AudioFlinger files reorganization
Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
new file mode 100644
index 0000000..1ceb850
--- /dev/null
+++ b/services/audioflinger/Threads.cpp
@@ -0,0 +1,4426 @@
+/*
+**
+** Copyright 2012, 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.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <fcntl.h>
+#include <sys/stat.h>
+#include <cutils/properties.h>
+#include <cutils/compiler.h>
+#include <utils/Log.h>
+
+#include <private/media/AudioTrackShared.h>
+#include <hardware/audio.h>
+#include <audio_effects/effect_ns.h>
+#include <audio_effects/effect_aec.h>
+#include <audio_utils/primitives.h>
+
+// NBAIO implementations
+#include <media/nbaio/AudioStreamOutSink.h>
+#include <media/nbaio/MonoPipe.h>
+#include <media/nbaio/MonoPipeReader.h>
+#include <media/nbaio/Pipe.h>
+#include <media/nbaio/PipeReader.h>
+#include <media/nbaio/SourceAudioBufferProvider.h>
+
+#include <powermanager/PowerManager.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
+
+#include "AudioFlinger.h"
+#include "AudioMixer.h"
+#include "FastMixer.h"
+#include "ServiceUtilities.h"
+#include "SchedulingPolicyService.h"
+
+#undef ADD_BATTERY_DATA
+
+#ifdef ADD_BATTERY_DATA
+#include <media/IMediaPlayerService.h>
+#include <media/IMediaDeathNotifier.h>
+#endif
+
+// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
+#ifdef DEBUG_CPU_USAGE
+#include <cpustats/CentralTendencyStatistics.h>
+#include <cpustats/ThreadCpuUsage.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 {
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+// allow less retry attempts on direct output thread.
+// direct outputs can be a scarce resource in audio hardware and should
+// be released as quickly as possible.
+static const int8_t kMaxTrackRetriesDirect = 2;
+
+// don't warn about blocked writes or record buffer overflows more often than this
+static const nsecs_t kWarningThrottleNs = seconds(5);
+
+// RecordThread loop sleep time upon application overrun or audio HAL read error
+static const int kRecordThreadSleepUs = 5000;
+
+// maximum time to wait for setParameters to complete
+static const nsecs_t kSetParametersTimeoutNs = seconds(2);
+
+// minimum sleep time for the mixer thread loop when tracks are active but in underrun
+static const uint32_t kMinThreadSleepTimeUs = 5000;
+// maximum divider applied to the active sleep time in the mixer thread loop
+static const uint32_t kMaxThreadSleepTimeShift = 2;
+
+// minimum normal mix buffer size, expressed in milliseconds rather than frames
+static const uint32_t kMinNormalMixBufferSizeMs = 20;
+// maximum normal mix buffer size
+static const uint32_t kMaxNormalMixBufferSizeMs = 24;
+
+// Whether to use fast mixer
+static const enum {
+ FastMixer_Never, // never initialize or use: for debugging only
+ FastMixer_Always, // always initialize and use, even if not needed: for debugging only
+ // normal mixer multiplier is 1
+ FastMixer_Static, // initialize if needed, then use all the time if initialized,
+ // multiplier is calculated based on min & max normal mixer buffer size
+ FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load,
+ // multiplier is calculated based on min & max normal mixer buffer size
+ // FIXME for FastMixer_Dynamic:
+ // Supporting this option will require fixing HALs that can't handle large writes.
+ // For example, one HAL implementation returns an error from a large write,
+ // and another HAL implementation corrupts memory, possibly in the sample rate converter.
+ // We could either fix the HAL implementations, or provide a wrapper that breaks
+ // up large writes into smaller ones, and the wrapper would need to deal with scheduler.
+} kUseFastMixer = FastMixer_Static;
+
+// Priorities for requestPriority
+static const int kPriorityAudioApp = 2;
+static const int kPriorityFastMixer = 3;
+
+// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
+// for the track. The client then sub-divides this into smaller buffers for its use.
+// Currently the client uses double-buffering by default, but doesn't tell us about that.
+// So for now we just assume that client is double-buffered.
+// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
+// N-buffering, so AudioFlinger could allocate the right amount of memory.
+// See the client's minBufCount and mNotificationFramesAct calculations for details.
+static const int kFastTrackMultiplier = 2;
+
+// ----------------------------------------------------------------------------
+
+#ifdef ADD_BATTERY_DATA
+// To collect the amplifier usage
+static void addBatteryData(uint32_t params) {
+ sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
+ if (service == NULL) {
+ // it already logged
+ return;
+ }
+
+ service->addBatteryData(params);
+}
+#endif
+
+
+// ----------------------------------------------------------------------------
+// CPU Stats
+// ----------------------------------------------------------------------------
+
+class CpuStats {
+public:
+ CpuStats();
+ void sample(const String8 &title);
+#ifdef DEBUG_CPU_USAGE
+private:
+ ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
+ CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
+
+ CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
+
+ int mCpuNum; // thread's current CPU number
+ int mCpukHz; // frequency of thread's current CPU in kHz
+#endif
+};
+
+CpuStats::CpuStats()
+#ifdef DEBUG_CPU_USAGE
+ : mCpuNum(-1), mCpukHz(-1)
+#endif
+{
+}
+
+void CpuStats::sample(const String8 &title) {
+#ifdef DEBUG_CPU_USAGE
+ // get current thread's delta CPU time in wall clock ns
+ double wcNs;
+ bool valid = mCpuUsage.sampleAndEnable(wcNs);
+
+ // record sample for wall clock statistics
+ if (valid) {
+ mWcStats.sample(wcNs);
+ }
+
+ // get the current CPU number
+ int cpuNum = sched_getcpu();
+
+ // get the current CPU frequency in kHz
+ int cpukHz = mCpuUsage.getCpukHz(cpuNum);
+
+ // check if either CPU number or frequency changed
+ if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
+ mCpuNum = cpuNum;
+ mCpukHz = cpukHz;
+ // ignore sample for purposes of cycles
+ valid = false;
+ }
+
+ // if no change in CPU number or frequency, then record sample for cycle statistics
+ if (valid && mCpukHz > 0) {
+ double cycles = wcNs * cpukHz * 0.000001;
+ mHzStats.sample(cycles);
+ }
+
+ unsigned n = mWcStats.n();
+ // mCpuUsage.elapsed() is expensive, so don't call it every loop
+ if ((n & 127) == 1) {
+ long long elapsed = mCpuUsage.elapsed();
+ if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
+ double perLoop = elapsed / (double) n;
+ double perLoop100 = perLoop * 0.01;
+ double perLoop1k = perLoop * 0.001;
+ double mean = mWcStats.mean();
+ double stddev = mWcStats.stddev();
+ double minimum = mWcStats.minimum();
+ double maximum = mWcStats.maximum();
+ double meanCycles = mHzStats.mean();
+ double stddevCycles = mHzStats.stddev();
+ double minCycles = mHzStats.minimum();
+ double maxCycles = mHzStats.maximum();
+ mCpuUsage.resetElapsed();
+ mWcStats.reset();
+ mHzStats.reset();
+ ALOGD("CPU usage for %s over past %.1f secs\n"
+ " (%u mixer loops at %.1f mean ms per loop):\n"
+ " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
+ " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
+ " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
+ title.string(),
+ elapsed * .000000001, n, perLoop * .000001,
+ mean * .001,
+ stddev * .001,
+ minimum * .001,
+ maximum * .001,
+ mean / perLoop100,
+ stddev / perLoop100,
+ minimum / perLoop100,
+ maximum / perLoop100,
+ meanCycles / perLoop1k,
+ stddevCycles / perLoop1k,
+ minCycles / perLoop1k,
+ maxCycles / perLoop1k);
+
+ }
+ }
+#endif
+};
+
+// ----------------------------------------------------------------------------
+// ThreadBase
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
+ : Thread(false /*canCallJava*/),
+ mType(type),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
+ // mChannelMask
+ mChannelCount(0),
+ mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+ mParamStatus(NO_ERROR),
+ mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
+ mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
+ // mName will be set by concrete (non-virtual) subclass
+ mDeathRecipient(new PMDeathRecipient(this))
+{
+}
+
+AudioFlinger::ThreadBase::~ThreadBase()
+{
+ mParamCond.broadcast();
+ // do not lock the mutex in destructor
+ releaseWakeLock_l();
+ if (mPowerManager != 0) {
+ sp<IBinder> binder = mPowerManager->asBinder();
+ binder->unlinkToDeath(mDeathRecipient);
+ }
+}
+
+void AudioFlinger::ThreadBase::exit()
+{
+ ALOGV("ThreadBase::exit");
+ // do any cleanup required for exit to succeed
+ preExit();
+ {
+ // This lock prevents the following race in thread (uniprocessor for illustration):
+ // if (!exitPending()) {
+ // // context switch from here to exit()
+ // // exit() calls requestExit(), what exitPending() observes
+ // // exit() calls signal(), which is dropped since no waiters
+ // // context switch back from exit() to here
+ // mWaitWorkCV.wait(...);
+ // // now thread is hung
+ // }
+ AutoMutex lock(mLock);
+ requestExit();
+ mWaitWorkCV.broadcast();
+ }
+ // When Thread::requestExitAndWait is made virtual and this method is renamed to
+ // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
+ requestExitAndWait();
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+ status_t status;
+
+ ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
+ Mutex::Autolock _l(mLock);
+
+ mNewParameters.add(keyValuePairs);
+ mWaitWorkCV.signal();
+ // wait condition with timeout in case the thread loop has exited
+ // before the request could be processed
+ if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
+ status = mParamStatus;
+ mWaitWorkCV.signal();
+ } else {
+ status = TIMED_OUT;
+ }
+ return status;
+}
+
+void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
+{
+ Mutex::Autolock _l(mLock);
+ sendIoConfigEvent_l(event, param);
+}
+
+// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param)
+{
+ IoConfigEvent *ioEvent = new IoConfigEvent(event, param);
+ mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent));
+ ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event,
+ param);
+ mWaitWorkCV.signal();
+}
+
+// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
+{
+ PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio);
+ mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent));
+ ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d",
+ mConfigEvents.size(), pid, tid, prio);
+ mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+ mLock.lock();
+ while (!mConfigEvents.isEmpty()) {
+ ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+ ConfigEvent *event = mConfigEvents[0];
+ mConfigEvents.removeAt(0);
+ // release mLock before locking AudioFlinger mLock: lock order is always
+ // AudioFlinger then ThreadBase to avoid cross deadlock
+ mLock.unlock();
+ switch(event->type()) {
+ case CFG_EVENT_PRIO: {
+ PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
+ int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio());
+ if (err != 0) {
+ ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
+ "error %d",
+ prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
+ }
+ } break;
+ case CFG_EVENT_IO: {
+ IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
+ mAudioFlinger->mLock.lock();
+ audioConfigChanged_l(ioEvent->event(), ioEvent->param());
+ mAudioFlinger->mLock.unlock();
+ } break;
+ default:
+ ALOGE("processConfigEvents() unknown event type %d", event->type());
+ break;
+ }
+ delete event;
+ mLock.lock();
+ }
+ mLock.unlock();
+}
+
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ bool locked = AudioFlinger::dumpTryLock(mLock);
+ if (!locked) {
+ snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
+ write(fd, buffer, strlen(buffer));
+ }
+
+ snprintf(buffer, SIZE, "io handle: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "TID: %d\n", getTid());
+ result.append(buffer);
+ snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
+ result.append(buffer);
+ result.append(" Index Command");
+ for (size_t i = 0; i < mNewParameters.size(); ++i) {
+ snprintf(buffer, SIZE, "\n %02d ", i);
+ result.append(buffer);
+ result.append(mNewParameters[i]);
+ }
+
+ snprintf(buffer, SIZE, "\n\nPending config events: \n");
+ result.append(buffer);
+ for (size_t i = 0; i < mConfigEvents.size(); i++) {
+ mConfigEvents[i]->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+
+ if (locked) {
+ mLock.unlock();
+ }
+}
+
+void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+ write(fd, buffer, strlen(buffer));
+
+ for (size_t i = 0; i < mEffectChains.size(); ++i) {
+ sp<EffectChain> chain = mEffectChains[i];
+ if (chain != 0) {
+ chain->dump(fd, args);
+ }
+ }
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock()
+{
+ Mutex::Autolock _l(mLock);
+ acquireWakeLock_l();
+}
+
+void AudioFlinger::ThreadBase::acquireWakeLock_l()
+{
+ if (mPowerManager == 0) {
+ // use checkService() to avoid blocking if power service is not up yet
+ sp<IBinder> binder =
+ defaultServiceManager()->checkService(String16("power"));
+ if (binder == 0) {
+ ALOGW("Thread %s cannot connect to the power manager service", mName);
+ } else {
+ mPowerManager = interface_cast<IPowerManager>(binder);
+ binder->linkToDeath(mDeathRecipient);
+ }
+ }
+ if (mPowerManager != 0) {
+ sp<IBinder> binder = new BBinder();
+ status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
+ binder,
+ String16(mName));
+ if (status == NO_ERROR) {
+ mWakeLockToken = binder;
+ }
+ ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+ }
+}
+
+void AudioFlinger::ThreadBase::releaseWakeLock()
+{
+ Mutex::Autolock _l(mLock);
+ releaseWakeLock_l();
+}
+
+void AudioFlinger::ThreadBase::releaseWakeLock_l()
+{
+ if (mWakeLockToken != 0) {
+ ALOGV("releaseWakeLock_l() %s", mName);
+ if (mPowerManager != 0) {
+ mPowerManager->releaseWakeLock(mWakeLockToken, 0);
+ }
+ mWakeLockToken.clear();
+ }
+}
+
+void AudioFlinger::ThreadBase::clearPowerManager()
+{
+ Mutex::Autolock _l(mLock);
+ releaseWakeLock_l();
+ mPowerManager.clear();
+}
+
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+{
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ thread->clearPowerManager();
+ }
+ ALOGW("power manager service died !!!");
+}
+
+void AudioFlinger::ThreadBase::setEffectSuspended(
+ const effect_uuid_t *type, bool suspend, int sessionId)
+{
+ Mutex::Autolock _l(mLock);
+ setEffectSuspended_l(type, suspend, sessionId);
+}
+
+void AudioFlinger::ThreadBase::setEffectSuspended_l(
+ const effect_uuid_t *type, bool suspend, int sessionId)
+{
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
+ if (chain != 0) {
+ if (type != NULL) {
+ chain->setEffectSuspended_l(type, suspend);
+ } else {
+ chain->setEffectSuspendedAll_l(suspend);
+ }
+ }
+
+ updateSuspendedSessions_l(type, suspend, sessionId);
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
+{
+ ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
+ if (index < 0) {
+ return;
+ }
+
+ const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
+ mSuspendedSessions.valueAt(index);
+
+ for (size_t i = 0; i < sessionEffects.size(); i++) {
+ sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
+ for (int j = 0; j < desc->mRefCount; j++) {
+ if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
+ chain->setEffectSuspendedAll_l(true);
+ } else {
+ ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
+ desc->mType.timeLow);
+ chain->setEffectSuspended_l(&desc->mType, true);
+ }
+ }
+ }
+}
+
+void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
+ bool suspend,
+ int sessionId)
+{
+ ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
+
+ KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
+
+ if (suspend) {
+ if (index >= 0) {
+ sessionEffects = mSuspendedSessions.valueAt(index);
+ } else {
+ mSuspendedSessions.add(sessionId, sessionEffects);
+ }
+ } else {
+ if (index < 0) {
+ return;
+ }
+ sessionEffects = mSuspendedSessions.valueAt(index);
+ }
+
+
+ int key = EffectChain::kKeyForSuspendAll;
+ if (type != NULL) {
+ key = type->timeLow;
+ }
+ index = sessionEffects.indexOfKey(key);
+
+ sp<SuspendedSessionDesc> desc;
+ if (suspend) {
+ if (index >= 0) {
+ desc = sessionEffects.valueAt(index);
+ } else {
+ desc = new SuspendedSessionDesc();
+ if (type != NULL) {
+ desc->mType = *type;
+ }
+ sessionEffects.add(key, desc);
+ ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
+ }
+ desc->mRefCount++;
+ } else {
+ if (index < 0) {
+ return;
+ }
+ desc = sessionEffects.valueAt(index);
+ if (--desc->mRefCount == 0) {
+ ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
+ sessionEffects.removeItemsAt(index);
+ if (sessionEffects.isEmpty()) {
+ ALOGV("updateSuspendedSessions_l() restore removing session %d",
+ sessionId);
+ mSuspendedSessions.removeItem(sessionId);
+ }
+ }
+ }
+ if (!sessionEffects.isEmpty()) {
+ mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
+ }
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
+ bool enabled,
+ int sessionId)
+{
+ Mutex::Autolock _l(mLock);
+ checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
+}
+
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
+ bool enabled,
+ int sessionId)
+{
+ if (mType != RECORD) {
+ // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
+ // another session. This gives the priority to well behaved effect control panels
+ // and applications not using global effects.
+ // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
+ // global effects
+ if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
+ setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
+ }
+ }
+
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
+ if (chain != 0) {
+ chain->checkSuspendOnEffectEnabled(effect, enabled);
+ }
+}
+
+// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
+ const sp<AudioFlinger::Client>& client,
+ const sp<IEffectClient>& effectClient,
+ int32_t priority,
+ int sessionId,
+ effect_descriptor_t *desc,
+ int *enabled,
+ status_t *status
+ )
+{
+ sp<EffectModule> effect;
+ sp<EffectHandle> handle;
+ status_t lStatus;
+ sp<EffectChain> chain;
+ bool chainCreated = false;
+ bool effectCreated = false;
+ bool effectRegistered = false;
+
+ lStatus = initCheck();
+ if (lStatus != NO_ERROR) {
+ ALOGW("createEffect_l() Audio driver not initialized.");
+ goto Exit;
+ }
+
+ // Do not allow effects with session ID 0 on direct output or duplicating threads
+ // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
+ ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
+ desc->name, sessionId);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ // Only Pre processor effects are allowed on input threads and only on input threads
+ if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
+ ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
+ desc->name, desc->flags, mType);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+
+ // check for existing effect chain with the requested audio session
+ chain = getEffectChain_l(sessionId);
+ if (chain == 0) {
+ // create a new chain for this session
+ ALOGV("createEffect_l() new effect chain for session %d", sessionId);
+ chain = new EffectChain(this, sessionId);
+ addEffectChain_l(chain);
+ chain->setStrategy(getStrategyForSession_l(sessionId));
+ chainCreated = true;
+ } else {
+ effect = chain->getEffectFromDesc_l(desc);
+ }
+
+ ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
+
+ if (effect == 0) {
+ int id = mAudioFlinger->nextUniqueId();
+ // Check CPU and memory usage
+ lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
+ if (lStatus != NO_ERROR) {
+ goto Exit;
+ }
+ effectRegistered = true;
+ // create a new effect module if none present in the chain
+ effect = new EffectModule(this, chain, desc, id, sessionId);
+ lStatus = effect->status();
+ if (lStatus != NO_ERROR) {
+ goto Exit;
+ }
+ lStatus = chain->addEffect_l(effect);
+ if (lStatus != NO_ERROR) {
+ goto Exit;
+ }
+ effectCreated = true;
+
+ effect->setDevice(mOutDevice);
+ effect->setDevice(mInDevice);
+ effect->setMode(mAudioFlinger->getMode());
+ effect->setAudioSource(mAudioSource);
+ }
+ // create effect handle and connect it to effect module
+ handle = new EffectHandle(effect, client, effectClient, priority);
+ lStatus = effect->addHandle(handle.get());
+ if (enabled != NULL) {
+ *enabled = (int)effect->isEnabled();
+ }
+ }
+
+Exit:
+ if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
+ Mutex::Autolock _l(mLock);
+ if (effectCreated) {
+ chain->removeEffect_l(effect);
+ }
+ if (effectRegistered) {
+ AudioSystem::unregisterEffect(effect->id());
+ }
+ if (chainCreated) {
+ removeEffectChain_l(chain);
+ }
+ handle.clear();
+ }
+
+ if (status != NULL) {
+ *status = lStatus;
+ }
+ return handle;
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
+{
+ Mutex::Autolock _l(mLock);
+ return getEffect_l(sessionId, effectId);
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
+{
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
+ return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
+}
+
+// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
+// PlaybackThread::mLock held
+status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
+{
+ // check for existing effect chain with the requested audio session
+ int sessionId = effect->sessionId();
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
+ bool chainCreated = false;
+
+ if (chain == 0) {
+ // create a new chain for this session
+ ALOGV("addEffect_l() new effect chain for session %d", sessionId);
+ chain = new EffectChain(this, sessionId);
+ addEffectChain_l(chain);
+ chain->setStrategy(getStrategyForSession_l(sessionId));
+ chainCreated = true;
+ }
+ ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
+
+ if (chain->getEffectFromId_l(effect->id()) != 0) {
+ ALOGW("addEffect_l() %p effect %s already present in chain %p",
+ this, effect->desc().name, chain.get());
+ return BAD_VALUE;
+ }
+
+ status_t status = chain->addEffect_l(effect);
+ if (status != NO_ERROR) {
+ if (chainCreated) {
+ removeEffectChain_l(chain);
+ }
+ return status;
+ }
+
+ effect->setDevice(mOutDevice);
+ effect->setDevice(mInDevice);
+ effect->setMode(mAudioFlinger->getMode());
+ effect->setAudioSource(mAudioSource);
+ return NO_ERROR;
+}
+
+void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
+
+ ALOGV("removeEffect_l() %p effect %p", this, effect.get());
+ effect_descriptor_t desc = effect->desc();
+ if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+ detachAuxEffect_l(effect->id());
+ }
+
+ sp<EffectChain> chain = effect->chain().promote();
+ if (chain != 0) {
+ // remove effect chain if removing last effect
+ if (chain->removeEffect_l(effect) == 0) {
+ removeEffectChain_l(chain);
+ }
+ } else {
+ ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
+ }
+}
+
+void AudioFlinger::ThreadBase::lockEffectChains_l(
+ Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+{
+ effectChains = mEffectChains;
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ mEffectChains[i]->lock();
+ }
+}
+
+void AudioFlinger::ThreadBase::unlockEffectChains(
+ const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+{
+ for (size_t i = 0; i < effectChains.size(); i++) {
+ effectChains[i]->unlock();
+ }
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
+{
+ Mutex::Autolock _l(mLock);
+ return getEffectChain_l(sessionId);
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
+{
+ size_t size = mEffectChains.size();
+ for (size_t i = 0; i < size; i++) {
+ if (mEffectChains[i]->sessionId() == sessionId) {
+ return mEffectChains[i];
+ }
+ }
+ return 0;
+}
+
+void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
+{
+ Mutex::Autolock _l(mLock);
+ size_t size = mEffectChains.size();
+ for (size_t i = 0; i < size; i++) {
+ mEffectChains[i]->setMode_l(mode);
+ }
+}
+
+void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
+ EffectHandle *handle,
+ bool unpinIfLast) {
+
+ Mutex::Autolock _l(mLock);
+ ALOGV("disconnectEffect() %p effect %p", this, effect.get());
+ // delete the effect module if removing last handle on it
+ if (effect->removeHandle(handle) == 0) {
+ if (!effect->isPinned() || unpinIfLast) {
+ removeEffect_l(effect);
+ AudioSystem::unregisterEffect(effect->id());
+ }
+ }
+}
+
+// ----------------------------------------------------------------------------
+// Playback
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output,
+ audio_io_handle_t id,
+ audio_devices_t device,
+ type_t type)
+ : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ // mStreamTypes[] initialized in constructor body
+ mOutput(output),
+ mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
+ mMixerStatus(MIXER_IDLE),
+ mMixerStatusIgnoringFastTracks(MIXER_IDLE),
+ standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
+ mScreenState(AudioFlinger::mScreenState),
+ // index 0 is reserved for normal mixer's submix
+ mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
+{
+ snprintf(mName, kNameLength, "AudioOut_%X", id);
+
+ // Assumes constructor is called by AudioFlinger with it's mLock held, but
+ // it would be safer to explicitly pass initial masterVolume/masterMute as
+ // parameter.
+ //
+ // If the HAL we are using has support for master volume or master mute,
+ // then do not attenuate or mute during mixing (just leave the volume at 1.0
+ // and the mute set to false).
+ mMasterVolume = audioFlinger->masterVolume_l();
+ mMasterMute = audioFlinger->masterMute_l();
+ if (mOutput && mOutput->audioHwDev) {
+ if (mOutput->audioHwDev->canSetMasterVolume()) {
+ mMasterVolume = 1.0;
+ }
+
+ if (mOutput->audioHwDev->canSetMasterMute()) {
+ mMasterMute = false;
+ }
+ }
+
+ readOutputParameters();
+
+ // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
+ // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
+ for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
+ stream = (audio_stream_type_t) (stream + 1)) {
+ mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
+ mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
+ }
+ // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
+ // because mAudioFlinger doesn't have one to copy from
+}
+
+AudioFlinger::PlaybackThread::~PlaybackThread()
+{
+ delete [] mMixBuffer;
+}
+
+void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
+{
+ dumpInternals(fd, args);
+ dumpTracks(fd, args);
+ dumpEffectChains(fd, args);
+}
+
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ result.appendFormat("Output thread %p stream volumes in dB:\n ", this);
+ for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
+ const stream_type_t *st = &mStreamTypes[i];
+ if (i > 0) {
+ result.appendFormat(", ");
+ }
+ result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
+ if (st->mute) {
+ result.append("M");
+ }
+ }
+ result.append("\n");
+ write(fd, result.string(), result.length());
+ result.clear();
+
+ snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
+ result.append(buffer);
+ Track::appendDumpHeader(result);
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> track = mTracks[i];
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+
+ snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
+ result.append(buffer);
+ Track::appendDumpHeader(result);
+ for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+ write(fd, result.string(), result.size());
+
+ // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
+ FastTrackUnderruns underruns = getFastTrackUnderruns(0);
+ fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
+ underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
+}
+
+void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
+ ns2ms(systemTime() - mLastWriteTime));
+ result.append(buffer);
+ snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
+
+ dumpBase(fd, args);
+}
+
+// Thread virtuals
+status_t AudioFlinger::PlaybackThread::readyToRun()
+{
+ status_t status = initCheck();
+ if (status == NO_ERROR) {
+ ALOGI("AudioFlinger's thread %p ready to run", this);
+ } else {
+ ALOGE("No working audio driver found.");
+ }
+ return status;
+}
+
+void AudioFlinger::PlaybackThread::onFirstRef()
+{
+ run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// ThreadBase virtuals
+void AudioFlinger::PlaybackThread::preExit()
+{
+ ALOGV(" preExit()");
+ // FIXME this is using hard-coded strings but in the future, this functionality will be
+ // converted to use audio HAL extensions required to support tunneling
+ mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
+}
+
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+ const sp<AudioFlinger::Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId,
+ IAudioFlinger::track_flags_t *flags,
+ pid_t tid,
+ status_t *status)
+{
+ sp<Track> track;
+ status_t lStatus;
+
+ bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
+
+ // client expresses a preference for FAST, but we get the final say
+ if (*flags & IAudioFlinger::TRACK_FAST) {
+ if (
+ // not timed
+ (!isTimed) &&
+ // either of these use cases:
+ (
+ // use case 1: shared buffer with any frame count
+ (
+ (sharedBuffer != 0)
+ ) ||
+ // use case 2: callback handler and frame count is default or at least as large as HAL
+ (
+ (tid != -1) &&
+ ((frameCount == 0) ||
+ (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+ )
+ ) &&
+ // PCM data
+ audio_is_linear_pcm(format) &&
+ // mono or stereo
+ ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
+ (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
+#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
+ // hardware sample rate
+ (sampleRate == mSampleRate) &&
+#endif
+ // normal mixer has an associated fast mixer
+ hasFastMixer() &&
+ // there are sufficient fast track slots available
+ (mFastTrackAvailMask != 0)
+ // FIXME test that MixerThread for this fast track has a capable output HAL
+ // FIXME add a permission test also?
