AudioFlinger.cpp revision 462fd2fa9eef642b0574aa7409de0bde3fec8d43
1023d2bbbbedc6ed991b11381a987673133be2c81Michael Gottesman/* 27ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** 37ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** Copyright 2007, The Android Open Source Project 47ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** 57ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** Licensed under the Apache License, Version 2.0 (the "License"); 67ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** you may not use this file except in compliance with the License. 77ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** You may obtain a copy of the License at 87ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** 97ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** http://www.apache.org/licenses/LICENSE-2.0 107ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** 117ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** Unless required by applicable law or agreed to in writing, software 127ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** distributed under the License is distributed on an "AS IS" BASIS, 137ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 147ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** See the License for the specific language governing permissions and 157ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman** limitations under the License. 167ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman*/ 177ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 187ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 197ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#define LOG_TAG "AudioFlinger" 207ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman//#define LOG_NDEBUG 0 217ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 227ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include "Configuration.h" 237ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <dirent.h> 247ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <math.h> 257ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <signal.h> 267ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <sys/time.h> 277ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <sys/resource.h> 287ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 297ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <binder/IPCThreadState.h> 307ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <binder/IServiceManager.h> 317ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <utils/Log.h> 327ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <utils/Trace.h> 337ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <binder/Parcel.h> 347ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <utils/String16.h> 357ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <utils/threads.h> 367ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <utils/Atomic.h> 377ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 387ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <cutils/bitops.h> 397ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <cutils/properties.h> 407ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 417ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <system/audio.h> 427ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <hardware/audio.h> 437ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 447ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include "AudioMixer.h" 457ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include "AudioFlinger.h" 46dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include "ServiceUtilities.h" 47dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines 48dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <media/EffectsFactoryApi.h> 49dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <audio_effects/effect_visualizer.h> 50dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <audio_effects/effect_ns.h> 51dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <audio_effects/effect_aec.h> 52dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines 53dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <audio_utils/primitives.h> 54dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines 55dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <powermanager/PowerManager.h> 567ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 577ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <common_time/cc_helper.h> 587ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 597ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#include <media/IMediaLogService.h> 60c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman 61dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <media/nbaio/Pipe.h> 62dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <media/nbaio/PipeReader.h> 63dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <media/AudioParameter.h> 64dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines#include <private/android_filesystem_config.h> 65dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines 66dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines// ---------------------------------------------------------------------------- 67dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines 68dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines// Note: the following macro is used for extremely verbose logging message. In 69dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 707ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman// 0; but one side effect of this is to turn all LOGV's as well. Some messages 717ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman// are so verbose that we want to suppress them even when we have ALOG_ASSERT 727ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman// turned on. Do not uncomment the #def below unless you really know what you 73dce4a407a24b04eebc6a376f8e62b41aaa7b071fStephen Hines// are doing and want to see all of the extremely verbose messages. 747ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman//#define VERY_VERY_VERBOSE_LOGGING 757ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#ifdef VERY_VERY_VERBOSE_LOGGING 767ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#define ALOGVV ALOGV 777ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#else 787ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#define ALOGVV(a...) do { } while(0) 797ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#endif 807ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 817ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmannamespace android { 827ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 837ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanstatic const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 847ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanstatic const char kHardwareLockedString[] = "Hardware lock is taken\n"; 857ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 867ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 877ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmannsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 887ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 89462e998f076b625507d134c8c341f8cf960d1eb0Michael Gottesmanuint32_t AudioFlinger::mScreenState; 90462e998f076b625507d134c8c341f8cf960d1eb0Michael Gottesman 91462e998f076b625507d134c8c341f8cf960d1eb0Michael Gottesman#ifdef TEE_SINK 927ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanbool AudioFlinger::mTeeSinkInputEnabled = false; 937ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanbool AudioFlinger::mTeeSinkOutputEnabled = false; 947ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanbool AudioFlinger::mTeeSinkTrackEnabled = false; 957ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 967ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmansize_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 977ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmansize_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 987ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmansize_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 99462e998f076b625507d134c8c341f8cf960d1eb0Michael Gottesman#endif 100462e998f076b625507d134c8c341f8cf960d1eb0Michael Gottesman 1017ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 1027ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman// we define a minimum time during which a global effect is considered enabled. 1037ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanstatic const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 1047ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 105c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman// ---------------------------------------------------------------------------- 1067ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 1077ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanstatic int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 1087ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman{ 1097ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman const hw_module_t *mod; 1107ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman int rc; 1117ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 1127ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 1137ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 1147ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 1157ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman if (rc) { 1167ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman goto out; 1177ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 1187ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman rc = audio_hw_device_open(mod, dev); 1197ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 1207ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 1217ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman if (rc) { 1227ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman goto out; 1237ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 1247ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 1257ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 1267ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman rc = BAD_VALUE; 1277ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman goto out; 1287ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 1297ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman return 0; 1307ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 131c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesmanout: 1327ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman *dev = NULL; 