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 51 extern "C" { 52 53 namespace 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/media" 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. 125 struct 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 10 145 typedef 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; 168 169 struct 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 }; 176 177 struct 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 write_counter_frames; 183 #if LOG_STREAMS_TO_FILES 184 int log_fd; 185 #endif // LOG_STREAMS_TO_FILES 186 }; 187 188 struct submix_stream_in { 189 struct audio_stream_in stream; 190 struct submix_audio_device *dev; 191 int route_handle; 192 bool input_standby; 193 bool output_standby_rec_thr; // output standby state as seen from record thread 194 // wall clock when recording starts 195 struct timespec record_start_time; 196 // how many frames have been requested to be read 197 uint64_t read_counter_frames; 198 199 #if ENABLE_LEGACY_INPUT_OPEN 200 // Number of references to this input stream. 201 volatile int32_t ref_count; 202 #endif // ENABLE_LEGACY_INPUT_OPEN 203 #if LOG_STREAMS_TO_FILES 204 int log_fd; 205 #endif // LOG_STREAMS_TO_FILES 206 207 volatile int16_t read_error_count; 208 }; 209 210 // Determine whether the specified sample rate is supported by the submix module. 211 static bool sample_rate_supported(const uint32_t sample_rate) 212 { 213 // Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp. 214 static const unsigned int supported_sample_rates[] = { 215 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 216 }; 217 bool return_value; 218 SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value); 219 return return_value; 220 } 221 222 // Determine whether the specified sample rate is supported, if it is return the specified sample 223 // rate, otherwise return the default sample rate for the submix module. 224 static uint32_t get_supported_sample_rate(uint32_t sample_rate) 225 { 226 return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ; 227 } 228 229 // Determine whether the specified channel in mask is supported by the submix module. 230 static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask) 231 { 232 // Set of channel in masks supported by Format_from_SR_C() 233 // frameworks/av/media/libnbaio/NAIO.cpp. 234 static const audio_channel_mask_t supported_channel_in_masks[] = { 235 AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, 236 }; 237 bool return_value; 238 SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value); 239 return return_value; 240 } 241 242 // Determine whether the specified channel in mask is supported, if it is return the specified 243 // channel in mask, otherwise return the default channel in mask for the submix module. 244 static audio_channel_mask_t get_supported_channel_in_mask( 245 const audio_channel_mask_t channel_in_mask) 246 { 247 return channel_in_mask_supported(channel_in_mask) ? channel_in_mask : 248 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO); 249 } 250 251 // Determine whether the specified channel out mask is supported by the submix module. 252 static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask) 253 { 254 // Set of channel out masks supported by Format_from_SR_C() 255 // frameworks/av/media/libnbaio/NAIO.cpp. 256 static const audio_channel_mask_t supported_channel_out_masks[] = { 257 AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO, 258 }; 259 bool return_value; 260 SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value); 261 return return_value; 262 } 263 264 // Determine whether the specified channel out mask is supported, if it is return the specified 265 // channel out mask, otherwise return the default channel out mask for the submix module. 266 static audio_channel_mask_t get_supported_channel_out_mask( 267 const audio_channel_mask_t channel_out_mask) 268 { 269 return channel_out_mask_supported(channel_out_mask) ? channel_out_mask : 270 static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO); 271 } 272 273 // Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the 274 // structure. 275 static struct submix_stream_out * audio_stream_out_get_submix_stream_out( 276 struct audio_stream_out * const stream) 277 { 278 ALOG_ASSERT(stream); 279 return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) - 280 offsetof(struct submix_stream_out, stream)); 281 } 282 283 // Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure. 284 static struct submix_stream_out * audio_stream_get_submix_stream_out( 285 struct audio_stream * const stream) 286 { 287 ALOG_ASSERT(stream); 288 return audio_stream_out_get_submix_stream_out( 289 reinterpret_cast<struct audio_stream_out *>(stream)); 290 } 291 292 // Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the 293 // structure. 294 static struct submix_stream_in * audio_stream_in_get_submix_stream_in( 295 struct audio_stream_in * const stream) 296 { 297 ALOG_ASSERT(stream); 298 return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) - 299 offsetof(struct submix_stream_in, stream)); 300 } 301 302 // Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure. 303 static struct submix_stream_in * audio_stream_get_submix_stream_in( 304 struct audio_stream * const stream) 305 { 306 ALOG_ASSERT(stream); 307 return audio_stream_in_get_submix_stream_in( 308 reinterpret_cast<struct audio_stream_in *>(stream)); 309 } 310 311 // Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within 312 // the structure. 313 static struct submix_audio_device * audio_hw_device_get_submix_audio_device( 314 struct audio_hw_device *device) 315 { 316 ALOG_ASSERT(device); 317 return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) - 318 offsetof(struct submix_audio_device, device)); 319 } 320 321 // Compare an audio_config with input channel mask and an audio_config with output channel mask 322 // returning false if they do *not* match, true otherwise. 323 static bool audio_config_compare(const audio_config * const input_config, 324 const audio_config * const output_config) 325 { 326 #if !ENABLE_CHANNEL_CONVERSION 327 const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask); 328 const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask); 329 if (input_channels != output_channels) { 330 ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d", 331 input_channels, output_channels); 332 return false; 333 } 334 #endif // !ENABLE_CHANNEL_CONVERSION 335 #if ENABLE_RESAMPLING 336 if (input_config->sample_rate != output_config->sample_rate && 337 audio_channel_count_from_in_mask(input_config->channel_mask) != 1) { 338 #else 339 if (input_config->sample_rate != output_config->sample_rate) { 340 #endif // ENABLE_RESAMPLING 341 ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul", 342 input_config->sample_rate, output_config->sample_rate); 343 return false; 344 } 345 if (input_config->format != output_config->format) { 346 ALOGE("audio_config_compare() format mismatch %x vs. %x", 347 input_config->format, output_config->format); 348 return false; 349 } 350 // This purposely ignores offload_info as it's not required for the submix device. 351 return true; 352 } 353 354 // If one doesn't exist, create a pipe for the submix audio device rsxadev of size 355 // buffer_size_frames and optionally associate "in" or "out" with the submix audio device. 