+ ) {
+ // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
+ if (frameCount == 0) {
+ frameCount = mFrameCount * kFastTrackMultiplier;
+ }
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
+ frameCount, mFrameCount);
+ } else {
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
+ "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
+ "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
+ isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
+ audio_is_linear_pcm(format),
+ channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
+ *flags &= ~IAudioFlinger::TRACK_FAST;
+ // For compatibility with AudioTrack calculation, buffer depth is forced
+ // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
+ // This is probably too conservative, but legacy application code may depend on it.
+ // If you change this calculation, also review the start threshold which is related.
+ uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
+ uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
+ if (minBufCount < 2) {
+ minBufCount = 2;
+ }
+ size_t minFrameCount = mNormalFrameCount * minBufCount;
+ if (frameCount < minFrameCount) {
+ frameCount = minFrameCount;
+ }
+ }
+ }
+
+ if (mType == DIRECT) {
+ if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
+ if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
+ ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
+ "for output %p with format %d",
+ sampleRate, format, channelMask, mOutput, mFormat);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ }
+ } else {
+ // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+ if (sampleRate > mSampleRate*2) {
+ ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ }
+
+ lStatus = initCheck();
+ if (lStatus != NO_ERROR) {
+ ALOGE("Audio driver not initialized.");
+ goto Exit;
+ }
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+
+ // all tracks in same audio session must share the same routing strategy otherwise
+ // conflicts will happen when tracks are moved from one output to another by audio policy
+ // manager
+ uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> t = mTracks[i];
+ if (t != 0 && !t->isOutputTrack()) {
+ uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
+ if (sessionId == t->sessionId() && strategy != actual) {
+ ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
+ strategy, actual);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ }
+ }
+
+ if (!isTimed) {
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId, *flags);
+ } else {
+ track = TimedTrack::create(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ }
+ if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
+ lStatus = NO_MEMORY;
+ goto Exit;
+ }
+ mTracks.add(track);
+
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
+ if (chain != 0) {
+ ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
+ track->setMainBuffer(chain->inBuffer());
+ chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
+ chain->incTrackCnt();
+ }
+
+ if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
+ pid_t callingPid = IPCThreadState::self()->getCallingPid();
+ // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
+ // so ask activity manager to do this on our behalf
+ sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
+ }
+ }
+
+ lStatus = NO_ERROR;
+
+Exit:
+ if (status) {
+ *status = lStatus;
+ }
+ return track;
+}
+
+uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
+{
+ return latency;
+}
+
+uint32_t AudioFlinger::PlaybackThread::latency() const
+{
+ Mutex::Autolock _l(mLock);
+ return latency_l();
+}
+uint32_t AudioFlinger::PlaybackThread::latency_l() const
+{
+ if (initCheck() == NO_ERROR) {
+ return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
+ } else {
+ return 0;
+ }
+}
+
+void AudioFlinger::PlaybackThread::setMasterVolume(float value)
+{
+ Mutex::Autolock _l(mLock);
+ // Don't apply master volume in SW if our HAL can do it for us.
+ if (mOutput && mOutput->audioHwDev &&
+ mOutput->audioHwDev->canSetMasterVolume()) {
+ mMasterVolume = 1.0;
+ } else {
+ mMasterVolume = value;
+ }
+}
+
+void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+{
+ Mutex::Autolock _l(mLock);
+ // Don't apply master mute in SW if our HAL can do it for us.
+ if (mOutput && mOutput->audioHwDev &&
+ mOutput->audioHwDev->canSetMasterMute()) {
+ mMasterMute = false;
+ } else {
+ mMasterMute = muted;
+ }
+}
+
+void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+{
+ Mutex::Autolock _l(mLock);
+ mStreamTypes[stream].volume = value;
+}
+
+void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+{
+ Mutex::Autolock _l(mLock);
+ mStreamTypes[stream].mute = muted;
+}
+
+float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
+{
+ Mutex::Autolock _l(mLock);
+ return mStreamTypes[stream].volume;
+}
+
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+{
+ status_t status = ALREADY_EXISTS;
+
+ // set retry count for buffer fill
+ track->mRetryCount = kMaxTrackStartupRetries;
+ if (mActiveTracks.indexOf(track) < 0) {
+ // the track is newly added, make sure it fills up all its
+ // buffers before playing. This is to ensure the client will
+ // effectively get the latency it requested.
+ track->mFillingUpStatus = Track::FS_FILLING;
+ track->mResetDone = false;
+ track->mPresentationCompleteFrames = 0;
+ mActiveTracks.add(track);
+ if (track->mainBuffer() != mMixBuffer) {
+ sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+ if (chain != 0) {
+ ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
+ track->sessionId());
+ chain->incActiveTrackCnt();
+ }
+ }
+
+ status = NO_ERROR;
+ }
+
+ ALOGV("mWaitWorkCV.broadcast");
+ mWaitWorkCV.broadcast();
+
+ return status;
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+{
+ track->mState = TrackBase::TERMINATED;
+ // active tracks are removed by threadLoop()
+ if (mActiveTracks.indexOf(track) < 0) {
+ removeTrack_l(track);
+ }
+}
+
+void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
+{
+ track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
+ mTracks.remove(track);
+ deleteTrackName_l(track->name());
+ // redundant as track is about to be destroyed, for dumpsys only
+ track->mName = -1;
+ if (track->isFastTrack()) {
+ int index = track->mFastIndex;
+ ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
+ ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
+ mFastTrackAvailMask |= 1 << index;
+ // redundant as track is about to be destroyed, for dumpsys only
+ track->mFastIndex = -1;
+ }
+ sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+ if (chain != 0) {
+ chain->decTrackCnt();
+ }
+}
+
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+{
+ String8 out_s8 = String8("");
+ char *s;
+
+ Mutex::Autolock _l(mLock);
+ if (initCheck() != NO_ERROR) {
+ return out_s8;
+ }
+
+ s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
+ out_s8 = String8(s);
+ free(s);
+ return out_s8;
+}
+
+// audioConfigChanged_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = NULL;
+
+ ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event,
+ param);
+
+ switch (event) {
+ case AudioSystem::OUTPUT_OPENED:
+ case AudioSystem::OUTPUT_CONFIG_CHANGED:
+ desc.channels = mChannelMask;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mNormalFrameCount; // FIXME see
+ // AudioFlinger::frameCount(audio_io_handle_t)
+ desc.latency = latency();
+ param2 = &desc;
+ break;
+
+ case AudioSystem::STREAM_CONFIG_CHANGED:
+ param2 = ¶m;
+ case AudioSystem::OUTPUT_CLOSED:
+ default:
+ break;
+ }
+ mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+ mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
+ mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
+ mChannelCount = (uint16_t)popcount(mChannelMask);
+ mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
+ mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
+ mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
+ if (mFrameCount & 15) {
+ ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
+ mFrameCount);
+ }
+
+ // Calculate size of normal mix buffer relative to the HAL output buffer size
+ double multiplier = 1.0;
+ if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
+ kUseFastMixer == FastMixer_Dynamic)) {
+ size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
+ size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
+ // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
+ minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
+ maxNormalFrameCount = maxNormalFrameCount & ~15;
+ if (maxNormalFrameCount < minNormalFrameCount) {
+ maxNormalFrameCount = minNormalFrameCount;
+ }
+ multiplier = (double) minNormalFrameCount / (double) mFrameCount;
+ if (multiplier <= 1.0) {
+ multiplier = 1.0;
+ } else if (multiplier <= 2.0) {
+ if (2 * mFrameCount <= maxNormalFrameCount) {
+ multiplier = 2.0;
+ } else {
+ multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
+ }
+ } else {
+ // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
+ // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
+ // track, but we sometimes have to do this to satisfy the maximum frame count
+ // constraint)
+ // FIXME this rounding up should not be done if no HAL SRC
+ uint32_t truncMult = (uint32_t) multiplier;
+ if ((truncMult & 1)) {
+ if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
+ ++truncMult;
+ }
+ }
+ multiplier = (double) truncMult;
+ }
+ }
+ mNormalFrameCount = multiplier * mFrameCount;
+ // round up to nearest 16 frames to satisfy AudioMixer
+ mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+ ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
+ mNormalFrameCount);
+
+ delete[] mMixBuffer;
+ mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
+ memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+
+ // force reconfiguration of effect chains and engines to take new buffer size and audio
+ // parameters into account
+ // Note that mLock is not held when readOutputParameters() is called from the constructor
+ // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
+ // matter.
+ // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
+ Vector< sp<EffectChain> > effectChains = mEffectChains;
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
+ }
+}
+
+
+status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
+{
+ if (halFrames == NULL || dspFrames == NULL) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ if (initCheck() != NO_ERROR) {
+ return INVALID_OPERATION;
+ }
+ size_t framesWritten = mBytesWritten / mFrameSize;
+ *halFrames = framesWritten;
+
+ if (isSuspended()) {
+ // return an estimation of rendered frames when the output is suspended
+ size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
+ *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
+ return NO_ERROR;
+ } else {
+ return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
+ }
+}
+
+uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
+{
+ Mutex::Autolock _l(mLock);
+ uint32_t result = 0;
+ if (getEffectChain_l(sessionId) != 0) {
+ result = EFFECT_SESSION;
+ }
+
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> track = mTracks[i];
+ if (sessionId == track->sessionId() &&
+ !(track->mCblk->flags & CBLK_INVALID)) {
+ result |= TRACK_SESSION;
+ break;
+ }
+ }
+
+ return result;
+}
+
+uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
+{
+ // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
+ // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
+ return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+ }
+ for (size_t i = 0; i < mTracks.size(); i++) {
+ sp<Track> track = mTracks[i];
+ if (sessionId == track->sessionId() &&
+ !(track->mCblk->flags & CBLK_INVALID)) {
+ return AudioSystem::getStrategyForStream(track->streamType());
+ }
+ }
+ return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+}
+
+
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+{
+ Mutex::Autolock _l(mLock);
+ return mOutput;
+}
+
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+{
+ Mutex::Autolock _l(mLock);
+ AudioStreamOut *output = mOutput;
+ mOutput = NULL;
+ // FIXME FastMixer might also have a raw ptr to mOutputSink;
+ // must push a NULL and wait for ack
+ mOutputSink.clear();
+ mPipeSink.clear();
+ mNormalSink.clear();
+ return output;
+}
+
+// this method must always be called either with ThreadBase mLock held or inside the thread loop
+audio_stream_t* AudioFlinger::PlaybackThread::stream() const
+{
+ if (mOutput == NULL) {
+ return NULL;
+ }
+ return &mOutput->stream->common;
+}
+
+uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
+{
+ return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
+}
+
+status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
+{
+ if (!isValidSyncEvent(event)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock _l(mLock);
+
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> track = mTracks[i];
+ if (event->triggerSession() == track->sessionId()) {
+ (void) track->setSyncEvent(event);
+ return NO_ERROR;
+ }
+ }
+
+ return NAME_NOT_FOUND;
+}
+
+bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+{
+ return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
+}
+
+void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
+ const Vector< sp<Track> >& tracksToRemove)
+{
+ size_t count = tracksToRemove.size();
+ if (CC_UNLIKELY(count)) {
+ for (size_t i = 0 ; i < count ; i++) {
+ const sp<Track>& track = tracksToRemove.itemAt(i);
+ if ((track->sharedBuffer() != 0) &&
+ (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) {
+ AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
+ }
+ }
+ }
+
+}
+
+void AudioFlinger::PlaybackThread::checkSilentMode_l()
+{
+ if (!mMasterMute) {
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("ro.audio.silent", value, "0") > 0) {
+ char *endptr;
+ unsigned long ul = strtoul(value, &endptr, 0);
+ if (*endptr == '\0' && ul != 0) {
+ ALOGD("Silence is golden");
+ // The setprop command will not allow a property to be changed after
+ // the first time it is set, so we don't have to worry about un-muting.
+ setMasterMute_l(true);
+ }
+ }
+ }
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_write()
+{
+ // FIXME rewrite to reduce number of system calls
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ int bytesWritten;
+
+ // If an NBAIO sink is present, use it to write the normal mixer's submix
+ if (mNormalSink != 0) {
+#define mBitShift 2 // FIXME
+ size_t count = mixBufferSize >> mBitShift;
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+ Tracer::traceBegin(ATRACE_TAG, "write");
+#endif
+ // update the setpoint when AudioFlinger::mScreenState changes
+ uint32_t screenState = AudioFlinger::mScreenState;
+ if (screenState != mScreenState) {
+ mScreenState = screenState;
+ MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+ if (pipe != NULL) {
+ pipe->setAvgFrames((mScreenState & 1) ?
+ (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+ }
+ }
+ ssize_t framesWritten = mNormalSink->write(mMixBuffer, count);
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+ Tracer::traceEnd(ATRACE_TAG);
+#endif
+ if (framesWritten > 0) {
+ bytesWritten = framesWritten << mBitShift;
+ } else {
+ bytesWritten = framesWritten;
+ }
+ // otherwise use the HAL / AudioStreamOut directly
+ } else {
+ // Direct output thread.
+ bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
+ }
+
+ if (bytesWritten > 0) {
+ mBytesWritten += mixBufferSize;
+ }
+ mNumWrites++;
+ mInWrite = false;
+}
+
+/*
+The derived values that are cached:
+ - mixBufferSize from frame count * frame size
+ - activeSleepTime from activeSleepTimeUs()
+ - idleSleepTime from idleSleepTimeUs()
+ - standbyDelay from mActiveSleepTimeUs (DIRECT only)
+ - maxPeriod from frame count and sample rate (MIXER only)
+
+The parameters that affect these derived values are:
+ - frame count
+ - frame size
+ - sample rate
+ - device type: A2DP or not
+ - device latency
+ - format: PCM or not
+ - active sleep time
+ - idle sleep time
+*/
+
+void AudioFlinger::PlaybackThread::cacheParameters_l()
+{
+ mixBufferSize = mNormalFrameCount * mFrameSize;
+ activeSleepTime = activeSleepTimeUs();
+ idleSleepTime = idleSleepTimeUs();
+}
+
+void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+{
+ ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
+ this, streamType, mTracks.size());
+ Mutex::Autolock _l(mLock);
+
+ size_t size = mTracks.size();
+ for (size_t i = 0; i < size; i++) {
+ sp<Track> t = mTracks[i];
+ if (t->streamType() == streamType) {
+ android_atomic_or(CBLK_INVALID, &t->mCblk->flags);
+ t->mCblk->cv.signal();
+ }
+ }
+}
+
+status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+ int session = chain->sessionId();
+ int16_t *buffer = mMixBuffer;
+ bool ownsBuffer = false;
+
+ ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
+ if (session > 0) {
+ // Only one effect chain can be present in direct output thread and it uses
+ // the mix buffer as input
+ if (mType != DIRECT) {
+ size_t numSamples = mNormalFrameCount * mChannelCount;
+ buffer = new int16_t[numSamples];
+ memset(buffer, 0, numSamples * sizeof(int16_t));
+ ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
+ ownsBuffer = true;
+ }
+
+ // Attach all tracks with same session ID to this chain.
