_duration.tv_sec*sample_rate))) 1298 * 1000000 / sample_rate 1299 - (record_duration.tv_nsec / 1000); 1300 1301 SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 1302 record_duration.tv_sec, record_duration.tv_nsec/1000000, 1303 projected_vs_observed_offset_us); 1304 if (projected_vs_observed_offset_us > 0) { 1305 usleep(projected_vs_observed_offset_us); 1306 } 1307 } 1308 1309 SUBMIX_ALOGV("in_read returns %zu", bytes); 1310 return bytes; 1311 1312} 1313 1314static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 1315{ 1316 (void)stream; 1317 return 0; 1318} 1319 1320static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1321{ 1322 (void)stream; 1323 (void)effect; 1324 return 0; 1325} 1326 1327static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1328{ 1329 (void)stream; 1330 (void)effect; 1331 return 0; 1332} 1333 1334static int adev_open_output_stream(struct audio_hw_device *dev, 1335 audio_io_handle_t handle, 1336 audio_devices_t devices, 1337 audio_output_flags_t flags, 1338 struct audio_config *config, 1339 struct audio_stream_out **stream_out, 1340 const char *address) 1341{ 1342 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1343 ALOGD("adev_open_output_stream(address=%s)", address); 1344 struct submix_stream_out *out; 1345 bool force_pipe_creation = false; 1346 (void)handle; 1347 (void)devices; 1348 (void)flags; 1349 1350 *stream_out = NULL; 1351 1352 // Make sure it's possible to open the device given the current audio config. 1353 submix_sanitize_config(config, false); 1354 1355 int route_idx = -1; 1356 1357 pthread_mutex_lock(&rsxadev->lock); 1358 1359 status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); 1360 if (res != OK) { 1361 ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); 1362 pthread_mutex_unlock(&rsxadev->lock); 1363 return res; 1364 } 1365 1366 if (!submix_open_validate_l(rsxadev, route_idx, config, false)) { 1367 ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address); 1368 pthread_mutex_unlock(&rsxadev->lock); 1369 return -EINVAL; 1370 } 1371 1372 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 1373 if (!out) { 1374 pthread_mutex_unlock(&rsxadev->lock); 1375 return -ENOMEM; 1376 } 1377 1378 // Initialize the function pointer tables (v-tables). 1379 out->stream.common.get_sample_rate = out_get_sample_rate; 1380 out->stream.common.set_sample_rate = out_set_sample_rate; 1381 out->stream.common.get_buffer_size = out_get_buffer_size; 1382 out->stream.common.get_channels = out_get_channels; 1383 out->stream.common.get_format = out_get_format; 1384 out->stream.common.set_format = out_set_format; 1385 out->stream.common.standby = out_standby; 1386 out->stream.common.dump = out_dump; 1387 out->stream.common.set_parameters = out_set_parameters; 1388 out->stream.common.get_parameters = out_get_parameters; 1389 out->stream.common.add_audio_effect = out_add_audio_effect; 1390 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1391 out->stream.get_latency = out_get_latency; 1392 out->stream.set_volume = out_set_volume; 1393 out->stream.write = out_write; 1394 out->stream.get_render_position = out_get_render_position; 1395 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1396 out->stream.get_presentation_position = out_get_presentation_position; 1397 1398#if ENABLE_RESAMPLING 1399 // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits 1400 // writes correctly. 1401 force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate 1402 != config->sample_rate; 1403#endif // ENABLE_RESAMPLING 1404 1405 // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so 1406 // that it's recreated. 1407 if ((rsxadev->routes[route_idx].rsxSink != NULL 1408 && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { 1409 submix_audio_device_release_pipe_l(rsxadev, route_idx); 1410 } 1411 1412 // Store a pointer to the device from the output stream. 