356 // Must be called with lock held on the submix_audio_device 357 static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev, 358 const struct audio_config * const config, 359 const size_t buffer_size_frames, 360 const uint32_t buffer_period_count, 361 struct submix_stream_in * const in, 362 struct submix_stream_out * const out, 363 const char *address, 364 int route_idx) 365 { 366 ALOG_ASSERT(in || out); 367 ALOG_ASSERT(route_idx > -1); 368 ALOG_ASSERT(route_idx < MAX_ROUTES); 369 ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx); 370 371 // Save a reference to the specified input or output stream and the associated channel 372 // mask. 373 if (in) { 374 in->route_handle = route_idx; 375 rsxadev->routes[route_idx].input = in; 376 rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask; 377 #if ENABLE_RESAMPLING 378 rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate; 379 // If the output isn't configured yet, set the output sample rate to the maximum supported 380 // sample rate such that the smallest possible input buffer is created, and put a default 381 // value for channel count 382 if (!rsxadev->routes[route_idx].output) { 383 rsxadev->routes[route_idx].config.output_sample_rate = 48000; 384 rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO; 385 } 386 #endif // ENABLE_RESAMPLING 387 } 388 if (out) { 389 out->route_handle = route_idx; 390 rsxadev->routes[route_idx].output = out; 391 rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask; 392 #if ENABLE_RESAMPLING 393 rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate; 394 #endif // ENABLE_RESAMPLING 395 } 396 // Save the address 397 strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN); 398 ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx); 399 // If a pipe isn't associated with the device, create one. 400 if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL) 401 { 402 struct submix_config * const device_config = &rsxadev->routes[route_idx].config; 403 uint32_t channel_count; 404 if (out) 405 channel_count = audio_channel_count_from_out_mask(config->channel_mask); 406 else 407 channel_count = audio_channel_count_from_in_mask(config->channel_mask); 408 #if ENABLE_CHANNEL_CONVERSION 409 // If channel conversion is enabled, allocate enough space for the maximum number of 410 // possible channels stored in the pipe for the situation when the number of channels in 411 // the output stream don't match the number in the input stream. 412 const uint32_t pipe_channel_count = max(channel_count, 2); 413 #else 414 const uint32_t pipe_channel_count = channel_count; 415 #endif // ENABLE_CHANNEL_CONVERSION 416 const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count, 417 config->format); 418 const NBAIO_Format offers[1] = {format}; 419 size_t numCounterOffers = 0; 420 // Create a MonoPipe with optional blocking set to true. 421 MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/); 422 // Negotiation between the source and sink cannot fail as the device open operation 423 // creates both ends of the pipe using the same audio format. 424 ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers); 425 ALOG_ASSERT(index == 0); 426 MonoPipeReader* source = new MonoPipeReader(sink); 427 numCounterOffers = 0; 428 index = source->negotiate(offers, 1, NULL, numCounterOffers); 429 ALOG_ASSERT(index == 0); 430 ALOGV("submix_audio_device_create_pipe_l(): created pipe"); 431 432 // Save references to the source and sink. 433 ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL); 434 ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL); 435 rsxadev->routes[route_idx].rsxSink = sink; 436 rsxadev->routes[route_idx].rsxSource = source; 437 // Store the sanitized audio format in the device so that it's possible to determine 438 // the format of the pipe source when opening the input device. 439 memcpy(&device_config->common, config, sizeof(device_config->common)); 440 device_config->buffer_size_frames = sink->maxFrames(); 441 device_config->buffer_period_size_frames = device_config->buffer_size_frames / 442 buffer_period_count; 443 if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream); 444 if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream); 445 #if ENABLE_CHANNEL_CONVERSION 446 // Calculate the pipe frame size based upon the number of channels. 447 device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) / 448 channel_count; 449 #endif // ENABLE_CHANNEL_CONVERSION 450 SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, " 451 "period size %zd", device_config->pipe_frame_size, 452 device_config->buffer_size_frames, device_config->buffer_period_size_frames); 453 } 454 } 455 456 // Release references to the sink and source. Input and output threads may maintain references 457 // to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use 458 // before they shutdown. 459 // Must be called with lock held on the submix_audio_device 460 static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev, 461 int route_idx) 462 { 463 ALOG_ASSERT(route_idx > -1); 464 ALOG_ASSERT(route_idx < MAX_ROUTES); 465 ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx, 466 rsxadev->routes[route_idx].address); 467 if (rsxadev->routes[route_idx].rsxSink != 0) { 468 rsxadev->routes[route_idx].rsxSink.clear(); 469 rsxadev->routes[route_idx].rsxSink = 0; 470 } 471 if (rsxadev->routes[route_idx].rsxSource != 0) { 472 rsxadev->routes[route_idx].rsxSource.clear(); 473 rsxadev->routes[route_idx].rsxSource = 0; 474 } 475 memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN); 476 #ifdef ENABLE_RESAMPLING 477 memset(rsxadev->routes[route_idx].resampler_buffer, 0, 478 sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES); 479 #endif 480 } 481 482 // Remove references to the specified input and output streams. When the device no longer 483 // references input and output streams destroy the associated pipe. 484 // Must be called with lock held on the submix_audio_device 485 static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev, 486 const struct submix_stream_in * const in, 487 const struct submix_stream_out * const out) 488 { 489 MonoPipe* sink; 490 ALOGV("submix_audio_device_destroy_pipe_l()"); 491 int route_idx = -1; 492 if (in != NULL) { 493 #if ENABLE_LEGACY_INPUT_OPEN 494 const_cast<struct submix_stream_in*>(in)->ref_count--; 495 route_idx = in->route_handle; 496 ALOG_ASSERT(rsxadev->routes[route_idx].input == in); 497 if (in->ref_count == 0) { 498 rsxadev->routes[route_idx].input = NULL; 499 } 500 ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count); 501 #else 502 rsxadev->input = NULL; 503 #endif // ENABLE_LEGACY_INPUT_OPEN 504 } 505 if (out != NULL) { 506 route_idx = out->route_handle; 507 ALOG_ASSERT(rsxadev->routes[route_idx].output == out); 508 rsxadev->routes[route_idx].output = NULL; 509 } 510 if (route_idx != -1 && 511 rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) { 512 submix_audio_device_release_pipe_l(rsxadev, route_idx); 513 ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed"); 514 } 515 } 516 517 // Sanitize the user specified audio config for a submix input / output stream. 518 static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format) 519 { 520 config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) : 521 get_supported_channel_out_mask(config->channel_mask); 522 config->sample_rate = get_supported_sample_rate(config->sample_rate); 523 config->format = DEFAULT_FORMAT; 524 } 525 526 // Verify a submix input or output stream can be opened. 