/* ** ** Copyright 2007, The Android Open Source Project ** ** Licensed under the Apache License, Version 2.0 (the "License"); ** you may not use this file except in compliance with the License. ** You may obtain a copy of the License at ** ** http://www.apache.org/licenses/LICENSE-2.0 ** ** Unless required by applicable law or agreed to in writing, software ** distributed under the License is distributed on an "AS IS" BASIS, ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. ** See the License for the specific language governing permissions and ** limitations under the License. ** ** This file was modified by DTS, Inc. The portions of the ** code that are surrounded by "DTS..." are copyrighted and ** licensed separately, as follows: ** ** (C) 2015 DTS, Inc. ** ** Licensed under the Apache License, Version 2.0 (the "License"); ** you may not use this file except in compliance with the License. ** You may obtain a copy of the License at ** ** http://www.apache.org/licenses/LICENSE-2.0 ** ** Unless required by applicable law or agreed to in writing, software ** distributed under the License is distributed on an "AS IS" BASIS, ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. ** See the License for the specific language governing permissions and ** limitations under the License. ** */ #define LOG_TAG "AudioFlinger" //#define LOG_NDEBUG 0 #include "Configuration.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "AudioMixer.h" #include "AudioFlinger.h" #include "ServiceUtilities.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #ifdef SRS_PROCESSING #include "postpro_patch.h" #endif // ---------------------------------------------------------------------------- // Note: the following macro is used for extremely verbose logging message. In // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to // 0; but one side effect of this is to turn all LOGV's as well. Some messages // are so verbose that we want to suppress them even when we have ALOG_ASSERT // turned on. Do not uncomment the #def below unless you really know what you // are doing and want to see all of the extremely verbose messages. //#define VERY_VERY_VERBOSE_LOGGING #ifdef VERY_VERY_VERBOSE_LOGGING #define ALOGVV ALOGV #else #define ALOGVV(a...) do { } while(0) #endif namespace android { static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; static const char kHardwareLockedString[] = "Hardware lock is taken\n"; static const char kClientLockedString[] = "Client lock is taken\n"; nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; uint32_t AudioFlinger::mScreenState; #ifdef TEE_SINK bool AudioFlinger::mTeeSinkInputEnabled = false; bool AudioFlinger::mTeeSinkOutputEnabled = false; bool AudioFlinger::mTeeSinkTrackEnabled = false; size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; #endif // In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off // we define a minimum time during which a global effect is considered enabled. static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); // ---------------------------------------------------------------------------- const char *formatToString(audio_format_t format) { switch (format & AUDIO_FORMAT_MAIN_MASK) { case AUDIO_FORMAT_PCM: switch (format) { case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; default: break; } break; case AUDIO_FORMAT_MP3: return "mp3"; case AUDIO_FORMAT_AMR_NB: return "amr-nb"; case AUDIO_FORMAT_AMR_WB: return "amr-wb"; case AUDIO_FORMAT_AAC: return "aac"; case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; case AUDIO_FORMAT_VORBIS: return "vorbis"; case AUDIO_FORMAT_OPUS: return "opus"; case AUDIO_FORMAT_AC3: return "ac-3"; case AUDIO_FORMAT_E_AC3: return "e-ac-3"; case AUDIO_FORMAT_PCM_OFFLOAD: switch (format) { case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD: return "pcm-16bit-offload"; case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD: return "pcm-24bit-offload"; default: break; } break; default: break; } return "unknown"; } static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) { const hw_module_t *mod; int rc; rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); if (rc) { goto out; } rc = audio_hw_device_open(mod, dev); ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); if (rc) { goto out; } if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); rc = BAD_VALUE; goto out; } return 0; out: *dev = NULL; return rc; } // ---------------------------------------------------------------------------- AudioFlinger::AudioFlinger() : BnAudioFlinger(), mPrimaryHardwareDev(NULL), mAudioHwDevs(NULL), mHardwareStatus(AUDIO_HW_IDLE), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), mMode(AUDIO_MODE_INVALID), mBtNrecIsOff(false), mIsLowRamDevice(true), mIsDeviceTypeKnown(false), mGlobalEffectEnableTime(0), mSystemReady(false) { getpid_cached = getpid(); char value[PROPERTY_VALUE_MAX]; bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); if (doLog) { mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); } #ifdef TEE_SINK (void) property_get("ro.debuggable", value, "0"); int debuggable = atoi(value); int teeEnabled = 0; if (debuggable) { (void) property_get("af.tee", value, "0"); teeEnabled = atoi(value); } // FIXME symbolic constants here if (teeEnabled & 1) { mTeeSinkInputEnabled = true; } if (teeEnabled & 2) { mTeeSinkOutputEnabled = true; } if (teeEnabled & 4) { mTeeSinkTrackEnabled = true; } #endif } void AudioFlinger::onFirstRef() { int rc = 0; Mutex::Autolock _l(mLock); /* TODO: move all this work into an Init() function */ char val_str[PROPERTY_VALUE_MAX] = { 0 }; if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { uint32_t int_val; if (1 == sscanf(val_str, "%u", &int_val)) { mStandbyTimeInNsecs = milliseconds(int_val); ALOGI("Using %u mSec as standby time.", int_val); } else { mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; ALOGI("Using default %u mSec as standby time.", (uint32_t)(mStandbyTimeInNsecs / 1000000)); } } mPatchPanel = new PatchPanel(this); mMode = AUDIO_MODE_NORMAL; } AudioFlinger::~AudioFlinger() { while (!mRecordThreads.isEmpty()) { // closeInput_nonvirtual() will remove specified entry from mRecordThreads closeInput_nonvirtual(mRecordThreads.keyAt(0)); } while (!mPlaybackThreads.isEmpty()) { // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); } for (size_t i = 0; i < mAudioHwDevs.size(); i++) { // no mHardwareLock needed, as there are no other references to this audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); delete mAudioHwDevs.valueAt(i); } // Tell media.log service about any old writers that still need to be unregistered sp binder = defaultServiceManager()->getService(String16("media.log")); if (binder != 0) { sp mediaLogService(interface_cast(binder)); for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { sp iMemory(mUnregisteredWriters.top()->getIMemory()); mUnregisteredWriters.pop(); mediaLogService->unregisterWriter(iMemory); } } } static const char * const audio_interfaces[] = { AUDIO_HARDWARE_MODULE_ID_PRIMARY, AUDIO_HARDWARE_MODULE_ID_A2DP, AUDIO_HARDWARE_MODULE_ID_USB, }; #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) AudioHwDevice* AudioFlinger::findSuitableHwDev_l( audio_module_handle_t module, audio_devices_t devices) { // if module is 0, the request comes from an old policy manager and we should load // well known modules if (module == 0) { ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { loadHwModule_l(audio_interfaces[i]); } // then try to find a module supporting the requested device. for (size_t i = 0; i < mAudioHwDevs.size(); i++) { AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); audio_hw_device_t *dev = audioHwDevice->hwDevice(); if ((dev->get_supported_devices != NULL) && (dev->get_supported_devices(dev) & devices) == devices) return audioHwDevice; } } else { // check a match for the requested module handle AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); if (audioHwDevice != NULL) { return audioHwDevice; } } return NULL; } void AudioFlinger::dumpClients(int fd, const Vector& args __unused) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; result.append("Clients:\n"); for (size_t i = 0; i < mClients.size(); ++i) { sp client = mClients.valueAt(i).promote(); if (client != 0) { snprintf(buffer, SIZE, " pid: %d\n", client->pid()); result.append(buffer); } } result.append("Notification Clients:\n"); for (size_t i = 0; i < mNotificationClients.size(); ++i) { snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); result.append(buffer); } result.append("Global session refs:\n"); result.append(" session pid count\n"); for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { AudioSessionRef *r = mAudioSessionRefs[i]; snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); result.append(buffer); } write(fd, result.string(), result.size()); } void AudioFlinger::dumpInternals(int fd, const Vector& args __unused) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; hardware_call_state hardwareStatus = mHardwareStatus; snprintf(buffer, SIZE, "Hardware status: %d\n" "Standby Time mSec: %u\n", hardwareStatus, (uint32_t)(mStandbyTimeInNsecs / 1000000)); result.append(buffer); write(fd, result.string(), result.size()); } void AudioFlinger::dumpPermissionDenial(int fd, const Vector& args __unused) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Permission Denial: " "can't dump AudioFlinger from