1413 out->dev = rsxadev; 1414 // Initialize the pipe. 1415 ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx); 1416 submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1417 DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx); 1418#if LOG_STREAMS_TO_FILES 1419 out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1420 LOG_STREAM_FILE_PERMISSIONS); 1421 ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", 1422 strerror(errno)); 1423 ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); 1424#endif // LOG_STREAMS_TO_FILES 1425 // Return the output stream. 1426 *stream_out = &out->stream; 1427 1428 pthread_mutex_unlock(&rsxadev->lock); 1429 return 0; 1430} 1431 1432static void adev_close_output_stream(struct audio_hw_device *dev, 1433 struct audio_stream_out *stream) 1434{ 1435 struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( 1436 const_cast<struct audio_hw_device*>(dev)); 1437 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 1438 1439 pthread_mutex_lock(&rsxadev->lock); 1440 ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address); 1441 submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out); 1442#if LOG_STREAMS_TO_FILES 1443 if (out->log_fd >= 0) close(out->log_fd); 1444#endif // LOG_STREAMS_TO_FILES 1445 1446 pthread_mutex_unlock(&rsxadev->lock); 1447 free(out); 1448} 1449 1450static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1451{ 1452 (void)dev; 1453 (void)kvpairs; 1454 return -ENOSYS; 1455} 1456 1457static char * adev_get_parameters(const struct audio_hw_device *dev, 1458 const char *keys) 1459{ 1460 (void)dev; 1461 (void)keys; 1462 return strdup("");; 1463} 1464 1465static int adev_init_check(const struct audio_hw_device *dev) 1466{ 1467 ALOGI("adev_init_check()"); 1468 (void)dev; 1469 return 0; 1470} 1471 1472static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 1473{ 1474 (void)dev; 1475 (void)volume; 1476 return -ENOSYS; 1477} 1478 1479static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 1480{ 1481 (void)dev; 1482 (void)volume; 1483 return -ENOSYS; 1484} 1485 1486static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 1487{ 1488 (void)dev; 1489 (void)volume; 1490 return -ENOSYS; 1491} 1492 1493static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 1494{ 1495 (void)dev; 1496 (void)muted; 1497 return -ENOSYS; 1498} 1499 1500static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 1501{ 1502 (void)dev; 1503 (void)muted; 1504 return -ENOSYS; 1505} 1506 1507static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 1508{ 1509 (void)dev; 1510 (void)mode; 1511 return 0; 1512} 1513 1514static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 1515{ 1516 (void)dev; 1517 (void)state; 1518 return -ENOSYS; 1519} 1520 1521static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 1522{ 1523 (void)dev; 1524 (void)state; 1525 return -ENOSYS; 1526} 1527 1528static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 1529 const struct audio_config *config) 1530{ 1531 if (audio_is_linear_pcm(config->format)) { 1532 size_t max_buffer_period_size_frames = 0; 1533 struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( 1534 const_cast<struct audio_hw_device*>(dev)); 1535 // look for the largest buffer period size 1536 for (int i = 0 ; i < MAX_ROUTES ; i++) { 1537 if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames) 1538 { 1539 max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames; 1540 } 1541 } 1542 const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * 1543 audio_bytes_per_sample(config->format); 1544 const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes; 1545 SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", 1546 buffer_size, buffer_period_size_frames); 1547 return buffer_size; 1548 } 1549 return 0; 1550} 1551 1552static int adev_open_input_stream(struct audio_hw_device *dev, 1553 audio_io_handle_t handle, 1554 audio_devices_t devices, 1555 struct audio_config *config, 1556 struct audio_stream_in **stream_in, 1557 audio_input_flags_t