527 // Must be called with lock held on the submix_audio_device 528 static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev, 529 int route_idx, 530 const struct audio_config * const config, 531 const bool opening_input) 532 { 533 bool input_open; 534 bool output_open; 535 audio_config pipe_config; 536 537 // Query the device for the current audio config and whether input and output streams are open. 538 output_open = rsxadev->routes[route_idx].output != NULL; 539 input_open = rsxadev->routes[route_idx].input != NULL; 540 memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config)); 541 542 // If the stream is already open, don't open it again. 543 if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) { 544 ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" : 545 "Output"); 546 return false; 547 } 548 549 SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x " 550 "%s_channel_mask=%x", config->sample_rate, config->format, 551 opening_input ? "in" : "out", config->channel_mask); 552 553 // If either stream is open, verify the existing audio config the pipe matches the user 554 // specified config. 555 if (input_open || output_open) { 556 const audio_config * const input_config = opening_input ? config : &pipe_config; 557 const audio_config * const output_config = opening_input ? &pipe_config : config; 558 // Get the channel mask of the open device. 559 pipe_config.channel_mask = 560 opening_input ? rsxadev->routes[route_idx].config.output_channel_mask : 561 rsxadev->routes[route_idx].config.input_channel_mask; 562 if (!audio_config_compare(input_config, output_config)) { 563 ALOGE("submix_open_validate_l(): Unsupported format."); 564 return false; 565 } 566 } 567 return true; 568 } 569 570 // Must be called with lock held on the submix_audio_device 571 static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev, 572 const char* address, /*in*/ 573 int *idx /*out*/) 574 { 575 // Do we already have a route for this address 576 int route_idx = -1; 577 int route_empty_idx = -1; // index of an empty route slot that can be used if needed 578 for (int i=0 ; i < MAX_ROUTES ; i++) { 579 if (strcmp(rsxadev->routes[i].address, "") == 0) { 580 route_empty_idx = i; 581 } 582 if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { 583 route_idx = i; 584 break; 585 } 586 } 587 588 if ((route_idx == -1) && (route_empty_idx == -1)) { 589 ALOGE("Cannot create new route for address %s, max number of routes reached", address); 590 return -ENOMEM; 591 } 592 if (route_idx == -1) { 593 route_idx = route_empty_idx; 594 } 595 *idx = route_idx; 596 return OK; 597 } 598 599 600 // Calculate the maximum size of the pipe buffer in frames for the specified stream. 601 static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream, 602 const struct submix_config *config, 603 const size_t pipe_frames, 604 const size_t stream_frame_size) 605 { 606 const size_t pipe_frame_size = config->pipe_frame_size; 607 const size_t max_frame_size = max(stream_frame_size, pipe_frame_size); 608 return (pipe_frames * config->pipe_frame_size) / max_frame_size; 609 } 610 611 /* audio HAL functions */ 612 613 static uint32_t out_get_sample_rate(const struct audio_stream *stream) 614 { 615 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 616 const_cast<struct audio_stream *>(stream)); 617 #if ENABLE_RESAMPLING 618 const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate; 619 #else 620 const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate; 621 #endif // ENABLE_RESAMPLING 622 SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s", 623 out_rate, out->dev->routes[out->route_handle].address); 624 return out_rate; 625 } 626 627 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) 628 { 629 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 630 #if ENABLE_RESAMPLING 631 // The sample rate of the stream can't be changed once it's set since this would change the 632 // output buffer size and hence break playback to the shared pipe. 633 if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) { 634 ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from " 635 "%u to %u for addr %s", 636 out->dev->routes[out->route_handle].config.output_sample_rate, rate, 637 out->dev->routes[out->route_handle].address); 638 return -ENOSYS; 639 } 640 #endif // ENABLE_RESAMPLING 641 if (!sample_rate_supported(rate)) { 642 ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate); 643 return -ENOSYS; 644 } 645 SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate); 646 out->dev->routes[out->route_handle].config.common.sample_rate = rate; 647 return 0; 648 } 649 650 static size_t out_get_buffer_size(const struct audio_stream *stream) 651 { 652 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 653 const_cast<struct audio_stream *>(stream)); 654 const struct submix_config * const config = &out->dev->routes[out->route_handle].config; 655 const size_t stream_frame_size = 656 audio_stream_out_frame_size((const struct audio_stream_out *)stream); 657 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 658 stream, config, config->buffer_period_size_frames, stream_frame_size); 659 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 660 SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames", 661 buffer_size_bytes, buffer_size_frames); 662 return buffer_size_bytes; 663 } 664 665 static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) 666 { 667 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 668 const_cast<struct audio_stream *>(stream)); 669 uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask; 670 SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask); 671 return channel_mask; 672 } 673 674 static audio_format_t out_get_format(const struct audio_stream *stream) 675 { 676 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out( 677 const_cast<struct audio_stream *>(stream)); 678 const audio_format_t format = out->dev->routes[out->route_handle].config.common.format; 679 SUBMIX_ALOGV("out_get_format() returns %x", format); 680 return format; 681 } 682 683 static int out_set_format(struct audio_stream *stream, audio_format_t format) 684 { 685 const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 686 if (format != out->dev->routes[out->route_handle].config.common.format) { 687 ALOGE("out_set_format(format=%x) format unsupported", format); 688 return -ENOSYS; 689 } 690 SUBMIX_ALOGV("out_set_format(format=%x)", format); 691 return 0; 692 } 693 694 static int out_standby(struct audio_stream *stream) 695 { 696 ALOGI("out_standby()"); 697 struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream); 698 struct submix_audio_device * const rsxadev = out->dev; 699 700 pthread_mutex_lock(&rsxadev->lock); 701 702 out->output_standby = true; 703 out->write_counter_frames = 0; 704 705 pthread_mutex_unlock(&rsxadev->lock); 706 707 return 0; 708 } 709 710 static int out_dump(const struct audio_stream *stream, int fd) 711 { 712 (void)stream; 713 (void)fd; 714 return 0; 715 } 716 717 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 718 { 719 int exiting = -1; 720 AudioParameter parms = AudioParameter(String8(kvpairs)); 721 SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs); 722 723 // FIXME this is using hard-coded strings but in the future, this