blob: c2320bc0ec7a71a28697783c79e85cd2acd9f7a8 [file] [log] [blame]
/*
**
** 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 <linux/futex.h>
#include <sys/stat.h>
#include <sys/syscall.h>
#include <cutils/properties.h>
#include <media/AudioParameter.h>
#include <media/AudioResamplerPublic.h>
#include <media/RecordBufferConverter.h>
#include <media/TypeConverter.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <private/media/AudioTrackShared.h>
#include <private/android_filesystem_config.h>
#include <audio_utils/mono_blend.h>
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
#include <audio_utils/minifloat.h>
#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_aec.h>
#include <system/audio.h>
// NBAIO implementations
#include <media/nbaio/AudioStreamInSource.h>
#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 <mediautils/BatteryNotifier.h>
#include <powermanager/PowerManager.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <media/audiohal/StreamHalInterface.h>
#include "AudioFlinger.h"
#include "FastMixer.h"
#include "FastCapture.h"
#include "ServiceUtilities.h"
#include "mediautils/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
#include "AutoPark.h"
#include <pthread.h>
#include "TypedLogger.h"
// ----------------------------------------------------------------------------
// 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
// TODO: Move these macro/inlines to a header file.
#define max(a, b) ((a) > (b) ? (a) : (b))
template <typename T>
static inline T min(const T& a, const T& b)
{
return a < b ? a : b;
}
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 in sendConfigEvent_l() for a status to be received
static const nsecs_t kConfigEventTimeoutNs = 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 sink buffer size, expressed in milliseconds rather than frames
// FIXME This should be based on experimentally observed scheduling jitter
static const uint32_t kMinNormalSinkBufferSizeMs = 20;
// maximum normal sink buffer size
static const uint32_t kMaxNormalSinkBufferSizeMs = 24;
// minimum capture buffer size in milliseconds to _not_ need a fast capture thread
// FIXME This should be based on experimentally observed scheduling jitter
static const uint32_t kMinNormalCaptureBufferSizeMs = 12;
// Offloaded output thread standby delay: allows track transition without going to standby
static const nsecs_t kOffloadStandbyDelayNs = seconds(1);
// Direct output thread minimum sleep time in idle or active(underrun) state
static const nsecs_t kDirectMinSleepTimeUs = 10000;
// The universal constant for ubiquitous 20ms value. The value of 20ms seems to provide a good
// balance between power consumption and latency, and allows threads to be scheduled reliably
// by the CFS scheduler.
// FIXME Express other hardcoded references to 20ms with references to this constant and move
// it appropriately.
#define FMS_20 20
// 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;
// Whether to use fast capture
static const enum {
FastCapture_Never, // never initialize or use: for debugging only
FastCapture_Always, // always initialize and use, even if not needed: for debugging only
FastCapture_Static, // initialize if needed, then use all the time if initialized
} kUseFastCapture = FastCapture_Static;
// Priorities for requestPriority
static const int kPriorityAudioApp = 2;
static const int kPriorityFastMixer = 3;
static const int kPriorityFastCapture = 3;
// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the
// track buffer in shared memory. Zero on input means to use a default value. For fast tracks,
// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'.
// This is the default value, if not specified by property.
static const int kFastTrackMultiplier = 2;
// The minimum and maximum allowed values
static const int kFastTrackMultiplierMin = 1;
static const int kFastTrackMultiplierMax = 2;
// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
static int sFastTrackMultiplier = kFastTrackMultiplier;
// See Thread::readOnlyHeap().
// Initially this heap is used to allocate client buffers for "fast" AudioRecord.
// Eventually it will be the single buffer that FastCapture writes into via HAL read(),
// and that all "fast" AudioRecord clients read from. In either case, the size can be small.
static const size_t kRecordThreadReadOnlyHeapSize = 0x4000;
// ----------------------------------------------------------------------------
static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
static void sFastTrackMultiplierInit()
{
char value[PROPERTY_VALUE_MAX];
if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
char *endptr;
unsigned long ul = strtoul(value, &endptr, 0);
if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
sFastTrackMultiplier = (int) ul;
}
}
}
// ----------------------------------------------------------------------------
#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
// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset
struct {
// call when you acquire a partial wakelock
void acquire(const sp<IBinder> &wakeLockToken) {
pthread_mutex_lock(&mLock);
if (wakeLockToken.get() == nullptr) {
adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
} else {
if (mCount == 0) {
adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
}
++mCount;
}
pthread_mutex_unlock(&mLock);
}
// call when you release a partial wakelock.
void release(const sp<IBinder> &wakeLockToken) {
if (wakeLockToken.get() == nullptr) {
return;
}
pthread_mutex_lock(&mLock);
if (--mCount < 0) {
ALOGE("negative wakelock count");
mCount = 0;
}
pthread_mutex_unlock(&mLock);
}
// retrieves the boottime timebase offset from monotonic.
int64_t getBoottimeOffset() {
pthread_mutex_lock(&mLock);
int64_t boottimeOffset = mBoottimeOffset;
pthread_mutex_unlock(&mLock);
return boottimeOffset;
}
// Adjusts the timebase offset between TIMEBASE_MONOTONIC
// and the selected timebase.
// Currently only TIMEBASE_BOOTTIME is allowed.
//
// This only needs to be called upon acquiring the first partial wakelock
// after all other partial wakelocks are released.