flags __unused, 1558 const char *address, 1559 audio_source_t source __unused) 1560{ 1561 struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); 1562 struct submix_stream_in *in; 1563 ALOGD("adev_open_input_stream(addr=%s)", address); 1564 (void)handle; 1565 (void)devices; 1566 1567 *stream_in = NULL; 1568 1569 // Do we already have a route for this address 1570 int route_idx = -1; 1571 1572 pthread_mutex_lock(&rsxadev->lock); 1573 1574 status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); 1575 if (res != OK) { 1576 ALOGE("Error %d looking for address=%s in adev_open_input_stream", res, address); 1577 pthread_mutex_unlock(&rsxadev->lock); 1578 return res; 1579 } 1580 1581 // Make sure it's possible to open the device given the current audio config. 1582 submix_sanitize_config(config, true); 1583 if (!submix_open_validate_l(rsxadev, route_idx, config, true)) { 1584 ALOGE("adev_open_input_stream(): Unable to open input stream."); 1585 pthread_mutex_unlock(&rsxadev->lock); 1586 return -EINVAL; 1587 } 1588 1589#if ENABLE_LEGACY_INPUT_OPEN 1590 in = rsxadev->routes[route_idx].input; 1591 if (in) { 1592 in->ref_count++; 1593 sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink; 1594 ALOG_ASSERT(sink != NULL); 1595 // If the sink has been shutdown, delete the pipe. 1596 if (sink != NULL) { 1597 if (sink->isShutdown()) { 1598 ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d", 1599 in->ref_count); 1600 submix_audio_device_release_pipe_l(rsxadev, in->route_handle); 1601 } else { 1602 ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count); 1603 } 1604 } else { 1605 ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count); 1606 } 1607 } 1608#else 1609 in = NULL; 1610#endif // ENABLE_LEGACY_INPUT_OPEN 1611 1612 if (!in) { 1613 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 1614 if (!in) return -ENOMEM; 1615 in->ref_count = 1; 1616 1617 // Initialize the function pointer tables (v-tables). 1618 in->stream.common.get_sample_rate = in_get_sample_rate; 1619 in->stream.common.set_sample_rate = in_set_sample_rate; 1620 in->stream.common.get_buffer_size = in_get_buffer_size; 1621 in->stream.common.get_channels = in_get_channels; 1622 in->stream.common.get_format = in_get_format; 1623 in->stream.common.set_format = in_set_format; 1624 in->stream.common.standby = in_standby; 1625 in->stream.common.dump = in_dump; 1626 in->stream.common.set_parameters = in_set_parameters; 1627 in->stream.common.get_parameters = in_get_parameters; 1628 in->stream.common.add_audio_effect = in_add_audio_effect; 1629 in->stream.common.remove_audio_effect = in_remove_audio_effect; 1630 in->stream.set_gain = in_set_gain; 1631 in->stream.read = in_read; 1632 in->stream.get_input_frames_lost = in_get_input_frames_lost; 1633 1634 in->dev = rsxadev; 1635#if LOG_STREAMS_TO_FILES 1636 in->log_fd = -1; 1637#endif 1638 } 1639 1640 // Initialize the input stream. 1641 in->read_counter_frames = 0; 1642 in->input_standby = true; 1643 if (rsxadev->routes[route_idx].output != NULL) { 1644 in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby; 1645 } else { 1646 in->output_standby_rec_thr = true; 1647 } 1648 1649 in->read_error_count = 0; 1650 // Initialize the pipe. 1651 ALOGV("adev_open_input_stream(): about to create pipe"); 1652 submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1653 DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); 1654#if LOG_STREAMS_TO_FILES 1655 if (in->log_fd >= 0) close(in->log_fd); 1656 in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1657 LOG_STREAM_FILE_PERMISSIONS); 1658 ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", 1659 strerror(errno)); 1660 ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); 1661#endif // LOG_STREAMS_TO_FILES 1662 // Return the input stream. 