functionality will be 724 // converted to use audio HAL extensions required to support tunneling 725 if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) { 726 struct submix_audio_device * const rsxadev = 727 audio_stream_get_submix_stream_out(stream)->dev; 728 pthread_mutex_lock(&rsxadev->lock); 729 { // using the sink 730 sp<MonoPipe> sink = 731 rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle] 732 .rsxSink; 733 if (sink == NULL) { 734 pthread_mutex_unlock(&rsxadev->lock); 735 return 0; 736 } 737 738 ALOGD("out_set_parameters(): shutting down MonoPipe sink"); 739 sink->shutdown(true); 740 } // done using the sink 741 pthread_mutex_unlock(&rsxadev->lock); 742 } 743 return 0; 744 } 745 746 static char * out_get_parameters(const struct audio_stream *stream, const char *keys) 747 { 748 (void)stream; 749 (void)keys; 750 return strdup(""); 751 } 752 753 static uint32_t out_get_latency(const struct audio_stream_out *stream) 754 { 755 const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out( 756 const_cast<struct audio_stream_out *>(stream)); 757 const struct submix_config * const config = &out->dev->routes[out->route_handle].config; 758 const size_t stream_frame_size = 759 audio_stream_out_frame_size(stream); 760 const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 761 &stream->common, config, config->buffer_size_frames, stream_frame_size); 762 const uint32_t sample_rate = out_get_sample_rate(&stream->common); 763 const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate; 764 SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u", 765 latency_ms, buffer_size_frames, sample_rate); 766 return latency_ms; 767 } 768 769 static int out_set_volume(struct audio_stream_out *stream, float left, 770 float right) 771 { 772 (void)stream; 773 (void)left; 774 (void)right; 775 return -ENOSYS; 776 } 777 778 static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, 779 size_t bytes) 780 { 781 SUBMIX_ALOGV("out_write(bytes=%zd)", bytes); 782 ssize_t written_frames = 0; 783 const size_t frame_size = audio_stream_out_frame_size(stream); 784 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 785 struct submix_audio_device * const rsxadev = out->dev; 786 const size_t frames = bytes / frame_size; 787 788 pthread_mutex_lock(&rsxadev->lock); 789 790 out->output_standby = false; 791 792 sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink; 793 if (sink != NULL) { 794 if (sink->isShutdown()) { 795 sink.clear(); 796 pthread_mutex_unlock(&rsxadev->lock); 797 SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write."); 798 // the pipe has already been shutdown, this buffer will be lost but we must 799 // simulate timing so we don't drain the output faster than realtime 800 usleep(frames * 1000000 / out_get_sample_rate(&stream->common)); 801 return bytes; 802 } 803 } else { 804 pthread_mutex_unlock(&rsxadev->lock); 805 ALOGE("out_write without a pipe!"); 806 ALOG_ASSERT("out_write without a pipe!"); 807 return 0; 808 } 809 810 // If the write to the sink would block when no input stream is present, flush enough frames 811 // from the pipe to make space to write the most recent data. 812 { 813 const size_t availableToWrite = sink->availableToWrite(); 814 sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource; 815 if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) { 816 static uint8_t flush_buffer[64]; 817 const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size; 818 size_t frames_to_flush_from_source = frames - availableToWrite; 819 SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking", 820 frames_to_flush_from_source); 821 while (frames_to_flush_from_source) { 822 const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames); 823 frames_to_flush_from_source -= flush_size; 824 // read does not block 825 source->read(flush_buffer, flush_size, AudioBufferProvider::kInvalidPTS); 826 } 827 } 828 } 829 830 pthread_mutex_unlock(&rsxadev->lock); 831 832 written_frames = sink->write(buffer, frames); 833 834 #if LOG_STREAMS_TO_FILES 835 if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size); 836 #endif // LOG_STREAMS_TO_FILES 837 838 if (written_frames < 0) { 839 if (written_frames == (ssize_t)NEGOTIATE) { 840 ALOGE("out_write() write to pipe returned NEGOTIATE"); 841 842 pthread_mutex_lock(&rsxadev->lock); 843 sink.clear(); 844 pthread_mutex_unlock(&rsxadev->lock); 845 846 written_frames = 0; 847 return 0; 848 } else { 849 // write() returned UNDERRUN or WOULD_BLOCK, retry 850 ALOGE("out_write() write to pipe returned unexpected %zd", written_frames); 851 written_frames = sink->write(buffer, frames); 852 } 853 } 854 855 pthread_mutex_lock(&rsxadev->lock); 856 sink.clear(); 857 if (written_frames > 0) { 858 out->write_counter_frames += written_frames; 859 } 860 pthread_mutex_unlock(&rsxadev->lock); 861 862 if (written_frames < 0) { 863 ALOGE("out_write() failed writing to pipe with %zd", written_frames); 864 return 0; 865 } 866 const ssize_t written_bytes = written_frames * frame_size; 867 SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames); 868 return written_bytes; 869 } 870 871 static int out_get_presentation_position(const struct audio_stream_out *stream, 872 uint64_t *frames, struct timespec *timestamp) 873 { 874 const submix_stream_out * out = reinterpret_cast<const struct submix_stream_out *> 875 (reinterpret_cast<const uint8_t *>(stream) - 876 offsetof(struct submix_stream_out, stream)); 877 struct submix_audio_device * const rsxadev = out->dev; 878 int ret = 0; 879 880 pthread_mutex_lock(&rsxadev->lock); 881 882 if (frames) { 883 const ssize_t frames_in_pipe = 884 rsxadev->routes[out->route_handle].rsxSource->availableToRead(); 885 if (CC_UNLIKELY(frames_in_pipe < 0)) { 886 *frames = out->write_counter_frames; 887 } else { 888 *frames = out->write_counter_frames > (uint64_t) frames_in_pipe ? 889 out->write_counter_frames - frames_in_pipe : 0; 890 } 891 } 892 if (timestamp) { 893 clock_gettime(CLOCK_MONOTONIC, timestamp); 894 } 895 896 pthread_mutex_unlock(&rsxadev->lock); 897 898 SUBMIX_ALOGV("out_get_presentation_position() got frames=%llu timestamp sec=%llu", 899 frames ? *frames : -1, timestamp ? timestamp->tv_sec : -1); 900 901 return ret; 902 } 903 904 static int out_get_render_position(const struct audio_stream_out *stream, 905 uint32_t *dsp_frames) 906 { 907 if (!dsp_frames) { 908 return -EINVAL; 909 } 910 uint64_t frames = 0; 911 int ret = out_get_presentation_position(stream, &frames, NULL); 912 if ((ret == 0) && dsp_frames) { 913 *dsp_frames = (uint32_t) frames; 914 } 915 return ret; 916 } 917 918 static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 919 { 920 (void)stream; 921 (void)effect; 922 return 0; 923 } 924 925 static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 926 { 927 (void)stream; 928 (void)effect; 929 return 0; 930 } 931 932 static int out_get_next_write_timestamp(const struct audio_stream_out *stream, 933 int64_t *timestamp) 934 { 935 (void)stream; 936 (void)timestamp; 937 return -EINVAL; 938 } 939 940 /** audio_stream_in implementation **/ 941 static uint32_t in_get_sample_rate(const struct audio_stream *stream) 942 { 943 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 944 const_cast<struct audio_stream*>(stream)); 945 #if ENABLE_RESAMPLING 946 const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate; 947 #else 948 const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate; 