AudioFlinger.cpp revision a3b09254d44cd8d66ec947abe547538c4cfeaa89
1a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang/* 2a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** 3a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** Copyright 2007, The Android Open Source Project 4a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** 5a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** Licensed under the Apache License, Version 2.0 (the "License"); 6a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** you may not use this file except in compliance with the License. 7a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** You may obtain a copy of the License at 8a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** 9a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** http://www.apache.org/licenses/LICENSE-2.0 10a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** 11a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** Unless required by applicable law or agreed to in writing, software 12a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** distributed under the License is distributed on an "AS IS" BASIS, 13a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** See the License for the specific language governing permissions and 15a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang** limitations under the License. 16a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang*/ 17a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 18a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 19a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#define LOG_TAG "AudioFlinger" 20a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang//#define LOG_NDEBUG 0 21a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 22a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <math.h> 23a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <signal.h> 24a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <sys/time.h> 25a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <sys/resource.h> 26a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 27a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <binder/IPCThreadState.h> 28a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <binder/IServiceManager.h> 29a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <utils/Log.h> 30a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <binder/Parcel.h> 31a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <binder/IPCThreadState.h> 32a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <utils/String16.h> 33a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <utils/threads.h> 34a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <utils/Atomic.h> 35a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 36a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <cutils/bitops.h> 37a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <cutils/properties.h> 38a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <cutils/compiler.h> 39a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 40a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <media/IMediaPlayerService.h> 41a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <media/IMediaDeathNotifier.h> 42a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 43a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <private/media/AudioTrackShared.h> 44a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <private/media/AudioEffectShared.h> 45a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 46a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <system/audio.h> 47a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <hardware/audio.h> 48a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 49a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include "AudioMixer.h" 50a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include "AudioFlinger.h" 51a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include "ServiceUtilities.h" 52a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 53a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <media/EffectsFactoryApi.h> 54a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <audio_effects/effect_visualizer.h> 55a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <audio_effects/effect_ns.h> 56a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <audio_effects/effect_aec.h> 57a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 58a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <audio_utils/primitives.h> 59a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 60a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <cpustats/ThreadCpuUsage.h> 61a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <powermanager/PowerManager.h> 62a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 64a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <common_time/cc_helper.h> 65a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang#include <common_time/local_clock.h> 66a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 67a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// ---------------------------------------------------------------------------- 68a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 69a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 70a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangnamespace android { 71a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 72a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 75a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const float MAX_GAIN = 4096.0f; 76a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const uint32_t MAX_GAIN_INT = 0x1000; 77a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 78a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// retry counts for buffer fill timeout 79a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// 50 * ~20msecs = 1 second 80a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const int8_t kMaxTrackRetries = 50; 81a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const int8_t kMaxTrackStartupRetries = 50; 82a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// allow less retry attempts on direct output thread. 83a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// direct outputs can be a scarce resource in audio hardware and should 84a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// be released as quickly as possible. 85a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const int8_t kMaxTrackRetriesDirect = 2; 86a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 87a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const int kDumpLockRetries = 50; 88a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const int kDumpLockSleepUs = 20000; 89a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 90a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang// don't warn about blocked writes or record buffer overflows more often than this 91a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuangstatic const nsecs_t kWarningThrottleNs = seconds(5); 92a760e55783ee30b0f3494e113a1c37003c9d1770Tsu Chiang Chuang 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 status_t result; 827 828 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 829 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 830 // check calling permissions 831 if (!settingsAllowed()) { 832 return PERMISSION_DENIED; 833 } 834 835 // ioHandle == 0 means the parameters are global to the audio hardware interface 836 if (ioHandle == 0) { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_SET_PARAMETER; 839 status_t final_result = NO_ERROR; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 audio_hw_device_t *dev = mAudioHwDevs[i]; 909 char *s = dev->get_parameters(dev, keys.string()); 910 out_s8 += String8(s ? s : ""); 911 free(s); 912 } 913 return out_s8; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 919 if (playbackThread != NULL) { 920 return playbackThread->getParameters(keys); 921 } 922 RecordThread *recordThread = checkRecordThread_l(ioHandle); 923 if (recordThread != NULL) { 924 return recordThread->getParameters(keys); 925 } 926 return String8(""); 927} 928 929size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 930{ 931 status_t ret = initCheck(); 932 if (ret != NO_ERROR) { 933 return 0; 934 } 935 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 938 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 return size; 941} 942 943unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 944{ 945 if (ioHandle == 0) { 946 return 0; 947 } 948 949 Mutex::Autolock _l(mLock); 950 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getInputFramesLost(); 954 } 955 return 0; 956} 957 958status_t AudioFlinger::setVoiceVolume(float value) 959{ 960 status_t ret = initCheck(); 961 if (ret != NO_ERROR) { 962 return ret; 963 } 964 965 // check calling permissions 966 if (!settingsAllowed()) { 967 return PERMISSION_DENIED; 968 } 969 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 972 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 975 return ret; 976} 977 978status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 979 audio_io_handle_t output) const 980{ 981 status_t status; 982 983 Mutex::Autolock _l(mLock); 984 985 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 986 if (playbackThread != NULL) { 987 return playbackThread->getRenderPosition(halFrames, dspFrames); 988 } 989 990 return BAD_VALUE; 991} 992 993void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 994{ 995 996 Mutex::Autolock _l(mLock); 997 998 pid_t pid = IPCThreadState::self()->getCallingPid(); 999 if (mNotificationClients.indexOfKey(pid) < 0) { 1000 sp<NotificationClient> notificationClient = new NotificationClient(this, 1001 client, 1002 pid); 1003 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1004 1005 mNotificationClients.add(pid, notificationClient); 1006 1007 sp<IBinder> binder = client->asBinder(); 1008 binder->linkToDeath(notificationClient); 1009 1010 // the config change is always sent from playback or record threads to avoid deadlock 1011 // with AudioSystem::gLock 1012 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1013 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1014 } 1015 1016 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1017 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1018 } 1019 } 1020} 1021 1022void AudioFlinger::removeNotificationClient(pid_t pid) 1023{ 1024 Mutex::Autolock _l(mLock); 1025 1026 mNotificationClients.removeItem(pid); 1027 1028 ALOGV("%d died, releasing its sessions", pid); 1029 size_t num = mAudioSessionRefs.size(); 1030 bool removed = false; 1031 for (size_t i = 0; i< num; ) { 1032 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1033 ALOGV(" pid %d @ %d", ref->pid, i); 1034 if (ref->pid == pid) { 1035 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1036 mAudioSessionRefs.removeAt(i); 1037 delete ref; 1038 removed = true; 1039 num--; 1040 } else { 1041 i++; 1042 } 1043 } 1044 if (removed) { 1045 purgeStaleEffects_l(); 1046 } 1047} 1048 1049// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1050void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1051{ 1052 size_t size = mNotificationClients.size(); 1053 for (size_t i = 0; i < size; i++) { 1054 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1055 param2); 1056 } 1057} 1058 1059// removeClient_l() must be called with AudioFlinger::mLock held 1060void AudioFlinger::removeClient_l(pid_t pid) 1061{ 1062 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1063 mClients.removeItem(pid); 1064} 1065 1066 1067// ---------------------------------------------------------------------------- 1068 1069AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1070 uint32_t device, type_t type) 1071 : Thread(false), 1072 mType(type), 1073 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1074 // mChannelMask 1075 mChannelCount(0), 1076 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1077 mParamStatus(NO_ERROR), 1078 mStandby(false), mId(id), 1079 mDevice(device), 1080 mDeathRecipient(new PMDeathRecipient(this)) 1081{ 1082} 1083 1084AudioFlinger::ThreadBase::~ThreadBase() 1085{ 1086 mParamCond.broadcast(); 1087 // do not lock the mutex in destructor 1088 releaseWakeLock_l(); 1089 if (mPowerManager != 0) { 1090 sp<IBinder> binder = mPowerManager->asBinder(); 1091 binder->unlinkToDeath(mDeathRecipient); 1092 } 1093} 1094 1095void AudioFlinger::ThreadBase::exit() 1096{ 1097 ALOGV("ThreadBase::exit"); 1098 { 1099 // This lock prevents the following race in thread (uniprocessor for illustration): 1100 // if (!exitPending()) { 1101 // // context switch from here to exit() 1102 // // exit() calls requestExit(), what exitPending() observes 1103 // // exit() calls signal(), which is dropped since no waiters 1104 // // context switch back from exit() to here 1105 // mWaitWorkCV.wait(...); 1106 // // now thread is hung 1107 // } 1108 AutoMutex lock(mLock); 1109 requestExit(); 1110 mWaitWorkCV.signal(); 1111 } 1112 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1113 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1114 requestExitAndWait(); 1115} 1116 1117status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1118{ 1119 status_t status; 1120 1121 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1122 Mutex::Autolock _l(mLock); 1123 1124 mNewParameters.add(keyValuePairs); 1125 mWaitWorkCV.signal(); 1126 // wait condition with timeout in case the thread loop has exited 1127 // before the request could be processed 1128 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1129 status = mParamStatus; 1130 mWaitWorkCV.signal(); 1131 } else { 1132 status = TIMED_OUT; 1133 } 1134 return status; 1135} 1136 1137void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1138{ 1139 Mutex::Autolock _l(mLock); 1140 sendConfigEvent_l(event, param); 1141} 1142 1143// sendConfigEvent_l() must be called with ThreadBase::mLock held 1144void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1145{ 1146 ConfigEvent configEvent; 1147 configEvent.mEvent = event; 1148 configEvent.mParam = param; 1149 mConfigEvents.add(configEvent); 1150 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1151 mWaitWorkCV.signal(); 1152} 1153 1154void AudioFlinger::ThreadBase::processConfigEvents() 1155{ 1156 mLock.lock(); 1157 while(!mConfigEvents.isEmpty()) { 1158 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1159 ConfigEvent configEvent = mConfigEvents[0]; 1160 mConfigEvents.removeAt(0); 1161 // release mLock before locking AudioFlinger mLock: lock order is always 1162 // AudioFlinger then ThreadBase to avoid cross deadlock 1163 mLock.unlock(); 1164 mAudioFlinger->mLock.lock(); 1165 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1166 mAudioFlinger->mLock.unlock(); 1167 mLock.lock(); 1168 } 1169 mLock.unlock(); 1170} 1171 1172status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1173{ 1174 const size_t SIZE = 256; 1175 char buffer[SIZE]; 1176 String8 result; 1177 1178 bool locked = tryLock(mLock); 1179 if (!locked) { 1180 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1181 write(fd, buffer, strlen(buffer)); 1182 } 1183 1184 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1185 result.append(buffer); 1186 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1187 result.append(buffer); 1188 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1189 result.append(buffer); 1190 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1197 result.append(buffer); 1198 1199 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1200 result.append(buffer); 1201 result.append(" Index Command"); 1202 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1203 snprintf(buffer, SIZE, "\n %02d ", i); 1204 result.append(buffer); 1205 result.append(mNewParameters[i]); 1206 } 1207 1208 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1209 result.append(buffer); 1210 snprintf(buffer, SIZE, " Index event param\n"); 1211 result.append(buffer); 1212 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1213 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1214 result.append(buffer); 1215 } 1216 result.append("\n"); 1217 1218 write(fd, result.string(), result.size()); 1219 1220 if (locked) { 1221 mLock.unlock(); 1222 } 1223 return NO_ERROR; 1224} 1225 1226status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1227{ 1228 const size_t SIZE = 256; 1229 char buffer[SIZE]; 1230 String8 result; 1231 1232 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1233 write(fd, buffer, strlen(buffer)); 1234 1235 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1236 sp<EffectChain> chain = mEffectChains[i]; 1237 if (chain != 0) { 1238 chain->dump(fd, args); 1239 } 1240 } 1241 return NO_ERROR; 1242} 1243 1244void AudioFlinger::ThreadBase::acquireWakeLock() 1245{ 1246 Mutex::Autolock _l(mLock); 1247 acquireWakeLock_l(); 1248} 1249 1250void AudioFlinger::ThreadBase::acquireWakeLock_l() 1251{ 1252 if (mPowerManager == 0) { 1253 // use checkService() to avoid blocking if power service is not up yet 1254 sp<IBinder> binder = 1255 defaultServiceManager()->checkService(String16("power")); 1256 if (binder == 0) { 1257 ALOGW("Thread %s cannot connect to the power manager service", mName); 1258 } else { 1259 mPowerManager = interface_cast<IPowerManager>(binder); 1260 binder->linkToDeath(mDeathRecipient); 1261 } 1262 } 1263 if (mPowerManager != 0) { 1264 sp<IBinder> binder = new BBinder(); 1265 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1266 binder, 1267 String16(mName)); 1268 if (status == NO_ERROR) { 1269 mWakeLockToken = binder; 1270 } 1271 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1272 } 1273} 1274 1275void AudioFlinger::ThreadBase::releaseWakeLock() 1276{ 1277 Mutex::Autolock _l(mLock); 1278 releaseWakeLock_l(); 1279} 1280 1281void AudioFlinger::ThreadBase::releaseWakeLock_l() 1282{ 1283 if (mWakeLockToken != 0) { 1284 ALOGV("releaseWakeLock_l() %s", mName); 1285 if (mPowerManager != 0) { 1286 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1287 } 1288 mWakeLockToken.clear(); 1289 } 1290} 1291 1292void AudioFlinger::ThreadBase::clearPowerManager() 1293{ 1294 Mutex::Autolock _l(mLock); 1295 releaseWakeLock_l(); 1296 mPowerManager.clear(); 1297} 1298 1299void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1300{ 1301 sp<ThreadBase> thread = mThread.promote(); 1302 if (thread != 0) { 1303 thread->clearPowerManager(); 1304 } 1305 ALOGW("power manager service died !!!"); 1306} 1307 1308void AudioFlinger::ThreadBase::setEffectSuspended( 1309 const effect_uuid_t *type, bool suspend, int sessionId) 1310{ 1311 Mutex::Autolock _l(mLock); 1312 setEffectSuspended_l(type, suspend, sessionId); 1313} 1314 1315void AudioFlinger::ThreadBase::setEffectSuspended_l( 1316 const effect_uuid_t *type, bool suspend, int sessionId) 1317{ 1318 sp<EffectChain> chain = getEffectChain_l(sessionId); 1319 if (chain != 0) { 1320 if (type != NULL) { 1321 chain->setEffectSuspended_l(type, suspend); 1322 } else { 1323 chain->setEffectSuspendedAll_l(suspend); 1324 } 1325 } 1326 1327 updateSuspendedSessions_l(type, suspend, sessionId); 1328} 1329 1330void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1331{ 1332 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1333 if (index < 0) { 1334 return; 1335 } 1336 1337 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1338 mSuspendedSessions.editValueAt(index); 1339 1340 for (size_t i = 0; i < sessionEffects.size(); i++) { 1341 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1342 for (int j = 0; j < desc->mRefCount; j++) { 1343 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1344 chain->setEffectSuspendedAll_l(true); 1345 } else { 1346 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1347 desc->mType.timeLow); 1348 chain->setEffectSuspended_l(&desc->mType, true); 1349 } 1350 } 1351 } 1352} 1353 1354void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1355 bool suspend, 1356 int sessionId) 1357{ 1358 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1359 1360 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1361 1362 if (suspend) { 1363 if (index >= 0) { 1364 sessionEffects = mSuspendedSessions.editValueAt(index); 1365 } else { 1366 mSuspendedSessions.add(sessionId, sessionEffects); 1367 } 1368 } else { 1369 if (index < 0) { 1370 return; 1371 } 1372 sessionEffects = mSuspendedSessions.editValueAt(index); 1373 } 1374 1375 1376 int key = EffectChain::kKeyForSuspendAll; 1377 if (type != NULL) { 1378 key = type->timeLow; 1379 } 1380 index = sessionEffects.indexOfKey(key); 1381 1382 sp <SuspendedSessionDesc> desc; 1383 if (suspend) { 1384 if (index >= 0) { 1385 desc = sessionEffects.valueAt(index); 1386 } else { 1387 desc = new SuspendedSessionDesc(); 1388 if (type != NULL) { 1389 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1390 } 1391 sessionEffects.add(key, desc); 1392 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1393 } 1394 desc->mRefCount++; 1395 } else { 1396 if (index < 0) { 1397 return; 1398 } 1399 desc = sessionEffects.valueAt(index); 1400 if (--desc->mRefCount == 0) { 1401 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1402 sessionEffects.removeItemsAt(index); 1403 if (sessionEffects.isEmpty()) { 1404 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1405 sessionId); 1406 mSuspendedSessions.removeItem(sessionId); 1407 } 1408 } 1409 } 1410 if (!sessionEffects.isEmpty()) { 1411 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1412 } 1413} 1414 1415void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1416 bool enabled, 1417 int sessionId) 1418{ 1419 Mutex::Autolock _l(mLock); 1420 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1421} 1422 1423void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1424 bool enabled, 1425 int sessionId) 1426{ 1427 if (mType != RECORD) { 1428 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1429 // another session. This gives the priority to well behaved effect control panels 1430 // and applications not using global effects. 1431 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1432 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1433 } 1434 } 1435 1436 sp<EffectChain> chain = getEffectChain_l(sessionId); 1437 if (chain != 0) { 1438 chain->checkSuspendOnEffectEnabled(effect, enabled); 1439 } 1440} 1441 1442// ---------------------------------------------------------------------------- 1443 1444AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1445 AudioStreamOut* output, 1446 audio_io_handle_t id, 1447 uint32_t device, 1448 type_t type) 1449 : ThreadBase(audioFlinger, id, device, type), 1450 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1451 // Assumes constructor is called by AudioFlinger with it's mLock held, 1452 // but it would be safer to explicitly pass initial masterMute as parameter 1453 mMasterMute(audioFlinger->masterMute_l()), 1454 // mStreamTypes[] initialized in constructor body 1455 mOutput(output), 1456 // Assumes constructor is called by AudioFlinger with it's mLock held, 1457 // but it would be safer to explicitly pass initial masterVolume as parameter 1458 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1459 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1460{ 1461 snprintf(mName, kNameLength, "AudioOut_%d", id); 1462 1463 readOutputParameters(); 1464 1465 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1466 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1467 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1468 stream = (audio_stream_type_t) (stream + 1)) { 1469 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1470 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1471 // initialized by stream_type_t default constructor 1472 // mStreamTypes[stream].valid = true; 1473 } 1474 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1475 // because mAudioFlinger doesn't have one to copy from 1476} 1477 1478AudioFlinger::PlaybackThread::~PlaybackThread() 1479{ 1480 delete [] mMixBuffer; 1481} 1482 1483status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1484{ 1485 dumpInternals(fd, args); 1486 dumpTracks(fd, args); 1487 dumpEffectChains(fd, args); 1488 return NO_ERROR; 1489} 1490 1491status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1492{ 1493 const size_t SIZE = 256; 1494 char buffer[SIZE]; 1495 String8 result; 1496 1497 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1498 result.append(buffer); 1499 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1500 for (size_t i = 0; i < mTracks.size(); ++i) { 1501 sp<Track> track = mTracks[i]; 1502 if (track != 0) { 1503 track->dump(buffer, SIZE); 1504 result.append(buffer); 1505 } 1506 } 1507 1508 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1509 result.append(buffer); 1510 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1511 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1512 sp<Track> track = mActiveTracks[i].promote(); 1513 if (track != 0) { 1514 track->dump(buffer, SIZE); 1515 result.append(buffer); 1516 } 1517 } 1518 write(fd, result.string(), result.size()); 1519 return NO_ERROR; 1520} 1521 1522status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1523{ 1524 const size_t SIZE = 256; 1525 char buffer[SIZE]; 1526 String8 result; 1527 1528 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1529 result.append(buffer); 1530 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1531 result.append(buffer); 1532 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1533 result.append(buffer); 1534 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1541 result.append(buffer); 1542 write(fd, result.string(), result.size()); 1543 1544 dumpBase(fd, args); 1545 1546 return NO_ERROR; 1547} 1548 1549// Thread virtuals 1550status_t AudioFlinger::PlaybackThread::readyToRun() 1551{ 1552 status_t status = initCheck(); 1553 if (status == NO_ERROR) { 1554 ALOGI("AudioFlinger's thread %p ready to run", this); 1555 } else { 1556 ALOGE("No working audio driver found."); 1557 } 1558 return status; 1559} 1560 1561void AudioFlinger::PlaybackThread::onFirstRef() 1562{ 1563 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1564} 1565 1566// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1567sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1568 const sp<AudioFlinger::Client>& client, 1569 audio_stream_type_t streamType, 1570 uint32_t sampleRate, 1571 audio_format_t format, 1572 uint32_t channelMask, 1573 int frameCount, 1574 const sp<IMemory>& sharedBuffer, 1575 int sessionId, 1576 bool isTimed, 1577 status_t *status) 1578{ 1579 sp<Track> track; 1580 status_t lStatus; 1581 1582 if (mType == DIRECT) { 1583 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1584 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1585 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1586 "for output %p with format %d", 1587 sampleRate, format, channelMask, mOutput, mFormat); 1588 lStatus = BAD_VALUE; 1589 goto Exit; 1590 } 1591 } 1592 } else { 1593 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1594 if (sampleRate > mSampleRate*2) { 1595 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1596 lStatus = BAD_VALUE; 1597 goto Exit; 1598 } 1599 } 1600 1601 lStatus = initCheck(); 1602 if (lStatus != NO_ERROR) { 1603 ALOGE("Audio driver not initialized."); 1604 goto Exit; 1605 } 1606 1607 { // scope for mLock 1608 Mutex::Autolock _l(mLock); 1609 1610 // all tracks in same audio session must share the same routing strategy otherwise 1611 // conflicts will happen when tracks are moved from one output to another by audio policy 1612 // manager 1613 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1614 for (size_t i = 0; i < mTracks.size(); ++i) { 1615 sp<Track> t = mTracks[i]; 1616 if (t != 0) { 1617 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1618 if (sessionId == t->sessionId() && strategy != actual) { 1619 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1620 strategy, actual); 1621 lStatus = BAD_VALUE; 1622 goto Exit; 1623 } 1624 } 1625 } 1626 1627 if (!isTimed) { 1628 track = new Track(this, client, streamType, sampleRate, format, 1629 channelMask, frameCount, sharedBuffer, sessionId); 1630 } else { 1631 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1632 channelMask, frameCount, sharedBuffer, sessionId); 1633 } 1634 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1635 lStatus = NO_MEMORY; 1636 goto Exit; 1637 } 1638 mTracks.add(track); 1639 1640 sp<EffectChain> chain = getEffectChain_l(sessionId); 1641 if (chain != 0) { 1642 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1643 track->setMainBuffer(chain->inBuffer()); 1644 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1645 chain->incTrackCnt(); 1646 } 1647 1648 // invalidate track immediately if the stream type was moved to another thread since 1649 // createTrack() was called by the client process. 