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v811_spc009/frameworks/av/services/audioflinger/AudioFlinger.cpp

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/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
// Define AUDIO_ARRAYS_STATIC_CHECK to check all audio arrays are correct
#define AUDIO_ARRAYS_STATIC_CHECK 1
#include "Configuration.h"
#include <dirent.h>
#include <math.h>
#include <signal.h>
#include <string>
#include <sys/time.h>
#include <sys/resource.h>
#include <thread>
#include <android/media/IAudioPolicyService.h>
#include <android/os/IExternalVibratorService.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <binder/Parcel.h>
#include <media/audiohal/DeviceHalInterface.h>
#include <media/audiohal/DevicesFactoryHalInterface.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <media/AudioParameter.h>
#include <media/MediaMetricsItem.h>
#include <media/TypeConverter.h>
#include <mediautils/TimeCheck.h>
#include <memunreachable/memunreachable.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <cutils/atomic.h>
#include <cutils/properties.h>
#include <system/audio.h>
#include <audiomanager/AudioManager.h>
#include "AudioFlinger.h"
#include "NBAIO_Tee.h"
#include <media/AudioResamplerPublic.h>
#include <system/audio_effects/effect_visualizer.h>
#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_aec.h>
#include <system/audio_effects/effect_hapticgenerator.h>
#include <audio_utils/primitives.h>
#include <powermanager/PowerManager.h>
#include <media/IMediaLogService.h>
#include <media/AidlConversion.h>
#include <media/AudioValidator.h>
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
#include <mediautils/BatteryNotifier.h>
#include <mediautils/MemoryLeakTrackUtil.h>
#include <mediautils/ServiceUtilities.h>
#include <mediautils/TimeCheck.h>
#include <private/android_filesystem_config.h>
//#define BUFLOG_NDEBUG 0
#include <BufLog.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
namespace android {
using media::IEffectClient;
using android::content::AttributionSourceState;
static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
static const char kHardwareLockedString[] = "Hardware lock is taken\n";
static const char kClientLockedString[] = "Client lock is taken\n";
static const char kNoEffectsFactory[] = "Effects Factory is absent\n";
nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
uint32_t AudioFlinger::mScreenState;
// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
// we define a minimum time during which a global effect is considered enabled.
static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
// Keep a strong reference to media.log service around forever.
// The service is within our parent process so it can never die in a way that we could observe.
// These two variables are const after initialization.
static sp<IBinder> sMediaLogServiceAsBinder;
static sp<IMediaLogService> sMediaLogService;
static pthread_once_t sMediaLogOnce = PTHREAD_ONCE_INIT;
static void sMediaLogInit()
{
sMediaLogServiceAsBinder = defaultServiceManager()->getService(String16("media.log"));
if (sMediaLogServiceAsBinder != 0) {
sMediaLogService = interface_cast<IMediaLogService>(sMediaLogServiceAsBinder);
}
}
// Keep a strong reference to external vibrator service
static sp<os::IExternalVibratorService> sExternalVibratorService;
static sp<os::IExternalVibratorService> getExternalVibratorService() {
if (sExternalVibratorService == 0) {
sp<IBinder> binder = defaultServiceManager()->getService(
String16("external_vibrator_service"));
if (binder != 0) {
sExternalVibratorService =
interface_cast<os::IExternalVibratorService>(binder);
}
}
return sExternalVibratorService;
}
class DevicesFactoryHalCallbackImpl : public DevicesFactoryHalCallback {
public:
void onNewDevicesAvailable() override {
// Start a detached thread to execute notification in parallel.
// This is done to prevent mutual blocking of audio_flinger and
// audio_policy services during system initialization.
std::thread notifier([]() {
AudioSystem::onNewAudioModulesAvailable();
});
notifier.detach();
}
};
// TODO b/182392769: use attribution source util
/* static */
AttributionSourceState AudioFlinger::checkAttributionSourcePackage(
const AttributionSourceState& attributionSource) {
Vector<String16> packages;
PermissionController{}.getPackagesForUid(attributionSource.uid, packages);
AttributionSourceState checkedAttributionSource = attributionSource;
if (!attributionSource.packageName.has_value()
|| attributionSource.packageName.value().size() == 0) {
if (!packages.isEmpty()) {
checkedAttributionSource.packageName =
std::move(legacy2aidl_String16_string(packages[0]).value());
}
} else {
String16 opPackageLegacy = VALUE_OR_FATAL(
aidl2legacy_string_view_String16(attributionSource.packageName.value_or("")));
if (std::find_if(packages.begin(), packages.end(),
[&opPackageLegacy](const auto& package) {
return opPackageLegacy == package; }) == packages.end()) {
ALOGW("The package name(%s) provided does not correspond to the uid %d",
attributionSource.packageName.value_or("").c_str(), attributionSource.uid);
}
}
return checkedAttributionSource;
}
// ----------------------------------------------------------------------------
std::string formatToString(audio_format_t format) {
std::string result;
FormatConverter::toString(format, result);
return result;
}
// ----------------------------------------------------------------------------
void AudioFlinger::instantiate() {
sp<IServiceManager> sm(defaultServiceManager());
sm->addService(String16(IAudioFlinger::DEFAULT_SERVICE_NAME),
new AudioFlingerServerAdapter(new AudioFlinger()), false,
IServiceManager::DUMP_FLAG_PRIORITY_DEFAULT);
}
AudioFlinger::AudioFlinger()
: mMediaLogNotifier(new AudioFlinger::MediaLogNotifier()),
mPrimaryHardwareDev(NULL),
mAudioHwDevs(NULL),
mHardwareStatus(AUDIO_HW_IDLE),
mMasterVolume(1.0f),
mMasterMute(false),
// mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false),
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
mTotalMemory(0),
mClientSharedHeapSize(kMinimumClientSharedHeapSizeBytes),
mGlobalEffectEnableTime(0),
mPatchPanel(this),
mDeviceEffectManager(this),
mSystemReady(false)
{
// Move the audio session unique ID generator start base as time passes to limit risk of
// generating the same ID again after an audioserver restart.
// This is important because clients will reuse previously allocated audio session IDs
// when reconnecting after an audioserver restart and newly allocated IDs may conflict with
// active clients.
// Moving the base by 1 for each elapsed second is a good compromise between avoiding overlap
// between allocation ranges and not reaching wrap around too soon.
timespec ts{};
clock_gettime(CLOCK_MONOTONIC, &ts);
// zero ID has a special meaning, so start allocation at least at AUDIO_UNIQUE_ID_USE_MAX
uint32_t movingBase = (uint32_t)std::max((long)1, ts.tv_sec);
// unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
mNextUniqueIds[use] =
((use == AUDIO_UNIQUE_ID_USE_SESSION || use == AUDIO_UNIQUE_ID_USE_CLIENT) ?
movingBase : 1) * AUDIO_UNIQUE_ID_USE_MAX;
}
#if 1
// FIXME See bug 165702394 and bug 168511485
const bool doLog = false;
#else
const bool doLog = property_get_bool("ro.test_harness", false);
#endif
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
(void) pthread_once(&sMediaLogOnce, sMediaLogInit);
}
// reset battery stats.
// if the audio service has crashed, battery stats could be left
// in bad state, reset the state upon service start.
BatteryNotifier::getInstance().noteResetAudio();
mDevicesFactoryHal = DevicesFactoryHalInterface::create();
mEffectsFactoryHal = EffectsFactoryHalInterface::create();
mMediaLogNotifier->run("MediaLogNotifier");
std::vector<pid_t> halPids;
mDevicesFactoryHal->getHalPids(&halPids);
TimeCheck::setAudioHalPids(halPids);
// Notify that we have started (also called when audioserver service restarts)
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CTOR)
.record();
}
void AudioFlinger::onFirstRef()
{
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
char val_str[PROPERTY_VALUE_MAX] = { 0 };
if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
uint32_t int_val;
if (1 == sscanf(val_str, "%u", &int_val)) {
mStandbyTimeInNsecs = milliseconds(int_val);
ALOGI("Using %u mSec as standby time.", int_val);
} else {
mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
ALOGI("Using default %u mSec as standby time.",
(uint32_t)(mStandbyTimeInNsecs / 1000000));
}
}
mMode = AUDIO_MODE_NORMAL;
gAudioFlinger = this; // we are already refcounted, store into atomic pointer.
mDevicesFactoryHalCallback = new DevicesFactoryHalCallbackImpl;
mDevicesFactoryHal->setCallbackOnce(mDevicesFactoryHalCallback);
}
status_t AudioFlinger::setAudioHalPids(const std::vector<pid_t>& pids) {
TimeCheck::setAudioHalPids(pids);
return NO_ERROR;
}
status_t AudioFlinger::setVibratorInfos(
const std::vector<media::AudioVibratorInfo>& vibratorInfos) {
Mutex::Autolock _l(mLock);
mAudioVibratorInfos = vibratorInfos;
return NO_ERROR;
}
status_t AudioFlinger::updateSecondaryOutputs(
const TrackSecondaryOutputsMap& trackSecondaryOutputs) {
Mutex::Autolock _l(mLock);
for (const auto& [trackId, secondaryOutputs] : trackSecondaryOutputs) {
size_t i = 0;
for (; i < mPlaybackThreads.size(); ++i) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
Mutex::Autolock _tl(thread->mLock);
sp<PlaybackThread::Track> track = thread->getTrackById_l(trackId);
if (track != nullptr) {
ALOGD("%s trackId: %u", __func__, trackId);
updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
break;
}
}
ALOGW_IF(i >= mPlaybackThreads.size(),
"%s cannot find track with id %u", __func__, trackId);
}
return NO_ERROR;
}
// getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
const media::AudioVibratorInfo* AudioFlinger::getDefaultVibratorInfo_l() {
if (mAudioVibratorInfos.empty()) {
return nullptr;
}
return &mAudioVibratorInfos.front();
}
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
// closeInput_nonvirtual() will remove specified entry from mRecordThreads
closeInput_nonvirtual(mRecordThreads.keyAt(0));
}
while (!mPlaybackThreads.isEmpty()) {
// closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
}
while (!mMmapThreads.isEmpty()) {
const audio_io_handle_t io = mMmapThreads.keyAt(0);
if (mMmapThreads.valueAt(0)->isOutput()) {
closeOutput_nonvirtual(io); // removes entry from mMmapThreads
} else {
closeInput_nonvirtual(io); // removes entry from mMmapThreads
}
}
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
// no mHardwareLock needed, as there are no other references to this
delete mAudioHwDevs.valueAt(i);
}
// Tell media.log service about any old writers that still need to be unregistered
if (sMediaLogService != 0) {
for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
mUnregisteredWriters.pop();
sMediaLogService->unregisterWriter(iMemory);
}
}
}
//static
__attribute__ ((visibility ("default")))
status_t MmapStreamInterface::openMmapStream(MmapStreamInterface::stream_direction_t direction,
const audio_attributes_t *attr,
audio_config_base_t *config,
const AudioClient& client,
audio_port_handle_t *deviceId,
audio_session_t *sessionId,
const sp<MmapStreamCallback>& callback,
sp<MmapStreamInterface>& interface,
audio_port_handle_t *handle)
{
// TODO: Use ServiceManager to get IAudioFlinger instead of by atomic pointer.
// This allows moving oboeservice (AAudio) to a separate process in the future.
sp<AudioFlinger> af = AudioFlinger::gAudioFlinger.load(); // either nullptr or singleton AF.