949 #endif // ENABLE_RESAMPLING 950 SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate); 951 return rate; 952 } 953 954 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) 955 { 956 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 957 #if ENABLE_RESAMPLING 958 // The sample rate of the stream can't be changed once it's set since this would change the 959 // input buffer size and hence break recording from the shared pipe. 960 if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) { 961 ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from " 962 "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate); 963 return -ENOSYS; 964 } 965 #endif // ENABLE_RESAMPLING 966 if (!sample_rate_supported(rate)) { 967 ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate); 968 return -ENOSYS; 969 } 970 in->dev->routes[in->route_handle].config.common.sample_rate = rate; 971 SUBMIX_ALOGV("in_set_sample_rate() set %u", rate); 972 return 0; 973 } 974 975 static size_t in_get_buffer_size(const struct audio_stream *stream) 976 { 977 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 978 const_cast<struct audio_stream*>(stream)); 979 const struct submix_config * const config = &in->dev->routes[in->route_handle].config; 980 const size_t stream_frame_size = 981 audio_stream_in_frame_size((const struct audio_stream_in *)stream); 982 size_t buffer_size_frames = calculate_stream_pipe_size_in_frames( 983 stream, config, config->buffer_period_size_frames, stream_frame_size); 984 #if ENABLE_RESAMPLING 985 // Scale the size of the buffer based upon the maximum number of frames that could be returned 986 // given the ratio of output to input sample rate. 987 buffer_size_frames = (size_t)(((float)buffer_size_frames * 988 (float)config->input_sample_rate) / 989 (float)config->output_sample_rate); 990 #endif // ENABLE_RESAMPLING 991 const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size; 992 SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes, 993 buffer_size_frames); 994 return buffer_size_bytes; 995 } 996 997 static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) 998 { 999 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 1000 const_cast<struct audio_stream*>(stream)); 1001 const audio_channel_mask_t channel_mask = 1002 in->dev->routes[in->route_handle].config.input_channel_mask; 1003 SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask); 1004 return channel_mask; 1005 } 1006 1007 static audio_format_t in_get_format(const struct audio_stream *stream) 1008 { 1009 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in( 1010 const_cast<struct audio_stream*>(stream)); 1011 const audio_format_t format = in->dev->routes[in->route_handle].config.common.format; 1012 SUBMIX_ALOGV("in_get_format() returns %x", format); 1013 return format; 1014 } 1015 1016 static int in_set_format(struct audio_stream *stream, audio_format_t format) 1017 { 1018 const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 1019 if (format != in->dev->routes[in->route_handle].config.common.format) { 1020 ALOGE("in_set_format(format=%x) format unsupported", format); 1021 return -ENOSYS; 1022 } 1023 SUBMIX_ALOGV("in_set_format(format=%x)", format); 1024 return 0; 1025 } 1026 1027 static int in_standby(struct audio_stream *stream) 1028 { 1029 ALOGI("in_standby()"); 1030 struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream); 1031 struct submix_audio_device * const rsxadev = in->dev; 1032 1033 pthread_mutex_lock(&rsxadev->lock); 1034 1035 in->input_standby = true; 1036 1037 pthread_mutex_unlock(&rsxadev->lock); 1038 1039 return 0; 1040 } 1041 1042 static int in_dump(const struct audio_stream *stream, int fd) 1043 { 1044 (void)stream; 1045 (void)fd; 1046 return 0; 1047 } 1048 1049 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 1050 { 1051 (void)stream; 1052 (void)kvpairs; 1053 return 0; 1054 } 1055 1056 static char * in_get_parameters(const struct audio_stream *stream, 1057 const char *keys) 1058 { 1059 (void)stream; 1060 (void)keys; 1061 return strdup(""); 1062 } 1063 1064 static int in_set_gain(struct audio_stream_in *stream, float gain) 1065 { 1066 (void)stream; 1067 (void)gain; 1068 return 0; 1069 } 1070 1071 static ssize_t in_read(struct audio_stream_in *stream, void* buffer, 1072 size_t bytes) 1073 { 1074 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1075 struct submix_audio_device * const rsxadev = in->dev; 1076 struct audio_config *format; 1077 const size_t frame_size = audio_stream_in_frame_size(stream); 1078 const size_t frames_to_read = bytes / frame_size; 1079 1080 SUBMIX_ALOGV("in_read bytes=%zu", bytes); 1081 pthread_mutex_lock(&rsxadev->lock); 1082 1083 const bool output_standby = rsxadev->routes[in->route_handle].output == NULL 1084 ? true : rsxadev->routes[in->route_handle].output->output_standby; 1085 const bool output_standby_transition = (in->output_standby_rec_thr != output_standby); 1086 in->output_standby_rec_thr = output_standby; 1087 1088 if (in->input_standby || output_standby_transition) { 1089 in->input_standby = false; 1090 // keep track of when we exit input standby (== first read == start "real recording") 1091 // or when we start recording silence, and reset projected time 1092 int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time); 1093 if (rc == 0) { 1094 in->read_counter_frames = 0; 1095 } 1096 } 1097 1098 in->read_counter_frames += frames_to_read; 1099 size_t remaining_frames = frames_to_read; 1100 1101 { 1102 // about to read from audio source 1103 sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource; 1104 if (source == NULL) { 1105 in->read_error_count++;// ok if it rolls over 1106 ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS, 1107 "no audio pipe yet we're trying to read! (not all errors will be logged)"); 1108 pthread_mutex_unlock(&rsxadev->lock); 1109 usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common)); 1110 memset(buffer, 0, bytes); 1111 return bytes; 1112 } 1113 1114 pthread_mutex_unlock(&rsxadev->lock); 1115 1116 // read the data from the pipe (it's non blocking) 1117 int attempts = 0; 1118 char* buff = (char*)buffer; 1119 #if ENABLE_CHANNEL_CONVERSION 1120 // Determine whether channel conversion is required. 1121 const uint32_t input_channels = audio_channel_count_from_in_mask( 1122 rsxadev->routes[in->route_handle].config.input_channel_mask); 1123 const uint32_t output_channels = audio_channel_count_from_out_mask( 1124 rsxadev->routes[in->route_handle].config.output_channel_mask); 1125 if (input_channels != output_channels) { 1126 SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d " 1127 "input channels", output_channels, input_channels); 1128 // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono. 1129 ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == 1130 AUDIO_FORMAT_PCM_16_BIT); 1131 ALOG_ASSERT((input_channels == 1 && output_channels == 2) || 1132 (input_channels == 2 && output_channels == 1)); 1133 } 1134 #endif // ENABLE_CHANNEL_CONVERSION 1135 1136 #if ENABLE_RESAMPLING 1137 const uint32_t input_sample_rate = in_get_sample_rate(&stream->common); 1138 const uint32_t output_sample_rate = 1139 rsxadev->routes[in->route_handle].config.output_sample_rate; 1140 const size_t resampler_buffer_size_frames = 1141 sizeof(rsxadev->routes[in->route_handle].resampler_buffer) / 1142 sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]); 1143 float resampler_ratio = 1.0f; 1144 // Determine whether resampling is required. 