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> track = mTracks[i];
+ if (session == track->sessionId()) {
+ ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
+ buffer);
+ track->setMainBuffer(buffer);
+ chain->incTrackCnt();
+ }
+ }
+
+ // indicate all active tracks in the chain
+ for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track == 0) {
+ continue;
+ }
+ if (session == track->sessionId()) {
+ ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
+ chain->incActiveTrackCnt();
+ }
+ }
+ }
+
+ chain->setInBuffer(buffer, ownsBuffer);
+ chain->setOutBuffer(mMixBuffer);
+ // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
+ // chains list in order to be processed last as it contains output stage effects
+ // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
+ // session AUDIO_SESSION_OUTPUT_STAGE to be processed
+ // after track specific effects and before output stage
+ // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
+ // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
+ // Effect chain for other sessions are inserted at beginning of effect
+ // chains list to be processed before output mix effects. Relative order between other
+ // sessions is not important
+ size_t size = mEffectChains.size();
+ size_t i = 0;
+ for (i = 0; i < size; i++) {
+ if (mEffectChains[i]->sessionId() < session) {
+ break;
+ }
+ }
+ mEffectChains.insertAt(chain, i);
+ checkSuspendOnAddEffectChain_l(chain);
+
+ return NO_ERROR;
+}
+
+size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+ int session = chain->sessionId();
+
+ ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
+
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ if (chain == mEffectChains[i]) {
+ mEffectChains.removeAt(i);
+ // detach all active tracks from the chain
+ for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track == 0) {
+ continue;
+ }
+ if (session == track->sessionId()) {
+ ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
+ chain.get(), session);
+ chain->decActiveTrackCnt();
+ }
+ }
+
+ // detach all tracks with same session ID from this chain
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> track = mTracks[i];
+ if (session == track->sessionId()) {
+ track->setMainBuffer(mMixBuffer);
+ chain->decTrackCnt();
+ }
+ }
+ break;
+ }
+ }
+ return mEffectChains.size();
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect(
+ const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+ Mutex::Autolock _l(mLock);
+ return attachAuxEffect_l(track, EffectId);
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
+ const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+ status_t status = NO_ERROR;
+
+ if (EffectId == 0) {
+ track->setAuxBuffer(0, NULL);
+ } else {
+ // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
+ sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+ if (effect != 0) {
+ if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+ track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
+ } else {
+ status = INVALID_OPERATION;
+ }
+ } else {
+ status = BAD_VALUE;
+ }
+ }
+ return status;
+}
+
+void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
+{
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> track = mTracks[i];
+ if (track->auxEffectId() == effectId) {
+ attachAuxEffect_l(track, 0);
+ }
+ }
+}
+
+bool AudioFlinger::PlaybackThread::threadLoop()
+{
+ Vector< sp<Track> > tracksToRemove;
+
+ standbyTime = systemTime();
+
+ // MIXER
+ nsecs_t lastWarning = 0;
+
+ // DUPLICATING
+ // FIXME could this be made local to while loop?
+ writeFrames = 0;
+
+ cacheParameters_l();
+ sleepTime = idleSleepTime;
+
+ if (mType == MIXER) {
+ sleepTimeShift = 0;
+ }
+
+ CpuStats cpuStats;
+ const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
+
+ acquireWakeLock();
+
+ while (!exitPending())
+ {
+ cpuStats.sample(myName);
+
+ Vector< sp<EffectChain> > effectChains;
+
+ processConfigEvents();
+
+ { // scope for mLock
+
+ Mutex::Autolock _l(mLock);
+
+ if (checkForNewParameters_l()) {
+ cacheParameters_l();
+ }
+
+ saveOutputTracks();
+
+ // put audio hardware into standby after short delay
+ if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ isSuspended())) {
+ if (!mStandby) {
+
+ threadLoop_standby();
+
+ mStandby = true;
+ }
+
+ if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+
+ clearOutputTracks();
+
+ if (exitPending()) {
+ break;
+ }
+
+ releaseWakeLock_l();
+ // wait until we have something to do...
+ ALOGV("%s going to sleep", myName.string());
+ mWaitWorkCV.wait(mLock);
+ ALOGV("%s waking up", myName.string());
+ acquireWakeLock_l();
+
+ mMixerStatus = MIXER_IDLE;
+ mMixerStatusIgnoringFastTracks = MIXER_IDLE;
+ mBytesWritten = 0;
+
+ checkSilentMode_l();
+
+ standbyTime = systemTime() + standbyDelay;
+ sleepTime = idleSleepTime;
+ if (mType == MIXER) {
+ sleepTimeShift = 0;
+ }
+
+ continue;
+ }
+ }
+
+ // mMixerStatusIgnoringFastTracks is also updated internally
+ mMixerStatus = prepareTracks_l(&tracksToRemove);
+
+ // prevent any changes in effect chain list and in each effect chain
+ // during mixing and effect process as the audio buffers could be deleted
+ // or modified if an effect is created or deleted
+ lockEffectChains_l(effectChains);
+ }
+
+ if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
+ threadLoop_mix();
+ } else {
+ threadLoop_sleepTime();
+ }
+
+ if (isSuspended()) {
+ sleepTime = suspendSleepTimeUs();
+ mBytesWritten += mixBufferSize;
+ }
+
+ // only process effects if we're going to write
+ if (sleepTime == 0) {
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ effectChains[i]->process_l();
+ }
+ }
+
+ // enable changes in effect chain
+ unlockEffectChains(effectChains);
+
+ // sleepTime == 0 means we must write to audio hardware
+ if (sleepTime == 0) {
+
+ threadLoop_write();
+
+if (mType == MIXER) {
+ // write blocked detection
+ nsecs_t now = systemTime();
+ nsecs_t delta = now - mLastWriteTime;
+ if (!mStandby && delta > maxPeriod) {
+ mNumDelayedWrites++;
+ if ((now - lastWarning) > kWarningThrottleNs) {
+#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
+ ScopedTrace st(ATRACE_TAG, "underrun");
+#endif
+ ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+ ns2ms(delta), mNumDelayedWrites, this);
+ lastWarning = now;
+ }
+ }
+}
+
+ mStandby = false;
+ } else {
+ usleep(sleepTime);
+ }
+
+ // Finally let go of removed track(s), without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock. This will also mutate and push a new fast mixer state.
+ threadLoop_removeTracks(tracksToRemove);
+ tracksToRemove.clear();
+
+ // FIXME I don't understand the need for this here;
+ // it was in the original code but maybe the
+ // assignment in saveOutputTracks() makes this unnecessary?
+ clearOutputTracks();
+
+ // Effect chains will be actually deleted here if they were removed from
+ // mEffectChains list during mixing or effects processing
+ effectChains.clear();
+
+ // FIXME Note that the above .clear() is no longer necessary since effectChains
+ // is now local to this block, but will keep it for now (at least until merge done).
+ }
+
+ // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ...
+ if (mType == MIXER || mType == DIRECT) {
+ // put output stream into standby mode
+ if (!mStandby) {
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ }
+ }
+
+ releaseWakeLock();
+
+ ALOGV("Thread %p type %d exiting", this, mType);
+ return false;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, audio_devices_t device, type_t type)
+ : PlaybackThread(audioFlinger, output, id, device, type),
+ // mAudioMixer below
+ // mFastMixer below
+ mFastMixerFutex(0)
+ // mOutputSink below
+ // mPipeSink below
+ // mNormalSink below
+{
+ ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
+ ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
+ "mFrameCount=%d, mNormalFrameCount=%d",
+ mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
+ mNormalFrameCount);
+ mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+
+ // FIXME - Current mixer implementation only supports stereo output
+ if (mChannelCount != FCC_2) {
+ ALOGE("Invalid audio hardware channel count %d", mChannelCount);
+ }
+
+ // create an NBAIO sink for the HAL output stream, and negotiate
+ mOutputSink = new AudioStreamOutSink(output->stream);
+ size_t numCounterOffers = 0;
+ const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
+ ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+
+ // initialize fast mixer depending on configuration
+ bool initFastMixer;
+ switch (kUseFastMixer) {
+ case FastMixer_Never:
+ initFastMixer = false;
+ break;
+ case FastMixer_Always:
+ initFastMixer = true;
+ break;
+ case FastMixer_Static:
+ case FastMixer_Dynamic:
+ initFastMixer = mFrameCount < mNormalFrameCount;
+ break;
+ }
+ if (initFastMixer) {
+
+ // create a MonoPipe to connect our submix to FastMixer
+ NBAIO_Format format = mOutputSink->format();
+ // This pipe depth compensates for scheduling latency of the normal mixer thread.
+ // When it wakes up after a maximum latency, it runs a few cycles quickly before
+ // finally blocking. Note the pipe implementation rounds up the request to a power of 2.
+ MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
+ const NBAIO_Format offers[1] = {format};
+ size_t numCounterOffers = 0;
+ ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ monoPipe->setAvgFrames((mScreenState & 1) ?
+ (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+ mPipeSink = monoPipe;
+
+#ifdef TEE_SINK_FRAMES
+ // create a Pipe to archive a copy of FastMixer's output for dumpsys
+ Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format);
+ numCounterOffers = 0;
+ index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ mTeeSink = teeSink;
+ PipeReader *teeSource = new PipeReader(*teeSink);
+ numCounterOffers = 0;
+ index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ mTeeSource = teeSource;
+#endif
+
+ // create fast mixer and configure it initially with just one fast track for our submix
+ mFastMixer = new FastMixer();
+ FastMixerStateQueue *sq = mFastMixer->sq();
+#ifdef STATE_QUEUE_DUMP
+ sq->setObserverDump(&mStateQueueObserverDump);
+ sq->setMutatorDump(&mStateQueueMutatorDump);
+#endif
+ FastMixerState *state = sq->begin();
+ FastTrack *fastTrack = &state->mFastTracks[0];
+ // wrap the source side of the MonoPipe to make it an AudioBufferProvider
+ fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
+ fastTrack->mVolumeProvider = NULL;
+ fastTrack->mGeneration++;
+ state->mFastTracksGen++;
+ state->mTrackMask = 1;
+ // fast mixer will use the HAL output sink
+ state->mOutputSink = mOutputSink.get();
+ state->mOutputSinkGen++;
+ state->mFrameCount = mFrameCount;
+ state->mCommand = FastMixerState::COLD_IDLE;
+ // already done in constructor initialization list
+ //mFastMixerFutex = 0;
+ state->mColdFutexAddr = &mFastMixerFutex;
+ state->mColdGen++;
+ state->mDumpState = &mFastMixerDumpState;
+ state->mTeeSink = mTeeSink.get();
+ sq->end();
+ sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+
+ // start the fast mixer
+ mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
+ pid_t tid = mFastMixer->getTid();
+ int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+ if (err != 0) {
+ ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+ kPriorityFastMixer, getpid_cached, tid, err);
+ }
+
+#ifdef AUDIO_WATCHDOG
+ // create and start the watchdog
+ mAudioWatchdog = new AudioWatchdog();
+ mAudioWatchdog->setDump(&mAudioWatchdogDump);
+ mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
+ tid = mAudioWatchdog->getTid();
+ err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+ if (err != 0) {
+ ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+ kPriorityFastMixer, getpid_cached, tid, err);
+ }
+#endif
+
+ } else {
+ mFastMixer = NULL;
+ }
+
+ switch (kUseFastMixer) {
+ case FastMixer_Never:
+ case FastMixer_Dynamic:
+ mNormalSink = mOutputSink;
+ break;
+ case FastMixer_Always:
+ mNormalSink = mPipeSink;
+ break;
+ case FastMixer_Static:
+ mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
+ break;
+ }
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+ if (mFastMixer != NULL) {
+ FastMixerStateQueue *sq = mFastMixer->sq();
+ FastMixerState *state = sq->begin();
+ if (state->mCommand == FastMixerState::COLD_IDLE) {
+ int32_t old = android_atomic_inc(&mFastMixerFutex);
+ if (old == -1) {
+ __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+ }
+ }
+ state->mCommand = FastMixerState::EXIT;
+ sq->end();
+ sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+ mFastMixer->join();
+ // Though the fast mixer thread has exited, it's state queue is still valid.
+ // We'll use that extract the final state which contains one remaining fast track
+ // corresponding to our sub-mix.
