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 50 extern "C" { 51 52 namespace 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 // limit for number of read error log entries to avoid spamming the logs 99 #define MAX_READ_ERROR_LOGS 5 100 101 // Common limits macros. 102 #ifndef min 103 #define min(a, b) ((a) < (b) ? (a) : (b)) 104 #endif // min 105 #ifndef max 106 #define max(a, b) ((a) > (b) ? (a) : (b)) 107 #endif // max 108 109 // Set *result_variable_ptr to true if value_to_find is present in the array array_to_search, 110 // otherwise set *result_variable_ptr to false. 111 #define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \ 112 { \ 113 size_t i; \ 114 *(result_variable_ptr) = false; \ 115 for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \ 116 if ((value_to_find) == (array_to_search)[i]) { \ 117 *(result_variable_ptr) = true; \ 118 break; \ 119 } \ 120 } \ 121 } 122 123 // Configuration of the submix pipe. 124 struct submix_config { 125 // Channel mask field in this data structure is set to either input_channel_mask or 126 // output_channel_mask depending upon the last stream to be opened on this device. 127 struct audio_config common; 128 // Input stream and output stream channel masks. This is required since input and output 129 // channel bitfields are not equivalent. 130 audio_channel_mask_t input_channel_mask; 131 audio_channel_mask_t output_channel_mask; 132 #if ENABLE_RESAMPLING 133 // Input stream and output stream sample rates. 134 uint32_t input_sample_rate; 135 uint32_t output_sample_rate; 136 #endif // ENABLE_RESAMPLING 137 size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe. 138 size_t buffer_size_frames; // Size of the audio pipe in frames. 139 // Maximum number of frames buffered by the input and output streams. 140 size_t buffer_period_size_frames; 141 }; 142 143 #define MAX_ROUTES 10 144 typedef struct route_config { 145 struct submix_config config; 146 char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; 147 // Pipe variables: they handle the ring buffer that "pipes" audio: 148 // - from the submix virtual audio output == what needs to be played 149 // remotely, seen as an output for AudioFlinger 150 // - to the virtual audio source == what is captured by the component 151 // which "records" the submix / virtual audio source, and handles it as needed. 152 // A usecase example is one where the component capturing the audio is then sending it over 153 // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a 154 // TV with Wifi Display capabilities), or to a wireless audio player. 155 sp<MonoPipe> rsxSink; 156 sp<MonoPipeReader> rsxSource; 157 // Pointers to the current input and output stream instances. rsxSink and rsxSource are 158 // destroyed if both and input and output streams are destroyed. 159 struct submix_stream_out *output; 160 struct submix_stream_in *input; 161 #if ENABLE_RESAMPLING 162 // Buffer used as temporary storage for resampled data prior to returning data to the output 163 // stream. 164 int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES]; 165 #endif // ENABLE_RESAMPLING 166 } route_config_t; 167 168 struct submix_audio_device { 169 struct audio_hw_device device; 170 route_config_t routes[MAX_ROUTES]; 171 // Device lock, also used to protect access to submix_audio_device from the input and output 172 // streams. 173 pthread_mutex_t lock; 174 }; 175 176 struct submix_stream_out { 177 struct audio_stream_out stream; 178 struct submix_audio_device *dev; 179 int route_handle; 180 bool output_standby; 181 #if LOG_STREAMS_TO_FILES 182 int log_fd; 183 #endif // LOG_STREAMS_TO_FILES 184 }; 185 186 struct submix_stream_in { 187 struct audio_stream_in stream; 188 struct submix_audio_device *dev; 189 int route_handle; 190 bool input_standby; 191 bool output_standby_rec_thr; // output standby state as seen from record thread 192 193 // wall clock when recording starts 194 struct timespec record_start_time; 195 // how many frames have been requested to be read 196 int64_t read_counter_frames; 197 198 #if ENABLE_LEGACY_INPUT_OPEN 199 // Number of references to this input stream. 200 volatile int32_t ref_count; 201 #endif // ENABLE_LEGACY_INPUT_OPEN 202 #if LOG_STREAMS_TO_FILES 203 int log_fd; 204 #endif // LOG_STREAMS_TO_FILES 205 206 volatile int16_t read_error_count; 207 }; 208 209 // Determine whether the specified sample rate is supported by the submix module. 210 static bool sample_rate_supported(const uint32_t sample_rate) 211 { 212 // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp. 213 static const unsigned int supported_sample_rates[] = { 214 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 215 }; 216 bool return_value; 217 SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value); 218 return return_value; 219 } 220 221 // Determine whether the specified sample rate is supported, if it is return the specified sample 222 // rate, otherwise return the default sample rate for the submix module. 223 static uint32_t get_supported_sample_rate(uint32_t sample_rate) 224 { 225 return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ; 226 } 227 228 // Determine whether the specified channel in mask is supported by the submix module. 229 static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask) 230 { 231 // Set of channel in masks supported by Format_from_SR_C() 232 // frameworks/av/media/libnbaio/NAIO.cpp. 233 static const audio_channel_mask_t supported_channel_in_masks[] = { 234 AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, 235 }; 236 bool return_value; 237 SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value); 238 return return_value; 239 } 240 241 // Determine whether the specified channel in mask is supported, if it is return the specified 242 // channel in mask, otherwise return the default channel in mask for the submix module. 243 static audio_channel_mask_t get_supported_channel_in_mask( 244 const audio_channel_mask_t channel_in_mask) 245 { 246 return channel_in_mask_supported(channel_in_mask) ? channel_in_mask : 247 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO); 248 } 249 250 // Determine whether the specified channel out mask is supported by the submix module. 251 static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask) 252 { 253 // Set of channel out masks supported by Format_from_SR_C() 254 // frameworks/av/media/libnbaio/NAIO.cpp. 255 static const audio_channel_mask_t supported_channel_out_masks[] = { 256 AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO, 257 }; 258 bool return_value; 259 SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value); 260 return return_value; 261 } 262 263 // Determine whether the specified channel out mask is supported, if it is return the specified 264 // channel out mask, otherwise return the default channel out mask for the submix module. 265 static audio_channel_mask_t get_supported_channel_out_mask( 266 const audio_channel_mask_t channel_out_mask) 267 { 268 return channel_out_mask_supported(channel_out_mask) ? channel_out_mask : 269 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO); 270 } 271 272 // Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the 273 // structure. 