1337ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman return rc; 134c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman} 1357ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 1367ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman// ---------------------------------------------------------------------------- 1377ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 138c3e6edba384e023da4e974faca4e28b2276d575fMichael GottesmanAudioFlinger::AudioFlinger() 1397ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman : BnAudioFlinger(), 1407ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mPrimaryHardwareDev(NULL), 1417ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mHardwareStatus(AUDIO_HW_IDLE), 1427ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mMasterVolume(1.0f), 1437ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mMasterMute(false), 1447ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mNextUniqueId(1), 1457ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mMode(AUDIO_MODE_INVALID), 1467ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mBtNrecIsOff(false), 147c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman mIsLowRamDevice(true), 1487ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mIsDeviceTypeKnown(false), 1497ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mGlobalEffectEnableTime(0) 1507ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman{ 1517ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman getpid_cached = getpid(); 1527ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman char value[PROPERTY_VALUE_MAX]; 1537ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 154c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman if (doLog) { 1557ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 1567ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 157c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman#ifdef TEE_SINK 1587ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman (void) property_get("ro.debuggable", value, "0"); 1597ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman int debuggable = atoi(value); 1607ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman int teeEnabled = 0; 1617ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman if (debuggable) { 1627ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman (void) property_get("af.tee", value, "0"); 1637ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman teeEnabled = atoi(value); 1647ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 165c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman if (teeEnabled & 1) { 1667ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mTeeSinkInputEnabled = true; 1677ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 1687ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman if (teeEnabled & 2) { 1697ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mTeeSinkOutputEnabled = true; 1707ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman } 171c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman if (teeEnabled & 4) { 1727ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman mTeeSinkTrackEnabled = true; 173c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman } 1747ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman#endif 1757ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman} 1767ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 1777ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesmanvoid AudioFlinger::onFirstRef() 178c3e6edba384e023da4e974faca4e28b2276d575fMichael Gottesman{ 1797ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman int rc = 0; 1807ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 1817ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman Mutex::Autolock _l(mLock); 1827ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman 1837ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman /* TODO: move all this work into an Init() function */ 1847ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman char val_str[PROPERTY_VALUE_MAX] = { 0 }; 1857ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 1867ec67156b060ee4e0aac35eed24088ebcbe40aeeMichael Gottesman uint32_t int_val; 187 if (1 == sscanf(val_str, "%u", &int_val)) { 188 mStandbyTimeInNsecs = milliseconds(int_val); 189 ALOGI("Using %u mSec as standby time.", int_val); 190 } else { 191 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 192 ALOGI("Using default %u mSec as standby time.", 193 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 194 } 195 } 196 197 mMode = AUDIO_MODE_NORMAL; 198} 199 200AudioFlinger::~AudioFlinger() 201{ 202 while (!mRecordThreads.isEmpty()) { 203 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 204 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 205 } 206 while (!mPlaybackThreads.isEmpty()) { 207 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 208 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 209 } 210 211 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 212 // no mHardwareLock needed, as there are no other references to this 213 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 214 delete mAudioHwDevs.valueAt(i); 215 } 216} 217 218static const char * const audio_interfaces[] = { 219 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 220 AUDIO_HARDWARE_MODULE_ID_A2DP, 221 AUDIO_HARDWARE_MODULE_ID_USB, 222}; 223#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 224 225AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 226 audio_module_handle_t module, 227 audio_devices_t devices) 228{ 229 // if module is 0, the request comes from an old policy manager and we should load 230 // well known modules 231 if (module == 0) { 232 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 233 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 234 loadHwModule_l(audio_interfaces[i]); 235 } 236 // then try to find a module supporting the requested device. 237 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 238 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 239 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 240 if ((dev->get_supported_devices != NULL) && 241 (dev->get_supported_devices(dev) & devices) == devices) 242 return audioHwDevice; 243 } 244 } else { 245 // check a match for the requested module handle 246 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 247 if (audioHwDevice != NULL) { 248 return audioHwDevice; 249 } 250 } 251 252 return NULL; 253} 254 255void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 256{ 257 const size_t SIZE = 256; 258 char buffer[SIZE]; 259 String8 result; 260 261 result.append("Clients:\n"); 262 for (size_t i = 0; i < mClients.size(); ++i) { 263 sp<Client> client = mClients.valueAt(i).promote(); 264 if (client != 0) { 265 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 266 result.append(buffer); 267 } 268 } 269 270 result.append("Notification Clients:\n"); 271 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 272 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 273 result.append(buffer); 274 } 275 276 result.append("Global session refs:\n"); 277 result.append(" session pid count\n"); 278 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 279 AudioSessionRef *r = mAudioSessionRefs[i]; 280 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 281 result.append(buffer); 282 } 283 write(fd, result.string(), result.size()); 284} 285 286 287void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 288{ 289 const size_t SIZE = 256; 290 char buffer[SIZE]; 291 String8 result; 292 hardware_call_state hardwareStatus = mHardwareStatus; 293 294 snprintf(buffer, SIZE, "Hardware status: %d\n" 295 "Standby Time mSec: %u\n", 296 hardwareStatus, 297 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300} 301 302void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 303{ 304 const size_t SIZE = 256; 305 char buffer[SIZE]; 306 String8 result; 307 snprintf(buffer, SIZE, "Permission Denial: " 308 "can't dump AudioFlinger from pid=%d, uid=%d\n", 309 IPCThreadState::self()->getCallingPid(), 310 IPCThreadState::self()->getCallingUid()); 311 result.append(buffer); 312 write(fd, result.string(), result.size()); 313} 314 315bool AudioFlinger::dumpTryLock(Mutex& mutex) 316{ 317 bool locked = false; 318 for (int i = 0; i < kDumpLockRetries; ++i) { 319 if (mutex.tryLock() == NO_ERROR) { 320 locked = true; 321 break; 322 } 323 usleep(kDumpLockSleepUs); 324 } 325 return locked; 326} 327 328status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 329{ 330 if (!dumpAllowed()) { 331 dumpPermissionDenial(fd, args); 332 } else { 333 // get state of hardware lock 334 bool hardwareLocked = dumpTryLock(mHardwareLock); 335 if (!hardwareLocked) { 336 String8 result(kHardwareLockedString); 337 write(fd, result.string(), result.size()); 338 } else { 339 mHardwareLock.unlock(); 340 } 341 342 bool locked = dumpTryLock(mLock); 343 344 // failed to lock - AudioFlinger is probably deadlocked 345 if (!locked) { 346 String8 result(kDeadlockedString); 347 write(fd, result.string(), result.size()); 348 } 349 350 dumpClients(fd, args); 351 dumpInternals(fd, args); 352 353 // dump playback threads 354 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 355 mPlaybackThreads.valueAt(i)->dump(fd, args); 356 } 357 358 // dump record threads 359 for (size_t i = 0; i < mRecordThreads.size(); i++) { 360 mRecordThreads.valueAt(i)->dump(fd, args); 361 } 362 363 // dump all hardware devs 364 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 365 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 366 dev->dump(dev, fd); 367 } 368 369#ifdef TEE_SINK 370 // dump the serially shared record tee sink 371 if (mRecordTeeSource != 0) { 372 dumpTee(fd, mRecordTeeSource); 373 } 374#endif 375 376 if (locked) { 377 mLock.unlock(); 378 } 379 380 // append a copy of media.log here by forwarding fd to it, but don't attempt 381 // to lookup the service if it's not running, as it will block for a second 382 if (mLogMemoryDealer != 0) { 383 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 384 if (binder != 0) { 385 fdprintf(fd, "\nmedia.log:\n"); 386 Vector<String16> args; 387 binder->dump(fd, args); 388 } 389 } 390 } 391 return NO_ERROR; 392} 393 394sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 395{ 396 // If pid is already in the mClients wp<> map, then use that entry 397 // (for which promote() is always != 0), otherwise create a new entry and Client. 