+ state = sq->begin();
+ ALOG_ASSERT(state->mTrackMask == 1);
+ FastTrack *fastTrack = &state->mFastTracks[0];
+ ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
+ delete fastTrack->mBufferProvider;
+ sq->end(false /*didModify*/);
+ delete mFastMixer;
+#ifdef AUDIO_WATCHDOG
+ if (mAudioWatchdog != 0) {
+ mAudioWatchdog->requestExit();
+ mAudioWatchdog->requestExitAndWait();
+ mAudioWatchdog.clear();
+ }
+#endif
+ }
+ delete mAudioMixer;
+}
+
+
+uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
+{
+ if (mFastMixer != NULL) {
+ MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+ latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
+ }
+ return latency;
+}
+
+
+void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
+{
+ PlaybackThread::threadLoop_removeTracks(tracksToRemove);
+}
+
+void AudioFlinger::MixerThread::threadLoop_write()
+{
+ // FIXME we should only do one push per cycle; confirm this is true
+ // Start the fast mixer if it's not already running
+ if (mFastMixer != NULL) {
+ FastMixerStateQueue *sq = mFastMixer->sq();
+ FastMixerState *state = sq->begin();
+ if (state->mCommand != FastMixerState::MIX_WRITE &&
+ (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
+ if (state->mCommand == FastMixerState::COLD_IDLE) {
+ int32_t old = android_atomic_inc(&mFastMixerFutex);
+ if (old == -1) {
+ __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+ }
+#ifdef AUDIO_WATCHDOG
+ if (mAudioWatchdog != 0) {
+ mAudioWatchdog->resume();
+ }
+#endif
+ }
+ state->mCommand = FastMixerState::MIX_WRITE;
+ sq->end();
+ sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+ if (kUseFastMixer == FastMixer_Dynamic) {
+ mNormalSink = mPipeSink;
+ }
+ } else {
+ sq->end(false /*didModify*/);
+ }
+ }
+ PlaybackThread::threadLoop_write();
+}
+
+void AudioFlinger::MixerThread::threadLoop_standby()
+{
+ // Idle the fast mixer if it's currently running
+ if (mFastMixer != NULL) {
+ FastMixerStateQueue *sq = mFastMixer->sq();
+ FastMixerState *state = sq->begin();
+ if (!(state->mCommand & FastMixerState::IDLE)) {
+ state->mCommand = FastMixerState::COLD_IDLE;
+ state->mColdFutexAddr = &mFastMixerFutex;
+ state->mColdGen++;
+ mFastMixerFutex = 0;
+ sq->end();
+ // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
+ sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
+ if (kUseFastMixer == FastMixer_Dynamic) {
+ mNormalSink = mOutputSink;
+ }
+#ifdef AUDIO_WATCHDOG
+ if (mAudioWatchdog != 0) {
+ mAudioWatchdog->pause();
+ }
+#endif
+ } else {
+ sq->end(false /*didModify*/);
+ }
+ }
+ PlaybackThread::threadLoop_standby();
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_standby()
+{
+ ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
+ mOutput->stream->common.standby(&mOutput->stream->common);
+}
+
+void AudioFlinger::MixerThread::threadLoop_mix()
+{
+ // obtain the presentation timestamp of the next output buffer
+ int64_t pts;
+ status_t status = INVALID_OPERATION;
+
+ if (mNormalSink != 0) {
+ status = mNormalSink->getNextWriteTimestamp(&pts);
+ } else {
+ status = mOutputSink->getNextWriteTimestamp(&pts);
+ }
+
+ if (status != NO_ERROR) {
+ pts = AudioBufferProvider::kInvalidPTS;
+ }
+
+ // mix buffers...
+ mAudioMixer->process(pts);
+ // increase sleep time progressively when application underrun condition clears.
+ // Only increase sleep time if the mixer is ready for two consecutive times to avoid
+ // that a steady state of alternating ready/not ready conditions keeps the sleep time
+ // such that we would underrun the audio HAL.
+ if ((sleepTime == 0) && (sleepTimeShift > 0)) {
+ sleepTimeShift--;
+ }
+ sleepTime = 0;
+ standbyTime = systemTime() + standbyDelay;
+ //TODO: delay standby when effects have a tail
+}
+
+void AudioFlinger::MixerThread::threadLoop_sleepTime()
+{
+ // If no tracks are ready, sleep once for the duration of an output
+ // buffer size, then write 0s to the output
+ if (sleepTime == 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ sleepTime = activeSleepTime >> sleepTimeShift;
+ if (sleepTime < kMinThreadSleepTimeUs) {
+ sleepTime = kMinThreadSleepTimeUs;
+ }
+ // reduce sleep time in case of consecutive application underruns to avoid
+ // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
+ // duration we would end up writing less data than needed by the audio HAL if
+ // the condition persists.
+ if (sleepTimeShift < kMaxThreadSleepTimeShift) {
+ sleepTimeShift++;
+ }
+ } else {
+ sleepTime = idleSleepTime;
+ }
+ } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
+ memset (mMixBuffer, 0, mixBufferSize);
+ sleepTime = 0;
+ ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
+ "anticipated start");
+ }
+ // TODO add standby time extension fct of effect tail
+}
+
+// prepareTracks_l() must be called with ThreadBase::mLock held
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+ Vector< sp<Track> > *tracksToRemove)
+{
+
+ mixer_state mixerStatus = MIXER_IDLE;
+ // find out which tracks need to be processed
+ size_t count = mActiveTracks.size();
+ size_t mixedTracks = 0;
+ size_t tracksWithEffect = 0;
+ // counts only _active_ fast tracks
+ size_t fastTracks = 0;
+ uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
+
+ float masterVolume = mMasterVolume;
+ bool masterMute = mMasterMute;
+
+ if (masterMute) {
+ masterVolume = 0;
+ }
+ // Delegate master volume control to effect in output mix effect chain if needed
+ sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
+ if (chain != 0) {
+ uint32_t v = (uint32_t)(masterVolume * (1 << 24));
+ chain->setVolume_l(&v, &v);
+ masterVolume = (float)((v + (1 << 23)) >> 24);
+ chain.clear();
+ }
+
+ // prepare a new state to push
+ FastMixerStateQueue *sq = NULL;
+ FastMixerState *state = NULL;
+ bool didModify = false;
+ FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
+ if (mFastMixer != NULL) {
+ sq = mFastMixer->sq();
+ state = sq->begin();
+ }
+
+ for (size_t i=0 ; i<count ; i++) {
+ sp<Track> t = mActiveTracks[i].promote();
+ if (t == 0) {
+ continue;
+ }
+
+ // this const just means the local variable doesn't change
+ Track* const track = t.get();
+
+ // process fast tracks
+ if (track->isFastTrack()) {
+
+ // It's theoretically possible (though unlikely) for a fast track to be created
+ // and then removed within the same normal mix cycle. This is not a problem, as
+ // the track never becomes active so it's fast mixer slot is never touched.
+ // The converse, of removing an (active) track and then creating a new track
+ // at the identical fast mixer slot within the same normal mix cycle,
+ // is impossible because the slot isn't marked available until the end of each cycle.
+ int j = track->mFastIndex;
+ ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks);
+ ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
+ FastTrack *fastTrack = &state->mFastTracks[j];
+
+ // Determine whether the track is currently in underrun condition,
+ // and whether it had a recent underrun.
+ FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
+ FastTrackUnderruns underruns = ftDump->mUnderruns;
+ uint32_t recentFull = (underruns.mBitFields.mFull -
+ track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
+ uint32_t recentPartial = (underruns.mBitFields.mPartial -
+ track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
+ uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
+ track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
+ uint32_t recentUnderruns = recentPartial + recentEmpty;
+ track->mObservedUnderruns = underruns;
+ // don't count underruns that occur while stopping or pausing
+ // or stopped which can occur when flush() is called while active
+ if (!(track->isStopping() || track->isPausing() || track->isStopped())) {
+ track->mUnderrunCount += recentUnderruns;
+ }
+
+ // This is similar to the state machine for normal tracks,
+ // with a few modifications for fast tracks.
+ bool isActive = true;
+ switch (track->mState) {
+ case TrackBase::STOPPING_1:
+ // track stays active in STOPPING_1 state until first underrun
+ if (recentUnderruns > 0) {
+ track->mState = TrackBase::STOPPING_2;
+ }
+ break;
+ case TrackBase::PAUSING:
+ // ramp down is not yet implemented
+ track->setPaused();
+ break;
+ case TrackBase::RESUMING:
+ // ramp up is not yet implemented
+ track->mState = TrackBase::ACTIVE;
+ break;
+ case TrackBase::ACTIVE:
+ if (recentFull > 0 || recentPartial > 0) {
+ // track has provided at least some frames recently: reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+ }
+ if (recentUnderruns == 0) {
+ // no recent underruns: stay active
+ break;
+ }
+ // there has recently been an underrun of some kind
+ if (track->sharedBuffer() == 0) {
+ // were any of the recent underruns "empty" (no frames available)?
+ if (recentEmpty == 0) {
+ // no, then ignore the partial underruns as they are allowed indefinitely
+ break;
+ }
+ // there has recently been an "empty" underrun: decrement the retry counter
+ if (--(track->mRetryCount) > 0) {
+ break;
+ }
+ // indicate to client process that the track was disabled because of underrun;
+ // it will then automatically call start() when data is available
+ android_atomic_or(CBLK_DISABLED, &track->mCblk->flags);
+ // remove from active list, but state remains ACTIVE [confusing but true]
+ isActive = false;
+ break;
+ }
+ // fall through
+ case TrackBase::STOPPING_2:
+ case TrackBase::PAUSED:
+ case TrackBase::TERMINATED:
+ case TrackBase::STOPPED:
+ case TrackBase::FLUSHED: // flush() while active
+ // Check for presentation complete if track is inactive
+ // We have consumed all the buffers of this track.
+ // This would be incomplete if we auto-paused on underrun
+ {
+ size_t audioHALFrames =
+ (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
+ size_t framesWritten = mBytesWritten / mFrameSize;
+ if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
+ // track stays in active list until presentation is complete
+ break;
+ }
+ }
+ if (track->isStopping_2()) {
+ track->mState = TrackBase::STOPPED;
+ }
+ if (track->isStopped()) {
+ // Can't reset directly, as fast mixer is still polling this track
+ // track->reset();
+ // So instead mark this track as needing to be reset after push with ack
+ resetMask |= 1 << i;
+ }
+ isActive = false;
+ break;
+ case TrackBase::IDLE:
+ default:
+ LOG_FATAL("unexpected track state %d", track->mState);
+ }
+
+ if (isActive) {
+ // was it previously inactive?
+ if (!(state->mTrackMask & (1 << j))) {
+ ExtendedAudioBufferProvider *eabp = track;
+ VolumeProvider *vp = track;
+ fastTrack->mBufferProvider = eabp;
+ fastTrack->mVolumeProvider = vp;
+ fastTrack->mSampleRate = track->mSampleRate;
+ fastTrack->mChannelMask = track->mChannelMask;
+ fastTrack->mGeneration++;
+ state->mTrackMask |= 1 << j;
+ didModify = true;
+ // no acknowledgement required for newly active tracks
+ }
+ // cache the combined master volume and stream type volume for fast mixer; this
+ // lacks any synchronization or barrier so VolumeProvider may read a stale value
+ track->mCachedVolume = track->isMuted() ?
+ 0 : masterVolume * mStreamTypes[track->streamType()].volume;
+ ++fastTracks;
+ } else {
+ // was it previously active?
+ if (state->mTrackMask & (1 << j)) {
+ fastTrack->mBufferProvider = NULL;
+ fastTrack->mGeneration++;
+ state->mTrackMask &= ~(1 << j);
+ didModify = true;
+ // If any fast tracks were removed, we must wait for acknowledgement
+ // because we're about to decrement the last sp<> on those tracks.
+ block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
+ } else {
+ LOG_FATAL("fast track %d should have been active", j);
+ }
+ tracksToRemove->add(track);
+ // Avoids a misleading display in dumpsys
+ track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
+ }
+ continue;
+ }
+
+ { // local variable scope to avoid goto warning
+
+ audio_track_cblk_t* cblk = track->cblk();
+
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ int name = track->name();
+ // make sure that we have enough frames to mix one full buffer.
+ // enforce this condition only once to enable draining the buffer in case the client
+ // app does not call stop() and relies on underrun to stop:
+ // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
+ // during last round
+ uint32_t minFrames = 1;
+ if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
+ (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
+ if (t->sampleRate() == mSampleRate) {
+ minFrames = mNormalFrameCount;
+ } else {
+ // +1 for rounding and +1 for additional sample needed for interpolation
+ minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+ // add frames already consumed but not yet released by the resampler
+ // because cblk->framesReady() will include these frames
+ minFrames += mAudioMixer->getUnreleasedFrames(track->name());
+ // the minimum track buffer size is normally twice the number of frames necessary
+ // to fill one buffer and the resampler should not leave more than one buffer worth
+ // of unreleased frames after each pass, but just in case...