//
// We do an empirical measurement of the offset rather than parsing
// /proc/timer_list since the latter is not a formal kernel ABI.
static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) {
int clockbase;
switch (timebase) {
case ExtendedTimestamp::TIMEBASE_BOOTTIME:
clockbase = SYSTEM_TIME_BOOTTIME;
break;
default:
LOG_ALWAYS_FATAL("invalid timebase %d", timebase);
break;
}
// try three times to get the clock offset, choose the one
// with the minimum gap in measurements.
const int tries = 3;
nsecs_t bestGap, measured;
for (int i = 0; i < tries; ++i) {
const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC);
const nsecs_t tbase = systemTime(clockbase);
const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC);
const nsecs_t gap = tmono2 - tmono;
if (i == 0 || gap < bestGap) {
bestGap = gap;
measured = tbase - ((tmono + tmono2) >> 1);
}
}
// to avoid micro-adjusting, we don't change the timebase
// unless it is significantly different.
//
// Assumption: It probably takes more than toleranceNs to
// suspend and resume the device.
static int64_t toleranceNs = 10000; // 10 us
if (llabs(*offset - measured) > toleranceNs) {
ALOGV("Adjusting timebase offset old: %lld new: %lld",
(long long)*offset, (long long)measured);
*offset = measured;
}
}
pthread_mutex_t mLock;
int32_t mCount;
int64_t mBoottimeOffset;
} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization
// ----------------------------------------------------------------------------
// 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
#ifndef DEBUG_CPU_USAGE
__unused
#endif
) {
#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
// ----------------------------------------------------------------------------
// static
const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
{
switch (type) {
case MIXER:
return "MIXER";
case DIRECT:
return "DIRECT";
case DUPLICATING:
return "DUPLICATING";
case RECORD:
return "RECORD";
case OFFLOAD:
return "OFFLOAD";
case MMAP:
return "MMAP";
default:
return "unknown";
}
}
std::string devicesToString(audio_devices_t devices)
{
std::string result;
if (devices & AUDIO_DEVICE_BIT_IN) {
InputDeviceConverter::maskToString(devices, result);
} else {
OutputDeviceConverter::maskToString(devices, result);
}
return result;
}
std::string inputFlagsToString(audio_input_flags_t flags)
{
std::string result;
InputFlagConverter::maskToString(flags, result);
return result;
}
std::string outputFlagsToString(audio_output_flags_t flags)
{
std::string result;
OutputFlagConverter::maskToString(flags, result);
return result;
}
const char *sourceToString(audio_source_t source)
{
switch (source) {
case AUDIO_SOURCE_DEFAULT: return "default";
case AUDIO_SOURCE_MIC: return "mic";
case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink";
case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink";
case AUDIO_SOURCE_VOICE_CALL: return "voice call";
case AUDIO_SOURCE_CAMCORDER: return "camcorder";
case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition";
case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication";
case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix";
case AUDIO_SOURCE_UNPROCESSED: return "unprocessed";
case AUDIO_SOURCE_FM_TUNER: return "FM tuner";
case AUDIO_SOURCE_HOTWORD: return "hotword";
default: return "unknown";
}
}
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady)
: Thread(false /*canCallJava*/),
mType(type),
mAudioFlinger(audioFlinger),
// mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
// are set by PlaybackThread::readOutputParameters_l() or
// RecordThread::readInputParameters_l()
//FIXME: mStandby should be true here. Is this some kind of hack?
mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
// mName will be set by concrete (non-virtual) subclass
mDeathRecipient(new PMDeathRecipient(this)),
mSystemReady(systemReady),
mSignalPending(false)
{
memset(&mPatch, 0, sizeof(struct audio_patch));
}
AudioFlinger::ThreadBase::~ThreadBase()
{
// mConfigEvents should be empty, but just in case it isn't, free the memory it owns
mConfigEvents.clear();
// do not lock the mutex in destructor
releaseWakeLock_l();
if (mPowerManager != 0) {
sp<IBinder> binder = IInterface::asBinder(mPowerManager);
binder->unlinkToDeath(mDeathRecipient);
}
}
status_t AudioFlinger::ThreadBase::readyToRun()
{
status_t status = initCheck();
if (status == NO_ERROR) {
ALOGI("AudioFlinger's thread %p tid=%d ready to run", this, getTid());
} else {
ALOGE("No working audio driver found.");
}
return status;
}
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)
{
ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
Mutex::Autolock _l(mLock);
return sendSetParameterConfigEvent_l(keyValuePairs);
}
// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
{
status_t status = NO_ERROR;
if (event->mRequiresSystemReady && !mSystemReady) {
event->mWaitStatus = false;
mPendingConfigEvents.add(event);
return status;
}
mConfigEvents.add(event);
ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType);
mWaitWorkCV.signal();
mLock.unlock();
{
Mutex::Autolock _l(event->mLock);
while (event->mWaitStatus) {
if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) {
event->mStatus = TIMED_OUT;
event->mWaitStatus = false;
}
}
status = event->mStatus;
}
mLock.lock();
return status;
}
void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid)
{
Mutex::Autolock _l(mLock);
sendIoConfigEvent_l(event, pid);
}
// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid)
{
sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid);
sendConfigEvent_l(configEvent);
}
void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
{
Mutex::Autolock _l(mLock);
sendPrioConfigEvent_l(pid, tid, prio, forApp);
}
// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(
pid_t pid, pid_t tid, int32_t prio, bool forApp)
{
sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio, forApp);
sendConfigEvent_l(configEvent);
}
// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
{
sp<ConfigEvent> configEvent;
AudioParameter param(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keyMonoOutput), value) == NO_ERROR) {
setMasterMono_l(value != 0);
if (param.size() == 1) {
return NO_ERROR; // should be a solo parameter - we don't pass down
}
param.remove(String8(AudioParameter::keyMonoOutput));
configEvent = new SetParameterConfigEvent(param.toString());
} else {
configEvent = new SetParameterConfigEvent(keyValuePair);
}
return sendConfigEvent_l(configEvent);
}
status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
const struct audio_patch *patch,
audio_patch_handle_t *handle)
{
Mutex::Autolock _l(mLock);
sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