398 sp<Client> client = mClients.valueFor(pid).promote(); 399 if (client == 0) { 400 client = new Client(this, pid); 401 mClients.add(pid, client); 402 } 403 404 return client; 405} 406 407sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 408{ 409 if (mLogMemoryDealer == 0) { 410 return new NBLog::Writer(); 411 } 412 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 413 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 414 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 415 if (binder != 0) { 416 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 417 } 418 return writer; 419} 420 421void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 422{ 423 if (writer == 0) { 424 return; 425 } 426 sp<IMemory> iMemory(writer->getIMemory()); 427 if (iMemory == 0) { 428 return; 429 } 430 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 431 if (binder != 0) { 432 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 433 // Now the media.log remote reference to IMemory is gone. 434 // When our last local reference to IMemory also drops to zero, 435 // the IMemory destructor will deallocate the region from mMemoryDealer. 436 } 437} 438 439// IAudioFlinger interface 440 441 442sp<IAudioTrack> AudioFlinger::createTrack( 443 audio_stream_type_t streamType, 444 uint32_t sampleRate, 445 audio_format_t format, 446 audio_channel_mask_t channelMask, 447 size_t frameCount, 448 IAudioFlinger::track_flags_t *flags, 449 const sp<IMemory>& sharedBuffer, 450 audio_io_handle_t output, 451 pid_t tid, 452 int *sessionId, 453 String8& name, 454 int clientUid, 455 status_t *status) 456{ 457 sp<PlaybackThread::Track> track; 458 sp<TrackHandle> trackHandle; 459 sp<Client> client; 460 status_t lStatus; 461 int lSessionId; 462 463 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 464 // but if someone uses binder directly they could bypass that and cause us to crash 465 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 466 ALOGE("createTrack() invalid stream type %d", streamType); 467 lStatus = BAD_VALUE; 468 goto Exit; 469 } 470 471 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 472 // and we don't yet support 8.24 or 32-bit PCM 473 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 474 ALOGE("createTrack() invalid format %d", format); 475 lStatus = BAD_VALUE; 476 goto Exit; 477 } 478 479 { 480 Mutex::Autolock _l(mLock); 481 PlaybackThread *thread = checkPlaybackThread_l(output); 482 PlaybackThread *effectThread = NULL; 483 if (thread == NULL) { 484 ALOGE("no playback thread found for output handle %d", output); 485 lStatus = BAD_VALUE; 486 goto Exit; 487 } 488 489 pid_t pid = IPCThreadState::self()->getCallingPid(); 490 491 client = registerPid_l(pid); 492 493 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 494 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 495 // check if an effect chain with the same session ID is present on another 496 // output thread and move it here. 497 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 498 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 499 if (mPlaybackThreads.keyAt(i) != output) { 500 uint32_t sessions = t->hasAudioSession(*sessionId); 501 if (sessions & PlaybackThread::EFFECT_SESSION) { 502 effectThread = t.get(); 503 break; 504 } 505 } 506 } 507 lSessionId = *sessionId; 508 } else { 509 // if no audio session id is provided, create one here 510 lSessionId = nextUniqueId(); 511 if (sessionId != NULL) { 512 *sessionId = lSessionId; 513 } 514 } 515 ALOGV("createTrack() lSessionId: %d", lSessionId); 516 517 track = thread->createTrack_l(client, streamType, sampleRate, format, 518 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 519 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 520 521 // move effect chain to this output thread if an effect on same session was waiting 522 // for a track to be created 523 if (lStatus == NO_ERROR && effectThread != NULL) { 524 // no risk of deadlock because AudioFlinger::mLock is held 525 Mutex::Autolock _dl(thread->mLock); 526 Mutex::Autolock _sl(effectThread->mLock); 527 moveEffectChain_l(lSessionId, effectThread, thread, true); 528 } 529 530 // Look for sync events awaiting for a session to be used. 531 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 532 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 533 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 534 if (lStatus == NO_ERROR) { 535 (void) track->setSyncEvent(mPendingSyncEvents[i]); 536 } else { 537 mPendingSyncEvents[i]->cancel(); 538 } 539 mPendingSyncEvents.removeAt(i); 540 i--; 541 } 542 } 543 } 544 545 } 546 547 if (lStatus == NO_ERROR) { 548 // s for server's pid, n for normal mixer name, f for fast index 549 name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, 550 track->fastIndex()); 551 trackHandle = new TrackHandle(track); 552 } else { 553 // remove local strong reference to Client before deleting the Track so that the Client 554 // destructor is called by the TrackBase destructor with mLock held 555 client.clear(); 556 track.clear(); 557 } 558 559Exit: 560 *status = lStatus; 561 return trackHandle; 562} 563 564uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 565{ 566 Mutex::Autolock _l(mLock); 567 PlaybackThread *thread = checkPlaybackThread_l(output); 568 if (thread == NULL) { 569 ALOGW("sampleRate() unknown thread %d", output); 570 return 0; 571 } 572 return thread->sampleRate(); 573} 574 575int AudioFlinger::channelCount(audio_io_handle_t output) const 576{ 577 Mutex::Autolock _l(mLock); 578 PlaybackThread *thread = checkPlaybackThread_l(output); 579 if (thread == NULL) { 580 ALOGW("channelCount() unknown thread %d", output); 581 return 0; 582 } 583 return thread->channelCount(); 584} 585 586audio_format_t AudioFlinger::format(audio_io_handle_t output) const 587{ 588 Mutex::Autolock _l(mLock); 589 PlaybackThread *thread = checkPlaybackThread_l(output); 590 if (thread == NULL) { 591 ALOGW("format() unknown thread %d", output); 592 return AUDIO_FORMAT_INVALID; 593 } 594 return thread->format(); 595} 596 597size_t AudioFlinger::frameCount(audio_io_handle_t output) const 598{ 599 Mutex::Autolock _l(mLock); 600 PlaybackThread *thread = checkPlaybackThread_l(output); 601 if (thread == NULL) { 602 ALOGW("frameCount() unknown thread %d", output); 603 return 0; 604 } 605 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 606 // should examine all callers and fix them to handle smaller counts 607 return thread->frameCount(); 608} 609 610uint32_t AudioFlinger::latency(audio_io_handle_t output) const 611{ 612 Mutex::Autolock _l(mLock); 613 PlaybackThread *thread = checkPlaybackThread_l(output); 614 if (thread == NULL) { 615 ALOGW("latency(): no playback thread found for output handle %d", output); 616 return 0; 617 } 618 return thread->latency(); 619} 620 621status_t AudioFlinger::setMasterVolume(float value) 622{ 623 status_t ret = initCheck(); 624 if (ret != NO_ERROR) { 625 return ret; 626 } 627 628 // check calling permissions 629 if (!settingsAllowed()) { 630 return PERMISSION_DENIED; 631 } 632 633 Mutex::Autolock _l(mLock); 634 mMasterVolume = value; 635 636 // Set master volume in the HALs which support it. 637 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 638 AutoMutex lock(mHardwareLock); 639 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 640 641 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 642 if (dev->canSetMasterVolume()) { 643 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 644 } 645 mHardwareStatus = AUDIO_HW_IDLE; 646 } 647 648 // Now set the master volume in each playback thread. Playback threads 649 // assigned to HALs which do not have master volume support will apply 650 // master volume during the mix operation. Threads with HALs which do 651 // support master volume will simply ignore the setting. 652 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 653 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 654 655 return NO_ERROR; 656} 657 658status_t AudioFlinger::setMode(audio_mode_t mode) 659{ 660 status_t ret = initCheck(); 661 if (ret != NO_ERROR) { 662 return ret; 663 } 664 665 // check calling permissions 666 if (!settingsAllowed()) { 667 return PERMISSION_DENIED; 668 } 669 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 670 ALOGW("Illegal value: setMode(%d)", mode); 671 return BAD_VALUE; 672 } 673 674 { // scope for the lock 675 AutoMutex lock(mHardwareLock); 676 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 677 mHardwareStatus = AUDIO_HW_SET_MODE; 678 ret = dev->set_mode(dev, mode); 679 mHardwareStatus = AUDIO_HW_IDLE; 680 } 681 682 if (NO_ERROR == ret) { 683 Mutex::Autolock _l(mLock); 684 mMode = mode; 685 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 686 mPlaybackThreads.valueAt(i)->setMode(mode); 687 } 688 689 return ret; 690} 691 692status_t AudioFlinger::setMicMute(bool state) 693{ 694 status_t ret = initCheck(); 695 if (ret != NO_ERROR) { 696 return ret; 697 } 698 699 // check calling permissions 700 if (!settingsAllowed()) { 701 return PERMISSION_DENIED; 702 } 703 704 AutoMutex lock(mHardwareLock); 705 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 706 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 707 ret = dev->set_mic_mute(dev, state); 708 mHardwareStatus = AUDIO_HW_IDLE; 709 return ret; 710} 711 712bool AudioFlinger::getMicMute() const 713{ 714 status_t ret = initCheck(); 715 if (ret != NO_ERROR) { 716 return false; 717 } 718 719 bool state = AUDIO_MODE_INVALID; 720 AutoMutex lock(mHardwareLock); 721 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 722 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 723 dev->get_mic_mute(dev, &state); 724 mHardwareStatus = AUDIO_HW_IDLE; 725 return state; 726} 727 728status_t AudioFlinger::setMasterMute(bool muted) 729{ 730 status_t ret = initCheck(); 731 if (ret != NO_ERROR) { 732 return ret; 733 } 734 735 // check calling permissions 736 if (!settingsAllowed()) { 737 return PERMISSION_DENIED; 738 } 739 740 Mutex::Autolock _l(mLock); 741 mMasterMute = muted; 742 743 // Set master mute in the HALs which support it. 744 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 745 AutoMutex lock(mHardwareLock); 746 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 747 748 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 749 if (dev->canSetMasterMute()) { 750 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 751 } 752 mHardwareStatus = AUDIO_HW_IDLE; 753 } 754 755 // Now set the master mute in each playback thread. Playback threads 756 // assigned to HALs which do not have master mute support will apply master 757 // mute during the mix operation. Threads with HALs which do support master 758 // mute will simply ignore the setting. 759 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 760 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 761 762 return NO_ERROR; 763} 764 765float AudioFlinger::masterVolume() const 766{ 767 Mutex::Autolock _l(mLock); 768 return masterVolume_l(); 769} 770 771bool AudioFlinger::masterMute() const 772{ 773 Mutex::Autolock _l(mLock); 774 return masterMute_l(); 775} 776 777float AudioFlinger::masterVolume_l() const 778{ 779 return mMasterVolume; 780} 781 782bool AudioFlinger::masterMute_l() const 783{ 784 return mMasterMute; 785} 786 787status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 788 audio_io_handle_t output) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 ALOGE("setStreamVolume() invalid stream %d", stream); 797 return BAD_VALUE; 798 } 799 800 AutoMutex lock(mLock); 801 PlaybackThread *thread = NULL; 802 if (output) { 803 thread = checkPlaybackThread_l(output); 804 if (thread == NULL) { 805 return BAD_VALUE; 806 } 807 } 808 809 mStreamTypes[stream].volume = value; 810 811 if (thread == NULL) { 812 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 813 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 814 } 815 } else { 816 thread->setStreamVolume(stream, value); 817 } 818 819 return NO_ERROR; 820} 821 822status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 823{ 824 // check calling permissions 825 if (!settingsAllowed()) { 826 return PERMISSION_DENIED; 827 } 828 829 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 830 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 831 ALOGE("setStreamMute() invalid stream %d", stream); 832 return BAD_VALUE; 833 } 834 835 AutoMutex lock(mLock); 836 mStreamTypes[stream].mute = muted; 837 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 838 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 839 840 return NO_ERROR; 841} 842 843float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 844{ 845 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 846 return 0.0f; 847 } 848 849 AutoMutex lock(mLock); 850 float volume; 851 if (output) { 852 PlaybackThread *thread = checkPlaybackThread_l(output); 853 if (thread == NULL) { 854 return 0.0f; 855 } 856 volume = thread->streamVolume(stream); 857 } else { 858 volume = streamVolume_l(stream); 859 } 860 861 return volume; 862} 863 864bool AudioFlinger::streamMute(audio_stream_type_t stream) const 865{ 866 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 867 return true; 868 } 869 870 AutoMutex lock(mLock); 871 return streamMute_l(stream); 872} 873 874status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 875{ 876 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 877 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 878 879 // check calling permissions 880 if (!settingsAllowed()) { 881 return PERMISSION_DENIED; 882 } 883 884 // ioHandle == 0 means the parameters are global to the audio hardware interface 885 if (ioHandle == 0) { 886 Mutex::Autolock _l(mLock); 887 status_t final_result = NO_ERROR; 888 { 889 AutoMutex lock(mHardwareLock); 890 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 891 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 892 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 893 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 894 final_result = result ?: final_result; 895 } 896 mHardwareStatus = AUDIO_HW_IDLE; 897 } 898 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 899 AudioParameter param = AudioParameter(keyValuePairs); 900 String8 value; 901 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 902 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 903 if (mBtNrecIsOff != btNrecIsOff) { 904 for (size_t i = 0; i < mRecordThreads.size(); i++) { 905 sp<RecordThread> thread = mRecordThreads.valueAt(i); 906 audio_devices_t device = thread->inDevice(); 907 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 908 // collect all of the thread's session IDs 909 KeyedVector<int, bool> ids = thread->sessionIds(); 910 // suspend effects associated with those session IDs 911 for (size_t j = 0; j < ids.size(); ++j) { 912 int sessionId = ids.keyAt(j); 913 thread->setEffectSuspended(FX_IID_AEC, 914 suspend, 915 sessionId); 916 thread->setEffectSuspended(FX_IID_NS, 917 suspend, 918 sessionId); 919 } 920 } 921 mBtNrecIsOff = btNrecIsOff; 922 } 923 } 924 String8 screenState; 925 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 926 bool isOff = screenState == "off"; 927 if (isOff != (AudioFlinger::mScreenState & 1)) { 928 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 929 } 930 } 931 return final_result; 932 } 933 934 // hold a strong ref on thread in case closeOutput() or closeInput() is called 935 // and the thread is exited once the lock is released 936 sp<ThreadBase> thread; 937 { 938 Mutex::Autolock _l(mLock); 939 thread = checkPlaybackThread_l(ioHandle); 940 if (thread == 0) { 941 thread = checkRecordThread_l(ioHandle); 942 } else if (thread == primaryPlaybackThread_l()) { 943 // indicate output device change to all input threads for pre processing 944 AudioParameter param = AudioParameter(keyValuePairs); 945 int value; 946 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 947 (value != 0)) { 948 for (size_t i = 0; i < mRecordThreads.size(); i++) { 949 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 950 } 951 } 952 } 953 } 954 if (thread != 0) { 955 return thread->setParameters(keyValuePairs); 956 } 957 return BAD_VALUE; 958} 959 960String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 961{ 962 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 963 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 964 965 Mutex::Autolock _l(mLock); 966 967 if (ioHandle == 0) { 968 String8 out_s8; 969 970 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 971 char *s; 972 { 973 AutoMutex lock(mHardwareLock); 974 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 975 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 976 s = dev->get_parameters(dev, keys.string()); 977 mHardwareStatus = AUDIO_HW_IDLE; 978 } 979 out_s8 += String8(s ? s : ""); 980 free(s); 981 } 982 return out_s8; 983 } 984 985 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 986 if (playbackThread != NULL) { 987 return playbackThread->getParameters(keys); 988 } 989 RecordThread *recordThread = checkRecordThread_l(ioHandle); 990 if (recordThread != NULL) { 991 return recordThread->getParameters(keys); 992 } 993 return String8(""); 994} 995 996size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 997 audio_channel_mask_t channelMask) const 998{ 999 status_t ret = initCheck(); 1000 if (ret != NO_ERROR) { 1001 return 0; 1002 } 1003 1004 AutoMutex lock(mHardwareLock); 1005 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1006 struct audio_config config; 1007 memset(&config, 0, sizeof(config)); 1008 config.sample_rate = sampleRate; 1009 config.channel_mask = channelMask; 1010 config.format = format; 1011 1012 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1013 size_t size = dev->get_input_buffer_size(dev, &config); 1014 mHardwareStatus = AUDIO_HW_IDLE; 1015 return size; 1016} 1017 1018unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1019{ 1020 Mutex::Autolock _l(mLock); 1021 1022 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1023 if (recordThread != NULL) { 1024 return recordThread->getInputFramesLost(); 1025 } 1026 return 0; 1027} 1028 1029status_t AudioFlinger::setVoiceVolume(float value) 1030{ 1031 status_t ret = initCheck(); 1032 if (ret != NO_ERROR) { 1033 return ret; 1034 } 1035 1036 // check calling permissions 1037 if (!settingsAllowed()) { 1038 return PERMISSION_DENIED; 1039 } 1040 1041 AutoMutex lock(mHardwareLock); 1042 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1043 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1044 ret = dev->set_voice_volume(dev, value); 1045 mHardwareStatus = AUDIO_HW_IDLE; 1046 1047 return ret; 1048} 1049 1050status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1051 audio_io_handle_t output) const 1052{ 1053 status_t status; 1054 1055 Mutex::Autolock _l(mLock); 1056 1057 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1058 if (playbackThread != NULL) { 1059 return playbackThread->getRenderPosition(halFrames, dspFrames); 1060 } 1061 1062 return BAD_VALUE; 1063} 1064 1065void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1066{ 1067 1068 Mutex::Autolock _l(mLock); 1069 1070 pid_t pid = IPCThreadState::self()->getCallingPid(); 1071 if (mNotificationClients.indexOfKey(pid) < 0) { 1072 sp<NotificationClient> notificationClient = new NotificationClient(this, 1073 client, 1074 pid); 1075 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1076 1077 mNotificationClients.add(pid, notificationClient); 1078 1079 sp<IBinder> binder = client->asBinder(); 1080 binder->linkToDeath(notificationClient); 1081 1082 // the config change is always sent from playback or record threads to avoid deadlock 1083 // with AudioSystem::gLock 1084 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1085 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1086 } 1087 1088 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1089 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1090 } 1091 } 1092} 1093 1094void AudioFlinger::removeNotificationClient(pid_t pid) 1095{ 1096 Mutex::Autolock _l(mLock); 1097 1098 mNotificationClients.removeItem(pid); 1099 1100 ALOGV("%d died, releasing its sessions", pid); 1101 size_t num = mAudioSessionRefs.size(); 1102 bool removed = false; 1103 for (size_t i = 0; i< num; ) { 1104 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1105 ALOGV(" pid %d @ %d", ref->mPid, i); 1106 if (ref->mPid == pid) { 1107 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1108 mAudioSessionRefs.removeAt(i); 1109 delete ref; 1110 removed = true; 1111 num--; 1112 } else { 1113 i++; 1114 } 1115 } 1116 if (removed) { 1117 purgeStaleEffects_l(); 1118 } 1119} 1120 1121// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1122void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1123{ 1124 size_t size = mNotificationClients.size(); 1125 for (size_t i = 0; i < size; i++) { 1126 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1127 param2); 1128 } 1129} 1130 1131// removeClient_l() must be called with AudioFlinger::mLock held 1132void AudioFlinger::removeClient_l(pid_t pid) 1133{ 1134 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1135 IPCThreadState::self()->getCallingPid()); 1136 mClients.removeItem(pid); 1137} 1138 1139// getEffectThread_l() must be called with AudioFlinger::mLock held 1140sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1141{ 1142 sp<PlaybackThread> thread; 1143 1144 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1145 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1146 ALOG_ASSERT(thread == 0); 1147 thread = mPlaybackThreads.valueAt(i); 1148 } 1149 } 1150 1151 return thread; 1152} 1153 1154 1155 1156// ---------------------------------------------------------------------------- 1157 1158AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1159 : RefBase(), 1160 mAudioFlinger(audioFlinger), 1161 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1162 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1163 mPid(pid), 1164 mTimedTrackCount(0) 1165{ 1166 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1167} 1168 1169// Client destructor must be called with AudioFlinger::mLock held 1170AudioFlinger::Client::~Client() 1171{ 1172 mAudioFlinger->removeClient_l(mPid); 1173} 1174 1175sp<MemoryDealer> AudioFlinger::Client::heap() const 1176{ 1177 return mMemoryDealer; 1178} 1179 1180// Reserve one of the limited slots for a timed audio track associated 1181// with this client 1182bool AudioFlinger::Client::reserveTimedTrack() 1183{ 1184 const int kMaxTimedTracksPerClient = 4; 1185 1186 Mutex::Autolock _l(mTimedTrackLock); 1187 1188 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1189 ALOGW("can not create timed track - pid %d has exceeded the limit", 1190 mPid); 1191 return false; 1192 } 1193 1194 mTimedTrackCount++; 1195 return true; 1196} 1197 1198// Release a slot for a timed audio track 1199void AudioFlinger::Client::releaseTimedTrack() 1200{ 1201 Mutex::Autolock _l(mTimedTrackLock); 1202 mTimedTrackCount--; 1203} 1204 1205// ---------------------------------------------------------------------------- 1206 1207AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1208 const sp<IAudioFlingerClient>& client, 1209 pid_t pid) 1210 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1211{ 1212} 1213 1214AudioFlinger::NotificationClient::~NotificationClient() 1215{ 1216} 1217 1218void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1219{ 1220 sp<NotificationClient> keep(this); 1221 mAudioFlinger->removeNotificationClient(mPid); 1222} 1223 1224 1225// ---------------------------------------------------------------------------- 1226 1227static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1228 return audio_is_remote_submix_device(inDevice); 1229} 1230 1231sp<IAudioRecord> AudioFlinger::openRecord( 1232 audio_io_handle_t input, 1233 uint32_t sampleRate, 1234 audio_format_t format, 1235 audio_channel_mask_t channelMask, 1236 size_t frameCount, 1237 IAudioFlinger::track_flags_t *flags, 1238 pid_t tid, 1239 int *sessionId, 1240 status_t *status) 1241{ 1242 sp<RecordThread::RecordTrack> recordTrack; 1243 sp<RecordHandle> recordHandle; 1244 sp<Client> client; 1245 status_t lStatus; 1246 RecordThread *thread; 1247 size_t inFrameCount; 1248 int lSessionId; 1249 1250 // check calling permissions 1251 if (!recordingAllowed()) { 1252 ALOGE("openRecord() permission denied: recording not allowed"); 1253 lStatus = PERMISSION_DENIED; 1254 goto Exit; 1255 } 1256 1257 if (format != AUDIO_FORMAT_PCM_16_BIT) { 1258 ALOGE("openRecord() invalid format %d", format); 1259 lStatus = BAD_VALUE; 1260 goto Exit; 1261 } 1262 1263 // add client to list 1264 { // scope for mLock 1265 Mutex::Autolock _l(mLock); 1266 thread = checkRecordThread_l(input); 1267 if (thread == NULL) { 1268 ALOGE("openRecord() checkRecordThread_l failed"); 1269 lStatus = BAD_VALUE; 1270 goto Exit; 1271 } 1272 1273 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1274 && !captureAudioOutputAllowed()) { 1275 ALOGE("openRecord() permission denied: capture not allowed"); 1276 lStatus = PERMISSION_DENIED; 1277 goto Exit; 1278 } 1279 1280 pid_t pid = IPCThreadState::self()->getCallingPid(); 1281 client = registerPid_l(pid); 1282 1283 // If no audio session id is provided, create one here 1284 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1285 lSessionId = *sessionId; 1286 } else { 1287 lSessionId = nextUniqueId(); 1288 if (sessionId != NULL) { 1289 *sessionId = lSessionId; 1290 } 1291 } 1292 // create new record track. 1293 // The record track uses one track in mHardwareMixerThread by convention. 1294 // TODO: the uid should be passed in as a parameter to openRecord 1295 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1296 frameCount, lSessionId, 1297 IPCThreadState::self()->getCallingUid(), 1298 flags, tid, &lStatus); 1299 LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR)); 1300 } 1301 1302 if (lStatus != NO_ERROR) { 1303 // remove local strong reference to Client before deleting the RecordTrack so that the 1304 // Client destructor is called by the TrackBase destructor with mLock held 1305 client.clear(); 1306 recordTrack.clear(); 1307 goto Exit; 1308 } 1309 1310 // return handle to client 1311 recordHandle = new RecordHandle(recordTrack); 1312 1313Exit: 1314 *status = lStatus; 1315 return recordHandle; 1316} 1317 1318 1319 1320// ---------------------------------------------------------------------------- 1321 1322audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1323{ 1324 if (!settingsAllowed()) { 1325 return 0; 1326 } 1327 Mutex::Autolock _l(mLock); 1328 return loadHwModule_l(name); 1329} 1330 1331// loadHwModule_l() must be called with AudioFlinger::mLock held 1332audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1333{ 1334 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1335 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1336 ALOGW("loadHwModule() module %s already loaded", name); 1337 return mAudioHwDevs.keyAt(i); 1338 } 1339 } 1340 1341 audio_hw_device_t *dev; 1342 1343 int rc = load_audio_interface(name, &dev); 1344 if (rc) { 1345 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1346 return 0; 1347 } 1348 1349 mHardwareStatus = AUDIO_HW_INIT; 1350 rc = dev->init_check(dev); 1351 mHardwareStatus = AUDIO_HW_IDLE; 1352 if (rc) { 1353 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1354 return 0; 1355 } 1356 1357 // Check and cache this HAL's level of support for master mute and master 1358 // volume. If this is the first HAL opened, and it supports the get 1359 // methods, use the initial values provided by the HAL as the current 1360 // master mute and volume settings. 1361 1362 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1363 { // scope for auto-lock pattern 1364 AutoMutex lock(mHardwareLock); 1365 1366 if (0 == mAudioHwDevs.size()) { 1367 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1368 if (NULL != dev->get_master_volume) { 1369 float mv; 1370 if (OK == dev->get_master_volume(dev, &mv)) { 1371 mMasterVolume = mv; 1372 } 1373 } 1374 1375 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1376 if (NULL != dev->get_master_mute) { 1377 bool mm; 1378 if (OK == dev->get_master_mute(dev, &mm)) { 1379 mMasterMute = mm; 1380 } 1381 } 1382 } 1383 1384 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1385 if ((NULL != dev->set_master_volume) && 1386 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1387 flags = static_cast<AudioHwDevice::Flags>(flags | 1388 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1389 } 1390 1391 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1392 if ((NULL != dev->set_master_mute) && 1393 (OK == dev->set_master_mute(dev, mMasterMute))) { 1394 flags = static_cast<AudioHwDevice::Flags>(flags | 1395 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1396 } 1397 1398 mHardwareStatus = AUDIO_HW_IDLE; 1399 } 1400 1401 audio_module_handle_t handle = nextUniqueId(); 1402 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1403 1404 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1405 name, dev->common.module->name, dev->common.module->id, handle); 1406 1407 return handle; 1408 1409} 1410 1411// ---------------------------------------------------------------------------- 1412 1413uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1414{ 1415 Mutex::Autolock _l(mLock); 1416 PlaybackThread *thread = primaryPlaybackThread_l(); 1417 return thread != NULL ? thread->sampleRate() : 0; 1418} 1419 1420size_t AudioFlinger::getPrimaryOutputFrameCount() 1421{ 1422 Mutex::Autolock _l(mLock); 1423 PlaybackThread *thread = primaryPlaybackThread_l(); 1424 return thread != NULL ? thread->frameCountHAL() : 0; 1425} 1426 1427// ---------------------------------------------------------------------------- 1428 1429status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1430{ 1431 uid_t uid = IPCThreadState::self()->getCallingUid(); 1432 if (uid != AID_SYSTEM) { 1433 return PERMISSION_DENIED; 1434 } 1435 Mutex::Autolock _l(mLock); 1436 if (mIsDeviceTypeKnown) { 1437 return INVALID_OPERATION; 1438 } 1439 mIsLowRamDevice = isLowRamDevice; 1440 mIsDeviceTypeKnown = true; 1441 return NO_ERROR; 1442} 1443 1444// ---------------------------------------------------------------------------- 1445 1446audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1447 audio_devices_t *pDevices, 1448 uint32_t *pSamplingRate, 1449 audio_format_t *pFormat, 1450 audio_channel_mask_t *pChannelMask, 1451 uint32_t *pLatencyMs, 1452 audio_output_flags_t flags, 1453 const audio_offload_info_t *offloadInfo) 1454{ 1455 struct audio_config config; 1456 memset(&config, 0, sizeof(config)); 1457 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1458 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1459 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1460 if (offloadInfo != NULL) { 1461 config.offload_info = *offloadInfo; 1462 } 1463 1464 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1465 module, 1466 (pDevices != NULL) ? *pDevices : 0, 1467 config.sample_rate, 1468 config.format, 1469 config.channel_mask, 1470 flags); 1471 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1472 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); 1473 1474 if (pDevices == NULL || *pDevices == 0) { 1475 return 0; 1476 } 1477 1478 Mutex::Autolock _l(mLock); 1479 1480 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); 1481 if (outHwDev == NULL) { 1482 return 0; 1483 } 1484 1485 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1486 audio_io_handle_t id = nextUniqueId(); 1487 1488 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1489 1490 audio_stream_out_t *outStream = NULL; 1491 status_t status = hwDevHal->open_output_stream(hwDevHal, 1492 id, 1493 *pDevices, 1494 (audio_output_flags_t)flags, 1495 &config, 1496 &outStream); 1497 1498 mHardwareStatus = AUDIO_HW_IDLE; 1499 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1500 "Channels %x, status %d", 1501 outStream, 1502 config.sample_rate, 1503 config.format, 1504 config.channel_mask, 1505 status); 1506 1507 if (status == NO_ERROR && outStream != NULL) { 1508 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1509 1510 PlaybackThread *thread; 1511 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1512 thread = new OffloadThread(this, output, id, *pDevices); 1513 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1514 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1515 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1516 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1517 thread = new DirectOutputThread(this, output, id, *pDevices); 1518 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1519 } else { 1520 thread = new MixerThread(this, output, id, *pDevices); 1521 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1522 } 1523 mPlaybackThreads.add(id, thread); 1524 1525 if (pSamplingRate != NULL) { 1526 *pSamplingRate = config.sample_rate; 1527 } 1528 if (pFormat != NULL) { 1529 *pFormat = config.format; 1530 } 1531 if (pChannelMask != NULL) { 1532 *pChannelMask = config.channel_mask; 1533 } 1534 if (pLatencyMs != NULL) { 1535 *pLatencyMs = thread->latency(); 1536 } 1537 1538 // notify client processes of the new output creation 1539 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1540 1541 // the first primary output opened designates the primary hw device 1542 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1543 ALOGI("Using module %d has the primary audio interface", module); 1544 mPrimaryHardwareDev = outHwDev; 1545 1546 AutoMutex lock(mHardwareLock); 1547 mHardwareStatus = AUDIO_HW_SET_MODE; 1548 hwDevHal->set_mode(hwDevHal, mMode); 1549 mHardwareStatus = AUDIO_HW_IDLE; 1550 } 1551 return id; 1552 } 1553 1554 return 0; 1555} 1556 1557audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1558 audio_io_handle_t output2) 1559{ 1560 Mutex::Autolock _l(mLock); 1561 MixerThread *thread1 = checkMixerThread_l(output1); 1562 MixerThread *thread2 = checkMixerThread_l(output2); 1563 1564 if (thread1 == NULL || thread2 == NULL) { 1565 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1566 output2); 1567 return 0; 1568 } 1569 1570 audio_io_handle_t id = nextUniqueId(); 1571 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1572 thread->addOutputTrack(thread2); 1573 mPlaybackThreads.add(id, thread); 1574 // notify client processes of the new output creation 1575 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1576 return id; 1577} 1578 1579status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1580{ 1581 return closeOutput_nonvirtual(output); 1582} 1583 1584status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1585{ 1586 // keep strong reference on the playback thread so that 1587 // it is not destroyed while exit() is executed 1588 sp<PlaybackThread> thread; 1589 { 1590 Mutex::Autolock _l(mLock); 1591 thread = checkPlaybackThread_l(output); 1592 if (thread == NULL) { 1593 return BAD_VALUE; 1594 } 1595 1596 ALOGV("closeOutput() %d", output); 1597 1598 if (thread->type() == ThreadBase::MIXER) { 1599 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1600 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1601 DuplicatingThread *dupThread = 1602 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1603 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1604 1605 } 1606 } 1607 } 1608 1609 1610 mPlaybackThreads.removeItem(output); 1611 // save all effects to the default thread 1612 if (mPlaybackThreads.size()) { 1613 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1614 if (dstThread != NULL) { 1615 // audioflinger lock is held here so the acquisition order of thread locks does not 1616 // matter 1617 Mutex::Autolock _dl(dstThread->mLock); 1618 Mutex::Autolock _sl(thread->mLock); 1619 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1620 for (size_t i = 0; i < effectChains.size(); i ++) { 1621 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1622 } 1623 } 1624 } 1625 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1626 } 1627 thread->exit(); 1628 // The thread entity (active unit of execution) is no longer running here, 1629 // but the ThreadBase container still exists. 1630 1631 if (thread->type() != ThreadBase::DUPLICATING) { 1632 AudioStreamOut *out = thread->clearOutput(); 1633 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1634 // from now on thread->mOutput is NULL 1635 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1636 delete out; 1637 } 1638 return NO_ERROR; 1639} 1640 1641status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1642{ 1643 Mutex::Autolock _l(mLock); 1644 PlaybackThread *thread = checkPlaybackThread_l(output); 1645 1646 if (thread == NULL) { 1647 return BAD_VALUE; 1648 } 1649 1650 ALOGV("suspendOutput() %d", output); 1651 thread->suspend(); 1652 1653 return NO_ERROR; 1654} 1655 1656status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1657{ 1658 Mutex::Autolock _l(mLock); 1659 PlaybackThread *thread = checkPlaybackThread_l(output); 1660 1661 if (thread == NULL) { 1662 return BAD_VALUE; 1663 } 1664 1665 ALOGV("restoreOutput() %d", output); 1666 1667 thread->restore(); 1668 1669 return NO_ERROR; 1670} 1671 1672audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1673 audio_devices_t *pDevices, 1674 uint32_t *pSamplingRate, 1675 audio_format_t *pFormat, 1676 audio_channel_mask_t *pChannelMask) 1677{ 1678 struct audio_config config; 1679 memset(&config, 0, sizeof(config)); 1680 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1681 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1682 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1683 1684 uint32_t reqSamplingRate = config.sample_rate; 1685 audio_format_t reqFormat = config.format; 1686 audio_channel_mask_t reqChannelMask = config.channel_mask; 1687 1688 if (pDevices == NULL || *pDevices == 0) { 1689 return 0; 1690 } 1691 1692 Mutex::Autolock _l(mLock); 1693 1694 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); 1695 if (inHwDev == NULL) { 1696 return 0; 1697 } 1698 1699 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1700 audio_io_handle_t id = nextUniqueId(); 1701 1702 audio_stream_in_t *inStream = NULL; 1703 status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1704 &inStream); 1705 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1706 "status %d", 1707 inStream, 1708 config.sample_rate, 1709 config.format, 1710 config.channel_mask, 1711 status); 1712 1713 // If the input could not be opened with the requested parameters and we can handle the 1714 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1715 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1716 if (status == BAD_VALUE && 1717 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1718 (config.sample_rate <= 2 * reqSamplingRate) && 1719 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) { 1720 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1721 inStream = NULL; 1722 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1723 } 1724 1725 if (status == NO_ERROR && inStream != NULL) { 1726 1727#ifdef TEE_SINK 1728 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1729 // or (re-)create if current Pipe is idle and does not match the new format 1730 sp<NBAIO_Sink> teeSink; 1731 enum { 1732 TEE_SINK_NO, // don't copy input 1733 TEE_SINK_NEW, // copy input using a new pipe 1734 TEE_SINK_OLD, // copy input using an existing pipe 1735 } kind; 1736 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1737 popcount(inStream->common.get_channels(&inStream->common))); 1738 if (!mTeeSinkInputEnabled) { 1739 kind = TEE_SINK_NO; 1740 } else if (format == Format_Invalid) { 1741 kind = TEE_SINK_NO; 1742 } else if (mRecordTeeSink == 0) { 1743 kind = TEE_SINK_NEW; 1744 } else if (mRecordTeeSink->getStrongCount() != 1) { 1745 kind = TEE_SINK_NO; 1746 } else if (format == mRecordTeeSink->format()) { 1747 kind = TEE_SINK_OLD; 1748 } else { 1749 kind = TEE_SINK_NEW; 1750 } 1751 switch (kind) { 1752 case TEE_SINK_NEW: { 1753 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1754 size_t numCounterOffers = 0; 1755 const NBAIO_Format offers[1] = {format}; 1756 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1757 ALOG_ASSERT(index == 0); 1758 PipeReader *pipeReader = new PipeReader(*pipe); 1759 numCounterOffers = 0; 1760 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1761 ALOG_ASSERT(index == 0); 1762 mRecordTeeSink = pipe; 1763 mRecordTeeSource = pipeReader; 1764 teeSink = pipe; 1765 } 1766 break; 1767 case TEE_SINK_OLD: 1768 teeSink = mRecordTeeSink; 1769 break; 1770 case TEE_SINK_NO: 1771 default: 1772 break; 1773 } 1774#endif 1775 1776 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1777 1778 // Start record thread 1779 // RecordThread requires both input and output device indication to forward to audio 1780 // pre processing modules 1781 RecordThread *thread = new RecordThread(this, 1782 input, 1783 reqSamplingRate, 1784 reqChannelMask, 1785 id, 1786 primaryOutputDevice_l(), 1787 *pDevices 1788#ifdef TEE_SINK 1789 , teeSink 1790#endif 1791 ); 1792 mRecordThreads.add(id, thread); 1793 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1794 if (pSamplingRate != NULL) { 1795 *pSamplingRate = reqSamplingRate; 1796 } 1797 if (pFormat != NULL) { 1798 *pFormat = config.format; 1799 } 1800 if (pChannelMask != NULL) { 1801 *pChannelMask = reqChannelMask; 1802 } 1803 1804 // notify client processes of the new input creation 1805 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1806 return id; 1807 } 1808 1809 return 0; 1810} 1811 1812status_t AudioFlinger::closeInput(audio_io_handle_t input) 1813{ 1814 return closeInput_nonvirtual(input); 1815} 1816 1817status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1818{ 1819 // keep strong reference on the record thread so that 1820 // it is not destroyed while exit() is executed 1821 sp<RecordThread> thread; 1822 { 1823 Mutex::Autolock _l(mLock); 1824 thread = checkRecordThread_l(input); 1825 if (thread == 0) { 1826 return BAD_VALUE; 1827 } 1828 1829 ALOGV("closeInput() %d", input); 1830 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1831 mRecordThreads.removeItem(input); 1832 } 1833 thread->exit(); 1834 // The thread entity (active unit of execution) is no longer running here, 1835 // but the ThreadBase container still exists. 1836 1837 AudioStreamIn *in = thread->clearInput(); 1838 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1839 // from now on thread->mInput is NULL 1840 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1841 delete in; 1842 1843 return NO_ERROR; 1844} 1845 1846status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1847{ 1848 Mutex::Autolock _l(mLock); 1849 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1850 1851 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1852 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1853 thread->invalidateTracks(stream); 1854 } 1855 1856 return NO_ERROR; 1857} 1858 1859 1860int AudioFlinger::newAudioSessionId() 1861{ 1862 return nextUniqueId(); 1863} 1864 1865void AudioFlinger::acquireAudioSessionId(int audioSession) 1866{ 1867 Mutex::Autolock _l(mLock); 1868 pid_t caller = IPCThreadState::self()->getCallingPid(); 1869 ALOGV("acquiring %d from %d", audioSession, caller); 1870 1871 // Ignore requests received from processes not known as notification client. The request 1872 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 1873 // called from a different pid leaving a stale session reference. Also we don't know how 1874 // to clear this reference if the client process dies. 1875 if (mNotificationClients.indexOfKey(caller) < 0) { 1876 ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 1877 return; 1878 } 1879 1880 size_t num = mAudioSessionRefs.size(); 1881 for (size_t i = 0; i< num; i++) { 1882 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1883 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1884 ref->mCnt++; 1885 ALOGV(" incremented refcount to %d", ref->mCnt); 1886 return; 1887 } 1888 } 1889 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1890 ALOGV(" added new entry for %d", audioSession); 1891} 1892 1893void AudioFlinger::releaseAudioSessionId(int audioSession) 1894{ 1895 Mutex::Autolock _l(mLock); 1896 pid_t caller = IPCThreadState::self()->getCallingPid(); 1897 ALOGV("releasing %d from %d", audioSession, caller); 1898 size_t num = mAudioSessionRefs.size(); 1899 for (size_t i = 0; i< num; i++) { 1900 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1901 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1902 ref->mCnt--; 1903 ALOGV(" decremented refcount to %d", ref->mCnt); 1904 if (ref->mCnt == 0) { 1905 mAudioSessionRefs.removeAt(i); 1906 delete ref; 1907 purgeStaleEffects_l(); 1908 } 1909 return; 1910 } 1911 } 1912 // If the caller is mediaserver it is likely that the session being released was acquired 1913 // on behalf of a process not in notification clients and we ignore the warning. 1914 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 1915} 1916 1917void AudioFlinger::purgeStaleEffects_l() { 1918 1919 ALOGV("purging stale effects"); 1920 1921 Vector< sp<EffectChain> > chains; 1922 1923 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1924 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1925 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1926 sp<EffectChain> ec = t->mEffectChains[j]; 1927 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1928 chains.push(ec); 1929 } 1930 } 1931 } 1932 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1933 sp<RecordThread> t = mRecordThreads.valueAt(i); 1934 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1935 sp<EffectChain> ec = t->mEffectChains[j]; 1936 chains.push(ec); 1937 } 1938 } 1939 1940 for (size_t i = 0; i < chains.size(); i++) { 1941 sp<EffectChain> ec = chains[i]; 1942 int sessionid = ec->sessionId(); 1943 sp<ThreadBase> t = ec->mThread.promote(); 1944 if (t == 0) { 1945 continue; 1946 } 1947 size_t numsessionrefs = mAudioSessionRefs.size(); 1948 bool found = false; 1949 for (size_t k = 0; k < numsessionrefs; k++) { 1950 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1951 if (ref->mSessionid == sessionid) { 1952 ALOGV(" session %d still exists for %d with %d refs", 1953 sessionid, ref->mPid, ref->mCnt); 1954 found = true; 1955 break; 1956 } 1957 } 1958 if (!found) { 1959 Mutex::Autolock _l(t->mLock); 1960 // remove all effects from the chain 1961 while (ec->mEffects.size()) { 1962 sp<EffectModule> effect = ec->mEffects[0]; 1963 effect->unPin(); 1964 t->removeEffect_l(effect); 1965 if (effect->purgeHandles()) { 1966 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1967 } 1968 AudioSystem::unregisterEffect(effect->id()); 1969 } 1970 } 1971 } 1972 return; 1973} 1974 1975// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1976AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1977{ 1978 return mPlaybackThreads.valueFor(output).get(); 1979} 1980 1981// checkMixerThread_l() must be called with AudioFlinger::mLock held 1982AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1983{ 1984 PlaybackThread *thread = checkPlaybackThread_l(output); 1985 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1986} 1987 1988// checkRecordThread_l() must be called with AudioFlinger::mLock held 1989AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1990{ 1991 return mRecordThreads.valueFor(input).get(); 1992} 1993 1994uint32_t AudioFlinger::nextUniqueId() 1995{ 1996 return android_atomic_inc(&mNextUniqueId); 1997} 1998 1999AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2000{ 2001 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2002 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2003 AudioStreamOut *output = thread->getOutput(); 2004 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2005 return thread; 2006 } 2007 } 2008 return NULL; 2009} 2010 2011audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2012{ 2013 PlaybackThread *thread = primaryPlaybackThread_l(); 2014 2015 if (thread == NULL) { 2016 return 0; 2017 } 2018 2019 return thread->outDevice(); 2020} 2021 2022sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2023 int triggerSession, 2024 int listenerSession, 2025 sync_event_callback_t callBack, 2026 void *cookie) 2027{ 2028 Mutex::Autolock _l(mLock); 2029 2030 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2031 status_t playStatus = NAME_NOT_FOUND; 2032 status_t recStatus = NAME_NOT_FOUND; 2033 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2034 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2035 if (playStatus == NO_ERROR) { 2036 return event; 2037 } 2038 } 2039 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2040 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2041 if (recStatus == NO_ERROR) { 2042 return event; 2043 } 2044 } 2045 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2046 mPendingSyncEvents.add(event); 2047 } else { 2048 ALOGV("createSyncEvent() invalid event %d", event->type()); 2049 event.clear(); 2050 } 2051 return event; 2052} 2053 2054// ---------------------------------------------------------------------------- 2055// Effect management 2056// ---------------------------------------------------------------------------- 2057 2058 2059status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2060{ 2061 Mutex::Autolock _l(mLock); 2062 return EffectQueryNumberEffects(numEffects); 2063} 2064 2065status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2066{ 2067 Mutex::Autolock _l(mLock); 2068 return EffectQueryEffect(index, descriptor); 2069} 2070 2071status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2072 effect_descriptor_t *descriptor) const 2073{ 2074 Mutex::Autolock _l(mLock); 2075 return EffectGetDescriptor(pUuid, descriptor); 2076} 2077 2078 2079sp<IEffect> AudioFlinger::createEffect( 2080 effect_descriptor_t *pDesc, 2081 const sp<IEffectClient>& effectClient, 2082 int32_t priority, 2083 audio_io_handle_t io, 2084 int sessionId, 2085 status_t *status, 2086 int *id, 2087 int *enabled) 2088{ 2089 status_t lStatus = NO_ERROR; 2090 sp<EffectHandle> handle; 2091 effect_descriptor_t desc; 2092 2093 pid_t pid = IPCThreadState::self()->getCallingPid(); 2094 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2095 pid, effectClient.get(), priority, sessionId, io); 2096 2097 if (pDesc == NULL) { 2098 lStatus = BAD_VALUE; 2099 goto Exit; 2100 } 2101 2102 // check audio settings permission for global effects 2103 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2104 lStatus = PERMISSION_DENIED; 2105 goto Exit; 2106 } 2107 2108 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2109 // that can only be created by audio policy manager (running in same process) 2110 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2111 lStatus = PERMISSION_DENIED; 2112 goto Exit; 2113 } 2114 2115 { 2116 if (!EffectIsNullUuid(&pDesc->uuid)) { 2117 // if uuid is specified, request effect descriptor 2118 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2119 if (lStatus < 0) { 2120 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2121 goto Exit; 2122 } 2123 } else { 2124 // if uuid is not specified, look for an available implementation 2125 // of the required type in effect factory 2126 if (EffectIsNullUuid(&pDesc->type)) { 2127 ALOGW("createEffect() no effect type"); 2128 lStatus = BAD_VALUE; 2129 goto Exit; 2130 } 2131 uint32_t numEffects = 0; 2132 effect_descriptor_t d; 2133 d.flags = 0; // prevent compiler warning 2134 bool found = false; 2135 2136 lStatus = EffectQueryNumberEffects(&numEffects); 2137 if (lStatus < 0) { 2138 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2139 goto Exit; 2140 } 2141 for (uint32_t i = 0; i < numEffects; i++) { 2142 lStatus = EffectQueryEffect(i, &desc); 2143 if (lStatus < 0) { 2144 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2145 continue; 2146 } 2147 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2148 // If matching type found save effect descriptor. If the session is 2149 // 0 and the effect is not auxiliary, continue enumeration in case 2150 // an auxiliary version of this effect type is available 2151 found = true; 2152 d = desc; 2153 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2154 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2155 break; 2156 } 2157 } 2158 } 2159 if (!found) { 2160 lStatus = BAD_VALUE; 2161 ALOGW("createEffect() effect not found"); 2162 goto Exit; 2163 } 2164 // For same effect type, chose auxiliary version over insert version if 2165 // connect to output mix (Compliance to OpenSL ES) 2166 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2167 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2168 desc = d; 2169 } 2170 } 2171 2172 // Do not allow auxiliary effects on a session different from 0 (output mix) 2173 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2174 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2175 lStatus = INVALID_OPERATION; 2176 goto Exit; 2177 } 2178 2179 // check recording permission for visualizer 2180 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2181 !recordingAllowed()) { 2182 lStatus = PERMISSION_DENIED; 2183 goto Exit; 2184 } 2185 2186 // return effect descriptor 2187 *pDesc = desc; 2188 if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2189 // if the output returned by getOutputForEffect() is removed before we lock the 2190 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2191 // and we will exit safely 2192 io = AudioSystem::getOutputForEffect(&desc); 2193 ALOGV("createEffect got output %d", io); 2194 } 2195 2196 Mutex::Autolock _l(mLock); 2197 2198 // If output is not specified try to find a matching audio session ID in one of the 2199 // output threads. 2200 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2201 // because of code checking output when entering the function. 2202 // Note: io is never 0 when creating an effect on an input 2203 if (io == 0) { 2204 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2205 // output must be specified by AudioPolicyManager when using session 2206 // AUDIO_SESSION_OUTPUT_STAGE 2207 lStatus = BAD_VALUE; 2208 goto Exit; 2209 } 2210 // look for the thread where the specified audio session is present 2211 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2212 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2213 io = mPlaybackThreads.keyAt(i); 2214 break; 2215 } 2216 } 2217 if (io == 0) { 2218 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2219 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2220 io = mRecordThreads.keyAt(i); 2221 break; 2222 } 2223 } 2224 } 2225 // If no output thread contains the requested session ID, default to 2226 // first output. The effect chain will be moved to the correct output 2227 // thread when a track with the same session ID is created 2228 if (io == 0 && mPlaybackThreads.size()) { 2229 io = mPlaybackThreads.keyAt(0); 2230 } 2231 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2232 } 2233 ThreadBase *thread = checkRecordThread_l(io); 2234 if (thread == NULL) { 2235 thread = checkPlaybackThread_l(io); 2236 if (thread == NULL) { 2237 ALOGE("createEffect() unknown output thread"); 2238 lStatus = BAD_VALUE; 2239 goto Exit; 2240 } 2241 } 2242 2243 sp<Client> client = registerPid_l(pid); 2244 2245 // create effect on selected output thread 2246 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2247 &desc, enabled, &lStatus); 2248 if (handle != 0 && id != NULL) { 2249 *id = handle->id(); 2250 } 2251 } 2252 2253Exit: 2254 *status = lStatus; 2255 return handle; 2256} 2257 2258status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2259 audio_io_handle_t dstOutput) 2260{ 2261 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2262 sessionId, srcOutput, dstOutput); 2263 Mutex::Autolock _l(mLock); 2264 if (srcOutput == dstOutput) { 2265 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2266 return NO_ERROR; 2267 } 2268 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2269 if (srcThread == NULL) { 2270 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2271 return BAD_VALUE; 2272 } 2273 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2274 if (dstThread == NULL) { 2275 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2276 return BAD_VALUE; 2277 } 2278 2279 Mutex::Autolock _dl(dstThread->mLock); 2280 Mutex::Autolock _sl(srcThread->mLock); 2281 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2282} 2283 2284// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2285status_t AudioFlinger::moveEffectChain_l(int sessionId, 2286 AudioFlinger::PlaybackThread *srcThread, 2287 AudioFlinger::PlaybackThread *dstThread, 2288 bool reRegister) 2289{ 2290 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2291 sessionId, srcThread, dstThread); 2292 2293 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2294 if (chain == 0) { 2295 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2296 sessionId, srcThread); 2297 return INVALID_OPERATION; 2298 } 2299 2300 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2301 // so that a new chain is created with correct parameters when first effect is added. This is 2302 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2303 // removed. 2304 srcThread->removeEffectChain_l(chain); 2305 2306 // transfer all effects one by one so that new effect chain is created on new thread with 2307 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2308 sp<EffectChain> dstChain; 2309 uint32_t strategy = 0; // prevent compiler warning 2310 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2311 Vector< sp<EffectModule> > removed; 2312 status_t status = NO_ERROR; 2313 while (effect != 0) { 2314 srcThread->removeEffect_l(effect); 2315 removed.add(effect); 2316 status = dstThread->addEffect_l(effect); 2317 if (status != NO_ERROR) { 2318 break; 2319 } 2320 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2321 if (effect->state() == EffectModule::ACTIVE || 2322 effect->state() == EffectModule::STOPPING) { 2323 effect->start(); 2324 } 2325 // if the move request is not received from audio policy manager, the effect must be 2326 // re-registered with the new strategy and output 2327 if (dstChain == 0) { 2328 dstChain = effect->chain().promote(); 2329 if (dstChain == 0) { 2330 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2331 status = NO_INIT; 2332 break; 2333 } 2334 strategy = dstChain->strategy(); 2335 } 2336 if (reRegister) { 2337 AudioSystem::unregisterEffect(effect->id()); 2338 AudioSystem::registerEffect(&effect->desc(), 2339 dstThread->id(), 2340 strategy, 2341 sessionId, 2342 effect->id()); 2343 } 2344 effect = chain->getEffectFromId_l(0); 2345 } 2346 2347 if (status != NO_ERROR) { 2348 for (size_t i = 0; i < removed.size(); i++) { 2349 srcThread->addEffect_l(removed[i]); 2350 if (dstChain != 0 && reRegister) { 2351 AudioSystem::unregisterEffect(removed[i]->id()); 2352 AudioSystem::registerEffect(&removed[i]->desc(), 2353 srcThread->id(), 2354 strategy, 2355 sessionId, 2356 removed[i]->id()); 2357 } 2358 } 2359 } 2360 2361 return status; 2362} 2363 2364bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2365{ 2366 if (mGlobalEffectEnableTime != 0 && 2367 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2368 return true; 2369 } 2370 2371 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2372 sp<EffectChain> ec = 2373 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2374 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2375 return true; 2376 } 2377 } 2378 return false; 2379} 2380 2381void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2382{ 2383 Mutex::Autolock _l(mLock); 2384 2385 mGlobalEffectEnableTime = systemTime(); 2386 2387 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2388 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2389 if (t->mType == ThreadBase::OFFLOAD) { 2390 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2391 } 2392 } 2393 2394} 2395 2396struct Entry { 2397#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2398 char mName[MAX_NAME]; 2399}; 2400 2401int comparEntry(const void *p1, const void *p2) 2402{ 2403 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2404} 2405 2406#ifdef TEE_SINK 2407void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2408{ 2409 NBAIO_Source *teeSource = source.get(); 2410 if (teeSource != NULL) { 2411 // .wav rotation 2412 // There is a benign race condition if 2 threads call this simultaneously. 2413 // They would both traverse the directory, but the result would simply be 2414 // failures at unlink() which are ignored. It's also unlikely since 2415 // normally dumpsys is only done by bugreport or from the command line. 2416 char teePath[32+256]; 2417 strcpy(teePath, "/data/misc/media"); 2418 size_t teePathLen = strlen(teePath); 2419 DIR *dir = opendir(teePath); 2420 teePath[teePathLen++] = '/'; 2421 if (dir != NULL) { 2422#define MAX_SORT 20 // number of entries to sort 2423#define MAX_KEEP 10 // number of entries to keep 2424 struct Entry entries[MAX_SORT]; 2425 size_t entryCount = 0; 2426 while (entryCount < MAX_SORT) { 2427 struct dirent de; 2428 struct dirent *result = NULL; 2429 int rc = readdir_r(dir, &de, &result); 2430 if (rc != 0) { 2431 ALOGW("readdir_r failed %d", rc); 2432 break; 2433 } 2434 if (result == NULL) { 2435 break; 2436 } 2437 if (result != &de) { 2438 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2439 break; 2440 } 2441 // ignore non .wav file entries 2442 size_t nameLen = strlen(de.d_name); 2443 if (nameLen <= 4 || nameLen >= MAX_NAME || 2444 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2445 continue; 2446 } 2447 strcpy(entries[entryCount++].mName, de.d_name); 2448 } 2449 (void) closedir(dir); 2450 if (entryCount > MAX_KEEP) { 2451 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2452 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2453 strcpy(&teePath[teePathLen], entries[i].mName); 2454 (void) unlink(teePath); 2455 } 2456 } 2457 } else { 2458 if (fd >= 0) { 2459 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2460 } 2461 } 2462 char teeTime[16]; 2463 struct timeval tv; 2464 gettimeofday(&tv, NULL); 2465 struct tm tm; 2466 localtime_r(&tv.tv_sec, &tm); 2467 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2468 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2469 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2470 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2471 if (teeFd >= 0) { 2472 char wavHeader[44]; 2473 memcpy(wavHeader, 2474 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 2475 sizeof(wavHeader)); 2476 NBAIO_Format format = teeSource->format(); 2477 unsigned channelCount = Format_channelCount(format); 2478 ALOG_ASSERT(channelCount <= FCC_2); 2479 uint32_t sampleRate = Format_sampleRate(format); 2480 wavHeader[22] = channelCount; // number of channels 2481 wavHeader[24] = sampleRate; // sample rate 2482 wavHeader[25] = sampleRate >> 8; 2483 wavHeader[32] = channelCount * 2; // block alignment 2484 write(teeFd, wavHeader, sizeof(wavHeader)); 2485 size_t total = 0; 2486 bool firstRead = true; 2487 for (;;) { 2488#define TEE_SINK_READ 1024 2489 short buffer[TEE_SINK_READ * FCC_2]; 2490 size_t count = TEE_SINK_READ; 2491 ssize_t actual = teeSource->read(buffer, count, 2492 AudioBufferProvider::kInvalidPTS); 2493 bool wasFirstRead = firstRead; 2494 firstRead = false; 2495 if (actual <= 0) { 2496 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2497 continue; 2498 } 2499 break; 2500 } 2501 ALOG_ASSERT(actual <= (ssize_t)count); 2502 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2503 total += actual; 2504 } 2505 lseek(teeFd, (off_t) 4, SEEK_SET); 2506 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2507 write(teeFd, &temp, sizeof(temp)); 2508 lseek(teeFd, (off_t) 40, SEEK_SET); 2509 temp = total * channelCount * sizeof(short); 2510 write(teeFd, &temp, sizeof(temp)); 2511 close(teeFd); 2512 if (fd >= 0) { 2513 fdprintf(fd, "tee copied to %s\n", teePath); 2514 } 2515 } else { 2516 if (fd >= 0) { 2517 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2518 } 2519 } 2520 } 2521} 2522#endif 2523 2524// ---------------------------------------------------------------------------- 2525 2526status_t AudioFlinger::onTransact( 2527 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2528{ 2529 return BnAudioFlinger::onTransact(code, data, reply, flags); 2530} 2531 2532}; // namespace android 2533