1145 if (input_sample_rate != output_sample_rate) { 1146 resampler_ratio = (float)output_sample_rate / (float)input_sample_rate; 1147 // Only support 16-bit PCM mono resampling. 1148 // NOTE: Resampling is performed after the channel conversion step. 1149 ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format == 1150 AUDIO_FORMAT_PCM_16_BIT); 1151 ALOG_ASSERT(audio_channel_count_from_in_mask( 1152 rsxadev->routes[in->route_handle].config.input_channel_mask) == 1); 1153 } 1154 #endif // ENABLE_RESAMPLING 1155 1156 while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) { 1157 ssize_t frames_read = -1977; 1158 size_t read_frames = remaining_frames; 1159 #if ENABLE_RESAMPLING 1160 char* const saved_buff = buff; 1161 if (resampler_ratio != 1.0f) { 1162 // Calculate the number of frames from the pipe that need to be read to generate 1163 // the data for the input stream read. 1164 const size_t frames_required_for_resampler = (size_t)( 1165 (float)read_frames * (float)resampler_ratio); 1166 read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames); 1167 // Read into the resampler buffer. 1168 buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer; 1169 } 1170 #endif // ENABLE_RESAMPLING 1171 #if ENABLE_CHANNEL_CONVERSION 1172 if (output_channels == 1 && input_channels == 2) { 1173 // Need to read half the requested frames since the converted output 1174 // data will take twice the space (mono->stereo). 1175 read_frames /= 2; 1176 } 1177 #endif // ENABLE_CHANNEL_CONVERSION 1178 1179 SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead()); 1180 1181 frames_read = source->read(buff, read_frames, AudioBufferProvider::kInvalidPTS); 1182 1183 SUBMIX_ALOGV("in_read(): frames read %zd", frames_read); 1184 1185 #if ENABLE_CHANNEL_CONVERSION 1186 // Perform in-place channel conversion. 1187 // NOTE: In the following "input stream" refers to the data returned by this function 1188 // and "output stream" refers to the data read from the pipe. 1189 if (input_channels != output_channels && frames_read > 0) { 1190 int16_t *data = (int16_t*)buff; 1191 if (output_channels == 2 && input_channels == 1) { 1192 // Offset into the output stream data in samples. 1193 ssize_t output_stream_offset = 0; 1194 for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read; 1195 input_stream_frame++, output_stream_offset += 2) { 1196 // Average the content from both channels. 1197 data[input_stream_frame] = ((int32_t)data[output_stream_offset] + 1198 (int32_t)data[output_stream_offset + 1]) / 2; 1199 } 1200 } else if (output_channels == 1 && input_channels == 2) { 1201 // Offset into the input stream data in samples. 1202 ssize_t input_stream_offset = (frames_read - 1) * 2; 1203 for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0; 1204 output_stream_frame--, input_stream_offset -= 2) { 1205 const short sample = data[output_stream_frame]; 1206 data[input_stream_offset] = sample; 1207 data[input_stream_offset + 1] = sample; 1208 } 1209 } 1210 } 1211 #endif // ENABLE_CHANNEL_CONVERSION 1212 1213 #if ENABLE_RESAMPLING 1214 if (resampler_ratio != 1.0f) { 1215 SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read); 1216 const int16_t * const data = (int16_t*)buff; 1217 int16_t * const resampled_buffer = (int16_t*)saved_buff; 1218 // Resample with *no* filtering - if the data from the ouptut stream was really 1219 // sampled at a different rate this will result in very nasty aliasing. 1220 const float output_stream_frames = (float)frames_read; 1221 size_t input_stream_frame = 0; 1222 for (float output_stream_frame = 0.0f; 1223 output_stream_frame < output_stream_frames && 1224 input_stream_frame < remaining_frames; 1225 output_stream_frame += resampler_ratio, input_stream_frame++) { 1226 resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame]; 1227 } 1228 ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames); 1229 SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame); 1230 frames_read = input_stream_frame; 1231 buff = saved_buff; 1232 } 1233 #endif // ENABLE_RESAMPLING 1234 1235 if (frames_read > 0) { 1236 #if LOG_STREAMS_TO_FILES 1237 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size); 1238 #endif // LOG_STREAMS_TO_FILES 1239 1240 remaining_frames -= frames_read; 1241 buff += frames_read * frame_size; 1242 SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu", 1243 attempts, frames_read, remaining_frames); 1244 } else { 1245 attempts++; 1246 SUBMIX_ALOGE(" in_read read returned %zd", frames_read); 1247 usleep(READ_ATTEMPT_SLEEP_MS * 1000); 1248 } 1249 } 1250 // done using the source 1251 pthread_mutex_lock(&rsxadev->lock); 1252 source.clear(); 1253 pthread_mutex_unlock(&rsxadev->lock); 1254 } 1255 1256 if (remaining_frames > 0) { 1257 const size_t remaining_bytes = remaining_frames * frame_size; 1258 SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames); 1259 memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes); 1260 } 1261 1262 // compute how much we need to sleep after reading the data by comparing the wall clock with 1263 // the projected time at which we should return. 1264 struct timespec time_after_read;// wall clock after reading from the pipe 1265 struct timespec record_duration;// observed record duration 1266 int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read); 1267 const uint32_t sample_rate = in_get_sample_rate(&stream->common); 1268 if (rc == 0) { 1269 // for how long have we been recording? 1270 record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec; 1271 record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec; 1272 if (record_duration.tv_nsec < 0) { 1273 record_duration.tv_sec--; 1274 record_duration.tv_nsec += 1000000000; 1275 } 1276 1277 // read_counter_frames contains the number of frames that have been read since the 1278 // beginning of recording (including this call): it's converted to usec and compared to 1279 // how long we've been recording for, which gives us how long we must wait to sync the 1280 // projected recording time, and the observed recording time. 1281 long projected_vs_observed_offset_us = 1282 ((int64_t)(in->read_counter_frames 1283 - (record_duration.tv_sec*sample_rate))) 1284 * 1000000 / sample_rate 1285 - (record_duration.tv_nsec / 1000); 1286 1287 SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus", 1288 record_duration.tv_sec, record_duration.tv_nsec/1000000, 1289 projected_vs_observed_offset_us); 1290 if (projected_vs_observed_offset_us > 0) { 1291 usleep(projected_vs_observed_offset_us); 1292 } 1293 } 1294 1295 SUBMIX_ALOGV("in_read returns %zu", bytes); 1296 return bytes; 1297 1298 } 1299 1300 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) 1301 { 1302 (void)stream; 1303 return 0; 1304 } 1305 1306 static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1307 { 1308 (void)stream; 1309 (void)effect; 1310 return 0; 1311 } 1312 1313 static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) 1314 { 1315 (void)stream; 1316 (void)effect; 1317 return 0; 1318 } 1319 1320 static int adev_open_output_stream(struct audio_hw_device *dev, 1321 audio_io_handle_t handle, 1322 audio_devices_t devices, 1323 audio_output_flags_t flags, 1324 struct audio_config *config, 1325 struct audio_stream_out **stream_out, 1326 const char *address) 1327 { 1328 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1329 ALOGD("adev_open_output_stream(address=%s)", address); 1330 struct submix_stream_out *out; 1331 bool force_pipe_creation = false; 1332 (void)handle; 1333 (void)devices; 1334 (void)flags; 1335 1336 *stream_out = NULL; 1337 1338 // Make sure it's possible to open the device given the current audio config. 