status_t status = sendConfigEvent_l(configEvent);
if (status == NO_ERROR) {
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)configEvent->mData.get();
*handle = data->mHandle;
}
return status;
}
status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
const audio_patch_handle_t handle)
{
Mutex::Autolock _l(mLock);
sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
return sendConfigEvent_l(configEvent);
}
// post condition: mConfigEvents.isEmpty()
void AudioFlinger::ThreadBase::processConfigEvents_l()
{
bool configChanged = false;
while (!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size());
sp<ConfigEvent> event = mConfigEvents[0];
mConfigEvents.removeAt(0);
switch (event->mType) {
case CFG_EVENT_PRIO: {
PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
// FIXME Need to understand why this has to be done asynchronously
int err = requestPriority(data->mPid, data->mTid, data->mPrio, data->mForApp,
true /*asynchronous*/);
if (err != 0) {
ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
data->mPrio, data->mPid, data->mTid, err);
}
} break;
case CFG_EVENT_IO: {
IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
ioConfigChanged(data->mEvent, data->mPid);
} break;
case CFG_EVENT_SET_PARAMETER: {
SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
configChanged = true;
mLocalLog.log("CFG_EVENT_SET_PARAMETER: (%s) configuration changed",
data->mKeyValuePairs.string());
}
} break;
case CFG_EVENT_CREATE_AUDIO_PATCH: {
const audio_devices_t oldDevice = getDevice();
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)event->mData.get();
event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
const audio_devices_t newDevice = getDevice();
mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)",
(unsigned)oldDevice, devicesToString(oldDevice).c_str(),
(unsigned)newDevice, devicesToString(newDevice).c_str());
} break;
case CFG_EVENT_RELEASE_AUDIO_PATCH: {
const audio_devices_t oldDevice = getDevice();
ReleaseAudioPatchConfigEventData *data =
(ReleaseAudioPatchConfigEventData *)event->mData.get();
event->mStatus = releaseAudioPatch_l(data->mHandle);
const audio_devices_t newDevice = getDevice();
mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)",
(unsigned)oldDevice, devicesToString(oldDevice).c_str(),
(unsigned)newDevice, devicesToString(newDevice).c_str());
} break;
default:
ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
break;
}
{
Mutex::Autolock _l(event->mLock);
if (event->mWaitStatus) {
event->mWaitStatus = false;
event->mCond.signal();
}
}
ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
}
if (configChanged) {
cacheParameters_l();
}
}
String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
String8 s;
const audio_channel_representation_t representation =
audio_channel_mask_get_representation(mask);
switch (representation) {
case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
if (output) {
if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
} else {
if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
}
const int len = s.length();
if (len > 2) {
(void) s.lockBuffer(len); // needed?
s.unlockBuffer(len - 2); // remove trailing ", "
}
return s;
}
case AUDIO_CHANNEL_REPRESENTATION_INDEX:
s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask));
return s;
default:
s.appendFormat("unknown mask, representation:%d bits:%#x",
representation, audio_channel_mask_get_bits(mask));
return s;
}
}
void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
this, mThreadName, getTid(), type(), threadTypeToString(type()));
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
dprintf(fd, " Thread may be deadlocked\n");
}
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat).c_str());
dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize);
dprintf(fd, " Channel count: %u\n", mChannelCount);
dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
channelMaskToString(mChannelMask, mType != RECORD).string());
dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat).c_str());
dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize);
dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
for (size_t i = 0; i < numConfig; i++) {
mConfigEvents[i]->dump(buffer, SIZE);
dprintf(fd, "\n %s", buffer);
}
dprintf(fd, "\n");
} else {
dprintf(fd, " none\n");
}
// Note: output device may be used by capture threads for effects such as AEC.
dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).c_str());
dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).c_str());
dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
if (locked) {
mLock.unlock();
}
}
void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
size_t numEffectChains = mEffectChains.size();
snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < numEffectChains; ++i) {
sp<EffectChain> chain = mEffectChains[i];
if (chain != 0) {
chain->dump(fd, args);
}
}
}
void AudioFlinger::ThreadBase::acquireWakeLock()
{
Mutex::Autolock _l(mLock);
acquireWakeLock_l();
}
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");
case MMAP:
return String16("Mmap");
default:
ALOG_ASSERT(false);
return String16("AudioUnknown");
}
}
void AudioFlinger::ThreadBase::acquireWakeLock_l()
{
getPowerManager_l();
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
// Uses AID_AUDIOSERVER for wakelock. updateWakeLockUids_l() updates with client uids.
status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
getWakeLockTag(),
String16("audioserver"),
true /* FIXME force oneway contrary to .aidl */);
if (status == NO_ERROR) {
mWakeLockToken = binder;
}
ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
}
gBoottime.acquire(mWakeLockToken);
mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] =
gBoottime.getBoottimeOffset();
}
void AudioFlinger::ThreadBase::releaseWakeLock()
{
Mutex::Autolock _l(mLock);
releaseWakeLock_l();
}
void AudioFlinger::ThreadBase::releaseWakeLock_l()
{
gBoottime.release(mWakeLockToken);
if (mWakeLockToken != 0) {
ALOGV("releaseWakeLock_l() %s", mThreadName);
if (mPowerManager != 0) {
mPowerManager->releaseWakeLock(mWakeLockToken, 0,
true /* FIXME force oneway contrary to .aidl */);
}
mWakeLockToken.clear();
}
}
void AudioFlinger::ThreadBase::getPowerManager_l() {
if (mSystemReady && 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", mThreadName);
} else {
mPowerManager = interface_cast<IPowerManager>(binder);
binder->linkToDeath(mDeathRecipient);
}
}
}
void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t> &uids) {
getPowerManager_l();
#if !LOG_NDEBUG
std::stringstream s;
for (uid_t uid : uids) {
s << uid << " ";
}
ALOGD("updateWakeLockUids_l %s uids:%s", mThreadName, s.str().c_str());
#endif
if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called.