status_t ret = NO_INIT;
if (af != 0) {
ret = af->openMmapStream(
direction, attr, config, client, deviceId,
sessionId, callback, interface, handle);
}
return ret;
}
status_t AudioFlinger::openMmapStream(MmapStreamInterface::stream_direction_t direction,
const audio_attributes_t *attr,
audio_config_base_t *config,
const AudioClient& client,
audio_port_handle_t *deviceId,
audio_session_t *sessionId,
const sp<MmapStreamCallback>& callback,
sp<MmapStreamInterface>& interface,
audio_port_handle_t *handle)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
audio_session_t actualSessionId = *sessionId;
if (actualSessionId == AUDIO_SESSION_ALLOCATE) {
actualSessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
}
audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT;
audio_io_handle_t io = AUDIO_IO_HANDLE_NONE;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
audio_attributes_t localAttr = *attr;
// TODO b/182392553: refactor or make clearer
pid_t clientPid =
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(client.attributionSource.pid));
bool updatePid = (clientPid == (pid_t)-1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
AttributionSourceState adjAttributionSource = client.attributionSource;
if (!isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
uid_t clientUid =
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(client.attributionSource.uid));
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
updatePid = true;
}
if (updatePid) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(clientPid != (pid_t)-1 && clientPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, clientPid);
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
}
adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
adjAttributionSource);
if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
audio_config_t fullConfig = AUDIO_CONFIG_INITIALIZER;
fullConfig.sample_rate = config->sample_rate;
fullConfig.channel_mask = config->channel_mask;
fullConfig.format = config->format;
std::vector<audio_io_handle_t> secondaryOutputs;
ret = AudioSystem::getOutputForAttr(&localAttr, &io,
actualSessionId,
&streamType, adjAttributionSource,
&fullConfig,
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
AUDIO_OUTPUT_FLAG_DIRECT),
deviceId, &portId, &secondaryOutputs);
ALOGW_IF(!secondaryOutputs.empty(),
"%s does not support secondary outputs, ignoring them", __func__);
} else {
ret = AudioSystem::getInputForAttr(&localAttr, &io,
RECORD_RIID_INVALID,
actualSessionId,
adjAttributionSource,
config,
AUDIO_INPUT_FLAG_MMAP_NOIRQ, deviceId, &portId);
}
if (ret != NO_ERROR) {
return ret;
}
// at this stage, a MmapThread was created when openOutput() or openInput() was called by
// audio policy manager and we can retrieve it
sp<MmapThread> thread = mMmapThreads.valueFor(io);
if (thread != 0) {
interface = new MmapThreadHandle(thread);
thread->configure(&localAttr, streamType, actualSessionId, callback, *deviceId, portId);
*handle = portId;
*sessionId = actualSessionId;
config->sample_rate = thread->sampleRate();
config->channel_mask = thread->channelMask();
config->format = thread->format();
} else {
if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
AudioSystem::releaseOutput(portId);
} else {
AudioSystem::releaseInput(portId);
}
ret = NO_INIT;
}
ALOGV("%s done status %d portId %d", __FUNCTION__, ret, portId);
return ret;
}
/* static */
int AudioFlinger::onExternalVibrationStart(const sp<os::ExternalVibration>& externalVibration) {
sp<os::IExternalVibratorService> evs = getExternalVibratorService();
if (evs != nullptr) {
int32_t ret;
binder::Status status = evs->onExternalVibrationStart(*externalVibration, &ret);
if (status.isOk()) {
ALOGD("%s, start external vibration with intensity as %d", __func__, ret);
return ret;
}
}
ALOGD("%s, start external vibration with intensity as MUTE due to %s",
__func__,
evs == nullptr ? "external vibration service not found"
: "error when querying intensity");
return static_cast<int>(os::HapticScale::MUTE);
}
/* static */
void AudioFlinger::onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration) {
sp<os::IExternalVibratorService> evs = getExternalVibratorService();
if (evs != 0) {
evs->onExternalVibrationStop(*externalVibration);
}
}
status_t AudioFlinger::addEffectToHal(audio_port_handle_t deviceId,
audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(hwModuleId);
if (audioHwDevice == nullptr) {
return NO_INIT;
}
return audioHwDevice->hwDevice()->addDeviceEffect(deviceId, effect);
}
status_t AudioFlinger::removeEffectFromHal(audio_port_handle_t deviceId,
audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(hwModuleId);
if (audioHwDevice == nullptr) {
return NO_INIT;
}
return audioHwDevice->hwDevice()->removeDeviceEffect(deviceId, effect);
}
static const char * const audio_interfaces[] = {
AUDIO_HARDWARE_MODULE_ID_PRIMARY,
AUDIO_HARDWARE_MODULE_ID_A2DP,
AUDIO_HARDWARE_MODULE_ID_USB,
};
AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
audio_devices_t deviceType)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
AutoMutex lock(mHardwareLock);
if (module == 0) {
ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
for (size_t i = 0; i < arraysize(audio_interfaces); i++) {
loadHwModule_l(audio_interfaces[i]);
}
// then try to find a module supporting the requested device.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
sp<DeviceHalInterface> dev = audioHwDevice->hwDevice();
uint32_t supportedDevices;
if (dev->getSupportedDevices(&supportedDevices) == OK &&
(supportedDevices & deviceType) == deviceType) {
return audioHwDevice;
}
}
} else {
// check a match for the requested module handle
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
if (audioHwDevice != NULL) {
return audioHwDevice;
}
}
return NULL;
}
void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
{
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
sp<Client> client = mClients.valueAt(i).promote();
if (client != 0) {
result.appendFormat(" pid: %d\n", client->pid());
}
}
result.append("Notification Clients:\n");
result.append(" pid uid name\n");
for (size_t i = 0; i < mNotificationClients.size(); ++i) {
const pid_t pid = mNotificationClients[i]->getPid();
const uid_t uid = mNotificationClients[i]->getUid();
const mediautils::UidInfo::Info info = mUidInfo.getInfo(uid);
result.appendFormat("%6d %6u %s\n", pid, uid, info.package.c_str());
}
result.append("Global session refs:\n");
result.append(" session cnt pid uid name\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
const mediautils::UidInfo::Info info = mUidInfo.getInfo(r->mUid);
result.appendFormat(" %7d %4d %7d %6u %s\n", r->mSessionid, r->mCnt, r->mPid,
r->mUid, info.package.c_str());
}
write(fd, result.string(), result.size());
}
void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
hardware_call_state hardwareStatus = mHardwareStatus;
snprintf(buffer, SIZE, "Hardware status: %d\n"
"Standby Time mSec: %u\n",
hardwareStatus,
(uint32_t)(mStandbyTimeInNsecs / 1000000));
result.append(buffer);
write(fd, result.string(), result.size());
}
void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump AudioFlinger from pid=%d, uid=%d\n",
IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
}
bool AudioFlinger::dumpTryLock(Mutex& mutex)
{
status_t err = mutex.timedLock(kDumpLockTimeoutNs);
return err == NO_ERROR;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
bool hardwareLocked = dumpTryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
const bool locked = dumpTryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
}
bool clientLocked = dumpTryLock(mClientLock);
if (!clientLocked) {
String8 result(kClientLockedString);
write(fd, result.string(), result.size());
}
if (mEffectsFactoryHal != 0) {
mEffectsFactoryHal->dumpEffects(fd);
} else {
String8 result(kNoEffectsFactory);
write(fd, result.string(), result.size());
}
dumpClients(fd, args);
if (clientLocked) {
mClientLock.unlock();
}
dumpInternals(fd, args);
// dump playback threads
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->dump(fd, args);
}
// dump record threads
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->dump(fd, args);
}
// dump mmap threads
for (size_t i = 0; i < mMmapThreads.size(); i++) {
mMmapThreads.valueAt(i)->dump(fd, args);
}
// dump orphan effect chains
if (mOrphanEffectChains.size() != 0) {
write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n"));
for (size_t i = 0; i < mOrphanEffectChains.size(); i++) {
mOrphanEffectChains.valueAt(i)->dump(fd, args);
}
}
// dump all hardware devs
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
dev->dump(fd);
}
mPatchPanel.dump(fd);
mDeviceEffectManager.dump(fd);
// dump external setParameters
auto dumpLogger = [fd](SimpleLog& logger, const char* name) {
dprintf(fd, "\n%s setParameters:\n", name);
logger.dump(fd, " " /* prefix */);
};
dumpLogger(mRejectedSetParameterLog, "Rejected");
dumpLogger(mAppSetParameterLog, "App");
dumpLogger(mSystemSetParameterLog, "System");
// dump historical threads in the last 10 seconds
const std::string threadLog = mThreadLog.dumpToString(
"Historical Thread Log ", 0 /* lines */,
audio_utils_get_real_time_ns() - 10 * 60 * NANOS_PER_SECOND);
write(fd, threadLog.c_str(), threadLog.size());
BUFLOG_RESET;
if (locked) {
mLock.unlock();
}
#ifdef TEE_SINK
// NBAIO_Tee dump is safe to call outside of AF lock.
NBAIO_Tee::dumpAll(fd, "_DUMP");
#endif
// append a copy of media.log here by forwarding fd to it, but don't attempt
// to lookup the service if it's not running, as it will block for a second
if (sMediaLogServiceAsBinder != 0) {
dprintf(fd, "\nmedia.log:\n");
Vector<String16> args;
sMediaLogServiceAsBinder->dump(fd, args);
}
// check for optional arguments
bool dumpMem = false;
bool unreachableMemory = false;
for (const auto &arg : args) {
if (arg == String16("-m")) {
dumpMem = true;
} else if (arg == String16("--unreachable")) {
unreachableMemory = true;
}
}
if (dumpMem) {
dprintf(fd, "\nDumping memory:\n");
std::string s = dumpMemoryAddresses(100 /* limit */);
write(fd, s.c_str(), s.size());
}
if (unreachableMemory) {
dprintf(fd, "\nDumping unreachable memory:\n");
// TODO - should limit be an argument parameter?
std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */);
write(fd, s.c_str(), s.size());
}
}
return NO_ERROR;
}
sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
{
Mutex::Autolock _cl(mClientLock);
// If pid is already in the mClients wp<> map, then use that entry
// (for which promote() is always != 0), otherwise create a new entry and Client.
sp<Client> client = mClients.valueFor(pid).promote();
if (client == 0) {
client = new Client(this, pid);
mClients.add(pid, client);
}
return client;
}
sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
{
// If there is no memory allocated for logs, return a no-op writer that does nothing.
// Similarly if we can't contact the media.log service, also return a no-op writer.
if (mLogMemoryDealer == 0 || sMediaLogService == 0) {
return new NBLog::Writer();
}
sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
// If allocation fails, consult the vector of previously unregistered writers
// and garbage-collect one or more them until an allocation succeeds
if (shared == 0) {
Mutex::Autolock _l(mUnregisteredWritersLock);
for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
{
// Pick the oldest stale writer to garbage-collect
sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
mUnregisteredWriters.removeAt(0);
sMediaLogService->unregisterWriter(iMemory);
// Now the media.log remote reference to IMemory is gone. When our last local
// reference to IMemory also drops to zero at end of this block,
// the IMemory destructor will deallocate the region from mLogMemoryDealer.
}
// Re-attempt the allocation
shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
if (shared != 0) {
goto success;
}
}
// Even after garbage-collecting all old writers, there is still not enough memory,
// so return a no-op writer
return new NBLog::Writer();
}
success:
NBLog::Shared *sharedRawPtr = (NBLog::Shared *) shared->unsecurePointer();
new((void *) sharedRawPtr) NBLog::Shared(); // placement new here, but the corresponding
// explicit destructor not needed since it is POD
sMediaLogService->registerWriter(shared, size, name);
return new NBLog::Writer(shared, size);
}
void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
{
if (writer == 0) {
return;
}
sp<IMemory> iMemory(writer->getIMemory());
if (iMemory == 0) {
return;
}
// Rather than removing the writer immediately, append it to a queue of old writers to
// be garbage-collected later. This allows us to continue to view old logs for a while.
Mutex::Autolock _l(mUnregisteredWritersLock);
mUnregisteredWriters.push(writer);
}
// IAudioFlinger interface
status_t AudioFlinger::createTrack(const media::CreateTrackRequest& _input,
media::CreateTrackResponse& _output)
{
// Local version of VALUE_OR_RETURN, specific to this method's calling conventions.
CreateTrackInput input = VALUE_OR_RETURN_STATUS(CreateTrackInput::fromAidl(_input));
CreateTrackOutput output;
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
status_t lStatus;
audio_stream_type_t streamType;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
std::vector<audio_io_handle_t> secondaryOutputs;
// TODO b/182392553: refactor or make clearer
pid_t clientPid =
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(input.clientInfo.attributionSource.pid));
bool updatePid = (clientPid == (pid_t)-1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
uid_t clientUid =
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(input.clientInfo.attributionSource.uid));
audio_io_handle_t effectThreadId = AUDIO_IO_HANDLE_NONE;
std::vector<int> effectIds;
audio_attributes_t localAttr = input.attr;
AttributionSourceState adjAttributionSource = input.clientInfo.attributionSource;
if (!isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
clientUid = callingUid;
updatePid = true;
}
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
if (updatePid) {
ALOGW_IF(clientPid != (pid_t)-1 && clientPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, clientPid);
clientPid = callingPid;
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
}
adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
adjAttributionSource);
audio_session_t sessionId = input.sessionId;
if (sessionId == AUDIO_SESSION_ALLOCATE) {
sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
} else if (audio_unique_id_get_use(sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
lStatus = BAD_VALUE;
goto Exit;
}
output.sessionId = sessionId;
output.outputId = AUDIO_IO_HANDLE_NONE;
output.selectedDeviceId = input.selectedDeviceId;
lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
adjAttributionSource, &input.config, input.flags,
&output.selectedDeviceId, &portId, &secondaryOutputs);
if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
goto Exit;
}
// client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
// but if someone uses binder directly they could bypass that and cause us to crash
if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
// further channel mask checks are performed by createTrack_l() depending on the thread type
if (!audio_is_output_channel(input.config.channel_mask)) {
ALOGE("createTrack() invalid channel mask %#x", input.config.channel_mask);
lStatus = BAD_VALUE;
goto Exit;
}
// further format checks are performed by createTrack_l() depending on the thread type
if (!audio_is_valid_format(input.config.format)) {
ALOGE("createTrack() invalid format %#x", input.config.format);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output.outputId);
lStatus = BAD_VALUE;
goto Exit;
}
client = registerPid(clientPid);
PlaybackThread *effectThread = NULL;
// check if an effect chain with the same session ID is present on another
// output thread and move it here.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output.outputId) {
uint32_t sessions = t->hasAudioSession(sessionId);
if (sessions & ThreadBase::EFFECT_SESSION) {
effectThread = t.get();
break;
}
}
}
ALOGV("createTrack() sessionId: %d", sessionId);
output.sampleRate = input.config.sample_rate;
output.frameCount = input.frameCount;
output.notificationFrameCount = input.notificationFrameCount;
output.flags = input.flags;
output.streamType = streamType;
track = thread->createTrack_l(client, streamType, localAttr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, &output.notificationFrameCount,
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
callingPid, adjAttributionSource, input.clientInfo.clientTid,
&lStatus, portId, input.audioTrackCallback);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
output.afFrameCount = thread->frameCount();
output.afSampleRate = thread->sampleRate();
output.afLatencyMs = thread->latency();
output.portId = portId;
if (lStatus == NO_ERROR) {
// Connect secondary outputs. Failure on a secondary output must not imped the primary
// Any secondary output setup failure will lead to a desync between the AP and AF until
// the track is destroyed.
updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
}
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
// no risk of deadlock because AudioFlinger::mLock is held
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
if (moveEffectChain_l(sessionId, effectThread, thread) == NO_ERROR) {
effectThreadId = thread->id();
effectIds = thread->getEffectIds_l(sessionId);
}
}
// Look for sync events awaiting for a session to be used.
for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
if (mPendingSyncEvents[i]->triggerSession() == sessionId) {
if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
if (lStatus == NO_ERROR) {
(void) track->setSyncEvent(mPendingSyncEvents[i]);
} else {
mPendingSyncEvents[i]->cancel();
}
mPendingSyncEvents.removeAt(i);
i--;
}
}
}
setAudioHwSyncForSession_l(thread, sessionId);
}
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the Track so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
// Don't hold mClientLock when releasing the reference on the track as the
// destructor will acquire it.
{
Mutex::Autolock _cl(mClientLock);
client.clear();
}
track.clear();
goto Exit;
}
// effectThreadId is not NONE if an effect chain corresponding to the track session
// was found on another thread and must be moved on this thread
if (effectThreadId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
}
output.audioTrack = new TrackHandle(track);
_output = VALUE_OR_FATAL(output.toAidl());
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(portId);
}
return lStatus;
}
uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
ThreadBase *thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("sampleRate() unknown thread %d", ioHandle);
return 0;
}
return thread->sampleRate();
}
audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("format() unknown thread %d", output);
return AUDIO_FORMAT_INVALID;
}
return thread->format();
}
size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
ThreadBase *thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("frameCount() unknown thread %d", ioHandle);
return 0;
}
// FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
// should examine all callers and fix them to handle smaller counts
return thread->frameCount();
}
size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
ThreadBase *thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("frameCountHAL() unknown thread %d", ioHandle);
return 0;
}
return thread->frameCountHAL();
}
uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("latency(): no playback thread found for output handle %d", output);
return 0;
}
return thread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
mMasterVolume = value;
// Set master volume in the HALs which support it.
{
AutoMutex lock(mHardwareLock);
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (dev->canSetMasterVolume()) {
dev->hwDevice()->setMasterVolume(value);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
}
// Now set the master volume in each playback thread. Playback threads
// assigned to HALs which do not have master volume support will apply
// master volume during the mix operation. Threads with HALs which do
// support master volume will simply ignore the setting.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
continue;
}
mPlaybackThreads.valueAt(i)->setMasterVolume(value);
}
return NO_ERROR;
}
status_t AudioFlinger::setMasterBalance(float balance)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// check range
if (isnan(balance) || fabs(balance) > 1.f) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
// short cut.
if (mMasterBalance == balance) return NO_ERROR;
mMasterBalance = balance;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
continue;
}
mPlaybackThreads.valueAt(i)->setMasterBalance(balance);
}
return NO_ERROR;
}
status_t AudioFlinger::setMode(audio_mode_t mode)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(mode) >= AUDIO_MODE_CNT) {
ALOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
{ // scope for the lock
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return INVALID_OPERATION;
}
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = dev->setMode(mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
}
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETMODE)
.set(AMEDIAMETRICS_PROP_AUDIOMODE, toString(mode))
.record();
return ret;
}
status_t AudioFlinger::setMicMute(bool state)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return INVALID_OPERATION;
}
sp<DeviceHalInterface> primaryDev = mPrimaryHardwareDev->hwDevice();
if (primaryDev == nullptr) {
ALOGW("%s: no primary HAL device", __func__);
return INVALID_OPERATION;
}
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
ret = primaryDev->setMicMute(state);
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
if (dev != primaryDev) {
(void)dev->setMicMute(state);
}
}
mHardwareStatus = AUDIO_HW_IDLE;
ALOGW_IF(ret != NO_ERROR, "%s: error %d setting state to HAL", __func__, ret);
return ret;
}
bool AudioFlinger::getMicMute() const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return false;
}
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return false;
}
sp<DeviceHalInterface> primaryDev = mPrimaryHardwareDev->hwDevice();
if (primaryDev == nullptr) {
ALOGW("%s: no primary HAL device", __func__);
return false;
}
bool state;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
ret = primaryDev->getMicMute(&state);
mHardwareStatus = AUDIO_HW_IDLE;
ALOGE_IF(ret != NO_ERROR, "%s: error %d getting state from HAL", __func__, ret);
return (ret == NO_ERROR) && state;
}
void AudioFlinger::setRecordSilenced(audio_port_handle_t portId, bool silenced)
{
ALOGV("AudioFlinger::setRecordSilenced(portId:%d, silenced:%d)", portId, silenced);
AutoMutex lock(mLock);
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads[i]->setRecordSilenced(portId, silenced);
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
mMmapThreads[i]->setRecordSilenced(portId, silenced);
}
}
status_t AudioFlinger::setMasterMute(bool muted)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
mMasterMute = muted;
// Set master mute in the HALs which support it.
{
AutoMutex lock(mHardwareLock);
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if (dev->canSetMasterMute()) {
dev->hwDevice()->setMasterMute(muted);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
}
// Now set the master mute in each playback thread. Playback threads
// assigned to HALs which do not have master mute support will apply master mute
// during the mix operation. Threads with HALs which do support master mute
// will simply ignore the setting.
Vector<VolumeInterface *> volumeInterfaces = getAllVolumeInterfaces_l();
for (size_t i = 0; i < volumeInterfaces.size(); i++) {
volumeInterfaces[i]->setMasterMute(muted);
}
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
Mutex::Autolock _l(mLock);
return masterVolume_l();
}
status_t AudioFlinger::getMasterBalance(float *balance) const
{
Mutex::Autolock _l(mLock);
*balance = getMasterBalance_l();
return NO_ERROR; // if called through binder, may return a transactional error
}
bool AudioFlinger::masterMute() const
{
Mutex::Autolock _l(mLock);
return masterMute_l();
}
float AudioFlinger::masterVolume_l() const
{
return mMasterVolume;
}
float AudioFlinger::getMasterBalance_l() const
{
return mMasterBalance;
}
bool AudioFlinger::masterMute_l() const
{
return mMasterMute;
}
status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
ALOGW("checkStreamType() invalid stream %d", stream);
return BAD_VALUE;
}
const uid_t callerUid = IPCThreadState::self()->getCallingUid();
if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && !isAudioServerUid(callerUid)) {
ALOGW("checkStreamType() uid %d cannot use internal stream type %d", callerUid, stream);
return PERMISSION_DENIED;
}
return NO_ERROR;
}
status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return status;
}
if (output == AUDIO_IO_HANDLE_NONE) {
return BAD_VALUE;
}
LOG_ALWAYS_FATAL_IF(stream == AUDIO_STREAM_PATCH && value != 1.0f,
"AUDIO_STREAM_PATCH must have full scale volume");
AutoMutex lock(mLock);
VolumeInterface *volumeInterface = getVolumeInterface_l(output);
if (volumeInterface == NULL) {
return BAD_VALUE;
}
volumeInterface->setStreamVolume(stream, value);
return NO_ERROR;
}
status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return status;
}
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH");
if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
ALOGE("setStreamMute() invalid stream %d", stream);
return BAD_VALUE;
}
AutoMutex lock(mLock);
mStreamTypes[stream].mute = muted;
Vector<VolumeInterface *> volumeInterfaces = getAllVolumeInterfaces_l();
for (size_t i = 0; i < volumeInterfaces.size(); i++) {
volumeInterfaces[i]->setStreamMute(stream, muted);
}
return NO_ERROR;
}
float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return 0.0f;
}
if (output == AUDIO_IO_HANDLE_NONE) {
return 0.0f;
}
AutoMutex lock(mLock);
VolumeInterface *volumeInterface = getVolumeInterface_l(output);
if (volumeInterface == NULL) {
return 0.0f;
}
return volumeInterface->streamVolume(stream);
}
bool AudioFlinger::streamMute(audio_stream_type_t stream) const
{
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return true;
}
AutoMutex lock(mLock);
return streamMute_l(stream);
}
void AudioFlinger::broadcastParametersToRecordThreads_l(const String8& keyValuePairs)
{
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
}
}
void AudioFlinger::updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices)
{
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->updateOutDevices(devices);
}
}
// forwardAudioHwSyncToDownstreamPatches_l() must be called with AudioFlinger::mLock held
void AudioFlinger::forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
std::function<bool(const sp<PlaybackThread>&)> useThread)
{
std::vector<PatchPanel::SoftwarePatch> swPatches;
if (mPatchPanel.getDownstreamSoftwarePatches(upStream, &swPatches) != OK) return;
ALOGV_IF(!swPatches.empty(), "%s found %zu downstream patches for stream ID %d",
__func__, swPatches.size(), upStream);
for (const auto& swPatch : swPatches) {
sp<PlaybackThread> downStream = checkPlaybackThread_l(swPatch.getPlaybackThreadHandle());
if (downStream != NULL && (useThread == nullptr || useThread(downStream))) {
downStream->setParameters(keyValuePairs);
}
}
}
// Update downstream patches for all playback threads attached to an MSD module
void AudioFlinger::updateDownStreamPatches_l(const struct audio_patch *patch,
const std::set<audio_io_handle_t> streams)
{
for (const audio_io_handle_t stream : streams) {
PlaybackThread *playbackThread = checkPlaybackThread_l(stream);
if (playbackThread == nullptr || !playbackThread->isMsdDevice()) {
continue;
}
playbackThread->setDownStreamPatch(patch);
playbackThread->sendIoConfigEvent(AUDIO_OUTPUT_CONFIG_CHANGED);
}
}
// Filter reserved keys from setParameters() before forwarding to audio HAL or acting upon.
// Some keys are used for audio routing and audio path configuration and should be reserved for use
// by audio policy and audio flinger for functional, privacy and security reasons.
void AudioFlinger::filterReservedParameters(String8& keyValuePairs, uid_t callingUid)
{
static const String8 kReservedParameters[] = {
String8(AudioParameter::keyRouting),
String8(AudioParameter::keySamplingRate),
String8(AudioParameter::keyFormat),
String8(AudioParameter::keyChannels),
String8(AudioParameter::keyFrameCount),
String8(AudioParameter::keyInputSource),
String8(AudioParameter::keyMonoOutput),
String8(AudioParameter::keyDeviceConnect),
String8(AudioParameter::keyDeviceDisconnect),
String8(AudioParameter::keyStreamSupportedFormats),
String8(AudioParameter::keyStreamSupportedChannels),
String8(AudioParameter::keyStreamSupportedSamplingRates),
};
if (isAudioServerUid(callingUid)) {
return; // no need to filter if audioserver.
}
AudioParameter param = AudioParameter(keyValuePairs);
String8 value;
AudioParameter rejectedParam;
for (auto& key : kReservedParameters) {
if (param.get(key, value) == NO_ERROR) {
rejectedParam.add(key, value);
param.remove(key);
}
}
logFilteredParameters(param.size() + rejectedParam.size(), keyValuePairs,
rejectedParam.size(), rejectedParam.toString(), callingUid);
keyValuePairs = param.toString();
}
void AudioFlinger::logFilteredParameters(size_t originalKVPSize, const String8& originalKVPs,
size_t rejectedKVPSize, const String8& rejectedKVPs,
uid_t callingUid) {
auto prefix = String8::format("UID %5d", callingUid);
auto suffix = String8::format("%zu KVP received: %s", originalKVPSize, originalKVPs.c_str());
if (rejectedKVPSize != 0) {
auto error = String8::format("%zu KVP rejected: %s", rejectedKVPSize, rejectedKVPs.c_str());
ALOGW("%s: %s, %s, %s", __func__, prefix.c_str(), error.c_str(), suffix.c_str());
mRejectedSetParameterLog.log("%s, %s, %s", prefix.c_str(), error.c_str(), suffix.c_str());
} else {
auto& logger = (isServiceUid(callingUid) ? mSystemSetParameterLog : mAppSetParameterLog);
logger.log("%s, %s", prefix.c_str(), suffix.c_str());
}
}
status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d calling uid %d",
ioHandle, keyValuePairs.string(),
IPCThreadState::self()->getCallingPid(), IPCThreadState::self()->getCallingUid());
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
String8 filteredKeyValuePairs = keyValuePairs;
filterReservedParameters(filteredKeyValuePairs, IPCThreadState::self()->getCallingUid());
ALOGV("%s: filtered keyvalue %s", __func__, filteredKeyValuePairs.string());
// AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
if (ioHandle == AUDIO_IO_HANDLE_NONE) {
Mutex::Autolock _l(mLock);
// result will remain NO_INIT if no audio device is present
status_t final_result = NO_INIT;
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_PARAMETER;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->setParameters(filteredKeyValuePairs);
// return success if at least one audio device accepts the parameters as not all
// HALs are requested to support all parameters. If no audio device supports the
// requested parameters, the last error is reported.
if (final_result != NO_ERROR) {
final_result = result;
}
}
mHardwareStatus = AUDIO_HW_IDLE;
}
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
AudioParameter param = AudioParameter(filteredKeyValuePairs);
String8 value;
if (param.get(String8(AudioParameter::keyBtNrec), value) == NO_ERROR) {
bool btNrecIsOff = (value == AudioParameter::valueOff);
if (mBtNrecIsOff.exchange(btNrecIsOff) != btNrecIsOff) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->checkBtNrec();
}
}
}
String8 screenState;
if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
bool isOff = (screenState == AudioParameter::valueOff);
if (isOff != (AudioFlinger::mScreenState & 1)) {
AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
}
}
return final_result;
}
// hold a strong ref on thread in case closeOutput() or closeInput() is called
// and the thread is exited once the lock is released
sp<ThreadBase> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(ioHandle);
if (thread == 0) {
thread = checkRecordThread_l(ioHandle);
if (thread == 0) {
thread = checkMmapThread_l(ioHandle);
}
} else if (thread == primaryPlaybackThread_l()) {
// indicate output device change to all input threads for pre processing
AudioParameter param = AudioParameter(filteredKeyValuePairs);
int value;
if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
(value != 0)) {
broadcastParametersToRecordThreads_l(filteredKeyValuePairs);
}
}
}
if (thread != 0) {
status_t result = thread->setParameters(filteredKeyValuePairs);
forwardParametersToDownstreamPatches_l(thread->id(), filteredKeyValuePairs);
return result;
}
return BAD_VALUE;
}
String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
ALOGVV("getParameters() io %d, keys %s, calling pid %d",
ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
if (ioHandle == AUDIO_IO_HANDLE_NONE) {
String8 out_s8;
AutoMutex lock(mHardwareLock);
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
String8 s;
mHardwareStatus = AUDIO_HW_GET_PARAMETER;
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->getParameters(keys, &s);
mHardwareStatus = AUDIO_HW_IDLE;
if (result == OK) out_s8 += s;
}
return out_s8;
}
ThreadBase *thread = (ThreadBase *)checkPlaybackThread_l(ioHandle);
if (thread == NULL) {
thread = (ThreadBase *)checkRecordThread_l(ioHandle);
if (thread == NULL) {
thread = (ThreadBase *)checkMmapThread_l(ioHandle);
if (thread == NULL) {
return String8("");
}
}
}
return thread->getParameters(keys);
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
if ((sampleRate == 0) ||
!audio_is_valid_format(format) || !audio_has_proportional_frames(format) ||
!audio_is_input_channel(channelMask)) {
return 0;
}
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return 0;
}
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
std::vector<audio_channel_mask_t> channelMasks = {channelMask};
if (channelMask != AUDIO_CHANNEL_IN_MONO)
channelMasks.push_back(AUDIO_CHANNEL_IN_MONO);
if (channelMask != AUDIO_CHANNEL_IN_STEREO)
channelMasks.push_back(AUDIO_CHANNEL_IN_STEREO);
std::vector<audio_format_t> formats = {format};
if (format != AUDIO_FORMAT_PCM_16_BIT)
formats.push_back(AUDIO_FORMAT_PCM_16_BIT);
std::vector<uint32_t> sampleRates = {sampleRate};
static const uint32_t SR_44100 = 44100;
static const uint32_t SR_48000 = 48000;
if (sampleRate != SR_48000)
sampleRates.push_back(SR_48000);
if (sampleRate != SR_44100)
sampleRates.push_back(SR_44100);
mHardwareStatus = AUDIO_HW_IDLE;
// Change parameters of the configuration each iteration until we find a
// configuration that the device will support.
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
for (auto testChannelMask : channelMasks) {
config.channel_mask = testChannelMask;
for (auto testFormat : formats) {
config.format = testFormat;
for (auto testSampleRate : sampleRates) {
config.sample_rate = testSampleRate;
size_t bytes = 0;
status_t result = dev->getInputBufferSize(&config, &bytes);
if (result != OK || bytes == 0) {
continue;
}
if (config.sample_rate != sampleRate || config.channel_mask != channelMask ||
config.format != format) {
uint32_t dstChannelCount = audio_channel_count_from_in_mask(channelMask);
uint32_t srcChannelCount =
audio_channel_count_from_in_mask(config.channel_mask);
size_t srcFrames =
bytes / audio_bytes_per_frame(srcChannelCount, config.format);
size_t dstFrames = destinationFramesPossible(
srcFrames, config.sample_rate, sampleRate);
bytes = dstFrames * audio_bytes_per_frame(dstChannelCount, format);
}
return bytes;
}
}
}
ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
"format %#x, channelMask %#x",sampleRate, format, channelMask);
return 0;
}
uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getInputFramesLost();
}
return 0;
}
status_t AudioFlinger::setVoiceVolume(float value)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return INVALID_OPERATION;
}
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
ret = dev->setVoiceVolume(value);
mHardwareStatus = AUDIO_HW_IDLE;
mediametrics::LogItem(mMetricsId)
.set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOICEVOLUME)
.set(AMEDIAMETRICS_PROP_VOICEVOLUME, (double)value)
.record();
return ret;
}
status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
return playbackThread->getRenderPosition(halFrames, dspFrames);
}
return BAD_VALUE;
}
void AudioFlinger::registerClient(const sp<media::IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
if (client == 0) {
return;
}
pid_t pid = IPCThreadState::self()->getCallingPid();
const uid_t uid = IPCThreadState::self()->getCallingUid();
{
Mutex::Autolock _cl(mClientLock);
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
pid,
uid);
ALOGV("registerClient() client %p, pid %d, uid %u",
notificationClient.get(), pid, uid);
mNotificationClients.add(pid, notificationClient);
sp<IBinder> binder = IInterface::asBinder(client);
binder->linkToDeath(notificationClient);
}
}
// mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
// ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
// the config change is always sent from playback or record threads to avoid deadlock
// with AudioSystem::gLock
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_REGISTERED, pid);
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_REGISTERED, pid);
}
}
void AudioFlinger::removeNotificationClient(pid_t pid)
{
std::vector< sp<AudioFlinger::EffectModule> > removedEffects;
{
Mutex::Autolock _l(mLock);
{
Mutex::Autolock _cl(mClientLock);
mNotificationClients.removeItem(pid);
}
ALOGV("%d died, releasing its sessions", pid);
size_t num = mAudioSessionRefs.size();
bool removed = false;
for (size_t i = 0; i < num; ) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
ALOGV(" pid %d @ %zu", ref->mPid, i);
if (ref->mPid == pid) {
ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
mAudioSessionRefs.removeAt(i);
delete ref;
removed = true;
num--;
} else {
i++;
}
}
if (removed) {
removedEffects = purgeStaleEffects_l();
}
}
for (auto& effect : removedEffects) {
effect->updatePolicyState();
}
}
void AudioFlinger::ioConfigChanged(audio_io_config_event event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid) {
media::AudioIoDescriptor descAidl = VALUE_OR_FATAL(
legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(ioDesc));
media::AudioIoConfigEvent eventAidl = VALUE_OR_FATAL(
legacy2aidl_audio_io_config_event_AudioIoConfigEvent(event));
Mutex::Autolock _l(mClientLock);
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(eventAidl,
descAidl);
}
}
}
// removeClient_l() must be called with AudioFlinger::mClientLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
ALOGV("removeClient_l() pid %d, calling pid %d", pid,
IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// getEffectThread_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::ThreadBase> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
int effectId)
{
sp<ThreadBase> thread;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, effectId) != 0) {
ALOG_ASSERT(thread == 0);
thread = mPlaybackThreads.valueAt(i);
}
}
if (thread != nullptr) {
return thread;
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
if (mRecordThreads.valueAt(i)->getEffect(sessionId, effectId) != 0) {
ALOG_ASSERT(thread == 0);
thread = mRecordThreads.valueAt(i);
}
}
if (thread != nullptr) {
return thread;
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
if (mMmapThreads.valueAt(i)->getEffect(sessionId, effectId) != 0) {
ALOG_ASSERT(thread == 0);
thread = mMmapThreads.valueAt(i);
}
}
return thread;
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mPid(pid)
{
mMemoryDealer = new MemoryDealer(
audioFlinger->getClientSharedHeapSize(),
(std::string("AudioFlinger::Client(") + std::to_string(pid) + ")").c_str());
}
// Client destructor must be called with AudioFlinger::mClientLock held
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient_l(mPid);
}
sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<media::IAudioFlingerClient>& client,
pid_t pid,
uid_t uid)
: mAudioFlinger(audioFlinger), mPid(pid), mUid(uid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
{
sp<NotificationClient> keep(this);
mAudioFlinger->removeNotificationClient(mPid);
}
// ----------------------------------------------------------------------------
AudioFlinger::MediaLogNotifier::MediaLogNotifier()
: mPendingRequests(false) {}
void AudioFlinger::MediaLogNotifier::requestMerge() {
AutoMutex _l(mMutex);
mPendingRequests = true;
mCond.signal();
}
bool AudioFlinger::MediaLogNotifier::threadLoop() {
// Should already have been checked, but just in case
if (sMediaLogService == 0) {
return false;
}
// Wait until there are pending requests
{
AutoMutex _l(mMutex);
mPendingRequests = false; // to ignore past requests
while (!mPendingRequests) {
mCond.wait(mMutex);
// TODO may also need an exitPending check
}
mPendingRequests = false;
}
// Execute the actual MediaLogService binder call and ignore extra requests for a while
sMediaLogService->requestMergeWakeup();
usleep(kPostTriggerSleepPeriod);
return true;
}
void AudioFlinger::requestLogMerge() {
mMediaLogNotifier->requestMerge();
}
// ----------------------------------------------------------------------------
status_t AudioFlinger::createRecord(const media::CreateRecordRequest& _input,
media::CreateRecordResponse& _output)
{
CreateRecordInput input = VALUE_OR_RETURN_STATUS(CreateRecordInput::fromAidl(_input));
CreateRecordOutput output;
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
status_t lStatus;
audio_session_t sessionId = input.sessionId;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
output.cblk.clear();
output.buffers.clear();
output.inputId = AUDIO_IO_HANDLE_NONE;
// TODO b/182392553: refactor or clean up
AttributionSourceState adjAttributionSource = input.clientInfo.attributionSource;
bool updatePid = (adjAttributionSource.pid == -1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
const uid_t currentUid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(
adjAttributionSource.uid));
if (!isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
ALOGW_IF(currentUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, currentUid);
adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
updatePid = true;
}
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
const pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(
adjAttributionSource.pid));
if (updatePid) {
ALOGW_IF(currentPid != (pid_t)-1 && currentPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, currentPid);
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
}
adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
adjAttributionSource);
// we don't yet support anything other than linear PCM
if (!audio_is_valid_format(input.config.format) || !audio_is_linear_pcm(input.config.format)) {
ALOGE("createRecord() invalid format %#x", input.config.format);
lStatus = BAD_VALUE;
goto Exit;
}
// further channel mask checks are performed by createRecordTrack_l()
if (!audio_is_input_channel(input.config.channel_mask)) {
ALOGE("createRecord() invalid channel mask %#x", input.config.channel_mask);
lStatus = BAD_VALUE;
goto Exit;
}
if (sessionId == AUDIO_SESSION_ALLOCATE) {
sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
} else if (audio_unique_id_get_use(sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
lStatus = BAD_VALUE;
goto Exit;
}
output.sessionId = sessionId;
output.selectedDeviceId = input.selectedDeviceId;
output.flags = input.flags;
client = registerPid(VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(adjAttributionSource.pid)));
// Not a conventional loop, but a retry loop for at most two iterations total.
// Try first maybe with FAST flag then try again without FAST flag if that fails.
// Exits loop via break on no error of got exit on error
// The sp<> references will be dropped when re-entering scope.
// The lack of indentation is deliberate, to reduce code churn and ease merges.
for (;;) {
// release previously opened input if retrying.
if (output.inputId != AUDIO_IO_HANDLE_NONE) {
recordTrack.clear();
AudioSystem::releaseInput(portId);
output.inputId = AUDIO_IO_HANDLE_NONE;
output.selectedDeviceId = input.selectedDeviceId;
portId = AUDIO_PORT_HANDLE_NONE;
}
lStatus = AudioSystem::getInputForAttr(&input.attr, &output.inputId,
input.riid,
sessionId,
// FIXME compare to AudioTrack
adjAttributionSource,
&input.config,
output.flags, &output.selectedDeviceId, &portId);
if (lStatus != NO_ERROR) {
ALOGE("createRecord() getInputForAttr return error %d", lStatus);
goto Exit;
}
{
Mutex::Autolock _l(mLock);
RecordThread *thread = checkRecordThread_l(output.inputId);
if (thread == NULL) {
ALOGW("createRecord() checkRecordThread_l failed, input handle %d", output.inputId);
lStatus = FAILED_TRANSACTION;
goto Exit;
}
ALOGV("createRecord() lSessionId: %d input %d", sessionId, output.inputId);
output.sampleRate = input.config.sample_rate;
output.frameCount = input.frameCount;
output.notificationFrameCount = input.notificationFrameCount;
recordTrack = thread->createRecordTrack_l(client, input.attr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, sessionId,
&output.notificationFrameCount,
callingPid, adjAttributionSource, &output.flags,
input.clientInfo.clientTid,
&lStatus, portId, input.maxSharedAudioHistoryMs);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
// lStatus == BAD_TYPE means FAST flag was rejected: request a new input from
// audio policy manager without FAST constraint
if (lStatus == BAD_TYPE) {
continue;
}
if (lStatus != NO_ERROR) {
goto Exit;
}
// Check if one effect chain was awaiting for an AudioRecord to be created on this
// session and move it to this thread.
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
Mutex::Autolock _l(thread->mLock);
thread->addEffectChain_l(chain);
}
break;
}
// End of retry loop.
// The lack of indentation is deliberate, to reduce code churn and ease merges.
}
output.cblk = recordTrack->getCblk();
output.buffers = recordTrack->getBuffers();
output.portId = portId;
output.audioRecord = new RecordHandle(recordTrack);
_output = VALUE_OR_FATAL(output.toAidl());
Exit:
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the RecordTrack so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
// Don't hold mClientLock when releasing the reference on the track as the
// destructor will acquire it.
{
Mutex::Autolock _cl(mClientLock);
client.clear();
}
recordTrack.clear();
if (output.inputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseInput(portId);
}
}
return lStatus;
}
// ----------------------------------------------------------------------------
audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
{
if (name == NULL) {
return AUDIO_MODULE_HANDLE_NONE;
}
if (!settingsAllowed()) {
return AUDIO_MODULE_HANDLE_NONE;
}
Mutex::Autolock _l(mLock);
AutoMutex lock(mHardwareLock);
return loadHwModule_l(name);
}
// loadHwModule_l() must be called with AudioFlinger::mLock and AudioFlinger::mHardwareLock held
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
ALOGW("loadHwModule() module %s already loaded", name);
return mAudioHwDevs.keyAt(i);
}
}
sp<DeviceHalInterface> dev;
int rc = mDevicesFactoryHal->openDevice(name, &dev);
if (rc) {
ALOGE("loadHwModule() error %d loading module %s", rc, name);
return AUDIO_MODULE_HANDLE_NONE;
}
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->initCheck();
mHardwareStatus = AUDIO_HW_IDLE;
if (rc) {
ALOGE("loadHwModule() init check error %d for module %s", rc, name);
return AUDIO_MODULE_HANDLE_NONE;
}
// Check and cache this HAL's level of support for master mute and master
// volume. If this is the first HAL opened, and it supports the get
// methods, use the initial values provided by the HAL as the current
// master mute and volume settings.
AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
if (0 == mAudioHwDevs.size()) {
mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
float mv;
if (OK == dev->getMasterVolume(&mv)) {
mMasterVolume = mv;
}
mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
bool mm;
if (OK == dev->getMasterMute(&mm)) {
mMasterMute = mm;
}
}
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (OK == dev->setMasterVolume(mMasterVolume)) {
flags = static_cast<AudioHwDevice::Flags>(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
}
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if (OK == dev->setMasterMute(mMasterMute)) {
flags = static_cast<AudioHwDevice::Flags>(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
}
mHardwareStatus = AUDIO_HW_IDLE;
if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_MSD) == 0) {
// An MSD module is inserted before hardware modules in order to mix encoded streams.
flags = static_cast<AudioHwDevice::Flags>(flags | AudioHwDevice::AHWD_IS_INSERT);
}
audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
AudioHwDevice *audioDevice = new AudioHwDevice(handle, name, dev, flags);
if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
mPrimaryHardwareDev = audioDevice;
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
mAudioHwDevs.add(handle, audioDevice);
ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle);
return handle;
}
// ----------------------------------------------------------------------------
uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = fastPlaybackThread_l();
return thread != NULL ? thread->sampleRate() : 0;
}
size_t AudioFlinger::getPrimaryOutputFrameCount()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = fastPlaybackThread_l();
return thread != NULL ? thread->frameCountHAL() : 0;
}
// ----------------------------------------------------------------------------
status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice, int64_t totalMemory)
{
uid_t uid = IPCThreadState::self()->getCallingUid();
if (!isAudioServerOrSystemServerUid(uid)) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
if (mIsDeviceTypeKnown) {
return INVALID_OPERATION;
}
mIsLowRamDevice = isLowRamDevice;
mTotalMemory = totalMemory;
// mIsLowRamDevice and mTotalMemory are obtained through ActivityManager;
// see ActivityManager.isLowRamDevice() and ActivityManager.getMemoryInfo().
// mIsLowRamDevice generally represent devices with less than 1GB of memory,
// though actual setting is determined through device configuration.
constexpr int64_t GB = 1024 * 1024 * 1024;
mClientSharedHeapSize =
isLowRamDevice ? kMinimumClientSharedHeapSizeBytes
: mTotalMemory < 2 * GB ? 4 * kMinimumClientSharedHeapSizeBytes
: mTotalMemory < 3 * GB ? 8 * kMinimumClientSharedHeapSizeBytes
: mTotalMemory < 4 * GB ? 16 * kMinimumClientSharedHeapSizeBytes
: 32 * kMinimumClientSharedHeapSizeBytes;
mIsDeviceTypeKnown = true;
// TODO: Cache the client shared heap size in a persistent property.
// It's possible that a native process or Java service or app accesses audioserver
// after it is registered by system server, but before AudioService updates
// the memory info. This would occur immediately after boot or an audioserver
// crash and restore. Before update from AudioService, the client would get the
// minimum heap size.
ALOGD("isLowRamDevice:%s totalMemory:%lld mClientSharedHeapSize:%zu",
(isLowRamDevice ? "true" : "false"),
(long long)mTotalMemory,
mClientSharedHeapSize.load());
return NO_ERROR;
}
size_t AudioFlinger::getClientSharedHeapSize() const
{
size_t heapSizeInBytes = property_get_int32("ro.af.client_heap_size_kbyte", 0) * 1024;
if (heapSizeInBytes != 0) { // read-only property overrides all.
return heapSizeInBytes;
}
return mClientSharedHeapSize;
}
status_t AudioFlinger::setAudioPortConfig(const struct audio_port_config *config)
{
ALOGV(__func__);
status_t status = AudioValidator::validateAudioPortConfig(*config);
if (status != NO_ERROR) {
return status;
}
audio_module_handle_t module;
if (config->type == AUDIO_PORT_TYPE_DEVICE) {
module = config->ext.device.hw_module;
} else {
module = config->ext.mix.hw_module;
}
Mutex::Autolock _l(mLock);
AutoMutex lock(mHardwareLock);
ssize_t index = mAudioHwDevs.indexOfKey(module);
if (index < 0) {
ALOGW("%s() bad hw module %d", __func__, module);
return BAD_VALUE;
}
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(index);
return audioHwDevice->hwDevice()->setAudioPortConfig(config);
}
audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
{
Mutex::Autolock _l(mLock);
ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
if (index >= 0) {
ALOGV("getAudioHwSyncForSession found ID %d for session %d",
mHwAvSyncIds.valueAt(index), sessionId);
return mHwAvSyncIds.valueAt(index);
}
sp<DeviceHalInterface> dev;
{
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return AUDIO_HW_SYNC_INVALID;
}
dev = mPrimaryHardwareDev->hwDevice();
}
if (dev == nullptr) {
return AUDIO_HW_SYNC_INVALID;
}
String8 reply;
AudioParameter param;
if (dev->getParameters(String8(AudioParameter::keyHwAvSync), &reply) == OK) {
param = AudioParameter(reply);
}
int value;
if (param.getInt(String8(AudioParameter::keyHwAvSync), value) != NO_ERROR) {
ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId);
return AUDIO_HW_SYNC_INVALID;
}
// allow only one session for a given HW A/V sync ID.
for (size_t i = 0; i < mHwAvSyncIds.size(); i++) {
if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) {
ALOGV("getAudioHwSyncForSession removing ID %d for session %d",
value, mHwAvSyncIds.keyAt(i));
mHwAvSyncIds.removeItemsAt(i);
break;
}
}
mHwAvSyncIds.add(sessionId, value);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
uint32_t sessions = thread->hasAudioSession(sessionId);
if (sessions & ThreadBase::TRACK_SESSION) {
AudioParameter param = AudioParameter();
param.addInt(String8(AudioParameter::keyStreamHwAvSync), value);
String8 keyValuePairs = param.toString();
thread->setParameters(keyValuePairs);
forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
[](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
break;
}
}
ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
return (audio_hw_sync_t)value;
}
status_t AudioFlinger::systemReady()
{
Mutex::Autolock _l(mLock);
ALOGI("%s", __FUNCTION__);
if (mSystemReady) {
ALOGW("%s called twice", __FUNCTION__);
return NO_ERROR;
}
mSystemReady = true;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
thread->systemReady();
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
thread->systemReady();
}
return NO_ERROR;
}
status_t AudioFlinger::getMicrophones(std::vector<media::MicrophoneInfo> *microphones)
{
AutoMutex lock(mHardwareLock);
status_t status = INVALID_OPERATION;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
std::vector<media::MicrophoneInfo> mics;
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_GET_MICROPHONES;
status_t devStatus = dev->hwDevice()->getMicrophones(&mics);
mHardwareStatus = AUDIO_HW_IDLE;
if (devStatus == NO_ERROR) {
microphones->insert(microphones->begin(), mics.begin(), mics.end());
// report success if at least one HW module supports the function.
status = NO_ERROR;
}
}
return status;
}
// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
{
ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
if (index >= 0) {
audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index);
ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
AudioParameter param = AudioParameter();
param.addInt(String8(AudioParameter::keyStreamHwAvSync), syncId);
String8 keyValuePairs = param.toString();
thread->setParameters(keyValuePairs);
forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
[](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
}
}
// ----------------------------------------------------------------------------
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags)
{
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, deviceType);
if (outHwDev == NULL) {
return 0;
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
} else {
// Audio Policy does not currently request a specific output handle.
// If this is ever needed, see openInput_l() for example code.
ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
return 0;
}
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
// FOR TESTING ONLY:
// This if statement allows overriding the audio policy settings
// and forcing a specific format or channel mask to the HAL/Sink device for testing.
if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
//config->format = AUDIO_FORMAT_PCM_FLOAT;
//config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
//config->format = AUDIO_FORMAT_PCM_32_BIT;
//config->format = AUDIO_FORMAT_PCM_8_24_BIT;
// ALOGV("openOutput_l() upgrading format to %#08x", config->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
//config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
//config->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
AudioStreamOut *outputStream = NULL;
status_t status = outHwDev->openOutputStream(
&outputStream,
*output,
deviceType,
flags,
config,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
if (status == NO_ERROR) {
if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
sp<MmapPlaybackThread> thread =
new MmapPlaybackThread(this, *output, outHwDev, outputStream, mSystemReady);
mMmapThreads.add(*output, thread);
ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p",
*output, thread.get());
return thread;
} else {
sp<PlaybackThread> thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
} else {
thread = new MixerThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p",
*output, thread.get());
}
mPlaybackThreads.add(*output, thread);
struct audio_patch patch;
mPatchPanel.notifyStreamOpened(outHwDev, *output, &patch);
if (thread->isMsdDevice()) {
thread->setDownStreamPatch(&patch);
}
return thread;
}
}
return 0;
}
status_t AudioFlinger::openOutput(const media::OpenOutputRequest& request,
media::OpenOutputResponse* response)
{
audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_module_handle_t(request.module));
audio_config_t config = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfig_audio_config_t(request.config));
sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
aidl2legacy_DeviceDescriptorBase(request.device));
audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_output_flags_t_mask(request.flags));
audio_io_handle_t output;
uint32_t latencyMs;
ALOGI("openOutput() this %p, module %d Device %s, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
this, module,
device->toString().c_str(),
config.sample_rate,
config.format,
config.channel_mask,
flags);
audio_devices_t deviceType = device->type();
const String8 address = String8(device->address().c_str());
if (deviceType == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
latencyMs = playbackThread->latency();
// notify client processes of the new output creation
playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
// the first primary output opened designates the primary hw device if no HW module
// named "primary" was already loaded.
AutoMutex lock(mHardwareLock);
if ((mPrimaryHardwareDev == nullptr) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d as the primary audio interface", module);
mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev;
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
MmapThread *mmapThread = (MmapThread *)thread.get();
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
response->flags = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
return NO_ERROR;
}
return NO_INIT;
}
audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
MixerThread *thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
output2);
return AUDIO_IO_HANDLE_NONE;
}
audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
return id;
}
status_t AudioFlinger::closeOutput(audio_io_handle_t output)
{
return closeOutput_nonvirtual(output);
}
status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
sp<PlaybackThread> playbackThread;
sp<MmapPlaybackThread> mmapThread;
{
Mutex::Autolock _l(mLock);
playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
ALOGV("closeOutput() %d", output);
dumpToThreadLog_l(playbackThread);
if (playbackThread->type() == ThreadBase::MIXER) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
DuplicatingThread *dupThread =
(DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
dupThread->removeOutputTrack((MixerThread *)playbackThread.get());
}
}
}
mPlaybackThreads.removeItem(output);
// save all effects to the default thread
if (mPlaybackThreads.size()) {
PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
if (dstThread != NULL) {
// audioflinger lock is held so order of thread lock acquisition doesn't matter
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(playbackThread->mLock);
Vector< sp<EffectChain> > effectChains = playbackThread->getEffectChains_l();
for (size_t i = 0; i < effectChains.size(); i ++) {
moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(),
dstThread);
}
}
}
} else {
mmapThread = (MmapPlaybackThread *)checkMmapThread_l(output);
if (mmapThread == 0) {
return BAD_VALUE;
}
dumpToThreadLog_l(mmapThread);
mMmapThreads.removeItem(output);
ALOGD("closing mmapThread %p", mmapThread.get());
}
const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
ioDesc->mIoHandle = output;
ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
mPatchPanel.notifyStreamClosed(output);
}
// The thread entity (active unit of execution) is no longer running here,
// but the ThreadBase container still exists.
if (playbackThread != 0) {
playbackThread->exit();
if (!playbackThread->isDuplicating()) {
closeOutputFinish(playbackThread);
}
} else if (mmapThread != 0) {
ALOGD("mmapThread exit()");
mmapThread->exit();
AudioStreamOut *out = mmapThread->clearOutput();
ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
// from now on thread->mOutput is NULL
delete out;
}
return NO_ERROR;
}
void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread)
{
AudioStreamOut *out = thread->clearOutput();
ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
// from now on thread->mOutput is NULL
delete out;
}
void AudioFlinger::closeThreadInternal_l(const sp<PlaybackThread>& thread)
{
mPlaybackThreads.removeItem(thread->mId);
thread->exit();
closeOutputFinish(thread);
}
status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("suspendOutput() %d", output);
thread->suspend();
return NO_ERROR;
}
status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("restoreOutput() %d", output);
thread->restore();
return NO_ERROR;
}
status_t AudioFlinger::openInput(const media::OpenInputRequest& request,
media::OpenInputResponse* response)
{
Mutex::Autolock _l(mLock);
if (request.device.type == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
audio_io_handle_t input = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_io_handle_t(request.input));
audio_config_t config = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfig_audio_config_t(request.config));
AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioDeviceTypeAddress(request.device));
sp<ThreadBase> thread = openInput_l(
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(request.module)),
&input,
&config,
device.mType,
device.address().c_str(),
VALUE_OR_RETURN_STATUS(aidl2legacy_AudioSourceType_audio_source_t(request.source)),
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_input_flags_t_mask(request.flags)),
AUDIO_DEVICE_NONE,
String8{});
response->input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
response->device = request.device;
if (thread != 0) {
// notify client processes of the new input creation
thread->ioConfigChanged(AUDIO_INPUT_OPENED);
return NO_ERROR;
}
return NO_INIT;
}
sp<AudioFlinger::ThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t devices,
const char* address,
audio_source_t source,
audio_input_flags_t flags,
audio_devices_t outputDevice,
const String8& outputDeviceAddress)
{
AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
if (inHwDev == NULL) {
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
// Audio Policy can request a specific handle for hardware hotword.
// The goal here is not to re-open an already opened input.
// It is to use a pre-assigned I/O handle.
if (*input == AUDIO_IO_HANDLE_NONE) {
*input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
} else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) {
ALOGE("openInput_l() requested input handle %d is invalid", *input);
return 0;
} else if (mRecordThreads.indexOfKey(*input) >= 0) {
// This should not happen in a transient state with current design.
ALOGE("openInput_l() requested input handle %d is already assigned", *input);
return 0;
}
audio_config_t halconfig = *config;
sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice();
sp<StreamInHalInterface> inStream;
status_t status = inHwHal->openInputStream(
*input, devices, &halconfig, flags, address, source,
outputDevice, outputDeviceAddress, &inStream);
ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
", Format %#x, Channels %#x, flags %#x, status %d addr %s",
inStream.get(),
devices,
halconfig.sample_rate,
halconfig.format,
halconfig.channel_mask,
flags,
status, address);
// If the input could not be opened with the requested parameters and we can handle the
// conversion internally, try to open again with the proposed parameters.
if (status == BAD_VALUE &&
audio_is_linear_pcm(config->format) &&
audio_is_linear_pcm(halconfig.format) &&
(halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_LIMIT) &&
(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream.clear();
status = inHwHal->openInputStream(
*input, devices, &halconfig, flags, address, source,
outputDevice, outputDeviceAddress, &inStream);
// FIXME log this new status; HAL should not propose any further changes
}
if (status == NO_ERROR && inStream != 0) {
AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
sp<MmapCaptureThread> thread =
new MmapCaptureThread(this, *input, inHwDev, inputStream, mSystemReady);
mMmapThreads.add(*input, thread);
ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
thread.get());
return thread;
} else {
// Start record thread
// RecordThread requires both input and output device indication to forward to audio
// pre processing modules
sp<RecordThread> thread = new RecordThread(this, inputStream, *input, mSystemReady);
mRecordThreads.add(*input, thread);
ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
return thread;
}
}
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
return closeInput_nonvirtual(input);
}
status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
sp<RecordThread> recordThread;
sp<MmapCaptureThread> mmapThread;
{
Mutex::Autolock _l(mLock);
recordThread = checkRecordThread_l(input);
if (recordThread != 0) {
ALOGV("closeInput() %d", input);
dumpToThreadLog_l(recordThread);
// If we still have effect chains, it means that a client still holds a handle
// on at least one effect. We must either move the chain to an existing thread with the
// same session ID or put it aside in case a new record thread is opened for a
// new capture on the same session
sp<EffectChain> chain;
{
Mutex::Autolock _sl(recordThread->mLock);
Vector< sp<EffectChain> > effectChains = recordThread->getEffectChains_l();
// Note: maximum one chain per record thread
if (effectChains.size() != 0) {
chain = effectChains[0];
}
}
if (chain != 0) {
// first check if a record thread is already opened with a client on same session.
// This should only happen in case of overlap between one thread tear down and the
// creation of its replacement
size_t i;
for (i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> t = mRecordThreads.valueAt(i);
if (t == recordThread) {
continue;
}
if (t->hasAudioSession(chain->sessionId()) != 0) {
Mutex::Autolock _l(t->mLock);
ALOGV("closeInput() found thread %d for effect session %d",
t->id(), chain->sessionId());
t->addEffectChain_l(chain);
break;
}
}
// put the chain aside if we could not find a record thread with the same session id
if (i == mRecordThreads.size()) {
putOrphanEffectChain_l(chain);
}
}
mRecordThreads.removeItem(input);
} else {
mmapThread = (MmapCaptureThread *)checkMmapThread_l(input);
if (mmapThread == 0) {
return BAD_VALUE;
}
dumpToThreadLog_l(mmapThread);
mMmapThreads.removeItem(input);
}
const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
ioDesc->mIoHandle = input;
ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc);
}
// FIXME: calling thread->exit() without mLock held should not be needed anymore now that
// we have a different lock for notification client
if (recordThread != 0) {
closeInputFinish(recordThread);
} else if (mmapThread != 0) {
mmapThread->exit();
AudioStreamIn *in = mmapThread->clearInput();
ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
// from now on thread->mInput is NULL
delete in;
}
return NO_ERROR;
}
void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread)
{
thread->exit();
AudioStreamIn *in = thread->clearInput();
ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
// from now on thread->mInput is NULL
delete in;
}
void AudioFlinger::closeThreadInternal_l(const sp<RecordThread>& thread)
{
mRecordThreads.removeItem(thread->mId);
closeInputFinish(thread);
}
status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
{
Mutex::Autolock _l(mLock);
ALOGV("invalidateStream() stream %d", stream);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
thread->invalidateTracks(stream);
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
mMmapThreads[i]->invalidateTracks(stream);
}
return NO_ERROR;
}
audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use)
{
// This is a binder API, so a malicious client could pass in a bad parameter.
// Check for that before calling the internal API nextUniqueId().
if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) {
ALOGE("newAudioUniqueId invalid use %d", use);
return AUDIO_UNIQUE_ID_ALLOCATE;
}
return nextUniqueId(use);
}
void AudioFlinger::acquireAudioSessionId(
audio_session_t audioSession, pid_t pid, uid_t uid)
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
const uid_t callerUid = IPCThreadState::self()->getCallingUid();
if (pid != (pid_t)-1 && isAudioServerOrMediaServerUid(callerUid)) {
caller = pid; // check must match releaseAudioSessionId()
}
if (uid == (uid_t)-1 || !isAudioServerOrMediaServerUid(callerUid)) {
uid = callerUid;
}
{
Mutex::Autolock _cl(mClientLock);
// Ignore requests received from processes not known as notification client. The request
// is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
// called from a different pid leaving a stale session reference. Also we don't know how
// to clear this reference if the client process dies.
if (mNotificationClients.indexOfKey(caller) < 0) {
ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
return;
}
}
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i < num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
if (ref->mSessionid == audioSession && ref->mPid == caller) {
ref->mCnt++;
ALOGV(" incremented refcount to %d", ref->mCnt);
return;
}
}
mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller, uid));
ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
{
std::vector< sp<EffectModule> > removedEffects;
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
const uid_t callerUid = IPCThreadState::self()->getCallingUid();
if (pid != (pid_t)-1 && isAudioServerOrMediaServerUid(callerUid)) {
caller = pid; // check must match acquireAudioSessionId()
}
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i < num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
if (ref->mSessionid == audioSession && ref->mPid == caller) {
ref->mCnt--;
ALOGV(" decremented refcount to %d", ref->mCnt);
if (ref->mCnt == 0) {
mAudioSessionRefs.removeAt(i);
delete ref;
std::vector< sp<EffectModule> > effects = purgeStaleEffects_l();
removedEffects.insert(removedEffects.end(), effects.begin(), effects.end());
}
goto Exit;
}
}
// If the caller is audioserver it is likely that the session being released was acquired
// on behalf of a process not in notification clients and we ignore the warning.
ALOGW_IF(!isAudioServerUid(callerUid),
"session id %d not found for pid %d", audioSession, caller);
}
Exit:
for (auto& effect : removedEffects) {
effect->updatePolicyState();
}
}
bool AudioFlinger::isSessionAcquired_l(audio_session_t audioSession)
{
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i < num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
if (ref->mSessionid == audioSession) {
return true;
}
}
return false;
}
std::vector<sp<AudioFlinger::EffectModule>> AudioFlinger::purgeStaleEffects_l() {
ALOGV("purging stale effects");
Vector< sp<EffectChain> > chains;
std::vector< sp<EffectModule> > removedEffects;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
Mutex::Autolock _l(t->mLock);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
if (!audio_is_global_session(ec->sessionId())) {
chains.push(ec);
}
}
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> t = mRecordThreads.valueAt(i);
Mutex::Autolock _l(t->mLock);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
chains.push(ec);
}
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
sp<MmapThread> t = mMmapThreads.valueAt(i);
Mutex::Autolock _l(t->mLock);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
chains.push(ec);
}
}
for (size_t i = 0; i < chains.size(); i++) {
sp<EffectChain> ec = chains[i];
int sessionid = ec->sessionId();
sp<ThreadBase> t = ec->thread().promote();
if (t == 0) {
continue;
}
size_t numsessionrefs = mAudioSessionRefs.size();
bool found = false;
for (size_t k = 0; k < numsessionrefs; k++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
if (ref->mSessionid == sessionid) {
ALOGV(" session %d still exists for %d with %d refs",
sessionid, ref->mPid, ref->mCnt);
found = true;
break;
}
}
if (!found) {
Mutex::Autolock _l(t->mLock);
// remove all effects from the chain
while (ec->mEffects.size()) {
sp<EffectModule> effect = ec->mEffects[0];
effect->unPin();
t->removeEffect_l(effect, /*release*/ true);
if (effect->purgeHandles()) {
effect->checkSuspendOnEffectEnabled(false, true /*threadLocked*/);
}
removedEffects.push_back(effect);
}
}
}
return removedEffects;
}
// dumpToThreadLog_l() must be called with AudioFlinger::mLock held
void AudioFlinger::dumpToThreadLog_l(const sp<ThreadBase> &thread)
{
constexpr int THREAD_DUMP_TIMEOUT_MS = 2;
audio_utils::FdToString fdToString("- ", THREAD_DUMP_TIMEOUT_MS);
const int fd = fdToString.fd();
if (fd >= 0) {
thread->dump(fd, {} /* args */);
mThreadLog.logs(-1 /* time */, fdToString.getStringAndClose());
}
}
// checkThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
{
ThreadBase *thread = checkMmapThread_l(ioHandle);
if (thread == 0) {
switch (audio_unique_id_get_use(ioHandle)) {
case AUDIO_UNIQUE_ID_USE_OUTPUT:
thread = checkPlaybackThread_l(ioHandle);
break;
case AUDIO_UNIQUE_ID_USE_INPUT:
thread = checkRecordThread_l(ioHandle);
break;
default:
break;
}
}
return thread;
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
return mPlaybackThreads.valueFor(output).get();
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
return mRecordThreads.valueFor(input).get();
}
// checkMmapThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MmapThread *AudioFlinger::checkMmapThread_l(audio_io_handle_t io) const
{
return mMmapThreads.valueFor(io).get();
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::VolumeInterface *AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const
{
VolumeInterface *volumeInterface = mPlaybackThreads.valueFor(output).get();
if (volumeInterface == nullptr) {
MmapThread *mmapThread = mMmapThreads.valueFor(output).get();
if (mmapThread != nullptr) {
if (mmapThread->isOutput()) {
MmapPlaybackThread *mmapPlaybackThread =
static_cast<MmapPlaybackThread *>(mmapThread);
volumeInterface = mmapPlaybackThread;
}
}
}
return volumeInterface;
}
Vector <AudioFlinger::VolumeInterface *> AudioFlinger::getAllVolumeInterfaces_l() const
{
Vector <VolumeInterface *> volumeInterfaces;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
volumeInterfaces.add(mPlaybackThreads.valueAt(i).get());
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
if (mMmapThreads.valueAt(i)->isOutput()) {
MmapPlaybackThread *mmapPlaybackThread =
static_cast<MmapPlaybackThread *>(mMmapThreads.valueAt(i).get());
volumeInterfaces.add(mmapPlaybackThread);
}
}
return volumeInterfaces;
}
audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use)
{
// This is the internal API, so it is OK to assert on bad parameter.
LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX);
const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1;
for (int retry = 0; retry < maxRetries; retry++) {
// The cast allows wraparound from max positive to min negative instead of abort
uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use],
(uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel);
ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED);
// allow wrap by skipping 0 and -1 for session ids
if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) {
ALOGW_IF(retry != 0, "unique ID overflow for use %d", use);
return (audio_unique_id_t) (base | use);
}
}
// We have no way of recovering from wraparound
LOG_ALWAYS_FATAL("unique ID overflow for use %d", use);
// TODO Use a floor after wraparound. This may need a mutex.
}
AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
{
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return nullptr;
}
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if(thread->isDuplicating()) {
continue;
}
AudioStreamOut *output = thread->getOutput();
if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
return thread;
}
}
return nullptr;
}
DeviceTypeSet AudioFlinger::primaryOutputDevice_l() const
{
PlaybackThread *thread = primaryPlaybackThread_l();
if (thread == NULL) {
return DeviceTypeSet();
}
return thread->outDeviceTypes();
}
AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const
{
size_t minFrameCount = 0;
PlaybackThread *minThread = NULL;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if (!thread->isDuplicating()) {
size_t frameCount = thread->frameCountHAL();
if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount ||
(frameCount == minFrameCount && thread->hasFastMixer() &&
/*minThread != NULL &&*/ !minThread->hasFastMixer()))) {
minFrameCount = frameCount;
minThread = thread;
}
}
}
return minThread;
}
AudioFlinger::ThreadBase *AudioFlinger::hapticPlaybackThread_l() const {
for (size_t i = 0; i < mPlaybackThreads.size(); ++i) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if (thread->hapticChannelMask() != AUDIO_CHANNEL_NONE) {
return thread;
}
}
return nullptr;
}
void AudioFlinger::updateSecondaryOutputsForTrack_l(
PlaybackThread::Track* track,
PlaybackThread* thread,
const std::vector<audio_io_handle_t> &secondaryOutputs) const {
TeePatches teePatches;
for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
if (secondaryThread == nullptr) {
ALOGE("no playback thread found for secondary output %d", thread->id());
continue;
}
size_t sourceFrameCount = thread->frameCount() * track->sampleRate()
/ thread->sampleRate();
size_t sinkFrameCount = secondaryThread->frameCount() * track->sampleRate()
/ secondaryThread->sampleRate();
// If the secondary output has just been opened, the first secondaryThread write
// will not block as it will fill the empty startup buffer of the HAL,
// so a second sink buffer needs to be ready for the immediate next blocking write.
// Additionally, have a margin of one main thread buffer as the scheduling jitter
// can reorder the writes (eg if thread A&B have the same write intervale,
// the scheduler could schedule AB...BA)
size_t frameCountToBeReady = 2 * sinkFrameCount + sourceFrameCount;
// Total secondary output buffer must be at least as the read frames plus
// the margin of a few buffers on both sides in case the
// threads scheduling has some jitter.
// That value should not impact latency as the secondary track is started before
// its buffer is full, see frameCountToBeReady.
size_t frameCount = frameCountToBeReady + 2 * (sourceFrameCount + sinkFrameCount);
// The frameCount should also not be smaller than the secondary thread min frame
// count
size_t minFrameCount = AudioSystem::calculateMinFrameCount(
[&] { Mutex::Autolock _l(secondaryThread->mLock);
return secondaryThread->latency_l(); }(),
secondaryThread->mNormalFrameCount,
secondaryThread->mSampleRate,
track->sampleRate(),
track->getSpeed());
frameCount = std::max(frameCount, minFrameCount);
using namespace std::chrono_literals;
auto inChannelMask = audio_channel_mask_out_to_in(track->channelMask());
sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
track->sampleRate(),
inChannelMask,
track->format(),
frameCount,
nullptr /* buffer */,
(size_t)0 /* bufferSize */,
AUDIO_INPUT_FLAG_DIRECT,
0ns /* timeout */);
status_t status = patchRecord->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchRecord init failed: %d", status);
continue;
}
// TODO: We could check compatibility of the secondaryThread with the PatchTrack
// for fast usage: thread has fast mixer, sample rate matches, etc.;
// for now, we exclude fast tracks by removing the Fast flag.
const audio_output_flags_t outputFlags =
(audio_output_flags_t)(track->getOutputFlags() & ~AUDIO_OUTPUT_FLAG_FAST);
sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
track->streamType(),
track->sampleRate(),
track->channelMask(),
track->format(),
frameCount,
patchRecord->buffer(),
patchRecord->bufferSize(),
outputFlags,
0ns /* timeout */,
frameCountToBeReady);
status = patchTrack->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchTrack init failed: %d", status);
continue;
}
teePatches.push_back({patchRecord, patchTrack});
secondaryThread->addPatchTrack(patchTrack);
// In case the downstream patchTrack on the secondaryThread temporarily outlives
// our created track, ensure the corresponding patchRecord is still alive.
patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
}
track->setTeePatches(std::move(teePatches));
}
sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
audio_session_t triggerSession,
audio_session_t listenerSession,
sync_event_callback_t callBack,
const wp<RefBase>& cookie)
{
Mutex::Autolock _l(mLock);
sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
status_t playStatus = NAME_NOT_FOUND;
status_t recStatus = NAME_NOT_FOUND;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
if (playStatus == NO_ERROR) {
return event;
}
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
if (recStatus == NO_ERROR) {
return event;
}
}
if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
mPendingSyncEvents.add(event);
} else {
ALOGV("createSyncEvent() invalid event %d", event->type());
event.clear();
}
return event;
}
// ----------------------------------------------------------------------------
// Effect management
// ----------------------------------------------------------------------------
sp<EffectsFactoryHalInterface> AudioFlinger::getEffectsFactory() {
return mEffectsFactoryHal;
}
status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
{
Mutex::Autolock _l(mLock);
if (mEffectsFactoryHal.get()) {
return mEffectsFactoryHal->queryNumberEffects(numEffects);
} else {
return -ENODEV;
}
}
status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
if (mEffectsFactoryHal.get()) {
return mEffectsFactoryHal->getDescriptor(index, descriptor);
} else {
return -ENODEV;
}
}
status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
const effect_uuid_t *pTypeUuid,
uint32_t preferredTypeFlag,
effect_descriptor_t *descriptor) const
{
if (pUuid == NULL || pTypeUuid == NULL || descriptor == NULL) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
if (!mEffectsFactoryHal.get()) {
return -ENODEV;
}
status_t status = NO_ERROR;
if (!EffectsFactoryHalInterface::isNullUuid(pUuid)) {
// If uuid is specified, request effect descriptor from that.
status = mEffectsFactoryHal->getDescriptor(pUuid, descriptor);
} else if (!EffectsFactoryHalInterface::isNullUuid(pTypeUuid)) {
// If uuid is not specified, look for an available implementation
// of the required type instead.
// Use a temporary descriptor to avoid modifying |descriptor| in the failure case.
effect_descriptor_t desc;
desc.flags = 0; // prevent compiler warning
uint32_t numEffects = 0;
status = mEffectsFactoryHal->queryNumberEffects(&numEffects);
if (status < 0) {
ALOGW("getEffectDescriptor() error %d from FactoryHal queryNumberEffects", status);
return status;
}
bool found = false;
for (uint32_t i = 0; i < numEffects; i++) {
status = mEffectsFactoryHal->getDescriptor(i, &desc);
if (status < 0) {
ALOGW("getEffectDescriptor() error %d from FactoryHal getDescriptor", status);
continue;
}
if (memcmp(&desc.type, pTypeUuid, sizeof(effect_uuid_t)) == 0) {
// If matching type found save effect descriptor.
found = true;
*descriptor = desc;
// If there's no preferred flag or this descriptor matches the preferred
// flag, success! If this descriptor doesn't match the preferred
// flag, continue enumeration in case a better matching version of this
// effect type is available. Note that this means if no effect with a
// correct flag is found, the descriptor returned will correspond to the
// last effect that at least had a matching type uuid (if any).
if (preferredTypeFlag == EFFECT_FLAG_TYPE_MASK ||
(desc.flags & EFFECT_FLAG_TYPE_MASK) == preferredTypeFlag) {
break;
}
}
}
if (!found) {
status = NAME_NOT_FOUND;
ALOGW("getEffectDescriptor(): Effect not found by type.");
}
} else {
status = BAD_VALUE;
ALOGE("getEffectDescriptor(): Either uuid or type uuid must be non-null UUIDs.");
}
return status;
}
status_t AudioFlinger::createEffect(const media::CreateEffectRequest& request,
media::CreateEffectResponse* response) {
const sp<IEffectClient>& effectClient = request.client;
const int32_t priority = request.priority;
const AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioDeviceTypeAddress(request.device));
AttributionSourceState adjAttributionSource = request.attributionSource;
const audio_session_t sessionId = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_session_t(request.sessionId));
audio_io_handle_t io = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_io_handle_t(request.output));
const effect_descriptor_t descIn = VALUE_OR_RETURN_STATUS(
aidl2legacy_EffectDescriptor_effect_descriptor_t(request.desc));
const bool probe = request.probe;
sp<EffectHandle> handle;
effect_descriptor_t descOut;
int enabledOut = 0;
int idOut = -1;
status_t lStatus = NO_ERROR;
// TODO b/182392553: refactor or make clearer
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(adjAttributionSource.pid));
if (currentPid == -1 || !isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(currentPid != -1 && currentPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, currentPid);
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
currentPid = callingPid;
}
adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(adjAttributionSource);
ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p",
adjAttributionSource.pid, effectClient.get(), priority, sessionId, io,
mEffectsFactoryHal.get());
if (mEffectsFactoryHal == 0) {
ALOGE("%s: no effects factory hal", __func__);
lStatus = NO_INIT;
goto Exit;
}
// check audio settings permission for global effects
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
if (!settingsAllowed()) {
ALOGE("%s: no permission for AUDIO_SESSION_OUTPUT_MIX", __func__);
lStatus = PERMISSION_DENIED;
goto Exit;
}
} else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
if (!isAudioServerUid(callingUid)) {
ALOGE("%s: only APM can create using AUDIO_SESSION_OUTPUT_STAGE", __func__);
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (io == AUDIO_IO_HANDLE_NONE) {
ALOGE("%s: APM must specify output when using AUDIO_SESSION_OUTPUT_STAGE", __func__);
lStatus = BAD_VALUE;
goto Exit;
}
} else if (sessionId == AUDIO_SESSION_DEVICE) {
if (!modifyDefaultAudioEffectsAllowed(adjAttributionSource)) {
ALOGE("%s: device effect permission denied for uid %d", __func__, callingUid);
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (io != AUDIO_IO_HANDLE_NONE) {
ALOGE("%s: io handle should not be specified for device effect", __func__);
lStatus = BAD_VALUE;
goto Exit;
}
} else {
// general sessionId.
if (audio_unique_id_get_use(sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
ALOGE("%s: invalid sessionId %d", __func__, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
// TODO: should we check if the callingUid (limited to pid) is in mAudioSessionRefs
// to prevent creating an effect when one doesn't actually have track with that session?
}
{
// Get the full effect descriptor from the uuid/type.
// If the session is the output mix, prefer an auxiliary effect,
// otherwise no preference.
uint32_t preferredType = (sessionId == AUDIO_SESSION_OUTPUT_MIX ?
EFFECT_FLAG_TYPE_AUXILIARY : EFFECT_FLAG_TYPE_MASK);
lStatus = getEffectDescriptor(&descIn.uuid, &descIn.type, preferredType, &descOut);
if (lStatus < 0) {
ALOGW("createEffect() error %d from getEffectDescriptor", lStatus);
goto Exit;
}
// Do not allow auxiliary effects on a session different from 0 (output mix)
if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
(descOut.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// check recording permission for visualizer
if ((memcmp(&descOut.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
// TODO: Do we need to start/stop op - i.e. is there recording being performed?
!recordingAllowed(adjAttributionSource)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
const bool hapticPlaybackRequired = EffectModule::isHapticGenerator(&descOut.type);
if (hapticPlaybackRequired
&& (sessionId == AUDIO_SESSION_DEVICE
|| sessionId == AUDIO_SESSION_OUTPUT_MIX
|| sessionId == AUDIO_SESSION_OUTPUT_STAGE)) {
// haptic-generating effect is only valid when the session id is a general session id
lStatus = INVALID_OPERATION;
goto Exit;
}
if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(io) below will detect it
// and we will exit safely
io = AudioSystem::getOutputForEffect(&descOut);
ALOGV("createEffect got output %d", io);
}
Mutex::Autolock _l(mLock);
if (sessionId == AUDIO_SESSION_DEVICE) {
sp<Client> client = registerPid(currentPid);
ALOGV("%s device type %#x address %s", __func__, device.mType, device.getAddress());
handle = mDeviceEffectManager.createEffect_l(
&descOut, device, client, effectClient, mPatchPanel.patches_l(),
&enabledOut, &lStatus, probe);
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
// remove local strong reference to Client with mClientLock held
Mutex::Autolock _cl(mClientLock);
client.clear();
} else {
// handle must be valid here, but check again to be safe.
if (handle.get() != nullptr) idOut = handle->id();
}
goto Register;
}
// If output is not specified try to find a matching audio session ID in one of the
// output threads.
// If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
// because of code checking output when entering the function.
// Note: io is never AUDIO_IO_HANDLE_NONE when creating an effect on an input by APM.
// An AudioEffect created from the Java API will have io as AUDIO_IO_HANDLE_NONE.
if (io == AUDIO_IO_HANDLE_NONE) {
// look for the thread where the specified audio session is present
io = findIoHandleBySessionId_l(sessionId, mPlaybackThreads);
if (io == AUDIO_IO_HANDLE_NONE) {
io = findIoHandleBySessionId_l(sessionId, mRecordThreads);
}
if (io == AUDIO_IO_HANDLE_NONE) {
io = findIoHandleBySessionId_l(sessionId, mMmapThreads);
}
// If you wish to create a Record preprocessing AudioEffect in Java,
// you MUST create an AudioRecord first and keep it alive so it is picked up above.
// Otherwise it will fail when created on a Playback thread by legacy
// handling below. Ditto with Mmap, the associated Mmap track must be created
// before creating the AudioEffect or the io handle must be specified.
//
// Detect if the effect is created after an AudioRecord is destroyed.
if (getOrphanEffectChain_l(sessionId).get() != nullptr) {
ALOGE("%s: effect %s with no specified io handle is denied because the AudioRecord"
" for session %d no longer exists",
__func__, descOut.name, sessionId);
lStatus = PERMISSION_DENIED;
goto Exit;
}
// Legacy handling of creating an effect on an expired or made-up
// session id. We think that it is a Playback effect.
//
// If no output thread contains the requested session ID, default to
// first output. The effect chain will be moved to the correct output
// thread when a track with the same session ID is created
if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
io = mPlaybackThreads.keyAt(0);
}
ALOGV("createEffect() got io %d for effect %s", io, descOut.name);
} else if (checkPlaybackThread_l(io) != nullptr) {
// allow only one effect chain per sessionId on mPlaybackThreads.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
const audio_io_handle_t checkIo = mPlaybackThreads.keyAt(i);
if (io == checkIo) {
if (hapticPlaybackRequired
&& mPlaybackThreads.valueAt(i)
->hapticChannelMask() == AUDIO_CHANNEL_NONE) {
ALOGE("%s: haptic playback thread is required while the required playback "
"thread(io=%d) doesn't support", __func__, (int)io);
lStatus = BAD_VALUE;
goto Exit;
}
continue;
}
const uint32_t sessionType =
mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId);
if ((sessionType & ThreadBase::EFFECT_SESSION) != 0) {
ALOGE("%s: effect %s io %d denied because session %d effect exists on io %d",
__func__, descOut.name, (int) io, (int) sessionId, (int) checkIo);
android_errorWriteLog(0x534e4554, "123237974");
lStatus = BAD_VALUE;
goto Exit;
}
}
}
ThreadBase *thread = checkRecordThread_l(io);
if (thread == NULL) {
thread = checkPlaybackThread_l(io);
if (thread == NULL) {
thread = checkMmapThread_l(io);
if (thread == NULL) {
ALOGE("createEffect() unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
}
} else {
// Check if one effect chain was awaiting for an effect to be created on this
// session and used it instead of creating a new one.
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
Mutex::Autolock _l(thread->mLock);
thread->addEffectChain_l(chain);
}
}
sp<Client> client = registerPid(currentPid);
// create effect on selected output thread
bool pinned = !audio_is_global_session(sessionId) && isSessionAcquired_l(sessionId);
ThreadBase *oriThread = nullptr;
if (hapticPlaybackRequired && thread->hapticChannelMask() == AUDIO_CHANNEL_NONE) {
ThreadBase *hapticThread = hapticPlaybackThread_l();
if (hapticThread == nullptr) {
ALOGE("%s haptic thread not found while it is required", __func__);
lStatus = INVALID_OPERATION;
goto Exit;
}
if (hapticThread != thread) {
// Force to use haptic thread for haptic-generating effect.
oriThread = thread;
thread = hapticThread;
}
}
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
&descOut, &enabledOut, &lStatus, pinned, probe);
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
// remove local strong reference to Client with mClientLock held
Mutex::Autolock _cl(mClientLock);
client.clear();
} else {
// handle must be valid here, but check again to be safe.
if (handle.get() != nullptr) idOut = handle->id();
// Invalidate audio session when haptic playback is created.
if (hapticPlaybackRequired && oriThread != nullptr) {
// invalidateTracksForAudioSession will trigger locking the thread.
oriThread->invalidateTracksForAudioSession(sessionId);
}
}
}
Register:
if (!probe && (lStatus == NO_ERROR || lStatus == ALREADY_EXISTS)) {
// Check CPU and memory usage
sp<EffectBase> effect = handle->effect().promote();
if (effect != nullptr) {
status_t rStatus = effect->updatePolicyState();
if (rStatus != NO_ERROR) {
lStatus = rStatus;
}
}
} else {
handle.clear();
}
response->id = idOut;
response->enabled = enabledOut != 0;
response->effect = handle;
response->desc = VALUE_OR_RETURN_STATUS(
legacy2aidl_effect_descriptor_t_EffectDescriptor(descOut));
Exit:
return lStatus;
}
status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput)
{
ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
sessionId, srcOutput, dstOutput);
Mutex::Autolock _l(mLock);
if (srcOutput == dstOutput) {
ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
return NO_ERROR;
}
PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
if (srcThread == NULL) {
ALOGW("moveEffects() bad srcOutput %d", srcOutput);
return BAD_VALUE;
}
PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
if (dstThread == NULL) {
ALOGW("moveEffects() bad dstOutput %d", dstOutput);
return BAD_VALUE;
}
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(srcThread->mLock);
return moveEffectChain_l(sessionId, srcThread, dstThread);
}
void AudioFlinger::setEffectSuspended(int effectId,
audio_session_t sessionId,
bool suspended)
{
Mutex::Autolock _l(mLock);
sp<ThreadBase> thread = getEffectThread_l(sessionId, effectId);
if (thread == nullptr) {
return;
}
Mutex::Autolock _sl(thread->mLock);
sp<EffectModule> effect = thread->getEffect_l(sessionId, effectId);
thread->setEffectSuspended_l(&effect->desc().type, suspended, sessionId);
}
// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
AudioFlinger::PlaybackThread *srcThread,
AudioFlinger::PlaybackThread *dstThread)
{
ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
sessionId, srcThread, dstThread);
sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
if (chain == 0) {
ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
sessionId, srcThread);
return INVALID_OPERATION;
}
// Check whether the destination thread and all effects in the chain are compatible
if (!chain->isCompatibleWithThread_l(dstThread)) {
ALOGW("moveEffectChain_l() effect chain failed because"
" destination thread %p is not compatible with effects in the chain",
dstThread);
return INVALID_OPERATION;
}
// remove chain first. This is useful only if reconfiguring effect chain on same output thread,
// so that a new chain is created with correct parameters when first effect is added. This is
// otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
// removed.
srcThread->removeEffectChain_l(chain);
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
sp<EffectChain> dstChain;
uint32_t strategy = 0; // prevent compiler warning
sp<EffectModule> effect = chain->getEffectFromId_l(0);
Vector< sp<EffectModule> > removed;
status_t status = NO_ERROR;
while (effect != 0) {
srcThread->removeEffect_l(effect);
removed.add(effect);
status = dstThread->addEffect_l(effect);
if (status != NO_ERROR) {
break;
}
// removeEffect_l() has stopped the effect if it was active so it must be restarted
if (effect->state() == EffectModule::ACTIVE ||
effect->state() == EffectModule::STOPPING) {
effect->start();
}
// if the move request is not received from audio policy manager, the effect must be
// re-registered with the new strategy and output
if (dstChain == 0) {
dstChain = effect->getCallback()->chain().promote();
if (dstChain == 0) {
ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
status = NO_INIT;
break;
}
strategy = dstChain->strategy();
}
effect = chain->getEffectFromId_l(0);
}
if (status != NO_ERROR) {
for (size_t i = 0; i < removed.size(); i++) {
srcThread->addEffect_l(removed[i]);
}
}
return status;
}
status_t AudioFlinger::moveAuxEffectToIo(int EffectId,
const sp<PlaybackThread>& dstThread,
sp<PlaybackThread> *srcThread)
{
status_t status = NO_ERROR;
Mutex::Autolock _l(mLock);
sp<PlaybackThread> thread =
static_cast<PlaybackThread *>(getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId).get());
if (EffectId != 0 && thread != 0 && dstThread != thread.get()) {
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(thread->mLock);
sp<EffectChain> srcChain = thread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
sp<EffectChain> dstChain;
if (srcChain == 0) {
return INVALID_OPERATION;
}
sp<EffectModule> effect = srcChain->getEffectFromId_l(EffectId);
if (effect == 0) {
return INVALID_OPERATION;
}
thread->removeEffect_l(effect);
status = dstThread->addEffect_l(effect);
if (status != NO_ERROR) {
thread->addEffect_l(effect);
status = INVALID_OPERATION;
goto Exit;
}
dstChain = effect->getCallback()->chain().promote();
if (dstChain == 0) {
thread->addEffect_l(effect);
status = INVALID_OPERATION;
}
Exit:
// removeEffect_l() has stopped the effect if it was active so it must be restarted
if (effect->state() == EffectModule::ACTIVE ||
effect->state() == EffectModule::STOPPING) {
effect->start();
}
}
if (status == NO_ERROR && srcThread != nullptr) {
*srcThread = thread;
}
return status;
}
bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
{
if (mGlobalEffectEnableTime != 0 &&
((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
return true;
}
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<EffectChain> ec =
mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (ec != 0 && ec->isNonOffloadableEnabled()) {
return true;
}
}
return false;
}
void AudioFlinger::onNonOffloadableGlobalEffectEnable()
{
Mutex::Autolock _l(mLock);
mGlobalEffectEnableTime = systemTime();
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (t->mType == ThreadBase::OFFLOAD) {
t->invalidateTracks(AUDIO_STREAM_MUSIC);
}
}
}
status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
{
// clear possible suspended state before parking the chain so that it starts in default state
// when attached to a new record thread
chain->setEffectSuspended_l(FX_IID_AEC, false);
chain->setEffectSuspended_l(FX_IID_NS, false);
audio_session_t session = chain->sessionId();
ssize_t index = mOrphanEffectChains.indexOfKey(session);
ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
if (index >= 0) {
ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
return ALREADY_EXISTS;
}
mOrphanEffectChains.add(session, chain);
return NO_ERROR;
}
sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session)
{
sp<EffectChain> chain;
ssize_t index = mOrphanEffectChains.indexOfKey(session);
ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
if (index >= 0) {
chain = mOrphanEffectChains.valueAt(index);
mOrphanEffectChains.removeItemsAt(index);
}
return chain;
}
bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
{
Mutex::Autolock _l(mLock);
audio_session_t session = effect->sessionId();
ssize_t index = mOrphanEffectChains.indexOfKey(session);
ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
if (index >= 0) {
sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
if (chain->removeEffect_l(effect, true) == 0) {
ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
mOrphanEffectChains.removeItemsAt(index);
}
return true;
}
return false;
}
// ----------------------------------------------------------------------------
status_t AudioFlinger::onTransactWrapper(TransactionCode code,
const Parcel& data,
uint32_t flags,
const std::function<status_t()>& delegate) {
(void) data;
(void) flags;
// make sure transactions reserved to AudioPolicyManager do not come from other processes
switch (code) {
case TransactionCode::SET_STREAM_VOLUME:
case TransactionCode::SET_STREAM_MUTE:
case TransactionCode::OPEN_OUTPUT:
case TransactionCode::OPEN_DUPLICATE_OUTPUT:
case TransactionCode::CLOSE_OUTPUT:
case TransactionCode::SUSPEND_OUTPUT:
case TransactionCode::RESTORE_OUTPUT:
case TransactionCode::OPEN_INPUT:
case TransactionCode::CLOSE_INPUT:
case TransactionCode::INVALIDATE_STREAM:
case TransactionCode::SET_VOICE_VOLUME:
case TransactionCode::MOVE_EFFECTS:
case TransactionCode::SET_EFFECT_SUSPENDED:
case TransactionCode::LOAD_HW_MODULE:
case TransactionCode::GET_AUDIO_PORT:
case TransactionCode::CREATE_AUDIO_PATCH:
case TransactionCode::RELEASE_AUDIO_PATCH:
case TransactionCode::LIST_AUDIO_PATCHES:
case TransactionCode::SET_AUDIO_PORT_CONFIG:
case TransactionCode::SET_RECORD_SILENCED:
ALOGW("%s: transaction %d received from PID %d",
__func__, code, IPCThreadState::self()->getCallingPid());
// return status only for non void methods
switch (code) {
case TransactionCode::SET_RECORD_SILENCED:
case TransactionCode::SET_EFFECT_SUSPENDED:
break;
default:
return INVALID_OPERATION;
}
// Fail silently in these cases.
return OK;
default:
break;
}
// make sure the following transactions come from system components
switch (code) {
case TransactionCode::SET_MASTER_VOLUME:
case TransactionCode::SET_MASTER_MUTE:
case TransactionCode::MASTER_MUTE:
case TransactionCode::SET_MODE:
case TransactionCode::SET_MIC_MUTE:
case TransactionCode::SET_LOW_RAM_DEVICE:
case TransactionCode::SYSTEM_READY:
case TransactionCode::SET_AUDIO_HAL_PIDS:
case TransactionCode::SET_VIBRATOR_INFOS:
case TransactionCode::UPDATE_SECONDARY_OUTPUTS: {
if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
__func__, code, IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
// return status only for non void methods
switch (code) {
case TransactionCode::SYSTEM_READY:
break;
default:
return INVALID_OPERATION;
}
// Fail silently in these cases.
return OK;
}
} break;
default:
break;
}
// List of relevant events that trigger log merging.
// Log merging should activate during audio activity of any kind. This are considered the
// most relevant events.
// TODO should select more wisely the items from the list
switch (code) {
case TransactionCode::CREATE_TRACK:
case TransactionCode::CREATE_RECORD:
case TransactionCode::SET_MASTER_VOLUME:
case TransactionCode::SET_MASTER_MUTE:
case TransactionCode::SET_MIC_MUTE:
case TransactionCode::SET_PARAMETERS:
case TransactionCode::CREATE_EFFECT:
case TransactionCode::SYSTEM_READY: {
requestLogMerge();
break;
}
default:
break;
}
std::string tag("IAudioFlinger command " +
std::to_string(static_cast<std::underlying_type_t<TransactionCode>>(code)));
TimeCheck check(tag.c_str());
// Make sure we connect to Audio Policy Service before calling into AudioFlinger:
// - AudioFlinger can call into Audio Policy Service with its global mutex held
// - If this is the first time Audio Policy Service is queried from inside audioserver process
// this will trigger Audio Policy Manager initialization.
// - Audio Policy Manager initialization calls into AudioFlinger which will try to lock
// its global mutex and a deadlock will occur.
if (IPCThreadState::self()->getCallingPid() != getpid()) {
AudioSystem::get_audio_policy_service();
}
return delegate();
}
} // namespace android
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