AudioFlinger.cpp revision e9dd0176933d6233916c84e18f3e8c0d644ca05d
16c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik/* //device/include/server/AudioFlinger/AudioFlinger.cpp 26c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** 36c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** Copyright 2007, The Android Open Source Project 46c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** 56c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** Licensed under the Apache License, Version 2.0 (the "License"); 66c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** you may not use this file except in compliance with the License. 76c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** You may obtain a copy of the License at 86c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** 96c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** http://www.apache.org/licenses/LICENSE-2.0 106c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** 116c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** Unless required by applicable law or agreed to in writing, software 126c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** distributed under the License is distributed on an "AS IS" BASIS, 136c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 146c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** See the License for the specific language governing permissions and 156c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik** limitations under the License. 166c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik*/ 175e00c7ce063116c11315639f0035aca8ad73e8ccChris Craik 186c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 190519c810a56bded1284fcb2ae40f438878c6585fChris Craik#define LOG_TAG "AudioFlinger" 206c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik//#define LOG_NDEBUG 0 21922d3a7f6f8c1c05a996ee3e91e8cbadfff560c9Chris Craik 226c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <math.h> 23922d3a7f6f8c1c05a996ee3e91e8cbadfff560c9Chris Craik#include <signal.h> 246c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <sys/time.h> 256c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <sys/resource.h> 266c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 276c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <binder/IPCThreadState.h> 286c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <binder/IServiceManager.h> 296c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <utils/Log.h> 306c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <binder/Parcel.h> 316c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <binder/IPCThreadState.h> 32031888744e24b5c7243ac99ec98b78aff5db1c78Chris Craik#include <utils/String16.h> 330519c810a56bded1284fcb2ae40f438878c6585fChris Craik#include <utils/threads.h> 34922d3a7f6f8c1c05a996ee3e91e8cbadfff560c9Chris Craik#include <utils/Atomic.h> 35031888744e24b5c7243ac99ec98b78aff5db1c78Chris Craik 366c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <cutils/bitops.h> 376c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <cutils/properties.h> 386c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <cutils/compiler.h> 396c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 406c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <media/IMediaPlayerService.h> 416c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik#include <media/IMediaDeathNotifier.h> 4253e51e4aa933f9603587e1780f446c18816bf9beChris Craik 4353e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <private/media/AudioTrackShared.h> 4453e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <private/media/AudioEffectShared.h> 45138c21fbec12bead3c7ca1f181c3fd35542ccb00Chris Craik 4653e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <system/audio.h> 47138c21fbec12bead3c7ca1f181c3fd35542ccb00Chris Craik#include <hardware/audio.h> 48138c21fbec12bead3c7ca1f181c3fd35542ccb00Chris Craik 4953e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include "AudioMixer.h" 50138c21fbec12bead3c7ca1f181c3fd35542ccb00Chris Craik#include "AudioFlinger.h" 5153e51e4aa933f9603587e1780f446c18816bf9beChris Craik 52138c21fbec12bead3c7ca1f181c3fd35542ccb00Chris Craik#include <media/EffectsFactoryApi.h> 5353e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <audio_effects/effect_visualizer.h> 5453e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <audio_effects/effect_ns.h> 5553e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <audio_effects/effect_aec.h> 5653e51e4aa933f9603587e1780f446c18816bf9beChris Craik 5753e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <audio_utils/primitives.h> 5853e51e4aa933f9603587e1780f446c18816bf9beChris Craik 5953e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <cpustats/ThreadCpuUsage.h> 6053e51e4aa933f9603587e1780f446c18816bf9beChris Craik#include <powermanager/PowerManager.h> 6153e51e4aa933f9603587e1780f446c18816bf9beChris Craik// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 6253e51e4aa933f9603587e1780f446c18816bf9beChris Craik 6353e51e4aa933f9603587e1780f446c18816bf9beChris Craik// ---------------------------------------------------------------------------- 6453e51e4aa933f9603587e1780f446c18816bf9beChris Craik 6553e51e4aa933f9603587e1780f446c18816bf9beChris Craik 6653e51e4aa933f9603587e1780f446c18816bf9beChris Craiknamespace android { 6753e51e4aa933f9603587e1780f446c18816bf9beChris Craik 6853e51e4aa933f9603587e1780f446c18816bf9beChris Craikstatic const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 695e00c7ce063116c11315639f0035aca8ad73e8ccChris Craikstatic const char kHardwareLockedString[] = "Hardware lock is taken\n"; 70e4db79de127cfe961195f52907af8451026eaa20Chris Craik 7153e51e4aa933f9603587e1780f446c18816bf9beChris Craik//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 726c15ffa196fc9b7724c189d833c3435d8db12266Chris Craikstatic const float MAX_GAIN = 4096.0f; 736c15ffa196fc9b7724c189d833c3435d8db12266Chris Craikstatic const uint32_t MAX_GAIN_INT = 0x1000; 746c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 750519c810a56bded1284fcb2ae40f438878c6585fChris Craik// retry counts for buffer fill timeout 766c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik// 50 * ~20msecs = 1 second 776c15ffa196fc9b7724c189d833c3435d8db12266Chris Craikstatic const int8_t kMaxTrackRetries = 50; 786c15ffa196fc9b7724c189d833c3435d8db12266Chris Craikstatic const int8_t kMaxTrackStartupRetries = 50; 796c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik// allow less retry attempts on direct output thread. 800519c810a56bded1284fcb2ae40f438878c6585fChris Craik// direct outputs can be a scarce resource in audio hardware and should 810519c810a56bded1284fcb2ae40f438878c6585fChris Craik// be released as quickly as possible. 820519c810a56bded1284fcb2ae40f438878c6585fChris Craikstatic const int8_t kMaxTrackRetriesDirect = 2; 830519c810a56bded1284fcb2ae40f438878c6585fChris Craik 840519c810a56bded1284fcb2ae40f438878c6585fChris Craikstatic const int kDumpLockRetries = 50; 850519c810a56bded1284fcb2ae40f438878c6585fChris Craikstatic const int kDumpLockSleepUs = 20000; 866c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 876c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik// don't warn about blocked writes or record buffer overflows more often than this 88f42bf3e2573bccb01babec448f925e9395bf224csergeyvstatic const nsecs_t kWarningThrottleNs = seconds(5); 89f42bf3e2573bccb01babec448f925e9395bf224csergeyv 90f42bf3e2573bccb01babec448f925e9395bf224csergeyv// RecordThread loop sleep time upon application overrun or audio HAL read error 916c15ffa196fc9b7724c189d833c3435d8db12266Chris Craikstatic const int kRecordThreadSleepUs = 5000; 926c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 93ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik// maximum time to wait for setParameters to complete 94ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craikstatic const nsecs_t kSetParametersTimeoutNs = seconds(2); 95ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik 96ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik// minimum sleep time for the mixer thread loop when tracks are active but in underrun 97ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craikstatic const uint32_t kMinThreadSleepTimeUs = 5000; 98ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik// maximum divider applied to the active sleep time in the mixer thread loop 99ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craikstatic const uint32_t kMaxThreadSleepTimeShift = 2; 100ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik 101ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik 102ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik// ---------------------------------------------------------------------------- 103ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik 104ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craikstatic bool recordingAllowed() { 105eb911c2b0e8edeb7595a98af4b9f1bd47de1381eChris Craik if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 106ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 107ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 108ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik return ok; 109ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik} 110ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craik 111ef2507439c08f4e9c4c9bba1c6243ca9df2ee827Chris Craikstatic bool settingsAllowed() { 1120519c810a56bded1284fcb2ae40f438878c6585fChris Craik if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 1130519c810a56bded1284fcb2ae40f438878c6585fChris Craik bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 1146c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 1156c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik return ok; 1166c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik} 1176c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 11830036092b40badecbe64d9c2bff4850132147f78Chris Craik// To collect the amplifier usage 119f27133df2d179c99d6bc1ae644af09e9153a0071Chris Craikstatic void addBatteryData(uint32_t params) { 120f27133df2d179c99d6bc1ae644af09e9153a0071Chris Craik sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 121f27133df2d179c99d6bc1ae644af09e9153a0071Chris Craik if (service == NULL) { 122f27133df2d179c99d6bc1ae644af09e9153a0071Chris Craik // it already logged 12326bf34200e40a0fa8c66366559aa016380cd8c6fChris Craik return; 124f27133df2d179c99d6bc1ae644af09e9153a0071Chris Craik } 1250519c810a56bded1284fcb2ae40f438878c6585fChris Craik 1260519c810a56bded1284fcb2ae40f438878c6585fChris Craik service->addBatteryData(params); 127117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik} 1286c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 129117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craikstatic int load_audio_interface(const char *if_name, const hw_module_t **mod, 130117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik audio_hw_device_t **dev) 131117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik{ 132117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik int rc; 133117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik 134117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik if (rc) 136117bdbcfa3e8306dad21e7e01fa71b00cdfa7265Chris Craik goto out; 137922d3a7f6f8c1c05a996ee3e91e8cbadfff560c9Chris Craik 138922d3a7f6f8c1c05a996ee3e91e8cbadfff560c9Chris Craik rc = audio_hw_device_open(*mod, dev); 1396c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 1406c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 1416c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik if (rc) 14253e51e4aa933f9603587e1780f446c18816bf9beChris Craik goto out; 14353e51e4aa933f9603587e1780f446c18816bf9beChris Craik 1446c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik return 0; 14553e51e4aa933f9603587e1780f446c18816bf9beChris Craik 14653e51e4aa933f9603587e1780f446c18816bf9beChris Craikout: 1476c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik *mod = NULL; 14853e51e4aa933f9603587e1780f446c18816bf9beChris Craik *dev = NULL; 14953e51e4aa933f9603587e1780f446c18816bf9beChris Craik return rc; 15053e51e4aa933f9603587e1780f446c18816bf9beChris Craik} 15153e51e4aa933f9603587e1780f446c18816bf9beChris Craik 15253e51e4aa933f9603587e1780f446c18816bf9beChris Craikstatic const char * const audio_interfaces[] = { 15353e51e4aa933f9603587e1780f446c18816bf9beChris Craik "primary", 1546c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik "a2dp", 1556c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik "usb", 156f42bf3e2573bccb01babec448f925e9395bf224csergeyv}; 15730036092b40badecbe64d9c2bff4850132147f78Chris Craik#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 15830036092b40badecbe64d9c2bff4850132147f78Chris Craik 15930036092b40badecbe64d9c2bff4850132147f78Chris Craik// ---------------------------------------------------------------------------- 16030036092b40badecbe64d9c2bff4850132147f78Chris Craik 16130036092b40badecbe64d9c2bff4850132147f78Chris CraikAudioFlinger::AudioFlinger() 16230036092b40badecbe64d9c2bff4850132147f78Chris Craik : BnAudioFlinger(), 1636c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 164031888744e24b5c7243ac99ec98b78aff5db1c78Chris Craik mBtNrecIsOff(false) 165031888744e24b5c7243ac99ec98b78aff5db1c78Chris Craik{ 1666c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik} 1676c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 16830036092b40badecbe64d9c2bff4850132147f78Chris Craikvoid AudioFlinger::onFirstRef() 1696c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik{ 1706c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik int rc = 0; 1716c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 1726c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik Mutex::Autolock _l(mLock); 1736c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 1746c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik /* TODO: move all this work into an Init() function */ 1756c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik mHardwareStatus = AUDIO_HW_IDLE; 1766c15ffa196fc9b7724c189d833c3435d8db12266Chris Craik 177 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 178 const hw_module_t *mod; 179 audio_hw_device_t *dev; 180 181 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 182 if (rc) 183 continue; 184 185 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 186 mod->name, mod->id); 187 mAudioHwDevs.push(dev); 188 189 if (!mPrimaryHardwareDev) { 190 mPrimaryHardwareDev = dev; 191 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 192 mod->name, mod->id, audio_interfaces[i]); 193 } 194 } 195 196 mHardwareStatus = AUDIO_HW_INIT; 197 198 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 199 ALOGE("Primary audio interface not found"); 200 return; 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 audio_hw_device_t *dev = mAudioHwDevs[i]; 205 206 mHardwareStatus = AUDIO_HW_INIT; 207 rc = dev->init_check(dev); 208 if (rc == 0) { 209 AutoMutex lock(mHardwareLock); 210 211 mMode = AUDIO_MODE_NORMAL; 212 mHardwareStatus = AUDIO_HW_SET_MODE; 213 dev->set_mode(dev, mMode); 214 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 215 dev->set_master_volume(dev, 1.0f); 216 mHardwareStatus = AUDIO_HW_IDLE; 217 } 218 } 219} 220 221status_t AudioFlinger::initCheck() const 222{ 223 Mutex::Autolock _l(mLock); 224 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 225 return NO_INIT; 226 return NO_ERROR; 227} 228 229AudioFlinger::~AudioFlinger() 230{ 231 int num_devs = mAudioHwDevs.size(); 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (int i = 0; i < num_devs; i++) { 243 audio_hw_device_t *dev = mAudioHwDevs[i]; 244 audio_hw_device_close(dev); 245 } 246 mAudioHwDevs.clear(); 247} 248 249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 250{ 251 /* first matching HW device is returned */ 252 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 253 audio_hw_device_t *dev = mAudioHwDevs[i]; 254 if ((dev->get_supported_devices(dev) & devices) == devices) 255 return dev; 256 } 257 return NULL; 258} 259 260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 261{ 262 const size_t SIZE = 256; 263 char buffer[SIZE]; 264 String8 result; 265 266 result.append("Clients:\n"); 267 for (size_t i = 0; i < mClients.size(); ++i) { 268 wp<Client> wClient = mClients.valueAt(i); 269 if (wClient != 0) { 270 sp<Client> client = wClient.promote(); 271 if (client != 0) { 272 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 273 result.append(buffer); 274 } 275 } 276 } 277 278 result.append("Global session refs:\n"); 279 result.append(" session pid cnt\n"); 280 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 281 AudioSessionRef *r = mAudioSessionRefs[i]; 282 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 283 result.append(buffer); 284 } 285 write(fd, result.string(), result.size()); 286 return NO_ERROR; 287} 288 289 290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 291{ 292 const size_t SIZE = 256; 293 char buffer[SIZE]; 294 String8 result; 295 hardware_call_state hardwareStatus = mHardwareStatus; 296 297 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300 return NO_ERROR; 301} 302 303status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 304{ 305 const size_t SIZE = 256; 306 char buffer[SIZE]; 307 String8 result; 308 snprintf(buffer, SIZE, "Permission Denial: " 309 "can't dump AudioFlinger from pid=%d, uid=%d\n", 310 IPCThreadState::self()->getCallingPid(), 311 IPCThreadState::self()->getCallingUid()); 312 result.append(buffer); 313 write(fd, result.string(), result.size()); 314 return NO_ERROR; 315} 316 317static bool tryLock(Mutex& mutex) 318{ 319 bool locked = false; 320 for (int i = 0; i < kDumpLockRetries; ++i) { 321 if (mutex.tryLock() == NO_ERROR) { 322 locked = true; 323 break; 324 } 325 usleep(kDumpLockSleepUs); 326 } 327 return locked; 328} 329 330status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 331{ 332 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 333 dumpPermissionDenial(fd, args); 334 } else { 335 // get state of hardware lock 336 bool hardwareLocked = tryLock(mHardwareLock); 337 if (!hardwareLocked) { 338 String8 result(kHardwareLockedString); 339 write(fd, result.string(), result.size()); 340 } else { 341 mHardwareLock.unlock(); 342 } 343 344 bool locked = tryLock(mLock); 345 346 // failed to lock - AudioFlinger is probably deadlocked 347 if (!locked) { 348 String8 result(kDeadlockedString); 349 write(fd, result.string(), result.size()); 350 } 351 352 dumpClients(fd, args); 353 dumpInternals(fd, args); 354 355 // dump playback threads 356 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 357 mPlaybackThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump record threads 361 for (size_t i = 0; i < mRecordThreads.size(); i++) { 362 mRecordThreads.valueAt(i)->dump(fd, args); 363 } 364 365 // dump all hardware devs 366 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 367 audio_hw_device_t *dev = mAudioHwDevs[i]; 368 dev->dump(dev, fd); 369 } 370 if (locked) mLock.unlock(); 371 } 372 return NO_ERROR; 373} 374 375 376// IAudioFlinger interface 377 378 379sp<IAudioTrack> AudioFlinger::createTrack( 380 pid_t pid, 381 audio_stream_type_t streamType, 382 uint32_t sampleRate, 383 audio_format_t format, 384 uint32_t channelMask, 385 int frameCount, 386 uint32_t flags, 387 const sp<IMemory>& sharedBuffer, 388 int output, 389 int *sessionId, 390 status_t *status) 391{ 392 sp<PlaybackThread::Track> track; 393 sp<TrackHandle> trackHandle; 394 sp<Client> client; 395 wp<Client> wclient; 396 status_t lStatus; 397 int lSessionId; 398 399 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 400 // but if someone uses binder directly they could bypass that and cause us to crash 401 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 402 ALOGE("createTrack() invalid stream type %d", streamType); 403 lStatus = BAD_VALUE; 404 goto Exit; 405 } 406 407 { 408 Mutex::Autolock _l(mLock); 409 PlaybackThread *thread = checkPlaybackThread_l(output); 410 PlaybackThread *effectThread = NULL; 411 if (thread == NULL) { 412 ALOGE("unknown output thread"); 413 lStatus = BAD_VALUE; 414 goto Exit; 415 } 416 417 wclient = mClients.valueFor(pid); 418 419 if (wclient != NULL) { 420 client = wclient.promote(); 421 } else { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 427 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 428 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 429 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 430 if (mPlaybackThreads.keyAt(i) != output) { 431 // prevent same audio session on different output threads 432 uint32_t sessions = t->hasAudioSession(*sessionId); 433 if (sessions & PlaybackThread::TRACK_SESSION) { 434 ALOGE("createTrack() session ID %d already in use", *sessionId); 435 lStatus = BAD_VALUE; 436 goto Exit; 437 } 438 // check if an effect with same session ID is waiting for a track to be created 439 if (sessions & PlaybackThread::EFFECT_SESSION) { 440 effectThread = t.get(); 441 } 442 } 443 } 444 lSessionId = *sessionId; 445 } else { 446 // if no audio session id is provided, create one here 447 lSessionId = nextUniqueId(); 448 if (sessionId != NULL) { 449 *sessionId = lSessionId; 450 } 451 } 452 ALOGV("createTrack() lSessionId: %d", lSessionId); 453 454 track = thread->createTrack_l(client, streamType, sampleRate, format, 455 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 456 457 // move effect chain to this output thread if an effect on same session was waiting 458 // for a track to be created 459 if (lStatus == NO_ERROR && effectThread != NULL) { 460 Mutex::Autolock _dl(thread->mLock); 461 Mutex::Autolock _sl(effectThread->mLock); 462 moveEffectChain_l(lSessionId, effectThread, thread, true); 463 } 464 } 465 if (lStatus == NO_ERROR) { 466 trackHandle = new TrackHandle(track); 467 } else { 468 // remove local strong reference to Client before deleting the Track so that the Client 469 // destructor is called by the TrackBase destructor with mLock held 470 client.clear(); 471 track.clear(); 472 } 473 474Exit: 475 if(status) { 476 *status = lStatus; 477 } 478 return trackHandle; 479} 480 481uint32_t AudioFlinger::sampleRate(int output) const 482{ 483 Mutex::Autolock _l(mLock); 484 PlaybackThread *thread = checkPlaybackThread_l(output); 485 if (thread == NULL) { 486 ALOGW("sampleRate() unknown thread %d", output); 487 return 0; 488 } 489 return thread->sampleRate(); 490} 491 492int AudioFlinger::channelCount(int output) const 493{ 494 Mutex::Autolock _l(mLock); 495 PlaybackThread *thread = checkPlaybackThread_l(output); 496 if (thread == NULL) { 497 ALOGW("channelCount() unknown thread %d", output); 498 return 0; 499 } 500 return thread->channelCount(); 501} 502 503audio_format_t AudioFlinger::format(int output) const 504{ 505 Mutex::Autolock _l(mLock); 506 PlaybackThread *thread = checkPlaybackThread_l(output); 507 if (thread == NULL) { 508 ALOGW("format() unknown thread %d", output); 509 return AUDIO_FORMAT_INVALID; 510 } 511 return thread->format(); 512} 513 514size_t AudioFlinger::frameCount(int output) const 515{ 516 Mutex::Autolock _l(mLock); 517 PlaybackThread *thread = checkPlaybackThread_l(output); 518 if (thread == NULL) { 519 ALOGW("frameCount() unknown thread %d", output); 520 return 0; 521 } 522 return thread->frameCount(); 523} 524 525uint32_t AudioFlinger::latency(int output) const 526{ 527 Mutex::Autolock _l(mLock); 528 PlaybackThread *thread = checkPlaybackThread_l(output); 529 if (thread == NULL) { 530 ALOGW("latency() unknown thread %d", output); 531 return 0; 532 } 533 return thread->latency(); 534} 535 536status_t AudioFlinger::setMasterVolume(float value) 537{ 538 status_t ret = initCheck(); 539 if (ret != NO_ERROR) { 540 return ret; 541 } 542 543 // check calling permissions 544 if (!settingsAllowed()) { 545 return PERMISSION_DENIED; 546 } 547 548 // when hw supports master volume, don't scale in sw mixer 549 { // scope for the lock 550 AutoMutex lock(mHardwareLock); 551 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 552 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 553 value = 1.0f; 554 } 555 mHardwareStatus = AUDIO_HW_IDLE; 556 } 557 558 Mutex::Autolock _l(mLock); 559 mMasterVolume = value; 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 561 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 562 563 return NO_ERROR; 564} 565 566status_t AudioFlinger::setMode(audio_mode_t mode) 567{ 568 status_t ret = initCheck(); 569 if (ret != NO_ERROR) { 570 return ret; 571 } 572 573 // check calling permissions 574 if (!settingsAllowed()) { 575 return PERMISSION_DENIED; 576 } 577 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 578 ALOGW("Illegal value: setMode(%d)", mode); 579 return BAD_VALUE; 580 } 581 582 { // scope for the lock 583 AutoMutex lock(mHardwareLock); 584 mHardwareStatus = AUDIO_HW_SET_MODE; 585 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 586 mHardwareStatus = AUDIO_HW_IDLE; 587 } 588 589 if (NO_ERROR == ret) { 590 Mutex::Autolock _l(mLock); 591 mMode = mode; 592 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 593 mPlaybackThreads.valueAt(i)->setMode(mode); 594 } 595 596 return ret; 597} 598 599status_t AudioFlinger::setMicMute(bool state) 600{ 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 AutoMutex lock(mHardwareLock); 612 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 613 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 614 mHardwareStatus = AUDIO_HW_IDLE; 615 return ret; 616} 617 618bool AudioFlinger::getMicMute() const 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return false; 623 } 624 625 bool state = AUDIO_MODE_INVALID; 626 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 627 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 628 mHardwareStatus = AUDIO_HW_IDLE; 629 return state; 630} 631 632status_t AudioFlinger::setMasterMute(bool muted) 633{ 634 // check calling permissions 635 if (!settingsAllowed()) { 636 return PERMISSION_DENIED; 637 } 638 639 Mutex::Autolock _l(mLock); 640 mMasterMute = muted; 641 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 643 644 return NO_ERROR; 645} 646 647float AudioFlinger::masterVolume() const 648{ 649 Mutex::Autolock _l(mLock); 650 return masterVolume_l(); 651} 652 653bool AudioFlinger::masterMute() const 654{ 655 Mutex::Autolock _l(mLock); 656 return masterMute_l(); 657} 658 659status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output) 660{ 661 // check calling permissions 662 if (!settingsAllowed()) { 663 return PERMISSION_DENIED; 664 } 665 666 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 667 ALOGE("setStreamVolume() invalid stream %d", stream); 668 return BAD_VALUE; 669 } 670 671 AutoMutex lock(mLock); 672 PlaybackThread *thread = NULL; 673 if (output) { 674 thread = checkPlaybackThread_l(output); 675 if (thread == NULL) { 676 return BAD_VALUE; 677 } 678 } 679 680 mStreamTypes[stream].volume = value; 681 682 if (thread == NULL) { 683 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 684 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 685 } 686 } else { 687 thread->setStreamVolume(stream, value); 688 } 689 690 return NO_ERROR; 691} 692 693status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 694{ 695 // check calling permissions 696 if (!settingsAllowed()) { 697 return PERMISSION_DENIED; 698 } 699 700 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 701 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 702 ALOGE("setStreamMute() invalid stream %d", stream); 703 return BAD_VALUE; 704 } 705 706 AutoMutex lock(mLock); 707 mStreamTypes[stream].mute = muted; 708 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 709 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 710 711 return NO_ERROR; 712} 713 714float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 715{ 716 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 717 return 0.0f; 718 } 719 720 AutoMutex lock(mLock); 721 float volume; 722 if (output) { 723 PlaybackThread *thread = checkPlaybackThread_l(output); 724 if (thread == NULL) { 725 return 0.0f; 726 } 727 volume = thread->streamVolume(stream); 728 } else { 729 volume = mStreamTypes[stream].volume; 730 } 731 732 return volume; 733} 734 735bool AudioFlinger::streamMute(audio_stream_type_t stream) const 736{ 737 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 738 return true; 739 } 740 741 return mStreamTypes[stream].mute; 742} 743 744status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 745{ 746 status_t result; 747 748 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 749 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 750 // check calling permissions 751 if (!settingsAllowed()) { 752 return PERMISSION_DENIED; 753 } 754 755 // ioHandle == 0 means the parameters are global to the audio hardware interface 756 if (ioHandle == 0) { 757 AutoMutex lock(mHardwareLock); 758 mHardwareStatus = AUDIO_SET_PARAMETER; 759 status_t final_result = NO_ERROR; 760 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 761 audio_hw_device_t *dev = mAudioHwDevs[i]; 762 result = dev->set_parameters(dev, keyValuePairs.string()); 763 final_result = result ?: final_result; 764 } 765 mHardwareStatus = AUDIO_HW_IDLE; 766 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 767 AudioParameter param = AudioParameter(keyValuePairs); 768 String8 value; 769 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 770 Mutex::Autolock _l(mLock); 771 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 772 if (mBtNrecIsOff != btNrecIsOff) { 773 for (size_t i = 0; i < mRecordThreads.size(); i++) { 774 