if (mSystemReady) {
ALOGE("no wake lock to update, but system ready!");
} else {
ALOGW("no wake lock to update, system not ready yet");
}
return;
}
if (mPowerManager != 0) {
std::vector<int> uidsAsInt(uids.begin(), uids.end()); // powermanager expects uids as ints
status_t status = mPowerManager->updateWakeLockUids(
mWakeLockToken, uidsAsInt.size(), uidsAsInt.data(),
true /* FIXME force oneway contrary to .aidl */);
ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status);
}
}
void AudioFlinger::ThreadBase::clearPowerManager()
{
Mutex::Autolock _l(mLock);
releaseWakeLock_l();
mPowerManager.clear();
}
void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
thread->clearPowerManager();
}
ALOGW("power manager service died !!!");
}
void AudioFlinger::ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, audio_session_t 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++) {
const 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,
audio_session_t 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,
audio_session_t sessionId)
{
Mutex::Autolock _l(mLock);
checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
}
void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
bool enabled,
audio_session_t 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);
}
}
// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::RecordThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// No global effect sessions on record threads
if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s",
desc->name, mThreadName);
return BAD_VALUE;
}
// only pre processing effects on record thread
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) {
ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s",
desc->name, mThreadName);
return BAD_VALUE;
}
// always allow effects without processing load or latency
if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) {
return NO_ERROR;
}
audio_input_flags_t flags = mInput->flags;
if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) {
if (flags & AUDIO_INPUT_FLAG_RAW) {
ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode",
desc->name, mThreadName);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode",
desc->name, mThreadName);
return BAD_VALUE;
}
}
return NO_ERROR;
}
// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// no preprocessing on playback threads
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) {
ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback"
" thread %s", desc->name, mThreadName);
return BAD_VALUE;
}
// always allow effects without processing load or latency
if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) {
return NO_ERROR;
}
switch (mType) {
case MIXER: {
#ifndef MULTICHANNEL_EFFECT_CHAIN
// Reject any effect on mixer multichannel sinks.
// TODO: fix both format and multichannel issues with effects.
if (mChannelCount != FCC_2) {
ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER"
" thread %s", desc->name, mChannelCount, mThreadName);
return BAD_VALUE;
}
#endif
audio_output_flags_t flags = mOutput->flags;
if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) {
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// global effects are applied only to non fast tracks if they are SW
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
break;
}
} else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
// only post processing on output stage session
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed"
" on output stage session", desc->name);
return BAD_VALUE;
}
} else {
// no restriction on effects applied on non fast tracks
if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) {
break;
}
}
if (flags & AUDIO_OUTPUT_FLAG_RAW) {
ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode",
desc->name);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread"
" in fast mode", desc->name);
return BAD_VALUE;
}
}
} break;
case OFFLOAD:
// nothing actionable on offload threads, if the effect:
// - is offloadable: the effect can be created
// - is NOT offloadable: the effect should still be created, but EffectHandle::enable()
// will take care of invalidating the tracks of the thread
break;
case DIRECT:
// Reject any effect on Direct output threads for now, since the format of
// mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s",
desc->name, mThreadName);
return BAD_VALUE;
case DUPLICATING:
#ifndef MULTICHANNEL_EFFECT_CHAIN
// Reject any effect on mixer multichannel sinks.
// TODO: fix both format and multichannel issues with effects.
if (mChannelCount != FCC_2) {
ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)"
" on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName);
return BAD_VALUE;
}
#endif
if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) {
ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING"
" thread %s", desc->name, mThreadName);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
ALOGW("checkEffectCompatibility_l(): post processing effect %s on"
" DUPLICATING thread %s", desc->name, mThreadName);
return BAD_VALUE;
}
if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) {
ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on"
" DUPLICATING thread %s", desc->name, mThreadName);
return BAD_VALUE;
}
break;
default:
LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
}
return NO_ERROR;
}
// 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,
audio_session_t sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status,
bool pinned)
{
sp<EffectModule> effect;
sp<EffectHandle> handle;
status_t lStatus;
sp<EffectChain> chain;
bool chainCreated = false;
bool effectCreated = false;
bool effectRegistered = false;
audio_unique_id_t effectId = AUDIO_UNIQUE_ID_USE_UNSPECIFIED;
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGW("createEffect_l() Audio driver not initialized.");
goto Exit;
}
ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
{ // scope for mLock
Mutex::Autolock _l(mLock);
lStatus = checkEffectCompatibility_l(desc, sessionId);
if (lStatus != NO_ERROR) {
goto Exit;
}
// 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) {
effectId = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
// Check CPU and memory usage
lStatus = AudioSystem::registerEffect(
desc, mId, chain->strategy(), sessionId, effectId);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectRegistered = true;
// create a new effect module if none present in the chain
lStatus = chain->createEffect_l(effect, this, desc, effectId, sessionId, pinned);
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 = handle->initCheck();
if (lStatus == OK) {
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(effectId);
}
if (chainCreated) {
removeEffectChain_l(chain);
}
// handle must be cleared by caller to avoid deadlock.