pid=%d, uid=%d\n", IPCThreadState::self()->getCallingPid(), IPCThreadState::self()->getCallingUid()); result.append(buffer); write(fd, result.string(), result.size()); } bool AudioFlinger::dumpTryLock(Mutex& mutex) { bool locked = false; for (int i = 0; i < kDumpLockRetries; ++i) { if (mutex.tryLock() == NO_ERROR) { locked = true; break; } usleep(kDumpLockSleepUs); } return locked; } status_t AudioFlinger::dump(int fd, const Vector& args) { if (!dumpAllowed()) { dumpPermissionDenial(fd, args); } else { // get state of hardware lock bool hardwareLocked = dumpTryLock(mHardwareLock); if (!hardwareLocked) { String8 result(kHardwareLockedString); write(fd, result.string(), result.size()); } else { mHardwareLock.unlock(); } bool locked = dumpTryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { String8 result(kDeadlockedString); write(fd, result.string(), result.size()); } bool clientLocked = dumpTryLock(mClientLock); if (!clientLocked) { String8 result(kClientLockedString); write(fd, result.string(), result.size()); } EffectDumpEffects(fd); dumpClients(fd, args); if (clientLocked) { mClientLock.unlock(); } dumpInternals(fd, args); // dump playback threads for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->dump(fd, args); } // dump record threads for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->dump(fd, args); } // dump orphan effect chains if (mOrphanEffectChains.size() != 0) { write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { mOrphanEffectChains.valueAt(i)->dump(fd, args); } } // dump all hardware devs for (size_t i = 0; i < mAudioHwDevs.size(); i++) { audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); dev->dump(dev, fd); } #ifdef TEE_SINK // dump the serially shared record tee sink if (mRecordTeeSource != 0) { dumpTee(fd, mRecordTeeSource); } #endif if (locked) { mLock.unlock(); } // append a copy of media.log here by forwarding fd to it, but don't attempt // to lookup the service if it's not running, as it will block for a second if (mLogMemoryDealer != 0) { sp binder = defaultServiceManager()->getService(String16("media.log")); if (binder != 0) { dprintf(fd, "\nmedia.log:\n"); Vector args; binder->dump(fd, args); } } } return NO_ERROR; } sp AudioFlinger::registerPid(pid_t pid) { Mutex::Autolock _cl(mClientLock); // If pid is already in the mClients wp<> map, then use that entry // (for which promote() is always != 0), otherwise create a new entry and Client. sp client = mClients.valueFor(pid).promote(); if (client == 0) { client = new Client(this, pid); mClients.add(pid, client); } return client; } sp AudioFlinger::newWriter_l(size_t size, const char *name) { // If there is no memory allocated for logs, return a dummy writer that does nothing if (mLogMemoryDealer == 0) { return new NBLog::Writer(); } sp binder = defaultServiceManager()->getService(String16("media.log")); // Similarly if we can't contact the media.log service, also return a dummy writer if (binder == 0) { return new NBLog::Writer(); } sp mediaLogService(interface_cast(binder)); sp shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); // If allocation fails, consult the vector of previously unregistered writers // and garbage-collect one or more them until an allocation succeeds if (shared == 0) { Mutex::Autolock _l(mUnregisteredWritersLock); for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { { // Pick the oldest stale writer to garbage-collect sp iMemory(mUnregisteredWriters[0]->getIMemory()); mUnregisteredWriters.removeAt(0); mediaLogService->unregisterWriter(iMemory); // Now the media.log remote reference to IMemory is gone. When our last local // reference to IMemory also drops to zero at end of this block, // the IMemory destructor will deallocate the region from mLogMemoryDealer. } // Re-attempt the allocation shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); if (shared != 0) { goto success; } } // Even after garbage-collecting all old writers, there is still not enough memory, // so return a dummy writer return new NBLog::Writer(); } success: mediaLogService->registerWriter(shared, size, name); return new NBLog::Writer(size, shared); } void AudioFlinger::unregisterWriter(const sp& writer) { if (writer == 0) { return; } sp iMemory(writer->getIMemory()); if (iMemory == 0) { return; } // Rather than removing the writer immediately, append it to a queue of old writers to // be garbage-collected later. This allows us to continue to view old logs for a while. Mutex::Autolock _l(mUnregisteredWritersLock); mUnregisteredWriters.push(writer); } // IAudioFlinger interface sp AudioFlinger::createTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *frameCount, IAudioFlinger::track_flags_t *flags, const sp& sharedBuffer, audio_io_handle_t output, pid_t tid, int *sessionId, int clientUid, status_t *status) { sp track; sp trackHandle; sp client; status_t lStatus; int lSessionId; // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, // but if someone uses binder directly they could bypass that and cause us to crash if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { ALOGE("createTrack() invalid stream type %d", streamType); lStatus = BAD_VALUE; goto Exit; } // further sample rate checks are performed by createTrack_l() depending on the thread type if (sampleRate == 0) { ALOGE("createTrack() invalid sample rate %u", sampleRate); lStatus = BAD_VALUE; goto Exit; } // further channel mask checks are performed by createTrack_l() depending on the thread type if (!audio_is_output_channel(channelMask)) { ALOGE("createTrack() invalid channel mask %#x", channelMask); lStatus = BAD_VALUE; goto Exit; } // further format checks are performed by createTrack_l() depending on the thread type if (!audio_is_valid_format(format)) { ALOGE("createTrack() invalid format %#x", format); lStatus = BAD_VALUE; goto Exit; } if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); lStatus = BAD_VALUE; goto Exit; } { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGE("no playback thread found for output handle %d", output); lStatus = BAD_VALUE; goto Exit; } pid_t pid = IPCThreadState::self()->getCallingPid(); client = registerPid(pid); PlaybackThread *effectThread = NULL; if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { lSessionId = *sessionId; // check if an effect chain with the same session ID is present on another // output thread and move it here. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); if (mPlaybackThreads.keyAt(i) != output) { uint32_t sessions = t->hasAudioSession(lSessionId); if (sessions & PlaybackThread::EFFECT_SESSION) { effectThread = t.get(); break; } } } } else { // if no audio session id is provided, create one here lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } ALOGV("createTrack() lSessionId: %d", lSessionId); track = thread->createTrack_l(client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless // move effect chain to this output thread if an effect on same session was waiting // for a track to be created if (lStatus == NO_ERROR && effectThread != NULL) { // no risk of deadlock because AudioFlinger::mLock is held Mutex::Autolock _dl(thread->mLock); Mutex::Autolock _sl(effectThread->mLock); moveEffectChain_l(lSessionId, effectThread, thread, true); } // Look for sync events awaiting for a session to be used. for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { if (lStatus == NO_ERROR) { (void) track->setSyncEvent(mPendingSyncEvents[i]); } else { mPendingSyncEvents[i]->cancel(); } mPendingSyncEvents.removeAt(i); i--; } } } setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); } if (lStatus != NO_ERROR) { // remove local strong reference to Client before deleting the Track so that the // Client destructor is called by the TrackBase destructor with mClientLock held // Don't hold mClientLock when releasing the reference on the track as the // destructor will acquire it. { Mutex::Autolock _cl(mClientLock); client.clear(); } track.clear(); goto Exit; } // return handle to client trackHandle = new TrackHandle(track); Exit: *status = lStatus; return trackHandle; } uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("sampleRate() unknown thread %d", output); return 0; } return thread->sampleRate(); } audio_format_t AudioFlinger::format(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("format() unknown thread %d", output); return AUDIO_FORMAT_INVALID; } return thread->format(); } size_t AudioFlinger::frameCount(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("frameCount() unknown thread %d", output); return 0; } // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; // should examine all callers and fix them to handle smaller counts return thread->frameCount(); } uint32_t AudioFlinger::latency(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("latency(): no playback thread found for output handle %d", output); return 0; } return thread->latency(); } status_t AudioFlinger::setMasterVolume(float value) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } Mutex::Autolock _l(mLock); mMasterVolume = value; // Set master volume in the HALs which support it. for (size_t i = 0; i < mAudioHwDevs.size(); i++) { AutoMutex lock(mHardwareLock); AudioHwDevice *dev = mAudioHwDevs.valueAt(i); mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; if (dev->canSetMasterVolume()) { dev->hwDevice()->set_master_volume(dev->hwDevice(), value); } mHardwareStatus = AUDIO_HW_IDLE; } // Now set the master volume in each playback thread. Playback threads // assigned to HALs which do not have master volume support will apply // master volume during the mix operation. Threads with HALs which do // support master volume will simply ignore the setting. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isDuplicating()) { continue; } mPlaybackThreads.valueAt(i)->setMasterVolume(value); } return NO_ERROR; } status_t AudioFlinger::setMode(audio_mode_t mode) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(mode) >= AUDIO_MODE_CNT) { ALOGW("Illegal value: setMode(%d)", mode); return BAD_VALUE; } { // scope for the lock AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_HW_SET_MODE; ret = dev->set_mode(dev, mode); mHardwareStatus = AUDIO_HW_IDLE; } if (NO_ERROR == ret) { Mutex::Autolock _l(mLock); mMode = mode; for (size_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setMode(mode); } return ret; } status_t AudioFlinger::setMicMute(bool state) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; for (size_t i = 0; i < mAudioHwDevs.size(); i++) { audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); status_t result = dev->set_mic_mute(dev, state); if (result != NO_ERROR) { ret = result; } } mHardwareStatus = AUDIO_HW_IDLE; return ret; } bool AudioFlinger::getMicMute() const { status_t ret = initCheck(); if (ret != NO_ERROR) { return false; } bool mute = true; bool state = AUDIO_MODE_INVALID; AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; for (size_t i = 0; i < mAudioHwDevs.size(); i++) { audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); status_t result = dev->get_mic_mute(dev, &state); if (result == NO_ERROR) { mute = mute && state; } } mHardwareStatus = AUDIO_HW_IDLE; return mute; } status_t AudioFlinger::setMasterMute(bool muted) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } Mutex::Autolock _l(mLock); mMasterMute = muted; // Set master mute in the HALs which support it. for (size_t i = 0; i < mAudioHwDevs.size(); i++) { AutoMutex lock(mHardwareLock); AudioHwDevice *dev = mAudioHwDevs.valueAt(i); mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; if (dev->canSetMasterMute()) { dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); } mHardwareStatus = AUDIO_HW_IDLE; } // Now set the master mute in each playback thread. Playback threads // assigned to HALs which do not have master mute support will apply master // mute during the mix operation. Threads with HALs which do support master // mute will simply ignore the setting. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isDuplicating()) { continue; } mPlaybackThreads.valueAt(i)->setMasterMute(muted); } return NO_ERROR; } float AudioFlinger::masterVolume() const { Mutex::Autolock _l(mLock); return masterVolume_l(); } bool AudioFlinger::masterMute() const { Mutex::Autolock _l(mLock); return masterMute_l(); } float AudioFlinger::masterVolume_l() const { return mMasterVolume; } bool AudioFlinger::masterMute_l() const { return mMasterMute; } status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const { if (uint32_t(stream) >= AUDIO_STREAM_CNT) { ALOGW("setStreamVolume() invalid stream %d", stream); return BAD_VALUE; } pid_t caller = IPCThreadState::self()->getCallingPid(); if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); return PERMISSION_DENIED; } return NO_ERROR; } status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, audio_io_handle_t output) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } status_t status = checkStreamType(stream); if (status != NO_ERROR) { return status; } ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); AutoMutex lock(mLock); PlaybackThread *thread = NULL; if (output != AUDIO_IO_HANDLE_NONE) { thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } } mStreamTypes[stream].volume = value; if (thread == NULL) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); } } else { thread->setStreamVolume(stream, value); } return NO_ERROR; } status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } status_t status = checkStreamType(stream); if (status != NO_ERROR) { return status; } ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { ALOGE("setStreamMute() invalid stream %d", stream); return BAD_VALUE; } AutoMutex lock(mLock); mStreamTypes[stream].mute = muted; for (size_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); return NO_ERROR; } float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const { status_t status = checkStreamType(stream); if (status != NO_ERROR) { return 0.0f; } AutoMutex lock(mLock); float volume; if (output != AUDIO_IO_HANDLE_NONE) { PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return 0.0f; } volume = thread->streamVolume(stream); } else { volume = streamVolume_l(stream); } return volume; } bool AudioFlinger::streamMute(audio_stream_type_t stream) const { status_t status = checkStreamType(stream); if (status != NO_ERROR) { return true; } AutoMutex lock(mLock); return streamMute_l(stream); } void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) { for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->setParameters(keyValuePairs); } } status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) { ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface if (ioHandle == AUDIO_IO_HANDLE_NONE) { Mutex::Autolock _l(mLock); status_t final_result = NO_ERROR; #ifdef SRS_PROCESSING POSTPRO_PATCH_PARAMS_SET(keyValuePairs); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); thread->setPostPro(); } #endif { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_PARAMETER; for (size_t i = 0; i < mAudioHwDevs.size(); i++) { audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); status_t result = dev->set_parameters(dev, keyValuePairs.string()); final_result = result ?: final_result; } mHardwareStatus = AUDIO_HW_IDLE; } AudioParameter param = AudioParameter(keyValuePairs); String8 value, key; key = String8("SND_CARD_STATUS"); if (param.get(key, value) == NO_ERROR) { ALOGV("Set keySoundCardStatus:%s", value.string()); if ((value.find("OFFLINE", 0) != -1) ) { ALOGV("OFFLINE detected - call InvalidateTracks()"); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); if( thread->getOutput()->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ){ thread->invalidateTracks(AUDIO_STREAM_MUSIC); } } } } // disable AEC and NS if the device is a BT SCO headset supporting those pre processings if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); if (mBtNrecIsOff != btNrecIsOff) { for (size_t i = 0; i < mRecordThreads.size(); i++) { sp thread = mRecordThreads.valueAt(i); audio_devices_t device = thread->inDevice(); bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; // collect all of the thread's session IDs KeyedVector ids = thread->sessionIds(); // suspend effects associated with those session IDs for (size_t j = 0; j < ids.size(); ++j) { int sessionId = ids.keyAt(j); thread->setEffectSuspended(FX_IID_AEC, suspend, sessionId); thread->setEffectSuspended(FX_IID_NS, suspend, sessionId); } } mBtNrecIsOff = btNrecIsOff; } } String8 screenState; if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { bool isOff = screenState == "off"; if (isOff != (AudioFlinger::mScreenState & 1)) { AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; } } return final_result; } // hold a strong ref on thread in case closeOutput() or closeInput() is called // and the thread is exited once the lock is released sp thread; { Mutex::Autolock _l(mLock); thread = checkPlaybackThread_l(ioHandle); if (thread == 0) { thread = checkRecordThread_l(ioHandle); } else if (thread == primaryPlaybackThread_l()) { // indicate output device change to all input threads for pre processing AudioParameter param = AudioParameter(keyValuePairs); int value; if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && (value != 0)) { broacastParametersToRecordThreads_l(keyValuePairs); } } } if (thread != 0) { return thread->setParameters(keyValuePairs); } return BAD_VALUE; } String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const { ALOGVV("getParameters() io %d, keys %s, calling pid %d", ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); if (ioHandle == AUDIO_IO_HANDLE_NONE) { String8 out_s8; #ifdef SRS_PROCESSING POSTPRO_PATCH_PARAMS_GET(keys, out_s8); #endif for (size_t i = 0; i < mAudioHwDevs.size(); i++) { char *s; { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_PARAMETER; audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); s = dev->get_parameters(dev, keys.string()); mHardwareStatus = AUDIO_HW_IDLE; } out_s8 += String8(s ? s : ""); free(s); } return out_s8; } PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); if (playbackThread != NULL) { return playbackThread->getParameters(keys); } RecordThread *recordThread = checkRecordThread_l(ioHandle); if (recordThread != NULL) { return recordThread->getParameters(keys); } return String8(""); } size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) const { status_t ret = initCheck(); if (ret != NO_ERROR) { return 0; } if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { return 0; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; audio_config_t config, proposed; memset(&proposed, 0, sizeof(proposed)); proposed.sample_rate = sampleRate; proposed.channel_mask = channelMask; proposed.format = format; audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); size_t frames; for (;;) { // Note: config is currently a const parameter for get_input_buffer_size() // but we use a copy from proposed in case config changes from