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/*
**
** 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
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include "Configuration.h"
#include <math.h>
#include <fcntl.h>
#include <sys/stat.h>
#include <cutils/properties.h>
#include <media/AudioParameter.h>
#include <utils/Log.h>
#include <utils/Trace.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"
#ifdef ADD_BATTERY_DATA
#include <media/IMediaPlayerService.h>
#include <media/IMediaDeathNotifier.h>
#endif
#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;
// Offloaded output thread standby delay: allows track transition without going to standby
static const nsecs_t kOffloadStandbyDelayNs = seconds(1);
// 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 N-buffering by default, but doesn't tell us about the value of N.
// So for now we just assume that client is double-buffered for fast tracks.
// FIXME It would be better for client to tell AudioFlinger the value of N,
// 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, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are
// set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
mParamStatus(NO_ERROR),
//FIXME: mStandby should be true here. Is this some kind of hack?
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()
{
// mConfigEvents should be empty, but just in case it isn't, free the memory it owns
for (size_t i = 0; i < mConfigEvents.size(); i++) {
delete mConfigEvents[i];
}
mConfigEvents.clear();
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);
// FIXME Need to understand why this has be done asynchronously
int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(),
true /*asynchronous*/);
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, "Channel Count: %u\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(int uid)
{
Mutex::Autolock _l(mLock);
acquireWakeLock_l(uid);
}
String16 AudioFlinger::ThreadBase::getWakeLockTag()
{
switch (mType) {
case MIXER:
return String16("AudioMix");
case DIRECT:
return String16("AudioDirectOut");
case DUPLICATING:
return String16("AudioDup");
case RECORD:
return String16("AudioIn");
case OFFLOAD:
return String16("AudioOffload");
default:
ALOG_ASSERT(false);
return String16("AudioUnknown");
}
}
void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
{
getPowerManager_l();
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
status_t status;
if (uid >= 0) {
status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
getWakeLockTag(),
String16("media"),
uid);
} else {
status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
getWakeLockTag(),
String16("media"));
}
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::updateWakeLockUids(const SortedVector<int> &uids) {
Mutex::Autolock _l(mLock);
updateWakeLockUids_l(uids);
}
void AudioFlinger::ThreadBase::getPowerManager_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);
}
}
}
void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
getPowerManager_l();
if (mWakeLockToken == NULL) {
ALOGE("no wake lock to update!");
return;
}
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
status_t status;
status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array());
ALOGV("acquireWakeLock_l() %s status %d", mName, status);
}
}
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;
}
// Allow global effects only on offloaded and mixer threads
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
switch (mType) {
case MIXER:
case OFFLOAD:
break;
case DIRECT:
case DUPLICATING:
case RECORD:
default:
ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName);
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;
}
effect->setOffloaded(mType == OFFLOAD, mId);
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;
ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(),
"addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x",
this, effect->desc().name, effect->desc().flags);
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;
}
effect->setOffloaded(mType == OFFLOAD, mId);
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),
mNormalFrameCount(0), mMixBuffer(NULL),
mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
mActiveTracksGeneration(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),
mBytesRemaining(0),
mCurrentWriteLength(0),
mUseAsyncWrite(false),
mWriteAckSequence(0),
mDrainSequence(0),
mSignalPending(false),
mScreenState(AudioFlinger::mScreenState),
// index 0 is reserved for normal mixer's submix
mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1),
// mLatchD, mLatchQ,
mLatchDValid(false), mLatchQValid(false)
{
snprintf(mName, kNameLength, "AudioOut_%X", id);
mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
// 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()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
delete [] mAllocMixBuffer;
}
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, "Normal frame count: %d\n", mNormalFrameCount);
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,
int uid,
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))
)
) &&
// 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 if (mType == OFFLOAD) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
"for output %p with format %d",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
} else {
if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) {
ALOGE("createTrack_l() Bad parameter: format %d \""
"for output %p with format %d",
format, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
// 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, uid, *flags);
} else {
track = TimedTrack::create(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId, uid);
}
if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
// track must be cleared from the caller as the caller has the AF lock
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;
broadcast_l();
}
void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
Mutex::Autolock _l(mLock);
mStreamTypes[stream].mute = muted;
broadcast_l();
}
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.