}
*status = lStatus;
return handle;
}
void AudioFlinger::ThreadBase::disconnectEffectHandle(EffectHandle *handle,
bool unpinIfLast)
{
bool remove = false;
sp<EffectModule> effect;
{
Mutex::Autolock _l(mLock);
effect = handle->effect().promote();
if (effect == 0) {
return;
}
// restore suspended effects if the disconnected handle was enabled and the last one.
remove = (effect->removeHandle(handle) == 0) && (!effect->isPinned() || unpinIfLast);
if (remove) {
removeEffect_l(effect, true);
}
}
if (remove) {
mAudioFlinger->updateOrphanEffectChains(effect);
AudioSystem::unregisterEffect(effect->id());
if (handle->enabled()) {
checkSuspendOnEffectEnabled(effect, false, effect->sessionId());
}
}
}
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
int effectId)
{
Mutex::Autolock _l(mLock);
return getEffect_l(sessionId, effectId);
}
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t 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
audio_session_t 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, bool release) {
ALOGV("%s %p effect %p", __FUNCTION__, 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, release) == 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(audio_session_t sessionId)
{
Mutex::Autolock _l(mLock);
return getEffectChain_l(sessionId);
}
sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t 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::getAudioPortConfig(struct audio_port_config *config)
{
config->type = AUDIO_PORT_TYPE_MIX;
config->ext.mix.handle = mId;
config->sample_rate = mSampleRate;
config->format = mFormat;
config->channel_mask = mChannelMask;
config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT;
}
void AudioFlinger::ThreadBase::systemReady()
{
Mutex::Autolock _l(mLock);
if (mSystemReady) {
return;
}
mSystemReady = true;
for (size_t i = 0; i < mPendingConfigEvents.size(); i++) {
sendConfigEvent_l(mPendingConfigEvents.editItemAt(i));
}
mPendingConfigEvents.clear();
}
template <typename T>
ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::add(const sp<T> &track) {
ssize_t index = mActiveTracks.indexOf(track);
if (index >= 0) {
ALOGW("ActiveTracks<T>::add track %p already there", track.get());
return index;
}
logTrack("add", track);
mActiveTracksGeneration++;
mLatestActiveTrack = track;
++mBatteryCounter[track->uid()].second;
mHasChanged = true;
return mActiveTracks.add(track);
}
template <typename T>
ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::remove(const sp<T> &track) {
ssize_t index = mActiveTracks.remove(track);
if (index < 0) {
ALOGW("ActiveTracks<T>::remove nonexistent track %p", track.get());
return index;
}
logTrack("remove", track);
mActiveTracksGeneration++;
--mBatteryCounter[track->uid()].second;
// mLatestActiveTrack is not cleared even if is the same as track.
mHasChanged = true;
return index;
}
template <typename T>
void AudioFlinger::ThreadBase::ActiveTracks<T>::clear() {
for (const sp<T> &track : mActiveTracks) {
BatteryNotifier::getInstance().noteStopAudio(track->uid());
logTrack("clear", track);
}
mLastActiveTracksGeneration = mActiveTracksGeneration;
if (!mActiveTracks.empty()) { mHasChanged = true; }
mActiveTracks.clear();
mLatestActiveTrack.clear();
mBatteryCounter.clear();
}
template <typename T>
void AudioFlinger::ThreadBase::ActiveTracks<T>::updatePowerState(
sp<ThreadBase> thread, bool force) {
// Updates ActiveTracks client uids to the thread wakelock.
if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) {
thread->updateWakeLockUids_l(getWakeLockUids());
mLastActiveTracksGeneration = mActiveTracksGeneration;
}
// Updates BatteryNotifier uids
for (auto it = mBatteryCounter.begin(); it != mBatteryCounter.end();) {
const uid_t uid = it->first;
ssize_t &previous = it->second.first;
ssize_t &current = it->second.second;
if (current > 0) {
if (previous == 0) {
BatteryNotifier::getInstance().noteStartAudio(uid);
}
previous = current;
++it;
} else if (current == 0) {
if (previous > 0) {
BatteryNotifier::getInstance().noteStopAudio(uid);
}
it = mBatteryCounter.erase(it); // std::map<> is stable on iterator erase.