sp<RecordThread> thread = mRecordThreads.valueAt(i); 775 RecordThread::RecordTrack *track = thread->track(); 776 if (track != NULL) { 777 audio_devices_t device = (audio_devices_t)( 778 thread->device() & AUDIO_DEVICE_IN_ALL); 779 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 780 thread->setEffectSuspended(FX_IID_AEC, 781 suspend, 782 track->sessionId()); 783 thread->setEffectSuspended(FX_IID_NS, 784 suspend, 785 track->sessionId()); 786 } 787 } 788 mBtNrecIsOff = btNrecIsOff; 789 } 790 } 791 return final_result; 792 } 793 794 // hold a strong ref on thread in case closeOutput() or closeInput() is called 795 // and the thread is exited once the lock is released 796 sp<ThreadBase> thread; 797 { 798 Mutex::Autolock _l(mLock); 799 thread = checkPlaybackThread_l(ioHandle); 800 if (thread == NULL) { 801 thread = checkRecordThread_l(ioHandle); 802 } else if (thread == primaryPlaybackThread_l()) { 803 // indicate output device change to all input threads for pre processing 804 AudioParameter param = AudioParameter(keyValuePairs); 805 int value; 806 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 807 for (size_t i = 0; i < mRecordThreads.size(); i++) { 808 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 809 } 810 } 811 } 812 } 813 if (thread != NULL) { 814 result = thread->setParameters(keyValuePairs); 815 return result; 816 } 817 return BAD_VALUE; 818} 819 820String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 821{ 822// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 823// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 824 825 if (ioHandle == 0) { 826 String8 out_s8; 827 828 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 829 audio_hw_device_t *dev = mAudioHwDevs[i]; 830 char *s = dev->get_parameters(dev, keys.string()); 831 out_s8 += String8(s); 832 free(s); 833 } 834 return out_s8; 835 } 836 837 Mutex::Autolock _l(mLock); 838 839 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 840 if (playbackThread != NULL) { 841 return playbackThread->getParameters(keys); 842 } 843 RecordThread *recordThread = checkRecordThread_l(ioHandle); 844 if (recordThread != NULL) { 845 return recordThread->getParameters(keys); 846 } 847 return String8(""); 848} 849 850size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) 851{ 852 status_t ret = initCheck(); 853 if (ret != NO_ERROR) { 854 return 0; 855 } 856 857 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 858} 859 860unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 861{ 862 if (ioHandle == 0) { 863 return 0; 864 } 865 866 Mutex::Autolock _l(mLock); 867 868 RecordThread *recordThread = checkRecordThread_l(ioHandle); 869 if (recordThread != NULL) { 870 return recordThread->getInputFramesLost(); 871 } 872 return 0; 873} 874 875status_t AudioFlinger::setVoiceVolume(float value) 876{ 877 status_t ret = initCheck(); 878 if (ret != NO_ERROR) { 879 return ret; 880 } 881 882 // check calling permissions 883 if (!settingsAllowed()) { 884 return PERMISSION_DENIED; 885 } 886 887 AutoMutex lock(mHardwareLock); 888 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 889 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 890 mHardwareStatus = AUDIO_HW_IDLE; 891 892 return ret; 893} 894 895status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 896{ 897 status_t status; 898 899 Mutex::Autolock _l(mLock); 900 901 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 902 if (playbackThread != NULL) { 903 return playbackThread->getRenderPosition(halFrames, dspFrames); 904 } 905 906 return BAD_VALUE; 907} 908 909void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 910{ 911 912 Mutex::Autolock _l(mLock); 913 914 int pid = IPCThreadState::self()->getCallingPid(); 915 if (mNotificationClients.indexOfKey(pid) < 0) { 916 sp<NotificationClient> notificationClient = new NotificationClient(this, 917 client, 918 pid); 919 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 920 921 mNotificationClients.add(pid, notificationClient); 922 923 sp<IBinder> binder = client->asBinder(); 924 binder->linkToDeath(notificationClient); 925 926 // the config change is always sent from playback or record threads to avoid deadlock 927 // with AudioSystem::gLock 928 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 929 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 930 } 931 932 for (size_t i = 0; i < mRecordThreads.size(); i++) { 933 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 934 } 935 } 936} 937 938void AudioFlinger::removeNotificationClient(pid_t pid) 939{ 940 Mutex::Autolock _l(mLock); 941 942 int index = mNotificationClients.indexOfKey(pid); 943 if (index >= 0) { 944 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 945 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 946 mNotificationClients.removeItem(pid); 947 } 948 949 ALOGV("%d died, releasing its sessions", pid); 950 int num = mAudioSessionRefs.size(); 951 bool removed = false; 952 for (int i = 0; i< num; i++) { 953 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 954 ALOGV(" pid %d @ %d", ref->pid, i); 955 if (ref->pid == pid) { 956 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 957 mAudioSessionRefs.removeAt(i); 958 delete ref; 959 removed = true; 960 i--; 961 num--; 962 } 963 } 964 if (removed) { 965 purgeStaleEffects_l(); 966 } 967} 968 969// audioConfigChanged_l() must be called with AudioFlinger::mLock held 970void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 971{ 972 size_t size = mNotificationClients.size(); 973 for (size_t i = 0; i < size; i++) { 974 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 975 } 976} 977 978// removeClient_l() must be called with AudioFlinger::mLock held 979void AudioFlinger::removeClient_l(pid_t pid) 980{ 981 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 982 mClients.removeItem(pid); 983} 984 985 986// ---------------------------------------------------------------------------- 987 988AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 989 : Thread(false), 990 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 991 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), 992 mDevice(device) 993{ 994 mDeathRecipient = new PMDeathRecipient(this); 995} 996 997AudioFlinger::ThreadBase::~ThreadBase() 998{ 999 mParamCond.broadcast(); 1000 // do not lock the mutex in destructor 1001 releaseWakeLock_l(); 1002 if (mPowerManager != 0) { 1003 sp<IBinder> binder = mPowerManager->asBinder(); 1004 binder->unlinkToDeath(mDeathRecipient); 1005 } 1006} 1007 1008void AudioFlinger::ThreadBase::exit() 1009{ 1010 // keep a strong ref on ourself so that we won't get 1011 // destroyed in the middle of requestExitAndWait() 1012 sp <ThreadBase> strongMe = this; 1013 1014 ALOGV("ThreadBase::exit"); 1015 { 1016 AutoMutex lock(mLock); 1017 mExiting = true; 1018 requestExit(); 1019 mWaitWorkCV.signal(); 1020 } 1021 requestExitAndWait(); 1022} 1023 1024uint32_t AudioFlinger::ThreadBase::sampleRate() const 1025{ 1026 return mSampleRate; 1027} 1028 1029int AudioFlinger::ThreadBase::channelCount() const 1030{ 1031 return (int)mChannelCount; 1032} 1033 1034audio_format_t AudioFlinger::ThreadBase::format() const 1035{ 1036 return mFormat; 1037} 1038 1039size_t AudioFlinger::ThreadBase::frameCount() const 1040{ 1041 return mFrameCount; 1042} 1043 1044status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1045{ 1046 status_t status; 1047 1048 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1049 Mutex::Autolock _l(mLock); 1050 1051 mNewParameters.add(keyValuePairs); 1052 mWaitWorkCV.signal(); 1053 // wait condition with timeout in case the thread loop has exited 1054 // before the request could be processed 1055 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1056 status = mParamStatus; 1057 mWaitWorkCV.signal(); 1058 } else { 1059 status = TIMED_OUT; 1060 } 1061 return status; 1062} 1063 1064void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1065{ 1066 Mutex::Autolock _l(mLock); 1067 sendConfigEvent_l(event, param); 1068} 1069 1070// sendConfigEvent_l() must be called with ThreadBase::mLock held 1071void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1072{ 1073 ConfigEvent configEvent; 1074 configEvent.mEvent = event; 1075 configEvent.mParam = param; 1076 mConfigEvents.add(configEvent); 1077 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1078 mWaitWorkCV.signal(); 1079} 1080 1081void AudioFlinger::ThreadBase::processConfigEvents() 1082{ 1083 mLock.lock(); 1084 while(!mConfigEvents.isEmpty()) { 1085 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1086 ConfigEvent configEvent = mConfigEvents[0]; 1087 mConfigEvents.removeAt(0); 1088 // release mLock before locking AudioFlinger mLock: lock order is always 1089 // AudioFlinger then ThreadBase to avoid cross deadlock 1090 mLock.unlock(); 1091 mAudioFlinger->mLock.lock(); 1092 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1093 mAudioFlinger->mLock.unlock(); 1094 mLock.lock(); 1095 } 1096 mLock.unlock(); 1097} 1098 1099status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1100{ 1101 const size_t SIZE = 256; 1102 char buffer[SIZE]; 1103 String8 result; 1104 1105 bool locked = tryLock(mLock); 1106 if (!locked) { 1107 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1108 write(fd, buffer, strlen(buffer)); 1109 } 1110 1111 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1112 result.append(buffer); 1113 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1114 result.append(buffer); 1115 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1116 result.append(buffer); 1117 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1118 result.append(buffer); 1119 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1120 result.append(buffer); 1121 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1122 result.append(buffer); 1123 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1124 result.append(buffer); 1125 1126 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1127 result.append(buffer); 1128 result.append(" Index Command"); 1129 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1130 snprintf(buffer, SIZE, "\n %02d ", i); 1131 result.append(buffer); 1132 result.append(mNewParameters[i]); 1133 } 1134 1135 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1136 result.append(buffer); 1137 snprintf(buffer, SIZE, " Index event param\n"); 1138 result.append(buffer); 1139 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1140 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1141 result.append(buffer); 1142 } 1143 result.append("\n"); 1144 1145 write(fd, result.string(), result.size()); 1146 1147 if (locked) { 1148 mLock.unlock(); 1149 } 1150 return NO_ERROR; 1151} 1152 1153status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1154{ 1155 const size_t SIZE = 256; 1156 char buffer[SIZE]; 1157 String8 result; 1158 1159 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1160 write(fd, buffer, strlen(buffer)); 1161 1162 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1163 sp<EffectChain> chain = mEffectChains[i]; 1164 if (chain != 0) { 1165 chain->dump(fd, args); 1166 } 1167 } 1168 return NO_ERROR; 1169} 1170 1171void AudioFlinger::ThreadBase::acquireWakeLock() 1172{ 1173 Mutex::Autolock _l(mLock); 1174 acquireWakeLock_l(); 1175} 1176 1177void AudioFlinger::ThreadBase::acquireWakeLock_l() 1178{ 1179 if (mPowerManager == 0) { 1180 // use checkService() to avoid blocking if power service is not up yet 1181 sp<IBinder> binder = 1182 defaultServiceManager()->checkService(String16("power")); 1183 if (binder == 0) { 1184 ALOGW("Thread %s cannot connect to the power manager service", mName); 1185 } else { 1186 mPowerManager = interface_cast<IPowerManager>(binder); 1187 binder->linkToDeath(mDeathRecipient); 1188 } 1189 } 1190 if (mPowerManager != 0) { 1191 sp<IBinder> binder = new BBinder(); 1192 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1193 binder, 1194 String16(mName)); 1195 if (status == NO_ERROR) { 1196 mWakeLockToken = binder; 1197 } 1198 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1199 } 1200} 1201 1202void AudioFlinger::ThreadBase::releaseWakeLock() 1203{ 1204 Mutex::Autolock _l(mLock); 1205 releaseWakeLock_l(); 1206} 1207 1208void AudioFlinger::ThreadBase::releaseWakeLock_l() 1209{ 1210 if (mWakeLockToken != 0) { 1211 ALOGV("releaseWakeLock_l() %s", mName); 1212 if (mPowerManager != 0) { 1213 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1214 } 1215 mWakeLockToken.clear(); 1216 } 1217} 1218 1219void AudioFlinger::ThreadBase::clearPowerManager() 1220{ 1221 Mutex::Autolock _l(mLock); 1222 releaseWakeLock_l(); 1223 mPowerManager.clear(); 1224} 1225 1226void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1227{ 1228 sp<ThreadBase> thread = mThread.promote(); 1229 if (thread != 0) { 1230 thread->clearPowerManager(); 1231 } 1232 ALOGW("power manager service died !!!"); 1233} 1234 1235void AudioFlinger::ThreadBase::setEffectSuspended( 1236 const effect_uuid_t *type, bool suspend, int sessionId) 1237{ 1238 Mutex::Autolock _l(mLock); 1239 setEffectSuspended_l(type, suspend, sessionId); 1240} 1241 1242void AudioFlinger::ThreadBase::setEffectSuspended_l( 1243 const effect_uuid_t *type, bool suspend, int sessionId) 1244{ 1245 sp<EffectChain> chain; 1246 chain = getEffectChain_l(sessionId); 1247 if (chain != 0) { 1248 if (type != NULL) { 1249 chain->setEffectSuspended_l(type, suspend); 1250 } else { 1251 chain->setEffectSuspendedAll_l(suspend); 1252 } 1253 } 1254 1255 updateSuspendedSessions_l(type, suspend, sessionId); 1256} 1257 1258void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1259{ 1260 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1261 if (index < 0) { 1262 return; 1263 } 1264 1265 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1266 mSuspendedSessions.editValueAt(index); 1267 1268 for (size_t i = 0; i < sessionEffects.size(); i++) { 1269 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1270 for (int j = 0; j < desc->mRefCount; j++) { 1271 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1272 chain->setEffectSuspendedAll_l(true); 1273 } else { 1274 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1275 desc->mType.timeLow); 1276 chain->setEffectSuspended_l(&desc->mType, true); 1277 } 1278 } 1279 } 1280} 1281 1282void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1283 bool suspend, 1284 int sessionId) 1285{ 1286 int index = mSuspendedSessions.indexOfKey(sessionId); 1287 1288 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1289 1290 if (suspend) { 1291 if (index >= 0) { 1292 sessionEffects = mSuspendedSessions.editValueAt(index); 1293 } else { 1294 mSuspendedSessions.add(sessionId, sessionEffects); 1295 } 1296 } else { 1297 if (index < 0) { 1298 return; 1299 } 1300 sessionEffects = mSuspendedSessions.editValueAt(index); 1301 } 1302 1303 1304 int key = EffectChain::kKeyForSuspendAll; 1305 if (type != NULL) { 1306 key = type->timeLow; 1307 } 1308 index = sessionEffects.indexOfKey(key); 1309 1310 sp <SuspendedSessionDesc> desc; 1311 if (suspend) { 1312 if (index >= 0) { 1313 desc = sessionEffects.valueAt(index); 1314 } else { 1315 desc = new SuspendedSessionDesc(); 1316 if (type != NULL) { 1317 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1318 } 1319 sessionEffects.add(key, desc); 1320 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1321 } 1322 desc->mRefCount++; 1323 } else { 1324 if (index < 0) { 1325 return; 1326 } 1327 desc = sessionEffects.valueAt(index); 1328 if (--desc->mRefCount == 0) { 1329 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1330 sessionEffects.removeItemsAt(index); 1331 if (sessionEffects.isEmpty()) { 1332 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1333 sessionId); 1334 mSuspendedSessions.removeItem(sessionId); 1335 } 1336 } 1337 } 1338 if (!sessionEffects.isEmpty()) { 1339 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1340 } 1341} 1342 1343void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1344 bool enabled, 1345 int sessionId) 1346{ 1347 Mutex::Autolock _l(mLock); 1348 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1349} 1350 1351void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1352 bool enabled, 1353 int sessionId) 1354{ 1355 if (mType != RECORD) { 1356 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1357 // another session. This gives the priority to well behaved effect control panels 1358 // and applications not using global effects. 1359 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1360 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1361 } 1362 } 1363 1364 sp<EffectChain> chain = getEffectChain_l(sessionId); 1365 if (chain != 0) { 1366 chain->checkSuspendOnEffectEnabled(effect, enabled); 1367 } 1368} 1369 1370// ---------------------------------------------------------------------------- 1371 1372AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1373 AudioStreamOut* output, 1374 int id, 1375 uint32_t device) 1376 : ThreadBase(audioFlinger, id, device), 1377 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1378 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1379{ 1380 snprintf(mName, kNameLength, "AudioOut_%d", id); 1381 1382 readOutputParameters(); 1383 1384 // Assumes constructor is called by AudioFlinger with it's mLock held, 1385 // but it would be safer to explicitly pass these as parameters 1386 mMasterVolume = mAudioFlinger->masterVolume_l(); 1387 mMasterMute = mAudioFlinger->masterMute_l(); 1388 1389 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1390 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1391 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1392 stream = (audio_stream_type_t) (stream + 1)) { 1393 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1394 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1395 // initialized by stream_type_t default constructor 1396 // mStreamTypes[stream].valid = true; 1397 } 1398} 1399 1400AudioFlinger::PlaybackThread::~PlaybackThread() 1401{ 1402 delete [] mMixBuffer; 1403} 1404 1405status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1406{ 1407 dumpInternals(fd, args); 1408 dumpTracks(fd, args); 1409 dumpEffectChains(fd, args); 1410 return NO_ERROR; 1411} 1412 1413status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1414{ 1415 const size_t SIZE = 256; 1416 char buffer[SIZE]; 1417 String8 result; 1418 1419 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1420 result.append(buffer); 1421 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1422 for (size_t i = 0; i < mTracks.size(); ++i) { 1423 sp<Track> track = mTracks[i]; 1424 if (track != 0) { 1425 track->dump(buffer, SIZE); 1426 result.append(buffer); 1427 } 1428 } 1429 1430 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1431 result.append(buffer); 1432 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1433 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1434 wp<Track> wTrack = mActiveTracks[i]; 1435 if (wTrack != 0) { 1436 sp<Track> track = wTrack.promote(); 1437 if (track != 0) { 1438 track->dump(buffer, SIZE); 1439 result.append(buffer); 1440 } 1441 } 1442 } 1443 write(fd, result.string(), result.size()); 1444 return NO_ERROR; 1445} 1446 1447status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1448{ 1449 const size_t SIZE = 256; 1450 char buffer[SIZE]; 1451 String8 result; 1452 1453 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1458 result.append(buffer); 1459 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1460 result.append(buffer); 1461 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1462 result.append(buffer); 1463 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1464 result.append(buffer); 1465 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1466 result.append(buffer); 1467 write(fd, result.string(), result.size()); 1468 1469 dumpBase(fd, args); 1470 1471 return NO_ERROR; 1472} 1473 1474// Thread virtuals 1475status_t AudioFlinger::PlaybackThread::readyToRun() 1476{ 1477 status_t status = initCheck(); 1478 if (status == NO_ERROR) { 1479 ALOGI("AudioFlinger's thread %p ready to run", this); 1480 } else { 1481 ALOGE("No working audio driver found."); 1482 } 1483 return status; 1484} 1485 1486void AudioFlinger::PlaybackThread::onFirstRef() 1487{ 1488 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1489} 1490 1491// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1492sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1493 const sp<AudioFlinger::Client>& client, 1494 audio_stream_type_t streamType, 1495 uint32_t sampleRate, 1496 audio_format_t format, 1497 uint32_t channelMask, 1498 int frameCount, 1499 const sp<IMemory>& sharedBuffer, 1500 int sessionId, 1501 status_t *status) 1502{ 1503 sp<Track> track; 1504 status_t lStatus; 1505 1506 if (mType == DIRECT) { 1507 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1508 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1509 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1510 "for output %p with format %d", 1511 sampleRate, format, channelMask, mOutput, mFormat); 1512 lStatus = BAD_VALUE; 1513 goto Exit; 1514 } 1515 } 1516 } else { 1517 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1518 if (sampleRate > mSampleRate*2) { 1519 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1520 lStatus = BAD_VALUE; 1521 goto Exit; 1522 } 1523 } 1524 1525 lStatus = initCheck(); 1526 if (lStatus != NO_ERROR) { 1527 ALOGE("Audio driver not initialized."); 1528 goto Exit; 1529 } 1530 1531 { // scope for mLock 1532 Mutex::Autolock _l(mLock); 1533 1534 // all tracks in same audio session must share the same routing strategy otherwise 1535 // conflicts will happen when tracks are moved from one output to another by audio policy 1536 // manager 1537 uint32_t strategy = 1538 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1539 for (size_t i = 0; i < mTracks.size(); ++i) { 1540 sp<Track> t = mTracks[i]; 1541 if (t != 0) { 1542 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1543 if (sessionId == t->sessionId() && strategy != actual) { 1544 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1545 strategy, actual); 1546 lStatus = BAD_VALUE; 1547 goto Exit; 1548 } 1549 } 1550 } 1551 1552 track = new Track(this, client, streamType, sampleRate, format, 1553 channelMask, frameCount, sharedBuffer, sessionId); 1554 if (track->getCblk() == NULL || track->name() < 0) { 1555 lStatus = NO_MEMORY; 1556 goto Exit; 1557 } 1558 mTracks.add(track); 1559 1560 sp<EffectChain> chain = getEffectChain_l(sessionId); 1561 if (chain != 0) { 1562 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1563 track->setMainBuffer(chain->inBuffer()); 1564 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1565 chain->incTrackCnt(); 1566 } 1567 1568 // invalidate track immediately if the stream type was moved to another thread since 1569 // createTrack() was called by the client process. 1570 if (!mStreamTypes[streamType].valid) { 1571 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1572 this, streamType); 1573 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1574 } 1575 } 1576 lStatus = NO_ERROR; 1577 1578Exit: 1579 if(status) { 1580 *status = lStatus; 1581 } 1582 return track; 1583} 1584 1585uint32_t AudioFlinger::PlaybackThread::latency() const 1586{ 1587 Mutex::Autolock _l(mLock); 1588 if (initCheck() == NO_ERROR) { 1589 return mOutput->stream->get_latency(mOutput->stream); 1590 } else { 1591 return 0; 1592 } 1593} 1594 1595status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1596{ 1597 mMasterVolume = value; 1598 return NO_ERROR; 1599} 1600 1601status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1602{ 1603 mMasterMute = muted; 1604 return NO_ERROR; 1605} 1606 1607float AudioFlinger::PlaybackThread::masterVolume() const 1608{ 1609 return mMasterVolume; 1610} 1611 1612bool AudioFlinger::PlaybackThread::masterMute() const 1613{ 1614 return mMasterMute; 1615} 1616 1617status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1618{ 1619 mStreamTypes[stream].volume = value; 1620 return NO_ERROR; 1621} 1622 1623status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1624{ 1625 mStreamTypes[stream].mute = muted; 1626 return NO_ERROR; 1627} 1628 1629float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1630{ 1631 return mStreamTypes[stream].volume; 1632} 1633 1634bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1635{ 1636 return mStreamTypes[stream].mute; 1637} 1638 1639// addTrack_l() must be called with ThreadBase::mLock held 1640status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1641{ 1642 status_t status = ALREADY_EXISTS; 1643 1644 // set retry count for buffer fill 1645 track->mRetryCount = kMaxTrackStartupRetries; 1646 if (mActiveTracks.indexOf(track) < 0) { 1647 // the track is newly added, make sure it fills up all its 1648 // buffers before playing. This is to ensure the client will 1649 // effectively get the latency it requested. 1650 track->mFillingUpStatus = Track::FS_FILLING; 1651 track->mResetDone = false; 1652 mActiveTracks.add(track); 1653 if (track->mainBuffer() != mMixBuffer) { 1654 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1655 if (chain != 0) { 1656 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1657 chain->incActiveTrackCnt(); 1658 } 1659 } 1660 1661 status = NO_ERROR; 1662 } 1663 1664 ALOGV("mWaitWorkCV.broadcast"); 1665 mWaitWorkCV.broadcast(); 1666 1667 return status; 1668} 1669 1670// destroyTrack_l() must be called with ThreadBase::mLock held 1671void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1672{ 1673 track->mState = TrackBase::TERMINATED; 1674 if (mActiveTracks.indexOf(track) < 0) { 1675 removeTrack_l(track); 1676 } 1677} 1678 1679void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1680{ 1681 mTracks.remove(track); 1682 deleteTrackName_l(track->name()); 1683 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1684 if (chain != 0) { 1685 chain->decTrackCnt(); 1686 } 1687} 1688 1689String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1690{ 1691 String8 out_s8 = String8(""); 1692 char *s; 1693 1694 Mutex::Autolock _l(mLock); 1695 if (initCheck() != NO_ERROR) { 1696 return out_s8; 1697 } 1698 1699 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1700 out_s8 = String8(s); 1701 free(s); 1702 return out_s8; 1703} 1704 1705// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1706void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1707 AudioSystem::OutputDescriptor desc; 1708 void *param2 = 0; 1709 1710 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1711 1712 switch (event) { 1713 case AudioSystem::OUTPUT_OPENED: 1714 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1715 desc.channels = mChannelMask; 1716 desc.samplingRate = mSampleRate; 1717 desc.format = mFormat; 1718 desc.frameCount = mFrameCount; 1719 desc.latency = latency(); 1720 param2 = &desc; 1721 break; 1722 1723 case AudioSystem::STREAM_CONFIG_CHANGED: 1724 param2 = ¶m; 1725 case AudioSystem::OUTPUT_CLOSED: 1726 default: 1727 break; 1728 } 1729 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1730} 1731 1732void AudioFlinger::PlaybackThread::readOutputParameters() 1733{ 1734 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1735 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1736 mChannelCount = (uint16_t)popcount(mChannelMask); 1737 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1738 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1739 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1740 1741 // FIXME - Current mixer implementation only supports stereo output: Always 1742 // Allocate a stereo buffer even if HW output is mono. 1743 delete[] mMixBuffer; 1744 mMixBuffer = new int16_t[mFrameCount * 2]; 1745 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1746 1747 // force reconfiguration of effect chains and engines to take new buffer size and audio 1748 // parameters into account 1749 // Note that mLock is not held when readOutputParameters() is called from the constructor 1750 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1751 // matter. 