if (!track->isOutputTrack()) {
TrackBase::track_state state = track->mState;
mLock.unlock();
status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId());
mLock.lock();
// abort track was stopped/paused while we released the lock
if (state != track->mState) {
if (status == NO_ERROR) {
mLock.unlock();
AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
mLock.lock();
}
return INVALID_OPERATION;
}
// abort if start is rejected by audio policy manager
if (status != NO_ERROR) {
return PERMISSION_DENIED;
}
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
#endif
}
track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
track->mResetDone = false;
track->mPresentationCompleteFrames = 0;
mActiveTracks.add(track);
mWakeLockUids.add(track->uid());
mActiveTracksGeneration++;
mLatestActiveTrack = track;
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("signal playback thread");
broadcast_l();
return status;
}
bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
track->terminate();
// active tracks are removed by threadLoop()
bool trackActive = (mActiveTracks.indexOf(track) >= 0);
track->mState = TrackBase::STOPPED;
if (!trackActive) {
removeTrack_l(track);
} else if (track->isFastTrack() || track->isOffloaded()) {
track->mState = TrackBase::STOPPING_1;
}
return trackActive;
}
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();
}
}
void AudioFlinger::PlaybackThread::broadcast_l()
{
// Thread could be blocked waiting for async
// so signal it to handle state changes immediately
// If threadLoop is currently unlocked a signal of mWaitWorkCV will
// be lost so we also flag to prevent it blocking on mWaitWorkCV
mSignalPending = true;
mWaitWorkCV.broadcast();
}
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
return String8();
}
char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
const String8 out_s8(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.channelMask = 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::writeCallback()
{
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->resetWriteBlocked();
}
void AudioFlinger::PlaybackThread::drainCallback()
{
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->resetDraining();
}
void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// reject out of sequence requests
if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) {
mWriteAckSequence &= ~1;
mWaitWorkCV.signal();
}
}
void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// reject out of sequence requests
if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
mDrainSequence &= ~1;
mWaitWorkCV.signal();
}
}
// static
int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
void *param,
void *cookie)
{
AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
ALOGV("asyncCallback() event %d", event);
switch (event) {
case STREAM_CBK_EVENT_WRITE_READY:
me->writeCallback();
break;
case STREAM_CBK_EVENT_DRAIN_READY:
me->drainCallback();
break;
default:
ALOGW("asyncCallback() unknown event %d", event);
break;
}
return 0;
}
void AudioFlinger::PlaybackThread::readOutputParameters()
{
// unfortunately we have no way of recovering from errors here, hence the LOG_FATAL
mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
if (!audio_is_output_channel(mChannelMask)) {
LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
}
if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) {
LOG_FATAL("HAL channel mask %#x not supported for mixed output; "
"must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask);
}
mChannelCount = popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
if (!audio_is_valid_format(mFormat)) {
LOG_FATAL("HAL format %d not valid for output", mFormat);
}
if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
mFormat);
}
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);
}
if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) &&
(mOutput->stream->set_callback != NULL)) {
if (mOutput->stream->set_callback(mOutput->stream,
AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
mUseAsyncWrite = true;
mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
}
}
// 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[] mAllocMixBuffer;
size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize;
mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1];
mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align);
memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize);
// 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->isInvalid()) {
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->isInvalid()) {
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 (count) {
for (size_t i = 0 ; i < count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
if (!track->isOutputTrack()) {
AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
#endif
if (track->isTerminated()) {
AudioSystem::releaseOutput(mId);
}
}
}
}
}
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
ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
{
// FIXME rewrite to reduce number of system calls
mLastWriteTime = systemTime();
mInWrite = true;
ssize_t 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 = mBytesRemaining >> mBitShift;
size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1;
ATRACE_BEGIN("write");
// 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 + offset, count);
ATRACE_END();
if (framesWritten > 0) {
bytesWritten = framesWritten << mBitShift;
} else {
bytesWritten = framesWritten;
}
status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
if (status == NO_ERROR) {
size_t totalFramesWritten = mNormalSink->framesWritten();
if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) {
mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition;
mLatchDValid = true;
}
}
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
size_t offset = (mCurrentWriteLength - mBytesRemaining);
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
mWriteAckSequence |= 1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->stream->write(mOutput->stream,
(char *)mMixBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
mWriteAckSequence &= ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
}
mNumWrites++;
mInWrite = false;
mStandby = false;
return bytesWritten;
}
void AudioFlinger::PlaybackThread::threadLoop_drain()
{
if (mOutput->stream->drain) {
ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
if (mUseAsyncWrite) {
ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
mDrainSequence |= 1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setDraining(mDrainSequence);
}
mOutput->stream->drain(mOutput->stream,
(mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY
: AUDIO_DRAIN_ALL);
}
}
void AudioFlinger::PlaybackThread::threadLoop_exit()
{
// Default implementation has nothing to do
}
/*
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) {
t->invalidate();
}
}
}
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;
int lastGeneration = 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();
// mNBLogWriter->log can only be called while thread mutex mLock is held.