} else /* (current < 0) */ {
LOG_ALWAYS_FATAL("negative battery count %zd", current);
}
}
}
template <typename T>
bool AudioFlinger::ThreadBase::ActiveTracks<T>::readAndClearHasChanged() {
const bool hasChanged = mHasChanged;
mHasChanged = false;
return hasChanged;
}
template <typename T>
void AudioFlinger::ThreadBase::ActiveTracks<T>::logTrack(
const char *funcName, const sp<T> &track) const {
if (mLocalLog != nullptr) {
String8 result;
track->appendDump(result, false /* active */);
mLocalLog->log("AT::%-10s(%p) %s", funcName, track.get(), result.string());
}
}
void AudioFlinger::ThreadBase::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();
}
// ----------------------------------------------------------------------------
// Playback
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
type_t type,
bool systemReady)
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady),
mNormalFrameCount(0), mSinkBuffer(NULL),
mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
mMixerBuffer(NULL),
mMixerBufferSize(0),
mMixerBufferFormat(AUDIO_FORMAT_INVALID),
mMixerBufferValid(false),
mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
mEffectBuffer(NULL),
mEffectBufferSize(0),
mEffectBufferFormat(AUDIO_FORMAT_INVALID),
mEffectBufferValid(false),
mSuspended(0), mBytesWritten(0),
mFramesWritten(0),
mSuspendedFrames(0),
mActiveTracks(&this->mLocalLog),
// mStreamTypes[] initialized in constructor body
mTracks(type == MIXER),
mOutput(output),
mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
mMixerStatus(MIXER_IDLE),
mMixerStatusIgnoringFastTracks(MIXER_IDLE),
mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs),
mBytesRemaining(0),
mCurrentWriteLength(0),
mUseAsyncWrite(false),
mWriteAckSequence(0),
mDrainSequence(0),
mScreenState(AudioFlinger::mScreenState),
// index 0 is reserved for normal mixer's submix
mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1),
mHwSupportsPause(false), mHwPaused(false), mFlushPending(false),
mLeftVolFloat(-1.0), mRightVolFloat(-1.0)
{
snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
// 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_l();
// ++ operator does not compile
for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_FOR_POLICY_CNT;
stream = (audio_stream_type_t) (stream + 1)) {
mStreamTypes[stream].volume = 0.0f;
mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
}
// Audio patch volume is always max
mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
}
AudioFlinger::PlaybackThread::~PlaybackThread()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
free(mSinkBuffer);
free(mMixerBuffer);
free(mEffectBuffer);
}
void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
dumpEffectChains(fd, args);
dprintf(fd, " Local log:\n");
mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
}
void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
{
String8 result;
result.appendFormat(" Stream volumes in dB: ");
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();
// These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
FastTrackUnderruns underruns = getFastTrackUnderruns(0);
dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
size_t numtracks = mTracks.size();
size_t numactive = mActiveTracks.size();
dprintf(fd, " %zu Tracks", numtracks);
size_t numactiveseen = 0;
const char *prefix = " ";
if (numtracks) {
dprintf(fd, " of which %zu are active\n", numactive);
result.append(prefix);
Track::appendDumpHeader(result);
for (size_t i = 0; i < numtracks; ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
bool active = mActiveTracks.indexOf(track) >= 0;
if (active) {
numactiveseen++;
}
result.append(prefix);
track->appendDump(result, active);
}
}
} else {
result.append("\n");
}
if (numactiveseen != numactive) {
// some tracks in the active list were not in the tracks list
result.append(" The following tracks are in the active list but"
" not in the track list\n");
result.append(prefix);
Track::appendDumpHeader(result);
for (size_t i = 0; i < numactive; ++i) {
sp<Track> track = mActiveTracks[i];
if (mTracks.indexOf(track) < 0) {
result.append(prefix);
track->appendDump(result, true /* active */);
}
}
}
write(fd, result.string(), result.size());
}
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
dumpBase(fd, args);
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
dprintf(fd, " Last write occurred (msecs): %llu\n",
(unsigned long long) ns2ms(systemTime() - mLastWriteTime));
dprintf(fd, " Total writes: %d\n", mNumWrites);
dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
dprintf(fd, " Suspend count: %d\n", mSuspended);
dprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs);
AudioStreamOut *output = mOutput;
audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n",
output, flags, outputFlagsToString(flags).c_str());
dprintf(fd, " Frames written: %lld\n", (long long)mFramesWritten);
dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames);
if (mPipeSink.get() != nullptr) {
dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten());
}
if (output != nullptr) {
dprintf(fd, " Hal stream dump:\n");
(void)output->stream->dump(fd);
}
}
// Thread virtuals
void AudioFlinger::PlaybackThread::onFirstRef()
{
run(mThreadName, 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
status_t result = mOutput->stream->setParameters(String8("exiting=1"));
ALOGE_IF(result != OK, "Error when setting parameters on exit: %d", result);
}
// 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,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
size_t *pNotificationFrameCount,
uint32_t notificationsPerBuffer,
float speed,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
audio_output_flags_t *flags,
pid_t tid,
uid_t uid,
status_t *status,
audio_port_handle_t portId)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
sp<Track> track;
status_t lStatus;
audio_output_flags_t outputFlags = mOutput->flags;
audio_output_flags_t requestedFlags = *flags;
if (*pSampleRate == 0) {
*pSampleRate = mSampleRate;
}
uint32_t sampleRate = *pSampleRate;
// special case for FAST flag considered OK if fast mixer is present
if (hasFastMixer()) {
outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST);
}
// Check if requested flags are compatible with output stream flags
if ((*flags & outputFlags) != *flags) {
ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)",
*flags, outputFlags);
*flags = (audio_output_flags_t)(*flags & outputFlags);
}
// client expresses a preference for FAST, but we get the final say
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (
// PCM data
audio_is_linear_pcm(format) &&
// TODO: extract as a data library function that checks that a computationally
// expensive downmixer is not required: isFastOutputChannelConversion()
(channelMask == mChannelMask ||
mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
(channelMask == AUDIO_CHANNEL_OUT_MONO
/* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// 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?
) {
// static tracks can have any nonzero framecount, streaming tracks check against minimum.
if (sharedBuffer == 0) {
// read the fast track multiplier property the first time it is needed
int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
if (ok != 0) {
ALOGE("%s pthread_once failed: %d", __func__, ok);
}
frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
}
// check compatibility with audio effects.