1650 if (!mStreamTypes[streamType].valid) { 1651 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1652 this, streamType); 1653 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1654 } 1655 } 1656 lStatus = NO_ERROR; 1657 1658Exit: 1659 if(status) { 1660 *status = lStatus; 1661 } 1662 return track; 1663} 1664 1665uint32_t AudioFlinger::PlaybackThread::latency() const 1666{ 1667 Mutex::Autolock _l(mLock); 1668 if (initCheck() == NO_ERROR) { 1669 return mOutput->stream->get_latency(mOutput->stream); 1670 } else { 1671 return 0; 1672 } 1673} 1674 1675void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1676{ 1677 Mutex::Autolock _l(mLock); 1678 mMasterVolume = value; 1679} 1680 1681void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 setMasterMute_l(muted); 1685} 1686 1687void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1688{ 1689 Mutex::Autolock _l(mLock); 1690 mStreamTypes[stream].volume = value; 1691} 1692 1693void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1694{ 1695 Mutex::Autolock _l(mLock); 1696 mStreamTypes[stream].mute = muted; 1697} 1698 1699float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1700{ 1701 Mutex::Autolock _l(mLock); 1702 return mStreamTypes[stream].volume; 1703} 1704 1705// addTrack_l() must be called with ThreadBase::mLock held 1706status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1707{ 1708 status_t status = ALREADY_EXISTS; 1709 1710 // set retry count for buffer fill 1711 track->mRetryCount = kMaxTrackStartupRetries; 1712 if (mActiveTracks.indexOf(track) < 0) { 1713 // the track is newly added, make sure it fills up all its 1714 // buffers before playing. This is to ensure the client will 1715 // effectively get the latency it requested. 1716 track->mFillingUpStatus = Track::FS_FILLING; 1717 track->mResetDone = false; 1718 mActiveTracks.add(track); 1719 if (track->mainBuffer() != mMixBuffer) { 1720 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1721 if (chain != 0) { 1722 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1723 chain->incActiveTrackCnt(); 1724 } 1725 } 1726 1727 status = NO_ERROR; 1728 } 1729 1730 ALOGV("mWaitWorkCV.broadcast"); 1731 mWaitWorkCV.broadcast(); 1732 1733 return status; 1734} 1735 1736// destroyTrack_l() must be called with ThreadBase::mLock held 1737void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1738{ 1739 track->mState = TrackBase::TERMINATED; 1740 if (mActiveTracks.indexOf(track) < 0) { 1741 removeTrack_l(track); 1742 } 1743} 1744 1745void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1746{ 1747 mTracks.remove(track); 1748 deleteTrackName_l(track->name()); 1749 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1750 if (chain != 0) { 1751 chain->decTrackCnt(); 1752 } 1753} 1754 1755String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1756{ 1757 String8 out_s8 = String8(""); 1758 char *s; 1759 1760 Mutex::Autolock _l(mLock); 1761 if (initCheck() != NO_ERROR) { 1762 return out_s8; 1763 } 1764 1765 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1766 out_s8 = String8(s); 1767 free(s); 1768 return out_s8; 1769} 1770 1771// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1772void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1773 AudioSystem::OutputDescriptor desc; 1774 void *param2 = NULL; 1775 1776 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1777 1778 switch (event) { 1779 case AudioSystem::OUTPUT_OPENED: 1780 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1781 desc.channels = mChannelMask; 1782 desc.samplingRate = mSampleRate; 1783 desc.format = mFormat; 1784 desc.frameCount = mFrameCount; 1785 desc.latency = latency(); 1786 param2 = &desc; 1787 break; 1788 1789 case AudioSystem::STREAM_CONFIG_CHANGED: 1790 param2 = ¶m; 1791 case AudioSystem::OUTPUT_CLOSED: 1792 default: 1793 break; 1794 } 1795 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1796} 1797 1798void AudioFlinger::PlaybackThread::readOutputParameters() 1799{ 1800 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1801 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1802 mChannelCount = (uint16_t)popcount(mChannelMask); 1803 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1804 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1805 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1806 1807 // FIXME - Current mixer implementation only supports stereo output: Always 1808 // Allocate a stereo buffer even if HW output is mono. 1809 delete[] mMixBuffer; 1810 mMixBuffer = new int16_t[mFrameCount * 2]; 1811 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1812 1813 // force reconfiguration of effect chains and engines to take new buffer size and audio 1814 // parameters into account 1815 // Note that mLock is not held when readOutputParameters() is called from the constructor 1816 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1817 // matter. 1818 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1819 Vector< sp<EffectChain> > effectChains = mEffectChains; 1820 for (size_t i = 0; i < effectChains.size(); i ++) { 1821 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1822 } 1823} 1824 1825status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1826{ 1827 if (halFrames == NULL || dspFrames == NULL) { 1828 return BAD_VALUE; 1829 } 1830 Mutex::Autolock _l(mLock); 1831 if (initCheck() != NO_ERROR) { 1832 return INVALID_OPERATION; 1833 } 1834 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1835 1836 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1837} 1838 1839uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 uint32_t result = 0; 1843 if (getEffectChain_l(sessionId) != 0) { 1844 result = EFFECT_SESSION; 1845 } 1846 1847 for (size_t i = 0; i < mTracks.size(); ++i) { 1848 sp<Track> track = mTracks[i]; 1849 if (sessionId == track->sessionId() && 1850 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1851 result |= TRACK_SESSION; 1852 break; 1853 } 1854 } 1855 1856 return result; 1857} 1858 1859uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1860{ 1861 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1862 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1863 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1864 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1865 } 1866 for (size_t i = 0; i < mTracks.size(); i++) { 1867 sp<Track> track = mTracks[i]; 1868 if (sessionId == track->sessionId() && 1869 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1870 return AudioSystem::getStrategyForStream(track->streamType()); 1871 } 1872 } 1873 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1874} 1875 1876 1877AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1878{ 1879 Mutex::Autolock _l(mLock); 1880 return mOutput; 1881} 1882 1883AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1884{ 1885 Mutex::Autolock _l(mLock); 1886 AudioStreamOut *output = mOutput; 1887 mOutput = NULL; 1888 return output; 1889} 1890 1891// this method must always be called either with ThreadBase mLock held or inside the thread loop 1892audio_stream_t* AudioFlinger::PlaybackThread::stream() 1893{ 1894 if (mOutput == NULL) { 1895 return NULL; 1896 } 1897 return &mOutput->stream->common; 1898} 1899 1900uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1901{ 1902 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1903 // decoding and transfer time. So sleeping for half of the latency would likely cause 1904 // underruns 1905 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1906 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1907 } else { 1908 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1909 } 1910} 1911 1912// ---------------------------------------------------------------------------- 1913 1914AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1915 audio_io_handle_t id, uint32_t device, type_t type) 1916 : PlaybackThread(audioFlinger, output, id, device, type), 1917 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1918 mPrevMixerStatus(MIXER_IDLE) 1919{ 1920 // FIXME - Current mixer implementation only supports stereo output 1921 if (mChannelCount == 1) { 1922 ALOGE("Invalid audio hardware channel count"); 1923 } 1924} 1925 1926AudioFlinger::MixerThread::~MixerThread() 1927{ 1928 delete mAudioMixer; 1929} 1930 1931class CpuStats { 1932public: 1933 void sample(); 1934#ifdef DEBUG_CPU_USAGE 1935private: 1936 ThreadCpuUsage mCpu; 1937#endif 1938}; 1939 1940void CpuStats::sample() { 1941#ifdef DEBUG_CPU_USAGE 1942 const CentralTendencyStatistics& stats = mCpu.statistics(); 1943 mCpu.sampleAndEnable(); 1944 unsigned n = stats.n(); 1945 // mCpu.elapsed() is expensive, so don't call it every loop 1946 if ((n & 127) == 1) { 1947 long long elapsed = mCpu.elapsed(); 1948 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1949 double perLoop = elapsed / (double) n; 1950 double perLoop100 = perLoop * 0.01; 1951 double mean = stats.mean(); 1952 double stddev = stats.stddev(); 1953 double minimum = stats.minimum(); 1954 double maximum = stats.maximum(); 1955 mCpu.resetStatistics(); 1956 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1957 elapsed * .000000001, n, perLoop * .000001, 1958 mean * .001, 1959 stddev * .001, 1960 minimum * .001, 1961 maximum * .001, 1962 mean / perLoop100, 1963 stddev / perLoop100, 1964 minimum / perLoop100, 1965 maximum / perLoop100); 1966 } 1967 } 1968#endif 1969}; 1970 1971void AudioFlinger::PlaybackThread::checkSilentMode_l() 1972{ 1973 if (!mMasterMute) { 1974 char value[PROPERTY_VALUE_MAX]; 1975 if (property_get("ro.audio.silent", value, "0") > 0) { 1976 char *endptr; 1977 unsigned long ul = strtoul(value, &endptr, 0); 1978 if (*endptr == '\0' && ul != 0) { 1979 ALOGD("Silence is golden"); 1980 // The setprop command will not allow a property to be changed after 1981 // the first time it is set, so we don't have to worry about un-muting. 1982 setMasterMute_l(true); 1983 } 1984 } 1985 } 1986} 1987 1988bool AudioFlinger::MixerThread::threadLoop() 1989{ 1990 Vector< sp<Track> > tracksToRemove; 1991 mixer_state mixerStatus = MIXER_IDLE; 1992 nsecs_t standbyTime = systemTime(); 1993 size_t mixBufferSize = mFrameCount * mFrameSize; 1994 // FIXME: Relaxed timing because of a certain device that can't meet latency 1995 // Should be reduced to 2x after the vendor fixes the driver issue 1996 // increase threshold again due to low power audio mode. The way this warning threshold is 1997 // calculated and its usefulness should be reconsidered anyway. 1998 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1999 nsecs_t lastWarning = 0; 2000 bool longStandbyExit = false; 2001 uint32_t activeSleepTime = activeSleepTimeUs(); 2002 uint32_t idleSleepTime = idleSleepTimeUs(); 2003 uint32_t sleepTime = idleSleepTime; 2004 uint32_t sleepTimeShift = 0; 2005 Vector< sp<EffectChain> > effectChains; 2006 CpuStats cpuStats; 2007 2008 acquireWakeLock(); 2009 2010 while (!exitPending()) 2011 { 2012 cpuStats.sample(); 2013 processConfigEvents(); 2014 2015 mixerStatus = MIXER_IDLE; 2016 { // scope for mLock 2017 2018 Mutex::Autolock _l(mLock); 2019 2020 if (checkForNewParameters_l()) { 2021 mixBufferSize = mFrameCount * mFrameSize; 2022 // FIXME: Relaxed timing because of a certain device that can't meet latency 2023 // Should be reduced to 2x after the vendor fixes the driver issue 2024 // increase threshold again due to low power audio mode. The way this warning 2025 // threshold is calculated and its usefulness should be reconsidered anyway. 2026 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2027 activeSleepTime = activeSleepTimeUs(); 2028 idleSleepTime = idleSleepTimeUs(); 2029 } 2030 2031 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2032 2033 // put audio hardware into standby after short delay 2034 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2035 mSuspended)) { 2036 if (!mStandby) { 2037 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2038 mOutput->stream->common.standby(&mOutput->stream->common); 2039 mStandby = true; 2040 mBytesWritten = 0; 2041 } 2042 2043 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2044 // we're about to wait, flush the binder command buffer 2045 IPCThreadState::self()->flushCommands(); 2046 2047 if (exitPending()) break; 2048 2049 releaseWakeLock_l(); 2050 // wait until we have something to do... 2051 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2052 mWaitWorkCV.wait(mLock); 2053 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2054 acquireWakeLock_l(); 2055 2056 mPrevMixerStatus = MIXER_IDLE; 2057 checkSilentMode_l(); 2058 2059 standbyTime = systemTime() + mStandbyTimeInNsecs; 2060 sleepTime = idleSleepTime; 2061 sleepTimeShift = 0; 2062 continue; 2063 } 2064 } 2065 2066 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2067 2068 // prevent any changes in effect chain list and in each effect chain 2069 // during mixing and effect process as the audio buffers could be deleted 2070 // or modified if an effect is created or deleted 2071 lockEffectChains_l(effectChains); 2072 } 2073 2074 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2075 // obtain the presentation timestamp of the next output buffer 2076 int64_t pts; 2077 status_t status = INVALID_OPERATION; 2078 2079 if (NULL != mOutput->stream->get_next_write_timestamp) { 2080 status = mOutput->stream->get_next_write_timestamp( 2081 mOutput->stream, &pts); 2082 } 2083 2084 if (status != NO_ERROR) { 2085 pts = AudioBufferProvider::kInvalidPTS; 2086 } 2087 2088 // mix buffers... 2089 mAudioMixer->process(pts); 2090 // increase sleep time progressively when application underrun condition clears. 2091 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2092 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2093 // such that we would underrun the audio HAL. 2094 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2095 sleepTimeShift--; 2096 } 2097 sleepTime = 0; 2098 standbyTime = systemTime() + mStandbyTimeInNsecs; 2099 //TODO: delay standby when effects have a tail 2100 } else { 2101 // If no tracks are ready, sleep once for the duration of an output 2102 // buffer size, then write 0s to the output 2103 if (sleepTime == 0) { 2104 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2105 sleepTime = activeSleepTime >> sleepTimeShift; 2106 if (sleepTime < kMinThreadSleepTimeUs) { 2107 sleepTime = kMinThreadSleepTimeUs; 2108 } 2109 // reduce sleep time in case of consecutive application underruns to avoid 2110 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2111 // duration we would end up writing less data than needed by the audio HAL if 2112 // the condition persists. 2113 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2114 sleepTimeShift++; 2115 } 2116 } else { 2117 sleepTime = idleSleepTime; 2118 } 2119 } else if (mBytesWritten != 0 || 2120 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2121 memset (mMixBuffer, 0, mixBufferSize); 2122 sleepTime = 0; 2123 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2124 } 2125 // TODO add standby time extension fct of effect tail 2126 } 2127 2128 if (mSuspended) { 2129 sleepTime = suspendSleepTimeUs(); 2130 } 2131 // sleepTime == 0 means we must write to audio hardware 2132 if (sleepTime == 0) { 2133 for (size_t i = 0; i < effectChains.size(); i ++) { 2134 effectChains[i]->process_l(); 2135 } 2136 // enable changes in effect chain 2137 unlockEffectChains(effectChains); 2138 mLastWriteTime = systemTime(); 2139 mInWrite = true; 2140 mBytesWritten += mixBufferSize; 2141 2142 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2143 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2144 mNumWrites++; 2145 mInWrite = false; 2146 nsecs_t now = systemTime(); 2147 nsecs_t delta = now - mLastWriteTime; 2148 if (!mStandby && delta > maxPeriod) { 2149 mNumDelayedWrites++; 2150 if ((now - lastWarning) > kWarningThrottleNs) { 2151 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2152 ns2ms(delta), mNumDelayedWrites, this); 2153 lastWarning = now; 2154 } 2155 if (mStandby) { 2156 longStandbyExit = true; 2157 } 2158 } 2159 mStandby = false; 2160 } else { 2161 // enable changes in effect chain 2162 unlockEffectChains(effectChains); 2163 usleep(sleepTime); 2164 } 2165 2166 // finally let go of all our tracks, without the lock held 2167 // since we can't guarantee the destructors won't acquire that 2168 // same lock. 2169 tracksToRemove.clear(); 2170 2171 // Effect chains will be actually deleted here if they were removed from 2172 // mEffectChains list during mixing or effects processing 2173 effectChains.clear(); 2174 } 2175 2176 if (!mStandby) { 2177 mOutput->stream->common.standby(&mOutput->stream->common); 2178 } 2179 2180 releaseWakeLock(); 2181 2182 ALOGV("MixerThread %p exiting", this); 2183 return false; 2184} 2185 2186// prepareTracks_l() must be called with ThreadBase::mLock held 2187AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2188 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2189{ 2190 2191 mixer_state mixerStatus = MIXER_IDLE; 2192 // find out which tracks need to be processed 2193 size_t count = activeTracks.size(); 2194 size_t mixedTracks = 0; 2195 size_t tracksWithEffect = 0; 2196 2197 float masterVolume = mMasterVolume; 2198 bool masterMute = mMasterMute; 2199 2200 if (masterMute) { 2201 masterVolume = 0; 2202 } 2203 // Delegate master volume control to effect in output mix effect chain if needed 2204 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2205 if (chain != 0) { 2206 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2207 chain->setVolume_l(&v, &v); 2208 masterVolume = (float)((v + (1 << 23)) >> 24); 2209 chain.clear(); 2210 } 2211 2212 for (size_t i=0 ; i<count ; i++) { 2213 sp<Track> t = activeTracks[i].promote(); 2214 if (t == 0) continue; 2215 2216 // this const just means the local variable doesn't change 2217 Track* const track = t.get(); 2218 audio_track_cblk_t* cblk = track->cblk(); 2219 2220 // The first time a track is added we wait 2221 // for all its buffers to be filled before processing it 2222 int name = track->name(); 2223 // make sure that we have enough frames to mix one full buffer. 2224 // enforce this condition only once to enable draining the buffer in case the client 2225 // app does not call stop() and relies on underrun to stop: 2226 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2227 // during last round 2228 uint32_t minFrames = 1; 2229 if (!track->isStopped() && !track->isPausing() && 2230 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2231 if (t->sampleRate() == (int)mSampleRate) { 2232 minFrames = mFrameCount; 2233 } else { 2234 // +1 for rounding and +1 for additional sample needed for interpolation 2235 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2236 // add frames already consumed but not yet released by the resampler 2237 // because cblk->framesReady() will include these frames 2238 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2239 // the minimum track buffer size is normally twice the number of frames necessary 2240 // to fill one buffer and the resampler should not leave more than one buffer worth 2241 // of unreleased frames after each pass, but just in case... 2242 ALOG_ASSERT(minFrames <= cblk->frameCount); 2243 } 2244 } 2245 if ((track->framesReady() >= minFrames) && track->isReady() && 2246 !track->isPaused() && !track->isTerminated()) 2247 { 2248 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2249 2250 mixedTracks++; 2251 2252 // track->mainBuffer() != mMixBuffer means there is an effect chain 2253 // connected to the track 2254 chain.clear(); 2255 if (track->mainBuffer() != mMixBuffer) { 2256 chain = getEffectChain_l(track->sessionId()); 2257 // Delegate volume control to effect in track effect chain if needed 2258 if (chain != 0) { 2259 tracksWithEffect++; 2260 } else { 2261 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2262 name, track->sessionId()); 2263 } 2264 } 2265 2266 2267 int param = AudioMixer::VOLUME; 2268 if (track->mFillingUpStatus == Track::FS_FILLED) { 2269 // no ramp for the first volume setting 2270 track->mFillingUpStatus = Track::FS_ACTIVE; 2271 if (track->mState == TrackBase::RESUMING) { 2272 track->mState = TrackBase::ACTIVE; 2273 param = AudioMixer::RAMP_VOLUME; 2274 } 2275 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2276 } else if (cblk->server != 0) { 2277 // If the track is stopped before the first frame was mixed, 2278 // do not apply ramp 2279 param = AudioMixer::RAMP_VOLUME; 2280 } 2281 2282 // compute volume for this track 2283 uint32_t vl, vr, va; 2284 if (track->isMuted() || track->isPausing() || 2285 mStreamTypes[track->streamType()].mute) { 2286 vl = vr = va = 0; 2287 if (track->isPausing()) { 2288 track->setPaused(); 2289 } 2290 } else { 2291 2292 // read original volumes with volume control 2293 float typeVolume = mStreamTypes[track->streamType()].volume; 2294 float v = masterVolume * typeVolume; 2295 uint32_t vlr = cblk->getVolumeLR(); 2296 vl = vlr & 0xFFFF; 2297 vr = vlr >> 16; 2298 // track volumes come from shared memory, so can't be trusted and must be clamped 2299 if (vl > MAX_GAIN_INT) { 2300 ALOGV("Track left volume out of range: %04X", vl); 2301 vl = MAX_GAIN_INT; 2302 } 2303 if (vr > MAX_GAIN_INT) { 2304 ALOGV("Track right volume out of range: %04X", vr); 2305 vr = MAX_GAIN_INT; 2306 } 2307 // now apply the master volume and stream type volume 2308 vl = (uint32_t)(v * vl) << 12; 2309 vr = (uint32_t)(v * vr) << 12; 2310 // assuming master volume and stream type volume each go up to 1.0, 2311 // vl and vr are now in 8.24 format 2312 2313 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2314 // send level comes from shared memory and so may be corrupt 2315 if (sendLevel > MAX_GAIN_INT) { 2316 ALOGV("Track send level out of range: %04X", sendLevel); 2317 sendLevel = MAX_GAIN_INT; 2318 } 2319 va = (uint32_t)(v * sendLevel); 2320 } 2321 // Delegate volume control to effect in track effect chain if needed 2322 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2323 // Do not ramp volume if volume is controlled by effect 2324 param = AudioMixer::VOLUME; 2325 track->mHasVolumeController = true; 2326 } else { 2327 // force no volume ramp when volume controller was just disabled or removed 2328 // from effect chain to avoid volume spike 2329 if (track->mHasVolumeController) { 2330 param = AudioMixer::VOLUME; 2331 } 2332 track->mHasVolumeController = false; 2333 } 2334 2335 // Convert volumes from 8.24 to 4.12 format 2336 // This additional clamping is needed in case chain->setVolume_l() overshot 2337 vl = (vl + (1 << 11)) >> 12; 2338 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2339 vr = (vr + (1 << 11)) >> 12; 2340 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2341 2342 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2343 2344 // XXX: these things DON'T need to be done each time 2345 mAudioMixer->setBufferProvider(name, track); 2346 mAudioMixer->enable(name); 2347 2348 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2349 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2350 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2351 mAudioMixer->setParameter( 2352 name, 2353 AudioMixer::TRACK, 2354 AudioMixer::FORMAT, (void *)track->format()); 2355 mAudioMixer->setParameter( 2356 name, 2357 AudioMixer::TRACK, 2358 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2359 mAudioMixer->setParameter( 2360 name, 2361 AudioMixer::RESAMPLE, 2362 AudioMixer::SAMPLE_RATE, 2363 (void *)(cblk->sampleRate)); 2364 mAudioMixer->setParameter( 2365 name, 2366 AudioMixer::TRACK, 2367 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2368 mAudioMixer->setParameter( 2369 name, 2370 AudioMixer::TRACK, 2371 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2372 2373 // reset retry count 2374 track->mRetryCount = kMaxTrackRetries; 2375 // If one track is ready, set the mixer ready if: 2376 // - the mixer was not ready during previous round OR 2377 // - no other track is not ready 2378 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2379 mixerStatus != MIXER_TRACKS_ENABLED) { 2380 mixerStatus = MIXER_TRACKS_READY; 2381 } 2382 } else { 2383 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2384 if (track->isStopped()) { 2385 track->reset(); 2386 } 2387 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2388 // We have consumed all the buffers of this track. 2389 // Remove it from the list of active tracks. 2390 tracksToRemove->add(track); 2391 } else { 2392 // No buffers for this track. Give it a few chances to 2393 // fill a buffer, then remove it from active list. 2394 if (--(track->mRetryCount) <= 0) { 2395 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2396 tracksToRemove->add(track); 2397 // indicate to client process that the track was disabled because of underrun 2398 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2399 // If one track is not ready, mark the mixer also not ready if: 2400 // - the mixer was ready during previous round OR 2401 // - no other track is ready 2402 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2403 mixerStatus != MIXER_TRACKS_READY) { 2404 mixerStatus = MIXER_TRACKS_ENABLED; 2405 } 2406 } 2407 mAudioMixer->disable(name); 2408 } 2409 } 2410 2411 // remove all the tracks that need to be... 2412 count = tracksToRemove->size(); 2413 if (CC_UNLIKELY(count)) { 2414 for (size_t i=0 ; i<count ; i++) { 2415 const sp<Track>& track = tracksToRemove->itemAt(i); 2416 mActiveTracks.remove(track); 2417 if (track->mainBuffer() != mMixBuffer) { 2418 chain = getEffectChain_l(track->sessionId()); 2419 if (chain != 0) { 2420 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2421 chain->decActiveTrackCnt(); 2422 } 2423 } 2424 if (track->isTerminated()) { 2425 removeTrack_l(track); 2426 } 2427 } 2428 } 2429 2430 // mix buffer must be cleared if all tracks are connected to an 2431 // effect chain as in this case the mixer will not write to 2432 // mix buffer and track effects will accumulate into it 2433 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2434 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2435 } 2436 2437 mPrevMixerStatus = mixerStatus; 2438 return mixerStatus; 2439} 2440 2441void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2442{ 2443 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2444 this, streamType, mTracks.size()); 2445 Mutex::Autolock _l(mLock); 2446 2447 size_t size = mTracks.size(); 2448 for (size_t i = 0; i < size; i++) { 2449 sp<Track> t = mTracks[i]; 2450 if (t->streamType() == streamType) { 2451 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2452 t->mCblk->cv.signal(); 2453 } 2454 } 2455} 2456 2457void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2458{ 2459 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2460 this, streamType, valid); 2461 Mutex::Autolock _l(mLock); 2462 2463 mStreamTypes[streamType].valid = valid; 2464} 2465 2466// getTrackName_l() must be called with ThreadBase::mLock held 2467int AudioFlinger::MixerThread::getTrackName_l() 2468{ 2469 return mAudioMixer->getTrackName(); 2470} 2471 2472// deleteTrackName_l() must be called with ThreadBase::mLock held 2473void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2474{ 2475 ALOGV("remove track (%d) and delete from mixer", name); 2476 mAudioMixer->deleteTrackName(name); 2477} 2478 2479// checkForNewParameters_l() must be called with ThreadBase::mLock held 2480bool AudioFlinger::MixerThread::checkForNewParameters_l() 2481{ 2482 bool reconfig = false; 2483 2484 while (!mNewParameters.isEmpty()) { 2485 status_t status = NO_ERROR; 2486 String8 keyValuePair = mNewParameters[0]; 2487 AudioParameter param = AudioParameter(keyValuePair); 2488 int value; 2489 2490 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2491 reconfig = true; 2492 } 2493 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2494 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2495 status = BAD_VALUE; 2496 } else { 2497 reconfig = true; 2498 } 2499 } 2500 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2501 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2502 status = BAD_VALUE; 2503 } else { 2504 reconfig = true; 2505 } 2506 } 2507 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2508 // do not accept frame count changes if tracks are open as the track buffer 2509 // size depends on frame count and correct behavior would not be guaranteed 2510 // if frame count is changed after track creation 2511 if (!mTracks.isEmpty()) { 2512 status = INVALID_OPERATION; 2513 } else { 2514 reconfig = true; 2515 } 2516 } 2517 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2518 // when changing the audio output device, call addBatteryData to notify 2519 // the change 2520 if ((int)mDevice != value) { 2521 uint32_t params = 0; 2522 // check whether speaker is on 2523 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2524 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2525 } 2526 2527 int deviceWithoutSpeaker 2528 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2529 // check if any other device (except speaker) is on 2530 if (value & deviceWithoutSpeaker ) { 2531 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2532 } 2533 2534 if (params != 0) { 2535 addBatteryData(params); 2536 } 2537 } 2538 2539 // forward device change to effects that have requested to be 2540 // aware of attached audio device. 