+ ALOG_ASSERT(minFrames <= cblk->frameCount);
+ }
+ }
+ if ((track->framesReady() >= minFrames) && track->isReady() &&
+ !track->isPaused() && !track->isTerminated())
+ {
+ ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
+ this);
+
+ mixedTracks++;
+
+ // track->mainBuffer() != mMixBuffer means there is an effect chain
+ // connected to the track
+ chain.clear();
+ if (track->mainBuffer() != mMixBuffer) {
+ chain = getEffectChain_l(track->sessionId());
+ // Delegate volume control to effect in track effect chain if needed
+ if (chain != 0) {
+ tracksWithEffect++;
+ } else {
+ ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
+ "session %d",
+ name, track->sessionId());
+ }
+ }
+
+
+ int param = AudioMixer::VOLUME;
+ if (track->mFillingUpStatus == Track::FS_FILLED) {
+ // no ramp for the first volume setting
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ param = AudioMixer::RAMP_VOLUME;
+ }
+ mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
+ } else if (cblk->server != 0) {
+ // If the track is stopped before the first frame was mixed,
+ // do not apply ramp
+ param = AudioMixer::RAMP_VOLUME;
+ }
+
+ // compute volume for this track
+ uint32_t vl, vr, va;
+ if (track->isMuted() || track->isPausing() ||
+ mStreamTypes[track->streamType()].mute) {
+ vl = vr = va = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+
+ // read original volumes with volume control
+ float typeVolume = mStreamTypes[track->streamType()].volume;
+ float v = masterVolume * typeVolume;
+ uint32_t vlr = cblk->getVolumeLR();
+ vl = vlr & 0xFFFF;
+ vr = vlr >> 16;
+ // track volumes come from shared memory, so can't be trusted and must be clamped
+ if (vl > MAX_GAIN_INT) {
+ ALOGV("Track left volume out of range: %04X", vl);
+ vl = MAX_GAIN_INT;
+ }
+ if (vr > MAX_GAIN_INT) {
+ ALOGV("Track right volume out of range: %04X", vr);
+ vr = MAX_GAIN_INT;
+ }
+ // now apply the master volume and stream type volume
+ vl = (uint32_t)(v * vl) << 12;
+ vr = (uint32_t)(v * vr) << 12;
+ // assuming master volume and stream type volume each go up to 1.0,
+ // vl and vr are now in 8.24 format
+
+ uint16_t sendLevel = cblk->getSendLevel_U4_12();
+ // send level comes from shared memory and so may be corrupt
+ if (sendLevel > MAX_GAIN_INT) {
+ ALOGV("Track send level out of range: %04X", sendLevel);
+ sendLevel = MAX_GAIN_INT;
+ }
+ va = (uint32_t)(v * sendLevel);
+ }
+ // Delegate volume control to effect in track effect chain if needed
+ if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
+ // Do not ramp volume if volume is controlled by effect
+ param = AudioMixer::VOLUME;
+ track->mHasVolumeController = true;
+ } else {
+ // force no volume ramp when volume controller was just disabled or removed
+ // from effect chain to avoid volume spike
+ if (track->mHasVolumeController) {
+ param = AudioMixer::VOLUME;
+ }
+ track->mHasVolumeController = false;
+ }
+
+ // Convert volumes from 8.24 to 4.12 format
+ // This additional clamping is needed in case chain->setVolume_l() overshot
+ vl = (vl + (1 << 11)) >> 12;
+ if (vl > MAX_GAIN_INT) {
+ vl = MAX_GAIN_INT;
+ }
+ vr = (vr + (1 << 11)) >> 12;
+ if (vr > MAX_GAIN_INT) {
+ vr = MAX_GAIN_INT;
+ }
+
+ if (va > MAX_GAIN_INT) {
+ va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
+ }
+
+ // XXX: these things DON'T need to be done each time
+ mAudioMixer->setBufferProvider(name, track);
+ mAudioMixer->enable(name);
+
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
+ mAudioMixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::FORMAT, (void *)track->format());
+ mAudioMixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
+ mAudioMixer->setParameter(
+ name,
+ AudioMixer::RESAMPLE,
+ AudioMixer::SAMPLE_RATE,
+ (void *)(cblk->sampleRate));
+ mAudioMixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
+ mAudioMixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+
+ // If one track is ready, set the mixer ready if:
+ // - the mixer was not ready during previous round OR
+ // - no other track is not ready
+ if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
+ mixerStatus != MIXER_TRACKS_ENABLED) {
+ mixerStatus = MIXER_TRACKS_READY;
+ }
+ } else {
+ // clear effect chain input buffer if an active track underruns to avoid sending
+ // previous audio buffer again to effects
+ chain = getEffectChain_l(track->sessionId());
+ if (chain != 0) {
+ chain->clearInputBuffer();
+ }
+
+ ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user,
+ cblk->server, this);
+ if ((track->sharedBuffer() != 0) || track->isTerminated() ||
+ track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ // TODO: use actual buffer filling status instead of latency when available from
+ // audio HAL
+ size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+ size_t framesWritten = mBytesWritten / mFrameSize;
+ if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+ if (track->isStopped()) {
+ track->reset();
+ }
+ tracksToRemove->add(track);
+ }
+ } else {
+ track->mUnderrunCount++;
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
+ tracksToRemove->add(track);
+ // indicate to client process that the track was disabled because of underrun;
+ // it will then automatically call start() when data is available
+ android_atomic_or(CBLK_DISABLED, &cblk->flags);
+ // If one track is not ready, mark the mixer also not ready if:
+ // - the mixer was ready during previous round OR
+ // - no other track is ready
+ } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
+ mixerStatus != MIXER_TRACKS_READY) {
+ mixerStatus = MIXER_TRACKS_ENABLED;
+ }
+ }
+ mAudioMixer->disable(name);
+ }
+
+ } // local variable scope to avoid goto warning
+track_is_ready: ;
+
+ }
+
+ // Push the new FastMixer state if necessary
+ bool pauseAudioWatchdog = false;
+ if (didModify) {
+ state->mFastTracksGen++;
+ // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
+ if (kUseFastMixer == FastMixer_Dynamic &&
+ state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
+ state->mCommand = FastMixerState::COLD_IDLE;
+ state->mColdFutexAddr = &mFastMixerFutex;
+ state->mColdGen++;
+ mFastMixerFutex = 0;
+ if (kUseFastMixer == FastMixer_Dynamic) {
+ mNormalSink = mOutputSink;
+ }
+ // If we go into cold idle, need to wait for acknowledgement
+ // so that fast mixer stops doing I/O.
+ block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
+ pauseAudioWatchdog = true;
+ }
+ sq->end();
+ }
+ if (sq != NULL) {
+ sq->end(didModify);
+ sq->push(block);
+ }
+#ifdef AUDIO_WATCHDOG
+ if (pauseAudioWatchdog && mAudioWatchdog != 0) {
+ mAudioWatchdog->pause();
+ }
+#endif
+
+ // Now perform the deferred reset on fast tracks that have stopped
+ while (resetMask != 0) {
+ size_t i = __builtin_ctz(resetMask);
+ ALOG_ASSERT(i < count);
+ resetMask &= ~(1 << i);
+ sp<Track> t = mActiveTracks[i].promote();
+ if (t == 0) {
+ continue;
+ }
+ Track* track = t.get();
+ ALOG_ASSERT(track->isFastTrack() && track->isStopped());
+ track->reset();
+ }
+
+ // remove all the tracks that need to be...
+ count = tracksToRemove->size();
+ if (CC_UNLIKELY(count)) {
+ for (size_t i=0 ; i<count ; i++) {
+ const sp<Track>& track = tracksToRemove->itemAt(i);
+ mActiveTracks.remove(track);
+ if (track->mainBuffer() != mMixBuffer) {
+ chain = getEffectChain_l(track->sessionId());
+ if (chain != 0) {
+ ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
+ track->sessionId());
+ chain->decActiveTrackCnt();
+ }
+ }
+ if (track->isTerminated()) {
+ removeTrack_l(track);
+ }
+ }
+ }
+
+ // mix buffer must be cleared if all tracks are connected to an
+ // effect chain as in this case the mixer will not write to
+ // mix buffer and track effects will accumulate into it
+ if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
+ (mixedTracks == 0 && fastTracks > 0)) {
+ // FIXME as a performance optimization, should remember previous zero status
+ memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+ }
+
+ // if any fast tracks, then status is ready
+ mMixerStatusIgnoringFastTracks = mixerStatus;
+ if (fastTracks > 0) {
+ mixerStatus = MIXER_TRACKS_READY;
+ }
+ return mixerStatus;
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
+{
+ return mAudioMixer->getTrackName(channelMask, sessionId);
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::MixerThread::deleteTrackName_l(int name)
+{
+ ALOGV("remove track (%d) and delete from mixer", name);
+ mAudioMixer->deleteTrackName(name);
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
+{
+ // if !&IDLE, holds the FastMixer state to restore after new parameters processed
+ FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
+ bool reconfig = false;
+
+ while (!mNewParameters.isEmpty()) {
+
+ if (mFastMixer != NULL) {
+ FastMixerStateQueue *sq = mFastMixer->sq();
+ FastMixerState *state = sq->begin();
+ if (!(state->mCommand & FastMixerState::IDLE)) {
+ previousCommand = state->mCommand;
+ state->mCommand = FastMixerState::HOT_IDLE;
+ sq->end();
+ sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
+ } else {
+ sq->end(false /*didModify*/);
+ }
+ }
+
+ status_t status = NO_ERROR;
+ String8 keyValuePair = mNewParameters[0];
+ AudioParameter param = AudioParameter(keyValuePair);
+ int value;
+
+ if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+ if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ if (value != AUDIO_CHANNEL_OUT_STEREO) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be guaranteed
+ // if frame count is changed after track creation
+ if (!mTracks.isEmpty()) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+#ifdef ADD_BATTERY_DATA
+ // when changing the audio output device, call addBatteryData to notify
+ // the change
+ if (mOutDevice != value) {
+ uint32_t params = 0;
+ // check whether speaker is on
+ if (value & AUDIO_DEVICE_OUT_SPEAKER) {
+ params |= IMediaPlayerService::kBatteryDataSpeakerOn;
+ }
+
+ audio_devices_t deviceWithoutSpeaker
+ = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
+ // check if any other device (except speaker) is on
+ if (value & deviceWithoutSpeaker ) {
+ params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
+ }
+
+ if (params != 0) {
+ addBatteryData(params);
+ }
+ }
+#endif
+
+ // forward device change to effects that have requested to be
+ // aware of attached audio device.
+ mOutDevice = value;
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ mEffectChains[i]->setDevice_l(mOutDevice);
+ }
+ }
+
+ if (status == NO_ERROR) {
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ keyValuePair.string());
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ keyValuePair.string());
+ }
+ if (status == NO_ERROR && reconfig) {
+ delete mAudioMixer;
+ // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+ mAudioMixer = NULL;
+ readOutputParameters();
+ mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+ for (size_t i = 0; i < mTracks.size() ; i++) {
+ int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
+ if (name < 0) {
+ break;
+ }
+ mTracks[i]->mName = name;
+ // limit track sample rate to 2 x new output sample rate
+ if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
+ mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
+ }
+ }
+ sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+ // already timed out waiting for the status and will never signal the condition.
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+ }
+
+ if (!(previousCommand & FastMixerState::IDLE)) {
+ ALOG_ASSERT(mFastMixer != NULL);
+ FastMixerStateQueue *sq = mFastMixer->sq();
+ FastMixerState *state = sq->begin();
+ ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
+ state->mCommand = previousCommand;
+ sq->end();
+ sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
+ }
+
+ return reconfig;
+}
+
+
+void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ PlaybackThread::dumpInternals(fd, args);
+
+ snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
+ FastMixerDumpState copy = mFastMixerDumpState;
+ copy.dump(fd);
+
+#ifdef STATE_QUEUE_DUMP
+ // Similar for state queue
+ StateQueueObserverDump observerCopy = mStateQueueObserverDump;
+ observerCopy.dump(fd);
+ StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
+ mutatorCopy.dump(fd);
+#endif
+
+ // Write the tee output to a .wav file
+ dumpTee(fd, mTeeSource, mId);
+
+#ifdef AUDIO_WATCHDOG
+ if (mAudioWatchdog != 0) {
+ // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
+ AudioWatchdogDump wdCopy = mAudioWatchdogDump;
+ wdCopy.dump(fd);
+ }
+#endif
+}
+
+uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
+{
+ return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
+}
+
+uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
+{
+ return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
+}
+
+void AudioFlinger::MixerThread::cacheParameters_l()
+{
+ PlaybackThread::cacheParameters_l();
+
+ // FIXME: Relaxed timing because of a certain device that can't meet latency
+ // Should be reduced to 2x after the vendor fixes the driver issue
+ // increase threshold again due to low power audio mode. The way this warning
+ // threshold is calculated and its usefulness should be reconsidered anyway.
+ maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
+ : PlaybackThread(audioFlinger, output, id, device, DIRECT)
+ // mLeftVolFloat, mRightVolFloat
+{
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
+ Vector< sp<Track> > *tracksToRemove
+)
+{
+ sp<Track> trackToRemove;
+
+ mixer_state mixerStatus = MIXER_IDLE;
+
+ // find out which tracks need to be processed
+ if (mActiveTracks.size() != 0) {
+ sp<Track> t = mActiveTracks[0].promote();
+ // The track died recently
+ if (t == 0) {
+ return MIXER_IDLE;
+ }
+
+ Track* const track = t.get();
+ audio_track_cblk_t* cblk = track->cblk();
+
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ uint32_t minFrames;
+ if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
+ minFrames = mNormalFrameCount;
+ } else {
+ minFrames = 1;
+ }
+ if ((track->framesReady() >= minFrames) && track->isReady() &&
+ !track->isPaused() && !track->isTerminated())
+ {
+ ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+ if (track->mFillingUpStatus == Track::FS_FILLED) {
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ mLeftVolFloat = mRightVolFloat = 0;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ }
+ }
+
+ // compute volume for this track
+ float left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing() ||
+ mStreamTypes[track->streamType()].mute) {
+ left = right = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->streamType()].volume;
+ float v = mMasterVolume * typeVolume;
+ uint32_t vlr = cblk->getVolumeLR();
+ float v_clamped = v * (vlr & 0xFFFF);
+ if (v_clamped > MAX_GAIN) {
+ v_clamped = MAX_GAIN;
+ }
+ left = v_clamped/MAX_GAIN;
+ v_clamped = v * (vlr >> 16);
+ if (v_clamped > MAX_GAIN) {
+ v_clamped = MAX_GAIN;
+ }
+ right = v_clamped/MAX_GAIN;
+ }
+
+ if (left != mLeftVolFloat || right != mRightVolFloat) {
+ mLeftVolFloat = left;
+ mRightVolFloat = right;
+
+ // Convert volumes from float to 8.24
+ uint32_t vl = (uint32_t)(left * (1 << 24));
+ uint32_t vr = (uint32_t)(right * (1 << 24));
+
+ // Delegate volume control to effect in track effect chain if needed
+ // only one effect chain can be present on DirectOutputThread, so if
+ // there is one, the track is connected to it
+ if (!mEffectChains.isEmpty()) {
+ // Do not ramp volume if volume is controlled by effect
+ mEffectChains[0]->setVolume_l(&vl, &vr);
+ left = (float)vl / (1 << 24);
+ right = (float)vr / (1 << 24);
+ }
+ mOutput->stream->set_volume(mOutput->stream, left, right);
+ }
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetriesDirect;
+ mActiveTrack = t;
+ mixerStatus = MIXER_TRACKS_READY;
+ } else {
+ // clear effect chain input buffer if an active track underruns to avoid sending
+ // previous audio buffer again to effects
+ if (!mEffectChains.isEmpty()) {
+ mEffectChains[0]->clearInputBuffer();
+ }
+
+ ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+ if ((track->sharedBuffer() != 0) || track->isTerminated() ||
+ track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ // TODO: implement behavior for compressed audio
+ size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+ size_t framesWritten = mBytesWritten / mFrameSize;
+ if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
+ if (track->isStopped()) {
+ track->reset();
+ }
+ trackToRemove = track;
+ }
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ trackToRemove = track;
+ } else {
+ mixerStatus = MIXER_TRACKS_ENABLED;
+ }
+ }
+ }
+ }
+
+ // FIXME merge this with similar code for removing multiple tracks
+ // remove all the tracks that need to be...