{ // scope for mLock
Mutex::Autolock _l(mLock);
for (audio_session_t session : {
AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_OUTPUT_MIX,
sessionId,
}) {
sp<EffectChain> chain = getEffectChain_l(session);
if (chain.get() != nullptr) {
audio_output_flags_t old = *flags;
chain->checkOutputFlagCompatibility(flags);
if (old != *flags) {
ALOGV("AUDIO_OUTPUT_FLAGS denied by effect, session=%d old=%#x new=%#x",
(int)session, (int)old, (int)*flags);
}
}
}
}
ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0,
"AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
"mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
"sampleRate=%u mSampleRate=%u "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
audio_is_linear_pcm(format),
channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
}
}
if (!audio_has_proportional_frames(format)) {
if (sharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
frameCount = mNormalFrameCount;
}
if (notificationFrameCount != frameCount) {
notificationFrameCount = frameCount;
}
} else if (sharedBuffer != 0) {
// FIXME: Ensure client side memory buffers need
// not have additional alignment beyond sample
// (e.g. 16 bit stereo accessed as 32 bit frame).
size_t alignment = audio_bytes_per_sample(format);
if (alignment & 1) {
// for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
size_t frameSize = channelCount * audio_bytes_per_sample(format);
if (channelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
if (((uintptr_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
sharedBuffer->pointer(), channelCount);
lStatus = BAD_VALUE;
goto Exit;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
frameCount = sharedBuffer->size() / frameSize;
} else {
size_t minFrameCount = 0;
// For fast tracks we try to respect the application's request for notifications per buffer.
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
if (notificationsPerBuffer > 0) {
// Avoid possible arithmetic overflow during multiplication.
if (notificationsPerBuffer > SIZE_MAX / mFrameCount) {
ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",
notificationsPerBuffer, mFrameCount);
} else {
minFrameCount = mFrameCount * notificationsPerBuffer;
}
}
} else {
// For normal PCM streaming tracks, update minimum frame count.
// 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 = latency_l();
if (latencyMs == 0) {
ALOGE("Error when retrieving output stream latency");
lStatus = UNKNOWN_ERROR;
goto Exit;
}
minFrameCount = AudioSystem::calculateMinFrameCount(latencyMs, mNormalFrameCount,
mSampleRate, sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
}
if (frameCount < minFrameCount) {
frameCount = minFrameCount;
}
}
// Make sure that application is notified with sufficient margin before underrun.
// The client can divide the AudioTrack buffer into sub-buffers,
// and expresses its desire to server as the notification frame count.
if (sharedBuffer == 0 && audio_is_linear_pcm(format)) {
size_t maxNotificationFrames;
if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
// notify every HAL buffer, regardless of the size of the track buffer
maxNotificationFrames = mFrameCount;
} else {
// For normal tracks, use at least double-buffering if no sample rate conversion,
// or at least triple-buffering if there is sample rate conversion
const int nBuffering = sampleRate == mSampleRate ? 2 : 3;
maxNotificationFrames = frameCount / nBuffering;
// If client requested a fast track but this was denied, then use the smaller maximum.
if (requestedFlags & AUDIO_OUTPUT_FLAG_FAST) {
size_t maxNotificationFramesFastDenied = FMS_20 * sampleRate / 1000;
if (maxNotificationFrames > maxNotificationFramesFastDenied) {
maxNotificationFrames = maxNotificationFramesFastDenied;
}
}
}
if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) {
if (notificationFrameCount == 0) {
ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",
maxNotificationFrames, frameCount);
} else {
ALOGW("Client adjusted notificationFrames from %zu to %zu for frameCount %zu",
notificationFrameCount, maxNotificationFrames, frameCount);
}
notificationFrameCount = maxNotificationFrames;
}
}
*pFrameCount = frameCount;
*pNotificationFrameCount = notificationFrameCount;
switch (mType) {
case DIRECT:
if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
"for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
}
break;
case OFFLOAD:
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
"for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
break;
default:
if (!audio_is_linear_pcm(format)) {
ALOGE("createTrack_l() Bad parameter: format %#x \""
"for output %p with format %#x",
format, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
break;
}
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() 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->isExternalTrack()) {
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;
}
}
}
track = new Track(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
sessionId, uid, *flags, TrackBase::TYPE_DEFAULT, portId);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
// 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 & AUDIO_OUTPUT_FLAG_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, true /*forApp*/);
}
}
lStatus = NO_ERROR;
Exit:
*status = lStatus;
return track;
}
template<typename T>
ssize_t AudioFlinger::PlaybackThread::Tracks<T>::add(const sp<T> &track)
{
const ssize_t index = mTracks.add(track);
if (index >= 0) {
// set name for track when adding.
int name;
if (mUnusedTrackNames.empty()) {
name = mTracks.size() - 1; // new name {0 ... size-1}.
} else {
// reuse smallest name for deleted track.
auto it = mUnusedTrackNames.begin();
name = *it;
(void)mUnusedTrackNames.erase(it);
}
track->setName(name);
} else {
LOG_ALWAYS_FATAL("cannot add track");
}
return index;
}
template<typename T>
ssize_t AudioFlinger::PlaybackThread::Tracks<T>::remove(const sp<T> &track)
{
const int name = track->name();
const ssize_t index = mTracks.remove(track);
if (index >= 0) {
// invalidate name when removing from mTracks.
LOG_ALWAYS_FATAL_IF(name < 0, "invalid name %d for track on mTracks", name);
if (mSaveDeletedTrackNames) {
// We can't directly access mAudioMixer since the caller may be outside of threadLoop.