2541 mDevice = (uint32_t)value; 2542 for (size_t i = 0; i < mEffectChains.size(); i++) { 2543 mEffectChains[i]->setDevice_l(mDevice); 2544 } 2545 } 2546 2547 if (status == NO_ERROR) { 2548 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2549 keyValuePair.string()); 2550 if (!mStandby && status == INVALID_OPERATION) { 2551 mOutput->stream->common.standby(&mOutput->stream->common); 2552 mStandby = true; 2553 mBytesWritten = 0; 2554 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2555 keyValuePair.string()); 2556 } 2557 if (status == NO_ERROR && reconfig) { 2558 delete mAudioMixer; 2559 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2560 mAudioMixer = NULL; 2561 readOutputParameters(); 2562 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2563 for (size_t i = 0; i < mTracks.size() ; i++) { 2564 int name = getTrackName_l(); 2565 if (name < 0) break; 2566 mTracks[i]->mName = name; 2567 // limit track sample rate to 2 x new output sample rate 2568 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2569 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2570 } 2571 } 2572 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2573 } 2574 } 2575 2576 mNewParameters.removeAt(0); 2577 2578 mParamStatus = status; 2579 mParamCond.signal(); 2580 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2581 // already timed out waiting for the status and will never signal the condition. 2582 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2583 } 2584 return reconfig; 2585} 2586 2587status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2588{ 2589 const size_t SIZE = 256; 2590 char buffer[SIZE]; 2591 String8 result; 2592 2593 PlaybackThread::dumpInternals(fd, args); 2594 2595 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2596 result.append(buffer); 2597 write(fd, result.string(), result.size()); 2598 return NO_ERROR; 2599} 2600 2601uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2602{ 2603 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2604} 2605 2606uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2607{ 2608 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2609} 2610 2611// ---------------------------------------------------------------------------- 2612AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2613 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2614 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2615 // mLeftVolFloat, mRightVolFloat 2616 // mLeftVolShort, mRightVolShort 2617{ 2618} 2619 2620AudioFlinger::DirectOutputThread::~DirectOutputThread() 2621{ 2622} 2623 2624void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2625{ 2626 // Do not apply volume on compressed audio 2627 if (!audio_is_linear_pcm(mFormat)) { 2628 return; 2629 } 2630 2631 // convert to signed 16 bit before volume calculation 2632 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2633 size_t count = mFrameCount * mChannelCount; 2634 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2635 int16_t *dst = mMixBuffer + count-1; 2636 while(count--) { 2637 *dst-- = (int16_t)(*src--^0x80) << 8; 2638 } 2639 } 2640 2641 size_t frameCount = mFrameCount; 2642 int16_t *out = mMixBuffer; 2643 if (ramp) { 2644 if (mChannelCount == 1) { 2645 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2646 int32_t vlInc = d / (int32_t)frameCount; 2647 int32_t vl = ((int32_t)mLeftVolShort << 16); 2648 do { 2649 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2650 out++; 2651 vl += vlInc; 2652 } while (--frameCount); 2653 2654 } else { 2655 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2656 int32_t vlInc = d / (int32_t)frameCount; 2657 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2658 int32_t vrInc = d / (int32_t)frameCount; 2659 int32_t vl = ((int32_t)mLeftVolShort << 16); 2660 int32_t vr = ((int32_t)mRightVolShort << 16); 2661 do { 2662 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2663 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2664 out += 2; 2665 vl += vlInc; 2666 vr += vrInc; 2667 } while (--frameCount); 2668 } 2669 } else { 2670 if (mChannelCount == 1) { 2671 do { 2672 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2673 out++; 2674 } while (--frameCount); 2675 } else { 2676 do { 2677 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2678 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2679 out += 2; 2680 } while (--frameCount); 2681 } 2682 } 2683 2684 // convert back to unsigned 8 bit after volume calculation 2685 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2686 size_t count = mFrameCount * mChannelCount; 2687 int16_t *src = mMixBuffer; 2688 uint8_t *dst = (uint8_t *)mMixBuffer; 2689 while(count--) { 2690 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2691 } 2692 } 2693 2694 mLeftVolShort = leftVol; 2695 mRightVolShort = rightVol; 2696} 2697 2698bool AudioFlinger::DirectOutputThread::threadLoop() 2699{ 2700 mixer_state mixerStatus = MIXER_IDLE; 2701 sp<Track> trackToRemove; 2702 sp<Track> activeTrack; 2703 nsecs_t standbyTime = systemTime(); 2704 size_t mixBufferSize = mFrameCount*mFrameSize; 2705 uint32_t activeSleepTime = activeSleepTimeUs(); 2706 uint32_t idleSleepTime = idleSleepTimeUs(); 2707 uint32_t sleepTime = idleSleepTime; 2708 // use shorter standby delay as on normal output to release 2709 // hardware resources as soon as possible 2710 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2711 2712 acquireWakeLock(); 2713 2714 while (!exitPending()) 2715 { 2716 bool rampVolume; 2717 uint16_t leftVol; 2718 uint16_t rightVol; 2719 Vector< sp<EffectChain> > effectChains; 2720 2721 processConfigEvents(); 2722 2723 mixerStatus = MIXER_IDLE; 2724 2725 { // scope for the mLock 2726 2727 Mutex::Autolock _l(mLock); 2728 2729 if (checkForNewParameters_l()) { 2730 mixBufferSize = mFrameCount*mFrameSize; 2731 activeSleepTime = activeSleepTimeUs(); 2732 idleSleepTime = idleSleepTimeUs(); 2733 standbyDelay = microseconds(activeSleepTime*2); 2734 } 2735 2736 // put audio hardware into standby after short delay 2737 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2738 mSuspended)) { 2739 // wait until we have something to do... 2740 if (!mStandby) { 2741 ALOGV("Audio hardware entering standby, mixer %p", this); 2742 mOutput->stream->common.standby(&mOutput->stream->common); 2743 mStandby = true; 2744 mBytesWritten = 0; 2745 } 2746 2747 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2748 // we're about to wait, flush the binder command buffer 2749 IPCThreadState::self()->flushCommands(); 2750 2751 if (exitPending()) break; 2752 2753 releaseWakeLock_l(); 2754 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2755 mWaitWorkCV.wait(mLock); 2756 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2757 acquireWakeLock_l(); 2758 2759 checkSilentMode_l(); 2760 2761 standbyTime = systemTime() + standbyDelay; 2762 sleepTime = idleSleepTime; 2763 continue; 2764 } 2765 } 2766 2767 effectChains = mEffectChains; 2768 2769 // find out which tracks need to be processed 2770 if (mActiveTracks.size() != 0) { 2771 sp<Track> t = mActiveTracks[0].promote(); 2772 if (t == 0) continue; 2773 2774 Track* const track = t.get(); 2775 audio_track_cblk_t* cblk = track->cblk(); 2776 2777 // The first time a track is added we wait 2778 // for all its buffers to be filled before processing it 2779 if (cblk->framesReady() && track->isReady() && 2780 !track->isPaused() && !track->isTerminated()) 2781 { 2782 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2783 2784 if (track->mFillingUpStatus == Track::FS_FILLED) { 2785 track->mFillingUpStatus = Track::FS_ACTIVE; 2786 mLeftVolFloat = mRightVolFloat = 0; 2787 mLeftVolShort = mRightVolShort = 0; 2788 if (track->mState == TrackBase::RESUMING) { 2789 track->mState = TrackBase::ACTIVE; 2790 rampVolume = true; 2791 } 2792 } else if (cblk->server != 0) { 2793 // If the track is stopped before the first frame was mixed, 2794 // do not apply ramp 2795 rampVolume = true; 2796 } 2797 // compute volume for this track 2798 float left, right; 2799 if (track->isMuted() || mMasterMute || track->isPausing() || 2800 mStreamTypes[track->streamType()].mute) { 2801 left = right = 0; 2802 if (track->isPausing()) { 2803 track->setPaused(); 2804 } 2805 } else { 2806 float typeVolume = mStreamTypes[track->streamType()].volume; 2807 float v = mMasterVolume * typeVolume; 2808 uint32_t vlr = cblk->getVolumeLR(); 2809 float v_clamped = v * (vlr & 0xFFFF); 2810 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2811 left = v_clamped/MAX_GAIN; 2812 v_clamped = v * (vlr >> 16); 2813 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2814 right = v_clamped/MAX_GAIN; 2815 } 2816 2817 if (left != mLeftVolFloat || right != mRightVolFloat) { 2818 mLeftVolFloat = left; 2819 mRightVolFloat = right; 2820 2821 // If audio HAL implements volume control, 2822 // force software volume to nominal value 2823 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2824 left = 1.0f; 2825 right = 1.0f; 2826 } 2827 2828 // Convert volumes from float to 8.24 2829 uint32_t vl = (uint32_t)(left * (1 << 24)); 2830 uint32_t vr = (uint32_t)(right * (1 << 24)); 2831 2832 // Delegate volume control to effect in track effect chain if needed 2833 // only one effect chain can be present on DirectOutputThread, so if 2834 // there is one, the track is connected to it 2835 if (!effectChains.isEmpty()) { 2836 // Do not ramp volume if volume is controlled by effect 2837 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2838 rampVolume = false; 2839 } 2840 } 2841 2842 // Convert volumes from 8.24 to 4.12 format 2843 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2844 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2845 leftVol = (uint16_t)v_clamped; 2846 v_clamped = (vr + (1 << 11)) >> 12; 2847 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2848 rightVol = (uint16_t)v_clamped; 2849 } else { 2850 leftVol = mLeftVolShort; 2851 rightVol = mRightVolShort; 2852 rampVolume = false; 2853 } 2854 2855 // reset retry count 2856 track->mRetryCount = kMaxTrackRetriesDirect; 2857 activeTrack = t; 2858 mixerStatus = MIXER_TRACKS_READY; 2859 } else { 2860 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2861 if (track->isStopped()) { 2862 track->reset(); 2863 } 2864 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2865 // We have consumed all the buffers of this track. 2866 // Remove it from the list of active tracks. 2867 trackToRemove = track; 2868 } else { 2869 // No buffers for this track. Give it a few chances to 2870 // fill a buffer, then remove it from active list. 2871 if (--(track->mRetryCount) <= 0) { 2872 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2873 trackToRemove = track; 2874 } else { 2875 mixerStatus = MIXER_TRACKS_ENABLED; 2876 } 2877 } 2878 } 2879 } 2880 2881 // remove all the tracks that need to be... 2882 if (CC_UNLIKELY(trackToRemove != 0)) { 2883 mActiveTracks.remove(trackToRemove); 2884 if (!effectChains.isEmpty()) { 2885 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2886 trackToRemove->sessionId()); 2887 effectChains[0]->decActiveTrackCnt(); 2888 } 2889 if (trackToRemove->isTerminated()) { 2890 removeTrack_l(trackToRemove); 2891 } 2892 } 2893 2894 lockEffectChains_l(effectChains); 2895 } 2896 2897 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2898 AudioBufferProvider::Buffer buffer; 2899 size_t frameCount = mFrameCount; 2900 int8_t *curBuf = (int8_t *)mMixBuffer; 2901 // output audio to hardware 2902 while (frameCount) { 2903 buffer.frameCount = frameCount; 2904 activeTrack->getNextBuffer(&buffer, 2905 AudioBufferProvider::kInvalidPTS); 2906 if (CC_UNLIKELY(buffer.raw == NULL)) { 2907 memset(curBuf, 0, frameCount * mFrameSize); 2908 break; 2909 } 2910 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2911 frameCount -= buffer.frameCount; 2912 curBuf += buffer.frameCount * mFrameSize; 2913 activeTrack->releaseBuffer(&buffer); 2914 } 2915 sleepTime = 0; 2916 standbyTime = systemTime() + standbyDelay; 2917 } else { 2918 if (sleepTime == 0) { 2919 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2920 sleepTime = activeSleepTime; 2921 } else { 2922 sleepTime = idleSleepTime; 2923 } 2924 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2925 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2926 sleepTime = 0; 2927 } 2928 } 2929 2930 if (mSuspended) { 2931 sleepTime = suspendSleepTimeUs(); 2932 } 2933 // sleepTime == 0 means we must write to audio hardware 2934 if (sleepTime == 0) { 2935 if (mixerStatus == MIXER_TRACKS_READY) { 2936 applyVolume(leftVol, rightVol, rampVolume); 2937 } 2938 for (size_t i = 0; i < effectChains.size(); i ++) { 2939 effectChains[i]->process_l(); 2940 } 2941 unlockEffectChains(effectChains); 2942 2943 mLastWriteTime = systemTime(); 2944 mInWrite = true; 2945 mBytesWritten += mixBufferSize; 2946 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2947 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2948 mNumWrites++; 2949 mInWrite = false; 2950 mStandby = false; 2951 } else { 2952 unlockEffectChains(effectChains); 2953 usleep(sleepTime); 2954 } 2955 2956 // finally let go of removed track, without the lock held 2957 // since we can't guarantee the destructors won't acquire that 2958 // same lock. 2959 trackToRemove.clear(); 2960 activeTrack.clear(); 2961 2962 // Effect chains will be actually deleted here if they were removed from 2963 // mEffectChains list during mixing or effects processing 2964 effectChains.clear(); 2965 } 2966 2967 if (!mStandby) { 2968 mOutput->stream->common.standby(&mOutput->stream->common); 2969 } 2970 2971 releaseWakeLock(); 2972 2973 ALOGV("DirectOutputThread %p exiting", this); 2974 return false; 2975} 2976 2977// getTrackName_l() must be called with ThreadBase::mLock held 2978int AudioFlinger::DirectOutputThread::getTrackName_l() 2979{ 2980 return 0; 2981} 2982 2983// deleteTrackName_l() must be called with ThreadBase::mLock held 2984void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2985{ 2986} 2987 2988// checkForNewParameters_l() must be called with ThreadBase::mLock held 2989bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2990{ 2991 bool reconfig = false; 2992 2993 while (!mNewParameters.isEmpty()) { 2994 status_t status = NO_ERROR; 2995 String8 keyValuePair = mNewParameters[0]; 2996 AudioParameter param = AudioParameter(keyValuePair); 2997 int value; 2998 2999 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3000 // do not accept frame count changes if tracks are open as the track buffer 3001 // size depends on frame count and correct behavior would not be garantied 3002 // if frame count is changed after track creation 3003 if (!mTracks.isEmpty()) { 3004 status = INVALID_OPERATION; 3005 } else { 3006 reconfig = true; 3007 } 3008 } 3009 if (status == NO_ERROR) { 3010 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3011 keyValuePair.string()); 3012 if (!mStandby && status == INVALID_OPERATION) { 3013 mOutput->stream->common.standby(&mOutput->stream->common); 3014 mStandby = true; 3015 mBytesWritten = 0; 3016 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3017 keyValuePair.string()); 3018 } 3019 if (status == NO_ERROR && reconfig) { 3020 readOutputParameters(); 3021 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3022 } 3023 } 3024 3025 mNewParameters.removeAt(0); 3026 3027 mParamStatus = status; 3028 mParamCond.signal(); 3029 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3030 // already timed out waiting for the status and will never signal the condition. 3031 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3032 } 3033 return reconfig; 3034} 3035 3036uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3037{ 3038 uint32_t time; 3039 if (audio_is_linear_pcm(mFormat)) { 3040 time = PlaybackThread::activeSleepTimeUs(); 3041 } else { 3042 time = 10000; 3043 } 3044 return time; 3045} 3046 3047uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3048{ 3049 uint32_t time; 3050 if (audio_is_linear_pcm(mFormat)) { 3051 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3052 } else { 3053 time = 10000; 3054 } 3055 return time; 3056} 3057 3058uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3059{ 3060 uint32_t time; 3061 if (audio_is_linear_pcm(mFormat)) { 3062 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3063 } else { 3064 time = 10000; 3065 } 3066 return time; 3067} 3068 3069 3070// ---------------------------------------------------------------------------- 3071 3072AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3073 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3074 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3075 mWaitTimeMs(UINT_MAX) 3076{ 3077 addOutputTrack(mainThread); 3078} 3079 3080AudioFlinger::DuplicatingThread::~DuplicatingThread() 3081{ 3082 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3083 mOutputTracks[i]->destroy(); 3084 } 3085} 3086 3087bool AudioFlinger::DuplicatingThread::threadLoop() 3088{ 3089 Vector< sp<Track> > tracksToRemove; 3090 mixer_state mixerStatus = MIXER_IDLE; 3091 nsecs_t standbyTime = systemTime(); 3092 size_t mixBufferSize = mFrameCount*mFrameSize; 3093 SortedVector< sp<OutputTrack> > outputTracks; 3094 uint32_t writeFrames = 0; 3095 uint32_t activeSleepTime = activeSleepTimeUs(); 3096 uint32_t idleSleepTime = idleSleepTimeUs(); 3097 uint32_t sleepTime = idleSleepTime; 3098 Vector< sp<EffectChain> > effectChains; 3099 3100 acquireWakeLock(); 3101 3102 while (!exitPending()) 3103 { 3104 processConfigEvents(); 3105 3106 mixerStatus = MIXER_IDLE; 3107 { // scope for the mLock 3108 3109 Mutex::Autolock _l(mLock); 3110 3111 if (checkForNewParameters_l()) { 3112 mixBufferSize = mFrameCount*mFrameSize; 3113 updateWaitTime(); 3114 activeSleepTime = activeSleepTimeUs(); 3115 idleSleepTime = idleSleepTimeUs(); 3116 } 3117 3118 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3119 3120 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3121 outputTracks.add(mOutputTracks[i]); 3122 } 3123 3124 // put audio hardware into standby after short delay 3125 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3126 mSuspended)) { 3127 if (!mStandby) { 3128 for (size_t i = 0; i < outputTracks.size(); i++) { 3129 outputTracks[i]->stop(); 3130 } 3131 mStandby = true; 3132 mBytesWritten = 0; 3133 } 3134 3135 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3136 // we're about to wait, flush the binder command buffer 3137 IPCThreadState::self()->flushCommands(); 3138 outputTracks.clear(); 3139 3140 if (exitPending()) break; 3141 3142 releaseWakeLock_l(); 3143 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3144 mWaitWorkCV.wait(mLock); 3145 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3146 acquireWakeLock_l(); 3147 3148 checkSilentMode_l(); 3149 3150 standbyTime = systemTime() + mStandbyTimeInNsecs; 3151 sleepTime = idleSleepTime; 3152 continue; 3153 } 3154 } 3155 3156 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3157 3158 // prevent any changes in effect chain list and in each effect chain 3159 // during mixing and effect process as the audio buffers could be deleted 3160 // or modified if an effect is created or deleted 3161 lockEffectChains_l(effectChains); 3162 } 3163 3164 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3165 // mix buffers... 3166 if (outputsReady(outputTracks)) { 3167 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3168 } else { 3169 memset(mMixBuffer, 0, mixBufferSize); 3170 } 3171 sleepTime = 0; 3172 writeFrames = mFrameCount; 3173 } else { 3174 if (sleepTime == 0) { 3175 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3176 sleepTime = activeSleepTime; 3177 } else { 3178 sleepTime = idleSleepTime; 3179 } 3180 } else if (mBytesWritten != 0) { 3181 // flush remaining overflow buffers in output tracks 3182 for (size_t i = 0; i < outputTracks.size(); i++) { 3183 if (outputTracks[i]->isActive()) { 3184 sleepTime = 0; 3185 writeFrames = 0; 3186 memset(mMixBuffer, 0, mixBufferSize); 3187 break; 3188 } 3189 } 3190 } 3191 } 3192 3193 if (mSuspended) { 3194 sleepTime = suspendSleepTimeUs(); 3195 } 3196 // sleepTime == 0 means we must write to audio hardware 3197 if (sleepTime == 0) { 3198 for (size_t i = 0; i < effectChains.size(); i ++) { 3199 effectChains[i]->process_l(); 3200 } 3201 // enable changes in effect chain 3202 unlockEffectChains(effectChains); 3203 3204 standbyTime = systemTime() + mStandbyTimeInNsecs; 3205 for (size_t i = 0; i < outputTracks.size(); i++) { 3206 outputTracks[i]->write(mMixBuffer, writeFrames); 3207 } 3208 mStandby = false; 3209 mBytesWritten += mixBufferSize; 3210 } else { 3211 // enable changes in effect chain 3212 unlockEffectChains(effectChains); 3213 usleep(sleepTime); 3214 } 3215 3216 // finally let go of all our tracks, without the lock held 3217 // since we can't guarantee the destructors won't acquire that 3218 // same lock. 3219 tracksToRemove.clear(); 3220 outputTracks.clear(); 3221 3222 // Effect chains will be actually deleted here if they were removed from 3223 // mEffectChains list during mixing or effects processing 3224 effectChains.clear(); 3225 } 3226 3227 releaseWakeLock(); 3228 3229 return false; 3230} 3231 3232void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3233{ 3234 // FIXME explain this formula 3235 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3236 OutputTrack *outputTrack = new OutputTrack(thread, 3237 this, 3238 mSampleRate, 3239 mFormat, 3240 mChannelMask, 3241 frameCount); 3242 if (outputTrack->cblk() != NULL) { 3243 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3244 mOutputTracks.add(outputTrack); 3245 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3246 updateWaitTime(); 3247 } 3248} 3249 3250void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3251{ 3252 Mutex::Autolock _l(mLock); 3253 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3254 if (mOutputTracks[i]->thread() == thread) { 3255 mOutputTracks[i]->destroy(); 3256 mOutputTracks.removeAt(i); 3257 updateWaitTime(); 3258 return; 3259 } 3260 } 3261 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3262} 3263 3264void AudioFlinger::DuplicatingThread::updateWaitTime() 3265{ 3266 mWaitTimeMs = UINT_MAX; 3267 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3268 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3269 if (strong != 0) { 3270 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3271 if (waitTimeMs < mWaitTimeMs) { 3272 mWaitTimeMs = waitTimeMs; 3273 } 3274 } 3275 } 3276} 3277 3278 3279bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3280{ 3281 for (size_t i = 0; i < outputTracks.size(); i++) { 3282 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3283 if (thread == 0) { 3284 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3285 return false; 3286 } 3287 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3288 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3289 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3290 return false; 3291 } 3292 } 3293 return true; 3294} 3295 3296uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3297{ 3298 return (mWaitTimeMs * 1000) / 2; 3299} 3300 3301// ---------------------------------------------------------------------------- 3302 3303// TrackBase constructor must be called with AudioFlinger::mLock held 3304AudioFlinger::ThreadBase::TrackBase::TrackBase( 3305 ThreadBase *thread, 3306 const sp<Client>& client, 3307 uint32_t sampleRate, 3308 audio_format_t format, 3309 uint32_t channelMask, 3310 int frameCount, 3311 const sp<IMemory>& sharedBuffer, 3312 int sessionId) 3313 : RefBase(), 3314 mThread(thread), 3315 mClient(client), 3316 mCblk(NULL), 3317 // mBuffer 3318 // mBufferEnd 3319 mFrameCount(0), 3320 mState(IDLE), 3321 mFormat(format), 3322 mStepServerFailed(false), 3323 mSessionId(sessionId) 3324 // mChannelCount 3325 // mChannelMask 3326{ 3327 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3328 3329 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3330 size_t size = sizeof(audio_track_cblk_t); 3331 uint8_t channelCount = popcount(channelMask); 3332 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3333 if (sharedBuffer == 0) { 3334 size += bufferSize; 3335 } 3336 3337 if (client != NULL) { 3338 mCblkMemory = client->heap()->allocate(size); 3339 if (mCblkMemory != 0) { 3340 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3341 if (mCblk != NULL) { // construct the shared structure in-place. 3342 new(mCblk) audio_track_cblk_t(); 3343 // clear all buffers 3344 mCblk->frameCount = frameCount; 3345 mCblk->sampleRate = sampleRate; 3346 mChannelCount = channelCount; 3347 mChannelMask = channelMask; 3348 if (sharedBuffer == 0) { 3349 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3350 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3351 // Force underrun condition to avoid false underrun callback until first data is 3352 // written to buffer (other flags are cleared) 3353 mCblk->flags = CBLK_UNDERRUN_ON; 3354 } else { 3355 mBuffer = sharedBuffer->pointer(); 3356 } 3357 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3358 } 3359 } else { 3360 ALOGE("not enough memory for AudioTrack size=%u", size); 3361 client->heap()->dump("AudioTrack"); 3362 return; 3363 } 3364 } else { 3365 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3366 // construct the shared structure in-place. 3367 new(mCblk) audio_track_cblk_t(); 3368 // clear all buffers 3369 mCblk->frameCount = frameCount; 3370 mCblk->sampleRate = sampleRate; 3371 mChannelCount = channelCount; 3372 mChannelMask = channelMask; 3373 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3374 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3375 // Force underrun condition to avoid false underrun callback until first data is 3376 // written to buffer (other flags are cleared) 3377 mCblk->flags = CBLK_UNDERRUN_ON; 3378 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3379 } 3380} 3381 3382AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3383{ 3384 if (mCblk != NULL) { 3385 if (mClient == 0) { 3386 delete mCblk; 3387 } else { 3388 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3389 } 3390 } 3391 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3392 if (mClient != 0) { 3393 // Client destructor must run with AudioFlinger mutex locked 3394 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3395 // If the client's reference count drops to zero, the associated destructor 3396 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3397 // relying on the automatic clear() at end of scope. 