// So if you need to log when mutex is unlocked, set logString to a non-NULL string,
// and then that string will be logged at the next convenient opportunity.
const char *logString = NULL;
checkSilentMode_l();
while (!exitPending())
{
cpuStats.sample(myName);
Vector< sp<EffectChain> > effectChains;
processConfigEvents();
{ // scope for mLock
Mutex::Autolock _l(mLock);
if (logString != NULL) {
mNBLogWriter->logTimestamp();
mNBLogWriter->log(logString);
logString = NULL;
}
if (mLatchDValid) {
mLatchQ = mLatchD;
mLatchDValid = false;
mLatchQValid = true;
}
if (checkForNewParameters_l()) {
cacheParameters_l();
}
saveOutputTracks();
if (mSignalPending) {
// A signal was raised while we were unlocked
mSignalPending = false;
} else if (waitingAsyncCallback_l()) {
if (exitPending()) {
break;
}
releaseWakeLock_l();
mWakeLockUids.clear();
mActiveTracksGeneration++;
ALOGV("wait async completion");
mWaitWorkCV.wait(mLock);
ALOGV("async completion/wake");
acquireWakeLock_l();
standbyTime = systemTime() + standbyDelay;
sleepTime = 0;
continue;
}
if ((!mActiveTracks.size() && systemTime() > standbyTime) ||
isSuspended()) {
// put audio hardware into standby after short delay
if (shouldStandby_l()) {
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();
mWakeLockUids.clear();
mActiveTracksGeneration++;
// 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;
mBytesRemaining = 0;
checkSilentMode_l();
standbyTime = systemTime() + standbyDelay;
sleepTime = idleSleepTime;
if (mType == MIXER) {
sleepTimeShift = 0;
}
continue;
}
}
// mMixerStatusIgnoringFastTracks is also updated internally
mMixerStatus = prepareTracks_l(&tracksToRemove);
// compare with previously applied list
if (lastGeneration != mActiveTracksGeneration) {
// update wakelock
updateWakeLockUids_l(mWakeLockUids);
lastGeneration = mActiveTracksGeneration;
}
// 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);
} // mLock scope ends
if (mBytesRemaining == 0) {
mCurrentWriteLength = 0;
if (mMixerStatus == MIXER_TRACKS_READY) {
// threadLoop_mix() sets mCurrentWriteLength
threadLoop_mix();
} else if ((mMixerStatus != MIXER_DRAIN_TRACK)
&& (mMixerStatus != MIXER_DRAIN_ALL)) {
// threadLoop_sleepTime sets sleepTime to 0 if data
// must be written to HAL
threadLoop_sleepTime();
if (sleepTime == 0) {
mCurrentWriteLength = mixBufferSize;
}
}
mBytesRemaining = mCurrentWriteLength;
if (isSuspended()) {
sleepTime = suspendSleepTimeUs();
// simulate write to HAL when suspended
mBytesWritten += mixBufferSize;
mBytesRemaining = 0;
}
// only process effects if we're going to write
if (sleepTime == 0 && mType != OFFLOAD) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
}
}
// Process effect chains for offloaded thread even if no audio
// was read from audio track: process only updates effect state
// and thus does have to be synchronized with audio writes but may have
// to be called while waiting for async write callback
if (mType == OFFLOAD) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
}
// enable changes in effect chain
unlockEffectChains(effectChains);
if (!waitingAsyncCallback()) {
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
if (mBytesRemaining) {
ssize_t ret = threadLoop_write();
if (ret < 0) {
mBytesRemaining = 0;
} else {
mBytesWritten += ret;
mBytesRemaining -= ret;
}
} else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
(mMixerStatus == MIXER_DRAIN_ALL)) {
threadLoop_drain();
}
if (mType == MIXER) {
// write blocked detection
nsecs_t now = systemTime();
nsecs_t delta = now - mLastWriteTime;
if (!mStandby && delta > maxPeriod) {
mNumDelayedWrites++;
if ((now - lastWarning) > kWarningThrottleNs) {
ATRACE_NAME("underrun");
ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
}
}
}
} 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).