1752 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1753 Vector< sp<EffectChain> > effectChains = mEffectChains; 1754 for (size_t i = 0; i < effectChains.size(); i ++) { 1755 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1756 } 1757} 1758 1759status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1760{ 1761 if (halFrames == 0 || dspFrames == 0) { 1762 return BAD_VALUE; 1763 } 1764 Mutex::Autolock _l(mLock); 1765 if (initCheck() != NO_ERROR) { 1766 return INVALID_OPERATION; 1767 } 1768 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1769 1770 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1771} 1772 1773uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1774{ 1775 Mutex::Autolock _l(mLock); 1776 uint32_t result = 0; 1777 if (getEffectChain_l(sessionId) != 0) { 1778 result = EFFECT_SESSION; 1779 } 1780 1781 for (size_t i = 0; i < mTracks.size(); ++i) { 1782 sp<Track> track = mTracks[i]; 1783 if (sessionId == track->sessionId() && 1784 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1785 result |= TRACK_SESSION; 1786 break; 1787 } 1788 } 1789 1790 return result; 1791} 1792 1793uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1794{ 1795 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1796 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1797 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1798 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1799 } 1800 for (size_t i = 0; i < mTracks.size(); i++) { 1801 sp<Track> track = mTracks[i]; 1802 if (sessionId == track->sessionId() && 1803 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1804 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1805 } 1806 } 1807 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1808} 1809 1810 1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1812{ 1813 Mutex::Autolock _l(mLock); 1814 return mOutput; 1815} 1816 1817AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1818{ 1819 Mutex::Autolock _l(mLock); 1820 AudioStreamOut *output = mOutput; 1821 mOutput = NULL; 1822 return output; 1823} 1824 1825// this method must always be called either with ThreadBase mLock held or inside the thread loop 1826audio_stream_t* AudioFlinger::PlaybackThread::stream() 1827{ 1828 if (mOutput == NULL) { 1829 return NULL; 1830 } 1831 return &mOutput->stream->common; 1832} 1833 1834uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1835{ 1836 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1837 // decoding and transfer time. So sleeping for half of the latency would likely cause 1838 // underruns 1839 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1840 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1841 } else { 1842 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1843 } 1844} 1845 1846// ---------------------------------------------------------------------------- 1847 1848AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1849 : PlaybackThread(audioFlinger, output, id, device), 1850 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1851{ 1852 mType = ThreadBase::MIXER; 1853 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1854 1855 // FIXME - Current mixer implementation only supports stereo output 1856 if (mChannelCount == 1) { 1857 ALOGE("Invalid audio hardware channel count"); 1858 } 1859} 1860 1861AudioFlinger::MixerThread::~MixerThread() 1862{ 1863 delete mAudioMixer; 1864} 1865 1866bool AudioFlinger::MixerThread::threadLoop() 1867{ 1868 Vector< sp<Track> > tracksToRemove; 1869 mixer_state mixerStatus = MIXER_IDLE; 1870 nsecs_t standbyTime = systemTime(); 1871 size_t mixBufferSize = mFrameCount * mFrameSize; 1872 // FIXME: Relaxed timing because of a certain device that can't meet latency 1873 // Should be reduced to 2x after the vendor fixes the driver issue 1874 // increase threshold again due to low power audio mode. The way this warning threshold is 1875 // calculated and its usefulness should be reconsidered anyway. 1876 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1877 nsecs_t lastWarning = 0; 1878 bool longStandbyExit = false; 1879 uint32_t activeSleepTime = activeSleepTimeUs(); 1880 uint32_t idleSleepTime = idleSleepTimeUs(); 1881 uint32_t sleepTime = idleSleepTime; 1882 uint32_t sleepTimeShift = 0; 1883 Vector< sp<EffectChain> > effectChains; 1884#ifdef DEBUG_CPU_USAGE 1885 ThreadCpuUsage cpu; 1886 const CentralTendencyStatistics& stats = cpu.statistics(); 1887#endif 1888 1889 acquireWakeLock(); 1890 1891 while (!exitPending()) 1892 { 1893#ifdef DEBUG_CPU_USAGE 1894 cpu.sampleAndEnable(); 1895 unsigned n = stats.n(); 1896 // cpu.elapsed() is expensive, so don't call it every loop 1897 if ((n & 127) == 1) { 1898 long long elapsed = cpu.elapsed(); 1899 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1900 double perLoop = elapsed / (double) n; 1901 double perLoop100 = perLoop * 0.01; 1902 double mean = stats.mean(); 1903 double stddev = stats.stddev(); 1904 double minimum = stats.minimum(); 1905 double maximum = stats.maximum(); 1906 cpu.resetStatistics(); 1907 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", 1908 elapsed * .000000001, n, perLoop * .000001, 1909 mean * .001, 1910 stddev * .001, 1911 minimum * .001, 1912 maximum * .001, 1913 mean / perLoop100, 1914 stddev / perLoop100, 1915 minimum / perLoop100, 1916 maximum / perLoop100); 1917 } 1918 } 1919#endif 1920 processConfigEvents(); 1921 1922 mixerStatus = MIXER_IDLE; 1923 { // scope for mLock 1924 1925 Mutex::Autolock _l(mLock); 1926 1927 if (checkForNewParameters_l()) { 1928 mixBufferSize = mFrameCount * mFrameSize; 1929 // FIXME: Relaxed timing because of a certain device that can't meet latency 1930 // Should be reduced to 2x after the vendor fixes the driver issue 1931 // increase threshold again due to low power audio mode. The way this warning 1932 // threshold is calculated and its usefulness should be reconsidered anyway. 1933 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1934 activeSleepTime = activeSleepTimeUs(); 1935 idleSleepTime = idleSleepTimeUs(); 1936 } 1937 1938 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1939 1940 // put audio hardware into standby after short delay 1941 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1942 mSuspended)) { 1943 if (!mStandby) { 1944 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1945 mOutput->stream->common.standby(&mOutput->stream->common); 1946 mStandby = true; 1947 mBytesWritten = 0; 1948 } 1949 1950 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1951 // we're about to wait, flush the binder command buffer 1952 IPCThreadState::self()->flushCommands(); 1953 1954 if (exitPending()) break; 1955 1956 releaseWakeLock_l(); 1957 // wait until we have something to do... 1958 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1959 mWaitWorkCV.wait(mLock); 1960 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1961 acquireWakeLock_l(); 1962 1963 mPrevMixerStatus = MIXER_IDLE; 1964 if (!mMasterMute) { 1965 char value[PROPERTY_VALUE_MAX]; 1966 property_get("ro.audio.silent", value, "0"); 1967 if (atoi(value)) { 1968 ALOGD("Silence is golden"); 1969 setMasterMute(true); 1970 } 1971 } 1972 1973 standbyTime = systemTime() + kStandbyTimeInNsecs; 1974 sleepTime = idleSleepTime; 1975 sleepTimeShift = 0; 1976 continue; 1977 } 1978 } 1979 1980 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1981 1982 // prevent any changes in effect chain list and in each effect chain 1983 // during mixing and effect process as the audio buffers could be deleted 1984 // or modified if an effect is created or deleted 1985 lockEffectChains_l(effectChains); 1986 } 1987 1988 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1989 // mix buffers... 1990 mAudioMixer->process(); 1991 // increase sleep time progressively when application underrun condition clears. 1992 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1993 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1994 // such that we would underrun the audio HAL. 1995 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1996 sleepTimeShift--; 1997 } 1998 sleepTime = 0; 1999 standbyTime = systemTime() + kStandbyTimeInNsecs; 2000 //TODO: delay standby when effects have a tail 2001 } else { 2002 // If no tracks are ready, sleep once for the duration of an output 2003 // buffer size, then write 0s to the output 2004 if (sleepTime == 0) { 2005 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2006 sleepTime = activeSleepTime >> sleepTimeShift; 2007 if (sleepTime < kMinThreadSleepTimeUs) { 2008 sleepTime = kMinThreadSleepTimeUs; 2009 } 2010 // reduce sleep time in case of consecutive application underruns to avoid 2011 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2012 // duration we would end up writing less data than needed by the audio HAL if 2013 // the condition persists. 2014 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2015 sleepTimeShift++; 2016 } 2017 } else { 2018 sleepTime = idleSleepTime; 2019 } 2020 } else if (mBytesWritten != 0 || 2021 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2022 memset (mMixBuffer, 0, mixBufferSize); 2023 sleepTime = 0; 2024 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2025 } 2026 // TODO add standby time extension fct of effect tail 2027 } 2028 2029 if (mSuspended) { 2030 sleepTime = suspendSleepTimeUs(); 2031 } 2032 // sleepTime == 0 means we must write to audio hardware 2033 if (sleepTime == 0) { 2034 for (size_t i = 0; i < effectChains.size(); i ++) { 2035 effectChains[i]->process_l(); 2036 } 2037 // enable changes in effect chain 2038 unlockEffectChains(effectChains); 2039 mLastWriteTime = systemTime(); 2040 mInWrite = true; 2041 mBytesWritten += mixBufferSize; 2042 2043 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2044 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2045 mNumWrites++; 2046 mInWrite = false; 2047 nsecs_t now = systemTime(); 2048 nsecs_t delta = now - mLastWriteTime; 2049 if (!mStandby && delta > maxPeriod) { 2050 mNumDelayedWrites++; 2051 if ((now - lastWarning) > kWarningThrottleNs) { 2052 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2053 ns2ms(delta), mNumDelayedWrites, this); 2054 lastWarning = now; 2055 } 2056 if (mStandby) { 2057 longStandbyExit = true; 2058 } 2059 } 2060 mStandby = false; 2061 } else { 2062 // enable changes in effect chain 2063 unlockEffectChains(effectChains); 2064 usleep(sleepTime); 2065 } 2066 2067 // finally let go of all our tracks, without the lock held 2068 // since we can't guarantee the destructors won't acquire that 2069 // same lock. 2070 tracksToRemove.clear(); 2071 2072 // Effect chains will be actually deleted here if they were removed from 2073 // mEffectChains list during mixing or effects processing 2074 effectChains.clear(); 2075 } 2076 2077 if (!mStandby) { 2078 mOutput->stream->common.standby(&mOutput->stream->common); 2079 } 2080 2081 releaseWakeLock(); 2082 2083 ALOGV("MixerThread %p exiting", this); 2084 return false; 2085} 2086 2087// prepareTracks_l() must be called with ThreadBase::mLock held 2088AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2089 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2090{ 2091 2092 mixer_state mixerStatus = MIXER_IDLE; 2093 // find out which tracks need to be processed 2094 size_t count = activeTracks.size(); 2095 size_t mixedTracks = 0; 2096 size_t tracksWithEffect = 0; 2097 2098 float masterVolume = mMasterVolume; 2099 bool masterMute = mMasterMute; 2100 2101 if (masterMute) { 2102 masterVolume = 0; 2103 } 2104 // Delegate master volume control to effect in output mix effect chain if needed 2105 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2106 if (chain != 0) { 2107 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2108 chain->setVolume_l(&v, &v); 2109 masterVolume = (float)((v + (1 << 23)) >> 24); 2110 chain.clear(); 2111 } 2112 2113 for (size_t i=0 ; i<count ; i++) { 2114 sp<Track> t = activeTracks[i].promote(); 2115 if (t == 0) continue; 2116 2117 // this const just means the local variable doesn't change 2118 Track* const track = t.get(); 2119 audio_track_cblk_t* cblk = track->cblk(); 2120 2121 // The first time a track is added we wait 2122 // for all its buffers to be filled before processing it 2123 int name = track->name(); 2124 // make sure that we have enough frames to mix one full buffer. 2125 // enforce this condition only once to enable draining the buffer in case the client 2126 // app does not call stop() and relies on underrun to stop: 2127 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2128 // during last round 2129 uint32_t minFrames = 1; 2130 if (!track->isStopped() && !track->isPausing() && 2131 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2132 if (t->sampleRate() == (int)mSampleRate) { 2133 minFrames = mFrameCount; 2134 } else { 2135 // +1 for rounding and +1 for additional sample needed for interpolation 2136 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2137 // add frames already consumed but not yet released by the resampler 2138 // because cblk->framesReady() will include these frames 2139 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2140 // the minimum track buffer size is normally twice the number of frames necessary 2141 // to fill one buffer and the resampler should not leave more than one buffer worth 2142 // of unreleased frames after each pass, but just in case... 2143 ALOG_ASSERT(minFrames <= cblk->frameCount); 2144 } 2145 } 2146 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2147 !track->isPaused() && !track->isTerminated()) 2148 { 2149 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2150 2151 mixedTracks++; 2152 2153 // track->mainBuffer() != mMixBuffer means there is an effect chain 2154 // connected to the track 2155 chain.clear(); 2156 if (track->mainBuffer() != mMixBuffer) { 2157 chain = getEffectChain_l(track->sessionId()); 2158 // Delegate volume control to effect in track effect chain if needed 2159 if (chain != 0) { 2160 tracksWithEffect++; 2161 } else { 2162 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2163 name, track->sessionId()); 2164 } 2165 } 2166 2167 2168 int param = AudioMixer::VOLUME; 2169 if (track->mFillingUpStatus == Track::FS_FILLED) { 2170 // no ramp for the first volume setting 2171 track->mFillingUpStatus = Track::FS_ACTIVE; 2172 if (track->mState == TrackBase::RESUMING) { 2173 track->mState = TrackBase::ACTIVE; 2174 param = AudioMixer::RAMP_VOLUME; 2175 } 2176 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2177 } else if (cblk->server != 0) { 2178 // If the track is stopped before the first frame was mixed, 2179 // do not apply ramp 2180 param = AudioMixer::RAMP_VOLUME; 2181 } 2182 2183 // compute volume for this track 2184 uint32_t vl, vr, va; 2185 if (track->isMuted() || track->isPausing() || 2186 mStreamTypes[track->type()].mute) { 2187 vl = vr = va = 0; 2188 if (track->isPausing()) { 2189 track->setPaused(); 2190 } 2191 } else { 2192 2193 // read original volumes with volume control 2194 float typeVolume = mStreamTypes[track->type()].volume; 2195 float v = masterVolume * typeVolume; 2196 uint32_t vlr = cblk->volumeLR; 2197 vl = vlr & 0xFFFF; 2198 vr = vlr >> 16; 2199 // track volumes come from shared memory, so can't be trusted and must be clamped 2200 if (vl > MAX_GAIN_INT) { 2201 ALOGV("Track left volume out of range: %04X", vl); 2202 vl = MAX_GAIN_INT; 2203 } 2204 if (vr > MAX_GAIN_INT) { 2205 ALOGV("Track right volume out of range: %04X", vr); 2206 vr = MAX_GAIN_INT; 2207 } 2208 // now apply the master volume and stream type volume 2209 vl = (uint32_t)(v * vl) << 12; 2210 vr = (uint32_t)(v * vr) << 12; 2211 // assuming master volume and stream type volume each go up to 1.0, 2212 // vl and vr are now in 8.24 format 2213 2214 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2215 // send level comes from shared memory and so may be corrupt 2216 if (sendLevel >= MAX_GAIN_INT) { 2217 ALOGV("Track send level out of range: %04X", sendLevel); 2218 sendLevel = MAX_GAIN_INT; 2219 } 2220 va = (uint32_t)(v * sendLevel); 2221 } 2222 // Delegate volume control to effect in track effect chain if needed 2223 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2224 // Do not ramp volume if volume is controlled by effect 2225 param = AudioMixer::VOLUME; 2226 track->mHasVolumeController = true; 2227 } else { 2228 // force no volume ramp when volume controller was just disabled or removed 2229 // from effect chain to avoid volume spike 2230 if (track->mHasVolumeController) { 2231 param = AudioMixer::VOLUME; 2232 } 2233 track->mHasVolumeController = false; 2234 } 2235 2236 // Convert volumes from 8.24 to 4.12 format 2237 int16_t left, right, aux; 2238 // This additional clamping is needed in case chain->setVolume_l() overshot 2239 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2240 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2241 left = int16_t(v_clamped); 2242 v_clamped = (vr + (1 << 11)) >> 12; 2243 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2244 right = int16_t(v_clamped); 2245 2246 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2247 aux = int16_t(va); 2248 2249 // XXX: these things DON'T need to be done each time 2250 mAudioMixer->setBufferProvider(name, track); 2251 mAudioMixer->enable(name); 2252 2253 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2254 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2255 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2256 mAudioMixer->setParameter( 2257 name, 2258 AudioMixer::TRACK, 2259 AudioMixer::FORMAT, (void *)track->format()); 2260 mAudioMixer->setParameter( 2261 name, 2262 AudioMixer::TRACK, 2263 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2264 mAudioMixer->setParameter( 2265 name, 2266 AudioMixer::RESAMPLE, 2267 AudioMixer::SAMPLE_RATE, 2268 (void *)(cblk->sampleRate)); 2269 mAudioMixer->setParameter( 2270 name, 2271 AudioMixer::TRACK, 2272 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2273 mAudioMixer->setParameter( 2274 name, 2275 AudioMixer::TRACK, 2276 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2277 2278 // reset retry count 2279 track->mRetryCount = kMaxTrackRetries; 2280 // If one track is ready, set the mixer ready if: 2281 // - the mixer was not ready during previous round OR 2282 // - no other track is not ready 2283 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2284 mixerStatus != MIXER_TRACKS_ENABLED) { 2285 mixerStatus = MIXER_TRACKS_READY; 2286 } 2287 } else { 2288 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2289 if (track->isStopped()) { 2290 track->reset(); 2291 } 2292 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2293 // We have consumed all the buffers of this track. 2294 // Remove it from the list of active tracks. 2295 tracksToRemove->add(track); 2296 } else { 2297 // No buffers for this track. Give it a few chances to 2298 // fill a buffer, then remove it from active list. 2299 if (--(track->mRetryCount) <= 0) { 2300 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2301 tracksToRemove->add(track); 2302 // indicate to client process that the track was disabled because of underrun 2303 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2304 // If one track is not ready, mark the mixer also not ready if: 2305 // - the mixer was ready during previous round OR 2306 // - no other track is ready 2307 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2308 mixerStatus != MIXER_TRACKS_READY) { 2309 mixerStatus = MIXER_TRACKS_ENABLED; 2310 } 2311 } 2312 mAudioMixer->disable(name); 2313 } 2314 } 2315 2316 // remove all the tracks that need to be... 2317 count = tracksToRemove->size(); 2318 if (CC_UNLIKELY(count)) { 2319 for (size_t i=0 ; i<count ; i++) { 2320 const sp<Track>& track = tracksToRemove->itemAt(i); 2321 mActiveTracks.remove(track); 2322 if (track->mainBuffer() != mMixBuffer) { 2323 chain = getEffectChain_l(track->sessionId()); 2324 if (chain != 0) { 2325 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2326 chain->decActiveTrackCnt(); 2327 } 2328 } 2329 if (track->isTerminated()) { 2330 removeTrack_l(track); 2331 } 2332 } 2333 } 2334 2335 // mix buffer must be cleared if all tracks are connected to an 2336 // effect chain as in this case the mixer will not write to 2337 // mix buffer and track effects will accumulate into it 2338 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2339 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2340 } 2341 2342 mPrevMixerStatus = mixerStatus; 2343 return mixerStatus; 2344} 2345 2346void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2347{ 2348 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2349 this, streamType, mTracks.size()); 2350 Mutex::Autolock _l(mLock); 2351 2352 size_t size = mTracks.size(); 2353 for (size_t i = 0; i < size; i++) { 2354 sp<Track> t = mTracks[i]; 2355 if (t->type() == streamType) { 2356 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2357 t->mCblk->cv.signal(); 2358 } 2359 } 2360} 2361 2362void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2363{ 2364 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2365 this, streamType, valid); 2366 Mutex::Autolock _l(mLock); 2367 2368 mStreamTypes[streamType].valid = valid; 2369} 2370 2371// getTrackName_l() must be called with ThreadBase::mLock held 2372int AudioFlinger::MixerThread::getTrackName_l() 2373{ 2374 return mAudioMixer->getTrackName(); 2375} 2376 2377// deleteTrackName_l() must be called with ThreadBase::mLock held 2378void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2379{ 2380 ALOGV("remove track (%d) and delete from mixer", name); 2381 mAudioMixer->deleteTrackName(name); 2382} 2383 2384// checkForNewParameters_l() must be called with ThreadBase::mLock held 2385bool AudioFlinger::MixerThread::checkForNewParameters_l() 2386{ 2387 bool reconfig = false; 2388 2389 while (!mNewParameters.isEmpty()) { 2390 status_t status = NO_ERROR; 2391 String8 keyValuePair = mNewParameters[0]; 2392 AudioParameter param = AudioParameter(keyValuePair); 2393 int value; 2394 2395 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2396 reconfig = true; 2397 } 2398 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2399 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2400 status = BAD_VALUE; 2401 } else { 2402 reconfig = true; 2403 } 2404 } 2405 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2406 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2407 status = BAD_VALUE; 2408 } else { 2409 reconfig = true; 2410 } 2411 } 2412 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2413 // do not accept frame count changes if tracks are open as the track buffer 2414 // size depends on frame count and correct behavior would not be guaranteed 2415 // if frame count is changed after track creation 2416 if (!mTracks.isEmpty()) { 2417 status = INVALID_OPERATION; 2418 } else { 2419 reconfig = true; 2420 } 2421 } 2422 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2423 // when changing the audio output device, call addBatteryData to notify 2424 // the change 2425 if ((int)mDevice != value) { 2426 uint32_t params = 0; 2427 // check whether speaker is on 2428 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2429 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2430 } 2431 2432 int deviceWithoutSpeaker 2433 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2434 // check if any other device (except speaker) is on 2435 if (value & deviceWithoutSpeaker ) { 2436 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2437 } 2438 2439 if (params != 0) { 2440 addBatteryData(params); 2441 } 2442 } 2443 2444 // forward device change to effects that have requested to be 2445 // aware of attached audio device. 2446 mDevice = (uint32_t)value; 2447 for (size_t i = 0; i < mEffectChains.size(); i++) { 2448 mEffectChains[i]->setDevice_l(mDevice); 2449 } 2450 } 2451 2452 if (status == NO_ERROR) { 2453 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2454 keyValuePair.string()); 2455 if (!mStandby && status == INVALID_OPERATION) { 2456 mOutput->stream->common.standby(&mOutput->stream->common); 2457 mStandby = true; 2458 mBytesWritten = 0; 2459 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2460 keyValuePair.string()); 2461 } 2462 if (status == NO_ERROR && reconfig) { 2463 delete mAudioMixer; 2464 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2465 mAudioMixer = NULL; 2466 readOutputParameters(); 2467 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2468 for (size_t i = 0; i < mTracks.size() ; i++) { 2469 int name = getTrackName_l(); 2470 if (name < 0) break; 2471 mTracks[i]->mName = name; 2472 // limit track sample rate to 2 x new output sample rate 2473 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2474 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2475 } 2476 } 2477 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2478 } 2479 } 2480 2481 mNewParameters.removeAt(0); 2482 2483 mParamStatus = status; 2484 mParamCond.signal(); 2485 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2486 // already timed out waiting for the status and will never signal the condition. 2487 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2488 } 2489 return reconfig; 2490} 2491 2492status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2493{ 2494 const size_t SIZE = 256; 2495 char buffer[SIZE]; 2496 String8 result; 2497 2498 PlaybackThread::dumpInternals(fd, args); 2499 2500 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2501 result.append(buffer); 2502 write(fd, result.string(), result.size()); 2503 return NO_ERROR; 2504} 2505 2506uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2507{ 2508 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2509} 2510 2511uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2512{ 2513 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2514} 2515 2516// ---------------------------------------------------------------------------- 2517AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2518 : PlaybackThread(audioFlinger, output, id, device) 2519{ 2520 mType = ThreadBase::DIRECT; 2521} 2522 2523AudioFlinger::DirectOutputThread::~DirectOutputThread() 2524{ 2525} 2526 2527static inline 2528int32_t mul(int16_t in, int16_t v) 2529{ 2530#if defined(__arm__) && !defined(__thumb__) 2531 int32_t out; 2532 asm( "smulbb %[out], %[in], %[v] \n" 2533 : [out]"=r"(out) 2534 : [in]"%r"(in), [v]"r"(v) 2535 : ); 2536 return out; 2537#else 2538 return in * int32_t(v); 2539#endif 2540} 2541 2542void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2543{ 2544 // Do not apply volume on compressed audio 2545 if (!audio_is_linear_pcm(mFormat)) { 2546 return; 2547 } 2548 2549 // convert to signed 16 bit before volume calculation 2550 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2551 size_t count = mFrameCount * mChannelCount; 2552 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2553 int16_t *dst = mMixBuffer + count-1; 2554 while(count--) { 2555 *dst-- = (int16_t)(*src--^0x80) << 8; 2556 } 2557 } 2558 2559 size_t frameCount = mFrameCount; 2560 int16_t *out = mMixBuffer; 2561 if (ramp) { 2562 if (mChannelCount == 1) { 2563 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2564 int32_t vlInc = d / (int32_t)frameCount; 2565 int32_t vl = ((int32_t)mLeftVolShort << 16); 2566 do { 2567 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2568 out++; 2569 vl += vlInc; 2570 } while (--frameCount); 2571 2572 } else { 2573 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2574 int32_t vlInc = d / (int32_t)frameCount; 2575 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2576 int32_t vrInc = d / (int32_t)frameCount; 2577 int32_t vl = ((int32_t)mLeftVolShort << 16); 2578 int32_t vr = ((int32_t)mRightVolShort << 16); 2579 do { 2580 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2581 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2582 out += 2; 2583 vl += vlInc; 2584 vr += vrInc; 2585 } while (--frameCount); 2586 } 2587 } else { 2588 if (mChannelCount == 1) { 2589 do { 2590 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2591 out++; 2592 } while (--frameCount); 2593 } else { 2594 do { 2595 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2596 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2597 out += 2; 2598 } while (--frameCount); 2599 } 2600 } 2601 2602 // convert back to unsigned 8 bit after volume calculation 2603 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2604 size_t count = mFrameCount * mChannelCount; 2605 int16_t *src = mMixBuffer; 2606 uint8_t *dst = (uint8_t *)mMixBuffer; 2607 while(count--) { 2608 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2609 } 2610 } 2611 2612 mLeftVolShort = leftVol; 2613 mRightVolShort = rightVol; 2614} 2615 2616bool AudioFlinger::DirectOutputThread::threadLoop() 2617{ 2618 mixer_state mixerStatus = MIXER_IDLE; 2619 sp<Track> trackToRemove; 2620 sp<Track> activeTrack; 2621 nsecs_t standbyTime = systemTime(); 2622 int8_t *curBuf; 2623 size_t mixBufferSize = mFrameCount*mFrameSize; 2624 uint32_t activeSleepTime = activeSleepTimeUs(); 2625 uint32_t idleSleepTime = idleSleepTimeUs(); 2626 uint32_t sleepTime = idleSleepTime; 2627 // use shorter standby delay as on normal output to release 2628 // hardware resources as soon as possible 2629 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2630 2631 acquireWakeLock(); 2632 2633 while (!exitPending()) 2634 { 2635 bool rampVolume; 2636 uint16_t leftVol; 2637 uint16_t rightVol; 2638 Vector< sp<EffectChain> > effectChains; 2639 2640 processConfigEvents(); 2641 2642 mixerStatus = MIXER_IDLE; 2643 2644 { // scope for the mLock 2645 2646 Mutex::Autolock _l(mLock); 2647 2648 if (checkForNewParameters_l()) { 2649 mixBufferSize = mFrameCount*mFrameSize; 2650 activeSleepTime = activeSleepTimeUs(); 2651 idleSleepTime = idleSleepTimeUs(); 2652 standbyDelay = microseconds(activeSleepTime*2); 2653 } 2654 2655 // put audio hardware into standby after short delay 2656 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2657 mSuspended)) { 2658 // wait until we have something to do... 2659 if (!mStandby) { 2660 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2661 mOutput->stream->common.standby(&mOutput->stream->common); 2662 mStandby = true; 2663 mBytesWritten = 0; 2664 } 2665 2666 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2667 // we're about to wait, flush the binder command buffer 2668 IPCThreadState::self()->flushCommands(); 2669 2670 if (exitPending()) break; 2671 2672 releaseWakeLock_l(); 2673 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2674 mWaitWorkCV.wait(mLock); 2675 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2676 acquireWakeLock_l(); 2677 2678 if (!mMasterMute) { 2679 char value[PROPERTY_VALUE_MAX]; 2680 property_get("ro.audio.silent", value, "0"); 2681 if (atoi(value)) { 2682 ALOGD("Silence is golden"); 2683 setMasterMute(true); 2684 } 2685 } 2686 2687 standbyTime = systemTime() + standbyDelay; 2688 sleepTime = idleSleepTime; 2689 continue; 2690 } 2691 } 2692 2693 effectChains = mEffectChains; 2694 2695 // find out which tracks need to be processed 2696 if (mActiveTracks.size() != 0) { 2697 sp<Track> t = mActiveTracks[0].promote(); 2698 if (t == 0) continue; 2699 2700 Track* const track = t.get(); 2701 audio_track_cblk_t* cblk = track->cblk(); 2702 2703 // The first time a track is added we wait 2704 // for all its buffers to be filled before processing it 2705 if (cblk->framesReady() && track->isReady() && 2706 !track->isPaused() && !track->isTerminated()) 2707 { 2708 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2709 2710 if (track->mFillingUpStatus == Track::FS_FILLED) { 2711 track->mFillingUpStatus = Track::FS_ACTIVE; 2712 mLeftVolFloat = mRightVolFloat = 0; 2713 mLeftVolShort = mRightVolShort = 0; 2714 if (track->mState == TrackBase::RESUMING) { 2715 track->mState = TrackBase::ACTIVE; 2716 rampVolume = true; 2717 } 2718 } else if (cblk->server != 0) { 2719 // If the track is stopped before the first frame was mixed, 2720 // do not apply ramp 2721 rampVolume = true; 2722 } 2723 // compute volume for this track 2724 float left, right; 2725 if (track->isMuted() || mMasterMute || track->isPausing() || 2726 mStreamTypes[track->type()].mute) { 2727 left = right = 0; 2728 if (track->isPausing()) { 2729 track->setPaused(); 2730 } 2731 } else { 2732 float typeVolume = mStreamTypes[track->type()].volume; 2733 float v = mMasterVolume * typeVolume; 2734 uint32_t vlr = cblk->volumeLR; 2735 float v_clamped = v * (vlr & 0xFFFF); 2736 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2737 left = v_clamped/MAX_GAIN; 2738 v_clamped = v * (vlr >> 16); 2739 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2740 right = v_clamped/MAX_GAIN; 2741 } 2742 2743 if (left != mLeftVolFloat || right != mRightVolFloat) { 2744 mLeftVolFloat = left; 2745 mRightVolFloat = right; 2746 2747 // If audio HAL implements volume control, 2748 // force software volume to nominal value 2749 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2750 left = 1.0f; 2751 right = 1.0f; 2752 } 2753 2754 // Convert volumes from float to 8.24 2755 uint32_t vl = (uint32_t)(left * (1 << 24)); 2756 uint32_t vr = (uint32_t)(right * (1 << 24)); 2757 2758 // Delegate volume control to effect in track effect chain if needed 2759 // only one effect chain can be present on DirectOutputThread, so if 2760 // there is one, the track is connected to it 2761 if (!effectChains.isEmpty()) { 2762 // Do not ramp volume if volume is controlled by effect 2763 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2764 rampVolume = false; 2765 } 2766 } 2767 2768 // Convert volumes from 8.24 to 4.12 format 2769 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2770 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2771 leftVol = (uint16_t)v_clamped; 2772 v_clamped = (vr + (1 << 11)) >> 12; 2773 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2774 rightVol = (uint16_t)v_clamped; 2775 } else { 2776 leftVol = mLeftVolShort; 2777 rightVol = mRightVolShort; 2778 rampVolume = false; 2779 } 2780 2781 // reset retry count 2782 track->mRetryCount = kMaxTrackRetriesDirect; 2783 activeTrack = t; 2784 mixerStatus = MIXER_TRACKS_READY; 2785 } else { 2786 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2787 if (track->isStopped()) { 2788 track->reset(); 2789 } 2790 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2791 // We have consumed all the buffers of this track. 2792 // Remove it from the list of active tracks. 2793 trackToRemove = track; 2794 } else { 2795 // No buffers for this track. Give it a few chances to 2796 // fill a buffer, then remove it from active list. 2797 if (--(track->mRetryCount) <= 0) { 2798 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2799 trackToRemove = track; 2800 } else { 2801 mixerStatus = MIXER_TRACKS_ENABLED; 2802 } 2803 } 2804 } 2805 } 2806 2807 // remove all the tracks that need to be... 2808 if (CC_UNLIKELY(trackToRemove != 0)) { 2809 mActiveTracks.remove(trackToRemove); 2810 if (!effectChains.isEmpty()) { 2811 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2812 trackToRemove->sessionId()); 2813 effectChains[0]->decActiveTrackCnt(); 2814 } 2815 if (trackToRemove->isTerminated()) { 2816 removeTrack_l(trackToRemove); 2817 } 2818 } 2819 2820 lockEffectChains_l(effectChains); 2821 } 2822 2823 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2824 AudioBufferProvider::Buffer buffer; 2825 size_t frameCount = mFrameCount; 2826 curBuf = (int8_t *)mMixBuffer; 2827 // output audio to hardware 2828 while (frameCount) { 2829 buffer.frameCount = frameCount; 2830 activeTrack->getNextBuffer(&buffer); 2831 if (CC_UNLIKELY(buffer.raw == NULL)) { 2832 memset(curBuf, 0, frameCount * mFrameSize); 2833 break; 2834 } 2835 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2836 frameCount -= buffer.frameCount; 2837 curBuf += buffer.frameCount * mFrameSize; 2838 activeTrack->releaseBuffer(&buffer); 2839 } 2840 sleepTime = 0; 2841 standbyTime = systemTime() + standbyDelay; 2842 } else { 2843 if (sleepTime == 0) { 2844 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2845 sleepTime = activeSleepTime; 2846 } else { 2847 sleepTime = idleSleepTime; 2848 } 2849 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2850 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2851 sleepTime = 0; 2852 } 2853 } 2854 2855 if (mSuspended) { 2856 sleepTime = suspendSleepTimeUs(); 2857 } 2858 // sleepTime == 0 means we must write to audio hardware 2859 if (sleepTime == 0) { 2860 if (mixerStatus == MIXER_TRACKS_READY) { 2861 applyVolume(leftVol, rightVol, rampVolume); 2862 } 2863 for (size_t i = 0; i < effectChains.size(); i ++) { 2864 effectChains[i]->process_l(); 2865 } 2866 unlockEffectChains(effectChains); 2867 2868 mLastWriteTime = systemTime(); 2869 mInWrite = true; 2870 mBytesWritten += mixBufferSize; 2871 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2872 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2873 mNumWrites++; 2874 mInWrite = false; 2875 mStandby = false; 2876 } else { 2877 unlockEffectChains(effectChains); 2878 usleep(sleepTime); 2879 } 2880 2881 // finally let go of removed track, without the lock held 2882 // since we can't guarantee the destructors won't acquire that 2883 // same lock. 2884 trackToRemove.clear(); 2885 activeTrack.clear(); 2886 2887 // Effect chains will be actually deleted here if they were removed from 2888 // mEffectChains list during mixing or effects processing 2889 effectChains.clear(); 2890 } 2891 2892 if (!mStandby) { 2893 mOutput->stream->common.standby(&mOutput->stream->common); 2894 } 2895 2896 releaseWakeLock(); 2897 2898 ALOGV("DirectOutputThread %p exiting", this); 2899 return false; 2900} 2901 2902// getTrackName_l() must be called with ThreadBase::mLock held 2903int AudioFlinger::DirectOutputThread::getTrackName_l() 2904{ 2905 return 0; 2906} 2907 2908// deleteTrackName_l() must be called with ThreadBase::mLock held 2909void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2910{ 2911} 2912 2913// checkForNewParameters_l() must be called with ThreadBase::mLock held 2914bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2915{ 2916 bool reconfig = false; 2917 2918 while (!mNewParameters.isEmpty()) { 2919 status_t status = NO_ERROR; 2920 String8 keyValuePair = mNewParameters[0]; 2921 AudioParameter param = AudioParameter(keyValuePair); 2922 int value; 2923 2924 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2925 // do not accept frame count changes if tracks are open as the track buffer 2926 // size depends on frame count and correct behavior would not be garantied 2927 // if frame count is changed after track creation 2928 if (!mTracks.isEmpty()) { 2929 status = INVALID_OPERATION; 2930 } else { 2931 reconfig = true; 2932 } 2933 } 2934 if (status == NO_ERROR) { 2935 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2936 keyValuePair.string()); 2937 if (!mStandby && status == INVALID_OPERATION) { 2938 mOutput->stream->common.standby(&mOutput->stream->common); 2939 mStandby = true; 2940 mBytesWritten = 0; 2941 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2942 keyValuePair.string()); 2943 } 2944 if (status == NO_ERROR && reconfig) { 2945 readOutputParameters(); 2946 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2947 } 2948 } 2949 2950 mNewParameters.removeAt(0); 2951 2952 mParamStatus = status; 2953 mParamCond.signal(); 2954 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2955 // already timed out waiting for the status and will never signal the condition. 2956 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2957 } 2958 return reconfig; 2959} 2960 2961uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2962{ 2963 uint32_t time; 2964 if (audio_is_linear_pcm(mFormat)) { 2965 time = PlaybackThread::activeSleepTimeUs(); 2966 } else { 2967 time = 10000; 2968 } 2969 return time; 2970} 2971 2972uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2973{ 2974 uint32_t time; 2975 if (audio_is_linear_pcm(mFormat)) { 2976 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2977 } else { 2978 time = 10000; 2979 } 2980 return time; 2981} 2982 2983uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2984{ 2985 uint32_t time; 2986 if (audio_is_linear_pcm(mFormat)) { 2987 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2988 } else { 2989 time = 10000; 2990 } 2991 return time; 2992} 2993 2994 2995// ---------------------------------------------------------------------------- 2996 2997AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2998 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2999{ 3000 mType = ThreadBase::DUPLICATING; 3001 addOutputTrack(mainThread); 3002} 3003 3004AudioFlinger::DuplicatingThread::~DuplicatingThread() 3005{ 3006 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3007 mOutputTracks[i]->destroy(); 3008 } 3009 mOutputTracks.clear(); 3010} 3011 3012bool AudioFlinger::DuplicatingThread::threadLoop() 3013{ 3014 Vector< sp<Track> > tracksToRemove; 3015 mixer_state mixerStatus = MIXER_IDLE; 3016 nsecs_t standbyTime = systemTime(); 3017 size_t mixBufferSize = mFrameCount*mFrameSize; 3018 SortedVector< sp<OutputTrack> > outputTracks; 3019 uint32_t writeFrames = 0; 3020 uint32_t activeSleepTime = activeSleepTimeUs(); 3021 uint32_t idleSleepTime = idleSleepTimeUs(); 3022 uint32_t sleepTime = idleSleepTime; 3023 Vector< sp<EffectChain> > effectChains; 3024 3025 acquireWakeLock(); 3026 3027 while (!exitPending()) 3028 { 3029 processConfigEvents(); 3030 3031 mixerStatus = MIXER_IDLE; 3032 { // scope for the mLock 3033 3034 Mutex::Autolock _l(mLock); 3035 3036 if (checkForNewParameters_l()) { 3037 mixBufferSize = mFrameCount*mFrameSize; 3038 updateWaitTime(); 3039 activeSleepTime = activeSleepTimeUs(); 3040 idleSleepTime = idleSleepTimeUs(); 3041 } 3042 3043 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3044 3045 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3046 outputTracks.add(mOutputTracks[i]); 3047 } 3048 3049 // put audio hardware into standby after short delay 3050 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3051 mSuspended)) { 3052 if (!mStandby) { 3053 for (size_t i = 0; i < outputTracks.size(); i++) { 3054 outputTracks[i]->stop(); 3055 } 3056 mStandby = true; 3057 mBytesWritten = 0; 3058 } 3059 3060 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3061 // we're about to wait, flush the binder command buffer 3062 IPCThreadState::self()->flushCommands(); 3063 outputTracks.clear(); 3064 3065 if (exitPending()) break; 3066 3067 releaseWakeLock_l(); 3068 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3069 mWaitWorkCV.wait(mLock); 3070 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3071 acquireWakeLock_l(); 3072 3073 mPrevMixerStatus = MIXER_IDLE; 3074 if (!mMasterMute) { 3075 char value[PROPERTY_VALUE_MAX]; 3076 property_get("ro.audio.silent", value, "0"); 3077 if (atoi(value)) { 3078 ALOGD("Silence is golden"); 3079 setMasterMute(true); 3080 } 3081 } 3082 3083 standbyTime = systemTime() + kStandbyTimeInNsecs; 3084 sleepTime = idleSleepTime; 3085 continue; 3086 } 3087 } 3088 3089 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3090 3091 // prevent any changes in effect chain list and in each effect chain 3092 // during mixing and effect process as the audio buffers could be deleted 3093 // or modified if an effect is created or deleted 3094 lockEffectChains_l(effectChains); 3095 } 3096 3097 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3098 // mix buffers... 3099 if (outputsReady(outputTracks)) { 3100 mAudioMixer->process(); 3101 } else { 3102 memset(mMixBuffer, 0, mixBufferSize); 3103 } 3104 sleepTime = 0; 3105 writeFrames = mFrameCount; 3106 } else { 3107 if (sleepTime == 0) { 3108 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3109 sleepTime = activeSleepTime; 3110 } else { 3111 sleepTime = idleSleepTime; 3112 } 3113 } else if (mBytesWritten != 0) { 3114 // flush remaining overflow buffers in output tracks 3115 for (size_t i = 0; i < outputTracks.size(); i++) { 3116 if (outputTracks[i]->isActive()) { 3117 sleepTime = 0; 3118 writeFrames = 0; 3119 memset(mMixBuffer, 0, mixBufferSize); 3120 break; 3121 } 3122 } 3123 } 3124 } 3125 3126 if (mSuspended) { 3127 sleepTime = suspendSleepTimeUs(); 3128 } 3129 // sleepTime == 0 means we must write to audio hardware 3130 if (sleepTime == 0) { 3131 for (size_t i = 0; i < effectChains.size(); i ++) { 3132 effectChains[i]->process_l(); 3133 } 3134 // enable changes in effect chain 3135 unlockEffectChains(effectChains); 3136 3137 standbyTime = systemTime() + kStandbyTimeInNsecs; 3138 for (size_t i = 0; i < outputTracks.size(); i++) { 3139 outputTracks[i]->write(mMixBuffer, writeFrames); 3140 } 3141 mStandby = false; 3142 mBytesWritten += mixBufferSize; 3143 } else { 3144 // enable changes in effect chain 3145 unlockEffectChains(effectChains); 3146 usleep(sleepTime); 3147 } 3148 3149 // finally let go of all our tracks, without the lock held 3150 // since we can't guarantee the destructors won't acquire that 3151 // same lock. 3152 tracksToRemove.clear(); 3153 outputTracks.clear(); 3154 3155 // Effect chains will be actually deleted here if they were removed from 3156 // mEffectChains list during mixing or effects processing 3157 effectChains.clear(); 3158 } 3159 3160 releaseWakeLock(); 3161 3162 return false; 3163} 3164 3165void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3166{ 3167 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3168 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3169 this, 3170 mSampleRate, 3171 mFormat, 3172 mChannelMask, 3173 frameCount); 3174 if (outputTrack->cblk() != NULL) { 3175 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3176 mOutputTracks.add(outputTrack); 3177 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3178 updateWaitTime(); 3179 } 3180} 3181 3182void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3183{ 3184 Mutex::Autolock _l(mLock); 3185 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3186 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3187 mOutputTracks[i]->destroy(); 3188 mOutputTracks.removeAt(i); 3189 updateWaitTime(); 3190 return; 3191 } 3192 } 3193 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3194} 3195 3196void AudioFlinger::DuplicatingThread::updateWaitTime() 3197{ 3198 mWaitTimeMs = UINT_MAX; 3199 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3200 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3201 if (strong != NULL) { 3202 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3203 if (waitTimeMs < mWaitTimeMs) { 3204 mWaitTimeMs = waitTimeMs; 3205 } 3206 } 3207 } 3208} 3209 3210 3211bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3212{ 3213 for (size_t i = 0; i < outputTracks.size(); i++) { 3214 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3215 if (thread == 0) { 3216 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3217 return false; 3218 } 3219 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3220 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3221 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3222 return false; 3223 } 3224 } 3225 return true; 3226} 3227 3228uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3229{ 3230 return (mWaitTimeMs * 1000) / 2; 3231} 3232 3233// ---------------------------------------------------------------------------- 3234 3235// TrackBase constructor must be called with AudioFlinger::mLock held 3236AudioFlinger::ThreadBase::TrackBase::TrackBase( 3237 const wp<ThreadBase>& thread, 3238 const sp<Client>& client, 3239 uint32_t sampleRate, 3240 audio_format_t format, 3241 uint32_t channelMask, 3242 int frameCount, 3243 uint32_t flags, 3244 const sp<IMemory>& sharedBuffer, 3245 int sessionId) 3246 : RefBase(), 3247 mThread(thread), 3248 mClient(client), 3249 mCblk(0), 3250 mFrameCount(0), 3251 mState(IDLE), 3252 mClientTid(-1), 3253 mFormat(format), 3254 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3255 mSessionId(sessionId) 3256{ 3257 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3258 3259 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3260 size_t size = sizeof(audio_track_cblk_t); 3261 uint8_t channelCount = popcount(channelMask); 3262 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3263 if (sharedBuffer == 0) { 3264 size += bufferSize; 3265 } 3266 3267 if (client != NULL) { 3268 mCblkMemory = client->heap()->allocate(size); 3269 if (mCblkMemory != 0) { 3270 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3271 if (mCblk) { // construct the shared structure in-place. 3272 new(mCblk) audio_track_cblk_t(); 3273 // clear all buffers 3274 mCblk->frameCount = frameCount; 3275 mCblk->sampleRate = sampleRate; 3276 mChannelCount = channelCount; 3277 mChannelMask = channelMask; 3278 if (sharedBuffer == 0) { 3279 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3280 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3281 // Force underrun condition to avoid false underrun callback until first data is 3282 // written to buffer (other flags are cleared) 3283 mCblk->flags = CBLK_UNDERRUN_ON; 3284 } else { 3285 mBuffer = sharedBuffer->pointer(); 3286 } 3287 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3288 } 3289 } else { 3290 ALOGE("not enough memory for AudioTrack size=%u", size); 3291 client->heap()->dump("AudioTrack"); 3292 return; 3293 } 3294 } else { 3295 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3296 // construct the shared structure in-place. 3297 new(mCblk) audio_track_cblk_t(); 3298 // clear all buffers 3299 mCblk->frameCount = frameCount; 3300 mCblk->sampleRate = sampleRate; 3301 mChannelCount = channelCount; 3302 mChannelMask = channelMask; 3303 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3304 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3305 // Force underrun condition to avoid false underrun callback until first data is 3306 // written to buffer (other flags are cleared) 3307 mCblk->flags = CBLK_UNDERRUN_ON; 3308 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3309 } 3310} 3311 3312AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3313{ 3314 if (mCblk) { 3315 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3316 if (mClient == NULL) { 3317 delete mCblk; 3318 } 3319 } 3320 mCblkMemory.clear(); // and free the shared memory 3321 if (mClient != NULL) { 3322 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3323 mClient.clear(); 3324 } 3325} 3326 3327void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3328{ 3329 buffer->raw = NULL; 3330 mFrameCount = buffer->frameCount; 3331 step(); 3332 buffer->frameCount = 0; 3333} 3334 3335bool AudioFlinger::ThreadBase::TrackBase::step() { 3336 bool result; 3337 audio_track_cblk_t* cblk = this->cblk(); 3338 3339 result = cblk->stepServer(mFrameCount); 3340 if (!result) { 3341 ALOGV("stepServer failed acquiring cblk mutex"); 3342 mFlags |= STEPSERVER_FAILED; 3343 } 3344 return result; 3345} 3346 3347void AudioFlinger::ThreadBase::TrackBase::reset() { 3348 audio_track_cblk_t* cblk = this->cblk(); 3349 3350 cblk->user = 0; 3351 cblk->server = 0; 3352 cblk->userBase = 0; 3353 cblk->serverBase = 0; 3354 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3355 ALOGV("TrackBase::reset"); 3356} 3357 3358sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3359{ 3360 return mCblkMemory; 3361} 3362 3363int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3364 return (int)mCblk->sampleRate; 3365} 3366 3367int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3368 return (const int)mChannelCount; 3369} 3370 3371uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3372 return mChannelMask; 3373} 3374 3375void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3376 audio_track_cblk_t* cblk = this->cblk(); 3377 size_t frameSize = cblk->frameSize; 3378 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3379 int8_t *bufferEnd = bufferStart + frames * frameSize; 3380 3381 // Check validity of returned pointer in case the track control block would have been corrupted. 3382 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3383 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3384 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3385 server %d, serverBase %d, user %d, userBase %d", 3386 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3387 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3388 return 0; 3389 } 3390 3391 return bufferStart; 3392} 3393 3394// ---------------------------------------------------------------------------- 3395 3396// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3397AudioFlinger::PlaybackThread::Track::Track( 3398 const wp<ThreadBase>& thread, 3399 const sp<Client>& client, 3400 audio_stream_type_t streamType, 3401 uint32_t sampleRate, 3402 audio_format_t format, 3403 uint32_t channelMask, 3404 int frameCount, 3405 const sp<IMemory>& sharedBuffer, 3406 int sessionId) 3407 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3408 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3409 mAuxEffectId(0), mHasVolumeController(false) 3410{ 3411 if (mCblk != NULL) { 3412 sp<ThreadBase> baseThread = thread.promote(); 3413 if (baseThread != 0) { 3414 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3415 mName = playbackThread->getTrackName_l(); 3416 mMainBuffer = playbackThread->mixBuffer(); 3417 } 3418 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3419 if (mName < 0) { 3420 ALOGE("no more track names available"); 3421 } 3422 mStreamType = streamType; 3423 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3424 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3425 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3426 } 3427} 3428 3429AudioFlinger::PlaybackThread::Track::~Track() 3430{ 3431 ALOGV("PlaybackThread::Track destructor"); 3432 sp<ThreadBase> thread = mThread.promote(); 3433 if (thread != 0) { 3434 Mutex::Autolock _l(thread->mLock); 3435 mState = TERMINATED; 3436 } 3437} 3438 3439void AudioFlinger::PlaybackThread::Track::destroy() 3440{ 3441 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3442 // by removing it from mTracks vector, so there is a risk that this Tracks's 3443 // desctructor is called. As the destructor needs to lock mLock, 3444 // we must acquire a strong reference on this Track before locking mLock 3445 // here so that the destructor is called only when exiting this function. 3446 // On the other hand, as long as Track::destroy() is only called by 3447 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3448 // this Track with its member mTrack. 3449 sp<Track> keep(this); 3450 { // scope for mLock 3451 sp<ThreadBase> thread = mThread.promote(); 3452 if (thread != 0) { 3453 if (!isOutputTrack()) { 3454 if (mState == ACTIVE || mState == RESUMING) { 3455 AudioSystem::stopOutput(thread->id(), 3456 (audio_stream_type_t)mStreamType, 3457 mSessionId); 3458 3459 // to track the speaker usage 3460 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3461 } 3462 AudioSystem::releaseOutput(thread->id()); 3463 } 3464 Mutex::Autolock _l(thread->mLock); 3465 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3466 playbackThread->destroyTrack_l(this); 3467 } 3468 } 3469} 3470 3471void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3472{ 3473 uint32_t vlr = mCblk->volumeLR; 3474 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", 3475 mName - AudioMixer::TRACK0, 3476 (mClient == NULL) ? getpid() : mClient->pid(), 3477 mStreamType, 3478 mFormat, 3479 mChannelMask, 3480 mSessionId, 3481 mFrameCount, 3482 mState, 3483 mMute, 3484 mFillingUpStatus, 3485 mCblk->sampleRate, 3486 vlr & 0xFFFF, 3487 vlr >> 16, 3488 mCblk->server, 3489 mCblk->user, 3490 (int)mMainBuffer, 3491 (int)mAuxBuffer); 3492} 3493 3494status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3495{ 3496 audio_track_cblk_t* cblk = this->cblk(); 3497 uint32_t framesReady; 3498 uint32_t framesReq = buffer->frameCount; 3499 3500 // Check if last stepServer failed, try to step now 3501 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3502 if (!step()) goto getNextBuffer_exit; 3503 ALOGV("stepServer recovered"); 3504 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3505 } 3506 3507 framesReady = cblk->framesReady(); 3508 3509 if (CC_LIKELY(framesReady)) { 3510 uint32_t s = cblk->server; 3511 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3512 3513 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3514 if (framesReq > framesReady) { 3515 framesReq = framesReady; 3516 } 3517 if (s + framesReq > bufferEnd) { 3518 framesReq = bufferEnd - s; 3519 } 3520 3521 buffer->raw = getBuffer(s, framesReq); 3522 if (buffer->raw == NULL) goto getNextBuffer_exit; 3523 3524 buffer->frameCount = framesReq; 3525 return NO_ERROR; 3526 } 3527 3528getNextBuffer_exit: 3529 buffer->raw = NULL; 3530 buffer->frameCount = 0; 3531 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3532 return NOT_ENOUGH_DATA; 3533} 3534 3535bool AudioFlinger::PlaybackThread::Track::isReady() const { 3536 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3537 3538 if (mCblk->framesReady() >= mCblk->frameCount || 3539 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3540 mFillingUpStatus = FS_FILLED; 3541 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3542 return true; 3543 } 3544 return false; 3545} 3546 3547status_t AudioFlinger::PlaybackThread::Track::start() 3548{ 3549 status_t status = NO_ERROR; 3550 ALOGV("start(%d), calling thread %d session %d", 3551 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3552 sp<ThreadBase> thread = mThread.promote(); 3553 if (thread != 0) { 3554 Mutex::Autolock _l(thread->mLock); 3555 track_state state = mState; 3556 // here the track could be either new, or restarted 3557 // in both cases "unstop" the track 3558 if (mState == PAUSED) { 3559 mState = TrackBase::RESUMING; 3560 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3561 } else { 3562 mState = TrackBase::ACTIVE; 3563 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3564 } 3565 3566 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3567 thread->mLock.unlock(); 3568 status = AudioSystem::startOutput(thread->id(), 3569 (audio_stream_type_t)mStreamType, 3570 mSessionId); 3571 thread->mLock.lock(); 3572 3573 // to track the speaker usage 3574 if (status == NO_ERROR) { 3575 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3576 } 3577 } 3578 if (status == NO_ERROR) { 3579 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3580 playbackThread->addTrack_l(this); 3581 } else { 3582 mState = state; 3583 } 3584 } else { 3585 status = BAD_VALUE; 3586 } 3587 return status; 3588} 3589 3590void AudioFlinger::PlaybackThread::Track::stop() 3591{ 3592 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3593 sp<ThreadBase> thread = mThread.promote(); 3594 if (thread != 0) { 3595 Mutex::Autolock _l(thread->mLock); 3596 track_state state = mState; 3597 if (mState > STOPPED) { 3598 mState = STOPPED; 3599 // If the track is not active (PAUSED and buffers full), flush buffers 3600 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3601 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3602 reset(); 3603 } 3604 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3605 } 3606 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3607 thread->mLock.unlock(); 3608 AudioSystem::stopOutput(thread->id(), 3609 (audio_stream_type_t)mStreamType, 3610 mSessionId); 3611 thread->mLock.lock(); 3612 3613 // to track the speaker usage 3614 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3615 } 3616 } 3617} 3618 3619void AudioFlinger::PlaybackThread::Track::pause() 3620{ 3621 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3622 sp<ThreadBase> thread = mThread.promote(); 3623 if (thread != 0) { 3624 Mutex::Autolock _l(thread->mLock); 3625 if (mState == ACTIVE || mState == RESUMING) { 3626 mState = PAUSING; 3627 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3628 if (!isOutputTrack()) { 3629 thread->mLock.unlock(); 3630 AudioSystem::stopOutput(thread->id(), 3631 (audio_stream_type_t)mStreamType, 3632 mSessionId); 3633 thread->mLock.lock(); 3634 3635 // to track the speaker usage 3636 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3637 } 3638 } 3639 } 3640} 3641 3642void AudioFlinger::PlaybackThread::Track::flush() 3643{ 3644 ALOGV("flush(%d)", mName); 3645 sp<ThreadBase> thread = mThread.promote(); 3646 if (thread != 0) { 3647 Mutex::Autolock _l(thread->mLock); 3648 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3649 return; 3650 } 3651 // No point remaining in PAUSED state after a flush => go to 3652 // STOPPED state 3653 mState = STOPPED; 3654 3655 // do not reset the track if it is still in the process of being stopped or paused. 3656 // this will be done by prepareTracks_l() when the track is stopped. 3657 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3658 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3659 reset(); 3660 } 3661 } 3662} 3663 3664void AudioFlinger::PlaybackThread::Track::reset() 3665{ 3666 // Do not reset twice to avoid discarding data written just after a flush and before 3667 // the audioflinger thread detects the track is stopped. 3668 if (!mResetDone) { 3669 TrackBase::reset(); 3670 // Force underrun condition to avoid false underrun callback until first data is 3671 // written to buffer 3672 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3673 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3674 mFillingUpStatus = FS_FILLING; 3675 mResetDone = true; 3676 } 3677} 3678 3679void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3680{ 3681 mMute = muted; 3682} 3683 3684status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3685{ 3686 status_t status = DEAD_OBJECT; 3687 sp<ThreadBase> thread = mThread.promote(); 3688 if (thread != 0) { 3689 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3690 status = playbackThread->attachAuxEffect(this, EffectId); 3691 } 3692 return status; 3693} 3694 3695void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3696{ 3697 mAuxEffectId = EffectId; 3698 mAuxBuffer = buffer; 3699} 3700 3701// ---------------------------------------------------------------------------- 3702 3703// RecordTrack constructor must be called with AudioFlinger::mLock held 3704AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3705 const wp<ThreadBase>& thread, 3706 const sp<Client>& client, 3707 uint32_t sampleRate, 3708 audio_format_t format, 3709 uint32_t channelMask, 3710 int frameCount, 3711 uint32_t flags, 3712 int sessionId) 3713 : TrackBase(thread, client, sampleRate, format, 3714 channelMask, frameCount, flags, 0, sessionId), 3715 mOverflow(false) 3716{ 3717 if (mCblk != NULL) { 3718 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3719 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3720 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3721 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3722 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3723 } else { 3724 mCblk->frameSize = sizeof(int8_t); 3725 } 3726 } 3727} 3728 3729AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3730{ 3731 sp<ThreadBase> thread = mThread.promote(); 3732 if (thread != 0) { 3733 AudioSystem::releaseInput(thread->id()); 3734 } 3735} 3736 3737status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3738{ 3739 audio_track_cblk_t* cblk = this->cblk(); 3740 uint32_t framesAvail; 3741 uint32_t framesReq = buffer->frameCount; 3742 3743 // Check if last stepServer failed, try to step now 3744 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3745 if (!step()) goto getNextBuffer_exit; 3746 ALOGV("stepServer recovered"); 3747 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3748 } 3749 3750 framesAvail = cblk->framesAvailable_l(); 3751 3752 if (CC_LIKELY(framesAvail)) { 3753 uint32_t s = cblk->server; 3754 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3755 3756 if (framesReq > framesAvail) { 3757 framesReq = framesAvail; 3758 } 3759 if (s + framesReq > bufferEnd) { 3760 framesReq = bufferEnd - s; 3761 } 3762 3763 buffer->raw = getBuffer(s, framesReq); 3764 if (buffer->raw == NULL) goto getNextBuffer_exit; 3765 3766 buffer->frameCount = framesReq; 3767 return NO_ERROR; 3768 } 3769 3770getNextBuffer_exit: 3771 buffer->raw = NULL; 3772 buffer->frameCount = 0; 3773 return NOT_ENOUGH_DATA; 3774} 3775 3776status_t AudioFlinger::RecordThread::RecordTrack::start() 3777{ 3778 sp<ThreadBase> thread = mThread.promote(); 3779 if (thread != 0) { 3780 RecordThread *recordThread = (RecordThread *)thread.get(); 3781 return recordThread->start(this); 3782 } else { 3783 return BAD_VALUE; 3784 } 3785} 3786 3787void AudioFlinger::RecordThread::RecordTrack::stop() 3788{ 3789 sp<ThreadBase> thread = mThread.promote(); 3790 if (thread != 0) { 3791 RecordThread *recordThread = (RecordThread *)thread.get(); 3792 recordThread->stop(this); 3793 TrackBase::reset(); 3794 // Force overerrun condition to avoid false overrun callback until first data is 3795 // read from buffer 3796 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3797 } 3798} 3799 3800void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3801{ 3802 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3803 (mClient == NULL) ? getpid() : mClient->pid(), 3804 mFormat, 3805 mChannelMask, 3806 mSessionId, 3807 mFrameCount, 3808 mState, 3809 mCblk->sampleRate, 3810 mCblk->server, 3811 mCblk->user); 3812} 3813 3814 3815// ---------------------------------------------------------------------------- 3816 3817AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3818 const wp<ThreadBase>& thread, 3819 DuplicatingThread *sourceThread, 3820 uint32_t sampleRate, 3821 audio_format_t format, 3822 uint32_t channelMask, 3823 int frameCount) 3824 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3825 mActive(false), mSourceThread(sourceThread) 3826{ 3827 3828 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3829 if (mCblk != NULL) { 3830 mCblk->flags |= CBLK_DIRECTION_OUT; 3831 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3832 mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT; 3833 mOutBuffer.frameCount = 0; 3834 playbackThread->mTracks.add(this); 3835 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3836 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3837 mCblk, mBuffer, mCblk->buffers, 3838 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3839 } else { 3840 ALOGW("Error creating output track on thread %p", playbackThread); 3841 } 3842} 3843 3844AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3845{ 3846 clearBufferQueue(); 3847} 3848 3849status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3850{ 3851 status_t status = Track::start(); 3852 if (status != NO_ERROR) { 3853 return status; 3854 } 3855 3856 mActive = true; 3857 mRetryCount = 127; 3858 return status; 3859} 3860 3861void AudioFlinger::PlaybackThread::OutputTrack::stop() 3862{ 3863 Track::stop(); 3864 clearBufferQueue(); 3865 mOutBuffer.frameCount = 0; 3866 mActive = false; 3867} 3868 3869bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3870{ 3871 Buffer *pInBuffer; 3872 Buffer inBuffer; 3873 uint32_t channelCount = mChannelCount; 3874 bool outputBufferFull = false; 3875 inBuffer.frameCount = frames; 3876 inBuffer.i16 = data; 3877 3878 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3879 3880 if (!mActive && frames != 0) { 3881 start(); 3882 sp<ThreadBase> thread = mThread.promote(); 3883 if (thread != 0) { 3884 MixerThread *mixerThread = (MixerThread *)thread.get(); 3885 if (mCblk->frameCount > frames){ 3886 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3887 uint32_t startFrames = (mCblk->frameCount - frames); 3888 pInBuffer = new Buffer; 3889 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3890 pInBuffer->frameCount = startFrames; 3891 pInBuffer->i16 = pInBuffer->mBuffer; 3892 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3893 mBufferQueue.add(pInBuffer); 3894 } else { 3895 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3896 } 3897 } 3898 } 3899 } 3900 3901 while (waitTimeLeftMs) { 3902 // First write pending buffers, then new data 3903 if (mBufferQueue.size()) { 3904 pInBuffer = mBufferQueue.itemAt(0); 3905 } else { 3906 pInBuffer = &inBuffer; 3907 } 3908 3909 if (pInBuffer->frameCount == 0) { 3910 break; 3911 } 3912 3913 if (mOutBuffer.frameCount == 0) { 3914 mOutBuffer.frameCount = pInBuffer->frameCount; 3915 nsecs_t startTime = systemTime(); 3916 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3917 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3918 outputBufferFull = true; 3919 break; 3920 } 3921 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3922 if (waitTimeLeftMs >= waitTimeMs) { 3923 waitTimeLeftMs -= waitTimeMs; 3924 } else { 3925 waitTimeLeftMs = 0; 3926 } 3927 } 3928 3929 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3930 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3931 mCblk->stepUser(outFrames); 3932 pInBuffer->frameCount -= outFrames; 3933 pInBuffer->i16 += outFrames * channelCount; 3934 mOutBuffer.frameCount -= outFrames; 3935 mOutBuffer.i16 += outFrames * channelCount; 3936 3937 if (pInBuffer->frameCount == 0) { 3938 if (mBufferQueue.size()) { 3939 mBufferQueue.removeAt(0); 3940 delete [] pInBuffer->mBuffer; 3941 delete pInBuffer; 3942 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3943 } else { 3944 break; 3945 } 3946 } 3947 } 3948 3949 // If we could not write all frames, allocate a buffer and queue it for next time. 3950 if (inBuffer.frameCount) { 3951 sp<ThreadBase> thread = mThread.promote(); 3952 if (thread != 0 && !thread->standby()) { 3953 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3954 pInBuffer = new Buffer; 3955 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3956 pInBuffer->frameCount = inBuffer.frameCount; 3957 pInBuffer->i16 = pInBuffer->mBuffer; 3958 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3959 mBufferQueue.add(pInBuffer); 3960 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3961 } else { 3962 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3963 } 3964 } 3965 } 3966 3967 // Calling write() with a 0 length buffer, means that no more data will be written: 3968 // If no more buffers are pending, fill output track buffer to make sure it is started 3969 // by output mixer. 3970 if (frames == 0 && mBufferQueue.size() == 0) { 3971 if (mCblk->user < mCblk->frameCount) { 3972 frames = mCblk->frameCount - mCblk->user; 3973 pInBuffer = new Buffer; 3974 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3975 pInBuffer->frameCount = frames; 3976 pInBuffer->i16 = pInBuffer->mBuffer; 3977 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3978 mBufferQueue.add(pInBuffer); 3979 } else if (mActive) { 3980 stop(); 3981 } 3982 } 3983 3984 return outputBufferFull; 3985} 3986 3987status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3988{ 3989 int active; 3990 status_t result; 3991 audio_track_cblk_t* cblk = mCblk; 3992 uint32_t framesReq = buffer->frameCount; 3993 3994// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3995 buffer->frameCount = 0; 3996 3997 uint32_t framesAvail = cblk->framesAvailable(); 3998 3999 4000 if (framesAvail == 0) { 4001 Mutex::Autolock _l(cblk->lock); 4002 goto start_loop_here; 4003 while (framesAvail == 0) { 4004 active = mActive; 4005 if (CC_UNLIKELY(!active)) { 4006 ALOGV("Not active and NO_MORE_BUFFERS"); 4007 return NO_MORE_BUFFERS; 4008 } 4009 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4010 if (result != NO_ERROR) { 4011 return NO_MORE_BUFFERS; 4012 } 4013 // read the server count again 4014 start_loop_here: 4015 framesAvail = cblk->framesAvailable_l(); 4016 } 4017 } 4018 4019// if (framesAvail < framesReq) { 4020// return NO_MORE_BUFFERS; 4021// } 4022 4023 if (framesReq > framesAvail) { 4024 framesReq = framesAvail; 4025 } 4026 4027 uint32_t u = cblk->user; 4028 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4029 4030 if (u + framesReq > bufferEnd) { 4031 framesReq = bufferEnd - u; 4032 } 4033 4034 buffer->frameCount = framesReq; 4035 buffer->raw = (void *)cblk->buffer(u); 4036 return NO_ERROR; 4037} 4038 4039 4040void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4041{ 4042 size_t size = mBufferQueue.size(); 4043 Buffer *pBuffer; 4044 4045 for (size_t i = 0; i < size; i++) { 4046 pBuffer = mBufferQueue.itemAt(i); 4047 delete [] pBuffer->mBuffer; 4048 delete pBuffer; 4049 } 4050 mBufferQueue.clear(); 4051} 4052 4053// ---------------------------------------------------------------------------- 4054 4055AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4056 : RefBase(), 4057 mAudioFlinger(audioFlinger), 4058 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4059 mPid(pid) 4060{ 4061 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4062} 4063 4064// Client destructor must be called with AudioFlinger::mLock held 4065AudioFlinger::Client::~Client() 4066{ 4067 mAudioFlinger->removeClient_l(mPid); 4068} 4069 4070const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4071{ 4072 return mMemoryDealer; 4073} 4074 4075// ---------------------------------------------------------------------------- 4076 4077AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4078 const sp<IAudioFlingerClient>& client, 4079 pid_t pid) 4080 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4081{ 4082} 4083 4084AudioFlinger::NotificationClient::~NotificationClient() 4085{ 4086 mClient.clear(); 4087} 4088 4089void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4090{ 4091 sp<NotificationClient> keep(this); 4092 { 4093 mAudioFlinger->removeNotificationClient(mPid); 4094 } 4095} 4096 4097// ---------------------------------------------------------------------------- 4098 4099AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4100 : BnAudioTrack(), 4101 mTrack(track) 4102{ 4103} 4104 4105AudioFlinger::TrackHandle::~TrackHandle() { 4106 // just stop the track on deletion, associated resources 4107 // will be freed from the main thread once all pending buffers have 4108 // been played. Unless it's not in the active track list, in which 4109 // case we free everything now... 4110 mTrack->destroy(); 4111} 4112 4113sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4114 return mTrack->getCblk(); 4115} 4116 4117status_t AudioFlinger::TrackHandle::start() { 4118 return mTrack->start(); 4119} 4120 4121void AudioFlinger::TrackHandle::stop() { 4122 mTrack->stop(); 4123} 4124 4125void AudioFlinger::TrackHandle::flush() { 4126 mTrack->flush(); 4127} 4128 4129void AudioFlinger::TrackHandle::mute(bool e) { 4130 mTrack->mute(e); 4131} 4132 4133void AudioFlinger::TrackHandle::pause() { 4134 mTrack->pause(); 4135} 4136 4137status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4138{ 4139 return mTrack->attachAuxEffect(EffectId); 4140} 4141 4142status_t AudioFlinger::TrackHandle::onTransact( 4143 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4144{ 4145 return BnAudioTrack::onTransact(code, data, reply, flags); 4146} 4147 4148// ---------------------------------------------------------------------------- 4149 4150sp<IAudioRecord> AudioFlinger::openRecord( 4151 pid_t pid, 4152 int input, 4153 uint32_t sampleRate, 4154 audio_format_t format, 4155 uint32_t channelMask, 4156 int frameCount, 4157 uint32_t flags, 4158 int *sessionId, 4159 status_t *status) 4160{ 4161 sp<RecordThread::RecordTrack> recordTrack; 4162 sp<RecordHandle> recordHandle; 4163 sp<Client> client; 4164 wp<Client> wclient; 4165 status_t lStatus; 4166 RecordThread *thread; 4167 size_t inFrameCount; 4168 int lSessionId; 4169 4170 // check calling permissions 4171 if (!recordingAllowed()) { 4172 lStatus = PERMISSION_DENIED; 4173 goto Exit; 4174 } 4175 4176 // add client to list 4177 { // scope for mLock 4178 Mutex::Autolock _l(mLock); 4179 thread = checkRecordThread_l(input); 4180 if (thread == NULL) { 4181 lStatus = BAD_VALUE; 4182 goto Exit; 4183 } 4184 4185 wclient = mClients.valueFor(pid); 4186 if (wclient != NULL) { 4187 client = wclient.promote(); 4188 } else { 4189 client = new Client(this, pid); 4190 mClients.add(pid, client); 4191 } 4192 4193 // If no audio session id is provided, create one here 4194 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4195 lSessionId = *sessionId; 4196 } else { 4197 lSessionId = nextUniqueId(); 4198 if (sessionId != NULL) { 4199 *sessionId = lSessionId; 4200 } 4201 } 4202 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4203 recordTrack = thread->createRecordTrack_l(client, 4204 sampleRate, 4205 format, 4206 channelMask, 4207 frameCount, 4208 flags, 4209 lSessionId, 4210 &lStatus); 4211 } 4212 if (lStatus != NO_ERROR) { 4213 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4214 // destructor is called by the TrackBase destructor with mLock held 4215 client.clear(); 4216 recordTrack.clear(); 4217 goto Exit; 4218 } 4219 4220 // return to handle to client 4221 recordHandle = new RecordHandle(recordTrack); 4222 lStatus = NO_ERROR; 4223 4224Exit: 4225 if (status) { 4226 *status = lStatus; 4227 } 4228 return recordHandle; 4229} 4230 4231// ---------------------------------------------------------------------------- 4232 4233AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4234 : BnAudioRecord(), 4235 mRecordTrack(recordTrack) 4236{ 4237} 4238 4239AudioFlinger::RecordHandle::~RecordHandle() { 4240 stop(); 4241} 4242 4243sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4244 return mRecordTrack->getCblk(); 4245} 4246 4247status_t AudioFlinger::RecordHandle::start() { 4248 ALOGV("RecordHandle::start()"); 4249 return mRecordTrack->start(); 4250} 4251 4252void AudioFlinger::RecordHandle::stop() { 4253 ALOGV("RecordHandle::stop()"); 4254 mRecordTrack->stop(); 4255} 4256 4257status_t AudioFlinger::RecordHandle::onTransact( 4258 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4259{ 4260 return BnAudioRecord::onTransact(code, data, reply, flags); 4261} 4262 4263// ---------------------------------------------------------------------------- 4264 4265AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4266 AudioStreamIn *input, 4267 uint32_t sampleRate, 4268 uint32_t channels, 4269 int id, 4270 uint32_t device) : 4271 ThreadBase(audioFlinger, id, device), 4272 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4273{ 4274 mType = ThreadBase::RECORD; 4275 4276 snprintf(mName, kNameLength, "AudioIn_%d", id); 4277 4278 mReqChannelCount = popcount(channels); 4279 mReqSampleRate = sampleRate; 4280 readInputParameters(); 4281} 4282 4283 4284AudioFlinger::RecordThread::~RecordThread() 4285{ 4286 delete[] mRsmpInBuffer; 4287 delete mResampler; 4288 delete[] mRsmpOutBuffer; 4289} 4290 4291void AudioFlinger::RecordThread::onFirstRef() 4292{ 4293 run(mName, PRIORITY_URGENT_AUDIO); 4294} 4295 4296status_t AudioFlinger::RecordThread::readyToRun() 4297{ 4298 status_t status = initCheck(); 4299 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4300 return status; 4301} 4302 4303bool AudioFlinger::RecordThread::threadLoop() 4304{ 4305 AudioBufferProvider::Buffer buffer; 4306 sp<RecordTrack> activeTrack; 4307 Vector< sp<EffectChain> > effectChains; 4308 4309 nsecs_t lastWarning = 0; 4310 4311 acquireWakeLock(); 4312 4313 // start recording 4314 while (!exitPending()) { 4315 4316 processConfigEvents(); 4317 4318 { // scope for mLock 4319 Mutex::Autolock _l(mLock); 4320 checkForNewParameters_l(); 4321 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4322 if (!mStandby) { 4323 mInput->stream->common.standby(&mInput->stream->common); 4324 mStandby = true; 4325 } 4326 4327 if (exitPending()) break; 4328 4329 releaseWakeLock_l(); 4330 ALOGV("RecordThread: loop stopping"); 4331 // go to sleep 4332 mWaitWorkCV.wait(mLock); 4333 ALOGV("RecordThread: loop starting"); 4334 acquireWakeLock_l(); 4335 continue; 4336 } 4337 if (mActiveTrack != 0) { 4338 if (mActiveTrack->mState == TrackBase::PAUSING) { 4339 if (!mStandby) { 4340 mInput->stream->common.standby(&mInput->stream->common); 4341 mStandby = true; 4342 } 4343 mActiveTrack.clear(); 4344 mStartStopCond.broadcast(); 4345 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4346 if (mReqChannelCount != mActiveTrack->channelCount()) { 4347 mActiveTrack.clear(); 4348 mStartStopCond.broadcast(); 4349 } else if (mBytesRead != 0) { 4350 // record start succeeds only if first read from audio input 4351 // succeeds 4352 if (mBytesRead > 0) { 4353 mActiveTrack->mState = TrackBase::ACTIVE; 4354 } else { 4355 mActiveTrack.clear(); 4356 } 4357 mStartStopCond.broadcast(); 4358 } 4359 mStandby = false; 4360 } 4361 } 4362 lockEffectChains_l(effectChains); 4363 } 4364 4365 if (mActiveTrack != 0) { 4366 if (mActiveTrack->mState != TrackBase::ACTIVE && 4367 mActiveTrack->mState != TrackBase::RESUMING) { 4368 unlockEffectChains(effectChains); 4369 usleep(kRecordThreadSleepUs); 4370 continue; 4371 } 4372 for (size_t i = 0; i < effectChains.size(); i ++) { 4373 effectChains[i]->process_l(); 4374 } 4375 4376 buffer.frameCount = mFrameCount; 4377 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4378 size_t framesOut = buffer.frameCount; 4379 if (mResampler == NULL) { 4380 // no resampling 4381 while (framesOut) { 4382 size_t framesIn = mFrameCount - mRsmpInIndex; 4383 if (framesIn) { 4384 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4385 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4386 if (framesIn > framesOut) 4387 framesIn = framesOut; 4388 mRsmpInIndex += framesIn; 4389 framesOut -= framesIn; 4390 if ((int)mChannelCount == mReqChannelCount || 4391 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4392 memcpy(dst, src, framesIn * mFrameSize); 4393 } else { 4394 int16_t *src16 = (int16_t *)src; 4395 int16_t *dst16 = (int16_t *)dst; 4396 if (mChannelCount == 1) { 4397 while (framesIn--) { 4398 *dst16++ = *src16; 4399 *dst16++ = *src16++; 4400 } 4401 } else { 4402 while (framesIn--) { 4403 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4404 src16 += 2; 4405 } 4406 } 4407 } 4408 } 4409 if (framesOut && mFrameCount == mRsmpInIndex) { 4410 if (framesOut == mFrameCount && 4411 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4412 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4413 framesOut = 0; 4414 } else { 4415 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4416 mRsmpInIndex = 0; 4417 } 4418 if (mBytesRead < 0) { 4419 ALOGE("Error reading audio input"); 4420 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4421 // Force input into standby so that it tries to 4422 // recover at next read attempt 4423 mInput->stream->common.standby(&mInput->stream->common); 4424 usleep(kRecordThreadSleepUs); 4425 } 4426 mRsmpInIndex = mFrameCount; 4427 framesOut = 0; 4428 buffer.frameCount = 0; 4429 } 4430 } 4431 } 4432 } else { 4433 // resampling 4434 4435 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4436 // alter output frame count as if we were expecting stereo samples 4437 if (mChannelCount == 1 && mReqChannelCount == 1) { 4438 framesOut >>= 1; 4439 } 4440 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4441 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4442 // are 32 bit aligned which should be always true. 4443 if (mChannelCount == 2 && mReqChannelCount == 1) { 4444 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4445 // the resampler always outputs stereo samples: do post stereo to mono conversion 4446 int16_t *src = (int16_t *)mRsmpOutBuffer; 4447 int16_t *dst = buffer.i16; 4448 while (framesOut--) { 4449 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4450 src += 2; 4451 } 4452 } else { 4453 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4454 } 4455 4456 } 4457 mActiveTrack->releaseBuffer(&buffer); 4458 mActiveTrack->overflow(); 4459 } 4460 // client isn't retrieving buffers fast enough 4461 else { 4462 if (!mActiveTrack->setOverflow()) { 4463 nsecs_t now = systemTime(); 4464 if ((now - lastWarning) > kWarningThrottleNs) { 4465 ALOGW("RecordThread: buffer overflow"); 4466 lastWarning = now; 4467 } 4468 } 4469 // Release the processor for a while before asking for a new buffer. 4470 // This will give the application more chance to read from the buffer and 4471 // clear the overflow. 4472 usleep(kRecordThreadSleepUs); 4473 } 4474 } 4475 // enable changes in effect chain 4476 unlockEffectChains(effectChains); 4477 effectChains.clear(); 4478 } 4479 4480 if (!mStandby) { 4481 mInput->stream->common.standby(&mInput->stream->common); 4482 } 4483 mActiveTrack.clear(); 4484 4485 mStartStopCond.broadcast(); 4486 4487 releaseWakeLock(); 4488 4489 ALOGV("RecordThread %p exiting", this); 4490 return false; 4491} 4492 4493 4494sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4495 const sp<AudioFlinger::Client>& client, 4496 uint32_t sampleRate, 4497 audio_format_t format, 4498 int channelMask, 4499 int frameCount, 4500 uint32_t flags, 4501 int sessionId, 4502 status_t *status) 4503{ 4504 sp<RecordTrack> track; 4505 status_t lStatus; 4506 4507 lStatus = initCheck(); 4508 if (lStatus != NO_ERROR) { 4509 ALOGE("Audio driver not initialized."); 4510 goto Exit; 4511 } 4512 4513 { // scope for mLock 4514 Mutex::Autolock _l(mLock); 4515 4516 track = new RecordTrack(this, client, sampleRate, 4517 format, channelMask, frameCount, flags, sessionId); 4518 4519 if (track->getCblk() == NULL) { 4520 lStatus = NO_MEMORY; 4521 goto Exit; 4522 } 4523 4524 mTrack = track.get(); 4525 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4526 bool suspend = audio_is_bluetooth_sco_device( 4527 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4528 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4529 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4530 } 4531 lStatus = NO_ERROR; 4532 4533Exit: 4534 if (status) { 4535 *status = lStatus; 4536 } 4537 return track; 4538} 4539 4540status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4541{ 4542 ALOGV("RecordThread::start"); 4543 sp <ThreadBase> strongMe = this; 4544 status_t status = NO_ERROR; 4545 { 4546 AutoMutex lock(mLock); 4547 if (mActiveTrack != 0) { 4548 if (recordTrack != mActiveTrack.get()) { 4549 status = -EBUSY; 4550 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4551 mActiveTrack->mState = TrackBase::ACTIVE; 4552 } 4553 return status; 4554 } 4555 4556 recordTrack->mState = TrackBase::IDLE; 4557 mActiveTrack = recordTrack; 4558 mLock.unlock(); 4559 status_t status = AudioSystem::startInput(mId); 4560 mLock.lock(); 4561 if (status != NO_ERROR) { 4562 mActiveTrack.clear(); 4563 return status; 4564 } 4565 mRsmpInIndex = mFrameCount; 4566 mBytesRead = 0; 4567 if (mResampler != NULL) { 4568 mResampler->reset(); 4569 } 4570 mActiveTrack->mState = TrackBase::RESUMING; 4571 // signal thread to start 4572 ALOGV("Signal record thread"); 4573 mWaitWorkCV.signal(); 4574 // do not wait for mStartStopCond if exiting 4575 if (mExiting) { 4576 mActiveTrack.clear(); 4577 status = INVALID_OPERATION; 4578 goto startError; 4579 } 4580 mStartStopCond.wait(mLock); 4581 if (mActiveTrack == 0) { 4582 ALOGV("Record failed to start"); 4583 status = BAD_VALUE; 4584 goto startError; 4585 } 4586 ALOGV("Record started OK"); 4587 return status; 4588 } 4589startError: 4590 AudioSystem::stopInput(mId); 4591 return status; 4592} 4593 4594void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4595 ALOGV("RecordThread::stop"); 4596 sp <ThreadBase> strongMe = this; 4597 { 4598 AutoMutex lock(mLock); 4599 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4600 mActiveTrack->mState = TrackBase::PAUSING; 4601 // do not wait for mStartStopCond if exiting 4602 if (mExiting) { 4603 return; 4604 } 4605 mStartStopCond.wait(mLock); 4606 // if we have been restarted, recordTrack == mActiveTrack.get() here 4607 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4608 mLock.unlock(); 4609 AudioSystem::stopInput(mId); 4610 mLock.lock(); 4611 ALOGV("Record stopped OK"); 4612 } 4613 } 4614 } 4615} 4616 4617status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4618{ 4619 const size_t SIZE = 256; 4620 char buffer[SIZE]; 4621 String8 result; 4622 pid_t pid = 0; 4623 4624 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4625 result.append(buffer); 4626 4627 if (mActiveTrack != 0) { 4628 result.append("Active Track:\n"); 4629 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4630 mActiveTrack->dump(buffer, SIZE); 4631 result.append(buffer); 4632 4633 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4634 result.append(buffer); 4635 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4636 result.append(buffer); 4637 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4638 result.append(buffer); 4639 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4640 result.append(buffer); 4641 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4642 result.append(buffer); 4643 4644 4645 } else { 4646 result.append("No record client\n"); 4647 } 4648 write(fd, result.string(), result.size()); 4649 4650 dumpBase(fd, args); 4651 dumpEffectChains(fd, args); 4652 4653 return NO_ERROR; 4654} 4655 4656status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4657{ 4658 size_t framesReq = buffer->frameCount; 4659 size_t framesReady = mFrameCount - mRsmpInIndex; 4660 int channelCount; 4661 4662 if (framesReady == 0) { 4663 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4664 if (mBytesRead < 0) { 4665 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4666 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4667 // Force input into standby so that it tries to 4668 // recover at next read attempt 4669 mInput->stream->common.standby(&mInput->stream->common); 4670 usleep(kRecordThreadSleepUs); 4671 } 4672 buffer->raw = NULL; 4673 buffer->frameCount = 0; 4674 return NOT_ENOUGH_DATA; 4675 } 4676 mRsmpInIndex = 0; 4677 framesReady = mFrameCount; 4678 } 4679 4680 if (framesReq > framesReady) { 4681 framesReq = framesReady; 4682 } 4683 4684 if (mChannelCount == 1 && mReqChannelCount == 2) { 4685 channelCount = 1; 4686 } else { 4687 channelCount = 2; 4688 } 4689 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4690 buffer->frameCount = framesReq; 4691 return NO_ERROR; 4692} 4693 4694void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4695{ 4696 mRsmpInIndex += buffer->frameCount; 4697 buffer->frameCount = 0; 4698} 4699 4700bool AudioFlinger::RecordThread::checkForNewParameters_l() 4701{ 4702 bool reconfig = false; 4703 4704 while (!mNewParameters.isEmpty()) { 4705 status_t status = NO_ERROR; 4706 String8 keyValuePair = mNewParameters[0]; 4707 AudioParameter param = AudioParameter(keyValuePair); 4708 int value; 4709 audio_format_t reqFormat = mFormat; 4710 int reqSamplingRate = mReqSampleRate; 4711 int reqChannelCount = mReqChannelCount; 4712 4713 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4714 reqSamplingRate = value; 4715 reconfig = true; 4716 } 4717 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4718 reqFormat = (audio_format_t) value; 4719 reconfig = true; 4720 } 4721 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4722 reqChannelCount = popcount(value); 4723 reconfig = true; 4724 } 4725 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4726 // do not accept frame count changes if tracks are open as the track buffer 4727 // size depends on frame count and correct behavior would not be garantied 4728 // if frame count is changed after track creation 4729 if (mActiveTrack != 0) { 4730 status = INVALID_OPERATION; 4731 } else { 4732 reconfig = true; 4733 } 4734 } 4735 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4736 // forward device change to effects that have requested to be 4737 // aware of attached audio device. 4738 for (size_t i = 0; i < mEffectChains.size(); i++) { 4739 mEffectChains[i]->setDevice_l(value); 4740 } 4741 // store input device and output device but do not forward output device to audio HAL. 4742 // Note that status is ignored by the caller for output device 4743 // (see AudioFlinger::setParameters() 4744 if (value & AUDIO_DEVICE_OUT_ALL) { 4745 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4746 status = BAD_VALUE; 4747 } else { 4748 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4749 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4750 if (mTrack != NULL) { 4751 bool suspend = audio_is_bluetooth_sco_device( 4752 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4753 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4754 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4755 } 4756 } 4757 mDevice |= (uint32_t)value; 4758 } 4759 if (status == NO_ERROR) { 4760 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4761 if (status == INVALID_OPERATION) { 4762 mInput->stream->common.standby(&mInput->stream->common); 4763 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4764 } 4765 if (reconfig) { 4766 if (status == BAD_VALUE && 4767 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4768 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4769 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4770 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4771 (reqChannelCount < 3)) { 4772 status = NO_ERROR; 4773 } 4774 if (status == NO_ERROR) { 4775 readInputParameters(); 4776 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4777 } 4778 } 4779 } 4780 4781 mNewParameters.removeAt(0); 4782 4783 mParamStatus = status; 4784 mParamCond.signal(); 4785 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4786 // already timed out waiting for the status and will never signal the condition. 4787 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4788 } 4789 return reconfig; 4790} 4791 4792String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4793{ 4794 char *s; 4795 String8 out_s8 = String8(); 4796 4797 Mutex::Autolock _l(mLock); 4798 if (initCheck() != NO_ERROR) { 4799 return out_s8; 4800 } 4801 4802 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4803 out_s8 = String8(s); 4804 free(s); 4805 return out_s8; 4806} 4807 4808void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4809 AudioSystem::OutputDescriptor desc; 4810 void *param2 = 0; 4811 4812 switch (event) { 4813 case AudioSystem::INPUT_OPENED: 4814 case AudioSystem::INPUT_CONFIG_CHANGED: 4815 desc.channels = mChannelMask; 4816 desc.samplingRate = mSampleRate; 4817 desc.format = mFormat; 4818 desc.frameCount = mFrameCount; 4819 desc.latency = 0; 4820 param2 = &desc; 4821 break; 4822 4823 case AudioSystem::INPUT_CLOSED: 4824 default: 4825 break; 4826 } 4827 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4828} 4829 4830void AudioFlinger::RecordThread::readInputParameters() 4831{ 4832 delete mRsmpInBuffer; 4833 // mRsmpInBuffer is always assigned a new[] below 4834 delete mRsmpOutBuffer; 4835 mRsmpOutBuffer = NULL; 4836 delete mResampler; 4837 mResampler = NULL; 4838 4839 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4840 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4841 mChannelCount = (uint16_t)popcount(mChannelMask); 4842 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4843 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4844 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4845 mFrameCount = mInputBytes / mFrameSize; 4846 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4847 4848 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4849 { 4850 int channelCount; 4851 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4852 // stereo to mono post process as the resampler always outputs stereo. 4853 if (mChannelCount == 1 && mReqChannelCount == 2) { 4854 channelCount = 1; 4855 } else { 4856 channelCount = 2; 4857 } 4858 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4859 mResampler->setSampleRate(mSampleRate); 4860 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4861 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4862 4863 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4864 if (mChannelCount == 1 && mReqChannelCount == 1) { 4865 mFrameCount >>= 1; 4866 } 4867 4868 } 4869 mRsmpInIndex = mFrameCount; 4870} 4871 4872unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4873{ 4874 Mutex::Autolock _l(mLock); 4875 if (initCheck() != NO_ERROR) { 4876 return 0; 4877 } 4878 4879 return mInput->stream->get_input_frames_lost(mInput->stream); 4880} 4881 4882uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4883{ 4884 Mutex::Autolock _l(mLock); 4885 uint32_t result = 0; 4886 if (getEffectChain_l(sessionId) != 0) { 4887 result = EFFECT_SESSION; 4888 } 4889 4890 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4891 result |= TRACK_SESSION; 4892 } 4893 4894 return result; 4895} 4896 4897AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4898{ 4899 Mutex::Autolock _l(mLock); 4900 return mTrack; 4901} 4902 4903AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4904{ 4905 Mutex::Autolock _l(mLock); 4906 return mInput; 4907} 4908 4909AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4910{ 4911 Mutex::Autolock _l(mLock); 4912 AudioStreamIn *input = mInput; 4913 mInput = NULL; 4914 return input; 4915} 4916 4917// this method must always be called either with ThreadBase mLock held or inside the thread loop 4918audio_stream_t* AudioFlinger::RecordThread::stream() 4919{ 4920 if (mInput == NULL) { 4921 return NULL; 4922 } 4923 return &mInput->stream->common; 4924} 4925 4926 4927// ---------------------------------------------------------------------------- 4928 4929int AudioFlinger::openOutput(uint32_t *pDevices, 4930 uint32_t *pSamplingRate, 4931 audio_format_t *pFormat, 4932 uint32_t *pChannels, 4933 uint32_t *pLatencyMs, 4934 uint32_t flags) 4935{ 4936 status_t status; 4937 PlaybackThread *thread = NULL; 4938 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4939 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4940 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4941 uint32_t channels = pChannels ? *pChannels : 0; 4942 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4943 audio_stream_out_t *outStream; 4944 audio_hw_device_t *outHwDev; 4945 4946 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4947 pDevices ? *pDevices : 0, 4948 samplingRate, 4949 format, 4950 channels, 4951 flags); 4952 4953 if (pDevices == NULL || *pDevices == 0) { 4954 return 0; 4955 } 4956 4957 Mutex::Autolock _l(mLock); 4958 4959 outHwDev = findSuitableHwDev_l(*pDevices); 4960 if (outHwDev == NULL) 4961 return 0; 4962 4963 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4964 &channels, &samplingRate, &outStream); 4965 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4966 outStream, 4967 samplingRate, 4968 format, 4969 channels, 4970 status); 4971 4972 mHardwareStatus = AUDIO_HW_IDLE; 4973 if (outStream != NULL) { 4974 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4975 int id = nextUniqueId(); 4976 4977 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4978 (format != AUDIO_FORMAT_PCM_16_BIT) || 4979 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4980 thread = new DirectOutputThread(this, output, id, *pDevices); 4981 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4982 } else { 4983 thread = new MixerThread(this, output, id, *pDevices); 4984 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4985 } 4986 mPlaybackThreads.add(id, thread); 4987 4988 if (pSamplingRate) *pSamplingRate = samplingRate; 4989 if (pFormat) *pFormat = format; 4990 if (pChannels) *pChannels = channels; 4991 if (pLatencyMs) *pLatencyMs = thread->latency(); 4992 4993 // notify client processes of the new output creation 4994 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4995 return id; 4996 } 4997 4998 return 0; 4999} 5000 5001int AudioFlinger::openDuplicateOutput(int output1, int output2) 5002{ 5003 Mutex::Autolock _l(mLock); 5004 MixerThread *thread1 = checkMixerThread_l(output1); 5005 MixerThread *thread2 = checkMixerThread_l(output2); 5006 5007 if (thread1 == NULL || thread2 == NULL) { 5008 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5009 return 0; 5010 } 5011 5012 int id = nextUniqueId(); 5013 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5014 thread->addOutputTrack(thread2); 5015 mPlaybackThreads.add(id, thread); 5016 // notify client processes of the new output creation 5017 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5018 return id; 5019} 5020 5021status_t AudioFlinger::closeOutput(int output) 5022{ 5023 // keep strong reference on the playback thread so that 5024 // it is not destroyed while exit() is executed 5025 sp <PlaybackThread> thread; 5026 { 5027 Mutex::Autolock _l(mLock); 5028 thread = checkPlaybackThread_l(output); 5029 if (thread == NULL) { 5030 return BAD_VALUE; 5031 } 5032 5033 ALOGV("closeOutput() %d", output); 5034 5035 if (thread->type() == ThreadBase::MIXER) { 5036 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5037 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5038 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5039 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5040 } 5041 } 5042 } 5043 void *param2 = 0; 5044 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5045 mPlaybackThreads.removeItem(output); 5046 } 5047 thread->exit(); 5048 5049 if (thread->type() != ThreadBase::DUPLICATING) { 5050 AudioStreamOut *out = thread->clearOutput(); 5051 assert(out != NULL); 5052 // from now on thread->mOutput is NULL 5053 out->hwDev->close_output_stream(out->hwDev, out->stream); 5054 delete out; 5055 } 5056 return NO_ERROR; 5057} 5058 5059status_t AudioFlinger::suspendOutput(int output) 5060{ 5061 Mutex::Autolock _l(mLock); 5062 PlaybackThread *thread = checkPlaybackThread_l(output); 5063 5064 if (thread == NULL) { 5065 return BAD_VALUE; 5066 } 5067 5068 ALOGV("suspendOutput() %d", output); 5069 thread->suspend(); 5070 5071 return NO_ERROR; 5072} 5073 5074status_t AudioFlinger::restoreOutput(int output) 5075{ 5076 Mutex::Autolock _l(mLock); 5077 PlaybackThread *thread = checkPlaybackThread_l(output); 5078 5079 if (thread == NULL) { 5080 return BAD_VALUE; 5081 } 5082 5083 ALOGV("restoreOutput() %d", output); 5084 5085 thread->restore(); 5086 5087 return NO_ERROR; 5088} 5089 5090int AudioFlinger::openInput(uint32_t *pDevices, 5091 uint32_t *pSamplingRate, 5092 audio_format_t *pFormat, 5093 uint32_t *pChannels, 5094 uint32_t acoustics) 5095{ 5096 status_t status; 5097 RecordThread *thread = NULL; 5098 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5099 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5100 uint32_t channels = pChannels ? *pChannels : 0; 5101 uint32_t reqSamplingRate = samplingRate; 5102 audio_format_t reqFormat = format; 5103 uint32_t reqChannels = channels; 5104 audio_stream_in_t *inStream; 5105 audio_hw_device_t *inHwDev; 5106 5107 if (pDevices == NULL || *pDevices == 0) { 5108 return 0; 5109 } 5110 5111 Mutex::Autolock _l(mLock); 5112 5113 inHwDev = findSuitableHwDev_l(*pDevices); 5114 if (inHwDev == NULL) 5115 return 0; 5116 5117 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5118 &channels, &samplingRate, 5119 (audio_in_acoustics_t)acoustics, 5120 &inStream); 5121 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5122 inStream, 5123 samplingRate, 5124 format, 5125 channels, 5126 acoustics, 5127 status); 5128 5129 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5130 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5131 // or stereo to mono conversions on 16 bit PCM inputs. 5132 if (inStream == NULL && status == BAD_VALUE && 5133 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5134 (samplingRate <= 2 * reqSamplingRate) && 5135 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5136 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5137 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5138 &channels, &samplingRate, 5139 (audio_in_acoustics_t)acoustics, 5140 &inStream); 5141 } 5142 5143 if (inStream != NULL) { 5144 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5145 5146 int id = nextUniqueId(); 5147 // Start record thread 5148 // RecorThread require both input and output device indication to forward to audio 5149 // pre processing modules 5150 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5151 thread = new RecordThread(this, 5152 input, 5153 reqSamplingRate, 5154 reqChannels, 5155 id, 5156 device); 5157 mRecordThreads.add(id, thread); 5158 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5159 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5160 if (pFormat) *pFormat = format; 5161 if (pChannels) *pChannels = reqChannels; 5162 5163 input->stream->common.standby(&input->stream->common); 5164 5165 // notify client processes of the new input creation 5166 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5167 return id; 5168 } 5169 5170 return 0; 5171} 5172 5173status_t AudioFlinger::closeInput(int input) 5174{ 5175 // keep strong reference on the record thread so that 5176 // it is not destroyed while exit() is executed 5177 sp <RecordThread> thread; 5178 { 5179 Mutex::Autolock _l(mLock); 5180 thread = checkRecordThread_l(input); 5181 if (thread == NULL) { 5182 return BAD_VALUE; 5183 } 5184 5185 ALOGV("closeInput() %d", input); 5186 void *param2 = 0; 5187 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5188 mRecordThreads.removeItem(input); 5189 } 5190 thread->exit(); 5191 5192 AudioStreamIn *in = thread->clearInput(); 5193 assert(in != NULL); 5194 // from now on thread->mInput is NULL 5195 in->hwDev->close_input_stream(in->hwDev, in->stream); 5196 delete in; 5197 5198 return NO_ERROR; 5199} 5200 5201status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5202{ 5203 Mutex::Autolock _l(mLock); 5204 MixerThread *dstThread = checkMixerThread_l(output); 5205 if (dstThread == NULL) { 5206 ALOGW("setStreamOutput() bad output id %d", output); 5207 return BAD_VALUE; 5208 } 5209 5210 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5211 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5212 5213 dstThread->setStreamValid(stream, true); 5214 5215 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5216 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5217 if (thread != dstThread && 5218 thread->type() != ThreadBase::DIRECT) { 5219 MixerThread *srcThread = (MixerThread *)thread; 5220 srcThread->setStreamValid(stream, false); 5221 srcThread->invalidateTracks(stream); 5222 } 5223 } 5224 5225 return NO_ERROR; 5226} 5227 5228 5229int AudioFlinger::newAudioSessionId() 5230{ 5231 return nextUniqueId(); 5232} 5233 5234void AudioFlinger::acquireAudioSessionId(int audioSession) 5235{ 5236 Mutex::Autolock _l(mLock); 5237 int caller = IPCThreadState::self()->getCallingPid(); 5238 ALOGV("acquiring %d from %d", audioSession, caller); 5239 int num = mAudioSessionRefs.size(); 5240 for (int i = 0; i< num; i++) { 5241 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5242 if (ref->sessionid == audioSession && ref->pid == caller) { 5243 ref->cnt++; 5244 ALOGV(" incremented refcount to %d", ref->cnt); 5245 return; 5246 } 5247 } 5248 AudioSessionRef *ref = new AudioSessionRef(); 5249 ref->sessionid = audioSession; 5250 ref->pid = caller; 5251 ref->cnt = 1; 5252 mAudioSessionRefs.push(ref); 5253 ALOGV(" added new entry for %d", ref->sessionid); 5254} 5255 5256void AudioFlinger::releaseAudioSessionId(int audioSession) 5257{ 5258 Mutex::Autolock _l(mLock); 5259 int caller = IPCThreadState::self()->getCallingPid(); 5260 ALOGV("releasing %d from %d", audioSession, caller); 5261 int num = mAudioSessionRefs.size(); 5262 for (int i = 0; i< num; i++) { 5263 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5264 if (ref->sessionid == audioSession && ref->pid == caller) { 5265 ref->cnt--; 5266 ALOGV(" decremented refcount to %d", ref->cnt); 5267 if (ref->cnt == 0) { 5268 mAudioSessionRefs.removeAt(i); 5269 delete ref; 5270 purgeStaleEffects_l(); 5271 } 5272 return; 5273 } 5274 } 5275 ALOGW("session id %d not found for pid %d", audioSession, caller); 5276} 5277 5278void AudioFlinger::purgeStaleEffects_l() { 5279 5280 ALOGV("purging stale effects"); 5281 5282 Vector< sp<EffectChain> > chains; 5283 5284 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5285 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5286 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5287 sp<EffectChain> ec = t->mEffectChains[j]; 5288 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5289 chains.push(ec); 5290 } 5291 } 5292 } 5293 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5294 sp<RecordThread> t = mRecordThreads.valueAt(i); 5295 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5296 sp<EffectChain> ec = t->mEffectChains[j]; 5297 chains.push(ec); 5298 } 5299 } 5300 5301 for (size_t i = 0; i < chains.size(); i++) { 5302 sp<EffectChain> ec = chains[i]; 5303 int sessionid = ec->sessionId(); 5304 sp<ThreadBase> t = ec->mThread.promote(); 5305 if (t == 0) { 5306 continue; 5307 } 5308 size_t numsessionrefs = mAudioSessionRefs.size(); 5309 bool found = false; 5310 for (size_t k = 0; k < numsessionrefs; k++) { 5311 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5312 if (ref->sessionid == sessionid) { 5313 ALOGV(" session %d still exists for %d with %d refs", 5314 sessionid, ref->pid, ref->cnt); 5315 found = true; 5316 break; 5317 } 5318 } 5319 if (!found) { 5320 // remove all effects from the chain 5321 while (ec->mEffects.size()) { 5322 sp<EffectModule> effect = ec->mEffects[0]; 5323 effect->unPin(); 5324 Mutex::Autolock _l (t->mLock); 5325 t->removeEffect_l(effect); 5326 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5327 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5328 if (handle != 0) { 5329 handle->mEffect.clear(); 5330 if (handle->mHasControl && handle->mEnabled) { 5331 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5332 } 5333 } 5334 } 5335 AudioSystem::unregisterEffect(effect->id()); 5336 } 5337 } 5338 } 5339 return; 5340} 5341 5342// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5343AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5344{ 5345 PlaybackThread *thread = NULL; 5346 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5347 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5348 } 5349 return thread; 5350} 5351 5352// checkMixerThread_l() must be called with AudioFlinger::mLock held 5353AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5354{ 5355 PlaybackThread *thread = checkPlaybackThread_l(output); 5356 if (thread != NULL) { 5357 if (thread->type() == ThreadBase::DIRECT) { 5358 thread = NULL; 5359 } 5360 } 5361 return (MixerThread *)thread; 5362} 5363 5364// checkRecordThread_l() must be called with AudioFlinger::mLock held 5365AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5366{ 5367 RecordThread *thread = NULL; 5368 if (mRecordThreads.indexOfKey(input) >= 0) { 5369 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5370 } 5371 return thread; 5372} 5373 5374uint32_t AudioFlinger::nextUniqueId() 5375{ 5376 return android_atomic_inc(&mNextUniqueId); 5377} 5378 5379AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5380{ 5381 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5382 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5383 AudioStreamOut *output = thread->getOutput(); 5384 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5385 return thread; 5386 } 5387 } 5388 return NULL; 5389} 5390 5391uint32_t AudioFlinger::primaryOutputDevice_l() 5392{ 5393 PlaybackThread *thread = primaryPlaybackThread_l(); 5394 5395 if (thread == NULL) { 5396 return 0; 5397 } 5398 5399 return thread->device(); 5400} 5401 5402 5403// ---------------------------------------------------------------------------- 5404// Effect management 5405// ---------------------------------------------------------------------------- 5406 5407 5408status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5409{ 5410 Mutex::Autolock _l(mLock); 5411 return EffectQueryNumberEffects(numEffects); 5412} 5413 5414status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5415{ 5416 Mutex::Autolock _l(mLock); 5417 return EffectQueryEffect(index, descriptor); 5418} 5419 5420status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5421{ 5422 Mutex::Autolock _l(mLock); 5423 return EffectGetDescriptor(pUuid, descriptor); 5424} 5425 5426 5427sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5428 effect_descriptor_t *pDesc, 5429 const sp<IEffectClient>& effectClient, 5430 int32_t priority, 5431 int io, 5432 int sessionId, 5433 status_t *status, 5434 int *id, 5435 int *enabled) 5436{ 5437 status_t lStatus = NO_ERROR; 5438 sp<EffectHandle> handle; 5439 effect_descriptor_t desc; 5440 sp<Client> client; 5441 wp<Client> wclient; 5442 5443 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5444 pid, effectClient.get(), priority, sessionId, io); 5445 5446 if (pDesc == NULL) { 5447 lStatus = BAD_VALUE; 5448 goto Exit; 5449 } 5450 5451 // check audio settings permission for global effects 5452 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5453 lStatus = PERMISSION_DENIED; 5454 goto Exit; 5455 } 5456 5457 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5458 // that can only be created by audio policy manager (running in same process) 5459 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5460 lStatus = PERMISSION_DENIED; 5461 goto Exit; 5462 } 5463 5464 if (io == 0) { 5465 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5466 // output must be specified by AudioPolicyManager when using session 5467 // AUDIO_SESSION_OUTPUT_STAGE 5468 lStatus = BAD_VALUE; 5469 goto Exit; 5470 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5471 // if the output returned by getOutputForEffect() is removed before we lock the 5472 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5473 // and we will exit safely 5474 io = AudioSystem::getOutputForEffect(&desc); 5475 } 5476 } 5477 5478 { 5479 Mutex::Autolock _l(mLock); 5480 5481 5482 if (!EffectIsNullUuid(&pDesc->uuid)) { 5483 // if uuid is specified, request effect descriptor 5484 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5485 if (lStatus < 0) { 5486 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5487 goto Exit; 5488 } 5489 } else { 5490 // if uuid is not specified, look for an available implementation 5491 // of the required type in effect factory 5492 if (EffectIsNullUuid(&pDesc->type)) { 5493 ALOGW("createEffect() no effect type"); 5494 lStatus = BAD_VALUE; 5495 goto Exit; 5496 } 5497 uint32_t numEffects = 0; 5498 effect_descriptor_t d; 5499 d.flags = 0; // prevent compiler warning 5500 bool found = false; 5501 5502 lStatus = EffectQueryNumberEffects(&numEffects); 5503 if (lStatus < 0) { 5504 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5505 goto Exit; 5506 } 5507 for (uint32_t i = 0; i < numEffects; i++) { 5508 lStatus = EffectQueryEffect(i, &desc); 5509 if (lStatus < 0) { 5510 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5511 continue; 5512 } 5513 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5514 // If matching type found save effect descriptor. If the session is 5515 // 0 and the effect is not auxiliary, continue enumeration in case 5516 // an auxiliary version of this effect type is available 5517 found = true; 5518 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5519 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5520 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5521 break; 5522 } 5523 } 5524 } 5525 if (!found) { 5526 lStatus = BAD_VALUE; 5527 ALOGW("createEffect() effect not found"); 5528 goto Exit; 5529 } 5530 // For same effect type, chose auxiliary version over insert version if 5531 // connect to output mix (Compliance to OpenSL ES) 5532 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5533 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5534 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5535 } 5536 } 5537 5538 // Do not allow auxiliary effects on a session different from 0 (output mix) 5539 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5540 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5541 lStatus = INVALID_OPERATION; 5542 goto Exit; 5543 } 5544 5545 // check recording permission for visualizer 5546 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5547 !recordingAllowed()) { 5548 lStatus = PERMISSION_DENIED; 5549 goto Exit; 5550 } 5551 5552 // return effect descriptor 5553 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5554 5555 // If output is not specified try to find a matching audio session ID in one of the 5556 // output threads. 5557 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5558 // because of code checking output when entering the function. 5559 // Note: io is never 0 when creating an effect on an input 5560 if (io == 0) { 5561 // look for the thread where the specified audio session is present 5562 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5563 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5564 io = mPlaybackThreads.keyAt(i); 5565 break; 5566 } 5567 } 5568 if (io == 0) { 5569 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5570 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5571 io = mRecordThreads.keyAt(i); 5572 break; 5573 } 5574 } 5575 } 5576 // If no output thread contains the requested session ID, default to 5577 // first output. The effect chain will be moved to the correct output 5578 // thread when a track with the same session ID is created 5579 if (io == 0 && mPlaybackThreads.size()) { 5580 io = mPlaybackThreads.keyAt(0); 5581 } 5582 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5583 } 5584 ThreadBase *thread = checkRecordThread_l(io); 5585 if (thread == NULL) { 5586 thread = checkPlaybackThread_l(io); 5587 if (thread == NULL) { 5588 ALOGE("createEffect() unknown output thread"); 5589 lStatus = BAD_VALUE; 5590 goto Exit; 5591 } 5592 } 5593 5594 wclient = mClients.valueFor(pid); 5595 5596 if (wclient != NULL) { 5597 client = wclient.promote(); 5598 } else { 5599 client = new Client(this, pid); 5600 mClients.add(pid, client); 5601 } 5602 5603 // create effect on selected output thread 5604 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5605 &desc, enabled, &lStatus); 5606 if (handle != 0 && id != NULL) { 5607 *id = handle->id(); 5608 } 5609 } 5610 5611Exit: 5612 if(status) { 5613 *status = lStatus; 5614 } 5615 return handle; 5616} 5617 5618status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5619{ 5620 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5621 sessionId, srcOutput, dstOutput); 5622 Mutex::Autolock _l(mLock); 5623 if (srcOutput == dstOutput) { 5624 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5625 return NO_ERROR; 5626 } 5627 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5628 if (srcThread == NULL) { 5629 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5630 return BAD_VALUE; 5631 } 5632 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5633 if (dstThread == NULL) { 5634 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5635 return BAD_VALUE; 5636 } 5637 5638 Mutex::Autolock _dl(dstThread->mLock); 5639 Mutex::Autolock _sl(srcThread->mLock); 5640 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5641 5642 return NO_ERROR; 5643} 5644 5645// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5646status_t AudioFlinger::moveEffectChain_l(int sessionId, 5647 AudioFlinger::PlaybackThread *srcThread, 5648 AudioFlinger::PlaybackThread *dstThread, 5649 bool reRegister) 5650{ 5651 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5652 sessionId, srcThread, dstThread); 5653 5654 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5655 if (chain == 0) { 5656 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5657 sessionId, srcThread); 5658 return INVALID_OPERATION; 5659 } 5660 5661 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5662 // so that a new chain is created with correct parameters when first effect is added. This is 5663 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5664 // removed. 5665 srcThread->removeEffectChain_l(chain); 5666 5667 // transfer all effects one by one so that new effect chain is created on new thread with 5668 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5669 int dstOutput = dstThread->id(); 5670 sp<EffectChain> dstChain; 5671 uint32_t strategy = 0; // prevent compiler warning 5672 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5673 while (effect != 0) { 5674 srcThread->removeEffect_l(effect); 5675 dstThread->addEffect_l(effect); 5676 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5677 if (effect->state() == EffectModule::ACTIVE || 5678 effect->state() == EffectModule::STOPPING) { 5679 effect->start(); 5680 } 5681 // if the move request is not received from audio policy manager, the effect must be 5682 // re-registered with the new strategy and output 5683 if (dstChain == 0) { 5684 dstChain = effect->chain().promote(); 5685 if (dstChain == 0) { 5686 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5687 srcThread->addEffect_l(effect); 5688 return NO_INIT; 5689 } 5690 strategy = dstChain->strategy(); 5691 } 5692 if (reRegister) { 5693 AudioSystem::unregisterEffect(effect->id()); 5694 AudioSystem::registerEffect(&effect->desc(), 5695 dstOutput, 5696 strategy, 5697 sessionId, 5698 effect->id()); 5699 } 5700 effect = chain->getEffectFromId_l(0); 5701 } 5702 5703 return NO_ERROR; 5704} 5705 5706 5707// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5708sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5709 const sp<AudioFlinger::Client>& client, 5710 const sp<IEffectClient>& effectClient, 5711 int32_t priority, 5712 int sessionId, 5713 effect_descriptor_t *desc, 5714 int *enabled, 5715 status_t *status 5716 ) 5717{ 5718 sp<EffectModule> effect; 5719 sp<EffectHandle> handle; 5720 status_t lStatus; 5721 sp<EffectChain> chain; 5722 bool chainCreated = false; 5723 bool effectCreated = false; 5724 bool effectRegistered = false; 5725 5726 lStatus = initCheck(); 5727 if (lStatus != NO_ERROR) { 5728 ALOGW("createEffect_l() Audio driver not initialized."); 5729 goto Exit; 5730 } 5731 5732 // Do not allow effects with session ID 0 on direct output or duplicating threads 5733 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5734 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5735 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5736 desc->name, sessionId); 5737 lStatus = BAD_VALUE; 5738 goto Exit; 5739 } 5740 // Only Pre processor effects are allowed on input threads and only on input threads 5741 if ((mType == RECORD && 5742 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5743 (mType != RECORD && 5744 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5745 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5746 desc->name, desc->flags, mType); 5747 lStatus = BAD_VALUE; 5748 goto Exit; 5749 } 5750 5751 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5752 5753 { // scope for mLock 5754 Mutex::Autolock _l(mLock); 5755 5756 // check for existing effect chain with the requested audio session 5757 chain = getEffectChain_l(sessionId); 5758 if (chain == 0) { 5759 // create a new chain for this session 5760 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5761 chain = new EffectChain(this, sessionId); 5762 addEffectChain_l(chain); 5763 chain->setStrategy(getStrategyForSession_l(sessionId)); 5764 chainCreated = true; 5765 } else { 5766 effect = chain->getEffectFromDesc_l(desc); 5767 } 5768 5769 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5770 5771 if (effect == 0) { 5772 int id = mAudioFlinger->nextUniqueId(); 5773 // Check CPU and memory usage 5774 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5775 if (lStatus != NO_ERROR) { 5776 goto Exit; 5777 } 5778 effectRegistered = true; 5779 // create a new effect module if none present in the chain 5780 effect = new EffectModule(this, chain, desc, id, sessionId); 5781 lStatus = effect->status(); 5782 if (lStatus != NO_ERROR) { 5783 goto Exit; 5784 } 5785 lStatus = chain->addEffect_l(effect); 5786 if (lStatus != NO_ERROR) { 5787 goto Exit; 5788 } 5789 effectCreated = true; 5790 5791 effect->setDevice(mDevice); 5792 effect->setMode(mAudioFlinger->getMode()); 5793 } 5794 // create effect handle and connect it to effect module 5795 handle = new EffectHandle(effect, client, effectClient, priority); 5796 lStatus = effect->addHandle(handle); 5797 if (enabled) { 5798 *enabled = (int)effect->isEnabled(); 5799 } 5800 } 5801 5802Exit: 5803 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5804 Mutex::Autolock _l(mLock); 5805 if (effectCreated) { 5806 chain->removeEffect_l(effect); 5807 } 5808 if (effectRegistered) { 5809 AudioSystem::unregisterEffect(effect->id()); 5810 } 5811 if (chainCreated) { 5812 removeEffectChain_l(chain); 5813 } 5814 handle.clear(); 5815 } 5816 5817 if(status) { 5818 *status = lStatus; 5819 } 5820 return handle; 5821} 5822 5823sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5824{ 5825 sp<EffectModule> effect; 5826 5827 sp<EffectChain> chain = getEffectChain_l(sessionId); 5828 if (chain != 0) { 5829 effect = chain->getEffectFromId_l(effectId); 5830 } 5831 return effect; 5832} 5833 5834// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5835// PlaybackThread::mLock held 5836status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5837{ 5838 // check for existing effect chain with the requested audio session 5839 int sessionId = effect->sessionId(); 5840 sp<EffectChain> chain = getEffectChain_l(sessionId); 5841 bool chainCreated = false; 5842 5843 if (chain == 0) { 5844 // create a new chain for this session 5845 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5846 chain = new EffectChain(this, sessionId); 5847 addEffectChain_l(chain); 5848 chain->setStrategy(getStrategyForSession_l(sessionId)); 5849 chainCreated = true; 5850 } 5851 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5852 5853 if (chain->getEffectFromId_l(effect->id()) != 0) { 5854 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5855 this, effect->desc().name, chain.get()); 5856 return BAD_VALUE; 5857 } 5858 5859 status_t status = chain->addEffect_l(effect); 5860 if (status != NO_ERROR) { 5861 if (chainCreated) { 5862 removeEffectChain_l(chain); 5863 } 5864 return status; 5865 } 5866 5867 effect->setDevice(mDevice); 5868 effect->setMode(mAudioFlinger->getMode()); 5869 return NO_ERROR; 5870} 5871 5872void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5873 5874 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5875 effect_descriptor_t desc = effect->desc(); 5876 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5877 detachAuxEffect_l(effect->id()); 5878 } 5879 5880 sp<EffectChain> chain = effect->chain().promote(); 5881 if (chain != 0) { 5882 // remove effect chain if removing last effect 5883 if (chain->removeEffect_l(effect) == 0) { 5884 removeEffectChain_l(chain); 5885 } 5886 } else { 5887 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5888 } 5889} 5890 5891void AudioFlinger::ThreadBase::lockEffectChains_l( 5892 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5893{ 5894 effectChains = mEffectChains; 5895 for (size_t i = 0; i < mEffectChains.size(); i++) { 5896 mEffectChains[i]->lock(); 5897 } 5898} 5899 5900void AudioFlinger::ThreadBase::unlockEffectChains( 5901 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5902{ 5903 for (size_t i = 0; i < effectChains.size(); i++) { 5904 effectChains[i]->unlock(); 5905 } 5906} 5907 5908sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5909{ 5910 Mutex::Autolock _l(mLock); 5911 return getEffectChain_l(sessionId); 5912} 5913 5914sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5915{ 5916 sp<EffectChain> chain; 5917 5918 size_t size = mEffectChains.size(); 5919 for (size_t i = 0; i < size; i++) { 5920 if (mEffectChains[i]->sessionId() == sessionId) { 5921 chain = mEffectChains[i]; 5922 break; 5923 } 5924 } 5925 return chain; 5926} 5927 5928void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5929{ 5930 Mutex::Autolock _l(mLock); 5931 size_t size = mEffectChains.size(); 5932 for (size_t i = 0; i < size; i++) { 5933 mEffectChains[i]->setMode_l(mode); 5934 } 5935} 5936 5937void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5938 const wp<EffectHandle>& handle, 5939 bool unpiniflast) { 5940 5941 Mutex::Autolock _l(mLock); 5942 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5943 // delete the effect module if removing last handle on it 5944 if (effect->removeHandle(handle) == 0) { 5945 if (!effect->isPinned() || unpiniflast) { 5946 removeEffect_l(effect); 5947 AudioSystem::unregisterEffect(effect->id()); 5948 } 5949 } 5950} 5951 5952status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5953{ 5954 int session = chain->sessionId(); 5955 int16_t *buffer = mMixBuffer; 5956 bool ownsBuffer = false; 5957 5958 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5959 if (session > 0) { 5960 // Only one effect chain can be present in direct output thread and it uses 5961 // the mix buffer as input 5962 if (mType != DIRECT) { 5963 size_t numSamples = mFrameCount * mChannelCount; 5964 buffer = new int16_t[numSamples]; 5965 memset(buffer, 0, numSamples * sizeof(int16_t)); 5966 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5967 ownsBuffer = true; 5968 } 5969 5970 // Attach all tracks with same session ID to this chain. 5971 for (size_t i = 0; i < mTracks.size(); ++i) { 5972 sp<Track> track = mTracks[i]; 5973 if (session == track->sessionId()) { 5974 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5975 track->setMainBuffer(buffer); 5976 chain->incTrackCnt(); 5977 } 5978 } 5979 5980 // indicate all active tracks in the chain 5981 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5982 sp<Track> track = mActiveTracks[i].promote(); 5983 if (track == 0) continue; 5984 if (session == track->sessionId()) { 5985 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5986 chain->incActiveTrackCnt(); 5987 } 5988 } 5989 } 5990 5991 chain->setInBuffer(buffer, ownsBuffer); 5992 chain->setOutBuffer(mMixBuffer); 5993 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5994 // chains list in order to be processed last as it contains output stage effects 5995 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5996 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5997 // after track specific effects and before output stage 5998 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5999 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6000 // Effect chain for other sessions are inserted at beginning of effect 6001 // chains list to be processed before output mix effects. Relative order between other 6002 // sessions is not important 6003 size_t size = mEffectChains.size(); 6004 size_t i = 0; 6005 for (i = 0; i < size; i++) { 6006 if (mEffectChains[i]->sessionId() < session) break; 6007 } 6008 mEffectChains.insertAt(chain, i); 6009 checkSuspendOnAddEffectChain_l(chain); 6010 6011 return NO_ERROR; 6012} 6013 6014size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6015{ 6016 int session = chain->sessionId(); 6017 6018 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6019 6020 for (size_t i = 0; i < mEffectChains.size(); i++) { 6021 if (chain == mEffectChains[i]) { 6022 mEffectChains.removeAt(i); 6023 // detach all active tracks from the chain 6024 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6025 sp<Track> track = mActiveTracks[i].promote(); 6026 if (track == 0) continue; 6027 if (session == track->sessionId()) { 6028 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6029 chain.get(), session); 6030 chain->decActiveTrackCnt(); 6031 } 6032 } 6033 6034 // detach all tracks with same session ID from this chain 6035 for (size_t i = 0; i < mTracks.size(); ++i) { 6036 sp<Track> track = mTracks[i]; 6037 if (session == track->sessionId()) { 6038 track->setMainBuffer(mMixBuffer); 6039 chain->decTrackCnt(); 6040 } 6041 } 6042 break; 6043 } 6044 } 6045 return mEffectChains.size(); 6046} 6047 6048status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6049 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6050{ 6051 Mutex::Autolock _l(mLock); 6052 return attachAuxEffect_l(track, EffectId); 6053} 6054 6055status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6056 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6057{ 6058 status_t status = NO_ERROR; 6059 6060 if (EffectId == 0) { 6061 track->setAuxBuffer(0, NULL); 6062 } else { 6063 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6064 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6065 if (effect != 0) { 6066 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6067 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6068 } else { 6069 status = INVALID_OPERATION; 6070 } 6071 } else { 6072 status = BAD_VALUE; 6073 } 6074 } 6075 return status; 6076} 6077 6078void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6079{ 6080 for (size_t i = 0; i < mTracks.size(); ++i) { 6081 sp<Track> track = mTracks[i]; 6082 if (track->auxEffectId() == effectId) { 6083 attachAuxEffect_l(track, 0); 6084 } 6085 } 6086} 6087 6088status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6089{ 6090 // only one chain per input thread 6091 if (mEffectChains.size() != 0) { 6092 return INVALID_OPERATION; 6093 } 6094 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6095 6096 chain->setInBuffer(NULL); 6097 chain->setOutBuffer(NULL); 6098 6099 checkSuspendOnAddEffectChain_l(chain); 6100 6101 mEffectChains.add(chain); 6102 6103 return NO_ERROR; 6104} 6105 6106size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6107{ 6108 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6109 ALOGW_IF(mEffectChains.size() != 1, 6110 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6111 chain.get(), mEffectChains.size(), this); 6112 if (mEffectChains.size() == 1) { 6113 mEffectChains.removeAt(0); 6114 } 6115 return 0; 6116} 6117 6118// ---------------------------------------------------------------------------- 6119// EffectModule implementation 6120// ---------------------------------------------------------------------------- 6121 6122#undef LOG_TAG 6123#define LOG_TAG "AudioFlinger::EffectModule" 6124 6125AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6126 const wp<AudioFlinger::EffectChain>& chain, 6127 effect_descriptor_t *desc, 6128 int id, 6129 int sessionId) 6130 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6131 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6132{ 6133 ALOGV("Constructor %p", this); 6134 int lStatus; 6135 sp<ThreadBase> thread = mThread.promote(); 6136 if (thread == 0) { 6137 return; 6138 } 6139 6140 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6141 6142 // create effect engine from effect factory 6143 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6144 6145 if (mStatus != NO_ERROR) { 6146 return; 6147 } 6148 lStatus = init(); 6149 if (lStatus < 0) { 6150 mStatus = lStatus; 6151 goto Error; 6152 } 6153 6154 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6155 mPinned = true; 6156 } 6157 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6158 return; 6159Error: 6160 EffectRelease(mEffectInterface); 6161 mEffectInterface = NULL; 6162 ALOGV("Constructor Error %d", mStatus); 6163} 6164 6165AudioFlinger::EffectModule::~EffectModule() 6166{ 6167 ALOGV("Destructor %p", this); 6168 if (mEffectInterface != NULL) { 6169 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6170 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6171 sp<ThreadBase> thread = mThread.promote(); 6172 if (thread != 0) { 6173 audio_stream_t *stream = thread->stream(); 6174 if (stream != NULL) { 6175 stream->remove_audio_effect(stream, mEffectInterface); 6176 } 6177 } 6178 } 6179 // release effect engine 6180 EffectRelease(mEffectInterface); 6181 } 6182} 6183 6184status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6185{ 6186 status_t status; 6187 6188 Mutex::Autolock _l(mLock); 6189 // First handle in mHandles has highest priority and controls the effect module 6190 int priority = handle->priority(); 6191 size_t size = mHandles.size(); 6192 sp<EffectHandle> h; 6193 size_t i; 6194 for (i = 0; i < size; i++) { 6195 h = mHandles[i].promote(); 6196 if (h == 0) continue; 6197 if (h->priority() <= priority) break; 6198 } 6199 // if inserted in first place, move effect control from previous owner to this handle 6200 if (i == 0) { 6201 bool enabled = false; 6202 if (h != 0) { 6203 enabled = h->enabled(); 6204 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6205 } 6206 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6207 status = NO_ERROR; 6208 } else { 6209 status = ALREADY_EXISTS; 6210 } 6211 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6212 mHandles.insertAt(handle, i); 6213 return status; 6214} 6215 6216size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6217{ 6218 Mutex::Autolock _l(mLock); 6219 size_t size = mHandles.size(); 6220 size_t i; 6221 for (i = 0; i < size; i++) { 6222 if (mHandles[i] == handle) break; 6223 } 6224 if (i == size) { 6225 return size; 6226 } 6227 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6228 6229 bool enabled = false; 6230 EffectHandle *hdl = handle.unsafe_get(); 6231 if (hdl) { 6232 ALOGV("removeHandle() unsafe_get OK"); 6233 enabled = hdl->enabled(); 6234 } 6235 mHandles.removeAt(i); 6236 size = mHandles.size(); 6237 // if removed from first place, move effect control from this handle to next in line 6238 if (i == 0 && size != 0) { 6239 sp<EffectHandle> h = mHandles[0].promote(); 6240 if (h != 0) { 6241 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6242 } 6243 } 6244 6245 // Prevent calls to process() and other functions on effect interface from now on. 6246 // The effect engine will be released by the destructor when the last strong reference on 6247 // this object is released which can happen after next process is called. 6248 if (size == 0 && !mPinned) { 6249 mState = DESTROYED; 6250 } 6251 6252 return size; 6253} 6254 6255sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6256{ 6257 Mutex::Autolock _l(mLock); 6258 sp<EffectHandle> handle; 6259 if (mHandles.size() != 0) { 6260 handle = mHandles[0].promote(); 6261 } 6262 return handle; 6263} 6264 6265void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6266{ 6267 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6268 // keep a strong reference on this EffectModule to avoid calling the 6269 // destructor before we exit 6270 sp<EffectModule> keep(this); 6271 { 6272 sp<ThreadBase> thread = mThread.promote(); 6273 if (thread != 0) { 6274 thread->disconnectEffect(keep, handle, unpiniflast); 6275 } 6276 } 6277} 6278 6279void AudioFlinger::EffectModule::updateState() { 6280 Mutex::Autolock _l(mLock); 6281 6282 switch (mState) { 6283 case RESTART: 6284 reset_l(); 6285 // FALL THROUGH 6286 6287 case STARTING: 6288 // clear auxiliary effect input buffer for next accumulation 6289 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6290 memset(mConfig.inputCfg.buffer.raw, 6291 0, 6292 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6293 } 6294 start_l(); 6295 mState = ACTIVE; 6296 break; 6297 case STOPPING: 6298 stop_l(); 6299 mDisableWaitCnt = mMaxDisableWaitCnt; 6300 mState = STOPPED; 6301 break; 6302 case STOPPED: 6303 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6304 // turn off sequence. 6305 if (--mDisableWaitCnt == 0) { 6306 reset_l(); 6307 mState = IDLE; 6308 } 6309 break; 6310 default: //IDLE , ACTIVE, DESTROYED 6311 break; 6312 } 6313} 6314 6315void AudioFlinger::EffectModule::process() 6316{ 6317 Mutex::Autolock _l(mLock); 6318 6319 if (mState == DESTROYED || mEffectInterface == NULL || 6320 mConfig.inputCfg.buffer.raw == NULL || 6321 mConfig.outputCfg.buffer.raw == NULL) { 6322 return; 6323 } 6324 6325 if (isProcessEnabled()) { 6326 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6327 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6328 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6329 mConfig.inputCfg.buffer.s32, 6330 mConfig.inputCfg.buffer.frameCount/2); 6331 } 6332 6333 // do the actual processing in the effect engine 6334 int ret = (*mEffectInterface)->process(mEffectInterface, 6335 &mConfig.inputCfg.buffer, 6336 &mConfig.outputCfg.buffer); 6337 6338 // force transition to IDLE state when engine is ready 6339 if (mState == STOPPED && ret == -ENODATA) { 6340 mDisableWaitCnt = 1; 6341 } 6342 6343 // clear auxiliary effect input buffer for next accumulation 6344 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6345 memset(mConfig.inputCfg.buffer.raw, 0, 6346 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6347 } 6348 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6349 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6350 // If an insert effect is idle and input buffer is different from output buffer, 6351 // accumulate input onto output 6352 sp<EffectChain> chain = mChain.promote(); 6353 if (chain != 0 && chain->activeTrackCnt() != 0) { 6354 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6355 int16_t *in = mConfig.inputCfg.buffer.s16; 6356 int16_t *out = mConfig.outputCfg.buffer.s16; 6357 for (size_t i = 0; i < frameCnt; i++) { 6358 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6359 } 6360 } 6361 } 6362} 6363 6364void AudioFlinger::EffectModule::reset_l() 6365{ 6366 if (mEffectInterface == NULL) { 6367 return; 6368 } 6369 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6370} 6371 6372status_t AudioFlinger::EffectModule::configure() 6373{ 6374 uint32_t channels; 6375 if (mEffectInterface == NULL) { 6376 return NO_INIT; 6377 } 6378 6379 sp<ThreadBase> thread = mThread.promote(); 6380 if (thread == 0) { 6381 return DEAD_OBJECT; 6382 } 6383 6384 // TODO: handle configuration of effects replacing track process 6385 if (thread->channelCount() == 1) { 6386 channels = AUDIO_CHANNEL_OUT_MONO; 6387 } else { 6388 channels = AUDIO_CHANNEL_OUT_STEREO; 6389 } 6390 6391 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6392 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6393 } else { 6394 mConfig.inputCfg.channels = channels; 6395 } 6396 mConfig.outputCfg.channels = channels; 6397 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6398 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6399 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6400 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6401 mConfig.inputCfg.bufferProvider.cookie = NULL; 6402 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6403 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6404 mConfig.outputCfg.bufferProvider.cookie = NULL; 6405 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6406 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6407 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6408 // Insert effect: 6409 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6410 // always overwrites output buffer: input buffer == output buffer 6411 // - in other sessions: 6412 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6413 // other effect: overwrites output buffer: input buffer == output buffer 6414 // Auxiliary effect: 6415 // accumulates in output buffer: input buffer != output buffer 6416 // Therefore: accumulate <=> input buffer != output buffer 6417 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6418 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6419 } else { 6420 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6421 } 6422 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6423 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6424 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6425 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6426 6427 ALOGV("configure() %p thread %p buffer %p framecount %d", 6428 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6429 6430 status_t cmdStatus; 6431 uint32_t size = sizeof(int); 6432 status_t status = (*mEffectInterface)->command(mEffectInterface, 6433 EFFECT_CMD_SET_CONFIG, 6434 sizeof(effect_config_t), 6435 &mConfig, 6436 &size, 6437 &cmdStatus); 6438 if (status == 0) { 6439 status = cmdStatus; 6440 } 6441 6442 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6443 (1000 * mConfig.outputCfg.buffer.frameCount); 6444 6445 return status; 6446} 6447 6448status_t AudioFlinger::EffectModule::init() 6449{ 6450 Mutex::Autolock _l(mLock); 6451 if (mEffectInterface == NULL) { 6452 return NO_INIT; 6453 } 6454 status_t cmdStatus; 6455 uint32_t size = sizeof(status_t); 6456 status_t status = (*mEffectInterface)->command(mEffectInterface, 6457 EFFECT_CMD_INIT, 6458 0, 6459 NULL, 6460 &size, 6461 &cmdStatus); 6462 if (status == 0) { 6463 status = cmdStatus; 6464 } 6465 return status; 6466} 6467 6468status_t AudioFlinger::EffectModule::start() 6469{ 6470 Mutex::Autolock _l(mLock); 6471 return start_l(); 6472} 6473 6474status_t AudioFlinger::EffectModule::start_l() 6475{ 6476 if (mEffectInterface == NULL) { 6477 return NO_INIT; 6478 } 6479 status_t cmdStatus; 6480 uint32_t size = sizeof(status_t); 6481 status_t status = (*mEffectInterface)->command(mEffectInterface, 6482 EFFECT_CMD_ENABLE, 6483 0, 6484 NULL, 6485 &size, 6486 &cmdStatus); 6487 if (status == 0) { 6488 status = cmdStatus; 6489 } 6490 if (status == 0 && 6491 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6492 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6493 sp<ThreadBase> thread = mThread.promote(); 6494 if (thread != 0) { 6495 audio_stream_t *stream = thread->stream(); 6496 if (stream != NULL) { 6497 stream->add_audio_effect(stream, mEffectInterface); 6498 } 6499 } 6500 } 6501 return status; 6502} 6503 6504status_t AudioFlinger::EffectModule::stop() 6505{ 6506 Mutex::Autolock _l(mLock); 6507 return stop_l(); 6508} 6509 6510status_t AudioFlinger::EffectModule::stop_l() 6511{ 6512 if (mEffectInterface == NULL) { 6513 return NO_INIT; 6514 } 6515 status_t cmdStatus; 6516 uint32_t size = sizeof(status_t); 6517 status_t status = (*mEffectInterface)->command(mEffectInterface, 6518 EFFECT_CMD_DISABLE, 6519 0, 6520 NULL, 6521 &size, 6522 &cmdStatus); 6523 if (status == 0) { 6524 status = cmdStatus; 6525 } 6526 if (status == 0 && 6527 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6528 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6529 sp<ThreadBase> thread = mThread.promote(); 6530 if (thread != 0) { 6531 audio_stream_t *stream = thread->stream(); 6532 if (stream != NULL) { 6533 stream->remove_audio_effect(stream, mEffectInterface); 6534 } 6535 } 6536 } 6537 return status; 6538} 6539 6540status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6541 uint32_t cmdSize, 6542 void *pCmdData, 6543 uint32_t *replySize, 6544 void *pReplyData) 6545{ 6546 Mutex::Autolock _l(mLock); 6547// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6548 6549 if (mState == DESTROYED || mEffectInterface == NULL) { 6550 return NO_INIT; 6551 } 6552 status_t status = (*mEffectInterface)->command(mEffectInterface, 6553 cmdCode, 6554 cmdSize, 6555 pCmdData, 6556 replySize, 6557 pReplyData); 6558 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6559 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6560 for (size_t i = 1; i < mHandles.size(); i++) { 6561 sp<EffectHandle> h = mHandles[i].promote(); 6562 if (h != 0) { 6563 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6564 } 6565 } 6566 } 6567 return status; 6568} 6569 6570status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6571{ 6572 6573 Mutex::Autolock _l(mLock); 6574 ALOGV("setEnabled %p enabled %d", this, enabled); 6575 6576 if (enabled != isEnabled()) { 6577 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6578 if (enabled && status != NO_ERROR) { 6579 return status; 6580 } 6581 6582 switch (mState) { 6583 // going from disabled to enabled 6584 case IDLE: 6585 mState = STARTING; 6586 break; 6587 case STOPPED: 6588 mState = RESTART; 6589 break; 6590 case STOPPING: 6591 mState = ACTIVE; 6592 break; 6593 6594 // going from enabled to disabled 6595 case RESTART: 6596 mState = STOPPED; 6597 break; 6598 case STARTING: 6599 mState = IDLE; 6600 break; 6601 case ACTIVE: 6602 mState = STOPPING; 6603 break; 6604 case DESTROYED: 6605 return NO_ERROR; // simply ignore as we are being destroyed 6606 } 6607 for (size_t i = 1; i < mHandles.size(); i++) { 6608 sp<EffectHandle> h = mHandles[i].promote(); 6609 if (h != 0) { 6610 h->setEnabled(enabled); 6611 } 6612 } 6613 } 6614 return NO_ERROR; 6615} 6616 6617bool AudioFlinger::EffectModule::isEnabled() 6618{ 6619 switch (mState) { 6620 case RESTART: 6621 case STARTING: 6622 case ACTIVE: 6623 return true; 6624 case IDLE: 6625 case STOPPING: 6626 case STOPPED: 6627 case DESTROYED: 6628 default: 6629 return false; 6630 } 6631} 6632 6633bool AudioFlinger::EffectModule::isProcessEnabled() 6634{ 6635 switch (mState) { 6636 case RESTART: 6637 case ACTIVE: 6638 case STOPPING: 6639 case STOPPED: 6640 return true; 6641 case IDLE: 6642 case STARTING: 6643 case DESTROYED: 6644 default: 6645 return false; 6646 } 6647} 6648 6649status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6650{ 6651 Mutex::Autolock _l(mLock); 6652 status_t status = NO_ERROR; 6653 6654 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6655 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6656 if (isProcessEnabled() && 6657 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6658 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6659 status_t cmdStatus; 6660 uint32_t volume[2]; 6661 uint32_t *pVolume = NULL; 6662 uint32_t size = sizeof(volume); 6663 volume[0] = *left; 6664 volume[1] = *right; 6665 if (controller) { 6666 pVolume = volume; 6667 } 6668 status = (*mEffectInterface)->command(mEffectInterface, 6669 EFFECT_CMD_SET_VOLUME, 6670 size, 6671 volume, 6672 &size, 6673 pVolume); 6674 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6675 *left = volume[0]; 6676 *right = volume[1]; 6677 } 6678 } 6679 return status; 6680} 6681 6682status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6683{ 6684 Mutex::Autolock _l(mLock); 6685 status_t status = NO_ERROR; 6686 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6687 // audio pre processing modules on RecordThread can receive both output and 6688 // input device indication in the same call 6689 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6690 if (dev) { 