3398 mClient.clear(); 3399 } 3400} 3401 3402void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3403{ 3404 buffer->raw = NULL; 3405 mFrameCount = buffer->frameCount; 3406 step(); 3407 buffer->frameCount = 0; 3408} 3409 3410bool AudioFlinger::ThreadBase::TrackBase::step() { 3411 bool result; 3412 audio_track_cblk_t* cblk = this->cblk(); 3413 3414 result = cblk->stepServer(mFrameCount); 3415 if (!result) { 3416 ALOGV("stepServer failed acquiring cblk mutex"); 3417 mStepServerFailed = true; 3418 } 3419 return result; 3420} 3421 3422void AudioFlinger::ThreadBase::TrackBase::reset() { 3423 audio_track_cblk_t* cblk = this->cblk(); 3424 3425 cblk->user = 0; 3426 cblk->server = 0; 3427 cblk->userBase = 0; 3428 cblk->serverBase = 0; 3429 mStepServerFailed = false; 3430 ALOGV("TrackBase::reset"); 3431} 3432 3433int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3434 return (int)mCblk->sampleRate; 3435} 3436 3437void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3438 audio_track_cblk_t* cblk = this->cblk(); 3439 size_t frameSize = cblk->frameSize; 3440 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3441 int8_t *bufferEnd = bufferStart + frames * frameSize; 3442 3443 // Check validity of returned pointer in case the track control block would have been corrupted. 3444 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3445 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3446 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3447 server %d, serverBase %d, user %d, userBase %d", 3448 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3449 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3450 return NULL; 3451 } 3452 3453 return bufferStart; 3454} 3455 3456// ---------------------------------------------------------------------------- 3457 3458// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3459AudioFlinger::PlaybackThread::Track::Track( 3460 PlaybackThread *thread, 3461 const sp<Client>& client, 3462 audio_stream_type_t streamType, 3463 uint32_t sampleRate, 3464 audio_format_t format, 3465 uint32_t channelMask, 3466 int frameCount, 3467 const sp<IMemory>& sharedBuffer, 3468 int sessionId) 3469 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3470 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3471 mAuxEffectId(0), mHasVolumeController(false) 3472{ 3473 if (mCblk != NULL) { 3474 if (thread != NULL) { 3475 mName = thread->getTrackName_l(); 3476 mMainBuffer = thread->mixBuffer(); 3477 } 3478 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3479 if (mName < 0) { 3480 ALOGE("no more track names available"); 3481 } 3482 mStreamType = streamType; 3483 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3484 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3485 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3486 } 3487} 3488 3489AudioFlinger::PlaybackThread::Track::~Track() 3490{ 3491 ALOGV("PlaybackThread::Track destructor"); 3492 sp<ThreadBase> thread = mThread.promote(); 3493 if (thread != 0) { 3494 Mutex::Autolock _l(thread->mLock); 3495 mState = TERMINATED; 3496 } 3497} 3498 3499void AudioFlinger::PlaybackThread::Track::destroy() 3500{ 3501 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3502 // by removing it from mTracks vector, so there is a risk that this Tracks's 3503 // destructor is called. As the destructor needs to lock mLock, 3504 // we must acquire a strong reference on this Track before locking mLock 3505 // here so that the destructor is called only when exiting this function. 3506 // On the other hand, as long as Track::destroy() is only called by 3507 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3508 // this Track with its member mTrack. 3509 sp<Track> keep(this); 3510 { // scope for mLock 3511 sp<ThreadBase> thread = mThread.promote(); 3512 if (thread != 0) { 3513 if (!isOutputTrack()) { 3514 if (mState == ACTIVE || mState == RESUMING) { 3515 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3516 3517 // to track the speaker usage 3518 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3519 } 3520 AudioSystem::releaseOutput(thread->id()); 3521 } 3522 Mutex::Autolock _l(thread->mLock); 3523 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3524 playbackThread->destroyTrack_l(this); 3525 } 3526 } 3527} 3528 3529void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3530{ 3531 uint32_t vlr = mCblk->getVolumeLR(); 3532 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3533 mName - AudioMixer::TRACK0, 3534 (mClient == 0) ? getpid_cached : mClient->pid(), 3535 mStreamType, 3536 mFormat, 3537 mChannelMask, 3538 mSessionId, 3539 mFrameCount, 3540 mState, 3541 mMute, 3542 mFillingUpStatus, 3543 mCblk->sampleRate, 3544 vlr & 0xFFFF, 3545 vlr >> 16, 3546 mCblk->server, 3547 mCblk->user, 3548 (int)mMainBuffer, 3549 (int)mAuxBuffer); 3550} 3551 3552status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3553 AudioBufferProvider::Buffer* buffer, int64_t pts) 3554{ 3555 audio_track_cblk_t* cblk = this->cblk(); 3556 uint32_t framesReady; 3557 uint32_t framesReq = buffer->frameCount; 3558 3559 // Check if last stepServer failed, try to step now 3560 if (mStepServerFailed) { 3561 if (!step()) goto getNextBuffer_exit; 3562 ALOGV("stepServer recovered"); 3563 mStepServerFailed = false; 3564 } 3565 3566 framesReady = cblk->framesReady(); 3567 3568 if (CC_LIKELY(framesReady)) { 3569 uint32_t s = cblk->server; 3570 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3571 3572 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3573 if (framesReq > framesReady) { 3574 framesReq = framesReady; 3575 } 3576 if (s + framesReq > bufferEnd) { 3577 framesReq = bufferEnd - s; 3578 } 3579 3580 buffer->raw = getBuffer(s, framesReq); 3581 if (buffer->raw == NULL) goto getNextBuffer_exit; 3582 3583 buffer->frameCount = framesReq; 3584 return NO_ERROR; 3585 } 3586 3587getNextBuffer_exit: 3588 buffer->raw = NULL; 3589 buffer->frameCount = 0; 3590 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3591 return NOT_ENOUGH_DATA; 3592} 3593 3594uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3595 return mCblk->framesReady(); 3596} 3597 3598bool AudioFlinger::PlaybackThread::Track::isReady() const { 3599 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3600 3601 if (framesReady() >= mCblk->frameCount || 3602 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3603 mFillingUpStatus = FS_FILLED; 3604 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3605 return true; 3606 } 3607 return false; 3608} 3609 3610status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3611{ 3612 status_t status = NO_ERROR; 3613 ALOGV("start(%d), calling pid %d session %d tid %d", 3614 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3615 sp<ThreadBase> thread = mThread.promote(); 3616 if (thread != 0) { 3617 Mutex::Autolock _l(thread->mLock); 3618 track_state state = mState; 3619 // here the track could be either new, or restarted 3620 // in both cases "unstop" the track 3621 if (mState == PAUSED) { 3622 mState = TrackBase::RESUMING; 3623 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3624 } else { 3625 mState = TrackBase::ACTIVE; 3626 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3627 } 3628 3629 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3630 thread->mLock.unlock(); 3631 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3632 thread->mLock.lock(); 3633 3634 // to track the speaker usage 3635 if (status == NO_ERROR) { 3636 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3637 } 3638 } 3639 if (status == NO_ERROR) { 3640 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3641 playbackThread->addTrack_l(this); 3642 } else { 3643 mState = state; 3644 } 3645 } else { 3646 status = BAD_VALUE; 3647 } 3648 return status; 3649} 3650 3651void AudioFlinger::PlaybackThread::Track::stop() 3652{ 3653 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3654 sp<ThreadBase> thread = mThread.promote(); 3655 if (thread != 0) { 3656 Mutex::Autolock _l(thread->mLock); 3657 track_state state = mState; 3658 if (mState > STOPPED) { 3659 mState = STOPPED; 3660 // If the track is not active (PAUSED and buffers full), flush buffers 3661 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3662 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3663 reset(); 3664 } 3665 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3666 } 3667 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3668 thread->mLock.unlock(); 3669 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3670 thread->mLock.lock(); 3671 3672 // to track the speaker usage 3673 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3674 } 3675 } 3676} 3677 3678void AudioFlinger::PlaybackThread::Track::pause() 3679{ 3680 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3681 sp<ThreadBase> thread = mThread.promote(); 3682 if (thread != 0) { 3683 Mutex::Autolock _l(thread->mLock); 3684 if (mState == ACTIVE || mState == RESUMING) { 3685 mState = PAUSING; 3686 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3687 if (!isOutputTrack()) { 3688 thread->mLock.unlock(); 3689 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3690 thread->mLock.lock(); 3691 3692 // to track the speaker usage 3693 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3694 } 3695 } 3696 } 3697} 3698 3699void AudioFlinger::PlaybackThread::Track::flush() 3700{ 3701 ALOGV("flush(%d)", mName); 3702 sp<ThreadBase> thread = mThread.promote(); 3703 if (thread != 0) { 3704 Mutex::Autolock _l(thread->mLock); 3705 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3706 return; 3707 } 3708 // No point remaining in PAUSED state after a flush => go to 3709 // STOPPED state 3710 mState = STOPPED; 3711 3712 // do not reset the track if it is still in the process of being stopped or paused. 3713 // this will be done by prepareTracks_l() when the track is stopped. 3714 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3715 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3716 reset(); 3717 } 3718 } 3719} 3720 3721void AudioFlinger::PlaybackThread::Track::reset() 3722{ 3723 // Do not reset twice to avoid discarding data written just after a flush and before 3724 // the audioflinger thread detects the track is stopped. 3725 if (!mResetDone) { 3726 TrackBase::reset(); 3727 // Force underrun condition to avoid false underrun callback until first data is 3728 // written to buffer 3729 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3730 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3731 mFillingUpStatus = FS_FILLING; 3732 mResetDone = true; 3733 } 3734} 3735 3736void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3737{ 3738 mMute = muted; 3739} 3740 3741status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3742{ 3743 status_t status = DEAD_OBJECT; 3744 sp<ThreadBase> thread = mThread.promote(); 3745 if (thread != 0) { 3746 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3747 status = playbackThread->attachAuxEffect(this, EffectId); 3748 } 3749 return status; 3750} 3751 3752void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3753{ 3754 mAuxEffectId = EffectId; 3755 mAuxBuffer = buffer; 3756} 3757 3758// timed audio tracks 3759 3760sp<AudioFlinger::PlaybackThread::TimedTrack> 3761AudioFlinger::PlaybackThread::TimedTrack::create( 3762 PlaybackThread *thread, 3763 const sp<Client>& client, 3764 audio_stream_type_t streamType, 3765 uint32_t sampleRate, 3766 audio_format_t format, 3767 uint32_t channelMask, 3768 int frameCount, 3769 const sp<IMemory>& sharedBuffer, 3770 int sessionId) { 3771 if (!client->reserveTimedTrack()) 3772 return NULL; 3773 3774 sp<TimedTrack> track = new TimedTrack( 3775 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3776 sharedBuffer, sessionId); 3777 3778 if (track == NULL) { 3779 client->releaseTimedTrack(); 3780 return NULL; 3781 } 3782 3783 return track; 3784} 3785 3786AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3787 PlaybackThread *thread, 3788 const sp<Client>& client, 3789 audio_stream_type_t streamType, 3790 uint32_t sampleRate, 3791 audio_format_t format, 3792 uint32_t channelMask, 3793 int frameCount, 3794 const sp<IMemory>& sharedBuffer, 3795 int sessionId) 3796 : Track(thread, client, streamType, sampleRate, format, channelMask, 3797 frameCount, sharedBuffer, sessionId), 3798 mTimedSilenceBuffer(NULL), 3799 mTimedSilenceBufferSize(0), 3800 mTimedAudioOutputOnTime(false), 3801 mMediaTimeTransformValid(false) 3802{ 3803 LocalClock lc; 3804 mLocalTimeFreq = lc.getLocalFreq(); 3805 3806 mLocalTimeToSampleTransform.a_zero = 0; 3807 mLocalTimeToSampleTransform.b_zero = 0; 3808 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3809 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3810 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3811 &mLocalTimeToSampleTransform.a_to_b_denom); 3812} 3813 3814AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3815 mClient->releaseTimedTrack(); 3816 delete [] mTimedSilenceBuffer; 3817} 3818 3819status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3820 size_t size, sp<IMemory>* buffer) { 3821 3822 Mutex::Autolock _l(mTimedBufferQueueLock); 3823 3824 trimTimedBufferQueue_l(); 3825 3826 // lazily initialize the shared memory heap for timed buffers 3827 if (mTimedMemoryDealer == NULL) { 3828 const int kTimedBufferHeapSize = 512 << 10; 3829 3830 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3831 "AudioFlingerTimed"); 3832 if (mTimedMemoryDealer == NULL) 3833 return NO_MEMORY; 3834 } 3835 3836 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3837 if (newBuffer == NULL) { 3838 newBuffer = mTimedMemoryDealer->allocate(size); 3839 if (newBuffer == NULL) 3840 return NO_MEMORY; 3841 } 3842 3843 *buffer = newBuffer; 3844 return NO_ERROR; 3845} 3846 3847// caller must hold mTimedBufferQueueLock 3848void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3849 int64_t mediaTimeNow; 3850 { 3851 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3852 if (!mMediaTimeTransformValid) 3853 return; 3854 3855 int64_t targetTimeNow; 3856 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3857 ? mCCHelper.getCommonTime(&targetTimeNow) 3858 : mCCHelper.getLocalTime(&targetTimeNow); 3859 3860 if (OK != res) 3861 return; 3862 3863 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3864 &mediaTimeNow)) { 3865 return; 3866 } 3867 } 3868 3869 size_t trimIndex; 3870 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3871 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3872 break; 3873 } 3874 3875 if (trimIndex) { 3876 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3877 } 3878} 3879 3880status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3881 const sp<IMemory>& buffer, int64_t pts) { 3882 3883 { 3884 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3885 if (!mMediaTimeTransformValid) 3886 return INVALID_OPERATION; 3887 } 3888 3889 Mutex::Autolock _l(mTimedBufferQueueLock); 3890 3891 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3892 3893 return NO_ERROR; 3894} 3895 3896status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3897 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3898 3899 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3900 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3901 target); 3902 3903 if (!(target == TimedAudioTrack::LOCAL_TIME || 3904 target == TimedAudioTrack::COMMON_TIME)) { 3905 return BAD_VALUE; 3906 } 3907 3908 Mutex::Autolock lock(mMediaTimeTransformLock); 3909 mMediaTimeTransform = xform; 3910 mMediaTimeTransformTarget = target; 3911 mMediaTimeTransformValid = true; 3912 3913 return NO_ERROR; 3914} 3915 3916#define min(a, b) ((a) < (b) ? (a) : (b)) 3917 3918// implementation of getNextBuffer for tracks whose buffers have timestamps 3919status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3920 AudioBufferProvider::Buffer* buffer, int64_t pts) 3921{ 3922 if (pts == AudioBufferProvider::kInvalidPTS) { 3923 buffer->raw = 0; 3924 buffer->frameCount = 0; 3925 return INVALID_OPERATION; 3926 } 3927 3928 Mutex::Autolock _l(mTimedBufferQueueLock); 3929 3930 while (true) { 3931 3932 // if we have no timed buffers, then fail 3933 if (mTimedBufferQueue.isEmpty()) { 3934 buffer->raw = 0; 3935 buffer->frameCount = 0; 3936 return NOT_ENOUGH_DATA; 3937 } 3938 3939 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3940 3941 // calculate the PTS of the head of the timed buffer queue expressed in 3942 // local time 3943 int64_t headLocalPTS; 3944 { 3945 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3946 3947 assert(mMediaTimeTransformValid); 3948 3949 if (mMediaTimeTransform.a_to_b_denom == 0) { 3950 // the transform represents a pause, so yield silence 3951 timedYieldSilence(buffer->frameCount, buffer); 3952 return NO_ERROR; 3953 } 3954 3955 int64_t transformedPTS; 3956 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3957 &transformedPTS)) { 3958 // the transform failed. this shouldn't happen, but if it does 3959 // then just drop this buffer 3960 ALOGW("timedGetNextBuffer transform failed"); 3961 buffer->raw = 0; 3962 buffer->frameCount = 0; 3963 mTimedBufferQueue.removeAt(0); 3964 return NO_ERROR; 3965 } 3966 3967 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3968 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3969 &headLocalPTS)) { 3970 buffer->raw = 0; 3971 buffer->frameCount = 0; 3972 return INVALID_OPERATION; 3973 } 3974 } else { 3975 headLocalPTS = transformedPTS; 3976 } 3977 } 3978 3979 // adjust the head buffer's PTS to reflect the portion of the head buffer 3980 // that has already been consumed 3981 int64_t effectivePTS = headLocalPTS + 3982 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3983 3984 // Calculate the delta in samples between the head of the input buffer 3985 // queue and the start of the next output buffer that will be written. 3986 // If the transformation fails because of over or underflow, it means 3987 // that the sample's position in the output stream is so far out of 3988 // whack that it should just be dropped. 3989 int64_t sampleDelta; 3990 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3991 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3992 mTimedBufferQueue.removeAt(0); 3993 continue; 3994 } 3995 if (!mLocalTimeToSampleTransform.doForwardTransform( 3996 (effectivePTS - pts) << 32, &sampleDelta)) { 3997 ALOGV("*** too late during sample rate transform: dropped buffer"); 3998 mTimedBufferQueue.removeAt(0); 3999 continue; 4000 } 4001 4002 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4003 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4004 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4005 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4006 4007 // if the delta between the ideal placement for the next input sample and 4008 // the current output position is within this threshold, then we will 4009 // concatenate the next input samples to the previous output 4010 const int64_t kSampleContinuityThreshold = 4011 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4012 4013 // if this is the first buffer of audio that we're emitting from this track 4014 // then it should be almost exactly on time. 4015 const int64_t kSampleStartupThreshold = 1LL << 32; 4016 4017 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4018 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4019 // the next input is close enough to being on time, so concatenate it 4020 // with the last output 4021 timedYieldSamples(buffer); 4022 4023 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4024 return NO_ERROR; 4025 } else if (sampleDelta > 0) { 4026 // the gap between the current output position and the proper start of 4027 // the next input sample is too big, so fill it with silence 4028 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4029 4030 timedYieldSilence(framesUntilNextInput, buffer); 4031 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4032 return NO_ERROR; 4033 } else { 4034 // the next input sample is late 4035 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4036 size_t onTimeSamplePosition = 4037 head.position() + lateFrames * mCblk->frameSize; 4038 4039 if (onTimeSamplePosition > head.buffer()->size()) { 4040 // all the remaining samples in the head are too late, so 4041 // drop it and move on 4042 ALOGV("*** too late: dropped buffer"); 4043 mTimedBufferQueue.removeAt(0); 4044 continue; 4045 } else { 4046 // skip over the late samples 4047 head.setPosition(onTimeSamplePosition); 4048 4049 // yield the available samples 4050 timedYieldSamples(buffer); 4051 4052 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4053 return NO_ERROR; 4054 } 4055 } 4056 } 4057} 4058 4059// Yield samples from the timed buffer queue head up to the given output 4060// buffer's capacity. 4061// 4062// Caller must hold mTimedBufferQueueLock 4063void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4064 AudioBufferProvider::Buffer* buffer) { 4065 4066 const TimedBuffer& head = mTimedBufferQueue[0]; 4067 4068 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4069 head.position()); 4070 4071 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4072 mCblk->frameSize); 4073 size_t framesRequested = buffer->frameCount; 4074 buffer->frameCount = min(framesLeftInHead, framesRequested); 4075 4076 mTimedAudioOutputOnTime = true; 4077} 4078 4079// Yield samples of silence up to the given output buffer's capacity 4080// 4081// Caller must hold mTimedBufferQueueLock 4082void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4083 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4084 4085 // lazily allocate a buffer filled with silence 4086 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4087 delete [] mTimedSilenceBuffer; 4088 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4089 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4090 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4091 } 4092 4093 buffer->raw = mTimedSilenceBuffer; 4094 size_t framesRequested = buffer->frameCount; 4095 buffer->frameCount = min(numFrames, framesRequested); 4096 4097 mTimedAudioOutputOnTime = false; 4098} 4099 4100void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4101 AudioBufferProvider::Buffer* buffer) { 4102 4103 Mutex::Autolock _l(mTimedBufferQueueLock); 4104 4105 // If the buffer which was just released is part of the buffer at the head 4106 // of the queue, be sure to update the amt of the buffer which has been 4107 // consumed. If the buffer being returned is not part of the head of the 4108 // queue, its either because the buffer is part of the silence buffer, or 4109 // because the head of the timed queue was trimmed after the mixer called 4110 // getNextBuffer but before the mixer called releaseBuffer. 4111 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4112 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4113 4114 void* start = head.buffer()->pointer(); 4115 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4116 4117 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4118 head.setPosition(head.position() + 4119 (buffer->frameCount * mCblk->frameSize)); 4120 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4121 mTimedBufferQueue.removeAt(0); 4122 } 4123 } 4124 } 4125 4126 buffer->raw = 0; 4127 buffer->frameCount = 0; 4128} 4129 4130uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4131 Mutex::Autolock _l(mTimedBufferQueueLock); 4132 4133 uint32_t frames = 0; 4134 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4135 const TimedBuffer& tb = mTimedBufferQueue[i]; 4136 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4137 } 4138 4139 return frames; 4140} 4141 4142AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4143 : mPTS(0), mPosition(0) {} 4144 4145AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4146 const sp<IMemory>& buffer, int64_t pts) 4147 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4148 4149// ---------------------------------------------------------------------------- 4150 4151// RecordTrack constructor must be called with AudioFlinger::mLock held 4152AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4153 RecordThread *thread, 4154 const sp<Client>& client, 4155 uint32_t sampleRate, 4156 audio_format_t format, 4157 uint32_t channelMask, 4158 int frameCount, 4159 int sessionId) 4160 : TrackBase(thread, client, sampleRate, format, 4161 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4162 mOverflow(false) 4163{ 4164 if (mCblk != NULL) { 4165 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4166 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4167 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4168 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4169 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4170 } else { 4171 mCblk->frameSize = sizeof(int8_t); 4172 } 4173 } 4174} 4175 4176AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4177{ 4178 sp<ThreadBase> thread = mThread.promote(); 4179 if (thread != 0) { 4180 AudioSystem::releaseInput(thread->id()); 4181 } 4182} 4183 4184status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4185{ 4186 audio_track_cblk_t* cblk = this->cblk(); 4187 uint32_t framesAvail; 4188 uint32_t framesReq = buffer->frameCount; 4189 4190 // Check if last stepServer failed, try to step now 4191 if (mStepServerFailed) { 4192 if (!step()) goto getNextBuffer_exit; 4193 ALOGV("stepServer recovered"); 4194 mStepServerFailed = false; 4195 } 4196 4197 framesAvail = cblk->framesAvailable_l(); 4198 4199 if (CC_LIKELY(framesAvail)) { 4200 uint32_t s = cblk->server; 4201 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4202 4203 if (framesReq > framesAvail) { 4204 framesReq = framesAvail; 4205 } 4206 if (s + framesReq > bufferEnd) { 4207 framesReq = bufferEnd - s; 4208 } 4209 4210 buffer->raw = getBuffer(s, framesReq); 4211 if (buffer->raw == NULL) goto getNextBuffer_exit; 4212 4213 buffer->frameCount = framesReq; 4214 return NO_ERROR; 4215 } 4216 4217getNextBuffer_exit: 4218 buffer->raw = NULL; 4219 buffer->frameCount = 0; 4220 return NOT_ENOUGH_DATA; 4221} 4222 4223status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4224{ 4225 sp<ThreadBase> thread = mThread.promote(); 4226 if (thread != 0) { 4227 RecordThread *recordThread = (RecordThread *)thread.get(); 4228 return recordThread->start(this, tid); 4229 } else { 4230 return BAD_VALUE; 4231 } 4232} 4233 4234void AudioFlinger::RecordThread::RecordTrack::stop() 4235{ 4236 sp<ThreadBase> thread = mThread.promote(); 4237 if (thread != 0) { 4238 RecordThread *recordThread = (RecordThread *)thread.get(); 4239 recordThread->stop(this); 4240 TrackBase::reset(); 4241 // Force overerrun condition to avoid false overrun callback until first data is 4242 // read from buffer 4243 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4244 } 4245} 4246 4247void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4248{ 4249 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4250 (mClient == 0) ? getpid_cached : mClient->pid(), 4251 mFormat, 4252 mChannelMask, 4253 mSessionId, 4254 mFrameCount, 4255 mState, 4256 mCblk->sampleRate, 4257 mCblk->server, 4258 mCblk->user); 4259} 4260 4261 4262// ---------------------------------------------------------------------------- 4263 4264AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4265 PlaybackThread *playbackThread, 4266 DuplicatingThread *sourceThread, 4267 uint32_t sampleRate, 4268 audio_format_t format, 4269 uint32_t channelMask, 4270 int frameCount) 4271 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4272 mActive(false), mSourceThread(sourceThread) 4273{ 4274 4275 if (mCblk != NULL) { 4276 mCblk->flags |= CBLK_DIRECTION_OUT; 4277 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4278 mOutBuffer.frameCount = 0; 4279 playbackThread->mTracks.add(this); 4280 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4281 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4282 mCblk, mBuffer, mCblk->buffers, 4283 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4284 } else { 4285 ALOGW("Error creating output track on thread %p", playbackThread); 4286 } 4287} 4288 4289AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4290{ 4291 clearBufferQueue(); 4292} 4293 4294status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4295{ 4296 status_t status = Track::start(tid); 4297 if (status != NO_ERROR) { 4298 return status; 4299 } 4300 4301 mActive = true; 4302 mRetryCount = 127; 4303 return status; 4304} 4305 4306void AudioFlinger::PlaybackThread::OutputTrack::stop() 4307{ 4308 Track::stop(); 4309 clearBufferQueue(); 4310 mOutBuffer.frameCount = 0; 4311 mActive = false; 4312} 4313 4314bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4315{ 4316 Buffer *pInBuffer; 4317 Buffer inBuffer; 4318 uint32_t channelCount = mChannelCount; 4319 bool outputBufferFull = false; 4320 inBuffer.frameCount = frames; 4321 inBuffer.i16 = data; 4322 4323 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4324 4325 if (!mActive && frames != 0) { 4326 start(0); 4327 sp<ThreadBase> thread = mThread.promote(); 4328 if (thread != 0) { 4329 MixerThread *mixerThread = (MixerThread *)thread.get(); 4330 if (mCblk->frameCount > frames){ 4331 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4332 uint32_t startFrames = (mCblk->frameCount - frames); 4333 pInBuffer = new Buffer; 4334 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4335 pInBuffer->frameCount = startFrames; 4336 pInBuffer->i16 = pInBuffer->mBuffer; 4337 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4338 mBufferQueue.add(pInBuffer); 4339 } else { 4340 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4341 } 4342 } 4343 } 4344 } 4345 4346 while (waitTimeLeftMs) { 4347 // First write pending buffers, then new data 4348 if (mBufferQueue.size()) { 4349 pInBuffer = mBufferQueue.itemAt(0); 4350 } else { 4351 pInBuffer = &inBuffer; 4352 } 4353 4354 if (pInBuffer->frameCount == 0) { 4355 break; 4356 } 4357 4358 if (mOutBuffer.frameCount == 0) { 4359 mOutBuffer.frameCount = pInBuffer->frameCount; 4360 nsecs_t startTime = systemTime(); 4361 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4362 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4363 outputBufferFull = true; 4364 break; 4365 } 4366 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4367 if (waitTimeLeftMs >= waitTimeMs) { 4368 waitTimeLeftMs -= waitTimeMs; 4369 } else { 4370 waitTimeLeftMs = 0; 4371 } 4372 } 4373 4374 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4375 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4376 mCblk->stepUser(outFrames); 4377 pInBuffer->frameCount -= outFrames; 4378 pInBuffer->i16 += outFrames * channelCount; 4379 mOutBuffer.frameCount -= outFrames; 4380 mOutBuffer.i16 += outFrames * channelCount; 4381 4382 if (pInBuffer->frameCount == 0) { 4383 if (mBufferQueue.size()) { 4384 mBufferQueue.removeAt(0); 4385 delete [] pInBuffer->mBuffer; 4386 delete pInBuffer; 4387 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4388 } else { 4389 break; 4390 } 4391 } 4392 } 4393 4394 // If we could not write all frames, allocate a buffer and queue it for next time. 4395 if (inBuffer.frameCount) { 4396 sp<ThreadBase> thread = mThread.promote(); 4397 if (thread != 0 && !thread->standby()) { 4398 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4399 pInBuffer = new Buffer; 4400 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4401 pInBuffer->frameCount = inBuffer.frameCount; 4402 pInBuffer->i16 = pInBuffer->mBuffer; 4403 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4404 mBufferQueue.add(pInBuffer); 4405 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4406 } else { 4407 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4408 } 4409 } 4410 } 4411 4412 // Calling write() with a 0 length buffer, means that no more data will be written: 4413 // If no more buffers are pending, fill output track buffer to make sure it is started 4414 // by output mixer. 4415 if (frames == 0 && mBufferQueue.size() == 0) { 4416 if (mCblk->user < mCblk->frameCount) { 4417 frames = mCblk->frameCount - mCblk->user; 4418 pInBuffer = new Buffer; 4419 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4420 pInBuffer->frameCount = frames; 4421 pInBuffer->i16 = pInBuffer->mBuffer; 4422 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4423 mBufferQueue.add(pInBuffer); 4424 } else if (mActive) { 4425 stop(); 4426 } 4427 } 4428 4429 return outputBufferFull; 4430} 4431 4432status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4433{ 4434 int active; 4435 status_t result; 4436 audio_track_cblk_t* cblk = mCblk; 4437 uint32_t framesReq = buffer->frameCount; 4438 4439// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4440 buffer->frameCount = 0; 4441 4442 uint32_t framesAvail = cblk->framesAvailable(); 4443 4444 4445 if (framesAvail == 0) { 4446 Mutex::Autolock _l(cblk->lock); 4447 goto start_loop_here; 4448 while (framesAvail == 0) { 4449 active = mActive; 4450 if (CC_UNLIKELY(!active)) { 4451 ALOGV("Not active and NO_MORE_BUFFERS"); 4452 return NO_MORE_BUFFERS; 4453 } 4454 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4455 if (result != NO_ERROR) { 4456 return NO_MORE_BUFFERS; 4457 } 4458 // read the server count again 4459 start_loop_here: 4460 framesAvail = cblk->framesAvailable_l(); 4461 } 4462 } 4463 4464// if (framesAvail < framesReq) { 4465// return NO_MORE_BUFFERS; 4466// } 4467 4468 if (framesReq > framesAvail) { 4469 framesReq = framesAvail; 4470 } 4471 4472 uint32_t u = cblk->user; 4473 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4474 4475 if (u + framesReq > bufferEnd) { 4476 framesReq = bufferEnd - u; 4477 } 4478 4479 buffer->frameCount = framesReq; 4480 buffer->raw = (void *)cblk->buffer(u); 4481 return NO_ERROR; 4482} 4483 4484 4485void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4486{ 4487 size_t size = mBufferQueue.size(); 4488 4489 for (size_t i = 0; i < size; i++) { 4490 Buffer *pBuffer = mBufferQueue.itemAt(i); 4491 delete [] pBuffer->mBuffer; 4492 delete pBuffer; 4493 } 4494 mBufferQueue.clear(); 4495} 4496 4497// ---------------------------------------------------------------------------- 4498 4499AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4500 : RefBase(), 4501 mAudioFlinger(audioFlinger), 4502 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4503 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4504 mPid(pid), 4505 mTimedTrackCount(0) 4506{ 4507 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4508} 4509 4510// Client destructor must be called with AudioFlinger::mLock held 4511AudioFlinger::Client::~Client() 4512{ 4513 mAudioFlinger->removeClient_l(mPid); 4514} 4515 4516sp<MemoryDealer> AudioFlinger::Client::heap() const 4517{ 4518 return mMemoryDealer; 4519} 4520 4521// Reserve one of the limited slots for a timed audio track associated 4522// with this client 4523bool AudioFlinger::Client::reserveTimedTrack() 4524{ 4525 const int kMaxTimedTracksPerClient = 4; 4526 4527 Mutex::Autolock _l(mTimedTrackLock); 4528 4529 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4530 ALOGW("can not create timed track - pid %d has exceeded the limit", 4531 mPid); 4532 return false; 4533 } 4534 4535 mTimedTrackCount++; 4536 return true; 4537} 4538 4539// Release a slot for a timed audio track 4540void AudioFlinger::Client::releaseTimedTrack() 4541{ 4542 Mutex::Autolock _l(mTimedTrackLock); 4543 mTimedTrackCount--; 4544} 4545 4546// ---------------------------------------------------------------------------- 4547 4548AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4549 const sp<IAudioFlingerClient>& client, 4550 pid_t pid) 4551 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4552{ 4553} 4554 4555AudioFlinger::NotificationClient::~NotificationClient() 4556{ 4557} 4558 4559void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4560{ 4561 sp<NotificationClient> keep(this); 4562 mAudioFlinger->removeNotificationClient(mPid); 4563} 4564 4565// ---------------------------------------------------------------------------- 4566 4567AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4568 : BnAudioTrack(), 4569 mTrack(track) 4570{ 4571} 4572 4573AudioFlinger::TrackHandle::~TrackHandle() { 4574 // just stop the track on deletion, associated resources 4575 // will be freed from the main thread once all pending buffers have 4576 // been played. Unless it's not in the active track list, in which 4577 // case we free everything now... 4578 mTrack->destroy(); 4579} 4580 4581sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4582 return mTrack->getCblk(); 4583} 4584 4585status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4586 return mTrack->start(tid); 4587} 4588 4589void AudioFlinger::TrackHandle::stop() { 4590 mTrack->stop(); 4591} 4592 4593void AudioFlinger::TrackHandle::flush() { 4594 mTrack->flush(); 4595} 4596 4597void AudioFlinger::TrackHandle::mute(bool e) { 4598 mTrack->mute(e); 4599} 4600 4601void AudioFlinger::TrackHandle::pause() { 4602 mTrack->pause(); 4603} 4604 4605status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4606{ 4607 return mTrack->attachAuxEffect(EffectId); 4608} 4609 4610status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4611 sp<IMemory>* buffer) { 4612 if (!mTrack->isTimedTrack()) 4613 return INVALID_OPERATION; 4614 4615 PlaybackThread::TimedTrack* tt = 4616 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4617 return tt->allocateTimedBuffer(size, buffer); 4618} 4619 4620status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4621 int64_t pts) { 4622 if (!mTrack->isTimedTrack()) 4623 return INVALID_OPERATION; 4624 4625 PlaybackThread::TimedTrack* tt = 4626 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4627 return tt->queueTimedBuffer(buffer, pts); 4628} 4629 4630status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4631 const LinearTransform& xform, int target) { 4632 4633 if (!mTrack->isTimedTrack()) 4634 return INVALID_OPERATION; 4635 4636 PlaybackThread::TimedTrack* tt = 4637 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4638 return tt->setMediaTimeTransform( 4639 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4640} 4641 4642status_t AudioFlinger::TrackHandle::onTransact( 4643 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4644{ 4645 return BnAudioTrack::onTransact(code, data, reply, flags); 4646} 4647 4648// ---------------------------------------------------------------------------- 4649 4650sp<IAudioRecord> AudioFlinger::openRecord( 4651 pid_t pid, 4652 audio_io_handle_t input, 4653 uint32_t sampleRate, 4654 audio_format_t format, 4655 uint32_t channelMask, 4656 int frameCount, 4657 // FIXME dead, remove from IAudioFlinger 4658 uint32_t flags, 4659 int *sessionId, 4660 status_t *status) 4661{ 4662 sp<RecordThread::RecordTrack> recordTrack; 4663 sp<RecordHandle> recordHandle; 4664 sp<Client> client; 4665 status_t lStatus; 4666 RecordThread *thread; 4667 size_t inFrameCount; 4668 int lSessionId; 4669 4670 // check calling permissions 4671 if (!recordingAllowed()) { 4672 lStatus = PERMISSION_DENIED; 4673 goto Exit; 4674 } 4675 4676 // add client to list 4677 { // scope for mLock 4678 Mutex::Autolock _l(mLock); 4679 thread = checkRecordThread_l(input); 4680 if (thread == NULL) { 4681 lStatus = BAD_VALUE; 4682 goto Exit; 4683 } 4684 4685 client = registerPid_l(pid); 4686 4687 // If no audio session id is provided, create one here 4688 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4689 lSessionId = *sessionId; 4690 } else { 4691 lSessionId = nextUniqueId(); 4692 if (sessionId != NULL) { 4693 *sessionId = lSessionId; 4694 } 4695 } 4696 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4697 recordTrack = thread->createRecordTrack_l(client, 4698 sampleRate, 4699 format, 4700 channelMask, 4701 frameCount, 4702 lSessionId, 4703 &lStatus); 4704 } 4705 if (lStatus != NO_ERROR) { 4706 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4707 // destructor is called by the TrackBase destructor with mLock held 4708 client.clear(); 4709 recordTrack.clear(); 4710 goto Exit; 4711 } 4712 4713 // return to handle to client 4714 recordHandle = new RecordHandle(recordTrack); 4715 lStatus = NO_ERROR; 4716 4717Exit: 4718 if (status) { 4719 *status = lStatus; 4720 } 4721 return recordHandle; 4722} 4723 4724// ---------------------------------------------------------------------------- 4725 4726AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4727 : BnAudioRecord(), 4728 mRecordTrack(recordTrack) 4729{ 4730} 4731 4732AudioFlinger::RecordHandle::~RecordHandle() { 4733 stop(); 4734} 4735 4736sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4737 return mRecordTrack->getCblk(); 4738} 4739 4740status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4741 ALOGV("RecordHandle::start()"); 4742 return mRecordTrack->start(tid); 4743} 4744 4745void AudioFlinger::RecordHandle::stop() { 4746 ALOGV("RecordHandle::stop()"); 4747 mRecordTrack->stop(); 4748} 4749 4750status_t AudioFlinger::RecordHandle::onTransact( 4751 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4752{ 4753 return BnAudioRecord::onTransact(code, data, reply, flags); 4754} 4755 4756// ---------------------------------------------------------------------------- 4757 4758AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4759 AudioStreamIn *input, 4760 uint32_t sampleRate, 4761 uint32_t channels, 4762 audio_io_handle_t id, 4763 uint32_t device) : 4764 ThreadBase(audioFlinger, id, device, RECORD), 4765 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4766 // mRsmpInIndex and mInputBytes set by readInputParameters() 4767 mReqChannelCount(popcount(channels)), 4768 mReqSampleRate(sampleRate) 4769 // mBytesRead is only meaningful while active, and so is cleared in start() 4770 // (but might be better to also clear here for dump?) 4771{ 4772 snprintf(mName, kNameLength, "AudioIn_%d", id); 4773 4774 readInputParameters(); 4775} 4776 4777 4778AudioFlinger::RecordThread::~RecordThread() 4779{ 4780 delete[] mRsmpInBuffer; 4781 delete mResampler; 4782 delete[] mRsmpOutBuffer; 4783} 4784 4785void AudioFlinger::RecordThread::onFirstRef() 4786{ 4787 run(mName, PRIORITY_URGENT_AUDIO); 4788} 4789 4790status_t AudioFlinger::RecordThread::readyToRun() 4791{ 4792 status_t status = initCheck(); 4793 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4794 return status; 4795} 4796 4797bool AudioFlinger::RecordThread::threadLoop() 4798{ 4799 AudioBufferProvider::Buffer buffer; 4800 sp<RecordTrack> activeTrack; 4801 Vector< sp<EffectChain> > effectChains; 4802 4803 nsecs_t lastWarning = 0; 4804 4805 acquireWakeLock(); 4806 4807 // start recording 4808 while (!exitPending()) { 4809 4810 processConfigEvents(); 4811 4812 { // scope for mLock 4813 Mutex::Autolock _l(mLock); 4814 checkForNewParameters_l(); 4815 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4816 if (!mStandby) { 4817 mInput->stream->common.standby(&mInput->stream->common); 4818 mStandby = true; 4819 } 4820 4821 if (exitPending()) break; 4822 4823 releaseWakeLock_l(); 4824 ALOGV("RecordThread: loop stopping"); 4825 // go to sleep 4826 mWaitWorkCV.wait(mLock); 4827 ALOGV("RecordThread: loop starting"); 4828 acquireWakeLock_l(); 4829 continue; 4830 } 4831 if (mActiveTrack != 0) { 4832 if (mActiveTrack->mState == TrackBase::PAUSING) { 4833 if (!mStandby) { 4834 mInput->stream->common.standby(&mInput->stream->common); 4835 mStandby = true; 4836 } 4837 mActiveTrack.clear(); 4838 mStartStopCond.broadcast(); 4839 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4840 if (mReqChannelCount != mActiveTrack->channelCount()) { 4841 mActiveTrack.clear(); 4842 mStartStopCond.broadcast(); 4843 } else if (mBytesRead != 0) { 4844 // record start succeeds only if first read from audio input 4845 // succeeds 4846 if (mBytesRead > 0) { 4847 mActiveTrack->mState = TrackBase::ACTIVE; 4848 } else { 4849 mActiveTrack.clear(); 4850 } 4851 mStartStopCond.broadcast(); 4852 } 4853 mStandby = false; 4854 } 4855 } 4856 lockEffectChains_l(effectChains); 4857 } 4858 4859 if (mActiveTrack != 0) { 4860 if (mActiveTrack->mState != TrackBase::ACTIVE && 4861 mActiveTrack->mState != TrackBase::RESUMING) { 4862 unlockEffectChains(effectChains); 4863 usleep(kRecordThreadSleepUs); 4864 continue; 4865 } 4866 for (size_t i = 0; i < effectChains.size(); i ++) { 4867 effectChains[i]->process_l(); 4868 } 4869 4870 buffer.frameCount = mFrameCount; 4871 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4872 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4873 size_t framesOut = buffer.frameCount; 4874 if (mResampler == NULL) { 4875 // no resampling 4876 while (framesOut) { 4877 size_t framesIn = mFrameCount - mRsmpInIndex; 4878 if (framesIn) { 4879 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4880 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4881 if (framesIn > framesOut) 4882 framesIn = framesOut; 4883 mRsmpInIndex += framesIn; 4884 framesOut -= framesIn; 4885 if ((int)mChannelCount == mReqChannelCount || 4886 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4887 memcpy(dst, src, framesIn * mFrameSize); 4888 } else { 4889 int16_t *src16 = (int16_t *)src; 4890 int16_t *dst16 = (int16_t *)dst; 4891 if (mChannelCount == 1) { 4892 while (framesIn--) { 4893 *dst16++ = *src16; 4894 *dst16++ = *src16++; 4895 } 4896 } else { 4897 while (framesIn--) { 4898 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4899 src16 += 2; 4900 } 4901 } 4902 } 4903 } 4904 if (framesOut && mFrameCount == mRsmpInIndex) { 4905 if (framesOut == mFrameCount && 4906 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4907 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4908 framesOut = 0; 4909 } else { 4910 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4911 mRsmpInIndex = 0; 4912 } 4913 if (mBytesRead < 0) { 4914 ALOGE("Error reading audio input"); 4915 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4916 // Force input into standby so that it tries to 4917 // recover at next read attempt 4918 mInput->stream->common.standby(&mInput->stream->common); 4919 usleep(kRecordThreadSleepUs); 4920 } 4921 mRsmpInIndex = mFrameCount; 4922 framesOut = 0; 4923 buffer.frameCount = 0; 4924 } 4925 } 4926 } 4927 } else { 4928 // resampling 4929 4930 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4931 // alter output frame count as if we were expecting stereo samples 4932 if (mChannelCount == 1 && mReqChannelCount == 1) { 4933 framesOut >>= 1; 4934 } 4935 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4936 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4937 // are 32 bit aligned which should be always true. 4938 if (mChannelCount == 2 && mReqChannelCount == 1) { 4939 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4940 // the resampler always outputs stereo samples: do post stereo to mono conversion 4941 int16_t *src = (int16_t *)mRsmpOutBuffer; 4942 int16_t *dst = buffer.i16; 4943 while (framesOut--) { 4944 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4945 src += 2; 4946 } 4947 } else { 4948 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4949 } 4950 4951 } 4952 mActiveTrack->releaseBuffer(&buffer); 4953 mActiveTrack->overflow(); 4954 } 4955 // client isn't retrieving buffers fast enough 4956 else { 4957 if (!mActiveTrack->setOverflow()) { 4958 nsecs_t now = systemTime(); 4959 if ((now - lastWarning) > kWarningThrottleNs) { 4960 ALOGW("RecordThread: buffer overflow"); 4961 lastWarning = now; 4962 } 4963 } 4964 // Release the processor for a while before asking for a new buffer. 4965 // This will give the application more chance to read from the buffer and 4966 // clear the overflow. 4967 usleep(kRecordThreadSleepUs); 4968 } 4969 } 4970 // enable changes in effect chain 4971 unlockEffectChains(effectChains); 4972 effectChains.clear(); 4973 } 4974 4975 if (!mStandby) { 4976 mInput->stream->common.standby(&mInput->stream->common); 4977 } 4978 mActiveTrack.clear(); 4979 4980 mStartStopCond.broadcast(); 4981 4982 releaseWakeLock(); 4983 4984 ALOGV("RecordThread %p exiting", this); 4985 return false; 4986} 4987 4988 4989sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4990 const sp<AudioFlinger::Client>& client, 4991 uint32_t sampleRate, 4992 audio_format_t format, 4993 int channelMask, 4994 int frameCount, 4995 int sessionId, 4996 status_t *status) 4997{ 4998 sp<RecordTrack> track; 4999 status_t lStatus; 5000 5001 lStatus = initCheck(); 5002 if (lStatus != NO_ERROR) { 5003 ALOGE("Audio driver not initialized."); 5004 goto Exit; 5005 } 5006 5007 { // scope for mLock 5008 Mutex::Autolock _l(mLock); 5009 5010 track = new RecordTrack(this, client, sampleRate, 5011 format, channelMask, frameCount, sessionId); 5012 5013 if (track->getCblk() == 0) { 5014 lStatus = NO_MEMORY; 5015 goto Exit; 5016 } 5017 5018 mTrack = track.get(); 5019 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5020 bool suspend = audio_is_bluetooth_sco_device( 5021 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5022 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5023 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5024 } 5025 lStatus = NO_ERROR; 5026 5027Exit: 5028 if (status) { 5029 *status = lStatus; 5030 } 5031 return track; 5032} 5033 5034status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5035{ 5036 ALOGV("RecordThread::start tid=%d", tid); 5037 sp <ThreadBase> strongMe = this; 5038 status_t status = NO_ERROR; 5039 { 5040 AutoMutex lock(mLock); 5041 if (mActiveTrack != 0) { 5042 if (recordTrack != mActiveTrack.get()) { 5043 status = -EBUSY; 5044 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5045 mActiveTrack->mState = TrackBase::ACTIVE; 5046 } 5047 return status; 5048 } 5049 5050 recordTrack->mState = TrackBase::IDLE; 5051 mActiveTrack = recordTrack; 5052 mLock.unlock(); 5053 status_t status = AudioSystem::startInput(mId); 5054 mLock.lock(); 5055 if (status != NO_ERROR) { 5056 mActiveTrack.clear(); 5057 return status; 5058 } 5059 mRsmpInIndex = mFrameCount; 5060 mBytesRead = 0; 5061 if (mResampler != NULL) { 5062 mResampler->reset(); 5063 } 5064 mActiveTrack->mState = TrackBase::RESUMING; 5065 // signal thread to start 5066 ALOGV("Signal record thread"); 5067 mWaitWorkCV.signal(); 5068 // do not wait for mStartStopCond if exiting 5069 if (exitPending()) { 5070 mActiveTrack.clear(); 5071 status = INVALID_OPERATION; 5072 goto startError; 5073 } 5074 mStartStopCond.wait(mLock); 5075 if (mActiveTrack == 0) { 5076 ALOGV("Record failed to start"); 5077 status = BAD_VALUE; 5078 goto startError; 5079 } 5080 ALOGV("Record started OK"); 5081 return status; 5082 } 5083startError: 5084 AudioSystem::stopInput(mId); 5085 return status; 5086} 5087 5088void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5089 ALOGV("RecordThread::stop"); 5090 sp <ThreadBase> strongMe = this; 5091 { 5092 AutoMutex lock(mLock); 5093 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5094 mActiveTrack->mState = TrackBase::PAUSING; 5095 // do not wait for mStartStopCond if exiting 5096 if (exitPending()) { 5097 return; 5098 } 5099 mStartStopCond.wait(mLock); 5100 // if we have been restarted, recordTrack == mActiveTrack.get() here 5101 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5102 mLock.unlock(); 5103 AudioSystem::stopInput(mId); 5104 mLock.lock(); 5105 ALOGV("Record stopped OK"); 5106 } 5107 } 5108 } 5109} 5110 5111status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5112{ 5113 const size_t SIZE = 256; 5114 char buffer[SIZE]; 5115 String8 result; 5116 5117 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5118 result.append(buffer); 5119 5120 if (mActiveTrack != 0) { 5121 result.append("Active Track:\n"); 5122 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5123 mActiveTrack->dump(buffer, SIZE); 5124 result.append(buffer); 5125 5126 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5127 result.append(buffer); 5128 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5129 result.append(buffer); 5130 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5131 result.append(buffer); 5132 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5133 result.append(buffer); 5134 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5135 result.append(buffer); 5136 5137 5138 } else { 5139 result.append("No record client\n"); 5140 } 5141 write(fd, result.string(), result.size()); 5142 5143 dumpBase(fd, args); 5144 dumpEffectChains(fd, args); 5145 5146 return NO_ERROR; 5147} 5148 5149status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5150{ 5151 size_t framesReq = buffer->frameCount; 5152 size_t framesReady = mFrameCount - mRsmpInIndex; 5153 int channelCount; 5154 5155 if (framesReady == 0) { 5156 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5157 if (mBytesRead < 0) { 5158 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5159 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5160 // Force input into standby so that it tries to 5161 // recover at next read attempt 5162 mInput->stream->common.standby(&mInput->stream->common); 5163 usleep(kRecordThreadSleepUs); 5164 } 5165 buffer->raw = NULL; 5166 buffer->frameCount = 0; 5167 return NOT_ENOUGH_DATA; 5168 } 5169 mRsmpInIndex = 0; 5170 framesReady = mFrameCount; 5171 } 5172 5173 if (framesReq > framesReady) { 5174 framesReq = framesReady; 5175 } 5176 5177 if (mChannelCount == 1 && mReqChannelCount == 2) { 5178 channelCount = 1; 5179 } else { 5180 channelCount = 2; 5181 } 5182 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5183 buffer->frameCount = framesReq; 5184 return NO_ERROR; 5185} 5186 5187void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5188{ 5189 mRsmpInIndex += buffer->frameCount; 5190 buffer->frameCount = 0; 5191} 5192 5193bool AudioFlinger::RecordThread::checkForNewParameters_l() 5194{ 5195 bool reconfig = false; 5196 5197 while (!mNewParameters.isEmpty()) { 5198 status_t status = NO_ERROR; 5199 String8 keyValuePair = mNewParameters[0]; 5200 AudioParameter param = AudioParameter(keyValuePair); 5201 int value; 5202 audio_format_t reqFormat = mFormat; 5203 int reqSamplingRate = mReqSampleRate; 5204 int reqChannelCount = mReqChannelCount; 5205 5206 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5207 reqSamplingRate = value; 5208 reconfig = true; 5209 } 5210 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5211 reqFormat = (audio_format_t) value; 5212 reconfig = true; 5213 } 5214 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5215 reqChannelCount = popcount(value); 5216 reconfig = true; 5217 } 5218 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5219 // do not accept frame count changes if tracks are open as the track buffer 5220 // size depends on frame count and correct behavior would not be guaranteed 5221 // if frame count is changed after track creation 5222 if (mActiveTrack != 0) { 5223 status = INVALID_OPERATION; 5224 } else { 5225 reconfig = true; 5226 } 5227 } 5228 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5229 // forward device change to effects that have requested to be 5230 // aware of attached audio device. 5231 for (size_t i = 0; i < mEffectChains.size(); i++) { 5232 mEffectChains[i]->setDevice_l(value); 5233 } 5234 // store input device and output device but do not forward output device to audio HAL. 5235 // Note that status is ignored by the caller for output device 5236 // (see AudioFlinger::setParameters() 5237 if (value & AUDIO_DEVICE_OUT_ALL) { 5238 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5239 status = BAD_VALUE; 5240 } else { 5241 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5242 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5243 if (mTrack != NULL) { 5244 bool suspend = audio_is_bluetooth_sco_device( 5245 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5246 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5247 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5248 } 5249 } 5250 mDevice |= (uint32_t)value; 5251 } 5252 if (status == NO_ERROR) { 5253 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5254 if (status == INVALID_OPERATION) { 5255 mInput->stream->common.standby(&mInput->stream->common); 5256 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5257 } 5258 if (reconfig) { 5259 if (status == BAD_VALUE && 5260 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5261 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5262 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5263 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5264 (reqChannelCount < 3)) { 5265 status = NO_ERROR; 5266 } 5267 if (status == NO_ERROR) { 5268 readInputParameters(); 5269 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5270 } 5271 } 5272 } 5273 5274 mNewParameters.removeAt(0); 5275 5276 mParamStatus = status; 5277 mParamCond.signal(); 5278 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5279 // already timed out waiting for the status and will never signal the condition. 5280 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5281 } 5282 return reconfig; 5283} 5284 5285String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5286{ 5287 char *s; 5288 String8 out_s8 = String8(); 5289 5290 Mutex::Autolock _l(mLock); 5291 if (initCheck() != NO_ERROR) { 5292 return out_s8; 5293 } 5294 5295 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5296 out_s8 = String8(s); 5297 free(s); 5298 return out_s8; 5299} 5300 5301void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5302 AudioSystem::OutputDescriptor desc; 5303 void *param2 = NULL; 5304 5305 switch (event) { 5306 case AudioSystem::INPUT_OPENED: 5307 case AudioSystem::INPUT_CONFIG_CHANGED: 5308 desc.channels = mChannelMask; 5309 desc.samplingRate = mSampleRate; 5310 desc.format = mFormat; 5311 desc.frameCount = mFrameCount; 5312 desc.latency = 0; 5313 param2 = &desc; 5314 break; 5315 5316 case AudioSystem::INPUT_CLOSED: 5317 default: 5318 break; 5319 } 5320 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5321} 5322 5323void AudioFlinger::RecordThread::readInputParameters() 5324{ 5325 delete mRsmpInBuffer; 5326 // mRsmpInBuffer is always assigned a new[] below 5327 delete mRsmpOutBuffer; 5328 mRsmpOutBuffer = NULL; 5329 delete mResampler; 5330 mResampler = NULL; 5331 5332 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5333 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5334 mChannelCount = (uint16_t)popcount(mChannelMask); 5335 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5336 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5337 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5338 mFrameCount = mInputBytes / mFrameSize; 5339 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5340 5341 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5342 { 5343 int channelCount; 5344 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5345 // stereo to mono post process as the resampler always outputs stereo. 5346 if (mChannelCount == 1 && mReqChannelCount == 2) { 5347 channelCount = 1; 5348 } else { 5349 channelCount = 2; 5350 } 5351 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5352 mResampler->setSampleRate(mSampleRate); 5353 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5354 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5355 5356 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5357 if (mChannelCount == 1 && mReqChannelCount == 1) { 5358 mFrameCount >>= 1; 5359 } 5360 5361 } 5362 mRsmpInIndex = mFrameCount; 5363} 5364 5365unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5366{ 5367 Mutex::Autolock _l(mLock); 5368 if (initCheck() != NO_ERROR) { 5369 return 0; 5370 } 5371 5372 return mInput->stream->get_input_frames_lost(mInput->stream); 5373} 5374 5375uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5376{ 5377 Mutex::Autolock _l(mLock); 5378 uint32_t result = 0; 5379 if (getEffectChain_l(sessionId) != 0) { 5380 result = EFFECT_SESSION; 5381 } 5382 5383 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5384 result |= TRACK_SESSION; 5385 } 5386 5387 return result; 5388} 5389 5390AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5391{ 5392 Mutex::Autolock _l(mLock); 5393 return mTrack; 5394} 5395 5396AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5397{ 5398 Mutex::Autolock _l(mLock); 5399 return mInput; 5400} 5401 5402AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5403{ 5404 Mutex::Autolock _l(mLock); 5405 AudioStreamIn *input = mInput; 5406 mInput = NULL; 5407 return input; 5408} 5409 5410// this method must always be called either with ThreadBase mLock held or inside the thread loop 5411audio_stream_t* AudioFlinger::RecordThread::stream() 5412{ 5413 if (mInput == NULL) { 5414 return NULL; 5415 } 5416 return &mInput->stream->common; 5417} 5418 5419 5420// ---------------------------------------------------------------------------- 5421 5422audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5423 uint32_t *pSamplingRate, 5424 audio_format_t *pFormat, 5425 uint32_t *pChannels, 5426 uint32_t *pLatencyMs, 5427 uint32_t flags) 5428{ 5429 status_t status; 5430 PlaybackThread *thread = NULL; 5431 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5432 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5433 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5434 uint32_t channels = pChannels ? *pChannels : 0; 5435 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5436 audio_stream_out_t *outStream; 5437 audio_hw_device_t *outHwDev; 5438 5439 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5440 pDevices ? *pDevices : 0, 5441 samplingRate, 5442 format, 5443 channels, 5444 flags); 5445 5446 if (pDevices == NULL || *pDevices == 0) { 5447 return 0; 5448 } 5449 5450 Mutex::Autolock _l(mLock); 5451 5452 outHwDev = findSuitableHwDev_l(*pDevices); 5453 if (outHwDev == NULL) 5454 return 0; 5455 5456 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5457 &channels, &samplingRate, &outStream); 5458 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5459 outStream, 5460 samplingRate, 5461 format, 5462 channels, 5463 status); 5464 5465 mHardwareStatus = AUDIO_HW_IDLE; 5466 if (outStream != NULL) { 5467 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5468 audio_io_handle_t id = nextUniqueId(); 5469 5470 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5471 (format != AUDIO_FORMAT_PCM_16_BIT) || 5472 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5473 thread = new DirectOutputThread(this, output, id, *pDevices); 5474 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5475 } else { 5476 thread = new MixerThread(this, output, id, *pDevices); 5477 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5478 } 5479 mPlaybackThreads.add(id, thread); 5480 5481 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5482 if (pFormat != NULL) *pFormat = format; 5483 if (pChannels != NULL) *pChannels = channels; 5484 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5485 5486 // notify client processes of the new output creation 5487 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5488 return id; 5489 } 5490 5491 return 0; 5492} 5493 5494audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5495 audio_io_handle_t output2) 5496{ 5497 Mutex::Autolock _l(mLock); 5498 MixerThread *thread1 = checkMixerThread_l(output1); 5499 MixerThread *thread2 = checkMixerThread_l(output2); 5500 5501 if (thread1 == NULL || thread2 == NULL) { 5502 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5503 return 0; 5504 } 5505 5506 audio_io_handle_t id = nextUniqueId(); 5507 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5508 thread->addOutputTrack(thread2); 5509 mPlaybackThreads.add(id, thread); 5510 // notify client processes of the new output creation 5511 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5512 return id; 5513} 5514 5515status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5516{ 5517 // keep strong reference on the playback thread so that 5518 // it is not destroyed while exit() is executed 5519 sp <PlaybackThread> thread; 5520 { 5521 Mutex::Autolock _l(mLock); 5522 thread = checkPlaybackThread_l(output); 5523 if (thread == NULL) { 5524 return BAD_VALUE; 5525 } 5526 5527 ALOGV("closeOutput() %d", output); 5528 5529 if (thread->type() == ThreadBase::MIXER) { 5530 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5531 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5532 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5533 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5534 } 5535 } 5536 } 5537 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5538 mPlaybackThreads.removeItem(output); 5539 } 5540 thread->exit(); 5541 // The thread entity (active unit of execution) is no longer running here, 5542 // but the ThreadBase container still exists. 5543 5544 if (thread->type() != ThreadBase::DUPLICATING) { 5545 AudioStreamOut *out = thread->clearOutput(); 5546 assert(out != NULL); 5547 // from now on thread->mOutput is NULL 5548 out->hwDev->close_output_stream(out->hwDev, out->stream); 5549 delete out; 5550 } 5551 return NO_ERROR; 5552} 5553 5554status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5555{ 5556 Mutex::Autolock _l(mLock); 5557 PlaybackThread *thread = checkPlaybackThread_l(output); 5558 5559 if (thread == NULL) { 5560 return BAD_VALUE; 5561 } 5562 5563 ALOGV("suspendOutput() %d", output); 5564 thread->suspend(); 5565 5566 return NO_ERROR; 5567} 5568 5569status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5570{ 5571 Mutex::Autolock _l(mLock); 5572 PlaybackThread *thread = checkPlaybackThread_l(output); 5573 5574 if (thread == NULL) { 5575 return BAD_VALUE; 5576 } 5577 5578 ALOGV("restoreOutput() %d", output); 5579 5580 thread->restore(); 5581 5582 return NO_ERROR; 5583} 5584 5585audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5586 uint32_t *pSamplingRate, 5587 audio_format_t *pFormat, 5588 uint32_t *pChannels, 5589 audio_in_acoustics_t acoustics) 5590{ 5591 status_t status; 5592 RecordThread *thread = NULL; 5593 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5594 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5595 uint32_t channels = pChannels ? *pChannels : 0; 5596 uint32_t reqSamplingRate = samplingRate; 5597 audio_format_t reqFormat = format; 5598 uint32_t reqChannels = channels; 5599 audio_stream_in_t *inStream; 5600 audio_hw_device_t *inHwDev; 5601 5602 if (pDevices == NULL || *pDevices == 0) { 5603 return 0; 5604 } 5605 5606 Mutex::Autolock _l(mLock); 5607 5608 inHwDev = findSuitableHwDev_l(*pDevices); 5609 if (inHwDev == NULL) 5610 return 0; 5611 5612 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5613 &channels, &samplingRate, 5614 acoustics, 5615 &inStream); 5616 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5617 inStream, 5618 samplingRate, 5619 format, 5620 channels, 5621 acoustics, 5622 status); 5623 5624 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5625 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5626 // or stereo to mono conversions on 16 bit PCM inputs. 5627 if (inStream == NULL && status == BAD_VALUE && 5628 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5629 (samplingRate <= 2 * reqSamplingRate) && 5630 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5631 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5632 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5633 &channels, &samplingRate, 5634 acoustics, 5635 &inStream); 5636 } 5637 5638 if (inStream != NULL) { 5639 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5640 5641 audio_io_handle_t id = nextUniqueId(); 5642 // Start record thread 5643 // RecorThread require both input and output device indication to forward to audio 5644 // pre processing modules 5645 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5646 thread = new RecordThread(this, 5647 input, 5648 reqSamplingRate, 5649 reqChannels, 5650 id, 5651 device); 5652 mRecordThreads.add(id, thread); 5653 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5654 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5655 if (pFormat != NULL) *pFormat = format; 5656 if (pChannels != NULL) *pChannels = reqChannels; 5657 5658 input->stream->common.standby(&input->stream->common); 5659 5660 // notify client processes of the new input creation 5661 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5662 return id; 5663 } 5664 5665 return 0; 5666} 5667 5668status_t AudioFlinger::closeInput(audio_io_handle_t input) 5669{ 5670 // keep strong reference on the record thread so that 5671 // it is not destroyed while exit() is executed 5672 sp <RecordThread> thread; 5673 { 5674 Mutex::Autolock _l(mLock); 5675 thread = checkRecordThread_l(input); 5676 if (thread == NULL) { 5677 return BAD_VALUE; 5678 } 5679 5680 ALOGV("closeInput() %d", input); 5681 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5682 mRecordThreads.removeItem(input); 5683 } 5684 thread->exit(); 5685 // The thread entity (active unit of execution) is no longer running here, 5686 // but the ThreadBase container still exists. 5687 5688 AudioStreamIn *in = thread->clearInput(); 5689 assert(in != NULL); 5690 // from now on thread->mInput is NULL 5691 in->hwDev->close_input_stream(in->hwDev, in->stream); 5692 delete in; 5693 5694 return NO_ERROR; 5695} 5696 5697status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5698{ 5699 Mutex::Autolock _l(mLock); 5700 MixerThread *dstThread = checkMixerThread_l(output); 5701 if (dstThread == NULL) { 5702 ALOGW("setStreamOutput() bad output id %d", output); 5703 return BAD_VALUE; 5704 } 5705 5706 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5707 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5708 5709 dstThread->setStreamValid(stream, true); 5710 5711 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5712 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5713 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5714 MixerThread *srcThread = (MixerThread *)thread; 5715 srcThread->setStreamValid(stream, false); 5716 srcThread->invalidateTracks(stream); 5717 } 5718 } 5719 5720 return NO_ERROR; 5721} 5722 5723 5724int AudioFlinger::newAudioSessionId() 5725{ 5726 return nextUniqueId(); 5727} 5728 5729void AudioFlinger::acquireAudioSessionId(int audioSession) 5730{ 5731 Mutex::Autolock _l(mLock); 5732 pid_t caller = IPCThreadState::self()->getCallingPid(); 5733 ALOGV("acquiring %d from %d", audioSession, caller); 5734 size_t num = mAudioSessionRefs.size(); 5735 for (size_t i = 0; i< num; i++) { 5736 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5737 if (ref->sessionid == audioSession && ref->pid == caller) { 5738 ref->cnt++; 5739 ALOGV(" incremented refcount to %d", ref->cnt); 5740 return; 5741 } 5742 } 5743 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5744 ALOGV(" added new entry for %d", audioSession); 5745} 5746 5747void AudioFlinger::releaseAudioSessionId(int audioSession) 5748{ 5749 Mutex::Autolock _l(mLock); 5750 pid_t caller = IPCThreadState::self()->getCallingPid(); 5751 ALOGV("releasing %d from %d", audioSession, caller); 5752 size_t num = mAudioSessionRefs.size(); 5753 for (size_t i = 0; i< num; i++) { 5754 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5755 if (ref->sessionid == audioSession && ref->pid == caller) { 5756 ref->cnt--; 5757 ALOGV(" decremented refcount to %d", ref->cnt); 5758 if (ref->cnt == 0) { 5759 mAudioSessionRefs.removeAt(i); 5760 delete ref; 5761 purgeStaleEffects_l(); 5762 } 5763 return; 5764 } 5765 } 5766 ALOGW("session id %d not found for pid %d", audioSession, caller); 5767} 5768 5769void AudioFlinger::purgeStaleEffects_l() { 5770 5771 ALOGV("purging stale effects"); 5772 5773 Vector< sp<EffectChain> > chains; 5774 5775 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5776 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5777 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5778 sp<EffectChain> ec = t->mEffectChains[j]; 5779 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5780 chains.push(ec); 5781 } 5782 } 5783 } 5784 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5785 sp<RecordThread> t = mRecordThreads.valueAt(i); 5786 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5787 sp<EffectChain> ec = t->mEffectChains[j]; 5788 chains.push(ec); 5789 } 5790 } 5791 5792 for (size_t i = 0; i < chains.size(); i++) { 5793 sp<EffectChain> ec = chains[i]; 5794 int sessionid = ec->sessionId(); 5795 sp<ThreadBase> t = ec->mThread.promote(); 5796 if (t == 0) { 5797 continue; 5798 } 5799 size_t numsessionrefs = mAudioSessionRefs.size(); 5800 bool found = false; 5801 for (size_t k = 0; k < numsessionrefs; k++) { 5802 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5803 if (ref->sessionid == sessionid) { 5804 ALOGV(" session %d still exists for %d with %d refs", 5805 sessionid, ref->pid, ref->cnt); 5806 found = true; 5807 break; 5808 } 5809 } 5810 if (!found) { 5811 // remove all effects from the chain 5812 while (ec->mEffects.size()) { 5813 sp<EffectModule> effect = ec->mEffects[0]; 5814 effect->unPin(); 5815 Mutex::Autolock _l (t->mLock); 5816 t->removeEffect_l(effect); 5817 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5818 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5819 if (handle != 0) { 5820 handle->mEffect.clear(); 5821 if (handle->mHasControl && handle->mEnabled) { 5822 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5823 } 5824 } 5825 } 5826 AudioSystem::unregisterEffect(effect->id()); 5827 } 5828 } 5829 } 5830 return; 5831} 5832 5833// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5834AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5835{ 5836 return mPlaybackThreads.valueFor(output).get(); 5837} 5838 5839// checkMixerThread_l() must be called with AudioFlinger::mLock held 5840AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5841{ 5842 PlaybackThread *thread = checkPlaybackThread_l(output); 5843 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5844} 5845 5846// checkRecordThread_l() must be called with AudioFlinger::mLock held 5847AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5848{ 5849 return mRecordThreads.valueFor(input).get(); 5850} 5851 5852uint32_t AudioFlinger::nextUniqueId() 5853{ 5854 return android_atomic_inc(&mNextUniqueId); 5855} 5856 5857AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5858{ 5859 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5860 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5861 AudioStreamOut *output = thread->getOutput(); 5862 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5863 return thread; 5864 } 5865 } 5866 return NULL; 5867} 5868 5869uint32_t AudioFlinger::primaryOutputDevice_l() 5870{ 5871 PlaybackThread *thread = primaryPlaybackThread_l(); 5872 5873 if (thread == NULL) { 5874 return 0; 5875 } 5876 5877 return thread->device(); 5878} 5879 5880 5881// ---------------------------------------------------------------------------- 5882// Effect management 5883// ---------------------------------------------------------------------------- 5884 5885 5886status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5887{ 5888 Mutex::Autolock _l(mLock); 5889 return EffectQueryNumberEffects(numEffects); 5890} 5891 5892status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5893{ 5894 Mutex::Autolock _l(mLock); 5895 return EffectQueryEffect(index, descriptor); 5896} 5897 5898status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5899 effect_descriptor_t *descriptor) const 5900{ 5901 Mutex::Autolock _l(mLock); 5902 return EffectGetDescriptor(pUuid, descriptor); 5903} 5904 5905 5906sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5907 effect_descriptor_t *pDesc, 5908 const sp<IEffectClient>& effectClient, 5909 int32_t priority, 5910 audio_io_handle_t io, 5911 int sessionId, 5912 status_t *status, 5913 int *id, 5914 int *enabled) 5915{ 5916 status_t lStatus = NO_ERROR; 5917 sp<EffectHandle> handle; 5918 effect_descriptor_t desc; 5919 5920 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5921 pid, effectClient.get(), priority, sessionId, io); 5922 5923 if (pDesc == NULL) { 5924 lStatus = BAD_VALUE; 5925 goto Exit; 5926 } 5927 5928 // check audio settings permission for global effects 5929 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5930 lStatus = PERMISSION_DENIED; 5931 goto Exit; 5932 } 5933 5934 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5935 // that can only be created by audio policy manager (running in same process) 5936 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5937 lStatus = PERMISSION_DENIED; 5938 goto Exit; 5939 } 5940 5941 if (io == 0) { 5942 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5943 // output must be specified by AudioPolicyManager when using session 5944 // AUDIO_SESSION_OUTPUT_STAGE 5945 lStatus = BAD_VALUE; 5946 goto Exit; 5947 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5948 // if the output returned by getOutputForEffect() is removed before we lock the 5949 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5950 // and we will exit safely 5951 io = AudioSystem::getOutputForEffect(&desc); 5952 } 5953 } 5954 5955 { 5956 Mutex::Autolock _l(mLock); 5957 5958 5959 if (!EffectIsNullUuid(&pDesc->uuid)) { 5960 // if uuid is specified, request effect descriptor 5961 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5962 if (lStatus < 0) { 5963 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5964 goto Exit; 5965 } 5966 } else { 5967 // if uuid is not specified, look for an available implementation 5968 // of the required type in effect factory 5969 if (EffectIsNullUuid(&pDesc->type)) { 5970 ALOGW("createEffect() no effect type"); 5971 lStatus = BAD_VALUE; 5972 goto Exit; 5973 } 5974 uint32_t numEffects = 0; 5975 effect_descriptor_t d; 5976 d.flags = 0; // prevent compiler warning 5977 bool found = false; 5978 5979 lStatus = EffectQueryNumberEffects(&numEffects); 5980 if (lStatus < 0) { 5981 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5982 goto Exit; 5983 } 5984 for (uint32_t i = 0; i < numEffects; i++) { 5985 lStatus = EffectQueryEffect(i, &desc); 5986 if (lStatus < 0) { 5987 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5988 continue; 5989 } 5990 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5991 // If matching type found save effect descriptor. If the session is 5992 // 0 and the effect is not auxiliary, continue enumeration in case 5993 // an auxiliary version of this effect type is available 5994 found = true; 5995 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5996 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5997 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5998 break; 5999 } 6000 } 6001 } 6002 if (!found) { 6003 lStatus = BAD_VALUE; 6004 ALOGW("createEffect() effect not found"); 6005 goto Exit; 6006 } 6007 // For same effect type, chose auxiliary version over insert version if 6008 // connect to output mix (Compliance to OpenSL ES) 6009 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6010 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6011 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6012 } 6013 } 6014 6015 // Do not allow auxiliary effects on a session different from 0 (output mix) 6016 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6017 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6018 lStatus = INVALID_OPERATION; 6019 goto Exit; 6020 } 6021 6022 // check recording permission for visualizer 6023 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6024 !recordingAllowed()) { 6025 lStatus = PERMISSION_DENIED; 6026 goto Exit; 6027 } 6028 6029 // return effect descriptor 6030 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6031 6032 // If output is not specified try to find a matching audio session ID in one of the 6033 // output threads. 6034 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6035 // because of code checking output when entering the function. 6036 // Note: io is never 0 when creating an effect on an input 6037 if (io == 0) { 6038 // look for the thread where the specified audio session is present 6039 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6040 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6041 io = mPlaybackThreads.keyAt(i); 6042 break; 6043 } 6044 } 6045 if (io == 0) { 6046 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6047 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6048 io = mRecordThreads.keyAt(i); 6049 break; 6050 } 6051 } 6052 } 6053 // If no output thread contains the requested session ID, default to 6054 // first output. The effect chain will be moved to the correct output 6055 // thread when a track with the same session ID is created 6056 if (io == 0 && mPlaybackThreads.size()) { 6057 io = mPlaybackThreads.keyAt(0); 6058 } 6059 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6060 } 6061 ThreadBase *thread = checkRecordThread_l(io); 6062 if (thread == NULL) { 6063 thread = checkPlaybackThread_l(io); 6064 if (thread == NULL) { 6065 ALOGE("createEffect() unknown output thread"); 6066 lStatus = BAD_VALUE; 6067 goto Exit; 6068 } 6069 } 6070 6071 sp<Client> client = registerPid_l(pid); 6072 6073 // create effect on selected output thread 6074 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6075 &desc, enabled, &lStatus); 6076 if (handle != 0 && id != NULL) { 6077 *id = handle->id(); 6078 } 6079 } 6080 6081Exit: 6082 if(status) { 6083 *status = lStatus; 6084 } 6085 return handle; 6086} 6087 6088status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6089 audio_io_handle_t dstOutput) 6090{ 6091 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6092 sessionId, srcOutput, dstOutput); 6093 Mutex::Autolock _l(mLock); 6094 if (srcOutput == dstOutput) { 6095 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6096 return NO_ERROR; 6097 } 6098 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6099 if (srcThread == NULL) { 6100 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6101 return BAD_VALUE; 6102 } 6103 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6104 if (dstThread == NULL) { 6105 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6106 return BAD_VALUE; 6107 } 6108 6109 Mutex::Autolock _dl(dstThread->mLock); 6110 Mutex::Autolock _sl(srcThread->mLock); 6111 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6112 6113 return NO_ERROR; 6114} 6115 6116// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6117status_t AudioFlinger::moveEffectChain_l(int sessionId, 6118 AudioFlinger::PlaybackThread *srcThread, 6119 AudioFlinger::PlaybackThread *dstThread, 6120 bool reRegister) 6121{ 6122 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6123 sessionId, srcThread, dstThread); 6124 6125 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6126 if (chain == 0) { 6127 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6128 sessionId, srcThread); 6129 return INVALID_OPERATION; 6130 } 6131 6132 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6133 // so that a new chain is created with correct parameters when first effect is added. This is 6134 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6135 // removed. 