the call. config = proposed; frames = dev->get_input_buffer_size(dev, &config); if (frames != 0) { break; // hal success, config is the result } // change one parameter of the configuration each iteration to a more "common" value // to see if the device will support it. if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { proposed.format = AUDIO_FORMAT_PCM_16_BIT; } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? } else { ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " "format %#x, channelMask 0x%X", sampleRate, format, channelMask); break; // retries failed, break out of loop with frames == 0. } } mHardwareStatus = AUDIO_HW_IDLE; if (frames > 0 && config.sample_rate != sampleRate) { frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); } return frames; // may be converted to bytes at the Java level. } uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const { Mutex::Autolock _l(mLock); RecordThread *recordThread = checkRecordThread_l(ioHandle); if (recordThread != NULL) { return recordThread->getInputFramesLost(); } return 0; } status_t AudioFlinger::setVoiceVolume(float value) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; ret = dev->set_voice_volume(dev, value); mHardwareStatus = AUDIO_HW_IDLE; return ret; } status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_io_handle_t output) const { status_t status; Mutex::Autolock _l(mLock); PlaybackThread *playbackThread = checkPlaybackThread_l(output); if (playbackThread != NULL) { return playbackThread->getRenderPosition(halFrames, dspFrames); } return BAD_VALUE; } void AudioFlinger::registerClient(const sp& client) { Mutex::Autolock _l(mLock); if (client == 0) { return; } pid_t pid = IPCThreadState::self()->getCallingPid(); { Mutex::Autolock _cl(mClientLock); if (mNotificationClients.indexOfKey(pid) < 0) { sp notificationClient = new NotificationClient(this, client, pid); ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); mNotificationClients.add(pid, notificationClient); sp binder = IInterface::asBinder(client); binder->linkToDeath(notificationClient); } } // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. // the config change is always sent from playback or record threads to avoid deadlock // with AudioSystem::gLock for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); } for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); } } void AudioFlinger::removeNotificationClient(pid_t pid) { Mutex::Autolock _l(mLock); { Mutex::Autolock _cl(mClientLock); mNotificationClients.removeItem(pid); } ALOGV("%d died, releasing its sessions", pid); size_t num = mAudioSessionRefs.size(); bool removed = false; for (size_t i = 0; i< num; ) { AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); ALOGV(" pid %d @ %d", ref->mPid, i); if (ref->mPid == pid) { ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); mAudioSessionRefs.removeAt(i); delete ref; removed = true; num--; } else { i++; } } if (removed) { purgeStaleEffects_l(); } } void AudioFlinger::ioConfigChanged(audio_io_config_event event, const sp& ioDesc, pid_t pid) { Mutex::Autolock _l(mClientLock); size_t size = mNotificationClients.size(); for (size_t i = 0; i < size; i++) { if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); } } } // removeClient_l() must be called with AudioFlinger::mClientLock held void AudioFlinger::removeClient_l(pid_t pid) { ALOGV("removeClient_l() pid %d, calling pid %d", pid, IPCThreadState::self()->getCallingPid()); mClients.removeItem(pid); } // getEffectThread_l() must be called with AudioFlinger::mLock held sp AudioFlinger::getEffectThread_l(int sessionId, int EffectId) { sp thread; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { ALOG_ASSERT(thread == 0); thread = mPlaybackThreads.valueAt(i); } } return thread; } void AudioFlinger::PlaybackThread::setPostPro() { Mutex::Autolock _l(mLock); if (mType == OFFLOAD) broadcast_l(); } // ---------------------------------------------------------------------------- AudioFlinger::Client::Client(const sp& audioFlinger, pid_t pid) : RefBase(), mAudioFlinger(audioFlinger), mPid(pid), mTimedTrackCount(0) { size_t heapSize = kClientSharedHeapSizeBytes; // Increase heap size on non low ram devices to limit risk of reconnection failure for // invalidated tracks if (!audioFlinger->isLowRamDevice()) { heapSize *= kClientSharedHeapSizeMultiplier; } mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); } // Client destructor must be called with AudioFlinger::mClientLock held AudioFlinger::Client::~Client() { mAudioFlinger->removeClient_l(mPid); } sp AudioFlinger::Client::heap() const { return mMemoryDealer; } // Reserve one of the limited slots for a timed audio track associated // with this client bool AudioFlinger::Client::reserveTimedTrack() { const int kMaxTimedTracksPerClient = 4; Mutex::Autolock _l(mTimedTrackLock); if (mTimedTrackCount >= kMaxTimedTracksPerClient) { ALOGW("can not create timed track - pid %d has exceeded the limit", mPid); return false; } mTimedTrackCount++; return true; } // Release a slot for a timed audio track void AudioFlinger::Client::releaseTimedTrack() { Mutex::Autolock _l(mTimedTrackLock); mTimedTrackCount--; } // ---------------------------------------------------------------------------- AudioFlinger::NotificationClient::NotificationClient(const sp& audioFlinger, const sp& client, pid_t pid) : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) { } AudioFlinger::NotificationClient::~NotificationClient() { } void AudioFlinger::NotificationClient::binderDied(const wp& who __unused) { sp keep(this); mAudioFlinger->removeNotificationClient(mPid); } // ---------------------------------------------------------------------------- static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { return audio_is_remote_submix_device(inDevice); } sp AudioFlinger::openRecord( audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const String16& opPackageName, size_t *frameCount, IAudioFlinger::track_flags_t *flags, pid_t tid, int clientUid, int *sessionId, size_t *notificationFrames, sp& cblk, sp& buffers, status_t *status) { sp recordTrack; sp recordHandle; sp client; status_t lStatus; int lSessionId; cblk.clear(); buffers.clear(); // check calling permissions if (!recordingAllowed(opPackageName)) { ALOGE("openRecord() permission denied: recording not allowed"); lStatus = PERMISSION_DENIED; goto Exit; } // further sample rate checks are performed by createRecordTrack_l() if (sampleRate == 0) { ALOGE("openRecord() invalid sample rate %u", sampleRate); lStatus = BAD_VALUE; goto Exit; } // we don't yet support anything other than linear PCM if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { ALOGE("openRecord() invalid format %#x", format); lStatus = BAD_VALUE; goto Exit; } // further channel mask checks are performed by createRecordTrack_l() if (!audio_is_input_channel(channelMask)) { ALOGE("openRecord() invalid channel mask %#x", channelMask); lStatus = BAD_VALUE; goto Exit; } { Mutex::Autolock _l(mLock); RecordThread *thread = checkRecordThread_l(input); if (thread == NULL) { ALOGE("openRecord() checkRecordThread_l failed"); lStatus = BAD_VALUE; goto Exit; } pid_t pid = IPCThreadState::self()->getCallingPid(); client = registerPid(pid); if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { lSessionId = *sessionId; } else { // if no audio session id is provided, create one here lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); // TODO: the uid should be passed in as a parameter to openRecord recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, notificationFrames, clientUid, flags, tid, &lStatus); LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); if (lStatus == NO_ERROR) { // Check if one effect chain was awaiting for an AudioRecord to be created on this // session and move it to this thread. sp chain = getOrphanEffectChain_l((audio_session_t)lSessionId); if (chain != 0) { Mutex::Autolock _l(thread->mLock); thread->addEffectChain_l(chain); } } } if (lStatus != NO_ERROR) { // remove local strong reference to Client before deleting the RecordTrack so that the // Client destructor is called by the TrackBase destructor with mClientLock held // Don't hold mClientLock when releasing the reference on the track as the // destructor will acquire it. { Mutex::Autolock _cl(mClientLock); client.clear(); } recordTrack.clear(); goto Exit; } cblk = recordTrack->getCblk(); buffers = recordTrack->getBuffers(); // return handle to client recordHandle = new RecordHandle(recordTrack); Exit: *status = lStatus; return recordHandle; } // ---------------------------------------------------------------------------- audio_module_handle_t AudioFlinger::loadHwModule(const char *name) { if (name == NULL) { return 0; } if (!settingsAllowed()) { return 0; } Mutex::Autolock _l(mLock); return loadHwModule_l(name); } // loadHwModule_l() must be called with AudioFlinger::mLock held audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) { for (size_t i = 0; i < mAudioHwDevs.size(); i++) { if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { ALOGW("loadHwModule() module %s already loaded", name); return mAudioHwDevs.keyAt(i); } } audio_hw_device_t *dev; int rc = load_audio_interface(name, &dev); if (rc) { ALOGI("loadHwModule() error %d loading module %s ", rc, name); return 0; } mHardwareStatus = AUDIO_HW_INIT; rc = dev->init_check(dev); mHardwareStatus = AUDIO_HW_IDLE; if (rc) { ALOGI("loadHwModule() init check error %d for