6691 status_t cmdStatus; 6692 uint32_t size = sizeof(status_t); 6693 6694 status = (*mEffectInterface)->command(mEffectInterface, 6695 EFFECT_CMD_SET_DEVICE, 6696 sizeof(uint32_t), 6697 &dev, 6698 &size, 6699 &cmdStatus); 6700 if (status == NO_ERROR) { 6701 status = cmdStatus; 6702 } 6703 } 6704 dev = device & AUDIO_DEVICE_IN_ALL; 6705 if (dev) { 6706 status_t cmdStatus; 6707 uint32_t size = sizeof(status_t); 6708 6709 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6710 EFFECT_CMD_SET_INPUT_DEVICE, 6711 sizeof(uint32_t), 6712 &dev, 6713 &size, 6714 &cmdStatus); 6715 if (status2 == NO_ERROR) { 6716 status2 = cmdStatus; 6717 } 6718 if (status == NO_ERROR) { 6719 status = status2; 6720 } 6721 } 6722 } 6723 return status; 6724} 6725 6726status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6727{ 6728 Mutex::Autolock _l(mLock); 6729 status_t status = NO_ERROR; 6730 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6731 status_t cmdStatus; 6732 uint32_t size = sizeof(status_t); 6733 status = (*mEffectInterface)->command(mEffectInterface, 6734 EFFECT_CMD_SET_AUDIO_MODE, 6735 sizeof(audio_mode_t), 6736 &mode, 6737 &size, 6738 &cmdStatus); 6739 if (status == NO_ERROR) { 6740 status = cmdStatus; 6741 } 6742 } 6743 return status; 6744} 6745 6746void AudioFlinger::EffectModule::setSuspended(bool suspended) 6747{ 6748 Mutex::Autolock _l(mLock); 6749 mSuspended = suspended; 6750} 6751 6752bool AudioFlinger::EffectModule::suspended() const 6753{ 6754 Mutex::Autolock _l(mLock); 6755 return mSuspended; 6756} 6757 6758status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6759{ 6760 const size_t SIZE = 256; 6761 char buffer[SIZE]; 6762 String8 result; 6763 6764 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6765 result.append(buffer); 6766 6767 bool locked = tryLock(mLock); 6768 // failed to lock - AudioFlinger is probably deadlocked 6769 if (!locked) { 6770 result.append("\t\tCould not lock Fx mutex:\n"); 6771 } 6772 6773 result.append("\t\tSession Status State Engine:\n"); 6774 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6775 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6776 result.append(buffer); 6777 6778 result.append("\t\tDescriptor:\n"); 6779 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6780 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6781 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6782 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6783 result.append(buffer); 6784 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6785 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6786 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6787 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6788 result.append(buffer); 6789 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6790 mDescriptor.apiVersion, 6791 mDescriptor.flags); 6792 result.append(buffer); 6793 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6794 mDescriptor.name); 6795 result.append(buffer); 6796 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6797 mDescriptor.implementor); 6798 result.append(buffer); 6799 6800 result.append("\t\t- Input configuration:\n"); 6801 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6802 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6803 (uint32_t)mConfig.inputCfg.buffer.raw, 6804 mConfig.inputCfg.buffer.frameCount, 6805 mConfig.inputCfg.samplingRate, 6806 mConfig.inputCfg.channels, 6807 mConfig.inputCfg.format); 6808 result.append(buffer); 6809 6810 result.append("\t\t- Output configuration:\n"); 6811 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6812 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6813 (uint32_t)mConfig.outputCfg.buffer.raw, 6814 mConfig.outputCfg.buffer.frameCount, 6815 mConfig.outputCfg.samplingRate, 6816 mConfig.outputCfg.channels, 6817 mConfig.outputCfg.format); 6818 result.append(buffer); 6819 6820 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6821 result.append(buffer); 6822 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6823 for (size_t i = 0; i < mHandles.size(); ++i) { 6824 sp<EffectHandle> handle = mHandles[i].promote(); 6825 if (handle != 0) { 6826 handle->dump(buffer, SIZE); 6827 result.append(buffer); 6828 } 6829 } 6830 6831 result.append("\n"); 6832 6833 write(fd, result.string(), result.length()); 6834 6835 if (locked) { 6836 mLock.unlock(); 6837 } 6838 6839 return NO_ERROR; 6840} 6841 6842// ---------------------------------------------------------------------------- 6843// EffectHandle implementation 6844// ---------------------------------------------------------------------------- 6845 6846#undef LOG_TAG 6847#define LOG_TAG "AudioFlinger::EffectHandle" 6848 6849AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6850 const sp<AudioFlinger::Client>& client, 6851 const sp<IEffectClient>& effectClient, 6852 int32_t priority) 6853 : BnEffect(), 6854 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6855 mPriority(priority), mHasControl(false), mEnabled(false) 6856{ 6857 ALOGV("constructor %p", this); 6858 6859 if (client == 0) { 6860 return; 6861 } 6862 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6863 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6864 if (mCblkMemory != 0) { 6865 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6866 6867 if (mCblk) { 6868 new(mCblk) effect_param_cblk_t(); 6869 mBuffer = (uint8_t *)mCblk + bufOffset; 6870 } 6871 } else { 6872 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6873 return; 6874 } 6875} 6876 6877AudioFlinger::EffectHandle::~EffectHandle() 6878{ 6879 ALOGV("Destructor %p", this); 6880 disconnect(false); 6881 ALOGV("Destructor DONE %p", this); 6882} 6883 6884status_t AudioFlinger::EffectHandle::enable() 6885{ 6886 ALOGV("enable %p", this); 6887 if (!mHasControl) return INVALID_OPERATION; 6888 if (mEffect == 0) return DEAD_OBJECT; 6889 6890 if (mEnabled) { 6891 return NO_ERROR; 6892 } 6893 6894 mEnabled = true; 6895 6896 sp<ThreadBase> thread = mEffect->thread().promote(); 6897 if (thread != 0) { 6898 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6899 } 6900 6901 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6902 if (mEffect->suspended()) { 6903 return NO_ERROR; 6904 } 6905 6906 status_t status = mEffect->setEnabled(true); 6907 if (status != NO_ERROR) { 6908 if (thread != 0) { 6909 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6910 } 6911 mEnabled = false; 6912 } 6913 return status; 6914} 6915 6916status_t AudioFlinger::EffectHandle::disable() 6917{ 6918 ALOGV("disable %p", this); 6919 if (!mHasControl) return INVALID_OPERATION; 6920 if (mEffect == 0) return DEAD_OBJECT; 6921 6922 if (!mEnabled) { 6923 return NO_ERROR; 6924 } 6925 mEnabled = false; 6926 6927 if (mEffect->suspended()) { 6928 return NO_ERROR; 6929 } 6930 6931 status_t status = mEffect->setEnabled(false); 6932 6933 sp<ThreadBase> thread = mEffect->thread().promote(); 6934 if (thread != 0) { 6935 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6936 } 6937 6938 return status; 6939} 6940 6941void AudioFlinger::EffectHandle::disconnect() 6942{ 6943 disconnect(true); 6944} 6945 6946void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6947{ 6948 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6949 if (mEffect == 0) { 6950 return; 6951 } 6952 mEffect->disconnect(this, unpiniflast); 6953 6954 if (mHasControl && mEnabled) { 6955 sp<ThreadBase> thread = mEffect->thread().promote(); 6956 if (thread != 0) { 6957 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6958 } 6959 } 6960 6961 // release sp on module => module destructor can be called now 6962 mEffect.clear(); 6963 if (mClient != 0) { 6964 if (mCblk) { 6965 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6966 } 6967 mCblkMemory.clear(); // and free the shared memory 6968 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6969 mClient.clear(); 6970 } 6971} 6972 6973status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6974 uint32_t cmdSize, 6975 void *pCmdData, 6976 uint32_t *replySize, 6977 void *pReplyData) 6978{ 6979// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6980// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6981 6982 // only get parameter command is permitted for applications not controlling the effect 6983 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6984 return INVALID_OPERATION; 6985 } 6986 if (mEffect == 0) return DEAD_OBJECT; 6987 if (mClient == 0) return INVALID_OPERATION; 6988 6989 // handle commands that are not forwarded transparently to effect engine 6990 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6991 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6992 // no risk to block the whole media server process or mixer threads is we are stuck here 6993 Mutex::Autolock _l(mCblk->lock); 6994 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6995 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6996 mCblk->serverIndex = 0; 6997 mCblk->clientIndex = 0; 6998 return BAD_VALUE; 6999 } 7000 status_t status = NO_ERROR; 7001 while (mCblk->serverIndex < mCblk->clientIndex) { 7002 int reply; 7003 uint32_t rsize = sizeof(int); 7004 int *p = (int *)(mBuffer + mCblk->serverIndex); 7005 int size = *p++; 7006 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7007 ALOGW("command(): invalid parameter block size"); 7008 break; 7009 } 7010 effect_param_t *param = (effect_param_t *)p; 7011 if (param->psize == 0 || param->vsize == 0) { 7012 ALOGW("command(): null parameter or value size"); 7013 mCblk->serverIndex += size; 7014 continue; 7015 } 7016 uint32_t psize = sizeof(effect_param_t) + 7017 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7018 param->vsize; 7019 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7020 psize, 7021 p, 7022 &rsize, 7023 &reply); 7024 // stop at first error encountered 7025 if (ret != NO_ERROR) { 7026 status = ret; 7027 *(int *)pReplyData = reply; 7028 break; 7029 } else if (reply != NO_ERROR) { 7030 *(int *)pReplyData = reply; 7031 break; 7032 } 7033 mCblk->serverIndex += size; 7034 } 7035 mCblk->serverIndex = 0; 7036 mCblk->clientIndex = 0; 7037 return status; 7038 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7039 *(int *)pReplyData = NO_ERROR; 7040 return enable(); 7041 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7042 *(int *)pReplyData = NO_ERROR; 7043 return disable(); 7044 } 7045 7046 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7047} 7048 7049sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7050 return mCblkMemory; 7051} 7052 7053void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7054{ 7055 ALOGV("setControl %p control %d", this, hasControl); 7056 7057 mHasControl = hasControl; 7058 mEnabled = enabled; 7059 7060 if (signal && mEffectClient != 0) { 7061 mEffectClient->controlStatusChanged(hasControl); 7062 } 7063} 7064 7065void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7066 uint32_t cmdSize, 7067 void *pCmdData, 7068 uint32_t replySize, 7069 void *pReplyData) 7070{ 7071 if (mEffectClient != 0) { 7072 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7073 } 7074} 7075 7076 7077 7078void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7079{ 7080 if (mEffectClient != 0) { 7081 mEffectClient->enableStatusChanged(enabled); 7082 } 7083} 7084 7085status_t AudioFlinger::EffectHandle::onTransact( 7086 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7087{ 7088 return BnEffect::onTransact(code, data, reply, flags); 7089} 7090 7091 7092void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7093{ 7094 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7095 7096 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7097 (mClient == NULL) ? getpid() : mClient->pid(), 7098 mPriority, 7099 mHasControl, 7100 !locked, 7101 mCblk ? mCblk->clientIndex : 0, 7102 mCblk ? mCblk->serverIndex : 0 7103 ); 7104 7105 if (locked) { 7106 mCblk->lock.unlock(); 7107 } 7108} 7109 7110#undef LOG_TAG 7111#define LOG_TAG "AudioFlinger::EffectChain" 7112 7113AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7114 int sessionId) 7115 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7116 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7117 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7118{ 7119 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7120 sp<ThreadBase> thread = mThread.promote(); 7121 if (thread == 0) { 7122 return; 7123 } 7124 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7125 thread->frameCount(); 7126} 7127 7128AudioFlinger::EffectChain::~EffectChain() 7129{ 7130 if (mOwnInBuffer) { 7131 delete mInBuffer; 7132 } 7133 7134} 7135 7136// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7137sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7138{ 7139 sp<EffectModule> effect; 7140 size_t size = mEffects.size(); 7141 7142 for (size_t i = 0; i < size; i++) { 7143 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7144 effect = mEffects[i]; 7145 break; 7146 } 7147 } 7148 return effect; 7149} 7150 7151// getEffectFromId_l() must be called with ThreadBase::mLock held 7152sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7153{ 7154 sp<EffectModule> effect; 7155 size_t size = mEffects.size(); 7156 7157 for (size_t i = 0; i < size; i++) { 7158 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7159 if (id == 0 || mEffects[i]->id() == id) { 7160 effect = mEffects[i]; 7161 break; 7162 } 7163 } 7164 return effect; 7165} 7166 7167// getEffectFromType_l() must be called with ThreadBase::mLock held 7168sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7169 const effect_uuid_t *type) 7170{ 7171 sp<EffectModule> effect; 7172 size_t size = mEffects.size(); 7173 7174 for (size_t i = 0; i < size; i++) { 7175 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7176 effect = mEffects[i]; 7177 break; 7178 } 7179 } 7180 return effect; 7181} 7182 7183// Must be called with EffectChain::mLock locked 7184void AudioFlinger::EffectChain::process_l() 7185{ 7186 sp<ThreadBase> thread = mThread.promote(); 7187 if (thread == 0) { 7188 ALOGW("process_l(): cannot promote mixer thread"); 7189 return; 7190 } 7191 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7192 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7193 // always process effects unless no more tracks are on the session and the effect tail 7194 // has been rendered 7195 bool doProcess = true; 7196 if (!isGlobalSession) { 7197 bool tracksOnSession = (trackCnt() != 0); 7198 7199 if (!tracksOnSession && mTailBufferCount == 0) { 7200 doProcess = false; 7201 } 7202 7203 if (activeTrackCnt() == 0) { 7204 // if no track is active and the effect tail has not been rendered, 7205 // the input buffer must be cleared here as the mixer process will not do it 7206 if (tracksOnSession || mTailBufferCount > 0) { 7207 size_t numSamples = thread->frameCount() * thread->channelCount(); 7208 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7209 if (mTailBufferCount > 0) { 7210 mTailBufferCount--; 7211 } 7212 } 7213 } 7214 } 7215 7216 size_t size = mEffects.size(); 7217 if (doProcess) { 7218 for (size_t i = 0; i < size; i++) { 7219 mEffects[i]->process(); 7220 } 7221 } 7222 for (size_t i = 0; i < size; i++) { 7223 mEffects[i]->updateState(); 7224 } 7225} 7226 7227// addEffect_l() must be called with PlaybackThread::mLock held 7228status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7229{ 7230 effect_descriptor_t desc = effect->desc(); 7231 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7232 7233 Mutex::Autolock _l(mLock); 7234 effect->setChain(this); 7235 sp<ThreadBase> thread = mThread.promote(); 7236 if (thread == 0) { 7237 return NO_INIT; 7238 } 7239 effect->setThread(thread); 7240 7241 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7242 // Auxiliary effects are inserted at the beginning of mEffects vector as 7243 // they are processed first and accumulated in chain input buffer 7244 mEffects.insertAt(effect, 0); 7245 7246 // the input buffer for auxiliary effect contains mono samples in 7247 // 32 bit format. This is to avoid saturation in AudoMixer 7248 // accumulation stage. Saturation is done in EffectModule::process() before 7249 // calling the process in effect engine 7250 size_t numSamples = thread->frameCount(); 7251 int32_t *buffer = new int32_t[numSamples]; 7252 memset(buffer, 0, numSamples * sizeof(int32_t)); 7253 effect->setInBuffer((int16_t *)buffer); 7254 // auxiliary effects output samples to chain input buffer for further processing 7255 // by insert effects 7256 effect->setOutBuffer(mInBuffer); 7257 } else { 7258 // Insert effects are inserted at the end of mEffects vector as they are processed 7259 // after track and auxiliary effects. 7260 // Insert effect order as a function of indicated preference: 7261 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7262 // another effect is present 7263 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7264 // last effect claiming first position 7265 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7266 // first effect claiming last position 7267 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7268 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7269 // already present 7270 7271 int size = (int)mEffects.size(); 7272 int idx_insert = size; 7273 int idx_insert_first = -1; 7274 int idx_insert_last = -1; 7275 7276 for (int i = 0; i < size; i++) { 7277 effect_descriptor_t d = mEffects[i]->desc(); 7278 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7279 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7280 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7281 // check invalid effect chaining combinations 7282 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7283 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7284 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7285 return INVALID_OPERATION; 7286 } 7287 // remember position of first insert effect and by default 7288 // select this as insert position for new effect 7289 if (idx_insert == size) { 7290 idx_insert = i; 7291 } 7292 // remember position of last insert effect claiming 7293 // first position 7294 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7295 idx_insert_first = i; 7296 } 7297 // remember position of first insert effect claiming 7298 // last position 7299 if (iPref == EFFECT_FLAG_INSERT_LAST && 7300 idx_insert_last == -1) { 7301 idx_insert_last = i; 7302 } 7303 } 7304 } 7305 7306 // modify idx_insert from first position if needed 7307 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7308 if (idx_insert_last != -1) { 7309 idx_insert = idx_insert_last; 7310 } else { 7311 idx_insert = size; 7312 } 7313 } else { 7314 if (idx_insert_first != -1) { 7315 idx_insert = idx_insert_first + 1; 7316 } 7317 } 7318 7319 // always read samples from chain input buffer 7320 effect->setInBuffer(mInBuffer); 7321 7322 // if last effect in the chain, output samples to chain 7323 // output buffer, otherwise to chain input buffer 7324 if (idx_insert == size) { 7325 if (idx_insert != 0) { 7326 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7327 mEffects[idx_insert-1]->configure(); 7328 } 7329 effect->setOutBuffer(mOutBuffer); 7330 } else { 7331 effect->setOutBuffer(mInBuffer); 7332 } 7333 mEffects.insertAt(effect, idx_insert); 7334 7335 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7336 } 7337 effect->configure(); 7338 return NO_ERROR; 7339} 7340 7341// removeEffect_l() must be called with PlaybackThread::mLock held 7342size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7343{ 7344 Mutex::Autolock _l(mLock); 7345 int size = (int)mEffects.size(); 7346 int i; 7347 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7348 7349 for (i = 0; i < size; i++) { 7350 if (effect == mEffects[i]) { 7351 // calling stop here will remove pre-processing effect from the audio HAL. 7352 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7353 // the middle of a read from audio HAL 7354 if (mEffects[i]->state() == EffectModule::ACTIVE || 7355 mEffects[i]->state() == EffectModule::STOPPING) { 7356 mEffects[i]->stop(); 7357 } 7358 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7359 delete[] effect->inBuffer(); 7360 } else { 7361 if (i == size - 1 && i != 0) { 7362 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7363 mEffects[i - 1]->configure(); 7364 } 7365 } 7366 mEffects.removeAt(i); 7367 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7368 break; 7369 } 7370 } 7371 7372 return mEffects.size(); 7373} 7374 7375// setDevice_l() must be called with PlaybackThread::mLock held 7376void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7377{ 7378 size_t size = mEffects.size(); 7379 for (size_t i = 0; i < size; i++) { 7380 mEffects[i]->setDevice(device); 7381 } 7382} 7383 7384// setMode_l() must be called with PlaybackThread::mLock held 7385void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7386{ 7387 size_t size = mEffects.size(); 7388 for (size_t i = 0; i < size; i++) { 7389 mEffects[i]->setMode(mode); 7390 } 7391} 7392 7393// setVolume_l() must be called with PlaybackThread::mLock held 7394bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7395{ 7396 uint32_t newLeft = *left; 7397 uint32_t newRight = *right; 7398 bool hasControl = false; 7399 int ctrlIdx = -1; 7400 size_t size = mEffects.size(); 7401 7402 // first update volume controller 7403 for (size_t i = size; i > 0; i--) { 7404 if (mEffects[i - 1]->isProcessEnabled() && 7405 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7406 ctrlIdx = i - 1; 7407 hasControl = true; 7408 break; 7409 } 7410 } 7411 7412 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7413 if (hasControl) { 7414 *left = mNewLeftVolume; 7415 *right = mNewRightVolume; 7416 } 7417 return hasControl; 7418 } 7419 7420 mVolumeCtrlIdx = ctrlIdx; 7421 mLeftVolume = newLeft; 7422 mRightVolume = newRight; 7423 7424 // second get volume update from volume controller 7425 if (ctrlIdx >= 0) { 7426 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7427 mNewLeftVolume = newLeft; 7428 mNewRightVolume = newRight; 7429 } 7430 // then indicate volume to all other effects in chain. 7431 // Pass altered volume to effects before volume controller 7432 // and requested volume to effects after controller 7433 uint32_t lVol = newLeft; 7434 uint32_t rVol = newRight; 7435 7436 for (size_t i = 0; i < size; i++) { 7437 if ((int)i == ctrlIdx) continue; 7438 // this also works for ctrlIdx == -1 when there is no volume controller 7439 if ((int)i > ctrlIdx) { 7440 lVol = *left; 7441 rVol = *right; 7442 } 7443 mEffects[i]->setVolume(&lVol, &rVol, false); 7444 } 7445 *left = newLeft; 7446 *right = newRight; 7447 7448 return hasControl; 7449} 7450 7451status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7452{ 7453 const size_t SIZE = 256; 7454 char buffer[SIZE]; 7455 String8 result; 7456 7457 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7458 result.append(buffer); 7459 7460 bool locked = tryLock(mLock); 7461 // failed to lock - AudioFlinger is probably deadlocked 7462 if (!locked) { 7463 result.append("\tCould not lock mutex:\n"); 7464 } 7465 7466 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7467 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7468 mEffects.size(), 7469 (uint32_t)mInBuffer, 7470 (uint32_t)mOutBuffer, 7471 mActiveTrackCnt); 7472 result.append(buffer); 7473 write(fd, result.string(), result.size()); 7474 7475 for (size_t i = 0; i < mEffects.size(); ++i) { 7476 sp<EffectModule> effect = mEffects[i]; 7477 if (effect != 0) { 7478 effect->dump(fd, args); 7479 } 7480 } 7481 7482 if (locked) { 7483 mLock.unlock(); 7484 } 7485 7486 return NO_ERROR; 7487} 7488 7489// must be called with ThreadBase::mLock held 7490void AudioFlinger::EffectChain::setEffectSuspended_l( 7491 const effect_uuid_t *type, bool suspend) 7492{ 7493 sp<SuspendedEffectDesc> desc; 7494 // use effect type UUID timelow as key as there is no real risk of identical 7495 // timeLow fields among effect type UUIDs. 7496 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7497 if (suspend) { 7498 if (index >= 0) { 7499 desc = mSuspendedEffects.valueAt(index); 7500 } else { 7501 desc = new SuspendedEffectDesc(); 7502 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7503 mSuspendedEffects.add(type->timeLow, desc); 7504 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7505 } 7506 if (desc->mRefCount++ == 0) { 7507 sp<EffectModule> effect = getEffectIfEnabled(type); 7508 if (effect != 0) { 7509 desc->mEffect = effect; 7510 effect->setSuspended(true); 7511 effect->setEnabled(false); 7512 } 7513 } 7514 } else { 7515 if (index < 0) { 7516 return; 7517 } 7518 desc = mSuspendedEffects.valueAt(index); 7519 if (desc->mRefCount <= 0) { 7520 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7521 desc->mRefCount = 1; 7522 } 7523 if (--desc->mRefCount == 0) { 7524 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7525 if (desc->mEffect != 0) { 7526 sp<EffectModule> effect = desc->mEffect.promote(); 7527 if (effect != 0) { 7528 effect->setSuspended(false); 7529 sp<EffectHandle> handle = effect->controlHandle(); 7530 if (handle != 0) { 7531 effect->setEnabled(handle->enabled()); 7532 } 7533 } 7534 desc->mEffect.clear(); 7535 } 7536 mSuspendedEffects.removeItemsAt(index); 7537 } 7538 } 7539} 7540 7541// must be called with ThreadBase::mLock held 7542void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7543{ 7544 sp<SuspendedEffectDesc> desc; 7545 7546 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7547 if (suspend) { 7548 if (index >= 0) { 7549 desc = mSuspendedEffects.valueAt(index); 7550 } else { 7551 desc = new SuspendedEffectDesc(); 7552 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7553 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7554 } 7555 if (desc->mRefCount++ == 0) { 7556 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7557 for (size_t i = 0; i < effects.size(); i++) { 7558 setEffectSuspended_l(&effects[i]->desc().type, true); 7559 } 7560 } 7561 } else { 7562 if (index < 0) { 7563 return; 7564 } 7565 desc = mSuspendedEffects.valueAt(index); 7566 if (desc->mRefCount <= 0) { 7567 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7568 desc->mRefCount = 1; 7569 } 7570 if (--desc->mRefCount == 0) { 7571 Vector<const effect_uuid_t *> types; 7572 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7573 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7574 continue; 7575 } 7576 types.add(&mSuspendedEffects.valueAt(i)->mType); 7577 } 7578 for (size_t i = 0; i < types.size(); i++) { 7579 setEffectSuspended_l(types[i], false); 7580 } 7581 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7582 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7583 } 7584 } 7585} 7586 7587 7588// The volume effect is used for automated tests only 7589#ifndef OPENSL_ES_H_ 7590static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7591 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7592const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7593#endif //OPENSL_ES_H_ 7594 7595bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7596{ 7597 // auxiliary effects and visualizer are never suspended on output mix 7598 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7599 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7600 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7601 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7602 return false; 7603 } 7604 return true; 7605} 7606 7607Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7608{ 7609 Vector< sp<EffectModule> > effects; 7610 for (size_t i = 0; i < mEffects.size(); i++) { 7611 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7612 continue; 7613 } 7614 effects.add(mEffects[i]); 7615 } 7616 return effects; 7617} 7618 7619sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7620 const effect_uuid_t *type) 7621{ 7622 sp<EffectModule> effect; 7623 effect = getEffectFromType_l(type); 7624 if (effect != 0 && !effect->isEnabled()) { 7625 effect.clear(); 7626 } 7627 return effect; 7628} 7629 7630void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7631 bool enabled) 7632{ 7633 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7634 if (enabled) { 7635 if (index < 0) { 7636 // if the effect is not suspend check if all effects are suspended 7637 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7638 if (index < 0) { 7639 return; 7640 } 7641 if (!isEffectEligibleForSuspend(effect->desc())) { 7642 return; 7643 } 7644 setEffectSuspended_l(&effect->desc().type, enabled); 7645 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7646 if (index < 0) { 7647 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7648 return; 7649 } 7650 } 7651 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7652 effect->desc().type.timeLow); 7653 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7654 // if effect is requested to suspended but was not yet enabled, supend it now. 7655 if (desc->mEffect == 0) { 7656 desc->mEffect = effect; 7657 effect->setEnabled(false); 7658 effect->setSuspended(true); 7659 } 7660 } else { 7661 if (index < 0) { 7662 return; 7663 } 7664 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7665 effect->desc().type.timeLow); 7666 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7667 desc->mEffect.clear(); 7668 effect->setSuspended(false); 7669 } 7670} 7671 7672#undef LOG_TAG 7673#define LOG_TAG "AudioFlinger" 7674 7675// ---------------------------------------------------------------------------- 7676 7677status_t AudioFlinger::onTransact( 7678 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7679{ 7680 return BnAudioFlinger::onTransact(code, data, reply, flags); 7681} 7682 7683}; // namespace android 7684