1339 submix_sanitize_config(config, false); 1340 1341 int route_idx = -1; 1342 1343 pthread_mutex_lock(&rsxadev->lock); 1344 1345 status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); 1346 if (res != OK) { 1347 ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); 1348 pthread_mutex_unlock(&rsxadev->lock); 1349 return res; 1350 } 1351 1352 if (!submix_open_validate_l(rsxadev, route_idx, config, false)) { 1353 ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address); 1354 pthread_mutex_unlock(&rsxadev->lock); 1355 return -EINVAL; 1356 } 1357 1358 out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out)); 1359 if (!out) { 1360 pthread_mutex_unlock(&rsxadev->lock); 1361 return -ENOMEM; 1362 } 1363 1364 // Initialize the function pointer tables (v-tables). 1365 out->stream.common.get_sample_rate = out_get_sample_rate; 1366 out->stream.common.set_sample_rate = out_set_sample_rate; 1367 out->stream.common.get_buffer_size = out_get_buffer_size; 1368 out->stream.common.get_channels = out_get_channels; 1369 out->stream.common.get_format = out_get_format; 1370 out->stream.common.set_format = out_set_format; 1371 out->stream.common.standby = out_standby; 1372 out->stream.common.dump = out_dump; 1373 out->stream.common.set_parameters = out_set_parameters; 1374 out->stream.common.get_parameters = out_get_parameters; 1375 out->stream.common.add_audio_effect = out_add_audio_effect; 1376 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1377 out->stream.get_latency = out_get_latency; 1378 out->stream.set_volume = out_set_volume; 1379 out->stream.write = out_write; 1380 out->stream.get_render_position = out_get_render_position; 1381 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1382 out->stream.get_presentation_position = out_get_presentation_position; 1383 1384 out->write_counter_frames = 0; 1385 1386 #if ENABLE_RESAMPLING 1387 // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits 1388 // writes correctly. 1389 force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate 1390 != config->sample_rate; 1391 #endif // ENABLE_RESAMPLING 1392 1393 // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so 1394 // that it's recreated. 1395 if ((rsxadev->routes[route_idx].rsxSink != NULL 1396 && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) { 1397 submix_audio_device_release_pipe_l(rsxadev, route_idx); 1398 } 1399 1400 // Store a pointer to the device from the output stream. 1401 out->dev = rsxadev; 1402 // Initialize the pipe. 1403 ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx); 1404 submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1405 DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx); 1406 #if LOG_STREAMS_TO_FILES 1407 out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1408 LOG_STREAM_FILE_PERMISSIONS); 1409 ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s", 1410 strerror(errno)); 1411 ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd); 1412 #endif // LOG_STREAMS_TO_FILES 1413 // Return the output stream. 1414 *stream_out = &out->stream; 1415 1416 pthread_mutex_unlock(&rsxadev->lock); 1417 return 0; 1418 } 1419 1420 static void adev_close_output_stream(struct audio_hw_device *dev, 1421 struct audio_stream_out *stream) 1422 { 1423 struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( 1424 const_cast<struct audio_hw_device*>(dev)); 1425 struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream); 1426 1427 pthread_mutex_lock(&rsxadev->lock); 1428 ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address); 1429 submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out); 1430 #if LOG_STREAMS_TO_FILES 1431 if (out->log_fd >= 0) close(out->log_fd); 1432 #endif // LOG_STREAMS_TO_FILES 1433 1434 pthread_mutex_unlock(&rsxadev->lock); 1435 free(out); 1436 } 1437 1438 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1439 { 1440 (void)dev; 1441 (void)kvpairs; 1442 return -ENOSYS; 1443 } 1444 1445 static char * adev_get_parameters(const struct audio_hw_device *dev, 1446 const char *keys) 1447 { 1448 (void)dev; 1449 (void)keys; 1450 return strdup("");; 1451 } 1452 1453 static int adev_init_check(const struct audio_hw_device *dev) 1454 { 1455 ALOGI("adev_init_check()"); 1456 (void)dev; 1457 return 0; 1458 } 1459 1460 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 1461 { 1462 (void)dev; 1463 (void)volume; 1464 return -ENOSYS; 1465 } 1466 1467 static int adev_set_master_volume(struct audio_hw_device *dev, float volume) 1468 { 1469 (void)dev; 1470 (void)volume; 1471 return -ENOSYS; 1472 } 1473 1474 static int adev_get_master_volume(struct audio_hw_device *dev, float *volume) 1475 { 1476 (void)dev; 1477 (void)volume; 1478 return -ENOSYS; 1479 } 1480 1481 static int adev_set_master_mute(struct audio_hw_device *dev, bool muted) 1482 { 1483 (void)dev; 1484 (void)muted; 1485 return -ENOSYS; 1486 } 1487 1488 static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) 1489 { 1490 (void)dev; 1491 (void)muted; 1492 return -ENOSYS; 1493 } 1494 1495 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 1496 { 1497 (void)dev; 1498 (void)mode; 1499 return 0; 1500 } 1501 1502 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 1503 { 1504 (void)dev; 1505 (void)state; 1506 return -ENOSYS; 1507 } 1508 1509 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 1510 { 1511 (void)dev; 1512 (void)state; 1513 return -ENOSYS; 1514 } 1515 1516 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, 1517 const struct audio_config *config) 1518 { 1519 if (audio_is_linear_pcm(config->format)) { 1520 size_t max_buffer_period_size_frames = 0; 1521 struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device( 1522 const_cast<struct audio_hw_device*>(dev)); 1523 // look for the largest buffer period size 1524 for (int i = 0 ; i < MAX_ROUTES ; i++) { 1525 if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames) 1526 { 1527 max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames; 1528 } 1529 } 1530 const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) * 1531 audio_bytes_per_sample(config->format); 1532 const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes; 1533 SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames", 1534 buffer_size, buffer_period_size_frames); 1535 return buffer_size; 1536 } 1537 return 0; 1538 } 1539 1540 static int adev_open_input_stream(struct audio_hw_device *dev, 1541 audio_io_handle_t handle, 1542 audio_devices_t devices, 1543 struct audio_config *config, 1544 struct audio_stream_in **stream_in, 1545 audio_input_flags_t flags __unused, 1546 const char *address, 1547 audio_source_t source __unused) 1548 { 1549 struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev); 1550 struct submix_stream_in *in; 1551 ALOGD("adev_open_input_stream(addr=%s)", address); 1552 (void)handle; 1553 (void)devices; 1554 1555 *stream_in = NULL; 1556 1557 // Do we already have a route for this address 1558 int route_idx = -1; 1559 1560 pthread_mutex_lock(&rsxadev->lock); 1561 1562 status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx); 1563 if (res != OK) { 1564 ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address); 1565 pthread_mutex_unlock(&rsxadev->lock); 1566 return res; 1567 } 1568 1569 // Make sure it's possible to open the device given the current audio config. 