module %s ", rc, name); return 0; } // Check and cache this HAL's level of support for master mute and master // volume. If this is the first HAL opened, and it supports the get // methods, use the initial values provided by the HAL as the current // master mute and volume settings. AudioHwDevice::Flags flags = static_cast(0); { // scope for auto-lock pattern AutoMutex lock(mHardwareLock); if (0 == mAudioHwDevs.size()) { mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; if (NULL != dev->get_master_volume) { float mv; if (OK == dev->get_master_volume(dev, &mv)) { mMasterVolume = mv; } } mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; if (NULL != dev->get_master_mute) { bool mm; if (OK == dev->get_master_mute(dev, &mm)) { mMasterMute = mm; } } } mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; if ((NULL != dev->set_master_volume) && (OK == dev->set_master_volume(dev, mMasterVolume))) { flags = static_cast(flags | AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); } mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; if ((NULL != dev->set_master_mute) && (OK == dev->set_master_mute(dev, mMasterMute))) { flags = static_cast(flags | AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); } mHardwareStatus = AUDIO_HW_IDLE; } audio_module_handle_t handle = nextUniqueId(); mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", name, dev->common.module->name, dev->common.module->id, handle); return handle; } // ---------------------------------------------------------------------------- uint32_t AudioFlinger::getPrimaryOutputSamplingRate() { Mutex::Autolock _l(mLock); PlaybackThread *thread = primaryPlaybackThread_l(); return thread != NULL ? thread->sampleRate() : 0; } size_t AudioFlinger::getPrimaryOutputFrameCount() { Mutex::Autolock _l(mLock); PlaybackThread *thread = primaryPlaybackThread_l(); return thread != NULL ? thread->frameCountHAL() : 0; } // ---------------------------------------------------------------------------- status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) { uid_t uid = IPCThreadState::self()->getCallingUid(); if (uid != AID_SYSTEM) { return PERMISSION_DENIED; } Mutex::Autolock _l(mLock); if (mIsDeviceTypeKnown) { return INVALID_OPERATION; } mIsLowRamDevice = isLowRamDevice; mIsDeviceTypeKnown = true; return NO_ERROR; } audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) { Mutex::Autolock _l(mLock); ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); if (index >= 0) { ALOGV("getAudioHwSyncForSession found ID %d for session %d", mHwAvSyncIds.valueAt(index), sessionId); return mHwAvSyncIds.valueAt(index); } audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); if (dev == NULL) { return AUDIO_HW_SYNC_INVALID; } char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); AudioParameter param = AudioParameter(String8(reply)); free(reply); int value; if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); return AUDIO_HW_SYNC_INVALID; } // allow only one session for a given HW A/V sync ID. for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { ALOGV("getAudioHwSyncForSession removing ID %d for session %d", value, mHwAvSyncIds.keyAt(i)); mHwAvSyncIds.removeItemsAt(i); break; } } mHwAvSyncIds.add(sessionId, value); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp thread = mPlaybackThreads.valueAt(i); uint32_t sessions = thread->hasAudioSession(sessionId); if (sessions & PlaybackThread::TRACK_SESSION) { AudioParameter param = AudioParameter(); param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); thread->setParameters(param.toString()); break; } } ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); return (audio_hw_sync_t)value; } status_t AudioFlinger::systemReady() { Mutex::Autolock _l(mLock); ALOGI("%s", __FUNCTION__); if (mSystemReady) { ALOGW("%s called twice", __FUNCTION__); return NO_ERROR; } mSystemReady = true; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); thread->systemReady(); } for (size_t i = 0; i < mRecordThreads.size(); i++) { ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); thread->systemReady(); } return NO_ERROR; } // setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) { ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); if (index >= 0) { audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); AudioParameter param = AudioParameter(); param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); thread->setParameters(param.toString()); } } // ---------------------------------------------------------------------------- sp AudioFlinger::openOutput_l(audio_module_handle_t module, audio_io_handle_t *output, audio_config_t *config, audio_devices_t devices, const String8& address, audio_output_flags_t flags) { AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); if (outHwDev == NULL) { return 0; } audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); if (*output == AUDIO_IO_HANDLE_NONE) { *output = nextUniqueId(); } mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; // FOR TESTING ONLY: // This if statement allows overriding the audio policy settings // and forcing a specific format or channel mask to the HAL/Sink device for testing. if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { // Check only for Normal Mixing mode if (kEnableExtendedPrecision) { // Specify format (uncomment one below to choose) //config->format = AUDIO_FORMAT_PCM_FLOAT; //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; //config->format = AUDIO_FORMAT_PCM_32_BIT; //config->format = AUDIO_FORMAT_PCM_8_24_BIT; // ALOGV("openOutput_l() upgrading format to %#08x", config->format); } if (kEnableExtendedChannels) { // Specify channel mask (uncomment one below to choose) //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch //config->channel_mask = audio_channel_mask_from_representation_and_bits( // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example } } AudioStreamOut *outputStream = NULL; status_t status = outHwDev->openOutputStream( &outputStream, *output, devices, flags, config, address.string()); mHardwareStatus = AUDIO_HW_IDLE; if (status == NO_ERROR) { PlaybackThread *thread; if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || !isValidPcmSinkFormat(config->format) || !isValidPcmSinkChannelMask(config->channel_mask)) { thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); ALOGV("openOutput_l() created direct output: ID %d thread %p ", *output, thread); //Check if this is DirectPCM, if so if (flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) { thread->mIsDirectPcm = true; } } else { thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); } mPlaybackThreads.add(*output, thread); return thread; } return 0; } status_t AudioFlinger::openOutput(audio_module_handle_t module, audio_io_handle_t *output, audio_config_t *config, audio_devices_t *devices, const String8& address, uint32_t *latencyMs, audio_output_flags_t flags) { ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", module, (devices != NULL) ? *devices : 0, config->sample_rate, config->format, config->channel_mask, flags); if (*devices == AUDIO_DEVICE_NONE) { return BAD_VALUE; } Mutex::Autolock _l(mLock); sp thread = openOutput_l(module, output, config, *devices, address, flags); if (thread != 0) { *latencyMs = thread->latency(); // notify client processes of the new output creation thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); // the first primary output opened designates the primary hw device if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { ALOGI("Using module %d has the primary audio interface", module); mPrimaryHardwareDev = thread->getOutput()->audioHwDev; AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MODE; mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); mHardwareStatus = AUDIO_HW_IDLE; } return NO_ERROR; } return NO_INIT; } audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) { Mutex::Autolock _l(mLock); MixerThread *thread1 = checkMixerThread_l(output1); MixerThread *thread2 = checkMixerThread_l(output2); if (thread1 == NULL || thread2 == NULL) { ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); return AUDIO_IO_HANDLE_NONE; } audio_io_handle_t id = nextUniqueId(); DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); thread->addOutputTrack(thread2); mPlaybackThreads.add(id, thread); // notify client processes of the new output creation thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); return id; } status_t AudioFlinger::closeOutput(audio_io_handle_t output) { return closeOutput_nonvirtual(output); } status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) { // keep strong reference on the playback thread so that // it is not destroyed while exit() is executed sp thread; { Mutex::Autolock _l(mLock); thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } ALOGV("closeOutput() %d", output); if (thread->type() == ThreadBase::MIXER) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isDuplicating()) { DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); } } } mPlaybackThreads.removeItem(output); // save all effects to the default thread if (mPlaybackThreads.size()) { PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); if (dstThread != NULL) { // audioflinger lock is held here so the acquisition order of thread locks does not // matter Mutex::Autolock _dl(dstThread->mLock); Mutex::Autolock _sl(thread->mLock); Vector< sp > effectChains = thread->getEffectChains_l(); for (size_t i = 0; i < effectChains.size(); i ++) { moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); } } } const sp ioDesc = new AudioIoDescriptor(); ioDesc->mIoHandle = output; ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); } thread->exit(); // The thread entity (active unit of execution) is no longer running here, // but the ThreadBase container still exists. if (!thread->isDuplicating()) { closeOutputFinish(thread); } return NO_ERROR; } void AudioFlinger::closeOutputFinish(sp thread) { AudioStreamOut *out = thread->clearOutput(); ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); // from now on thread->mOutput is NULL out->hwDev()->close_output_stream(out->hwDev(), out->stream); delete out; } void AudioFlinger::closeOutputInternal_l(sp thread) { mPlaybackThreads.removeItem(thread->mId); thread->exit(); closeOutputFinish(thread); } status_t AudioFlinger::suspendOutput(audio_io_handle_t output) { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } ALOGV("suspendOutput() %d", output); thread->suspend(); return NO_ERROR; } status_t AudioFlinger::restoreOutput(audio_io_handle_t output) { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } ALOGV("restoreOutput() %d", output); thread->restore(); return NO_ERROR; } status_t AudioFlinger::openInput(audio_module_handle_t module, audio_io_handle_t *input, audio_config_t *config, audio_devices_t *devices, const String8& address, audio_source_t source, audio_input_flags_t flags) { Mutex::Autolock _l(mLock); if (*devices == AUDIO_DEVICE_NONE) { return BAD_VALUE; } sp thread = openInput_l(module, input, config, *devices, address, source, flags); if (thread != 0) { // notify client processes of the new input creation thread->ioConfigChanged(AUDIO_INPUT_OPENED); return NO_ERROR; } return NO_INIT; } sp AudioFlinger::openInput_l(audio_module_handle_t module, audio_io_handle_t *input, audio_config_t *config, audio_devices_t devices, const String8& address, audio_source_t source, audio_input_flags_t flags) { AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); if (inHwDev == NULL) { *input = AUDIO_IO_HANDLE_NONE; return 0; } if (*input == AUDIO_IO_HANDLE_NONE) { *input = nextUniqueId(); } audio_config_t halconfig = *config; audio_hw_device_t *inHwHal = inHwDev->hwDevice(); audio_stream_in_t *inStream = NULL; status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, &inStream, flags, address.string(), source); ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" ", Format %#x, Channels %x, flags %#x, status %d addr %s", inStream, halconfig.sample_rate, halconfig.format, halconfig.channel_mask, flags, status, address.string()); // If the input could not be opened with the requested parameters and we can handle the // conversion internally, try to open again with the proposed parameters. if (status == BAD_VALUE && audio_is_linear_pcm(config->format) && audio_is_linear_pcm(halconfig.format) && (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { // FIXME describe the change proposed by HAL (save old values so we can log them here) ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); inStream = NULL; status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, &inStream, flags, address.string(), source); // FIXME log this new status; HAL should not propose any further changes } if (status == NO_ERROR && inStream != NULL) { #ifdef TEE_SINK // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, // or (re-)create if current Pipe is idle and does not match the new format sp teeSink; enum { TEE_SINK_NO, // don't copy input TEE_SINK_NEW, // copy input using a new pipe TEE_SINK_OLD, // copy input using an existing pipe } kind; NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); if (!mTeeSinkInputEnabled) { kind = TEE_SINK_NO; } else if (!Format_isValid(format)) { kind = TEE_SINK_NO; } else if (mRecordTeeSink == 0) { kind = TEE_SINK_NEW; } else if (mRecordTeeSink->getStrongCount() != 1) { kind = TEE_SINK_NO; } else if (Format_isEqual(format, mRecordTeeSink->format())) { kind = TEE_SINK_OLD; } else { kind = TEE_SINK_NEW; } switch (kind) { case TEE_SINK_NEW: { Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); size_t numCounterOffers = 0; const NBAIO_Format offers[1] = {format}; ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); ALOG_ASSERT(index == 0); PipeReader *pipeReader = new PipeReader(*pipe); numCounterOffers = 0; index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); ALOG_ASSERT(index == 0); mRecordTeeSink = pipe; mRecordTeeSource = pipeReader; teeSink = pipe; } break; case TEE_SINK_OLD: teeSink = mRecordTeeSink; break; case TEE_SINK_NO: default: break; } #endif AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); // Start record thread // RecordThread requires both input and output device indication to forward to audio // pre processing modules sp thread = new RecordThread(this, inputStream, *input, primaryOutputDevice_l(), devices, mSystemReady #ifdef TEE_SINK , teeSink #endif ); mRecordThreads.add(*input, thread); ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); return thread; } *input = AUDIO_IO_HANDLE_NONE; return 0; } status_t AudioFlinger::closeInput(audio_io_handle_t input) { return closeInput_nonvirtual(input); } status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) { // keep strong reference on the record thread so that // it is not destroyed while exit() is executed sp thread; { Mutex::Autolock _l(mLock); thread = checkRecordThread_l(input); if (thread == 0) { return BAD_VALUE; } ALOGV("closeInput() %d", input); // If we still have effect chains, it means that a client still holds a handle // on at least one effect. We must either move the chain to an existing thread with the // same session ID or put it aside in case a new record thread is opened for a // new capture on the same session sp chain; { Mutex::Autolock _sl(thread->mLock); Vector< sp > effectChains = thread->getEffectChains_l(); // Note: maximum one chain per record thread if (effectChains.size() != 0) { chain = effectChains[0]; } } if (chain != 0) { // first check if a record thread is already opened with a client on the same session. // This should only happen in case of overlap between one thread tear down and the // creation of its replacement size_t i; for (i = 0; i < mRecordThreads.size(); i++) { sp t = mRecordThreads.valueAt(i); if (t == thread) { continue; } if (t->hasAudioSession(chain->sessionId()) != 0) { Mutex::Autolock _l(t->mLock); ALOGV("closeInput() found thread %d for effect session %d", t->id(), chain->sessionId()); t->addEffectChain_l(chain); break; } } // put the chain aside if we could not find a record thread with the same session id. if (i == mRecordThreads.size()) { putOrphanEffectChain_l(chain); } } const sp ioDesc = new AudioIoDescriptor(); ioDesc->mIoHandle = input; ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); mRecordThreads.removeItem(input); } // FIXME: calling thread->exit() without mLock held should not be needed anymore now that // we have a different lock for notification client closeInputFinish(thread); return NO_ERROR; } void AudioFlinger::closeInputFinish(sp thread) { thread->exit(); AudioStreamIn *in = thread->clearInput(); ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); // from now on thread->mInput is NULL in->hwDev()->close_input_stream(in->hwDev(), in->stream); delete in; } void AudioFlinger::closeInputInternal_l(sp thread) { mRecordThreads.removeItem(thread->mId); closeInputFinish(thread); } status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) { Mutex::Autolock _l(mLock); ALOGV("invalidateStream() stream %d", stream); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); thread->invalidateTracks(stream); } return NO_ERROR; } audio_unique_id_t AudioFlinger::newAudioUniqueId() { return nextUniqueId(); } void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) { Mutex::Autolock _l(mLock); pid_t caller = IPCThreadState::self()->getCallingPid(); ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); if (pid != -1 && (caller == getpid_cached)) { caller = pid; } { Mutex::Autolock _cl(mClientLock); // Ignore requests received from processes not known as notification client. The request // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be // called from a different pid leaving a stale session reference. Also we don't know how // to clear this reference if the client process dies. if (mNotificationClients.indexOfKey(caller) < 0) { ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); return; } } size_t num = mAudioSessionRefs.size(); for (size_t i = 0; i< num; i++) { AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); if (ref->mSessionid == audioSession && ref->mPid == caller) { ref->mCnt++; ALOGV(" incremented refcount to %d", ref->mCnt); return; } } mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); ALOGV(" added new entry for %d", audioSession); } void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) { Mutex::Autolock _l(mLock); pid_t caller = IPCThreadState::self()->getCallingPid(); ALOGV("releasing %d from %d for %d", audioSession, caller, pid); if (pid != -1 && (caller == getpid_cached)) { caller = pid; } size_t num = mAudioSessionRefs.size(); for (size_t i = 0; i< num; i++) { AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); if (ref->mSessionid == audioSession && ref->mPid == caller) { ref->mCnt--; ALOGV(" decremented refcount to %d", ref->mCnt); if (ref->mCnt == 0) { mAudioSessionRefs.removeAt(i); delete ref; purgeStaleEffects_l(); } return; } } // If the caller is mediaserver it is likely that the session being released was acquired // on behalf of a process not in notification clients and we ignore the warning. ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); } void AudioFlinger::purgeStaleEffects_l() { ALOGV("purging stale effects"); Vector< sp > chains; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); for (size_t j = 0; j < t->mEffectChains.size(); j++) { sp ec = t->mEffectChains[j]; if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { chains.push(ec); } } } for (size_t i = 0; i < mRecordThreads.size(); i++) { sp t = mRecordThreads.valueAt(i); for (size_t j = 0; j < t->mEffectChains.size(); j++) { sp ec = t->mEffectChains[j]; chains.push(ec); } } for (size_t i = 0; i < chains.size(); i++) { sp ec = chains[i]; int sessionid = ec->sessionId(); sp t = ec->mThread.promote(); if (t == 0) { continue; } size_t numsessionrefs = mAudioSessionRefs.size(); bool found = false; for (size_t k = 0; k < numsessionrefs; k++) { AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); if (ref->mSessionid == sessionid) { ALOGV(" session %d still exists for %d with %d refs", sessionid, ref->mPid, ref->mCnt); found = true; break; } } if (!found) { Mutex::Autolock _l(t->mLock); // remove all effects from the chain while (ec->mEffects.size()) { sp effect = ec->mEffects[0]; effect->unPin(); t->removeEffect_l(effect); if (effect->purgeHandles()) { t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); } AudioSystem::unregisterEffect(effect->id()); } } } return; } // checkPlaybackThread_l() must be called with AudioFlinger::mLock held AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const { return mPlaybackThreads.valueFor(output).get(); } // checkMixerThread_l() must be called with AudioFlinger::mLock held AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const { PlaybackThread *thread = checkPlaybackThread_l(output); return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; } // checkRecordThread_l() must be called with AudioFlinger::mLock held AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const { return mRecordThreads.valueFor(input).get(); } uint32_t AudioFlinger::nextUniqueId() { return (uint32_t) android_atomic_inc(&mNextUniqueId); } AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); if(thread->isDuplicating()) { continue; } AudioStreamOut *output = thread->getOutput(); if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { return thread; } } return NULL; } audio_devices_t AudioFlinger::primaryOutputDevice_l() const { PlaybackThread *thread = primaryPlaybackThread_l(); if (thread == NULL) { return 0; } return thread->outDevice(); } sp AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, int triggerSession, int listenerSession, sync_event_callback_t callBack, wp cookie) { Mutex::Autolock _l(mLock); sp event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); status_t playStatus = NAME_NOT_FOUND; status_t recStatus = NAME_NOT_FOUND; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); if (playStatus == NO_ERROR) { return event; } } for (size_t i = 0; i < mRecordThreads.size(); i++) { recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); if (recStatus == NO_ERROR) { return event; } } if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { mPendingSyncEvents.add(event); } else { ALOGV("createSyncEvent() invalid event %d", event->type()); event.clear(); } return event; } // ---------------------------------------------------------------------------- // Effect management // ---------------------------------------------------------------------------- status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const { Mutex::Autolock _l(mLock); return EffectQueryNumberEffects(numEffects); } status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const { Mutex::Autolock _l(mLock); return EffectQueryEffect(index, descriptor); } status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, effect_descriptor_t *descriptor) const { Mutex::Autolock _l(mLock); return EffectGetDescriptor(pUuid, descriptor); } sp AudioFlinger::createEffect( effect_descriptor_t *pDesc, const sp& effectClient, int32_t priority, audio_io_handle_t io, int sessionId, const String16& opPackageName, status_t *status, int *id, int *enabled) { status_t lStatus = NO_ERROR; sp handle; effect_descriptor_t desc; pid_t pid = IPCThreadState::self()->getCallingPid(); ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", pid, effectClient.get(), priority, sessionId, io); if (pDesc == NULL) { lStatus = BAD_VALUE; goto Exit; } // check audio settings permission for global effects if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects // that can only be created by audio policy manager (running in same process) if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { lStatus = PERMISSION_DENIED; goto Exit; } { if (!EffectIsNullUuid(&pDesc->uuid)) { // if uuid is specified, request effect descriptor lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); if (lStatus < 0) { ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); goto Exit; } } else { // if uuid is not specified, look for an available implementation // of the required type in effect factory if (EffectIsNullUuid(&pDesc->type)) { ALOGW("createEffect() no effect type"); lStatus = BAD_VALUE; goto Exit; } uint32_t numEffects = 0; effect_descriptor_t d; d.flags = 0; // prevent compiler warning bool found = false; lStatus = EffectQueryNumberEffects(&numEffects); if (lStatus < 0) { ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); goto Exit; } for (uint32_t i = 0; i < numEffects; i++) { lStatus = EffectQueryEffect(i, &desc); if (lStatus < 0) { ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); continue; } if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { // If matching type found save effect descriptor. If the session is // 0 and the effect is not auxiliary, continue enumeration in case // an auxiliary version of this effect type is available found = true; d = desc; if (sessionId != AUDIO_SESSION_OUTPUT_MIX || (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { break; } } } if (!found) { lStatus = BAD_VALUE; ALOGW("createEffect() effect not found"); goto Exit; } // For same effect type, chose auxiliary version over insert version if // connect to output mix (Compliance to OpenSL ES) if (sessionId == AUDIO_SESSION_OUTPUT_MIX && (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { desc = d; } } // Do not allow auxiliary effects on a session different from 0 (output mix) if (sessionId != AUDIO_SESSION_OUTPUT_MIX && (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { lStatus = INVALID_OPERATION; goto Exit; } // check recording permission for visualizer if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && !recordingAllowed(opPackageName)) { lStatus = PERMISSION_DENIED; goto Exit; } // return effect descriptor *pDesc = desc; if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { // if the output returned by getOutputForEffect() is removed before we lock the // mutex below, the call to checkPlaybackThread_l(io) below will detect it // and we will exit safely io = AudioSystem::getOutputForEffect(&desc); ALOGV("createEffect got output %d", io); } Mutex::Autolock _l(mLock); // If output is not specified try to find a matching audio session ID in one of the // output threads. // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX // because of code checking output when entering the function. // Note: io is never 0 when creating an effect on an input if (io == AUDIO_IO_HANDLE_NONE) { if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { // output must be specified by AudioPolicyManager when using session // AUDIO_SESSION_OUTPUT_STAGE lStatus = BAD_VALUE; goto Exit; } // look for the thread where the specified audio session is present for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { io = mPlaybackThreads.keyAt(i); break; } } if (io == 0) { for (size_t i = 0; i < mRecordThreads.size(); i++) { if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { io = mRecordThreads.keyAt(i); break; } } } // If no output thread contains the requested session ID, default to // first output. The effect chain will be moved to the correct output // thread when a track with the same session ID is created if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { io = mPlaybackThreads.keyAt(0); } ALOGV("createEffect() got io %d for effect %s", io, desc.name); } ThreadBase *thread = checkRecordThread_l(io); if (thread == NULL) { thread = checkPlaybackThread_l(io); if (thread == NULL) { ALOGE("createEffect() unknown output thread"); lStatus = BAD_VALUE; goto Exit; } } else { // Check if one effect chain was awaiting for an effect to be created on this // session and used it instead of creating a new one. sp chain = getOrphanEffectChain_l((audio_session_t)sessionId); if (chain != 0) { Mutex::Autolock _l(thread->mLock); thread->addEffectChain_l(chain); } } sp client = registerPid(pid); // create effect on selected output thread handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus); if (handle != 0 && id != NULL) { *id = handle->id(); } if (handle == 0) { // remove local strong reference to Client with mClientLock held Mutex::Autolock _cl(mClientLock); client.clear(); } } Exit: *status = lStatus; return handle; } status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, audio_io_handle_t dstOutput) { ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", sessionId, srcOutput, dstOutput); Mutex::Autolock _l(mLock); if (srcOutput == dstOutput) { ALOGW("moveEffects() same dst and src outputs %d", dstOutput); return NO_ERROR; } PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); if (srcThread == NULL) { ALOGW("moveEffects() bad srcOutput %d", srcOutput); return BAD_VALUE; } PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); if (dstThread == NULL) { ALOGW("moveEffects() bad dstOutput %d", dstOutput); return BAD_VALUE; } Mutex::Autolock _dl(dstThread->mLock); Mutex::Autolock _sl(srcThread->mLock); return moveEffectChain_l(sessionId, srcThread, dstThread, false); } // moveEffectChain_l must be called with both srcThread and dstThread mLocks held status_t AudioFlinger::moveEffectChain_l(int sessionId, AudioFlinger::PlaybackThread *srcThread, AudioFlinger::PlaybackThread *dstThread, bool reRegister) { ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", sessionId, srcThread, dstThread); sp chain = srcThread->getEffectChain_l(sessionId); if (chain == 0) { ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", sessionId, srcThread); return INVALID_OPERATION; } // Check whether the destination thread has a channel count of FCC_2, which is // currently required for (most) effects. Prevent moving the effect chain here rather // than disabling the addEffect_l() call in dstThread below. if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && dstThread->mChannelCount != FCC_2) { ALOGW("moveEffectChain_l() effect chain failed because" " destination thread %p channel count(%u) != %u", dstThread, dstThread->mChannelCount, FCC_2); return INVALID_OPERATION; } // remove chain first. This is useful only if reconfiguring effect chain on same output thread, // so that a new chain is created with correct parameters when first effect is added. This is // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is // removed. srcThread->removeEffectChain_l(chain); // transfer all effects one by one so that new effect chain is created on new thread with // correct buffer sizes and audio parameters and effect engines reconfigured accordingly sp dstChain; uint32_t strategy = 0; // prevent compiler warning sp effect = chain->getEffectFromId_l(0); Vector< sp > removed; status_t status = NO_ERROR; while (effect != 0) { srcThread->removeEffect_l(effect); removed.add(effect); status = dstThread->addEffect_l(effect); if (status != NO_ERROR) { break; } // removeEffect_l() has stopped the effect if it was active so it must be restarted if (effect->state() == EffectModule::ACTIVE || effect->state() == EffectModule::STOPPING) { effect->start(); } // if the move request is not received from audio policy manager, the effect must be // re-registered with the new strategy and output if (dstChain == 0) { dstChain = effect->chain().promote(); if (dstChain == 0) { ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); status = NO_INIT; break; } strategy = dstChain->strategy(); } if (reRegister) { AudioSystem::unregisterEffect(effect->id()); AudioSystem::registerEffect(&effect->desc(), dstThread->id(), strategy, sessionId, effect->id()); AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); } effect = chain->getEffectFromId_l(0); } if (status != NO_ERROR) { for (size_t i = 0; i < removed.size(); i++) { srcThread->addEffect_l(removed[i]); if (dstChain != 0 && reRegister) { AudioSystem::unregisterEffect(removed[i]->id()); AudioSystem::registerEffect(&removed[i]->desc(), srcThread->id(), strategy, sessionId, removed[i]->id()); AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); } } } return status; } bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() { if (mGlobalEffectEnableTime != 0 && ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { return true; } for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp ec = mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); if (ec != 0 && ec->isNonOffloadableEnabled()) { return true; } } return false; } void AudioFlinger::onNonOffloadableGlobalEffectEnable() { Mutex::Autolock _l(mLock); mGlobalEffectEnableTime = systemTime(); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); if (t->mType == ThreadBase::OFFLOAD) { t->invalidateTracks(AUDIO_STREAM_MUSIC); } } } status_t AudioFlinger::putOrphanEffectChain_l(const sp& chain) { audio_session_t session = (audio_session_t)chain->sessionId(); ssize_t index = mOrphanEffectChains.indexOfKey(session); ALOGV("putOrphanEffectChain_l session %d index %d", session, index); if (index >= 0) { ALOGW("putOrphanEffectChain_l chain for session %d already present", session); return ALREADY_EXISTS; } mOrphanEffectChains.add(session, chain); return NO_ERROR; } sp AudioFlinger::getOrphanEffectChain_l(audio_session_t session) { sp chain; ssize_t index = mOrphanEffectChains.indexOfKey(session); ALOGV("getOrphanEffectChain_l session %d index %d", session, index); if (index >= 0) { chain = mOrphanEffectChains.valueAt(index); mOrphanEffectChains.removeItemsAt(index); } return chain; } bool AudioFlinger::updateOrphanEffectChains(const sp& effect) { Mutex::Autolock _l(mLock); audio_session_t session = (audio_session_t)effect->sessionId(); ssize_t index = mOrphanEffectChains.indexOfKey(session); ALOGV("updateOrphanEffectChains session %d index %d", session, index); if (index >= 0) { sp chain = mOrphanEffectChains.valueAt(index); if (chain->removeEffect_l(effect) == 0) { ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); mOrphanEffectChains.removeItemsAt(index); } return true; } return false; } struct Entry { #define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 char mFileName[TEE_MAX_FILENAME]; }; int comparEntry(const void *p1, const void *p2) { return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); } #ifdef TEE_SINK void AudioFlinger::dumpTee(int fd, const sp& source, audio_io_handle_t id) { NBAIO_Source *teeSource = source.get(); if (teeSource != NULL) { // .wav rotation // There is a benign race condition if 2 threads call this simultaneously. // They would both traverse the directory, but the result would simply be // failures at unlink() which are ignored. It's also unlikely since // normally dumpsys is only done by bugreport or from the command line. char teePath[32+256]; strcpy(teePath, "/data/misc/media"); size_t teePathLen = strlen(teePath); DIR *dir = opendir(teePath); teePath[teePathLen++] = '/'; if (dir != NULL) { #define TEE_MAX_SORT 20 // number of entries to sort #define TEE_MAX_KEEP 10 // number of entries to keep struct Entry entries[TEE_MAX_SORT]; size_t entryCount = 0; while (entryCount < TEE_MAX_SORT) { struct dirent de; struct dirent *result = NULL; int rc = readdir_r(dir, &de, &result); if (rc != 0) { ALOGW("readdir_r failed %d", rc); break; } if (result == NULL) { break; } if (result != &de) { ALOGW("readdir_r returned unexpected result %p != %p", result, &de); break; } // ignore non .wav file entries size_t nameLen = strlen(de.d_name); if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || strcmp(&de.d_name[nameLen - 4], ".wav")) { continue; } strcpy(entries[entryCount++].mFileName, de.d_name); } (void) closedir(dir); if (entryCount > TEE_MAX_KEEP) { qsort(entries, entryCount, sizeof(Entry), comparEntry); for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { strcpy(&teePath[teePathLen], entries[i].mFileName); (void) unlink(teePath); } } } else { if (fd >= 0) { dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); } } char teeTime[16]; struct timeval tv; gettimeofday(&tv, NULL); struct tm tm; localtime_r(&tv.tv_sec, &tm); strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); if (teeFd >= 0) { // FIXME use libsndfile char wavHeader[44]; memcpy(wavHeader, "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", sizeof(wavHeader)); NBAIO_Format format = teeSource->format(); unsigned channelCount = Format_channelCount(format); uint32_t sampleRate = Format_sampleRate(format); size_t frameSize = Format_frameSize(format); wavHeader[22] = channelCount; // number of channels wavHeader[24] = sampleRate; // sample rate wavHeader[25] = sampleRate >> 8; wavHeader[32] = frameSize; // block alignment wavHeader[33] = frameSize >> 8; write(teeFd, wavHeader, sizeof(wavHeader)); size_t total = 0; bool firstRead = true; #define TEE_SINK_READ 1024 // frames per I/O operation void *buffer = malloc(TEE_SINK_READ * frameSize); ALOG_ASSERT(buffer != NULL); for (;;) { size_t count = TEE_SINK_READ; ssize_t actual = teeSource->read(buffer, count, AudioBufferProvider::kInvalidPTS); bool wasFirstRead = firstRead; firstRead = false; if (actual <= 0) { if (actual == (ssize_t) OVERRUN && wasFirstRead) { continue; } break; } ALOG_ASSERT(actual <= (ssize_t)count); write(teeFd, buffer, actual * frameSize); total += actual; } free(buffer); lseek(teeFd, (off_t) 4, SEEK_SET); uint32_t temp = 44 + total * frameSize - 8; // FIXME not big-endian safe write(teeFd, &temp, sizeof(temp)); lseek(teeFd, (off_t) 40, SEEK_SET); temp = total * frameSize; // FIXME not big-endian safe write(teeFd, &temp, sizeof(temp)); close(teeFd); if (fd >= 0) { dprintf(fd, "tee copied to %s\n", teePath); } } else { if (fd >= 0) { dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); } } } } #endif // ---------------------------------------------------------------------------- status_t AudioFlinger::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioFlinger::onTransact(code, data, reply, flags); } } // namespace android