274 static struct submix_stream_out * audio_stream_out_get_submix_stream_out( 275 struct audio_stream_out * const stream) 276 { 277 ALOG_ASSERT(stream); 278 return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) - 279 offsetof(struct submix_stream_out, stream)); 280 } 281 282 // Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure. 283 static struct submix_stream_out * audio_stream_get_submix_stream_out( 284 struct audio_stream * const stream) 285 { 286 ALOG_ASSERT(stream); 287 return audio_stream_out_get_submix_stream_out( 288 reinterpret_cast<struct audio_stream_out *>(stream)); 289 } 290 291 // Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the 292 // structure. 293 static struct submix_stream_in * audio_stream_in_get_submix_stream_in( 294 struct audio_stream_in * const stream) 295 { 296 ALOG_ASSERT(stream); 297 return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) - 298 offsetof(struct submix_stream_in, stream)); 299 } 300 301 // Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure. 302 static struct submix_stream_in * audio_stream_get_submix_stream_in( 303 struct audio_stream * const stream) 304 { 305 ALOG_ASSERT(stream); 306 return audio_stream_in_get_submix_stream_in( 307 reinterpret_cast<struct audio_stream_in *>(stream)); 308 } 309 310 // Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within 311 // the structure. 312 static struct submix_audio_device * audio_hw_device_get_submix_audio_device( 313 struct audio_hw_device *device) 314 { 315 ALOG_ASSERT(device); 316 return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) - 317 offsetof(struct submix_audio_device, device)); 318 } 319 320 // Compare an audio_config with input channel mask and an audio_config with output channel mask 321 // returning false if they do *not* match, true otherwise. 322 static bool audio_config_compare(const audio_config * const input_config, 323 const audio_config * const output_config) 324 { 325 #if !ENABLE_CHANNEL_CONVERSION 326 const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask); 327 const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask); 328 if (input_channels != output_channels) { 329 ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d", 330 input_channels, output_channels); 331 return false; 332 } 333 #endif // !ENABLE_CHANNEL_CONVERSION 334 #if ENABLE_RESAMPLING 335 if (input_config->sample_rate != output_config->sample_rate && 336 audio_channel_count_from_in_mask(input_config->channel_mask) != 1) { 337 #else 338 if (input_config->sample_rate != output_config->sample_rate) { 339 #endif // ENABLE_RESAMPLING 340 ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", 341 input_config->sample_rate, output_config->sample_rate); 342 return false; 343 } 344 if (input_config->format != output_config->format) { 345 ALOGE("audio_config_compare() format mismatch %x vs. %x", 346 input_config->format, output_config->format); 347 return false; 348 } 349 // This purposely ignores offload_info as it's not required for the submix device. 350 return true; 351 } 352 353 // If one doesn't exist, create a pipe for the submix audio device rsxadev of size 354 // buffer_size_frames and optionally associate "in" or "out" with the submix audio device. 355 // Must be called with lock held on the submix_audio_device 356 static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev, 357 const struct audio_config * const config, 358 const size_t buffer_size_frames, 359 const uint32_t buffer_period_count, 360 struct submix_stream_in * const in, 361 struct submix_stream_out * const out, 362 const char *address, 363 int route_idx) 364 { 365 ALOG_ASSERT(in || out); 366 ALOG_ASSERT(route_idx > -1); 367 ALOG_ASSERT(route_idx < MAX_ROUTES); 368 ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx); 369 370 // Save a reference to the specified input or output stream and the associated channel 371 // mask. 372 if (in) { 373 in->route_handle = route_idx; 374 rsxadev->routes[route_idx].input = in; 375 rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask; 376 #if ENABLE_RESAMPLING 377 rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate; 378 // If the output isn't configured yet, set the output sample rate to the maximum supported 379 // sample rate such that the smallest possible input buffer is created, and put a default 380 // value for channel count 381 if (!rsxadev->routes[route_idx].output) { 382 rsxadev->routes[route_idx].config.output_sample_rate = 48000; 383 rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO; 384 } 385 #endif // ENABLE_RESAMPLING 386 } 387 if (out) { 388 out->route_handle = route_idx; 389 rsxadev->routes[route_idx].output = out; 390 rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask; 391 #if ENABLE_RESAMPLING 392 rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate; 393 #endif // ENABLE_RESAMPLING 394 } 395 // Save the address 396 strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN); 397 ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx); 398 // If a pipe isn't associated with the device, create one. 399 if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL) 400 { 401 struct submix_config * const device_config = &rsxadev->routes[route_idx].config; 402 uint32_t channel_count; 403 if (out) 404 channel_count = audio_channel_count_from_out_mask(config->channel_mask); 405 else 406 channel_count = audio_channel_count_from_in_mask(config->channel_mask); 407 #if ENABLE_CHANNEL_CONVERSION 408 // If channel conversion is enabled, allocate enough space for the maximum number of 409 // possible channels stored in the pipe for the situation when the number of channels in 410 // the output stream don't match the number in the input stream. 411 const uint32_t pipe_channel_count = max(channel_count, 2); 412 #else 413 const uint32_t pipe_channel_count = channel_count; 414 #endif // ENABLE_CHANNEL_CONVERSION 415 const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, 416 config->format); 417 const NBAIO_Format offers[1] = {format}; 418 size_t numCounterOffers = 0; 419 // Create a MonoPipe with optional blocking set to true. 420 MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); 421 // Negotiation between the source and sink cannot fail as the device open operation 422 // creates both ends of the pipe using the same audio format. 423 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 424 ALOG_ASSERT(index == 0); 425 MonoPipeReader* source = new MonoPipeReader(sink); 426 numCounterOffers = 0; 427 index = source->negotiate(offers, 1, NULL, numCounterOffers); 428 ALOG_ASSERT(index == 0); 429 ALOGV("submix_audio_device_create_pipe_l(): created pipe"); 430 431 // Save references to the source and sink. 432 ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL); 433 ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL); 434 rsxadev->routes[route_idx].rsxSink = sink; 435 rsxadev->routes[route_idx].rsxSource = source; 436 // Store the sanitized audio format in the device so that it's possible to determine 437 // the format of the pipe source when opening the input device. 438 memcpy(&device_config->common, config, sizeof(device_config->common)); 439 device_config->buffer_size_frames = sink->maxFrames(); 440 device_config->buffer_period_size_frames = device_config->buffer_size_frames / 441 buffer_period_count; 442 if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); 443 if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); 444 #if ENABLE_CHANNEL_CONVERSION 445 // Calculate the pipe frame size based upon the number of channels. 446 device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / 447 channel_count; 448 #endif // ENABLE_CHANNEL_CONVERSION 449 SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, " 450 "period size %zd", device_config->pipe_frame_size, 451 device_config->buffer_size_frames, device_config->buffer_period_size_frames); 452 } 453 } 454 455 // Release references to the sink and source. Input and output threads may maintain references 456 // to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use 457 // before they shutdown. 458 // Must be called with lock held on the submix_audio_device 459 static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev, 460 int route_idx) 461 { 462 ALOG_ASSERT(route_idx > -1); 463 ALOG_ASSERT(route_idx < MAX_ROUTES); 464 ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx, 465 rsxadev->routes[route_idx].address); 466 if (rsxadev->routes[route_idx].rsxSink != 0) { 467 rsxadev->routes[route_idx].rsxSink.clear(); 468 rsxadev->routes[route_idx].rsxSink = 0; 469 } 470 if (rsxadev->routes[route_idx].rsxSource != 0) { 471 rsxadev->routes[route_idx].rsxSource.clear(); 472 rsxadev->routes[route_idx].rsxSource = 0; 473 } 474 memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); 475 #ifdef ENABLE_RESAMPLING 476 memset(rsxadev->routes[route_idx].resampler_buffer, 0, 477 sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); 478 #endif 479 } 480 481 // Remove references to the specified input and output streams. When the device no longer 482 // references input and output streams destroy the associated pipe. 483 // Must be called with lock held on the submix_audio_device 484 static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev, 485 const struct submix_stream_in * const in, 486 const struct submix_stream_out * const out) 487 { 488 MonoPipe* sink; 489 ALOGV("submix_audio_device_destroy_pipe_l()"); 490 int route_idx = -1; 491 if (in != NULL) { 492 #if ENABLE_LEGACY_INPUT_OPEN 493 const_cast<struct submix_stream_in*>(in)->ref_count--; 494 route_idx = in->route_handle; 495 ALOG_ASSERT(rsxadev->routes[route_idx].input == in); 496 if (in->ref_count == 0) { 497 rsxadev->routes[route_idx].input = NULL; 498 } 499 ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count); 500 #else 501 rsxadev->input = NULL; 502 #endif // ENABLE_LEGACY_INPUT_OPEN 503 } 504 if (out != NULL) { 505 route_idx = out->route_handle; 506 ALOG_ASSERT(rsxadev->routes[route_idx].output == out); 507 rsxadev->routes[route_idx].output = NULL; 508 } 509 if (route_idx != -1 && 510 rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) { 511 submix_audio_device_release_pipe_l(rsxadev, route_idx); 512 ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed"); 513 } 514 } 515 516 // Sanitize the user specified audio config for a submix input / output stream. 517 static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) 518 { 519 config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : 520 get_supported_channel_out_mask(config->channel_mask); 521 config->sample_rate = get_supported_sample_rate(config->sample_rate); 522 config->format = DEFAULT_FORMAT; 523 } 524 525 // Verify a submix input or output stream can be opened. 526 // Must be called with lock held on the submix_audio_device 527 static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev, 528 int route_idx, 529 const struct audio_config * const config, 530 const bool opening_input) 531 { 532 bool input_open; 533 bool output_open; 534 audio_config pipe_config; 535 536 // Query the device for the current audio config and whether input and output streams are open. 537 output_open = rsxadev->routes[route_idx].output != NULL; 538 input_open = rsxadev->routes[route_idx].input != NULL; 539 memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config)); 540 541 // If the stream is already open, don't open it again. 542 if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { 543 ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" : 544 "Output"); 545 return false; 546 } 547 548 SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x " 549 "%s_channel_mask=%x", config->sample_rate, config->format, 550 opening_input ? "in" : "out", config->channel_mask); 551 552 // If either stream is open, verify the existing audio config the pipe matches the user 553 // specified config. 554 if (input_open || output_open) { 555 const audio_config * const input_config = opening_input ? config : &pipe_config; 556 const audio_config * const output_config = opening_input ? &pipe_config : config; 557 // Get the channel mask of the open device. 558 pipe_config.channel_mask = 559 opening_input ? rsxadev->routes[route_idx].config.output_channel_mask : 560 rsxadev->routes[route_idx].config.input_channel_mask; 561 if (!audio_config_compare(input_config, output_config)) { 562 ALOGE("submix_open_validate_l(): Unsupported format."); 563 return false; 564 } 565 } 566 return true; 567 } 568 569 // Must be called with lock held on the submix_audio_device 570 static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev, 571 const char* address, /*in*/ 572 int *idx /*out*/) 573 { 574 // Do we already have a route for this address 575 int route_idx = -1; 576 int route_empty_idx = -1; // index of an empty route slot that can be used if needed 577 for (int i=0 ; i < MAX_ROUTES ; i++) { 578 if (strcmp(rsxadev->routes[i].address, "") == 0) { 579 route_empty_idx = i; 580 } 581 if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { 582 route_idx = i; 583 break; 584 } 585 } 586 587 if ((route_idx == -1) && (route_empty_idx == -1)) { 588 ALOGE("Cannot create new route for address %s, max number of routes reached", address); 589 return -ENOMEM; 590 } 591 if (route_idx == -1) { 592 route_idx = route_empty_idx; 593 } 594 *idx = route_idx; 595 return OK; 596 } 597 598 599 // Calculate the maximum size of the pipe buffer in frames for the specified stream. 600 static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, 601 const struct submix_config *config, 602 const size_t pipe_frames, 603 const size_t stream_frame_size) 604 { 605 const size_t pipe_frame_size = config->pipe_frame_size; 606 const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); 607 return (pipe_frames * config->pipe_frame_size) / max_frame_size; 608 } 609 610 /* audio HAL functions */ 611 612 static uint32_t out_get_sample_rate(const struct audio_stream *stream) 613 { 614 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 615 const_cast<struct audio_stream *>(stream)); 616 #if ENABLE_RESAMPLING 617 const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; 618 #else 619 const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate; 620 #endif // ENABLE_RESAMPLING 621 SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", 622 out_rate, out->dev->routes[out->route_handle].address); 623 return out_rate; 624 } 625 626 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 627 { 628 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 629 #if ENABLE_RESAMPLING 630 // The sample rate of the stream can't be changed once it's set since this would change the 631 // output buffer size and hence break playback to the shared pipe. 632 if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) { 633 ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " 634 "%u to %u for addr %s", 635 out->dev->routes[out->route_handle].config.output_sample_rate, rate, 636 out->dev->routes[out->route_handle].address); 637 return -ENOSYS; 638 } 639 #endif // ENABLE_RESAMPLING 640 if (!sample_rate_supported(rate)) { 641 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 642 return -ENOSYS; 643 } 644 SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); 645 out->dev->routes[out->route_handle].config.common.sample_rate = rate; 646 return 0; 647 } 648 649 static size_t out_get_buffer_size(const struct audio_stream *stream) 650 { 651 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 652 const_cast<struct audio_stream *>(stream)); 653 const struct submix_config * const config = &out->dev->routes[out->route_handle].config; 654 const size_t stream_frame_size = 655 audio_stream_out_frame_size((const struct audio_stream_out *)stream); 656 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 657 stream, config, config->buffer_period_size_frames, stream_frame_size); 658 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 659 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", 660 buffer_size_bytes, buffer_size_frames); 661 return buffer_size_bytes; 662 } 663 664 static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 665 { 666 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 667 const_cast<struct audio_stream *>(stream)); 668 uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask; 669 SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); 670 return channel_mask; 671 } 672 673 static audio_format_t out_get_format(const struct audio_stream *stream) 674 { 675 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 676 const_cast<struct audio_stream *>(stream)); 677 const audio_format_t format = out->dev->routes[out->route_handle].config.common.format; 678 SUBMIX_ALOGV("out_get_format() returns %x", format); 679 return format; 680 } 681 682 static int out_set_format(struct audio_stream *stream, audio_format_t format) 683 { 684 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 685 if (format != out->dev->routes[out->route_handle].config.common.format) { 686 ALOGE("out_set_format(format=%x) format unsupported", format); 687 return -ENOSYS; 688 } 689 SUBMIX_ALOGV("out_set_format(format=%x)", format); 690 return 0; 691 } 692 693 static int out_standby(struct audio_stream *stream) 694 { 695 ALOGI("out_standby()"); 696 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 697 struct submix_audio_device * const rsxadev = out->dev; 698 699 pthread_mutex_lock(&rsxadev->lock); 700 701 out->output_standby = true; 702 703 pthread_mutex_unlock(&rsxadev->lock); 704 705 return 0; 706 } 707 708 static int out_dump(const struct audio_stream *stream, int fd) 709 { 710 (void)stream; 711 (void)fd; 712 return 0; 713 } 714 715 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 716 { 717 int exiting = -1; 718 AudioParameter parms = AudioParameter(String8(kvpairs)); 719 SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs); 720 721 // FIXME this is using hard-coded strings but in the future, this functionality will be 722 // converted to use audio HAL extensions required to support tunneling 723 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { 724 struct submix_audio_device * const rsxadev = 725 audio_stream_get_submix_stream_out(stream)->dev; 726 pthread_mutex_lock(&rsxadev->lock); 727 { // using the sink 728 sp<MonoPipe> sink = 729 rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle] 730 .rsxSink; 731 if (sink == NULL) { 732 pthread_mutex_unlock(&rsxadev->lock); 733 return 0; 734 } 735 736 ALOGD("out_set_parameters(): shutting down MonoPipe sink"); 737 sink->shutdown(true); 738 } // done using the sink 739 pthread_mutex_unlock(&rsxadev->lock); 740 } 741 return 0; 742 } 743 744 static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 745 { 746 (void)stream; 747 (void)keys; 748 return strdup(""); 749 } 750 751 static uint32_t out_get_latency(const struct audio_stream_out *stream) 752 { 753 const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( 754 const_cast<struct audio_stream_out *>(stream)); 755 const struct submix_config * const config = &out->dev->routes[out->route_handle].config; 756 const size_t stream_frame_size = 757 audio_stream_out_frame_size(stream); 758 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 759 &stream->common, config, config->buffer_size_frames, stream_frame_size); 760 const uint32_t sample_rate = out_get_sample_rate(&stream->common); 761 const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate; 762 SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", 763 latency_ms, buffer_size_frames, sample_rate); 764 return latency_ms; 765 } 766 767 static int out_set_volume(struct audio_stream_out *stream, float left, 768 float right) 769 { 770 (void)stream; 771 (void)left; 772 (void)right; 773 return -ENOSYS; 774 } 775 776 static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 777 size_t bytes) 778 { 779 SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); 780 ssize_t written_frames = 0; 781 const size_t frame_size = audio_stream_out_frame_size(stream); 782 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 783 struct submix_audio_device * const rsxadev = out->dev; 784 const size_t frames = bytes / frame_size; 785 786 pthread_mutex_lock(&rsxadev->lock); 787 788 out->output_standby = false; 789 790 sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink; 791 if (sink != NULL) { 792 if (sink->isShutdown()) { 793 sink.clear(); 794 pthread_mutex_unlock(&rsxadev->lock); 795 SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); 796 // the pipe has already been shutdown, this buffer will be lost but we must 797 // simulate timing so we don't drain the output faster than realtime 798 usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); 799 return bytes; 800 } 801 } else { 802 pthread_mutex_unlock(&rsxadev->lock); 803 ALOGE("out_write without a pipe!"); 804 ALOG_ASSERT("out_write without a pipe!"); 805 return 0; 806 } 807 808 // If the write to the sink would block when no input stream is present, flush enough frames 809 // from the pipe to make space to write the most recent data. 810 { 811 const size_t availableToWrite = sink->availableToWrite(); 812 sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource; 813 if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) { 814 static uint8_t flush_buffer[64]; 815 const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; 816 size_t frames_to_flush_from_source = frames - availableToWrite; 817 SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", 818 frames_to_flush_from_source); 819 while (frames_to_flush_from_source) { 820 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); 821 frames_to_flush_from_source -= flush_size; 822 // read does not block 823 source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); 824 } 825 } 826 } 827 828 pthread_mutex_unlock(&rsxadev->lock); 829 830 written_frames = sink->write(buffer, frames); 831 832 #if LOG_STREAMS_TO_FILES 833 if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); 834 #endif // LOG_STREAMS_TO_FILES 835 836 if (written_frames < 0) { 837 if (written_frames == (ssize_t)NEGOTIATE) { 838 ALOGE("out_write() write to pipe returned NEGOTIATE"); 839 840 pthread_mutex_lock(&rsxadev->lock); 841 sink.clear(); 842 pthread_mutex_unlock(&rsxadev->lock); 843 844 written_frames = 0; 845 return 0; 846 } else { 847 // write() returned UNDERRUN or WOULD_BLOCK, retry 848 ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); 849 written_frames = sink->write(buffer, frames); 850 } 851 } 852 853 pthread_mutex_lock(&rsxadev->lock); 854 sink.clear(); 855 pthread_mutex_unlock(&rsxadev->lock); 856 857 if (written_frames < 0) { 858 ALOGE("out_write() failed writing to pipe with %zd", written_frames); 859 return 0; 860 } 861 const ssize_t written_bytes = written_frames * frame_size; 862 SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); 863 return written_bytes; 864 } 865 866 static int out_get_render_position(const struct audio_stream_out *stream, 867 uint32_t *dsp_frames) 868 { 869 (void)stream; 870 (void)dsp_frames; 871 return -EINVAL; 872 } 873 874 static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 875 { 876 (void)stream; 877 (void)effect; 878 return 0; 879 } 880 881 static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 882 { 883 (void)stream; 884 (void)effect; 885 return 0; 886 } 887 888 static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 889 int64_t *timestamp) 890 { 891 (void)stream; 892 (void)timestamp; 893 return -EINVAL; 894 } 895 896 /** audio_stream_in implementation **/ 897 static uint32_t in_get_sample_rate(const struct audio_stream *stream) 898 { 899 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 900 const_cast<struct audio_stream*>(stream)); 901 #if ENABLE_RESAMPLING 902 const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; 903 #else 904 const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate; 905 #endif // ENABLE_RESAMPLING 906 SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); 907 return rate; 908 } 909 910 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 911 { 912 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 913 #if ENABLE_RESAMPLING 914 // The sample rate of the stream can't be changed once it's set since this would change the 915 // input buffer size and hence break recording from the shared pipe. 916 if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) { 917 ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " 918 "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); 919 return -ENOSYS; 920 } 921 #endif // ENABLE_RESAMPLING 922 if (!sample_rate_supported(rate)) { 923 ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); 924 return -ENOSYS; 925 } 926 in->dev->routes[in->route_handle].config.common.sample_rate = rate; 927 SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); 928 return 0; 929 } 930 931 static size_t in_get_buffer_size(const struct audio_stream *stream) 932 { 933 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 934 const_cast<struct audio_stream*>(stream)); 935 const struct submix_config * const config = &in->dev->routes[in->route_handle].config; 936 const size_t stream_frame_size = 937 audio_stream_in_frame_size((const struct audio_stream_in *)stream); 938 size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 939 stream, config, config->buffer_period_size_frames, stream_frame_size); 940 #if ENABLE_RESAMPLING 941 // Scale the size of the buffer based upon the maximum number of frames that could be returned 942 // given the ratio of output to input sample rate. 943 buffer_size_frames = (size_t)(((float)buffer_size_frames * 944 (float)config->input_sample_rate) / 945 (float)config->output_sample_rate); 946 #endif // ENABLE_RESAMPLING 947 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 948 SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, 949 buffer_size_frames); 950 return buffer_size_bytes; 951 } 952 953 static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 954 { 955 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 956 const_cast<struct audio_stream*>(stream)); 957 const audio_channel_mask_t channel_mask = 958 in->dev->routes[in->route_handle].config.input_channel_mask; 959 SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); 960 return channel_mask; 961 } 962 963 static audio_format_t in_get_format(const struct audio_stream *stream) 964 { 965 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 966 const_cast<struct audio_stream*>(stream)); 967 const audio_format_t format = in->dev->routes[in->route_handle].config.common.format; 968 SUBMIX_ALOGV("in_get_format() returns %x", format); 969 return format; 970 } 971 972 static int in_set_format(struct audio_stream *stream, audio_format_t format) 973 { 974 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 975 if (format != in->dev->routes[in->route_handle].config.common.format) { 976 ALOGE("in_set_format(format=%x) format unsupported", format); 977 return -ENOSYS; 978 } 979 SUBMIX_ALOGV("in_set_format(format=%x)", format); 980 return 0; 981 } 982 983 static int in_standby(struct audio_stream *stream) 984 { 985 ALOGI("in_standby()"); 986 struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 987 struct submix_audio_device * const rsxadev = in->dev; 988 989 pthread_mutex_lock(&rsxadev->lock); 990 991 in->input_standby = true; 992 993 pthread_mutex_unlock(&rsxadev->lock); 994 995 return 0; 996 } 997 998 static int in_dump(const struct audio_stream *stream, int fd) 999 { 1000 (void)stream; 1001 (void)fd; 1002 return 0; 1003 } 1004 1005 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 1006 { 1007 (void)stream; 1008 (void)kvpairs; 1009 return 0; 1010 } 1011 1012 static char * in_get_parameters(const struct audio_stream *stream, 1013 const char *keys) 1014 { 1015 (void)stream; 1016 (void)keys; 1017 return strdup(""); 1018 } 1019 1020 static int in_set_gain(struct audio_stream_in *stream, float gain) 1021 { 1022 (void)stream; 1023 (void)gain; 1024 return 0; 1025 } 1026 1027 static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 1028 size_t bytes) 1029 { 1030 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1031 struct submix_audio_device * const rsxadev = in->dev; 1032 struct audio_config *format; 1033 const size_t frame_size = audio_stream_in_frame_size(stream); 1034 const size_t frames_to_read = bytes / frame_size; 1035 1036 SUBMIX_ALOGV("in_read bytes=%zu", bytes); 1037 pthread_mutex_lock(&rsxadev->lock); 1038 1039 const bool output_standby = rsxadev->routes[in->route_handle].output == NULL 1040 ? true : rsxadev->routes[in->route_handle].output->output_standby; 1041 const bool output_standby_transition = (in->output_standby_rec_thr != output_standby); 1042 in->output_standby_rec_thr = output_standby; 1043 1044 if (in->input_standby || output_standby_transition) { 1045 in->input_standby = false; 1046 // keep track of when we exit input standby (== first read == start "real recording") 1047 // or when we start recording silence, and reset projected time 1048 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 1049 if (rc == 0) { 1050 in->read_counter_frames = 0; 1051 } 1052 } 1053 1054 in->read_counter_frames += frames_to_read; 1055 size_t remaining_frames = frames_to_read; 1056 1057 { 1058 // about to read from audio source 1059 sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource; 1060 if (source == NULL) { 1061 in->read_error_count++;// ok if it rolls over 1062 ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS, 1063 "no audio pipe yet we're trying to read! (not all errors will be logged)"); 1064 pthread_mutex_unlock(&rsxadev->lock); 1065 usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); 1066 memset(buffer, 0, bytes); 1067 return bytes; 1068 } 1069 1070 pthread_mutex_unlock(&rsxadev->lock); 1071 1072 // read the data from the pipe (it's non blocking) 1073 int attempts = 0; 1074 char* buff = (char*)buffer; 1075 #if ENABLE_CHANNEL_CONVERSION 1076 // Determine whether channel conversion is required. 1077 const uint32_t input_channels = audio_channel_count_from_in_mask( 1078 rsxadev->routes[in->route_handle].config.input_channel_mask); 1079 const uint32_t output_channels = audio_channel_count_from_out_mask( 1080 rsxadev->routes[in->route_handle].config.output_channel_mask); 1081 if (input_channels != output_channels) { 1082 SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " 1083 "input channels", output_channels, input_channels); 1084 // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. 1085 ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == 1086 AUDIO_FORMAT_PCM_16_BIT); 1087 ALOG_ASSERT((input_channels == 1 && output_channels == 2) || 1088 (input_channels == 2 && output_channels == 1)); 1089 } 1090 #endif // ENABLE_CHANNEL_CONVERSION 1091 1092 #if ENABLE_RESAMPLING 1093 const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); 1094 const uint32_t output_sample_rate = 1095 rsxadev->routes[in->route_handle].config.output_sample_rate; 1096 const size_t resampler_buffer_size_frames = 1097 sizeof(rsxadev->routes[in->route_handle].resampler_buffer) / 1098 sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]); 1099 float resampler_ratio = 1.0f; 1100 // Determine whether resampling is required. 1101 if (input_sample_rate != output_sample_rate) { 1102 resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; 1103 // Only support 16-bit PCM mono resampling. 1104 // NOTE: Resampling is performed after the channel conversion step. 1105 ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == 1106 AUDIO_FORMAT_PCM_16_BIT); 1107 ALOG_ASSERT(audio_channel_count_from_in_mask( 1108 rsxadev->routes[in->route_handle].config.input_channel_mask) == 1); 1109 } 1110 #endif // ENABLE_RESAMPLING 1111 1112 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 1113 ssize_t frames_read = -1977; 1114 size_t read_frames = remaining_frames; 1115 #if ENABLE_RESAMPLING 1116 char* const saved_buff = buff; 1117 if (resampler_ratio != 1.0f) { 1118 // Calculate the number of frames from the pipe that need to be read to generate 1119 // the data for the input stream read. 1120 const size_t frames_required_for_resampler = (size_t)( 1121 (float)read_frames * (float)resampler_ratio); 1122 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); 1123 // Read into the resampler buffer. 1124 buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; 1125 } 1126 #endif // ENABLE_RESAMPLING 1127 #if ENABLE_CHANNEL_CONVERSION 1128 if (output_channels == 1 && input_channels == 2) { 1129 // Need to read half the requested frames since the converted output 1130 // data will take twice the space (mono->stereo). 1131 read_frames /= 2; 1132 } 1133 #endif // ENABLE_CHANNEL_CONVERSION 1134 1135 SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); 1136 1137 frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS); 1138 1139 SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); 1140 1141 #if ENABLE_CHANNEL_CONVERSION 1142 // Perform in-place channel conversion. 1143 // NOTE: In the following "input stream" refers to the data returned by this function 1144 // and "output stream" refers to the data read from the pipe. 1145 if (input_channels != output_channels && frames_read > 0) { 1146 int16_t *data = (int16_t*)buff; 1147 if (output_channels == 2 && input_channels == 1) { 1148 // Offset into the output stream data in samples. 1149 ssize_t output_stream_offset = 0; 1150 for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; 1151 input_stream_frame++, output_stream_offset += 2) { 1152 // Average the content from both channels. 1153 data[input_stream_frame] = ((int32_t)data[output_stream_offset] + 1154 (int32_t)data[output_stream_offset + 1]) / 2; 1155 } 1156 } else if (output_channels == 1 && input_channels == 2) { 1157 // Offset into the input stream data in samples. 1158 ssize_t input_stream_offset = (frames_read - 1) * 2; 1159 for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; 1160 output_stream_frame--, input_stream_offset -= 2) { 1161 const short sample = data[output_stream_frame]; 1162 data[input_stream_offset] = sample; 1163 data[input_stream_offset + 1] = sample; 1164 } 1165 } 1166 } 1167 #endif // ENABLE_CHANNEL_CONVERSION 1168 1169 #if ENABLE_RESAMPLING 1170 if (resampler_ratio != 1.0f) { 1171 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); 1172 const int16_t * const data = (int16_t*)buff; 1173 int16_t * const resampled_buffer = (int16_t*)saved_buff; 1174 // Resample with *no* filtering - if the data from the ouptut stream was really 1175 // sampled at a different rate this will result in very nasty aliasing. 1176 const float output_stream_frames = (float)frames_read; 1177 size_t input_stream_frame = 0; 1178 for (float output_stream_frame = 0.0f; 1179 output_stream_frame < output_stream_frames && 1180 input_stream_frame < remaining_frames; 1181 output_stream_frame += resampler_ratio, input_stream_frame++) { 1182 resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; 1183 } 1184 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); 1185 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); 1186 frames_read = input_stream_frame; 1187 buff = saved_buff; 1188 } 1189 #endif // ENABLE_RESAMPLING 1190 1191 if (frames_read > 0) { 1192 #if LOG_STREAMS_TO_FILES 1193 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); 1194 #endif // LOG_STREAMS_TO_FILES 1195 1196 remaining_frames -= frames_read; 1197 buff += frames_read * frame_size; 1198 SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", 1199 attempts, frames_read, remaining_frames); 1200 } else { 1201 attempts++; 1202 SUBMIX_ALOGE(" in_read read returned %zd", frames_read); 1203 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 1204 } 1205 } 1206 // done using the source 1207 pthread_mutex_lock(&rsxadev->lock); 1208 source.clear(); 1209 pthread_mutex_unlock(&rsxadev->lock); 1210 } 1211 1212 if (remaining_frames > 0) { 1213 const size_t remaining_bytes = remaining_frames * frame_size; 1214 SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames); 1215 memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); 1216 } 1217 1218 // compute how much we need to sleep after reading the data by comparing the wall clock with 1219 // the projected time at which we should return. 1220 struct timespec time_after_read;// wall clock after reading from the pipe 1221 struct timespec record_duration;// observed record duration 1222 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 1223 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 1224 if (rc == 0) { 1225 // for how long have we been recording? 1226 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 1227 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 1228 if (record_duration.tv_nsec < 0) { 1229 record_duration.tv_sec--; 1230 record_duration.tv_nsec += 1000000000; 1231 } 1232 1233 // read_counter_frames contains the number of frames that have been read since the 1234 // beginning of recording (including this call): it's converted to usec and compared to 1235 // how long we've been recording for, which gives us how long we must wait to sync the 1236 // projected recording time, and the observed recording time. 1237 long projected_vs_observed_offset_us = 1238 ((int64_t)(in->read_counter_frames 1239 - (record_duration.tv_sec*sample_rate))) 1240 * 1000000 / sample_rate 1241 - (record_duration.tv_nsec / 1000); 1242 1243 SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 1244 record_duration.tv_sec, record_duration.tv_nsec/1000000, 1245 projected_vs_observed_offset_us); 1246 if (projected_vs_observed_offset_us > 0) { 1247 usleep(projected_vs_observed_offset_us); 1248 } 1249 } 1250 1251 SUBMIX_ALOGV("in_read returns %zu", bytes); 1252 return bytes; 1253 1254 } 1255 1256 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 1257 { 1258 (void)stream; 1259 return 0; 1260 } 1261 1262 static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1263 { 1264 (void)stream; 1265 (void)effect; 1266 return 0; 1267 } 1268 1269 static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1270 { 1271 (void)stream; 1272 (void)effect; 1273 return 0; 1274 } 1275 1276 static int adev_open_output_stream(struct audio_hw_device *dev, 1277 audio_io_handle_t handle, 1278 audio_devices_t devices, 1279 audio_output_flags_t flags, 1280 struct audio_config *config, 1281 struct audio_stream_out **stream_out, 1282 const char *address) 1283 { 1284 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1285 ALOGD("adev_open_output_stream(address=%s)", address); 1286 struct submix_stream_out *out; 1287 bool force_pipe_creation = false; 1288 (void)handle; 1289 (void)devices; 1290 (void)flags; 1291 1292 *stream_out = NULL; 1293 1294 // Make sure it's possible to open the device given the current audio config. 1295 submix_sanitize_config(config, false); 1296 1297 int route_idx = -1; 1298 1299 pthread_mutex_lock(&rsxadev->lock); 1300 1301 status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); 1302 if (res != OK) { 1303 ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); 1304 pthread_mutex_unlock(&rsxadev->lock); 1305 return res; 1306 } 1307 1308 if (!submix_open_validate_l(rsxadev, route_idx, config, false)) { 1309 ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address); 1310 pthread_mutex_unlock(&rsxadev->lock); 1311 return -EINVAL; 1312 } 1313 1314 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 1315 if (!out) { 1316 pthread_mutex_unlock(&rsxadev->lock); 1317 return -ENOMEM; 1318 } 1319 1320 // Initialize the function pointer tables (v-tables). 1321 out->stream.common.get_sample_rate = out_get_sample_rate; 1322 out->stream.common.set_sample_rate = out_set_sample_rate; 1323 out->stream.common.get_buffer_size = out_get_buffer_size; 1324 out->stream.common.get_channels = out_get_channels; 1325 out->stream.common.get_format = out_get_format; 1326 out->stream.common.set_format = out_set_format; 1327 out->stream.common.standby = out_standby; 1328 out->stream.common.dump = out_dump; 1329 out->stream.common.set_parameters = out_set_parameters; 1330 out->stream.common.get_parameters = out_get_parameters; 1331 out->stream.common.add_audio_effect = out_add_audio_effect; 1332 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1333 out->stream.get_latency = out_get_latency; 1334 out->stream.set_volume = out_set_volume; 1335 out->stream.write = out_write; 1336 out->stream.get_render_position = out_get_render_position; 1337 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1338 1339 #if ENABLE_RESAMPLING 1340 // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits 1341 // writes correctly. 1342 force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate 1343 != config->sample_rate; 1344 #endif // ENABLE_RESAMPLING 1345 1346 // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so 1347 // that it's recreated. 1348 if ((rsxadev->routes[route_idx].rsxSink != NULL 1349 && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { 1350 submix_audio_device_release_pipe_l(rsxadev, route_idx); 1351 } 1352 1353 // Store a pointer to the device from the output stream. 1354 out->dev = rsxadev; 1355 // Initialize the pipe. 1356 ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx); 1357 submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1358 DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx); 1359 #if LOG_STREAMS_TO_FILES 1360 out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1361 LOG_STREAM_FILE_PERMISSIONS); 1362 ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", 1363 strerror(errno)); 1364 ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); 1365 #endif // LOG_STREAMS_TO_FILES 1366 // Return the output stream. 1367 *stream_out = &out->stream; 1368 1369 pthread_mutex_unlock(&rsxadev->lock); 1370 return 0; 1371 } 1372 1373 static void adev_close_output_stream(struct audio_hw_device *dev, 1374 struct audio_stream_out *stream) 1375 { 1376 struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( 1377 const_cast<struct audio_hw_device*>(dev)); 1378 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 1379 1380 pthread_mutex_lock(&rsxadev->lock); 1381 ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address); 1382 submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out); 1383 #if LOG_STREAMS_TO_FILES 1384 if (out->log_fd >= 0) close(out->log_fd); 1385 #endif // LOG_STREAMS_TO_FILES 1386 1387 pthread_mutex_unlock(&rsxadev->lock); 1388 free(out); 1389 } 1390 1391 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1392 { 1393 (void)dev; 1394 (void)kvpairs; 1395 return -ENOSYS; 1396 } 1397 1398 static char * adev_get_parameters(const struct audio_hw_device *dev, 1399 const char *keys) 1400 { 1401 (void)dev; 1402 (void)keys; 1403 return strdup("");; 1404 } 1405 1406 static int adev_init_check(const struct audio_hw_device *dev) 1407 { 1408 ALOGI("adev_init_check()"); 1409 (void)dev; 1410 return 0; 1411 } 1412 1413 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 1414 { 1415 (void)dev; 1416 (void)volume; 1417 return -ENOSYS; 1418 } 1419 1420 static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 1421 { 1422 (void)dev; 1423 (void)volume; 1424 return -ENOSYS; 1425 } 1426 1427 static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 1428 { 1429 (void)dev; 1430 (void)volume; 1431 return -ENOSYS; 1432 } 1433 1434 static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 1435 { 1436 (void)dev; 1437 (void)muted; 1438 return -ENOSYS; 1439 } 1440 1441 static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 1442 { 1443 (void)dev; 1444 (void)muted; 1445 return -ENOSYS; 1446 } 1447 1448 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 1449 { 1450 (void)dev; 1451 (void)mode; 1452 return 0; 1453 } 1454 1455 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 1456 { 1457 (void)dev; 1458 (void)state; 1459 return -ENOSYS; 1460 } 1461 1462 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 1463 { 1464 (void)dev; 1465 (void)state; 1466 return -ENOSYS; 1467 } 1468 1469 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 1470 const struct audio_config *config) 1471 { 1472 if (audio_is_linear_pcm(config->format)) { 1473 size_t max_buffer_period_size_frames = 0; 1474 struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( 1475 const_cast<struct audio_hw_device*>(dev)); 1476 // look for the largest buffer period size 1477 for (int i = 0 ; i < MAX_ROUTES ; i++) { 1478 if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames) 1479 { 1480 max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames; 1481 } 1482 } 1483 const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * 1484 audio_bytes_per_sample(config->format); 1485 const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes; 1486 SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", 1487 buffer_size, buffer_period_size_frames); 1488 return buffer_size; 1489 } 1490 return 0; 1491 } 1492 1493 static int adev_open_input_stream(struct audio_hw_device *dev, 1494 audio_io_handle_t handle, 1495 audio_devices_t devices, 1496 struct audio_config *config, 1497 struct audio_stream_in **stream_in, 1498 audio_input_flags_t flags __unused, 1499 const char *address, 1500 audio_source_t source __unused) 1501 { 1502 struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); 1503 struct submix_stream_in *in; 1504 ALOGD("adev_open_input_stream(addr=%s)", address); 1505 (void)handle; 1506 (void)devices; 1507 1508 *stream_in = NULL; 1509 1510 // Do we already have a route for this address 1511 int route_idx = -1; 1512 1513 pthread_mutex_lock(&rsxadev->lock); 1514 1515 status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); 1516 if (res != OK) { 1517 ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); 1518 pthread_mutex_unlock(&rsxadev->lock); 1519 return res; 1520 } 1521 1522 // Make sure it's possible to open the device given the current audio config. 1523 submix_sanitize_config(config, true); 1524 if (!submix_open_validate_l(rsxadev, route_idx, config, true)) { 1525 ALOGE("adev_open_input_stream(): Unable to open input stream."); 1526 pthread_mutex_unlock(&rsxadev->lock); 1527 return -EINVAL; 1528 } 1529 1530 #if ENABLE_LEGACY_INPUT_OPEN 1531 in = rsxadev->routes[route_idx].input; 1532 if (in) { 1533 in->ref_count++; 1534 sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink; 1535 ALOG_ASSERT(sink != NULL); 1536 // If the sink has been shutdown, delete the pipe. 1537 if (sink != NULL) { 1538 if (sink->isShutdown()) { 1539 ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d", 1540 in->ref_count); 1541 submix_audio_device_release_pipe_l(rsxadev, in->route_handle); 1542 } else { 1543 ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count); 1544 } 1545 } else { 1546 ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count); 1547 } 1548 } 1549 #else 1550 in = NULL; 1551 #endif // ENABLE_LEGACY_INPUT_OPEN 1552 1553 if (!in) { 1554 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 1555 if (!in) return -ENOMEM; 1556 in->ref_count = 1; 1557 1558 // Initialize the function pointer tables (v-tables). 1559 in->stream.common.get_sample_rate = in_get_sample_rate; 1560 in->stream.common.set_sample_rate = in_set_sample_rate; 1561 in->stream.common.get_buffer_size = in_get_buffer_size; 1562 in->stream.common.get_channels = in_get_channels; 1563 in->stream.common.get_format = in_get_format; 1564 in->stream.common.set_format = in_set_format; 1565 in->stream.common.standby = in_standby; 1566 in->stream.common.dump = in_dump; 1567 in->stream.common.set_parameters = in_set_parameters; 1568 in->stream.common.get_parameters = in_get_parameters; 1569 in->stream.common.add_audio_effect = in_add_audio_effect; 1570 in->stream.common.remove_audio_effect = in_remove_audio_effect; 1571 in->stream.set_gain = in_set_gain; 1572 in->stream.read = in_read; 1573 in->stream.get_input_frames_lost = in_get_input_frames_lost; 1574 1575 in->dev = rsxadev; 1576 #if LOG_STREAMS_TO_FILES 1577 in->log_fd = -1; 1578 #endif 1579 } 1580 1581 // Initialize the input stream. 1582 in->read_counter_frames = 0; 1583 in->input_standby = true; 1584 if (rsxadev->routes[route_idx].output != NULL) { 1585 in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby; 1586 } else { 1587 in->output_standby_rec_thr = true; 1588 } 1589 1590 in->read_error_count = 0; 1591 // Initialize the pipe. 1592 ALOGV("adev_open_input_stream(): about to create pipe"); 1593 submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1594 DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); 1595 #if LOG_STREAMS_TO_FILES 1596 if (in->log_fd >= 0) close(in->log_fd); 1597 in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1598 LOG_STREAM_FILE_PERMISSIONS); 1599 ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", 1600 strerror(errno)); 1601 ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); 1602 #endif // LOG_STREAMS_TO_FILES 1603 // Return the input stream. 1604 *stream_in = &in->stream; 1605 1606 pthread_mutex_unlock(&rsxadev->lock); 1607 return 0; 1608 } 1609 1610 static void adev_close_input_stream(struct audio_hw_device *dev, 1611 struct audio_stream_in *stream) 1612 { 1613 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1614 1615 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1616 ALOGD("adev_close_input_stream()"); 1617 pthread_mutex_lock(&rsxadev->lock); 1618 submix_audio_device_destroy_pipe_l(rsxadev, in, NULL); 1619 #if LOG_STREAMS_TO_FILES 1620 if (in->log_fd >= 0) close(in->log_fd); 1621 #endif // LOG_STREAMS_TO_FILES 1622 #if ENABLE_LEGACY_INPUT_OPEN 1623 if (in->ref_count == 0) free(in); 1624 #else 1625 free(in); 1626 #endif // ENABLE_LEGACY_INPUT_OPEN 1627 1628 pthread_mutex_unlock(&rsxadev->lock); 1629 } 1630 1631 static int adev_dump(const audio_hw_device_t *device, int fd) 1632 { 1633 const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device); 1634 reinterpret_cast<const struct submix_audio_device *>( 1635 reinterpret_cast<const uint8_t *>(device) - 1636 offsetof(struct submix_audio_device, device)); 1637 char msg[100]; 1638 int n = sprintf(msg, "\nReroute submix audio module:\n"); 1639 write(fd, &msg, n); 1640 for (int i=0 ; i < MAX_ROUTES ; i++) { 1641 n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i, 1642 rsxadev->routes[i].config.input_sample_rate, 1643 rsxadev->routes[i].config.output_sample_rate, 1644 rsxadev->routes[i].address); 1645 write(fd, &msg, n); 1646 } 1647 return 0; 1648 } 1649 1650 static int adev_close(hw_device_t *device) 1651 { 1652 ALOGI("adev_close()"); 1653 free(device); 1654 return 0; 1655 } 1656 1657 static int adev_open(const hw_module_t* module, const char* name, 1658 hw_device_t** device) 1659 { 1660 ALOGI("adev_open(name=%s)", name); 1661 struct submix_audio_device *rsxadev; 1662 1663 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 1664 return -EINVAL; 1665 1666 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 1667 if (!rsxadev) 1668 return -ENOMEM; 1669 1670 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 1671 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 1672 rsxadev->device.common.module = (struct hw_module_t *) module; 1673 rsxadev->device.common.close = adev_close; 1674 1675 rsxadev->device.init_check = adev_init_check; 1676 rsxadev->device.set_voice_volume = adev_set_voice_volume; 1677 rsxadev->device.set_master_volume = adev_set_master_volume; 1678 rsxadev->device.get_master_volume = adev_get_master_volume; 1679 rsxadev->device.set_master_mute = adev_set_master_mute; 1680 rsxadev->device.get_master_mute = adev_get_master_mute; 1681 rsxadev->device.set_mode = adev_set_mode; 1682 rsxadev->device.set_mic_mute = adev_set_mic_mute; 1683 rsxadev->device.get_mic_mute = adev_get_mic_mute; 1684 rsxadev->device.set_parameters = adev_set_parameters; 1685 rsxadev->device.get_parameters = adev_get_parameters; 1686 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 1687 rsxadev->device.open_output_stream = adev_open_output_stream; 1688 rsxadev->device.close_output_stream = adev_close_output_stream; 1689 rsxadev->device.open_input_stream = adev_open_input_stream; 1690 rsxadev->device.close_input_stream = adev_close_input_stream; 1691 rsxadev->device.dump = adev_dump; 1692 1693 for (int i=0 ; i < MAX_ROUTES ; i++) { 1694 memset(&rsxadev->routes[i], 0, sizeof(route_config)); 1695 strcpy(rsxadev->routes[i].address, ""); 1696 } 1697 1698 *device = &rsxadev->device.common; 1699 1700 return 0; 1701 } 1702 1703 static struct hw_module_methods_t hal_module_methods = { 1704 /* open */ adev_open, 1705 }; 1706 1707 struct audio_module HAL_MODULE_INFO_SYM = { 1708 /* common */ { 1709 /* tag */ HARDWARE_MODULE_TAG, 1710 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 1711 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 1712 /* id */ AUDIO_HARDWARE_MODULE_ID, 1713 /* name */ "Wifi Display audio HAL", 1714 /* author */ "The Android Open Source Project", 1715 /* methods */ &hal_module_methods, 1716 /* dso */ NULL, 1717 /* reserved */ { 0 }, 1718 }, 1719 }; 1720 1721 } //namespace android 1722 1723 } //extern "C" 1724