+ if (CC_UNLIKELY(trackToRemove != 0)) {
+ tracksToRemove->add(trackToRemove);
+ mActiveTracks.remove(trackToRemove);
+ if (!mEffectChains.isEmpty()) {
+ ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(),
+ trackToRemove->sessionId());
+ mEffectChains[0]->decActiveTrackCnt();
+ }
+ if (trackToRemove->isTerminated()) {
+ removeTrack_l(trackToRemove);
+ }
+ }
+
+ return mixerStatus;
+}
+
+void AudioFlinger::DirectOutputThread::threadLoop_mix()
+{
+ AudioBufferProvider::Buffer buffer;
+ size_t frameCount = mFrameCount;
+ int8_t *curBuf = (int8_t *)mMixBuffer;
+ // output audio to hardware
+ while (frameCount) {
+ buffer.frameCount = frameCount;
+ mActiveTrack->getNextBuffer(&buffer);
+ if (CC_UNLIKELY(buffer.raw == NULL)) {
+ memset(curBuf, 0, frameCount * mFrameSize);
+ break;
+ }
+ memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+ frameCount -= buffer.frameCount;
+ curBuf += buffer.frameCount * mFrameSize;
+ mActiveTrack->releaseBuffer(&buffer);
+ }
+ sleepTime = 0;
+ standbyTime = systemTime() + standbyDelay;
+ mActiveTrack.clear();
+
+}
+
+void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
+{
+ if (sleepTime == 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ sleepTime = activeSleepTime;
+ } else {
+ sleepTime = idleSleepTime;
+ }
+ } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
+ memset(mMixBuffer, 0, mFrameCount * mFrameSize);
+ sleepTime = 0;
+ }
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
+ int sessionId)
+{
+ return 0;
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+{
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
+{
+ bool reconfig = false;
+
+ while (!mNewParameters.isEmpty()) {
+ status_t status = NO_ERROR;
+ String8 keyValuePair = mNewParameters[0];
+ AudioParameter param = AudioParameter(keyValuePair);
+ int value;
+
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (!mTracks.isEmpty()) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ keyValuePair.string());
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ keyValuePair.string());
+ }
+ if (status == NO_ERROR && reconfig) {
+ readOutputParameters();
+ sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+ // already timed out waiting for the status and will never signal the condition.
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+ }
+ return reconfig;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
+{
+ uint32_t time;
+ if (audio_is_linear_pcm(mFormat)) {
+ time = PlaybackThread::activeSleepTimeUs();
+ } else {
+ time = 10000;
+ }
+ return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
+{
+ uint32_t time;
+ if (audio_is_linear_pcm(mFormat)) {
+ time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
+ } else {
+ time = 10000;
+ }
+ return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
+{
+ uint32_t time;
+ if (audio_is_linear_pcm(mFormat)) {
+ time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
+ } else {
+ time = 10000;
+ }
+ return time;
+}
+
+void AudioFlinger::DirectOutputThread::cacheParameters_l()
+{
+ PlaybackThread::cacheParameters_l();
+
+ // use shorter standby delay as on normal output to release
+ // hardware resources as soon as possible
+ standbyDelay = microseconds(activeSleepTime*2);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+ AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
+ : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
+ DUPLICATING),
+ mWaitTimeMs(UINT_MAX)
+{
+ addOutputTrack(mainThread);
+}
+
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
+{
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ mOutputTracks[i]->destroy();
+ }
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_mix()
+{
+ // mix buffers...
+ if (outputsReady(outputTracks)) {
+ mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
+ } else {
+ memset(mMixBuffer, 0, mixBufferSize);
+ }
+ sleepTime = 0;
+ writeFrames = mNormalFrameCount;
+ standbyTime = systemTime() + standbyDelay;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
+{
+ if (sleepTime == 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ sleepTime = activeSleepTime;
+ } else {
+ sleepTime = idleSleepTime;
+ }
+ } else if (mBytesWritten != 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ writeFrames = mNormalFrameCount;
+ memset(mMixBuffer, 0, mixBufferSize);
+ } else {
+ // flush remaining overflow buffers in output tracks
+ writeFrames = 0;
+ }
+ sleepTime = 0;
+ }
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_write()
+{
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->write(mMixBuffer, writeFrames);
+ }
+ mBytesWritten += mixBufferSize;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_standby()
+{
+ // DuplicatingThread implements standby by stopping all tracks
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->stop();
+ }
+}
+
+void AudioFlinger::DuplicatingThread::saveOutputTracks()
+{
+ outputTracks = mOutputTracks;
+}
+
+void AudioFlinger::DuplicatingThread::clearOutputTracks()
+{
+ outputTracks.clear();
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+ Mutex::Autolock _l(mLock);
+ // FIXME explain this formula
+ size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
+ OutputTrack *outputTrack = new OutputTrack(thread,
+ this,
+ mSampleRate,
+ mFormat,
+ mChannelMask,
+ frameCount);
+ if (outputTrack->cblk() != NULL) {
+ thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
+ mOutputTracks.add(outputTrack);
+ ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+ updateWaitTime_l();
+ }
+}
+
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+{
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ if (mOutputTracks[i]->thread() == thread) {
+ mOutputTracks[i]->destroy();
+ mOutputTracks.removeAt(i);
+ updateWaitTime_l();
+ return;
+ }
+ }
+ ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
+}
+
+// caller must hold mLock
+void AudioFlinger::DuplicatingThread::updateWaitTime_l()
+{
+ mWaitTimeMs = UINT_MAX;
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
+ if (strong != 0) {
+ uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
+ if (waitTimeMs < mWaitTimeMs) {
+ mWaitTimeMs = waitTimeMs;
+ }
+ }
+ }
+}
+
+
+bool AudioFlinger::DuplicatingThread::outputsReady(
+ const SortedVector< sp<OutputTrack> > &outputTracks)
+{
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ sp<ThreadBase> thread = outputTracks[i]->thread().promote();
+ if (thread == 0) {
+ ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
+ outputTracks[i].get());
+ return false;
+ }
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ // see note at standby() declaration
+ if (playbackThread->standby() && !playbackThread->isSuspended()) {
+ ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
+ thread.get());
+ return false;
+ }
+ }
+ return true;
+}
+
+uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
+{
+ return (mWaitTimeMs * 1000) / 2;
+}
+
+void AudioFlinger::DuplicatingThread::cacheParameters_l()
+{
+ // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
+ updateWaitTime_l();
+
+ MixerThread::cacheParameters_l();
+}
+
+// ----------------------------------------------------------------------------
+// Record
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamIn *input,
+ uint32_t sampleRate,
+ audio_channel_mask_t channelMask,
+ audio_io_handle_t id,
+ audio_devices_t device,
+ const sp<NBAIO_Sink>& teeSink) :
+ ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD),
+ mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+ // mRsmpInIndex and mInputBytes set by readInputParameters()
+ mReqChannelCount(popcount(channelMask)),
+ mReqSampleRate(sampleRate),
+ // mBytesRead is only meaningful while active, and so is cleared in start()
+ // (but might be better to also clear here for dump?)
+ mTeeSink(teeSink)
+{
+ snprintf(mName, kNameLength, "AudioIn_%X", id);
+
+ readInputParameters();
+
+}
+
+
+AudioFlinger::RecordThread::~RecordThread()
+{
+ delete[] mRsmpInBuffer;
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
+}
+
+void AudioFlinger::RecordThread::onFirstRef()
+{
+ run(mName, PRIORITY_URGENT_AUDIO);
+}
+
+status_t AudioFlinger::RecordThread::readyToRun()
+{
+ status_t status = initCheck();
+ ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
+ return status;
+}
+
+bool AudioFlinger::RecordThread::threadLoop()
+{
+ AudioBufferProvider::Buffer buffer;
+ sp<RecordTrack> activeTrack;
+ Vector< sp<EffectChain> > effectChains;
+
+ nsecs_t lastWarning = 0;
+
+ inputStandBy();
+ acquireWakeLock();
+
+ // used to verify we've read at least once before evaluating how many bytes were read
+ bool readOnce = false;
+
+ // start recording
+ while (!exitPending()) {
+
+ processConfigEvents();
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+ checkForNewParameters_l();
+ if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+ standby();
+
+ if (exitPending()) {
+ break;
+ }
+
+ releaseWakeLock_l();
+ ALOGV("RecordThread: loop stopping");
+ // go to sleep
+ mWaitWorkCV.wait(mLock);
+ ALOGV("RecordThread: loop starting");
+ acquireWakeLock_l();
+ continue;
+ }
+ if (mActiveTrack != 0) {
+ if (mActiveTrack->mState == TrackBase::PAUSING) {
+ standby();
+ mActiveTrack.clear();
+ mStartStopCond.broadcast();
+ } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+ if (mReqChannelCount != mActiveTrack->channelCount()) {
+ mActiveTrack.clear();
+ mStartStopCond.broadcast();
+ } else if (readOnce) {
+ // record start succeeds only if first read from audio input
+ // succeeds
+ if (mBytesRead >= 0) {
+ mActiveTrack->mState = TrackBase::ACTIVE;
+ } else {
+ mActiveTrack.clear();
+ }
+ mStartStopCond.broadcast();
+ }
+ mStandby = false;
+ } else if (mActiveTrack->mState == TrackBase::TERMINATED) {
+ removeTrack_l(mActiveTrack);
+ mActiveTrack.clear();
+ }
+ }
+ lockEffectChains_l(effectChains);
+ }
+
+ if (mActiveTrack != 0) {
+ if (mActiveTrack->mState != TrackBase::ACTIVE &&
+ mActiveTrack->mState != TrackBase::RESUMING) {
+ unlockEffectChains(effectChains);
+ usleep(kRecordThreadSleepUs);
+ continue;
+ }
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ effectChains[i]->process_l();
+ }
+
+ buffer.frameCount = mFrameCount;
+ if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ readOnce = true;
+ size_t framesOut = buffer.frameCount;
+ if (mResampler == NULL) {
+ // no resampling
+ while (framesOut) {
+ size_t framesIn = mFrameCount - mRsmpInIndex;
+ if (framesIn) {
+ int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+ int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
+ mActiveTrack->mFrameSize;
+ if (framesIn > framesOut)
+ framesIn = framesOut;
+ mRsmpInIndex += framesIn;
+ framesOut -= framesIn;
+ if (mChannelCount == mReqChannelCount ||
+ mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+ memcpy(dst, src, framesIn * mFrameSize);
+ } else {
+ if (mChannelCount == 1) {
+ upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
+ (int16_t *)src, framesIn);
+ } else {
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+ (int16_t *)src, framesIn);
+ }
+ }
+ }
+ if (framesOut && mFrameCount == mRsmpInIndex) {
+ void *readInto;
+ if (framesOut == mFrameCount &&
+ (mChannelCount == mReqChannelCount ||
+ mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+ readInto = buffer.raw;
+ framesOut = 0;
+ } else {
+ readInto = mRsmpInBuffer;
+ mRsmpInIndex = 0;
+ }
+ mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes);
+ if (mBytesRead <= 0) {
+ if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
+ {
+ ALOGE("Error reading audio input");
+ // Force input into standby so that it tries to
+ // recover at next read attempt
+ inputStandBy();
+ usleep(kRecordThreadSleepUs);
+ }
+ mRsmpInIndex = mFrameCount;
+ framesOut = 0;
+ buffer.frameCount = 0;
+ } else if (mTeeSink != 0) {
+ (void) mTeeSink->write(readInto,
+ mBytesRead >> Format_frameBitShift(mTeeSink->format()));
+ }
+ }
+ }
+ } else {
+ // resampling
+
+ memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+ // alter output frame count as if we were expecting stereo samples
+ if (mChannelCount == 1 && mReqChannelCount == 1) {
+ framesOut >>= 1;
+ }
+ mResampler->resample(mRsmpOutBuffer, framesOut,
+ this /* AudioBufferProvider* */);
+ // ditherAndClamp() works as long as all buffers returned by
+ // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
+ if (mChannelCount == 2 && mReqChannelCount == 1) {
+ ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ // the resampler always outputs stereo samples:
+ // do post stereo to mono conversion
+ downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
+ framesOut);
+ } else {
+ ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ }
+
+ }
+ if (mFramestoDrop == 0) {
+ mActiveTrack->releaseBuffer(&buffer);
+ } else {
+ if (mFramestoDrop > 0) {
+ mFramestoDrop -= buffer.frameCount;
+ if (mFramestoDrop <= 0) {
+ clearSyncStartEvent();
+ }
+ } else {
+ mFramestoDrop += buffer.frameCount;
+ if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
+ mSyncStartEvent->isCancelled()) {
+ ALOGW("Synced record %s, session %d, trigger session %d",
+ (mFramestoDrop >= 0) ? "timed out" : "cancelled",
+ mActiveTrack->sessionId(),
+ (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
+ clearSyncStartEvent();
+ }
+ }
+ }
+ mActiveTrack->clearOverflow();
+ }
+ // client isn't retrieving buffers fast enough
+ else {
+ if (!mActiveTrack->setOverflow()) {
+ nsecs_t now = systemTime();
+ if ((now - lastWarning) > kWarningThrottleNs) {
+ ALOGW("RecordThread: buffer overflow");
+ lastWarning = now;
+ }
+ }
+ // Release the processor for a while before asking for a new buffer.
+ // This will give the application more chance to read from the buffer and
+ // clear the overflow.