1570 submix_sanitize_config(config, true); 1571 if (!submix_open_validate_l(rsxadev, route_idx, config, true)) { 1572 ALOGE("adev_open_input_stream(): Unable to open input stream."); 1573 pthread_mutex_unlock(&rsxadev->lock); 1574 return -EINVAL; 1575 } 1576 1577 #if ENABLE_LEGACY_INPUT_OPEN 1578 in = rsxadev->routes[route_idx].input; 1579 if (in) { 1580 in->ref_count++; 1581 sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink; 1582 ALOG_ASSERT(sink != NULL); 1583 // If the sink has been shutdown, delete the pipe. 1584 if (sink != NULL) { 1585 if (sink->isShutdown()) { 1586 ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d", 1587 in->ref_count); 1588 submix_audio_device_release_pipe_l(rsxadev, in->route_handle); 1589 } else { 1590 ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count); 1591 } 1592 } else { 1593 ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count); 1594 } 1595 } 1596 #else 1597 in = NULL; 1598 #endif // ENABLE_LEGACY_INPUT_OPEN 1599 1600 if (!in) { 1601 in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in)); 1602 if (!in) return -ENOMEM; 1603 in->ref_count = 1; 1604 1605 // Initialize the function pointer tables (v-tables). 1606 in->stream.common.get_sample_rate = in_get_sample_rate; 1607 in->stream.common.set_sample_rate = in_set_sample_rate; 1608 in->stream.common.get_buffer_size = in_get_buffer_size; 1609 in->stream.common.get_channels = in_get_channels; 1610 in->stream.common.get_format = in_get_format; 1611 in->stream.common.set_format = in_set_format; 1612 in->stream.common.standby = in_standby; 1613 in->stream.common.dump = in_dump; 1614 in->stream.common.set_parameters = in_set_parameters; 1615 in->stream.common.get_parameters = in_get_parameters; 1616 in->stream.common.add_audio_effect = in_add_audio_effect; 1617 in->stream.common.remove_audio_effect = in_remove_audio_effect; 1618 in->stream.set_gain = in_set_gain; 1619 in->stream.read = in_read; 1620 in->stream.get_input_frames_lost = in_get_input_frames_lost; 1621 1622 in->dev = rsxadev; 1623 #if LOG_STREAMS_TO_FILES 1624 in->log_fd = -1; 1625 #endif 1626 } 1627 1628 // Initialize the input stream. 1629 in->read_counter_frames = 0; 1630 in->input_standby = true; 1631 if (rsxadev->routes[route_idx].output != NULL) { 1632 in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby; 1633 } else { 1634 in->output_standby_rec_thr = true; 1635 } 1636 1637 in->read_error_count = 0; 1638 // Initialize the pipe. 1639 ALOGV("adev_open_input_stream(): about to create pipe"); 1640 submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES, 1641 DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx); 1642 #if LOG_STREAMS_TO_FILES 1643 if (in->log_fd >= 0) close(in->log_fd); 1644 in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY, 1645 LOG_STREAM_FILE_PERMISSIONS); 1646 ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s", 1647 strerror(errno)); 1648 ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd); 1649 #endif // LOG_STREAMS_TO_FILES 1650 // Return the input stream. 1651 *stream_in = &in->stream; 1652 1653 pthread_mutex_unlock(&rsxadev->lock); 1654 return 0; 1655 } 1656 1657 static void adev_close_input_stream(struct audio_hw_device *dev, 1658 struct audio_stream_in *stream) 1659 { 1660 struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev); 1661 1662 struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream); 1663 ALOGD("adev_close_input_stream()"); 1664 pthread_mutex_lock(&rsxadev->lock); 1665 submix_audio_device_destroy_pipe_l(rsxadev, in, NULL); 1666 #if LOG_STREAMS_TO_FILES 1667 if (in->log_fd >= 0) close(in->log_fd); 1668 #endif // LOG_STREAMS_TO_FILES 1669 #if ENABLE_LEGACY_INPUT_OPEN 1670 if (in->ref_count == 0) free(in); 1671 #else 1672 free(in); 1673 #endif // ENABLE_LEGACY_INPUT_OPEN 1674 1675 pthread_mutex_unlock(&rsxadev->lock); 1676 } 1677 1678 static int adev_dump(const audio_hw_device_t *device, int fd) 1679 { 1680 const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device); 1681 reinterpret_cast<const struct submix_audio_device *>( 1682 reinterpret_cast<const uint8_t *>(device) - 1683 offsetof(struct submix_audio_device, device)); 1684 char msg[100]; 1685 int n = sprintf(msg, "\nReroute submix audio module:\n"); 1686 write(fd, &msg, n); 1687 for (int i=0 ; i < MAX_ROUTES ; i++) { 1688 n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i, 1689 rsxadev->routes[i].config.input_sample_rate, 1690 rsxadev->routes[i].config.output_sample_rate, 1691 rsxadev->routes[i].address); 1692 write(fd, &msg, n); 1693 } 1694 return 0; 1695 } 1696 1697 static int adev_close(hw_device_t *device) 1698 { 1699 ALOGI("adev_close()"); 1700 free(device); 1701 return 0; 1702 } 1703 1704 static int adev_open(const hw_module_t* module, const char* name, 1705 hw_device_t** device) 1706 { 1707 ALOGI("adev_open(name=%s)", name); 1708 struct submix_audio_device *rsxadev; 1709 1710 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) 1711 return -EINVAL; 1712 1713 rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device)); 1714 if (!rsxadev) 1715 return -ENOMEM; 1716 1717 rsxadev->device.common.tag = HARDWARE_DEVICE_TAG; 1718 rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 1719 rsxadev->device.common.module = (struct hw_module_t *) module; 1720 rsxadev->device.common.close = adev_close; 1721 1722 rsxadev->device.init_check = adev_init_check; 1723 rsxadev->device.set_voice_volume = adev_set_voice_volume; 1724 rsxadev->device.set_master_volume = adev_set_master_volume; 1725 rsxadev->device.get_master_volume = adev_get_master_volume; 1726 rsxadev->device.set_master_mute = adev_set_master_mute; 1727 rsxadev->device.get_master_mute = adev_get_master_mute; 1728 rsxadev->device.set_mode = adev_set_mode; 1729 rsxadev->device.set_mic_mute = adev_set_mic_mute; 1730 rsxadev->device.get_mic_mute = adev_get_mic_mute; 1731 rsxadev->device.set_parameters = adev_set_parameters; 1732 rsxadev->device.get_parameters = adev_get_parameters; 1733 rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size; 1734 rsxadev->device.open_output_stream = adev_open_output_stream; 1735 rsxadev->device.close_output_stream = adev_close_output_stream; 1736 rsxadev->device.open_input_stream = adev_open_input_stream; 1737 rsxadev->device.close_input_stream = adev_close_input_stream; 1738 rsxadev->device.dump = adev_dump; 1739 1740 for (int i=0 ; i < MAX_ROUTES ; i++) { 1741 memset(&rsxadev->routes[i], 0, sizeof(route_config)); 1742 strcpy(rsxadev->routes[i].address, ""); 1743 } 1744 1745 *device = &rsxadev->device.common; 1746 1747 return 0; 1748 } 1749 1750 static struct hw_module_methods_t hal_module_methods = { 1751 /* open */ adev_open, 1752 }; 1753 1754 struct audio_module HAL_MODULE_INFO_SYM = { 1755 /* common */ { 1756 /* tag */ HARDWARE_MODULE_TAG, 1757 /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1, 1758 /* hal_api_version */ HARDWARE_HAL_API_VERSION, 1759 /* id */ AUDIO_HARDWARE_MODULE_ID, 1760 /* name */ "Wifi Display audio HAL", 1761 /* author */ "The Android Open Source Project", 1762 /* methods */ &hal_module_methods, 1763 /* dso */ NULL, 1764 /* reserved */ { 0 }, 1765 }, 1766 }; 1767 1768 } //namespace android 1769 1770 } //extern "C" 1771