1663 *stream_in = &in->stream; 1664 1665 pthread_mutex_unlock(&rsxadev->lock); 1666 return 0; 1667} 1668 1669static void adev_close_input_stream(struct audio_hw_device *dev, 1670 struct audio_stream_in *stream) 1671{ 1672 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1673 1674 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1675 ALOGD("adev_close_input_stream()"); 1676 pthread_mutex_lock(&rsxadev->lock); 1677 submix_audio_device_destroy_pipe_l(rsxadev, in, NULL); 1678#if LOG_STREAMS_TO_FILES 1679 if (in->log_fd >= 0) close(in->log_fd); 1680#endif // LOG_STREAMS_TO_FILES 1681#if ENABLE_LEGACY_INPUT_OPEN 1682 if (in->ref_count == 0) free(in); 1683#else 1684 free(in); 1685#endif // ENABLE_LEGACY_INPUT_OPEN 1686 1687 pthread_mutex_unlock(&rsxadev->lock); 1688} 1689 1690static int adev_dump(const audio_hw_device_t *device, int fd) 1691{ 1692 const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device); 1693 reinterpret_cast<const struct submix_audio_device *>( 1694 reinterpret_cast<const uint8_t *>(device) - 1695 offsetof(struct submix_audio_device, device)); 1696 char msg[100]; 1697 int n = sprintf(msg, "\nReroute submix audio module:\n"); 1698 write(fd, &msg, n); 1699 for (int i=0 ; i < MAX_ROUTES ; i++) { 1700 n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i, 1701 rsxadev->routes[i].config.input_sample_rate, 1702 rsxadev->routes[i].config.output_sample_rate, 1703 rsxadev->routes[i].address); 1704 write(fd, &msg, n); 1705 } 1706 return 0; 1707} 1708 1709static int adev_close(hw_device_t *device) 1710{ 1711 ALOGI("adev_close()"); 1712 free(device); 1713 return 0; 1714} 1715 1716static int adev_open(const hw_module_t* module, const char* name, 1717 hw_device_t** device) 1718{ 1719 ALOGI("adev_open(name=%s)", name); 1720 struct submix_audio_device *rsxadev; 1721 1722 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 1723 return -EINVAL; 1724 1725 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 1726 if (!rsxadev) 1727 return -ENOMEM; 1728 1729 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 1730 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 1731 rsxadev->device.common.module = (struct hw_module_t *) module; 1732 rsxadev->device.common.close = adev_close; 1733 1734 rsxadev->device.init_check = adev_init_check; 1735 rsxadev->device.set_voice_volume = adev_set_voice_volume; 1736 rsxadev->device.set_master_volume = adev_set_master_volume; 1737 rsxadev->device.get_master_volume = adev_get_master_volume; 1738 rsxadev->device.set_master_mute = adev_set_master_mute; 1739 rsxadev->device.get_master_mute = adev_get_master_mute; 1740 rsxadev->device.set_mode = adev_set_mode; 1741 rsxadev->device.set_mic_mute = adev_set_mic_mute; 1742 rsxadev->device.get_mic_mute = adev_get_mic_mute; 1743 rsxadev->device.set_parameters = adev_set_parameters; 1744 rsxadev->device.get_parameters = adev_get_parameters; 1745 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 1746 rsxadev->device.open_output_stream = adev_open_output_stream; 1747 rsxadev->device.close_output_stream = adev_close_output_stream; 1748 rsxadev->device.open_input_stream = adev_open_input_stream; 1749 rsxadev->device.close_input_stream = adev_close_input_stream; 1750 rsxadev->device.dump = adev_dump; 1751 1752 for (int i=0 ; i < MAX_ROUTES ; i++) { 1753 memset(&rsxadev->routes[i], 0, sizeof(route_config)); 1754 strcpy(rsxadev->routes[i].address, ""); 1755 } 1756 1757 *device = &rsxadev->device.common; 1758 1759 return 0; 1760} 1761 1762static struct hw_module_methods_t hal_module_methods = { 1763 /* open */ adev_open, 1764}; 1765 1766struct audio_module HAL_MODULE_INFO_SYM = { 1767 /* common */ { 1768 /* tag */ HARDWARE_MODULE_TAG, 1769 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 1770 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 1771 /* id */ AUDIO_HARDWARE_MODULE_ID, 1772 /* name */ "Wifi Display audio HAL", 1773 /* author */ "The Android Open Source Project", 1774 /* methods */ &hal_module_methods, 1775 /* dso */ NULL, 1776 /* reserved */ { 0 }, 1777 }, 1778}; 1779 1780} //namespace android 1781 1782} //extern "C" 1783