+ usleep(kRecordThreadSleepUs);
+ }
+ }
+ // enable changes in effect chain
+ unlockEffectChains(effectChains);
+ effectChains.clear();
+ }
+
+ standby();
+
+ {
+ Mutex::Autolock _l(mLock);
+ mActiveTrack.clear();
+ mStartStopCond.broadcast();
+ }
+
+ releaseWakeLock();
+
+ ALOGV("RecordThread %p exiting", this);
+ return false;
+}
+
+void AudioFlinger::RecordThread::standby()
+{
+ if (!mStandby) {
+ inputStandBy();
+ mStandby = true;
+ }
+}
+
+void AudioFlinger::RecordThread::inputStandBy()
+{
+ mInput->stream->common.standby(&mInput->stream->common);
+}
+
+sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
+ const sp<AudioFlinger::Client>& client,
+ uint32_t sampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t frameCount,
+ int sessionId,
+ IAudioFlinger::track_flags_t flags,
+ pid_t tid,
+ status_t *status)
+{
+ sp<RecordTrack> track;
+ status_t lStatus;
+
+ lStatus = initCheck();
+ if (lStatus != NO_ERROR) {
+ ALOGE("Audio driver not initialized.");
+ goto Exit;
+ }
+
+ // FIXME use flags and tid similar to createTrack_l()
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+
+ track = new RecordTrack(this, client, sampleRate,
+ format, channelMask, frameCount, sessionId);
+
+ if (track->getCblk() == 0) {
+ lStatus = NO_MEMORY;
+ goto Exit;
+ }
+ mTracks.add(track);
+
+ // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
+ bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+ mAudioFlinger->btNrecIsOff();
+ setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
+ setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
+ }
+ lStatus = NO_ERROR;
+
+Exit:
+ if (status) {
+ *status = lStatus;
+ }
+ return track;
+}
+
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
+ AudioSystem::sync_event_t event,
+ int triggerSession)
+{
+ ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
+ sp<ThreadBase> strongMe = this;
+ status_t status = NO_ERROR;
+
+ if (event == AudioSystem::SYNC_EVENT_NONE) {
+ clearSyncStartEvent();
+ } else if (event != AudioSystem::SYNC_EVENT_SAME) {
+ mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
+ triggerSession,
+ recordTrack->sessionId(),
+ syncStartEventCallback,
+ this);
+ // Sync event can be cancelled by the trigger session if the track is not in a
+ // compatible state in which case we start record immediately
+ if (mSyncStartEvent->isCancelled()) {
+ clearSyncStartEvent();
+ } else {
+ // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
+ mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
+ }
+ }
+
+ {
+ AutoMutex lock(mLock);
+ if (mActiveTrack != 0) {
+ if (recordTrack != mActiveTrack.get()) {
+ status = -EBUSY;
+ } else if (mActiveTrack->mState == TrackBase::PAUSING) {
+ mActiveTrack->mState = TrackBase::ACTIVE;
+ }
+ return status;
+ }
+
+ recordTrack->mState = TrackBase::IDLE;
+ mActiveTrack = recordTrack;
+ mLock.unlock();
+ status_t status = AudioSystem::startInput(mId);
+ mLock.lock();
+ if (status != NO_ERROR) {
+ mActiveTrack.clear();
+ clearSyncStartEvent();
+ return status;
+ }
+ mRsmpInIndex = mFrameCount;
+ mBytesRead = 0;
+ if (mResampler != NULL) {
+ mResampler->reset();
+ }
+ mActiveTrack->mState = TrackBase::RESUMING;
+ // signal thread to start
+ ALOGV("Signal record thread");
+ mWaitWorkCV.broadcast();
+ // do not wait for mStartStopCond if exiting
+ if (exitPending()) {
+ mActiveTrack.clear();
+ status = INVALID_OPERATION;
+ goto startError;
+ }
+ mStartStopCond.wait(mLock);
+ if (mActiveTrack == 0) {
+ ALOGV("Record failed to start");
+ status = BAD_VALUE;
+ goto startError;
+ }
+ ALOGV("Record started OK");
+ return status;
+ }
+startError:
+ AudioSystem::stopInput(mId);
+ clearSyncStartEvent();
+ return status;
+}
+
+void AudioFlinger::RecordThread::clearSyncStartEvent()
+{
+ if (mSyncStartEvent != 0) {
+ mSyncStartEvent->cancel();
+ }
+ mSyncStartEvent.clear();
+ mFramestoDrop = 0;
+}
+
+void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
+{
+ sp<SyncEvent> strongEvent = event.promote();
+
+ if (strongEvent != 0) {
+ RecordThread *me = (RecordThread *)strongEvent->cookie();
+ me->handleSyncStartEvent(strongEvent);
+ }
+}
+
+void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event)
+{
+ if (event == mSyncStartEvent) {
+ // TODO: use actual buffer filling status instead of 2 buffers when info is available
+ // from audio HAL
+ mFramestoDrop = mFrameCount * 2;
+ }
+}
+
+bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) {
+ ALOGV("RecordThread::stop");
+ if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
+ return false;
+ }
+ recordTrack->mState = TrackBase::PAUSING;
+ // do not wait for mStartStopCond if exiting
+ if (exitPending()) {
+ return true;
+ }
+ mStartStopCond.wait(mLock);
+ // if we have been restarted, recordTrack == mActiveTrack.get() here
+ if (exitPending() || recordTrack != mActiveTrack.get()) {
+ ALOGV("Record stopped OK");
+ return true;
+ }
+ return false;
+}
+
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+{
+ return false;
+}
+
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+{
+#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
+ if (!isValidSyncEvent(event)) {
+ return BAD_VALUE;
+ }
+
+ int eventSession = event->triggerSession();
+ status_t ret = NAME_NOT_FOUND;
+
+ Mutex::Autolock _l(mLock);
+
+ for (size_t i = 0; i < mTracks.size(); i++) {
+ sp<RecordTrack> track = mTracks[i];
+ if (eventSession == track->sessionId()) {
+ (void) track->setSyncEvent(event);
+ ret = NO_ERROR;
+ }
+ }
+ return ret;
+#else
+ return BAD_VALUE;
+#endif
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
+{
+ track->mState = TrackBase::TERMINATED;
+ // active tracks are removed by threadLoop()
+ if (mActiveTrack != track) {
+ removeTrack_l(track);
+ }
+}
+
+void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
+{
+ mTracks.remove(track);
+ // need anything related to effects here?
+}
+
+void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
+{
+ dumpInternals(fd, args);
+ dumpTracks(fd, args);
+ dumpEffectChains(fd, args);
+}
+
+void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
+ result.append(buffer);
+
+ if (mActiveTrack != 0) {
+ snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
+ result.append(buffer);
+ } else {
+ result.append("No active record client\n");
+ }
+
+ write(fd, result.string(), result.size());
+
+ dumpBase(fd, args);
+}
+
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
+ result.append(buffer);
+ RecordTrack::appendDumpHeader(result);
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<RecordTrack> track = mTracks[i];
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+
+ if (mActiveTrack != 0) {
+ snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
+ result.append(buffer);
+ RecordTrack::appendDumpHeader(result);
+ mActiveTrack->dump(buffer, SIZE);
+ result.append(buffer);
+
+ }
+ write(fd, result.string(), result.size());
+}
+
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+ size_t framesReq = buffer->frameCount;
+ size_t framesReady = mFrameCount - mRsmpInIndex;
+ int channelCount;
+
+ if (framesReady == 0) {
+ mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
+ if (mBytesRead <= 0) {
+ if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
+ ALOGE("RecordThread::getNextBuffer() Error reading audio input");
+ // Force input into standby so that it tries to
+ // recover at next read attempt
+ inputStandBy();
+ usleep(kRecordThreadSleepUs);
+ }
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+ mRsmpInIndex = 0;
+ framesReady = mFrameCount;
+ }
+
+ if (framesReq > framesReady) {
+ framesReq = framesReady;
+ }
+
+ if (mChannelCount == 1 && mReqChannelCount == 2) {
+ channelCount = 1;
+ } else {
+ channelCount = 2;
+ }
+ buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+}
+
+// AudioBufferProvider interface
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ mRsmpInIndex += buffer->frameCount;
+ buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+ bool reconfig = false;
+
+ while (!mNewParameters.isEmpty()) {
+ status_t status = NO_ERROR;
+ String8 keyValuePair = mNewParameters[0];
+ AudioParameter param = AudioParameter(keyValuePair);
+ int value;
+ audio_format_t reqFormat = mFormat;
+ uint32_t reqSamplingRate = mReqSampleRate;
+ uint32_t reqChannelCount = mReqChannelCount;
+
+ if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+ reqSamplingRate = value;
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+ reqFormat = (audio_format_t) value;
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ reqChannelCount = popcount(value);
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be guaranteed
+ // if frame count is changed after track creation
+ if (mActiveTrack != 0) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+ // forward device change to effects that have requested to be
+ // aware of attached audio device.
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ mEffectChains[i]->setDevice_l(value);
+ }
+
+ // store input device and output device but do not forward output device to audio HAL.
+ // Note that status is ignored by the caller for output device
+ // (see AudioFlinger::setParameters()
+ if (audio_is_output_devices(value)) {
+ mOutDevice = value;
+ status = BAD_VALUE;
+ } else {
+ mInDevice = value;
+ // disable AEC and NS if the device is a BT SCO headset supporting those
+ // pre processings
+ if (mTracks.size() > 0) {
+ bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+ mAudioFlinger->btNrecIsOff();
+ for (size_t i = 0; i < mTracks.size(); i++) {
+ sp<RecordTrack> track = mTracks[i];
+ setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
+ setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
+ }
+ }
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
+ mAudioSource != (audio_source_t)value) {
+ // forward device change to effects that have requested to be
+ // aware of attached audio device.
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ mEffectChains[i]->setAudioSource_l((audio_source_t)value);
+ }
+ mAudioSource = (audio_source_t)value;
+ }
+ if (status == NO_ERROR) {
+ status = mInput->stream->common.set_parameters(&mInput->stream->common,
+ keyValuePair.string());
+ if (status == INVALID_OPERATION) {
+ inputStandBy();
+ status = mInput->stream->common.set_parameters(&mInput->stream->common,
+ keyValuePair.string());
+ }
+ if (reconfig) {
+ if (status == BAD_VALUE &&
+ reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
+ reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
+ ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common)
+ <= (2 * reqSamplingRate)) &&
+ popcount(mInput->stream->common.get_channels(&mInput->stream->common))
+ <= FCC_2 &&
+ (reqChannelCount <= FCC_2)) {
+ status = NO_ERROR;
+ }
+ if (status == NO_ERROR) {
+ readInputParameters();
+ sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
+ }
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ // wait for condition with time out in case the thread calling ThreadBase::setParameters()
+ // already timed out waiting for the status and will never signal the condition.
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
+ }
+ return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+ char *s;
+ String8 out_s8 = String8();
+
+ Mutex::Autolock _l(mLock);
+ if (initCheck() != NO_ERROR) {
+ return out_s8;
+ }
+
+ s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
+ out_s8 = String8(s);
+ free(s);
+ return out_s8;
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = NULL;
+
+ switch (event) {
+ case AudioSystem::INPUT_OPENED:
+ case AudioSystem::INPUT_CONFIG_CHANGED:
+ desc.channels = mChannelMask;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mFrameCount;
+ desc.latency = 0;
+ param2 = &desc;
+ break;
+
+ case AudioSystem::INPUT_CLOSED:
+ default:
+ break;
+ }
+ mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+ delete mRsmpInBuffer;
+ // mRsmpInBuffer is always assigned a new[] below
+ delete mRsmpOutBuffer;
+ mRsmpOutBuffer = NULL;
+ delete mResampler;
+ mResampler = NULL;
+
+ mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
+ mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
+ mChannelCount = (uint16_t)popcount(mChannelMask);
+ mFormat = mInput->stream->common.get_format(&mInput->stream->common);
+ mFrameSize = audio_stream_frame_size(&mInput->stream->common);
+ mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
+ mFrameCount = mInputBytes / mFrameSize;
+ mNormalFrameCount = mFrameCount; // not used by record, but used by input effects
+ mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+ if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
+ {
+ int channelCount;
+ // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+ // stereo to mono post process as the resampler always outputs stereo.
+ if (mChannelCount == 1 && mReqChannelCount == 2) {
+ channelCount = 1;
+ } else {
+ channelCount = 2;
+ }
+ mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+ mResampler->setSampleRate(mSampleRate);
+ mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+ mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+ // optmization: if mono to mono, alter input frame count as if we were inputing
+ // stereo samples
+ if (mChannelCount == 1 && mReqChannelCount == 1) {
+ mFrameCount >>= 1;
+ }
+
+ }
+ mRsmpInIndex = mFrameCount;
+}
+
+unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+{
+ Mutex::Autolock _l(mLock);
+ if (initCheck() != NO_ERROR) {
+ return 0;
+ }
+
+ return mInput->stream->get_input_frames_lost(mInput->stream);
+}
+
+uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
+{
+ Mutex::Autolock _l(mLock);
+ uint32_t result = 0;
+ if (getEffectChain_l(sessionId) != 0) {
+ result = EFFECT_SESSION;
+ }
+
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ if (sessionId == mTracks[i]->sessionId()) {
+ result |= TRACK_SESSION;
+ break;
+ }
+ }
+
+ return result;
+}
+
+KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
+{
+ KeyedVector<int, bool> ids;
+ Mutex::Autolock _l(mLock);
+ for (size_t j = 0; j < mTracks.size(); ++j) {
+ sp<RecordThread::RecordTrack> track = mTracks[j];
+ int sessionId = track->sessionId();
+ if (ids.indexOfKey(sessionId) < 0) {
+ ids.add(sessionId, true);
+ }
+ }
+ return ids;
+}
+
+AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
+{
+ Mutex::Autolock _l(mLock);
+ AudioStreamIn *input = mInput;
+ mInput = NULL;
+ return input;
+}
+
+// this method must always be called either with ThreadBase mLock held or inside the thread loop
+audio_stream_t* AudioFlinger::RecordThread::stream() const
+{
+ if (mInput == NULL) {
+ return NULL;
+ }
+ return &mInput->stream->common;
+}
+
+status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+ // only one chain per input thread
+ if (mEffectChains.size() != 0) {
+ return INVALID_OPERATION;
+ }
+ ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
+
+ chain->setInBuffer(NULL);
+ chain->setOutBuffer(NULL);
+
+ checkSuspendOnAddEffectChain_l(chain);
+
+ mEffectChains.add(chain);
+
+ return NO_ERROR;
+}
+
+size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+ ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
+ ALOGW_IF(mEffectChains.size() != 1,
+ "removeEffectChain_l() %p invalid chain size %d on thread %p",
+ chain.get(), mEffectChains.size(), this);
+ if (mEffectChains.size() == 1) {
+ mEffectChains.removeAt(0);
+ }
+ return 0;
+}
+
+}; // namespace android