6136 srcThread->removeEffectChain_l(chain); 6137 6138 // transfer all effects one by one so that new effect chain is created on new thread with 6139 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6140 audio_io_handle_t dstOutput = dstThread->id(); 6141 sp<EffectChain> dstChain; 6142 uint32_t strategy = 0; // prevent compiler warning 6143 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6144 while (effect != 0) { 6145 srcThread->removeEffect_l(effect); 6146 dstThread->addEffect_l(effect); 6147 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6148 if (effect->state() == EffectModule::ACTIVE || 6149 effect->state() == EffectModule::STOPPING) { 6150 effect->start(); 6151 } 6152 // if the move request is not received from audio policy manager, the effect must be 6153 // re-registered with the new strategy and output 6154 if (dstChain == 0) { 6155 dstChain = effect->chain().promote(); 6156 if (dstChain == 0) { 6157 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6158 srcThread->addEffect_l(effect); 6159 return NO_INIT; 6160 } 6161 strategy = dstChain->strategy(); 6162 } 6163 if (reRegister) { 6164 AudioSystem::unregisterEffect(effect->id()); 6165 AudioSystem::registerEffect(&effect->desc(), 6166 dstOutput, 6167 strategy, 6168 sessionId, 6169 effect->id()); 6170 } 6171 effect = chain->getEffectFromId_l(0); 6172 } 6173 6174 return NO_ERROR; 6175} 6176 6177 6178// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6179sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6180 const sp<AudioFlinger::Client>& client, 6181 const sp<IEffectClient>& effectClient, 6182 int32_t priority, 6183 int sessionId, 6184 effect_descriptor_t *desc, 6185 int *enabled, 6186 status_t *status 6187 ) 6188{ 6189 sp<EffectModule> effect; 6190 sp<EffectHandle> handle; 6191 status_t lStatus; 6192 sp<EffectChain> chain; 6193 bool chainCreated = false; 6194 bool effectCreated = false; 6195 bool effectRegistered = false; 6196 6197 lStatus = initCheck(); 6198 if (lStatus != NO_ERROR) { 6199 ALOGW("createEffect_l() Audio driver not initialized."); 6200 goto Exit; 6201 } 6202 6203 // Do not allow effects with session ID 0 on direct output or duplicating threads 6204 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6205 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6206 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6207 desc->name, sessionId); 6208 lStatus = BAD_VALUE; 6209 goto Exit; 6210 } 6211 // Only Pre processor effects are allowed on input threads and only on input threads 6212 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6213 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6214 desc->name, desc->flags, mType); 6215 lStatus = BAD_VALUE; 6216 goto Exit; 6217 } 6218 6219 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6220 6221 { // scope for mLock 6222 Mutex::Autolock _l(mLock); 6223 6224 // check for existing effect chain with the requested audio session 6225 chain = getEffectChain_l(sessionId); 6226 if (chain == 0) { 6227 // create a new chain for this session 6228 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6229 chain = new EffectChain(this, sessionId); 6230 addEffectChain_l(chain); 6231 chain->setStrategy(getStrategyForSession_l(sessionId)); 6232 chainCreated = true; 6233 } else { 6234 effect = chain->getEffectFromDesc_l(desc); 6235 } 6236 6237 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6238 6239 if (effect == 0) { 6240 int id = mAudioFlinger->nextUniqueId(); 6241 // Check CPU and memory usage 6242 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6243 if (lStatus != NO_ERROR) { 6244 goto Exit; 6245 } 6246 effectRegistered = true; 6247 // create a new effect module if none present in the chain 6248 effect = new EffectModule(this, chain, desc, id, sessionId); 6249 lStatus = effect->status(); 6250 if (lStatus != NO_ERROR) { 6251 goto Exit; 6252 } 6253 lStatus = chain->addEffect_l(effect); 6254 if (lStatus != NO_ERROR) { 6255 goto Exit; 6256 } 6257 effectCreated = true; 6258 6259 effect->setDevice(mDevice); 6260 effect->setMode(mAudioFlinger->getMode()); 6261 } 6262 // create effect handle and connect it to effect module 6263 handle = new EffectHandle(effect, client, effectClient, priority); 6264 lStatus = effect->addHandle(handle); 6265 if (enabled != NULL) { 6266 *enabled = (int)effect->isEnabled(); 6267 } 6268 } 6269 6270Exit: 6271 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6272 Mutex::Autolock _l(mLock); 6273 if (effectCreated) { 6274 chain->removeEffect_l(effect); 6275 } 6276 if (effectRegistered) { 6277 AudioSystem::unregisterEffect(effect->id()); 6278 } 6279 if (chainCreated) { 6280 removeEffectChain_l(chain); 6281 } 6282 handle.clear(); 6283 } 6284 6285 if(status) { 6286 *status = lStatus; 6287 } 6288 return handle; 6289} 6290 6291sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6292{ 6293 sp<EffectChain> chain = getEffectChain_l(sessionId); 6294 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6295} 6296 6297// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6298// PlaybackThread::mLock held 6299status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6300{ 6301 // check for existing effect chain with the requested audio session 6302 int sessionId = effect->sessionId(); 6303 sp<EffectChain> chain = getEffectChain_l(sessionId); 6304 bool chainCreated = false; 6305 6306 if (chain == 0) { 6307 // create a new chain for this session 6308 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6309 chain = new EffectChain(this, sessionId); 6310 addEffectChain_l(chain); 6311 chain->setStrategy(getStrategyForSession_l(sessionId)); 6312 chainCreated = true; 6313 } 6314 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6315 6316 if (chain->getEffectFromId_l(effect->id()) != 0) { 6317 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6318 this, effect->desc().name, chain.get()); 6319 return BAD_VALUE; 6320 } 6321 6322 status_t status = chain->addEffect_l(effect); 6323 if (status != NO_ERROR) { 6324 if (chainCreated) { 6325 removeEffectChain_l(chain); 6326 } 6327 return status; 6328 } 6329 6330 effect->setDevice(mDevice); 6331 effect->setMode(mAudioFlinger->getMode()); 6332 return NO_ERROR; 6333} 6334 6335void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6336 6337 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6338 effect_descriptor_t desc = effect->desc(); 6339 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6340 detachAuxEffect_l(effect->id()); 6341 } 6342 6343 sp<EffectChain> chain = effect->chain().promote(); 6344 if (chain != 0) { 6345 // remove effect chain if removing last effect 6346 if (chain->removeEffect_l(effect) == 0) { 6347 removeEffectChain_l(chain); 6348 } 6349 } else { 6350 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6351 } 6352} 6353 6354void AudioFlinger::ThreadBase::lockEffectChains_l( 6355 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6356{ 6357 effectChains = mEffectChains; 6358 for (size_t i = 0; i < mEffectChains.size(); i++) { 6359 mEffectChains[i]->lock(); 6360 } 6361} 6362 6363void AudioFlinger::ThreadBase::unlockEffectChains( 6364 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6365{ 6366 for (size_t i = 0; i < effectChains.size(); i++) { 6367 effectChains[i]->unlock(); 6368 } 6369} 6370 6371sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6372{ 6373 Mutex::Autolock _l(mLock); 6374 return getEffectChain_l(sessionId); 6375} 6376 6377sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6378{ 6379 size_t size = mEffectChains.size(); 6380 for (size_t i = 0; i < size; i++) { 6381 if (mEffectChains[i]->sessionId() == sessionId) { 6382 return mEffectChains[i]; 6383 } 6384 } 6385 return 0; 6386} 6387 6388void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6389{ 6390 Mutex::Autolock _l(mLock); 6391 size_t size = mEffectChains.size(); 6392 for (size_t i = 0; i < size; i++) { 6393 mEffectChains[i]->setMode_l(mode); 6394 } 6395} 6396 6397void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6398 const wp<EffectHandle>& handle, 6399 bool unpinIfLast) { 6400 6401 Mutex::Autolock _l(mLock); 6402 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6403 // delete the effect module if removing last handle on it 6404 if (effect->removeHandle(handle) == 0) { 6405 if (!effect->isPinned() || unpinIfLast) { 6406 removeEffect_l(effect); 6407 AudioSystem::unregisterEffect(effect->id()); 6408 } 6409 } 6410} 6411 6412status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6413{ 6414 int session = chain->sessionId(); 6415 int16_t *buffer = mMixBuffer; 6416 bool ownsBuffer = false; 6417 6418 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6419 if (session > 0) { 6420 // Only one effect chain can be present in direct output thread and it uses 6421 // the mix buffer as input 6422 if (mType != DIRECT) { 6423 size_t numSamples = mFrameCount * mChannelCount; 6424 buffer = new int16_t[numSamples]; 6425 memset(buffer, 0, numSamples * sizeof(int16_t)); 6426 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6427 ownsBuffer = true; 6428 } 6429 6430 // Attach all tracks with same session ID to this chain. 6431 for (size_t i = 0; i < mTracks.size(); ++i) { 6432 sp<Track> track = mTracks[i]; 6433 if (session == track->sessionId()) { 6434 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6435 track->setMainBuffer(buffer); 6436 chain->incTrackCnt(); 6437 } 6438 } 6439 6440 // indicate all active tracks in the chain 6441 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6442 sp<Track> track = mActiveTracks[i].promote(); 6443 if (track == 0) continue; 6444 if (session == track->sessionId()) { 6445 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6446 chain->incActiveTrackCnt(); 6447 } 6448 } 6449 } 6450 6451 chain->setInBuffer(buffer, ownsBuffer); 6452 chain->setOutBuffer(mMixBuffer); 6453 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6454 // chains list in order to be processed last as it contains output stage effects 6455 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6456 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6457 // after track specific effects and before output stage 6458 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6459 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6460 // Effect chain for other sessions are inserted at beginning of effect 6461 // chains list to be processed before output mix effects. Relative order between other 6462 // sessions is not important 6463 size_t size = mEffectChains.size(); 6464 size_t i = 0; 6465 for (i = 0; i < size; i++) { 6466 if (mEffectChains[i]->sessionId() < session) break; 6467 } 6468 mEffectChains.insertAt(chain, i); 6469 checkSuspendOnAddEffectChain_l(chain); 6470 6471 return NO_ERROR; 6472} 6473 6474size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6475{ 6476 int session = chain->sessionId(); 6477 6478 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6479 6480 for (size_t i = 0; i < mEffectChains.size(); i++) { 6481 if (chain == mEffectChains[i]) { 6482 mEffectChains.removeAt(i); 6483 // detach all active tracks from the chain 6484 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6485 sp<Track> track = mActiveTracks[i].promote(); 6486 if (track == 0) continue; 6487 if (session == track->sessionId()) { 6488 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6489 chain.get(), session); 6490 chain->decActiveTrackCnt(); 6491 } 6492 } 6493 6494 // detach all tracks with same session ID from this chain 6495 for (size_t i = 0; i < mTracks.size(); ++i) { 6496 sp<Track> track = mTracks[i]; 6497 if (session == track->sessionId()) { 6498 track->setMainBuffer(mMixBuffer); 6499 chain->decTrackCnt(); 6500 } 6501 } 6502 break; 6503 } 6504 } 6505 return mEffectChains.size(); 6506} 6507 6508status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6509 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6510{ 6511 Mutex::Autolock _l(mLock); 6512 return attachAuxEffect_l(track, EffectId); 6513} 6514 6515status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6516 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6517{ 6518 status_t status = NO_ERROR; 6519 6520 if (EffectId == 0) { 6521 track->setAuxBuffer(0, NULL); 6522 } else { 6523 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6524 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6525 if (effect != 0) { 6526 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6527 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6528 } else { 6529 status = INVALID_OPERATION; 6530 } 6531 } else { 6532 status = BAD_VALUE; 6533 } 6534 } 6535 return status; 6536} 6537 6538void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6539{ 6540 for (size_t i = 0; i < mTracks.size(); ++i) { 6541 sp<Track> track = mTracks[i]; 6542 if (track->auxEffectId() == effectId) { 6543 attachAuxEffect_l(track, 0); 6544 } 6545 } 6546} 6547 6548status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6549{ 6550 // only one chain per input thread 6551 if (mEffectChains.size() != 0) { 6552 return INVALID_OPERATION; 6553 } 6554 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6555 6556 chain->setInBuffer(NULL); 6557 chain->setOutBuffer(NULL); 6558 6559 checkSuspendOnAddEffectChain_l(chain); 6560 6561 mEffectChains.add(chain); 6562 6563 return NO_ERROR; 6564} 6565 6566size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6567{ 6568 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6569 ALOGW_IF(mEffectChains.size() != 1, 6570 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6571 chain.get(), mEffectChains.size(), this); 6572 if (mEffectChains.size() == 1) { 6573 mEffectChains.removeAt(0); 6574 } 6575 return 0; 6576} 6577 6578// ---------------------------------------------------------------------------- 6579// EffectModule implementation 6580// ---------------------------------------------------------------------------- 6581 6582#undef LOG_TAG 6583#define LOG_TAG "AudioFlinger::EffectModule" 6584 6585AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6586 const wp<AudioFlinger::EffectChain>& chain, 6587 effect_descriptor_t *desc, 6588 int id, 6589 int sessionId) 6590 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6591 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6592{ 6593 ALOGV("Constructor %p", this); 6594 int lStatus; 6595 if (thread == NULL) { 6596 return; 6597 } 6598 6599 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6600 6601 // create effect engine from effect factory 6602 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6603 6604 if (mStatus != NO_ERROR) { 6605 return; 6606 } 6607 lStatus = init(); 6608 if (lStatus < 0) { 6609 mStatus = lStatus; 6610 goto Error; 6611 } 6612 6613 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6614 mPinned = true; 6615 } 6616 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6617 return; 6618Error: 6619 EffectRelease(mEffectInterface); 6620 mEffectInterface = NULL; 6621 ALOGV("Constructor Error %d", mStatus); 6622} 6623 6624AudioFlinger::EffectModule::~EffectModule() 6625{ 6626 ALOGV("Destructor %p", this); 6627 if (mEffectInterface != NULL) { 6628 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6629 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6630 sp<ThreadBase> thread = mThread.promote(); 6631 if (thread != 0) { 6632 audio_stream_t *stream = thread->stream(); 6633 if (stream != NULL) { 6634 stream->remove_audio_effect(stream, mEffectInterface); 6635 } 6636 } 6637 } 6638 // release effect engine 6639 EffectRelease(mEffectInterface); 6640 } 6641} 6642 6643status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6644{ 6645 status_t status; 6646 6647 Mutex::Autolock _l(mLock); 6648 int priority = handle->priority(); 6649 size_t size = mHandles.size(); 6650 sp<EffectHandle> h; 6651 size_t i; 6652 for (i = 0; i < size; i++) { 6653 h = mHandles[i].promote(); 6654 if (h == 0) continue; 6655 if (h->priority() <= priority) break; 6656 } 6657 // if inserted in first place, move effect control from previous owner to this handle 6658 if (i == 0) { 6659 bool enabled = false; 6660 if (h != 0) { 6661 enabled = h->enabled(); 6662 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6663 } 6664 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6665 status = NO_ERROR; 6666 } else { 6667 status = ALREADY_EXISTS; 6668 } 6669 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6670 mHandles.insertAt(handle, i); 6671 return status; 6672} 6673 6674size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6675{ 6676 Mutex::Autolock _l(mLock); 6677 size_t size = mHandles.size(); 6678 size_t i; 6679 for (i = 0; i < size; i++) { 6680 if (mHandles[i] == handle) break; 6681 } 6682 if (i == size) { 6683 return size; 6684 } 6685 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6686 6687 bool enabled = false; 6688 EffectHandle *hdl = handle.unsafe_get(); 6689 if (hdl != NULL) { 6690 ALOGV("removeHandle() unsafe_get OK"); 6691 enabled = hdl->enabled(); 6692 } 6693 mHandles.removeAt(i); 6694 size = mHandles.size(); 6695 // if removed from first place, move effect control from this handle to next in line 6696 if (i == 0 && size != 0) { 6697 sp<EffectHandle> h = mHandles[0].promote(); 6698 if (h != 0) { 6699 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6700 } 6701 } 6702 6703 // Prevent calls to process() and other functions on effect interface from now on. 6704 // The effect engine will be released by the destructor when the last strong reference on 6705 // this object is released which can happen after next process is called. 6706 if (size == 0 && !mPinned) { 6707 mState = DESTROYED; 6708 } 6709 6710 return size; 6711} 6712 6713sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6714{ 6715 Mutex::Autolock _l(mLock); 6716 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6717} 6718 6719void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6720{ 6721 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6722 // keep a strong reference on this EffectModule to avoid calling the 6723 // destructor before we exit 6724 sp<EffectModule> keep(this); 6725 { 6726 sp<ThreadBase> thread = mThread.promote(); 6727 if (thread != 0) { 6728 thread->disconnectEffect(keep, handle, unpinIfLast); 6729 } 6730 } 6731} 6732 6733void AudioFlinger::EffectModule::updateState() { 6734 Mutex::Autolock _l(mLock); 6735 6736 switch (mState) { 6737 case RESTART: 6738 reset_l(); 6739 // FALL THROUGH 6740 6741 case STARTING: 6742 // clear auxiliary effect input buffer for next accumulation 6743 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6744 memset(mConfig.inputCfg.buffer.raw, 6745 0, 6746 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6747 } 6748 start_l(); 6749 mState = ACTIVE; 6750 break; 6751 case STOPPING: 6752 stop_l(); 6753 mDisableWaitCnt = mMaxDisableWaitCnt; 6754 mState = STOPPED; 6755 break; 6756 case STOPPED: 6757 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6758 // turn off sequence. 6759 if (--mDisableWaitCnt == 0) { 6760 reset_l(); 6761 mState = IDLE; 6762 } 6763 break; 6764 default: //IDLE , ACTIVE, DESTROYED 6765 break; 6766 } 6767} 6768 6769void AudioFlinger::EffectModule::process() 6770{ 6771 Mutex::Autolock _l(mLock); 6772 6773 if (mState == DESTROYED || mEffectInterface == NULL || 6774 mConfig.inputCfg.buffer.raw == NULL || 6775 mConfig.outputCfg.buffer.raw == NULL) { 6776 return; 6777 } 6778 6779 if (isProcessEnabled()) { 6780 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6781 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6782 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6783 mConfig.inputCfg.buffer.s32, 6784 mConfig.inputCfg.buffer.frameCount/2); 6785 } 6786 6787 // do the actual processing in the effect engine 6788 int ret = (*mEffectInterface)->process(mEffectInterface, 6789 &mConfig.inputCfg.buffer, 6790 &mConfig.outputCfg.buffer); 6791 6792 // force transition to IDLE state when engine is ready 6793 if (mState == STOPPED && ret == -ENODATA) { 6794 mDisableWaitCnt = 1; 6795 } 6796 6797 // clear auxiliary effect input buffer for next accumulation 6798 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6799 memset(mConfig.inputCfg.buffer.raw, 0, 6800 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6801 } 6802 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6803 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6804 // If an insert effect is idle and input buffer is different from output buffer, 6805 // accumulate input onto output 6806 sp<EffectChain> chain = mChain.promote(); 6807 if (chain != 0 && chain->activeTrackCnt() != 0) { 6808 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6809 int16_t *in = mConfig.inputCfg.buffer.s16; 6810 int16_t *out = mConfig.outputCfg.buffer.s16; 6811 for (size_t i = 0; i < frameCnt; i++) { 6812 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6813 } 6814 } 6815 } 6816} 6817 6818void AudioFlinger::EffectModule::reset_l() 6819{ 6820 if (mEffectInterface == NULL) { 6821 return; 6822 } 6823 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6824} 6825 6826status_t AudioFlinger::EffectModule::configure() 6827{ 6828 uint32_t channels; 6829 if (mEffectInterface == NULL) { 6830 return NO_INIT; 6831 } 6832 6833 sp<ThreadBase> thread = mThread.promote(); 6834 if (thread == 0) { 6835 return DEAD_OBJECT; 6836 } 6837 6838 // TODO: handle configuration of effects replacing track process 6839 if (thread->channelCount() == 1) { 6840 channels = AUDIO_CHANNEL_OUT_MONO; 6841 } else { 6842 channels = AUDIO_CHANNEL_OUT_STEREO; 6843 } 6844 6845 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6846 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6847 } else { 6848 mConfig.inputCfg.channels = channels; 6849 } 6850 mConfig.outputCfg.channels = channels; 6851 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6852 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6853 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6854 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6855 mConfig.inputCfg.bufferProvider.cookie = NULL; 6856 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6857 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6858 mConfig.outputCfg.bufferProvider.cookie = NULL; 6859 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6860 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6861 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6862 // Insert effect: 6863 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6864 // always overwrites output buffer: input buffer == output buffer 6865 // - in other sessions: 6866 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6867 // other effect: overwrites output buffer: input buffer == output buffer 6868 // Auxiliary effect: 6869 // accumulates in output buffer: input buffer != output buffer 6870 // Therefore: accumulate <=> input buffer != output buffer 6871 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6872 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6873 } else { 6874 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6875 } 6876 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6877 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6878 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6879 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6880 6881 ALOGV("configure() %p thread %p buffer %p framecount %d", 6882 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6883 6884 status_t cmdStatus; 6885 uint32_t size = sizeof(int); 6886 status_t status = (*mEffectInterface)->command(mEffectInterface, 6887 EFFECT_CMD_SET_CONFIG, 6888 sizeof(effect_config_t), 6889 &mConfig, 6890 &size, 6891 &cmdStatus); 6892 if (status == 0) { 6893 status = cmdStatus; 6894 } 6895 6896 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6897 (1000 * mConfig.outputCfg.buffer.frameCount); 6898 6899 return status; 6900} 6901 6902status_t AudioFlinger::EffectModule::init() 6903{ 6904 Mutex::Autolock _l(mLock); 6905 if (mEffectInterface == NULL) { 6906 return NO_INIT; 6907 } 6908 status_t cmdStatus; 6909 uint32_t size = sizeof(status_t); 6910 status_t status = (*mEffectInterface)->command(mEffectInterface, 6911 EFFECT_CMD_INIT, 6912 0, 6913 NULL, 6914 &size, 6915 &cmdStatus); 6916 if (status == 0) { 6917 status = cmdStatus; 6918 } 6919 return status; 6920} 6921 6922status_t AudioFlinger::EffectModule::start() 6923{ 6924 Mutex::Autolock _l(mLock); 6925 return start_l(); 6926} 6927 6928status_t AudioFlinger::EffectModule::start_l() 6929{ 6930 if (mEffectInterface == NULL) { 6931 return NO_INIT; 6932 } 6933 status_t cmdStatus; 6934 uint32_t size = sizeof(status_t); 6935 status_t status = (*mEffectInterface)->command(mEffectInterface, 6936 EFFECT_CMD_ENABLE, 6937 0, 6938 NULL, 6939 &size, 6940 &cmdStatus); 6941 if (status == 0) { 6942 status = cmdStatus; 6943 } 6944 if (status == 0 && 6945 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6946 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6947 sp<ThreadBase> thread = mThread.promote(); 6948 if (thread != 0) { 6949 audio_stream_t *stream = thread->stream(); 6950 if (stream != NULL) { 6951 stream->add_audio_effect(stream, mEffectInterface); 6952 } 6953 } 6954 } 6955 return status; 6956} 6957 6958status_t AudioFlinger::EffectModule::stop() 6959{ 6960 Mutex::Autolock _l(mLock); 6961 return stop_l(); 6962} 6963 6964status_t AudioFlinger::EffectModule::stop_l() 6965{ 6966 if (mEffectInterface == NULL) { 6967 return NO_INIT; 6968 } 6969 status_t cmdStatus; 6970 uint32_t size = sizeof(status_t); 6971 status_t status = (*mEffectInterface)->command(mEffectInterface, 6972 EFFECT_CMD_DISABLE, 6973 0, 6974 NULL, 6975 &size, 6976 &cmdStatus); 6977 if (status == 0) { 6978 status = cmdStatus; 6979 } 6980 if (status == 0 && 6981 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6982 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6983 sp<ThreadBase> thread = mThread.promote(); 6984 if (thread != 0) { 6985 audio_stream_t *stream = thread->stream(); 6986 if (stream != NULL) { 6987 stream->remove_audio_effect(stream, mEffectInterface); 6988 } 6989 } 6990 } 6991 return status; 6992} 6993 6994status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6995 uint32_t cmdSize, 6996 void *pCmdData, 6997 uint32_t *replySize, 6998 void *pReplyData) 6999{ 7000 Mutex::Autolock _l(mLock); 7001// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7002 7003 if (mState == DESTROYED || mEffectInterface == NULL) { 7004 return NO_INIT; 7005 } 7006 status_t status = (*mEffectInterface)->command(mEffectInterface, 7007 cmdCode, 7008 cmdSize, 7009 pCmdData, 7010 replySize, 7011 pReplyData); 7012 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7013 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7014 for (size_t i = 1; i < mHandles.size(); i++) { 7015 sp<EffectHandle> h = mHandles[i].promote(); 7016 if (h != 0) { 7017 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7018 } 7019 } 7020 } 7021 return status; 7022} 7023 7024status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7025{ 7026 7027 Mutex::Autolock _l(mLock); 7028 ALOGV("setEnabled %p enabled %d", this, enabled); 7029 7030 if (enabled != isEnabled()) { 7031 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7032 if (enabled && status != NO_ERROR) { 7033 return status; 7034 } 7035 7036 switch (mState) { 7037 // going from disabled to enabled 7038 case IDLE: 7039 mState = STARTING; 7040 break; 7041 case STOPPED: 7042 mState = RESTART; 7043 break; 7044 case STOPPING: 7045 mState = ACTIVE; 7046 break; 7047 7048 // going from enabled to disabled 7049 case RESTART: 7050 mState = STOPPED; 7051 break; 7052 case STARTING: 7053 mState = IDLE; 7054 break; 7055 case ACTIVE: 7056 mState = STOPPING; 7057 break; 7058 case DESTROYED: 7059 return NO_ERROR; // simply ignore as we are being destroyed 7060 } 7061 for (size_t i = 1; i < mHandles.size(); i++) { 7062 sp<EffectHandle> h = mHandles[i].promote(); 7063 if (h != 0) { 7064 h->setEnabled(enabled); 7065 } 7066 } 7067 } 7068 return NO_ERROR; 7069} 7070 7071bool AudioFlinger::EffectModule::isEnabled() const 7072{ 7073 switch (mState) { 7074 case RESTART: 7075 case STARTING: 7076 case ACTIVE: 7077 return true; 7078 case IDLE: 7079 case STOPPING: 7080 case STOPPED: 7081 case DESTROYED: 7082 default: 7083 return false; 7084 } 7085} 7086 7087bool AudioFlinger::EffectModule::isProcessEnabled() const 7088{ 7089 switch (mState) { 7090 case RESTART: 7091 case ACTIVE: 7092 case STOPPING: 7093 case STOPPED: 7094 return true; 7095 case IDLE: 7096 case STARTING: 7097 case DESTROYED: 7098 default: 7099 return false; 7100 } 7101} 7102 7103status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7104{ 7105 Mutex::Autolock _l(mLock); 7106 status_t status = NO_ERROR; 7107 7108 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7109 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7110 if (isProcessEnabled() && 7111 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7112 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7113 status_t cmdStatus; 7114 uint32_t volume[2]; 7115 uint32_t *pVolume = NULL; 7116 uint32_t size = sizeof(volume); 7117 volume[0] = *left; 7118 volume[1] = *right; 7119 if (controller) { 7120 pVolume = volume; 7121 } 7122 status = (*mEffectInterface)->command(mEffectInterface, 7123 EFFECT_CMD_SET_VOLUME, 7124 size, 7125 volume, 7126 &size, 7127 pVolume); 7128 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7129 *left = volume[0]; 7130 *right = volume[1]; 7131 } 7132 } 7133 return status; 7134} 7135 7136status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7137{ 7138 Mutex::Autolock _l(mLock); 7139 status_t status = NO_ERROR; 7140 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7141 // audio pre processing modules on RecordThread can receive both output and 7142 // input device indication in the same call 7143 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7144 if (dev) { 7145 status_t cmdStatus; 7146 uint32_t size = sizeof(status_t); 7147 7148 status = (*mEffectInterface)->command(mEffectInterface, 7149 EFFECT_CMD_SET_DEVICE, 7150 sizeof(uint32_t), 7151 &dev, 7152 &size, 7153 &cmdStatus); 7154 if (status == NO_ERROR) { 7155 status = cmdStatus; 7156 } 7157 } 7158 dev = device & AUDIO_DEVICE_IN_ALL; 7159 if (dev) { 7160 status_t cmdStatus; 7161 uint32_t size = sizeof(status_t); 7162 7163 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7164 EFFECT_CMD_SET_INPUT_DEVICE, 7165 sizeof(uint32_t), 7166 &dev, 7167 &size, 7168 &cmdStatus); 7169 if (status2 == NO_ERROR) { 7170 status2 = cmdStatus; 7171 } 7172 if (status == NO_ERROR) { 7173 status = status2; 7174 } 7175 } 7176 } 7177 return status; 7178} 7179 7180status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7181{ 7182 Mutex::Autolock _l(mLock); 7183 status_t status = NO_ERROR; 7184 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7185 status_t cmdStatus; 7186 uint32_t size = sizeof(status_t); 7187 status = (*mEffectInterface)->command(mEffectInterface, 7188 EFFECT_CMD_SET_AUDIO_MODE, 7189 sizeof(audio_mode_t), 7190 &mode, 7191 &size, 7192 &cmdStatus); 7193 if (status == NO_ERROR) { 7194 status = cmdStatus; 7195 } 7196 } 7197 return status; 7198} 7199 7200void AudioFlinger::EffectModule::setSuspended(bool suspended) 7201{ 7202 Mutex::Autolock _l(mLock); 7203 mSuspended = suspended; 7204} 7205 7206bool AudioFlinger::EffectModule::suspended() const 7207{ 7208 Mutex::Autolock _l(mLock); 7209 return mSuspended; 7210} 7211 7212status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7213{ 7214 const size_t SIZE = 256; 7215 char buffer[SIZE]; 7216 String8 result; 7217 7218 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7219 result.append(buffer); 7220 7221 bool locked = tryLock(mLock); 7222 // failed to lock - AudioFlinger is probably deadlocked 7223 if (!locked) { 7224 result.append("\t\tCould not lock Fx mutex:\n"); 7225 } 7226 7227 result.append("\t\tSession Status State Engine:\n"); 7228 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7229 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7230 result.append(buffer); 7231 7232 result.append("\t\tDescriptor:\n"); 7233 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7234 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7235 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7236 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7237 result.append(buffer); 7238 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7239 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7240 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7241 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7242 result.append(buffer); 7243 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7244 mDescriptor.apiVersion, 7245 mDescriptor.flags); 7246 result.append(buffer); 7247 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7248 mDescriptor.name); 7249 result.append(buffer); 7250 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7251 mDescriptor.implementor); 7252 result.append(buffer); 7253 7254 result.append("\t\t- Input configuration:\n"); 7255 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7256 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7257 (uint32_t)mConfig.inputCfg.buffer.raw, 7258 mConfig.inputCfg.buffer.frameCount, 7259 mConfig.inputCfg.samplingRate, 7260 mConfig.inputCfg.channels, 7261 mConfig.inputCfg.format); 7262 result.append(buffer); 7263 7264 result.append("\t\t- Output configuration:\n"); 7265 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7266 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7267 (uint32_t)mConfig.outputCfg.buffer.raw, 7268 mConfig.outputCfg.buffer.frameCount, 7269 mConfig.outputCfg.samplingRate, 7270 mConfig.outputCfg.channels, 7271 mConfig.outputCfg.format); 7272 result.append(buffer); 7273 7274 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7275 result.append(buffer); 7276 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7277 for (size_t i = 0; i < mHandles.size(); ++i) { 7278 sp<EffectHandle> handle = mHandles[i].promote(); 7279 if (handle != 0) { 7280 handle->dump(buffer, SIZE); 7281 result.append(buffer); 7282 } 7283 } 7284 7285 result.append("\n"); 7286 7287 write(fd, result.string(), result.length()); 7288 7289 if (locked) { 7290 mLock.unlock(); 7291 } 7292 7293 return NO_ERROR; 7294} 7295 7296// ---------------------------------------------------------------------------- 7297// EffectHandle implementation 7298// ---------------------------------------------------------------------------- 7299 7300#undef LOG_TAG 7301#define LOG_TAG "AudioFlinger::EffectHandle" 7302 7303AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7304 const sp<AudioFlinger::Client>& client, 7305 const sp<IEffectClient>& effectClient, 7306 int32_t priority) 7307 : BnEffect(), 7308 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7309 mPriority(priority), mHasControl(false), mEnabled(false) 7310{ 7311 ALOGV("constructor %p", this); 7312 7313 if (client == 0) { 7314 return; 7315 } 7316 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7317 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7318 if (mCblkMemory != 0) { 7319 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7320 7321 if (mCblk != NULL) { 7322 new(mCblk) effect_param_cblk_t(); 7323 mBuffer = (uint8_t *)mCblk + bufOffset; 7324 } 7325 } else { 7326 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7327 return; 7328 } 7329} 7330 7331AudioFlinger::EffectHandle::~EffectHandle() 7332{ 7333 ALOGV("Destructor %p", this); 7334 disconnect(false); 7335 ALOGV("Destructor DONE %p", this); 7336} 7337 7338status_t AudioFlinger::EffectHandle::enable() 7339{ 7340 ALOGV("enable %p", this); 7341 if (!mHasControl) return INVALID_OPERATION; 7342 if (mEffect == 0) return DEAD_OBJECT; 7343 7344 if (mEnabled) { 7345 return NO_ERROR; 7346 } 7347 7348 mEnabled = true; 7349 7350 sp<ThreadBase> thread = mEffect->thread().promote(); 7351 if (thread != 0) { 7352 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7353 } 7354 7355 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7356 if (mEffect->suspended()) { 7357 return NO_ERROR; 7358 } 7359 7360 status_t status = mEffect->setEnabled(true); 7361 if (status != NO_ERROR) { 7362 if (thread != 0) { 7363 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7364 } 7365 mEnabled = false; 7366 } 7367 return status; 7368} 7369 7370status_t AudioFlinger::EffectHandle::disable() 7371{ 7372 ALOGV("disable %p", this); 7373 if (!mHasControl) return INVALID_OPERATION; 7374 if (mEffect == 0) return DEAD_OBJECT; 7375 7376 if (!mEnabled) { 7377 return NO_ERROR; 7378 } 7379 mEnabled = false; 7380 7381 if (mEffect->suspended()) { 7382 return NO_ERROR; 7383 } 7384 7385 status_t status = mEffect->setEnabled(false); 7386 7387 sp<ThreadBase> thread = mEffect->thread().promote(); 7388 if (thread != 0) { 7389 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7390 } 7391 7392 return status; 7393} 7394 7395void AudioFlinger::EffectHandle::disconnect() 7396{ 7397 disconnect(true); 7398} 7399 7400void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7401{ 7402 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7403 if (mEffect == 0) { 7404 return; 7405 } 7406 mEffect->disconnect(this, unpinIfLast); 7407 7408 if (mHasControl && mEnabled) { 7409 sp<ThreadBase> thread = mEffect->thread().promote(); 7410 if (thread != 0) { 7411 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7412 } 7413 } 7414 7415 // release sp on module => module destructor can be called now 7416 mEffect.clear(); 7417 if (mClient != 0) { 7418 if (mCblk != NULL) { 7419 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7420 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7421 } 7422 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7423 // Client destructor must run with AudioFlinger mutex locked 7424 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7425 mClient.clear(); 7426 } 7427} 7428 7429status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7430 uint32_t cmdSize, 7431 void *pCmdData, 7432 uint32_t *replySize, 7433 void *pReplyData) 7434{ 7435// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7436// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7437 7438 // only get parameter command is permitted for applications not controlling the effect 7439 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7440 return INVALID_OPERATION; 7441 } 7442 if (mEffect == 0) return DEAD_OBJECT; 7443 if (mClient == 0) return INVALID_OPERATION; 7444 7445 // handle commands that are not forwarded transparently to effect engine 7446 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7447 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7448 // no risk to block the whole media server process or mixer threads is we are stuck here 7449 Mutex::Autolock _l(mCblk->lock); 7450 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7451 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7452 mCblk->serverIndex = 0; 7453 mCblk->clientIndex = 0; 7454 return BAD_VALUE; 7455 } 7456 status_t status = NO_ERROR; 7457 while (mCblk->serverIndex < mCblk->clientIndex) { 7458 int reply; 7459 uint32_t rsize = sizeof(int); 7460 int *p = (int *)(mBuffer + mCblk->serverIndex); 7461 int size = *p++; 7462 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7463 ALOGW("command(): invalid parameter block size"); 7464 break; 7465 } 7466 effect_param_t *param = (effect_param_t *)p; 7467 if (param->psize == 0 || param->vsize == 0) { 7468 ALOGW("command(): null parameter or value size"); 7469 mCblk->serverIndex += size; 7470 continue; 7471 } 7472 uint32_t psize = sizeof(effect_param_t) + 7473 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7474 param->vsize; 7475 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7476 psize, 7477 p, 7478 &rsize, 7479 &reply); 7480 // stop at first error encountered 7481 if (ret != NO_ERROR) { 7482 status = ret; 7483 *(int *)pReplyData = reply; 7484 break; 7485 } else if (reply != NO_ERROR) { 7486 *(int *)pReplyData = reply; 7487 break; 7488 } 7489 mCblk->serverIndex += size; 7490 } 7491 mCblk->serverIndex = 0; 7492 mCblk->clientIndex = 0; 7493 return status; 7494 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7495 *(int *)pReplyData = NO_ERROR; 7496 return enable(); 7497 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7498 *(int *)pReplyData = NO_ERROR; 7499 return disable(); 7500 } 7501 7502 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7503} 7504 7505void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7506{ 7507 ALOGV("setControl %p control %d", this, hasControl); 7508 7509 mHasControl = hasControl; 7510 mEnabled = enabled; 7511 7512 if (signal && mEffectClient != 0) { 7513 mEffectClient->controlStatusChanged(hasControl); 7514 } 7515} 7516 7517void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7518 uint32_t cmdSize, 7519 void *pCmdData, 7520 uint32_t replySize, 7521 void *pReplyData) 7522{ 7523 if (mEffectClient != 0) { 7524 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7525 } 7526} 7527 7528 7529 7530void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7531{ 7532 if (mEffectClient != 0) { 7533 mEffectClient->enableStatusChanged(enabled); 7534 } 7535} 7536 7537status_t AudioFlinger::EffectHandle::onTransact( 7538 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7539{ 7540 return BnEffect::onTransact(code, data, reply, flags); 7541} 7542 7543 7544void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7545{ 7546 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7547 7548 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7549 (mClient == 0) ? getpid_cached : mClient->pid(), 7550 mPriority, 7551 mHasControl, 7552 !locked, 7553 mCblk ? mCblk->clientIndex : 0, 7554 mCblk ? mCblk->serverIndex : 0 7555 ); 7556 7557 if (locked) { 7558 mCblk->lock.unlock(); 7559 } 7560} 7561 7562#undef LOG_TAG 7563#define LOG_TAG "AudioFlinger::EffectChain" 7564 7565AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7566 int sessionId) 7567 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7568 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7569 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7570{ 7571 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7572 if (thread == NULL) { 7573 return; 7574 } 7575 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7576 thread->frameCount(); 7577} 7578 7579AudioFlinger::EffectChain::~EffectChain() 7580{ 7581 if (mOwnInBuffer) { 7582 delete mInBuffer; 7583 } 7584 7585} 7586 7587// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7588sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7589{ 7590 size_t size = mEffects.size(); 7591 7592 for (size_t i = 0; i < size; i++) { 7593 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7594 return mEffects[i]; 7595 } 7596 } 7597 return 0; 7598} 7599 7600// getEffectFromId_l() must be called with ThreadBase::mLock held 7601sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7602{ 7603 size_t size = mEffects.size(); 7604 7605 for (size_t i = 0; i < size; i++) { 7606 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7607 if (id == 0 || mEffects[i]->id() == id) { 7608 return mEffects[i]; 7609 } 7610 } 7611 return 0; 7612} 7613 7614// getEffectFromType_l() must be called with ThreadBase::mLock held 7615sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7616 const effect_uuid_t *type) 7617{ 7618 size_t size = mEffects.size(); 7619 7620 for (size_t i = 0; i < size; i++) { 7621 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7622 return mEffects[i]; 7623 } 7624 } 7625 return 0; 7626} 7627 7628// Must be called with EffectChain::mLock locked 7629void AudioFlinger::EffectChain::process_l() 7630{ 7631 sp<ThreadBase> thread = mThread.promote(); 7632 if (thread == 0) { 7633 ALOGW("process_l(): cannot promote mixer thread"); 7634 return; 7635 } 7636 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7637 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7638 // always process effects unless no more tracks are on the session and the effect tail 7639 // has been rendered 7640 bool doProcess = true; 7641 if (!isGlobalSession) { 7642 bool tracksOnSession = (trackCnt() != 0); 7643 7644 if (!tracksOnSession && mTailBufferCount == 0) { 7645 doProcess = false; 7646 } 7647 7648 if (activeTrackCnt() == 0) { 7649 // if no track is active and the effect tail has not been rendered, 7650 // the input buffer must be cleared here as the mixer process will not do it 7651 if (tracksOnSession || mTailBufferCount > 0) { 7652 size_t numSamples = thread->frameCount() * thread->channelCount(); 7653 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7654 if (mTailBufferCount > 0) { 7655 mTailBufferCount--; 7656 } 7657 } 7658 } 7659 } 7660 7661 size_t size = mEffects.size(); 7662 if (doProcess) { 7663 for (size_t i = 0; i < size; i++) { 7664 mEffects[i]->process(); 7665 } 7666 } 7667 for (size_t i = 0; i < size; i++) { 7668 mEffects[i]->updateState(); 7669 } 7670} 7671 7672// addEffect_l() must be called with PlaybackThread::mLock held 7673status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7674{ 7675 effect_descriptor_t desc = effect->desc(); 7676 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7677 7678 Mutex::Autolock _l(mLock); 7679 effect->setChain(this); 7680 sp<ThreadBase> thread = mThread.promote(); 7681 if (thread == 0) { 7682 return NO_INIT; 7683 } 7684 effect->setThread(thread); 7685 7686 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7687 // Auxiliary effects are inserted at the beginning of mEffects vector as 7688 // they are processed first and accumulated in chain input buffer 7689 mEffects.insertAt(effect, 0); 7690 7691 // the input buffer for auxiliary effect contains mono samples in 7692 // 32 bit format. This is to avoid saturation in AudoMixer 7693 // accumulation stage. Saturation is done in EffectModule::process() before 7694 // calling the process in effect engine 7695 size_t numSamples = thread->frameCount(); 7696 int32_t *buffer = new int32_t[numSamples]; 7697 memset(buffer, 0, numSamples * sizeof(int32_t)); 7698 effect->setInBuffer((int16_t *)buffer); 7699 // auxiliary effects output samples to chain input buffer for further processing 7700 // by insert effects 7701 effect->setOutBuffer(mInBuffer); 7702 } else { 7703 // Insert effects are inserted at the end of mEffects vector as they are processed 7704 // after track and auxiliary effects. 7705 // Insert effect order as a function of indicated preference: 7706 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7707 // another effect is present 7708 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7709 // last effect claiming first position 7710 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7711 // first effect claiming last position 7712 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7713 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7714 // already present 7715 7716 size_t size = mEffects.size(); 7717 size_t idx_insert = size; 7718 ssize_t idx_insert_first = -1; 7719 ssize_t idx_insert_last = -1; 7720 7721 for (size_t i = 0; i < size; i++) { 7722 effect_descriptor_t d = mEffects[i]->desc(); 7723 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7724 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7725 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7726 // check invalid effect chaining combinations 7727 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7728 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7729 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7730 return INVALID_OPERATION; 7731 } 7732 // remember position of first insert effect and by default 7733 // select this as insert position for new effect 7734 if (idx_insert == size) { 7735 idx_insert = i; 7736 } 7737 // remember position of last insert effect claiming 7738 // first position 7739 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7740 idx_insert_first = i; 7741 } 7742 // remember position of first insert effect claiming 7743 // last position 7744 if (iPref == EFFECT_FLAG_INSERT_LAST && 7745 idx_insert_last == -1) { 7746 idx_insert_last = i; 7747 } 7748 } 7749 } 7750 7751 // modify idx_insert from first position if needed 7752 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7753 if (idx_insert_last != -1) { 7754 idx_insert = idx_insert_last; 7755 } else { 7756 idx_insert = size; 7757 } 7758 } else { 7759 if (idx_insert_first != -1) { 7760 idx_insert = idx_insert_first + 1; 7761 } 7762 } 7763 7764 // always read samples from chain input buffer 7765 effect->setInBuffer(mInBuffer); 7766 7767 // if last effect in the chain, output samples to chain 7768 // output buffer, otherwise to chain input buffer 7769 if (idx_insert == size) { 7770 if (idx_insert != 0) { 7771 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7772 mEffects[idx_insert-1]->configure(); 7773 } 7774 effect->setOutBuffer(mOutBuffer); 7775 } else { 7776 effect->setOutBuffer(mInBuffer); 7777 } 7778 mEffects.insertAt(effect, idx_insert); 7779 7780 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7781 } 7782 effect->configure(); 7783 return NO_ERROR; 7784} 7785 7786// removeEffect_l() must be called with PlaybackThread::mLock held 7787size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7788{ 7789 Mutex::Autolock _l(mLock); 7790 size_t size = mEffects.size(); 7791 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7792 7793 for (size_t i = 0; i < size; i++) { 7794 if (effect == mEffects[i]) { 7795 // calling stop here will remove pre-processing effect from the audio HAL. 7796 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7797 // the middle of a read from audio HAL 7798 if (mEffects[i]->state() == EffectModule::ACTIVE || 7799 mEffects[i]->state() == EffectModule::STOPPING) { 7800 mEffects[i]->stop(); 7801 } 7802 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7803 delete[] effect->inBuffer(); 7804 } else { 7805 if (i == size - 1 && i != 0) { 7806 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7807 mEffects[i - 1]->configure(); 7808 } 7809 } 7810 mEffects.removeAt(i); 7811 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7812 break; 7813 } 7814 } 7815 7816 return mEffects.size(); 7817} 7818 7819// setDevice_l() must be called with PlaybackThread::mLock held 7820void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7821{ 7822 size_t size = mEffects.size(); 7823 for (size_t i = 0; i < size; i++) { 7824 mEffects[i]->setDevice(device); 7825 } 7826} 7827 7828// setMode_l() must be called with PlaybackThread::mLock held 7829void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7830{ 7831 size_t size = mEffects.size(); 7832 for (size_t i = 0; i < size; i++) { 7833 mEffects[i]->setMode(mode); 7834 } 7835} 7836 7837// setVolume_l() must be called with PlaybackThread::mLock held 7838bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7839{ 7840 uint32_t newLeft = *left; 7841 uint32_t newRight = *right; 7842 bool hasControl = false; 7843 int ctrlIdx = -1; 7844 size_t size = mEffects.size(); 7845 7846 // first update volume controller 7847 for (size_t i = size; i > 0; i--) { 7848 if (mEffects[i - 1]->isProcessEnabled() && 7849 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7850 ctrlIdx = i - 1; 7851 hasControl = true; 7852 break; 7853 } 7854 } 7855 7856 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7857 if (hasControl) { 7858 *left = mNewLeftVolume; 7859 *right = mNewRightVolume; 7860 } 7861 return hasControl; 7862 } 7863 7864 mVolumeCtrlIdx = ctrlIdx; 7865 mLeftVolume = newLeft; 7866 mRightVolume = newRight; 7867 7868 // second get volume update from volume controller 7869 if (ctrlIdx >= 0) { 7870 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7871 mNewLeftVolume = newLeft; 7872 mNewRightVolume = newRight; 7873 } 7874 // then indicate volume to all other effects in chain. 7875 // Pass altered volume to effects before volume controller 7876 // and requested volume to effects after controller 7877 uint32_t lVol = newLeft; 7878 uint32_t rVol = newRight; 7879 7880 for (size_t i = 0; i < size; i++) { 7881 if ((int)i == ctrlIdx) continue; 7882 // this also works for ctrlIdx == -1 when there is no volume controller 7883 if ((int)i > ctrlIdx) { 7884 lVol = *left; 7885 rVol = *right; 7886 } 7887 mEffects[i]->setVolume(&lVol, &rVol, false); 7888 } 7889 *left = newLeft; 7890 *right = newRight; 7891 7892 return hasControl; 7893} 7894 7895status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7896{ 7897 const size_t SIZE = 256; 7898 char buffer[SIZE]; 7899 String8 result; 7900 7901 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7902 result.append(buffer); 7903 7904 bool locked = tryLock(mLock); 7905 // failed to lock - AudioFlinger is probably deadlocked 7906 if (!locked) { 7907 result.append("\tCould not lock mutex:\n"); 7908 } 7909 7910 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7911 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7912 mEffects.size(), 7913 (uint32_t)mInBuffer, 7914 (uint32_t)mOutBuffer, 7915 mActiveTrackCnt); 7916 result.append(buffer); 7917 write(fd, result.string(), result.size()); 7918 7919 for (size_t i = 0; i < mEffects.size(); ++i) { 7920 sp<EffectModule> effect = mEffects[i]; 7921 if (effect != 0) { 7922 effect->dump(fd, args); 7923 } 7924 } 7925 7926 if (locked) { 7927 mLock.unlock(); 7928 } 7929 7930 return NO_ERROR; 7931} 7932 7933// must be called with ThreadBase::mLock held 7934void AudioFlinger::EffectChain::setEffectSuspended_l( 7935 const effect_uuid_t *type, bool suspend) 7936{ 7937 sp<SuspendedEffectDesc> desc; 7938 // use effect type UUID timelow as key as there is no real risk of identical 7939 // timeLow fields among effect type UUIDs. 7940 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7941 if (suspend) { 7942 if (index >= 0) { 7943 desc = mSuspendedEffects.valueAt(index); 7944 } else { 7945 desc = new SuspendedEffectDesc(); 7946 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7947 mSuspendedEffects.add(type->timeLow, desc); 7948 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7949 } 7950 if (desc->mRefCount++ == 0) { 7951 sp<EffectModule> effect = getEffectIfEnabled(type); 7952 if (effect != 0) { 7953 desc->mEffect = effect; 7954 effect->setSuspended(true); 7955 effect->setEnabled(false); 7956 } 7957 } 7958 } else { 7959 if (index < 0) { 7960 return; 7961 } 7962 desc = mSuspendedEffects.valueAt(index); 7963 if (desc->mRefCount <= 0) { 7964 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7965 desc->mRefCount = 1; 7966 } 7967 if (--desc->mRefCount == 0) { 7968 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7969 if (desc->mEffect != 0) { 7970 sp<EffectModule> effect = desc->mEffect.promote(); 7971 if (effect != 0) { 7972 effect->setSuspended(false); 7973 sp<EffectHandle> handle = effect->controlHandle(); 7974 if (handle != 0) { 7975 effect->setEnabled(handle->enabled()); 7976 } 7977 } 7978 desc->mEffect.clear(); 7979 } 7980 mSuspendedEffects.removeItemsAt(index); 7981 } 7982 } 7983} 7984 7985// must be called with ThreadBase::mLock held 7986void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7987{ 7988 sp<SuspendedEffectDesc> desc; 7989 7990 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7991 if (suspend) { 7992 if (index >= 0) { 7993 desc = mSuspendedEffects.valueAt(index); 7994 } else { 7995 desc = new SuspendedEffectDesc(); 7996 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7997 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7998 } 7999 if (desc->mRefCount++ == 0) { 8000 Vector< sp<EffectModule> > effects; 8001 getSuspendEligibleEffects(effects); 8002 for (size_t i = 0; i < effects.size(); i++) { 8003 setEffectSuspended_l(&effects[i]->desc().type, true); 8004 } 8005 } 8006 } else { 8007 if (index < 0) { 8008 return; 8009 } 8010 desc = mSuspendedEffects.valueAt(index); 8011 if (desc->mRefCount <= 0) { 8012 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8013 desc->mRefCount = 1; 8014 } 8015 if (--desc->mRefCount == 0) { 8016 Vector<const effect_uuid_t *> types; 8017 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8018 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8019 continue; 8020 } 8021 types.add(&mSuspendedEffects.valueAt(i)->mType); 8022 } 8023 for (size_t i = 0; i < types.size(); i++) { 8024 setEffectSuspended_l(types[i], false); 8025 } 8026 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8027 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8028 } 8029 } 8030} 8031 8032 8033// The volume effect is used for automated tests only 8034#ifndef OPENSL_ES_H_ 8035static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8036 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8037const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8038#endif //OPENSL_ES_H_ 8039 8040bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8041{ 8042 // auxiliary effects and visualizer are never suspended on output mix 8043 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8044 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8045 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8046 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8047 return false; 8048 } 8049 return true; 8050} 8051 8052void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8053{ 8054 effects.clear(); 8055 for (size_t i = 0; i < mEffects.size(); i++) { 8056 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8057 effects.add(mEffects[i]); 8058 } 8059 } 8060} 8061 8062sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8063 const effect_uuid_t *type) 8064{ 8065 sp<EffectModule> effect = getEffectFromType_l(type); 8066 return effect != 0 && effect->isEnabled() ? effect : 0; 8067} 8068 8069void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8070 bool enabled) 8071{ 8072 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8073 if (enabled) { 8074 if (index < 0) { 8075 // if the effect is not suspend check if all effects are suspended 8076 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8077 if (index < 0) { 8078 return; 8079 } 8080 if (!isEffectEligibleForSuspend(effect->desc())) { 8081 return; 8082 } 8083 setEffectSuspended_l(&effect->desc().type, enabled); 8084 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8085 if (index < 0) { 8086 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8087 return; 8088 } 8089 } 8090 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8091 effect->desc().type.timeLow); 8092 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8093 // if effect is requested to suspended but was not yet enabled, supend it now. 8094 if (desc->mEffect == 0) { 8095 desc->mEffect = effect; 8096 effect->setEnabled(false); 8097 effect->setSuspended(true); 8098 } 8099 } else { 8100 if (index < 0) { 8101 return; 8102 } 8103 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8104 effect->desc().type.timeLow); 8105 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8106 desc->mEffect.clear(); 8107 effect->setSuspended(false); 8108 } 8109} 8110 8111#undef LOG_TAG 8112#define LOG_TAG "AudioFlinger" 8113 8114// ---------------------------------------------------------------------------- 8115 8116status_t AudioFlinger::onTransact( 8117 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8118{ 8119 return BnAudioFlinger::onTransact(code, data, reply, flags); 8120} 8121 8122}; // namespace android 8123