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