}
threadLoop_exit();
// for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ...
if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) {
// put output stream into standby mode
if (!mStandby) {
mOutput->stream->common.standby(&mOutput->stream->common);
}
}
releaseWakeLock();
mWakeLockUids.clear();
mActiveTracksGeneration++;
ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
// removeTracks_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
{
size_t count = tracksToRemove.size();
if (count) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
mActiveTracks.remove(track);
mWakeLockUids.remove(track->uid());
mActiveTracksGeneration++;
ALOGV("removeTracks_l removing track on session %d", track->sessionId());
sp<EffectChain> 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);
}
}
}
}
status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
{
if (mNormalSink != 0) {
return mNormalSink->getTimestamp(timestamp);
}
if (mType == OFFLOAD && mOutput->stream->get_presentation_position) {
uint64_t position64;
int ret = mOutput->stream->get_presentation_position(
mOutput->stream, &position64, ×tamp.mTime);
if (ret == 0) {
timestamp.mPosition = (uint32_t)position64;
return NO_ERROR;
}
}
return INVALID_OPERATION;
}
// ----------------------------------------------------------------------------
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=%u, 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
if (mTeeSinkOutputEnabled) {
// create a Pipe to archive a copy of FastMixer's output for dumpsys
Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, 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;
#ifdef TEE_SINK
state->mTeeSink = mTeeSink.get();
#endif
mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
state->mNBLogWriter = mFastMixerNBLogWriter.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
}
mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
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);
}
ssize_t 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;
mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
if (kUseFastMixer == FastMixer_Dynamic) {
mNormalSink = mPipeSink;
}
} else {
sq->end(false /*didModify*/);
}
}
return 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();
}
// Empty implementation for standard mixer
// Overridden for offloaded playback
void AudioFlinger::PlaybackThread::flushOutput_l()
{
}
bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
{
return false;
}
bool AudioFlinger::PlaybackThread::shouldStandby_l()
{
return !mStandby;
}
bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
{
Mutex::Autolock _l(mLock);
return waitingAsyncCallback_l();
}
// 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);
if (mUseAsyncWrite != 0) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
mDrainSequence = (mDrainSequence + 2) & ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
mCallbackThread->setDraining(mDrainSequence);
}
}
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);
mCurrentWriteLength = mixBufferSize;
// 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++) {
const 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()) &&
recentUnderruns > 0) {
// FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
}
// 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->isTerminated()) {
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->mFlags);
// 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::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 = 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
size_t desiredFrames;
uint32_t sr = track->sampleRate();
if (sr == mSampleRate) {
desiredFrames = mNormalFrameCount;
} else {
// +1 for rounding and +1 for additional sample needed for interpolation
desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
// add frames already consumed but not yet released by the resampler
// because cblk->framesReady() will include these frames
desiredFrames += 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(desiredFrames <= cblk->frameCount_);
}
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
minFrames = desiredFrames;
}
size_t framesReady = track->framesReady();
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, 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);
// FIXME should not make a decision based on mServer
} else if (cblk->mServer != 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->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;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
uint32_t vlr = proxy->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 = proxy->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());
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * 2;
uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
if (reqSampleRate == 0) {
reqSampleRate = mSampleRate;
} else if (reqSampleRate > maxSampleRate) {
reqSampleRate = maxSampleRate;
}
mAudioMixer->setParameter(
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)reqSampleRate);
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 {
if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
}
// 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 s=%08x [NOT READY] on thread %p", name, cblk->mServer, 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 {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
ALOGI("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->mFlags);
// 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;
}
}
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...
removeTracks_l(*tracksToRemove);
// 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 ((mBytesRemaining == 0) && ((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 ((audio_channel_mask_t) 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.
if (value != AUDIO_DEVICE_NONE) {
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) {
readOutputParameters();
delete mAudioMixer;
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;
}
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
const 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
#ifdef TEE_SINK
// Write the tee output to a .wav file
dumpTee(fd, mTeeSource, mId);
#endif
#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(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output, audio_io_handle_t id, uint32_t device,
ThreadBase::type_t type)
: PlaybackThread(audioFlinger, output, id, device, type)
// mLeftVolFloat, mRightVolFloat
{
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
{
audio_track_cblk_t* cblk = track->cblk();
float left, right;
if (mMasterMute || mStreamTypes[track->streamType()].mute) {
left = right = 0;
} else {
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
uint32_t vlr = proxy->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 (lastTrack) {
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()) {
mEffectChains[0]->setVolume_l(&vl, &vr);
left = (float)vl / (1 << 24);
right = (float)vr / (1 << 24);
}
if (mOutput->stream->set_volume) {
mOutput->stream->set_volume(mOutput->stream, left, right);
}
}
}
}
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
Vector< sp<Track> > *tracksToRemove
)
{
size_t count = mActiveTracks.size();
mixer_state mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
for (size_t i = 0; i < count; i++) {
sp<Track> t = mActiveTracks[i].promote();
// The track died recently
if (t == 0) {
continue;
}
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// Only consider last track started for volume and mixer state control.
// In theory an older track could underrun and restart after the new one starts
// but as we only care about the transition phase between two tracks on a
// direct output, it is not a problem to ignore the underrun case.
sp<Track> l = mLatestActiveTrack.promote();
bool last = l.get() == track;
// 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 s=%08x [OK]", track->name(), cblk->mServer);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
// make sure processVolume_l() will apply new volume even if 0
mLeftVolFloat = mRightVolFloat = -1.0;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
}
}
// compute volume for this track
processVolume_l(track, last);
if (last) {
// reset retry count
track->mRetryCount = kMaxTrackRetriesDirect;
mActiveTrack = t;
mixerStatus = MIXER_TRACKS_READY;
}
} else {
// clear effect chain input buffer if the last active track started underruns
// to avoid sending previous audio buffer again to effects
if (!mEffectChains.isEmpty() && last) {
mEffectChains[0]->clearInputBuffer();
}
ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
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 || !last ||
track->presentationComplete(framesWritten, audioHALFrames)) {
if (track->isStopped()) {
track->reset();
}
tracksToRemove->add(track);
}
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
// Only consider last track started for mixer state control
if (--(track->mRetryCount) <= 0) {
ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
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->mFlags);
} else if (last) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
}
}
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
return mixerStatus;
}
void AudioFlinger::DirectOutputThread::threadLoop_mix()
{
size_t frameCount = mFrameCount;
int8_t *curBuf = (int8_t *)mMixBuffer;
// output audio to hardware
while (frameCount) {
AudioBufferProvider::Buffer buffer;
buffer.frameCount = frameCount;
mActiveTrack->getNextBuffer(&buffer);
if (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);
}
mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer;
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
if (audio_is_linear_pcm(mFormat)) {
standbyDelay = microseconds(activeSleepTime*2);
} else {
standbyDelay = kOffloadStandbyDelayNs;
}
}
// ----------------------------------------------------------------------------
AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
const wp<AudioFlinger::PlaybackThread>& playbackThread)
: Thread(false /*canCallJava*/),
mPlaybackThread(playbackThread),
mWriteAckSequence(0),
mDrainSequence(0)
{
}
AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
{
}
void AudioFlinger::AsyncCallbackThread::onFirstRef()
{
run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
}
bool AudioFlinger::AsyncCallbackThread::threadLoop()
{
while (!exitPending()) {
uint32_t writeAckSequence;
uint32_t drainSequence;
{
Mutex::Autolock _l(mLock);
while (!((mWriteAckSequence & 1) ||
(mDrainSequence & 1) ||
exitPending())) {
mWaitWorkCV.wait(mLock);
}
if (exitPending()) {
break;
}
ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d",
mWriteAckSequence, mDrainSequence);
writeAckSequence = mWriteAckSequence;
mWriteAckSequence &= ~1;
drainSequence = mDrainSequence;
mDrainSequence &= ~1;
}
{
sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
if (playbackThread != 0) {
if (writeAckSequence & 1) {
playbackThread->resetWriteBlocked(writeAckSequence >> 1);
}
if (drainSequence & 1) {
playbackThread->resetDraining(drainSequence >> 1);
}
}
}
}
return false;
}
void AudioFlinger::AsyncCallbackThread::exit()
{
ALOGV("AsyncCallbackThread::exit");
Mutex::Autolock _l(mLock);
requestExit();
mWaitWorkCV.broadcast();
}
void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// bit 0 is cleared
mWriteAckSequence = sequence << 1;
}
void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
{
Mutex::Autolock _l(mLock);
// ignore unexpected callbacks
if (mWriteAckSequence & 2) {
mWriteAckSequence |= 1;
mWaitWorkCV.signal();
}
}
void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// bit 0 is cleared
mDrainSequence = sequence << 1;
}
void AudioFlinger::AsyncCallbackThread::resetDraining()
{
Mutex::Autolock _l(mLock);
// ignore unexpected callbacks
if (mDrainSequence & 2) {
mDrainSequence |= 1;
mWaitWorkCV.signal();
}
}
// ----------------------------------------------------------------------------
AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
: DirectOutputThread(audioFlinger, output, id, device, OFFLOAD),
mHwPaused(false),
mFlushPending(false),
mPausedBytesRemaining(0)
{
//FIXME: mStandby should be set to true by ThreadBase constructor
mStandby = true;
}
void AudioFlinger::OffloadThread::threadLoop_exit()
{
if (mFlushPending || mHwPaused) {
// If a flush is pending or track was paused, just discard buffered data
flushHw_l();
} else {
mMixerStatus = MIXER_DRAIN_ALL;
threadLoop_drain();
}
mCallbackThread->exit();
PlaybackThread::threadLoop_exit();
}
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
Vector< sp<Track> > *tracksToRemove
)
{
size_t count = mActiveTracks.size();
mixer_state mixerStatus = MIXER_IDLE;
bool doHwPause = false;
bool doHwResume = false;
ALOGV("OffloadThread::prepareTracks_l active tracks %d", count);
// find out which tracks need to be processed
for (size_t i = 0; i < count; i++) {
sp<Track> t = mActiveTracks[i].promote();
// The track died recently
if (t == 0) {
continue;
}
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// Only consider last track started for volume and mixer state control.
// In theory an older track could underrun and restart after the new one starts
// but as we only care about the transition phase between two tracks on a
// direct output, it is not a problem to ignore the underrun case.
sp<Track> l = mLatestActiveTrack.promote();
bool last = l.get() == track;
if (track->isPausing()) {
track->setPaused();
if (last) {
if (!mHwPaused) {
doHwPause = true;
mHwPaused = true;
}
// If we were part way through writing the mixbuffer to
// the HAL we must save this until we resume
// BUG - this will be wrong if a different track is made active,
// in that case we want to discard the pending data in the
// mixbuffer and tell the client to present it again when the
// track is resumed
mPausedWriteLength = mCurrentWriteLength;
mPausedBytesRemaining = mBytesRemaining;
mBytesRemaining = 0; // stop writing
}
tracksToRemove->add(track);
} else if (track->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
// make sure processVolume_l() will apply new volume even if 0
mLeftVolFloat = mRightVolFloat = -1.0;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
if (last) {
if (mPausedBytesRemaining) {
// Need to continue write that was interrupted
mCurrentWriteLength = mPausedWriteLength;
mBytesRemaining = mPausedBytesRemaining;
mPausedBytesRemaining = 0;
}
if (mHwPaused) {
doHwResume = true;
mHwPaused = false;
// threadLoop_mix() will handle the case that we need to
// resume an interrupted write
}
// enable write to audio HAL
sleepTime = 0;
}
}
}
if (last) {
sp<Track> previousTrack = mPreviousTrack.promote();
if (previousTrack != 0) {
if (track != previousTrack.get()) {
// Flush any data still being written from last track
mBytesRemaining = 0;
if (mPausedBytesRemaining) {
// Last track was paused so we also need to flush saved
// mixbuffer state and invalidate track so that it will
// re-submit that unwritten data when it is next resumed
mPausedBytesRemaining = 0;
// Invalidate is a bit drastic - would be more efficient
// to have a flag to tell client that some of the
// previously written data was lost
previousTrack->invalidate();
}
// flush data already sent to the DSP if changing audio session as audio
// comes from a different source. Also invalidate previous track to force a
// seek when resuming.
if (previousTrack->sessionId() != track->sessionId()) {
previousTrack->invalidate();
mFlushPending = true;
}
}
}
mPreviousTrack = track;
// reset retry count
track->mRetryCount = kMaxTrackRetriesOffload;
mActiveTrack = t;
mixerStatus = MIXER_TRACKS_READY;
}
} else {
ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
if (track->isStopping_1()) {
// Hardware buffer can hold a large amount of audio so we must
// wait for all current track's data to drain before we say
// that the track is stopped.
if (mBytesRemaining == 0) {
// Only start draining when all data in mixbuffer
// has been written
ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
track->mState = TrackBase::STOPPING_2; // so presentation completes after drain
// do not drain if no data was ever sent to HAL (mStandby == true)
if (last && !mStandby) {
// do not modify drain sequence if we are already draining. This happens
// when resuming from pause after drain.
if ((mDrainSequence & 1) == 0) {
sleepTime = 0;
standbyTime = systemTime() + standbyDelay;
mixerStatus = MIXER_DRAIN_TRACK;
mDrainSequence += 2;
}
if (mHwPaused) {
// It is possible to move from PAUSED to STOPPING_1 without
// a resume so we must ensure hardware is running
doHwResume = true;
mHwPaused = false;
}
}
}
} else if (track->isStopping_2()) {
// Drain has completed or we are in standby, signal presentation complete
if (!(mDrainSequence & 1) || !last || mStandby) {
track->mState = TrackBase::STOPPED;
size_t audioHALFrames =
(mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
size_t framesWritten =
mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
track->presentationComplete(framesWritten, audioHALFrames);
track->reset();
tracksToRemove->add(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("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
track->name());
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->mFlags);
} else if (last){
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
}
// compute volume for this track
processVolume_l(track, last);
}
// make sure the pause/flush/resume sequence is executed in the right order.
// If a flush is pending and a track is active but the HW is not paused, force a HW pause
// before flush and then resume HW. This can happen in case of pause/flush/resume
// if resume is received before pause is executed.
if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
mOutput->stream->pause(mOutput->stream);
if (!doHwPause) {
doHwResume = true;
}
}
if (mFlushPending) {
flushHw_l();
mFlushPending = false;
}
if (!mStandby && doHwResume) {
mOutput->stream->resume(mOutput->stream);
}
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
return mixerStatus;
}
void AudioFlinger::OffloadThread::flushOutput_l()
{
mFlushPending = true;
}
// must be called with thread mutex locked
bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
{
ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
mWriteAckSequence, mDrainSequence);
if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) {
return true;
}
return false;
}
// must be called with thread mutex locked
bool AudioFlinger::OffloadThread::shouldStandby_l()
{
bool TrackPaused = false;
// do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
// after a timeout and we will enter standby then.
if (mTracks.size() > 0) {
TrackPaused = mTracks[mTracks.size() - 1]->isPaused();
}
return !mStandby && !TrackPaused;
}
bool AudioFlinger::OffloadThread::waitingAsyncCallback()
{
Mutex::Autolock _l(mLock);
return waitingAsyncCallback_l();
}
void AudioFlinger::OffloadThread::flushHw_l()
{
mOutput->stream->flush(mOutput->stream);
// Flush anything still waiting in the mixbuffer
mCurrentWriteLength = 0;
mBytesRemaining = 0;
mPausedWriteLength = 0;
mPausedBytesRemaining = 0;
if (mUseAsyncWrite) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
mDrainSequence = (mDrainSequence + 2) & ~1;
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
mCallbackThread->setDraining(mDrainSequence);
}
}
// ----------------------------------------------------------------------------
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;
mCurrentWriteLength = mixBufferSize;
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;
}
}
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(mMixBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)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,
IPCThreadState::self()->getCallingUid());
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 outDevice,
audio_devices_t inDevice
#ifdef TEE_SINK
, const sp<NBAIO_Sink>& teeSink
#endif
) :
ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
// mRsmpInIndex and mBufferSize 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?)
#ifdef TEE_SINK
, mTeeSink(teeSink)
#endif
{
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();
{
Mutex::Autolock _l(mLock);
activeTrack = mActiveTrack;
acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1);
}
// 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 && activeTrack != mActiveTrack) {
SortedVector<int> tmp;
tmp.add(mActiveTrack->uid());
updateWakeLockUids_l(tmp);
}
activeTrack = mActiveTrack;
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(mActiveTrack != 0 ? mActiveTrack->uid() : -1);
continue;
}
if (mActiveTrack != 0) {
if (mActiveTrack->isTerminated()) {
removeTrack_l(mActiveTrack);
mActiveTrack.clear();
} else 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;
}
}
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;
status_t status = mActiveTrack->getNextBuffer(&buffer);
if (status == 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) {
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) {
readInto = buffer.raw;
framesOut = 0;
} else {
readInto = mRsmpInBuffer;
mRsmpInIndex = 0;
}
mBytesRead = mInput->stream->read(mInput->stream, readInto,
mBufferSize);
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;
}
#ifdef TEE_SINK
else if (mTeeSink != 0) {
(void) mTeeSink->write(readInto,
mBytesRead >> Format_frameBitShift(mTeeSink->format()));
}
#endif
}
}
} else {
// resampling
// resampler accumulates, but we only have one source track
memset(mRsmpOutBuffer, 0, framesOut * FCC_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) {
// temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
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);
}
// now done with mRsmpOutBuffer
}
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);
for (size_t i = 0; i < mTracks.size(); i++) {
sp<RecordTrack> track = mTracks[i];
track->invalidate();
}
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,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status)
{
sp<RecordTrack> track;
status_t lStatus;
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createRecordTrack_l() audio driver not initialized");
goto Exit;
}
// client expresses a preference for FAST, but we get the final say
if (*flags & IAudioFlinger::TRACK_FAST) {
if (
// use case: callback handler and frame count is default or at least as large as HAL
(
(tid != -1) &&
((frameCount == 0) ||
(frameCount >= mFrameCount))
) &&
// FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format)
// mono or stereo
( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
(channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// record thread has an associated fast recorder
hasFastRecorder()
// FIXME test that RecordThread for this fast track has a capable output HAL
// FIXME add a permission test also?
) {
// if frameCount not specified, then it defaults to fast recorder (HAL) frame count
if (frameCount == 0) {
frameCount = mFrameCount * kFastTrackMultiplier;
}
ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d "
"mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
"hasFastRecorder=%d tid=%d",
frameCount, mFrameCount, format,
audio_is_linear_pcm(format),
channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid);
*flags &= ~IAudioFlinger::TRACK_FAST;
// For compatibility with AudioRecord calculation, buffer depth is forced
// to be at least 2 x the record thread 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 = 50; // FIXME mInput->stream->get_latency(mInput->stream);
size_t mNormalFrameCount = 2048; // FIXME
uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
if (minBufCount < 2) {
minBufCount = 2;
}
size_t minFrameCount = mNormalFrameCount * minBufCount;
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
}
// 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, uid);
if (track->getCblk() == 0) {
ALOGE("createRecordTrack_l() no control block");
lStatus = NO_MEMORY;
// track must be cleared from the caller as the caller has the AF lock
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);
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;
}
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(RecordThread::RecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
AutoMutex _l(mLock);
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->terminate();
track->mState = TrackBase::STOPPED;
// 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, "Buffer size: %u bytes\n", mBufferSize);
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, mBufferSize);
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) {
if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
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 &&
(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)
{
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
return String8();
}
char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
const String8 out_s8(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.channelMask = 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 = popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
}
mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
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 * FCC_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