// Instead, we add to mDeletedTrackNames which is solely used for mAudioMixer update,
// to be handled when MixerThread::prepareTracks_l() next changes mAudioMixer.
mDeletedTrackNames.emplace(name);
}
mUnusedTrackNames.emplace(name);
track->setName(T::TRACK_NAME_PENDING);
} else {
LOG_ALWAYS_FATAL_IF(name >= 0,
"valid name %d for track not in mTracks (returned %zd)", name, index);
}
return index;
}
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
{
uint32_t latency;
if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) {
return correctLatency_l(latency);
}
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)
{
if (isDuplicating()) {
return;
}
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;
}
void AudioFlinger::PlaybackThread::setVolumeForOutput_l(float left, float right) const
{
mOutput->stream->setVolume(left, right);
}
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
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->isExternalTrack()) {
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
}
// set retry count for buffer fill
if (track->isOffloaded()) {
if (track->isStopping_1()) {
track->mRetryCount = kMaxTrackStopRetriesOffload;
} else {
track->mRetryCount = kMaxTrackStartupRetriesOffload;
}
track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
} else {
track->mRetryCount = kMaxTrackStartupRetries;
track->mFillingUpStatus =
track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
}
track->mResetDone = false;
track->mPresentationCompleteFrames = 0;
mActiveTracks.add(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;
}
onAddNewTrack_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->isDirect()) {
track->mState = TrackBase::STOPPING_1;
}
return trackActive;
}
void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
{
track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
String8 result;
track->appendDump(result, false /* active */);
mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.string());
mTracks.remove(track);
if (track->isFastTrack()) {
int index = track->mFastIndex;
ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks);
ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
mFastTrackAvailMask |= 1 << index;
// redundant as track is about to be destroyed, for dumpsys only
track->mFastIndex = -1;
}
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->decTrackCnt();
}
}
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
Mutex::Autolock _l(mLock);
String8 out_s8;
if (initCheck() == NO_ERROR && mOutput->stream->getParameters(keys, &out_s8) == OK) {
return out_s8;
}
return String8();
}
void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
desc->mIoHandle = mId;
switch (event) {
case AUDIO_OUTPUT_OPENED:
case AUDIO_OUTPUT_REGISTERED:
case AUDIO_OUTPUT_CONFIG_CHANGED:
desc->mPatch = mPatch;
desc->mChannelMask = mChannelMask;
desc->mSamplingRate = mSampleRate;
desc->mFormat = mFormat;
desc->mFrameCount = mNormalFrameCount; // FIXME see
// AudioFlinger::frameCount(audio_io_handle_t)
desc->mFrameCountHAL = mFrameCount;
desc->mLatency = latency_l();
break;
case AUDIO_OUTPUT_CLOSED:
default:
break;
}
mAudioFlinger->ioConfigChanged(event, desc, pid);
}
void AudioFlinger::PlaybackThread::onWriteReady()
{
mCallbackThread->resetWriteBlocked();
}
void AudioFlinger::PlaybackThread::onDrainReady()
{
mCallbackThread->resetDraining();
}
void AudioFlinger::PlaybackThread::onError()
{
mCallbackThread->setAsyncError();
}
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();
}
}
void AudioFlinger::PlaybackThread::readOutputParameters_l()
{
// unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
mSampleRate = mOutput->getSampleRate();
mChannelMask = mOutput->getChannelMask();
if (!audio_is_output_channel(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
}
if ((mType == MIXER || mType == DUPLICATING)
&& !isValidPcmSinkChannelMask(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
mChannelMask);
}
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
// Get actual HAL format.
status_t result = mOutput->stream->getFormat(&mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result);
// Get format from the shim, which will be different than the HAL format
// if playing compressed audio over HDMI passthrough.
mFormat = mOutput->getFormat();
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
}
if ((mType == MIXER || mType == DUPLICATING)
&& !isValidPcmSinkFormat(mFormat)) {
LOG_FATAL("HAL format %#x not supported for mixed output",
mFormat);
}
mFrameSize = mOutput->getFrameSize();
result = mOutput->stream->getBufferSize(&mBufferSize);
LOG_ALWAYS_FATAL_IF(result != OK,
"Error when retrieving output stream buffer size: %d", result);
mFrameCount = mBufferSize / mFrameSize;
if (mFrameCount & 15) {
ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
mFrameCount);
}
if (mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) {
if (mOutput->stream->setCallback(this) == OK) {
mUseAsyncWrite = true;
mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
}
}
mHwSupportsPause = false;
if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) {
bool supportsPause = false, supportsResume = false;
if (mOutput->stream->supportsPauseAndResume(&supportsPause, &supportsResume) == OK) {
if (supportsPause && supportsResume) {
mHwSupportsPause = true;
} else if (supportsPause) {
ALOGW("direct output implements pause but not resume");
} else if (supportsResume) {
ALOGW("direct output implements resume but not pause");
}
}
}
if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) {
LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume");
}
if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
// For best precision, we use float instead of the associated output
// device format (typically PCM 16 bit).
mFormat = AUDIO_FORMAT_PCM_FLOAT;
mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
mBufferSize = mFrameSize * mFrameCount;
// TODO: We currently use the associated output device channel mask and sample rate.
// (1) Perhaps use the ORed channel mask of all downstream MixerThreads
// (if a valid mask) to avoid premature downmix.
// (2) Perhaps use the maximum sample rate of all downstream MixerThreads
// instead of the output device sample rate to avoid loss of high frequency information.
// This may need to be updated as MixerThread/OutputTracks are added and not here.
}
// Calculate size of normal sink buffer relative to the HAL output buffer size
double multiplier = 1.0;
if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
kUseFastMixer == FastMixer_Dynamic)) {
size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * 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 {
multiplier = floor(multiplier);
}
}
mNormalFrameCount = multiplier * mFrameCount;
// round up to nearest 16 frames to satisfy AudioMixer
if (mType == MIXER || mType == DUPLICATING) {
mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
}
ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
mNormalFrameCount);
// Check if we want to throttle the processing to no more than 2x normal rate
mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */);
mThreadThrottleTimeMs = 0;
mThreadThrottleEndMs = 0;
mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate);