blob: bc1654e6746c675ca5f1e9cdeeaafee09ef46098 [file] [log] [blame]
/*
**
** 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
#include "Configuration.h"
#include <dirent.h>
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <cutils/multiuser.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/TypeConverter.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 "AudioFlinger.h"
#include "ServiceUtilities.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 <audio_utils/primitives.h>
#include <audio_utils/string.h>
#include <powermanager/PowerManager.h>
#include <media/IMediaLogService.h>
#include <media/MemoryLeakTrackUtil.h>
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
#include <media/AudioParameter.h>
#include <mediautils/BatteryNotifier.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 {
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;
#ifdef TEE_SINK
bool AudioFlinger::mTeeSinkInputEnabled = false;
bool AudioFlinger::mTeeSinkOutputEnabled = false;
bool AudioFlinger::mTeeSinkTrackEnabled = false;
size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
#endif
// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
// we define a minimum time during which a global effect is considered enabled.
static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
Mutex gLock;
wp<AudioFlinger> gAudioFlinger;
// 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);
}
}
// ----------------------------------------------------------------------------
std::string formatToString(audio_format_t format) {
std::string result;
FormatConverter::toString(format, result);
return result;
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
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),
mSystemReady(false)
{
// 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++) {
// zero ID has a special meaning, so unavailable
mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX;
}
getpid_cached = getpid();
const bool doLog = property_get_bool("ro.test_harness", false);
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");
#ifdef TEE_SINK
char value[PROPERTY_VALUE_MAX];
(void) property_get("ro.debuggable", value, "0");
int debuggable = atoi(value);
int teeEnabled = 0;
if (debuggable) {
(void) property_get("af.tee", value, "0");
teeEnabled = atoi(value);
}
// FIXME symbolic constants here
if (teeEnabled & 1) {
mTeeSinkInputEnabled = true;
}
if (teeEnabled & 2) {
mTeeSinkOutputEnabled = true;
}
if (teeEnabled & 4) {
mTeeSinkTrackEnabled = true;
}
#endif
}
void AudioFlinger::onFirstRef()
{
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
char val_str[PROPERTY_VALUE_MAX] = { 0 };
if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
uint32_t int_val;
if (1 == sscanf(val_str, "%u", &int_val)) {
mStandbyTimeInNsecs = milliseconds(int_val);
ALOGI("Using %u mSec as standby time.", int_val);
} else {
mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
ALOGI("Using default %u mSec as standby time.",
(uint32_t)(mStandbyTimeInNsecs / 1000000));
}
}
mPatchPanel = new PatchPanel(this);
mMode = AUDIO_MODE_NORMAL;
gAudioFlinger = this;
}
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
// closeInput_nonvirtual() will remove specified entry from mRecordThreads
closeInput_nonvirtual(mRecordThreads.keyAt(0));
}
while (!mPlaybackThreads.isEmpty()) {
// closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
}
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
// no mHardwareLock needed, as there are no other references to this
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)
{
sp<AudioFlinger> af;
{
Mutex::Autolock _l(gLock);
af = gAudioFlinger.promote();
}
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;
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;
ret = AudioSystem::getOutputForAttr(attr, &io,
actualSessionId,
&streamType, client.clientPid, client.clientUid,
&fullConfig,
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
AUDIO_OUTPUT_FLAG_DIRECT),
deviceId, &portId);
} else {
ret = AudioSystem::getInputForAttr(attr, &io,
actualSessionId,
client.clientPid,
client.clientUid,
client.packageName,
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(attr, streamType, actualSessionId, callback, *deviceId, portId);
*handle = portId;
*sessionId = actualSessionId;
} else {
if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
AudioSystem::releaseOutput(io, streamType, actualSessionId);
} else {
AudioSystem::releaseInput(portId);
}
ret = NO_INIT;
}
ALOGV("%s done status %d portId %d", __FUNCTION__, ret, portId);
return ret;
}
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 devices)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
if (module == 0) {
ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
for (size_t i = 0; i < 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 & devices) == devices) {
return audioHwDevice;
}
}
} else {
// check a match for the requested module handle
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
if (audioHwDevice != NULL) {
return audioHwDevice;
}
}
return NULL;
}
void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
sp<Client> client = mClients.valueAt(i).promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
result.append("Notification Clients:\n");
for (size_t i = 0; i < mNotificationClients.size(); ++i) {
snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i));
result.append(buffer);
}
result.append("Global session refs:\n");
result.append(" session pid count\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
result.append(buffer);
}
write(fd, result.string(), result.size());
}
void AudioFlinger::dumpInternals(int fd, const Vector<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)
{
bool locked = false;
for (int i = 0; i < kDumpLockRetries; ++i) {
if (mutex.tryLock() == NO_ERROR) {
locked = true;
break;
}
usleep(kDumpLockSleepUs);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<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();
}
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);
}
#ifdef TEE_SINK
// dump the serially shared record tee sink
if (mRecordTeeSource != 0) {
dumpTee(fd, mRecordTeeSource, AUDIO_IO_HANDLE_NONE, 'C');
}
#endif
BUFLOG_RESET;
if (locked) {
mLock.unlock();
}
// append a copy of media.log here by forwarding fd to it, but don't attempt
// to lookup the service if it's not running, as it will block for a second
if (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 dummy writer that does nothing.
// Similarly if we can't contact the media.log service, also return a dummy 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 dummy writer
return new NBLog::Writer();
}
success:
NBLog::Shared *sharedRawPtr = (NBLog::Shared *) shared->pointer();
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
sp<IAudioTrack> AudioFlinger::createTrack(const CreateTrackInput& input,
CreateTrackOutput& output,
status_t *status)
{
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;
bool updatePid = (input.clientInfo.clientPid == -1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
uid_t clientUid = input.clientInfo.clientUid;
if (!isTrustedCallingUid(callingUid)) {
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
clientUid = callingUid;
updatePid = true;
}
pid_t clientPid = input.clientInfo.clientPid;
if (updatePid) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(clientPid != -1 && clientPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, clientPid);
clientPid = callingPid;
}
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(&input.attr, &output.outputId, sessionId, &streamType,
clientPid, clientUid, &input.config, input.flags,
&output.selectedDeviceId, &portId);
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;
track = thread->createTrack_l(client, streamType, input.attr, &output.sampleRate,
input.config.format, input.config.channel_mask,
&output.frameCount, &output.notificationFrameCount,
input.notificationsPerBuffer, input.speed,
input.sharedBuffer, sessionId, &output.flags,
input.clientInfo.clientTid, clientUid, &lStatus, portId);
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();
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
// no risk of deadlock because AudioFlinger::mLock is held
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
moveEffectChain_l(sessionId, effectThread, thread, true);
}
// Look for sync events awaiting for a session to be used.
for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
if (mPendingSyncEvents[i]->triggerSession() == 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;
}
// return handle to client
trackHandle = new TrackHandle(track);
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(output.outputId, streamType, sessionId);
}
*status = lStatus;
return trackHandle;
}
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.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (dev->canSetMasterVolume()) {
dev->hwDevice()->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::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);
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);
}
return ret;
}
status_t AudioFlinger::setMicMute(bool state)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->setMicMute(state);
if (result != NO_ERROR) {
ret = result;
}
}
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return false;
}
bool mute = true;
bool state = AUDIO_MODE_INVALID;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->getMicMute(&state);
if (result == NO_ERROR) {
mute = mute && state;
}
}
mHardwareStatus = AUDIO_HW_IDLE;
return mute;
}
void AudioFlinger::setRecordSilenced(uid_t uid, bool silenced)
{
ALOGV("AudioFlinger::setRecordSilenced(uid:%d, silenced:%d)", uid, silenced);
AutoMutex lock(mLock);
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads[i]->setRecordSilenced(uid, silenced);
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
mMmapThreads[i]->setRecordSilenced(uid, 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.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if (dev->canSetMasterMute()) {
dev->hwDevice()->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();
}
bool AudioFlinger::masterMute() const
{
Mutex::Autolock _l(mLock);
return masterMute_l();
}
float AudioFlinger::masterVolume_l() const
{
return mMasterVolume;
}
bool AudioFlinger::masterMute_l() const
{
return mMasterMute;
}
status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
ALOGW("checkStreamType() invalid stream %d", stream);
return BAD_VALUE;
}
pid_t caller = IPCThreadState::self()->getCallingPid();
if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
ALOGW("checkStreamType() pid %d cannot use internal stream type %d", caller, stream);
return PERMISSION_DENIED;
}
return NO_ERROR;
}
status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return status;
}
if (output == AUDIO_IO_HANDLE_NONE) {
return BAD_VALUE;
}
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH 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::broacastParametersToRecordThreads_l(const String8& keyValuePairs)
{
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
}
}
// 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::keyStreamConnect),
String8(AudioParameter::keyStreamDisconnect),
String8(AudioParameter::keyStreamSupportedFormats),
String8(AudioParameter::keyStreamSupportedChannels),
String8(AudioParameter::keyStreamSupportedSamplingRates),
};
// multiuser friendly app ID check for requests coming from audioserver
if (multiuser_get_app_id(callingUid) == AID_AUDIOSERVER) {
return;
}
AudioParameter param = AudioParameter(keyValuePairs);
String8 value;
for (auto& key : kReservedParameters) {
if (param.get(key, value) == NO_ERROR) {
ALOGW("%s: filtering key %s value %s from uid %d",
__func__, key.string(), value.string(), callingUid);
param.remove(key);
}
}
keyValuePairs = param.toString();
}
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)) {
broacastParametersToRecordThreads_l(filteredKeyValuePairs);
}
}
}
if (thread != 0) {
return thread->setParameters(filteredKeyValuePairs);
}
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;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
String8 s;
status_t result;
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_PARAMETER;
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
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);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
audio_config_t config, proposed;
memset(&proposed, 0, sizeof(proposed));
proposed.sample_rate = sampleRate;
proposed.channel_mask = channelMask;
proposed.format = format;
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
size_t frames;
for (;;) {
// Note: config is currently a const parameter for get_input_buffer_size()
// but we use a copy from proposed in case config changes from the call.
config = proposed;
status_t result = dev->getInputBufferSize(&config, &frames);
if (result == OK && frames != 0) {
break; // hal success, config is the result
}
// change one parameter of the configuration each iteration to a more "common" value
// to see if the device will support it.
if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
proposed.format = AUDIO_FORMAT_PCM_16_BIT;
} else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw?
} else {
ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
"format %#x, channelMask 0x%X",
sampleRate, format, channelMask);
break; // retries failed, break out of loop with frames == 0.
}
}
mHardwareStatus = AUDIO_HW_IDLE;
if (frames > 0 && config.sample_rate != sampleRate) {
frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
}
return frames; // may be converted to bytes at the Java level.
}
uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getInputFramesLost();
}
return 0;
}
status_t AudioFlinger::setVoiceVolume(float value)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
ret = dev->setVoiceVolume(value);
mHardwareStatus = AUDIO_HW_IDLE;
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<IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
if (client == 0) {
return;
}
pid_t pid = IPCThreadState::self()->getCallingPid();
{
Mutex::Autolock _cl(mClientLock);
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
pid);
ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
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)
{
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) {
purgeStaleEffects_l();
}
}
void AudioFlinger::ioConfigChanged(audio_io_config_event event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid)
{
Mutex::Autolock _l(mClientLock);
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc);
}
}
}
// removeClient_l() must be called with AudioFlinger::mClientLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
ALOGV("removeClient_l() pid %d, calling pid %d", pid,
IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// getEffectThread_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
int EffectId)
{
sp<PlaybackThread> thread;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
ALOG_ASSERT(thread == 0);
thread = mPlaybackThreads.valueAt(i);
}
}
return thread;
}
// ----------------------------------------------------------------------------
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<IAudioFlingerClient>& client,
pid_t pid)
: mAudioFlinger(audioFlinger), mPid(pid), 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();
}
// ----------------------------------------------------------------------------
sp<media::IAudioRecord> AudioFlinger::createRecord(const CreateRecordInput& input,
CreateRecordOutput& output,
status_t *status)
{
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;
bool updatePid = (input.clientInfo.clientPid == -1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
uid_t clientUid = input.clientInfo.clientUid;
if (!isTrustedCallingUid(callingUid)) {
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
clientUid = callingUid;
updatePid = true;
}
pid_t clientPid = input.clientInfo.clientPid;
if (updatePid) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(clientPid != -1 && clientPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, clientPid);
clientPid = callingPid;
}
// 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(clientPid);
// 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,
sessionId,
// FIXME compare to AudioTrack
clientPid,
clientUid,
input.opPackageName,
&input.config,
output.flags, &output.selectedDeviceId, &portId);
{
Mutex::Autolock _l(mLock);
RecordThread *thread = checkRecordThread_l(output.inputId);
if (thread == NULL) {
ALOGE("createRecord() checkRecordThread_l failed");
lStatus = BAD_VALUE;
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,
clientUid, &output.flags,
input.clientInfo.clientTid,
&lStatus, portId);
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();
// return handle to client
recordHandle = new RecordHandle(recordTrack);
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);
}
}
*status = lStatus;
return recordHandle;
}
// ----------------------------------------------------------------------------
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);
return loadHwModule_l(name);
}
// loadHwModule_l() must be called with AudioFlinger::mLock held
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
ALOGW("loadHwModule() module %s already loaded", name);
return mAudioHwDevs.keyAt(i);
}
}
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);
{ // scope for auto-lock pattern
AutoMutex lock(mHardwareLock);
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;
}
audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
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 (uid != AID_SYSTEM) {
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;
}
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 = mPrimaryHardwareDev->hwDevice();
if (dev == NULL) {
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);
thread->setParameters(param.toString());
break;
}
}
ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
return (audio_hw_sync_t)value;
}
status_t AudioFlinger::systemReady()
{
Mutex::Autolock _l(mLock);
ALOGI("%s", __FUNCTION__);
if (mSystemReady) {
ALOGW("%s called twice", __FUNCTION__);
return NO_ERROR;
}
mSystemReady = true;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
thread->systemReady();
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
thread->systemReady();
}
return NO_ERROR;
}
status_t AudioFlinger::getMicrophones(std::vector<media::MicrophoneInfo> *microphones)
{
AutoMutex lock(mHardwareLock);
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
status_t status = dev->getMicrophones(microphones);
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);
thread->setParameters(param.toString());
}
}
// ----------------------------------------------------------------------------
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_output_flags_t flags)
{
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
if (outHwDev == NULL) {
return 0;
}
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,
devices,
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,
devices, AUDIO_DEVICE_NONE, 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, devices, 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, devices, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
} else {
thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p",
*output, thread.get());
}
mPlaybackThreads.add(*output, thread);
return thread;
}
}
return 0;
}
status_t AudioFlinger::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
ALOGI("openOutput() this %p, module %d Device %#x, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
this, module,
(devices != NULL) ? *devices : 0,
config->sample_rate,
config->format,
config->channel_mask,
flags);
if (devices == NULL || *devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
sp<ThreadBase> thread = openOutput_l(module, output, config, *devices, 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 ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d as the primary audio interface", module);
mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->setMode(mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
MmapThread *mmapThread = (MmapThread *)thread.get();
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
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);
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, true);
}
}
}
} else {
mmapThread = (MmapPlaybackThread *)checkMmapThread_l(output);
if (mmapThread == 0) {
return BAD_VALUE;
}
mMmapThreads.removeItem(output);
ALOGD("closing mmapThread %p", mmapThread.get());
}
const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
ioDesc->mIoHandle = output;
ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
}
// 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::closeOutputInternal_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(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
audio_source_t source,
audio_input_flags_t flags)
{
Mutex::Autolock _l(mLock);
if (*devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
sp<ThreadBase> thread = openInput_l(module, input, config, *devices, address, source, flags);
if (thread != 0) {
// notify client processes of the new input creation
thread->ioConfigChanged(AUDIO_INPUT_OPENED);
return NO_ERROR;
}
return NO_INIT;
}
sp<AudioFlinger::ThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_source_t source,
audio_input_flags_t flags)
{
AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
if (inHwDev == NULL) {
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
// 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.string(), source, &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.string());
// If the input could not be opened with the requested parameters and we can handle the
// conversion internally, try to open again with the proposed parameters.
if (status == BAD_VALUE &&
audio_is_linear_pcm(config->format) &&
audio_is_linear_pcm(halconfig.format) &&
(halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) &&
(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) {
// 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.string(), source, &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,
primaryOutputDevice_l(), devices, mSystemReady);
mMmapThreads.add(*input, thread);
ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
thread.get());
return thread;
} else {
#ifdef TEE_SINK
// Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
// or (re-)create if current Pipe is idle and does not match the new format
sp<NBAIO_Sink> teeSink;
enum {
TEE_SINK_NO, // don't copy input
TEE_SINK_NEW, // copy input using a new pipe
TEE_SINK_OLD, // copy input using an existing pipe
} kind;
NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
} else if (!Format_isValid(format)) {
kind = TEE_SINK_NO;
} else if (mRecordTeeSink == 0) {
kind = TEE_SINK_NEW;
} else if (mRecordTeeSink->getStrongCount() != 1) {
kind = TEE_SINK_NO;
} else if (Format_isEqual(format, mRecordTeeSink->format())) {
kind = TEE_SINK_OLD;
} else {
kind = TEE_SINK_NEW;
}
switch (kind) {
case TEE_SINK_NEW: {
Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {format};
ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
PipeReader *pipeReader = new PipeReader(*pipe);
numCounterOffers = 0;
index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
mRecordTeeSink = pipe;
mRecordTeeSource = pipeReader;
teeSink = pipe;
}
break;
case TEE_SINK_OLD:
teeSink = mRecordTeeSink;
break;
case TEE_SINK_NO:
default:
break;
}
#endif
// 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,
primaryOutputDevice_l(),
devices,
mSystemReady
#ifdef TEE_SINK
, teeSink
#endif
);
mRecordThreads.add(*input, thread);
ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
return thread;
}
}
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
return closeInput_nonvirtual(input);
}
status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
sp<RecordThread> recordThread;
sp<MmapCaptureThread> mmapThread;
{
Mutex::Autolock _l(mLock);
recordThread = checkRecordThread_l(input);
if (recordThread != 0) {
ALOGV("closeInput() %d", input);
// If we still have effect chains, it means that a client still holds a handle
// on at least one effect. We must either move the chain to an existing thread with the
// same session ID or put it aside in case a new record thread is opened for a
// new capture on the same session
sp<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;
}
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::closeInputInternal_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)
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
if (pid != -1 && (caller == getpid_cached)) {
caller = pid;
}
{
Mutex::Autolock _cl(mClientLock);
// Ignore requests received from processes not known as notification client. The request
// is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
// called from a different pid leaving a stale session reference. Also we don't know how
// to clear this reference if the client process dies.
if (mNotificationClients.indexOfKey(caller) < 0) {
ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
return;
}
}